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Diffstat (limited to 'packages/linux/linux-rp-2.6.17/asoc-v0.12.4_2.6.17.patch')
| -rw-r--r-- | packages/linux/linux-rp-2.6.17/asoc-v0.12.4_2.6.17.patch | 31713 |
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diff --git a/packages/linux/linux-rp-2.6.17/asoc-v0.12.4_2.6.17.patch b/packages/linux/linux-rp-2.6.17/asoc-v0.12.4_2.6.17.patch deleted file mode 100644 index 4f9672299b..0000000000 --- a/packages/linux/linux-rp-2.6.17/asoc-v0.12.4_2.6.17.patch +++ /dev/null @@ -1,31713 +0,0 @@ -Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/DAI.txt -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/DAI.txt -@@ -0,0 +1,546 @@ -+ASoC currently supports the three main Digital Audio Interfaces (DAI) found on -+SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM. -+ -+ -+AC97 -+==== -+ -+ AC97 is a five wire interface commonly found on many PC sound cards. It is -+now also popular in many portable devices. This DAI has a reset line and time -+multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines. -+The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the -+frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 -+frame is 21uS long and is divided into 13 time slots. -+ -+The AC97 specification can be found at :- -+http://www.intel.com/design/chipsets/audio/ac97_r23.pdf -+ -+ -+I2S -+=== -+ -+ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and -+Rx lines are used for audio transmision, whilst the bit clock (BCLK) and -+left/right clock (LRC) synchronise the link. I2S is flexible in that either the -+controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock -+usually varies depending on the sample rate and the master system clock -+(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate -+ADC and DAC LRCLK's, this allows for similtanious capture and playback at -+different sample rates. -+ -+I2S has several different operating modes:- -+ -+ o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC -+ transition. -+ -+ o Left Justified - MSB is transmitted on transition of LRC. -+ -+ o Right Justified - MSB is transmitted sample size BCLK's before LRC -+ transition. -+ -+PCM -+=== -+ -+PCM is another 4 wire interface, very similar to I2S, that can support a more -+flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used -+to synchronise the link whilst the Tx and Rx lines are used to transmit and -+receive the audio data. Bit clock usually varies depending on sample rate -+whilst sync runs at the sample rate. PCM also supports Time Division -+Multiplexing (TDM) in that several devices can use the bus similtaniuosly (This -+is sometimes referred to as network mode). -+ -+Common PCM operating modes:- -+ -+ o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. -+ -+ o Mode B - MSB is transmitted on rising edge of FRAME/SYNC. -+ -+ -+ASoC DAI Configuration -+====================== -+ -+Every CODEC DAI and SoC DAI must have their capabilities defined in order to -+be configured together at runtime when the audio and clocking parameters are -+known. This is achieved by creating an array of struct snd_soc_hw_mode in the -+the CODEC and SoC interface drivers. Each element in the array describes a DAI -+mode and each mode is usually based upon the DAI system clock to sample rate -+ratio (FS). -+ -+i.e. 48k sample rate @ 256 FS = sytem clock of 12.288 MHz -+ 48000 * 256 = 12288000 -+ -+The CPU and Codec DAI modes are then ANDed together at runtime to determine the -+rutime DAI configuration for both the Codec and CPU. -+ -+When creating a new codec or SoC DAI it's probably best to start of with a few -+sample rates first and then test your interface. -+ -+struct snd_soc_dai_mode is defined (in soc.h) as:- -+ -+/* SoC DAI mode */ -+struct snd_soc_dai_mode { -+ u16 fmt; /* SND_SOC_DAIFMT_* */ -+ u16 tdm; /* SND_SOC_HWTDM_* */ -+ u64 pcmfmt; /* SNDRV_PCM_FMTBIT_* */ -+ u16 pcmrate; /* SND_SOC_HWRATE_* */ -+ u16 pcmdir:2; /* SND_SOC_HWDIR_* */ -+ u16 flags:8; /* hw flags */ -+ u16 fs; /* mclk to rate divider */ -+ u64 bfs; /* mclk to bclk dividers */ -+ unsigned long priv; /* private mode data */ -+}; -+ -+fmt: -+---- -+This field defines the DAI mode hardware format (e.g. I2S settings) and -+supports the following settings:- -+ -+ 1) hardware DAI formats -+ -+#define SND_SOC_DAIFMT_I2S (1 << 0) /* I2S mode */ -+#define SND_SOC_DAIFMT_RIGHT_J (1 << 1) /* Right justified mode */ -+#define SND_SOC_DAIFMT_LEFT_J (1 << 2) /* Left Justified mode */ -+#define SND_SOC_DAIFMT_DSP_A (1 << 3) /* L data msb after FRM */ -+#define SND_SOC_DAIFMT_DSP_B (1 << 4) /* L data msb during FRM */ -+#define SND_SOC_DAIFMT_AC97 (1 << 5) /* AC97 */ -+ -+ 2) hw DAI signal inversions -+ -+#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */ -+#define SND_SOC_DAIFMT_NB_IF (1 << 9) /* normal bclk + inv frm */ -+#define SND_SOC_DAIFMT_IB_NF (1 << 10) /* invert bclk + nor frm */ -+#define SND_SOC_DAIFMT_IB_IF (1 << 11) /* invert bclk + frm */ -+ -+ 3) hw clock masters -+ This is wrt the codec, the inverse is true for the interface -+ i.e. if the codec is clk and frm master then the interface is -+ clk and frame slave. -+ -+#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & frm master */ -+#define SND_SOC_DAIFMT_CBS_CFM (1 << 13) /* codec clk slave & frm master */ -+#define SND_SOC_DAIFMT_CBM_CFS (1 << 14) /* codec clk master & frame slave */ -+#define SND_SOC_DAIFMT_CBS_CFS (1 << 15) /* codec clk & frm slave */ -+ -+At least one option from each section must be selected. Multiple selections are -+also supported e.g. -+ -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ -+ SND_SOC_DAIFMT_IB_IF -+ -+ -+tdm: -+------ -+This field defines the Time Division Multiplexing left and right word -+positions for the DAI mode if applicable. Set to SND_SOC_DAITDM_LRDW(0,0) for -+no TDM. -+ -+ -+pcmfmt: -+--------- -+The hardware PCM format. This describes the PCM formats supported by the DAI -+mode e.g. -+ -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ -+ SNDRV_PCM_FORMAT_S24_3LE -+ -+pcmrate: -+---------- -+The PCM sample rates supported by the DAI mode. e.g. -+ -+ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 -+ -+ -+pcmdir: -+--------- -+The stream directions supported by this mode. e.g. playback and capture -+ -+ -+flags: -+-------- -+The DAI hardware flags supported by the mode. -+ -+/* use bfs mclk divider mode (BCLK = MCLK / x) */ -+#define SND_SOC_DAI_BFS_DIV 0x1 -+/* use bfs rate mulitplier (BCLK = RATE * x)*/ -+#define SND_SOC_DAI_BFS_RATE 0x2 -+/* use bfs rcw multiplier (BCLK = RATE * CHN * WORD SIZE) */ -+#define SND_SOC_DAI_BFS_RCW 0x4 -+/* capture and playback can use different clocks */ -+#define SND_SOC_DAI_ASYNC 0x8 -+ -+NOTE: Bitclock division and mulitiplication modes can be safely matched by the -+core logic. -+ -+ -+fs: -+----- -+The FS supported by this DAI mode FS is the ratio between the system clock and -+the sample rate. See above -+ -+bfs: -+------ -+BFS is the ratio of BCLK to MCLK or the ratio of BCLK to sample rate (this -+depends on the codec or CPU DAI). -+ -+The BFS supported by the DAI mode. This can either be the ratio between the -+bitclock (BCLK) and the sample rate OR the ratio between the system clock and -+the sample rate. Depends on the flags above. -+ -+priv: -+----- -+private codec mode data. -+ -+ -+ -+Examples -+======== -+ -+Note that Codec DAI and CPU DAI examples are interchangeable in these examples -+as long as the bus master is reversed. i.e. -+ -+ SND_SOC_DAIFMT_CBM_CFM would become SND_SOC_DAIFMT_CBS_CFS -+ and vice versa. -+ -+This applies to all SND_SOC_DAIFMT_CB*_CF*. -+ -+Example 1 -+--------- -+ -+Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a -+BCLK of either MCLK/2 or MCLK/4. -+ -+ /* codec master */ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(2) | SND_SOC_FSBD(4), -+ } -+ -+ -+Example 2 -+--------- -+Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a -+BCLK of either Rate * 32 or Rate * 64. -+ -+ /* codec master */ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 32, -+ }, -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ -+ -+Example 3 -+--------- -+Codec that runs at 8k & 48k @ 256FS in master mode, can generate a BCLK that -+is a multiple of Rate * channels * word size. (RCW) i.e. -+ -+ BCLK = 8000 * 2 * 16 (8k, stereo, 16bit) -+ = 256kHz -+ -+This codecs supports a RCW multiple of 1,2 -+ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 256, -+ .bfs = SND_SOC_FSBW(1) | SND_SOC_FSBW(2), -+ } -+ -+ -+Example 4 -+--------- -+Codec that only runs at 8k & 48k @ 256FS in master mode, can generate a -+BCLK of either Rate * 32 or Rate * 64. Codec can also run in slave mode as long -+as BCLK is rate * 32 or rate * 64. -+ -+ /* codec master */ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 32, -+ }, -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ -+ /* codec slave */ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmdir = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = 32, -+ }, -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmdir = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = 64, -+ }, -+ -+ -+Example 5 -+--------- -+Codec that only runs at 8k, 16k, 32k, 48k, 96k @ 128FS, 192FS & 256FS in master -+mode and can generate a BCLK of MCLK / (1,2,4,8,16). Codec can also run in slave -+mode as and does not care about FS or BCLK (as long as there is enough bandwidth). -+ -+ #define CODEC_FSB \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ -+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) -+ -+ #define CODEC_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) -+ -+ /* codec master @ 128, 192 & 256 FS */ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = CODEC_RATES, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 128, -+ .bfs = CODEC_FSB, -+ }, -+ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = CODEC_RATES, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 192, -+ .bfs = CODEC_FSB -+ }, -+ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = CODEC_RATES, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = CODEC_FSB, -+ }, -+ -+ /* codec slave */ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = CODEC_RATES, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+ -+ -+Example 6 -+--------- -+Codec that only runs at 8k, 44.1k, 48k @ different FS in master mode (for use -+with a fixed MCLK) and can generate a BCLK of MCLK / (1,2,4,8,16). -+Codec can also run in slave mode as and does not care about FS or BCLK (as long -+as there is enough bandwidth). Codec can support 16, 24 and 32 bit PCM sample -+sizes. -+ -+ #define CODEC_FSB \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ -+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) -+ -+ #define CODEC_PCM_FORMATS \ -+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ -+ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | SNDRV_PCM_FORMAT_S32_LE) -+ -+ /* codec master */ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1536, -+ .bfs = CODEC_FSB, -+ }, -+ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 272, -+ .bfs = CODEC_FSB, -+ }, -+ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = CODEC_FSB, -+ }, -+ -+ /* codec slave */ -+ { -+ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, -+ .pcmrate = CODEC_RATES, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+ -+ -+Example 7 -+--------- -+AC97 Codec that does not support VRA (i.e only runs at 48k). -+ -+ #define AC97_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+ #define AC97_PCM_FORMATS \ -+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S18_3LE | \ -+ SNDRV_PCM_FORMAT_S20_3LE) -+ -+ /* AC97 with no VRA */ -+ { -+ .pcmfmt = AC97_PCM_FORMATS, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ } -+ -+ -+Example 8 -+--------- -+ -+CPU DAI that supports 8k - 48k @ 256FS and BCLK = MCLK / 4 in master mode. -+Slave mode (CPU DAI is FRAME master) supports 8k - 96k at any FS as long as -+BCLK = 64 * rate. (Intel XScale I2S controller). -+ -+ #define PXA_I2S_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF) -+ -+ #define PXA_I2S_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+ #define PXA_I2S_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -+ -+ /* priv is divider */ -+ static struct snd_soc_dai_mode pxa2xx_i2s_modes[] = { -+ /* pxa2xx I2S frame and clock master modes */ -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0x48, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0x34, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0x24, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0x1a, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0xd, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0xc, -+ }, -+ -+ /* pxa2xx I2S frame master and clock slave mode */ -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = PXA_I2S_RATES, -+ .pcmdir = PXA_I2S_DIR, -+ .fs = SND_SOC_FS_ALL, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .bfs = 64, -+ .priv = 0x48, -+ }, -+}; -Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/clocking.txt -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/clocking.txt -@@ -0,0 +1,314 @@ -+Audio Clocking -+============== -+ -+This text describes the audio clocking terms in ASoC and digital audio in -+general. Note: Audio clocking can be complex ! -+ -+ -+Master Clock -+------------ -+ -+Every audio subsystem is driven by a master clock (sometimes refered to as MCLK -+or SYSCLK). This audio master clock can be derived from a number of sources -+(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct -+audio playback and capture sample rates. -+ -+Some master clocks (e.g. PLL's and CPU based clocks) are configuarble in that -+their speed can be altered by software (depending on the system use and to save -+power). Other master clocks are fixed at at set frequency (i.e. crystals). -+ -+ -+DAI Clocks -+---------- -+The Digital Audio Interface is usually driven by a Bit Clock (often referred to -+as BCLK). This clock is used to drive the digital audio data across the link -+between the codec and CPU. -+ -+The DAI also has a frame clock to signal the start of each audio frame. This -+clock is sometimes referred to as LRC (left right clock) or FRAME. This clock -+runs at exactly the sample rate (LRC = Rate). -+ -+Bit Clock can be generated as follows:- -+ -+BCLK = MCLK / x -+ -+ or -+ -+BCLK = LRC * x -+ -+ or -+ -+BCLK = LRC * Channels * Word Size -+ -+This relationship depends on the codec or SoC CPU in particular. ASoC can quite -+easily match BCLK generated by division (SND_SOC_DAI_BFS_DIV) with BCLK by -+multiplication (SND_SOC_DAI_BFS_RATE) or BCLK generated by -+Rate * Channels * Word size (RCW or SND_SOC_DAI_BFS_RCW). -+ -+ -+ASoC Clocking -+------------- -+ -+The ASoC core determines the clocking for each particular configuration at -+runtime. This is to allow for dynamic audio clocking wereby the audio clock is -+variable and depends on the system state or device usage scenario. i.e. a voice -+call requires slower clocks (and hence less power) than MP3 playback. -+ -+ASoC will call the config_sysclock() function for the target machine during the -+audio parameters configuration. The function is responsible for then clocking -+the machine audio subsytem and returning the audio clock speed to the core. -+This function should also call the codec and cpu DAI clock_config() functions -+to configure their respective internal clocking if required. -+ -+ -+ASoC Clocking Control Flow -+-------------------------- -+ -+The ASoC core will call the machine drivers config_sysclock() when most of the -+DAI capabilities are known. The machine driver is then responsible for calling -+the codec and/or CPU DAI drivers with the selected capabilities and the current -+MCLK. Note that the machine driver is also resonsible for setting the MCLK (and -+enabling it). -+ -+ (1) Match Codec and CPU DAI capabilities. At this point we have -+ matched the majority of the DAI fields and now need to make sure this -+ mode is currently clockable. -+ -+ (2) machine->config_sysclk() is now called with the matched DAI FS, sample -+ rate and BCLK master. This function then gets/sets the current audio -+ clock (depening on usage) and calls the codec and CPUI DAI drivers with -+ the FS, rate, BCLK master and MCLK. -+ -+ (3) Codec/CPU DAI config_sysclock(). This function checks that the FS, rate, -+ BCLK master and MCLK are acceptable for the codec or CPU DAI. It also -+ sets the DAI internal state to work with said clocks. -+ -+The config_sysclk() functions for CPU, codec and machine should return the MCLK -+on success and 0 on failure. -+ -+ -+Examples (b = BCLK, l = LRC) -+============================ -+ -+Example 1 -+--------- -+ -+Simple codec that only runs at 48k @ 256FS in master mode. -+ -+CPU only runs as slave DAI, however it generates a variable MCLK. -+ -+ -------- --------- -+ | | <----mclk--- | | -+ | Codec |b -----------> | CPU | -+ | |l -----------> | | -+ | | | | -+ -------- --------- -+ -+The codec driver has the following config_sysclock() -+ -+ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+ { -+ /* make sure clock is 256 * rate */ -+ if(info->rate << 8 == clk) { -+ dai->mclk = clk; -+ return clk; -+ } -+ -+ return 0; -+ } -+ -+The CPU I2S DAI driver has the following config_sysclk() -+ -+ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+ { -+ /* can we support this clk */ -+ if(set_audio_clk(clk) < 0) -+ return -EINVAL; -+ -+ dai->mclk = clk; -+ return dai->clk; -+ } -+ -+The machine driver config_sysclk() in this example is as follows:- -+ -+ unsigned int machine_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+ { -+ int clk = info->rate * info->fs; -+ -+ /* check that CPU can deliver clock */ -+ if(rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk) < 0) -+ return -EINVAL; -+ -+ /* can codec work with this clock */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk); -+ } -+ -+ -+Example 2 -+--------- -+ -+Codec that can master at 8k and 48k at various FS (and hence supports a fixed -+set of input MCLK's) and can also be slave at various FS . -+ -+The CPU can master at 8k and 48k @256 FS and can be slave at any FS. -+ -+MCLK is a 12.288MHz crystal on this machine. -+ -+ -------- --------- -+ | | <---xtal---> | | -+ | Codec |b <----------> | CPU | -+ | |l <----------> | | -+ | | | | -+ -------- --------- -+ -+ -+The codec driver has the following config_sysclock() -+ -+ /* supported input clocks */ -+ const static int hifi_clks[] = {11289600, 12000000, 12288000, -+ 16934400, 18432000}; -+ -+ static unsigned int config_hsysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+ { -+ int i; -+ -+ /* is clk supported */ -+ for(i = 0; i < ARRAY_SIZE(hifi_clks); i++) { -+ if(clk == hifi_clks[i]) { -+ dai->mclk = clk; -+ return clk; -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+ } -+ -+The CPU I2S DAI driver has the following config_sysclk() -+ -+ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+ { -+ /* are we master or slave */ -+ if (info->bclk_master & -+ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) { -+ -+ /* we can only master @ 256FS */ -+ if(info->rate << 8 == clk) { -+ dai->mclk = clk; -+ return dai->mclk; -+ } -+ } else { -+ /* slave we can run at any FS */ -+ dai->mclk = clk; -+ return dai->mclk; -+ } -+ -+ /* not supported */ -+ return dai->clk; -+ } -+ -+The machine driver config_sysclk() in this example is as follows:- -+ -+ unsigned int machine_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+ { -+ int clk = 12288000; /* 12.288MHz */ -+ -+ /* who's driving the link */ -+ if (info->bclk_master & -+ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) { -+ /* codec master */ -+ -+ /* check that CPU can work with clock */ -+ if(rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk) < 0) -+ return -EINVAL; -+ -+ /* can codec work with this clock */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk); -+ } else { -+ /* cpu master */ -+ -+ /* check that codec can work with clock */ -+ if(rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk) < 0) -+ return -EINVAL; -+ -+ /* can CPU work with this clock */ -+ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk); -+ } -+ } -+ -+ -+ -+Example 3 -+--------- -+ -+Codec that masters at 8k ... 48k @256 FS. Codec can also be slave and -+doesn't care about FS. The codec has an internal PLL and dividers to generate -+the necessary internal clocks (for 256FS). -+ -+CPU can only be slave and doesn't care about FS. -+ -+MCLK is a non controllable 13MHz clock from the CPU. -+ -+ -+ -------- --------- -+ | | <----mclk--- | | -+ | Codec |b <----------> | CPU | -+ | |l <----------> | | -+ | | | | -+ -------- --------- -+ -+The codec driver has the following config_sysclock() -+ -+ /* valid PCM clock dividers * 2 */ -+ static int pcm_divs[] = {2, 6, 11, 4, 8, 12, 16}; -+ -+ static unsigned int config_vsysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+ { -+ int i, j, best_clk = info->fs * info->rate; -+ -+ /* can we run at this clk without the PLL ? */ -+ for (i = 0; i < ARRAY_SIZE(pcm_divs); i++) { -+ if ((best_clk >> 1) * pcm_divs[i] == clk) { -+ dai->pll_in = 0; -+ dai->clk_div = pcm_divs[i]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ -+ /* now check for PLL support */ -+ for (i = 0; i < ARRAY_SIZE(pll_div); i++) { -+ if (pll_div[i].pll_in == clk) { -+ for (j = 0; j < ARRAY_SIZE(pcm_divs); j++) { -+ if (pll_div[i].pll_out == pcm_divs[j] * (best_clk >> 1)) { -+ dai->pll_in = clk; -+ dai->pll_out = pll_div[i].pll_out; -+ dai->clk_div = pcm_divs[j]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+ } -+ -+ -+The CPU I2S DAI driver has the does not need a config_sysclk() as it can slave -+at any FS. -+ -+ unsigned int config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+ { -+ /* codec has pll that generates mclk from 13MHz xtal */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000); -+ } -Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/codec.txt -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/codec.txt -@@ -0,0 +1,232 @@ -+ASoC Codec Driver -+================= -+ -+The codec driver is generic and hardware independent code that configures the -+codec to provide audio capture and playback. It should contain no code that is -+specific to the target platform or machine. All platform and machine specific -+code should be added to the platform and machine drivers respectively. -+ -+Each codec driver must provide the following features:- -+ -+ 1) Digital audio interface (DAI) description -+ 2) Digital audio interface configuration -+ 3) PCM's description -+ 4) Codec control IO - using I2C, 3 Wire(SPI) or both API's -+ 5) Mixers and audio controls -+ 6) Sysclk configuration -+ 7) Codec audio operations -+ -+Optionally, codec drivers can also provide:- -+ -+ 8) DAPM description. -+ 9) DAPM event handler. -+10) DAC Digital mute control. -+ -+It's probably best to use this guide in conjuction with the existing codec -+driver code in sound/soc/codecs/ -+ -+ASoC Codec driver breakdown -+=========================== -+ -+1 - Digital Audio Interface (DAI) description -+--------------------------------------------- -+The DAI is a digital audio data transfer link between the codec and host SoC -+CPU. It typically has data transfer capabilities in both directions -+(playback and capture) and can run at a variety of different speeds. -+Supported interfaces currently include AC97, I2S and generic PCM style links. -+Please read DAI.txt for implementation information. -+ -+ -+2 - Digital Audio Interface (DAI) configuration -+----------------------------------------------- -+DAI configuration is handled by the codec_pcm_prepare function and is -+responsible for configuring and starting the DAI on the codec. This can be -+called multiple times and is atomic. It can access the runtime parameters. -+ -+This usually consists of a large function with numerous switch statements to -+set up each configuration option. These options are set by the core at runtime. -+ -+ -+3 - Codec PCM's -+--------------- -+Each codec must have it's PCM's defined. This defines the number of channels, -+stream names, callbacks and codec name. It is also used to register the DAI -+with the ASoC core. The PCM structure also associates the DAI capabilities with -+the ALSA PCM. -+ -+e.g. -+ -+static struct snd_soc_pcm_codec wm8731_pcm_client = { -+ .name = "WM8731", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = wm8731_config_sysclk, -+ .ops = { -+ .prepare = wm8731_pcm_prepare, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8731_hwfmt), -+ .modes = &wm8731_hwfmt[0], -+ }, -+}; -+ -+ -+4 - Codec control IO -+-------------------- -+The codec can ususally be controlled via an I2C or SPI style interface (AC97 -+combines control with data in the DAI). The codec drivers will have to provide -+functions to read and write the codec registers along with supplying a register -+cache:- -+ -+ /* IO control data and register cache */ -+ void *control_data; /* codec control (i2c/3wire) data */ -+ void *reg_cache; -+ -+Codec read/write should do any data formatting and call the hardware read write -+below to perform the IO. These functions are called by the core and alsa when -+performing DAPM or changing the mixer:- -+ -+ unsigned int (*read)(struct snd_soc_codec *, unsigned int); -+ int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); -+ -+Codec hardware IO functions - usually points to either the I2C, SPI or AC97 -+read/write:- -+ -+ hw_write_t hw_write; -+ hw_read_t hw_read; -+ -+ -+5 - Mixers and audio controls -+----------------------------- -+All the codec mixers and audio controls can be defined using the convenience -+macros defined in soc.h. -+ -+ #define SOC_SINGLE(xname, reg, shift, mask, invert) -+ -+Defines a single control as follows:- -+ -+ xname = Control name e.g. "Playback Volume" -+ reg = codec register -+ shift = control bit(s) offset in register -+ mask = control bit size(s) e.g. mask of 7 = 3 bits -+ invert = the control is inverted -+ -+Other macros include:- -+ -+ #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) -+ -+A stereo control -+ -+ #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) -+ -+A stereo control spanning 2 registers -+ -+ #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) -+ -+Defines an single enumerated control as follows:- -+ -+ xreg = register -+ xshift = control bit(s) offset in register -+ xmask = control bit(s) size -+ xtexts = pointer to array of strings that describe each setting -+ -+ #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) -+ -+Defines a stereo enumerated control -+ -+ -+6 - System clock configuration. -+------------------------------- -+The system clock that drives the audio subsystem can change depending on sample -+rate and the system power state. i.e. -+ -+o Higher sample rates sometimes need a higher system clock. -+o Low system power states can sometimes limit the available clocks. -+ -+This function is a callback that the machine driver can call to set and -+determine if the clock and sample rate combination is supported by the codec at -+the present time (and system state). -+ -+NOTE: If the codec has a PLL then it has a lot more flexability wrt clock and -+sample rate combinations. -+ -+Your config_sysclock function should return the MCLK if it's a valid -+combination for your codec else 0; -+ -+Please read clocking.txt now. -+ -+ -+7 - Codec Audio Operations -+-------------------------- -+The codec driver also supports the following alsa operations:- -+ -+/* SoC audio ops */ -+struct snd_soc_ops { -+ int (*startup)(snd_pcm_substream_t *); -+ void (*shutdown)(snd_pcm_substream_t *); -+ int (*hw_params)(snd_pcm_substream_t *, snd_pcm_hw_params_t *); -+ int (*hw_free)(snd_pcm_substream_t *); -+ int (*prepare)(snd_pcm_substream_t *); -+}; -+ -+Please refer to the alsa driver PCM documentation for details. -+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm -+ -+ -+8 - DAPM description. -+--------------------- -+The Dynamic Audio Power Management description describes the codec's power -+components, their relationships and registers to the ASoC core. Please read -+dapm.txt for details of building the description. -+ -+Please also see the examples in other codec drivers. -+ -+ -+9 - DAPM event handler -+---------------------- -+This function is a callback that handles codec domain PM calls and system -+domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep -+when not in use. -+ -+Power states:- -+ -+ SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, clk and osc on, active */ -+ -+ SNDRV_CTL_POWER_D1: /* partial On */ -+ SNDRV_CTL_POWER_D2: /* partial On */ -+ -+ SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, inactive */ -+ -+ SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ -+ -+ -+10 - Codec DAC digital mute control. -+------------------------------------ -+Most codecs have a digital mute before the DAC's that can be used to minimise -+any system noise. The mute stops any digital data from entering the DAC. -+ -+A callback can be created that is called by the core for each codec DAI when the -+mute is applied or freed. -+ -+i.e. -+ -+static int wm8974_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; -+ if(mute) -+ wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); -+ else -+ wm8974_write(codec, WM8974_DAC, mute_reg); -+ return 0; -+} -Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/dapm.txt -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/dapm.txt -@@ -0,0 +1,297 @@ -+Dynamic Audio Power Management for Portable Devices -+=================================================== -+ -+1. Description -+============== -+ -+Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices -+to use the minimum amount of power within the audio subsystem at all times. It -+is independent of other kernel PM and as such, can easily co-exist with the -+other PM systems. -+ -+DAPM is also completely transparent to all user space applications as all power -+switching is done within the ASoC core. No code changes or recompiling are -+required for user space applications. DAPM makes power switching descisions based -+upon any audio stream (capture/playback) activity and audio mixer settings -+within the device. -+ -+DAPM spans the whole machine. It covers power control within the entire audio -+subsystem, this includes internal codec power blocks and machine level power -+systems. -+ -+There are 4 power domains within DAPM -+ -+ 1. Codec domain - VREF, VMID (core codec and audio power) -+ Usually controlled at codec probe/remove and suspend/resume, although -+ can be set at stream time if power is not needed for sidetone, etc. -+ -+ 2. Platform/Machine domain - physically connected inputs and outputs -+ Is platform/machine and user action specific, is configured by the -+ machine driver and responds to asynchronous events e.g when HP -+ are inserted -+ -+ 3. Path domain - audio susbsystem signal paths -+ Automatically set when mixer and mux settings are changed by the user. -+ e.g. alsamixer, amixer. -+ -+ 4. Stream domain - DAC's and ADC's. -+ Enabled and disabled when stream playback/capture is started and -+ stopped respectively. e.g. aplay, arecord. -+ -+All DAPM power switching descisons are made automatically by consulting an audio -+routing map of the whole machine. This map is specific to each machine and -+consists of the interconnections between every audio component (including -+internal codec components). All audio components that effect power are called -+widgets hereafter. -+ -+ -+2. DAPM Widgets -+=============== -+ -+Audio DAPM widgets fall into a number of types:- -+ -+ o Mixer - Mixes several analog signals into a single analog signal. -+ o Mux - An analog switch that outputs only 1 of it's inputs. -+ o PGA - A programmable gain amplifier or attenuation widget. -+ o ADC - Analog to Digital Converter -+ o DAC - Digital to Analog Converter -+ o Switch - An analog switch -+ o Input - A codec input pin -+ o Output - A codec output pin -+ o Headphone - Headphone (and optional Jack) -+ o Mic - Mic (and optional Jack) -+ o Line - Line Input/Output (and optional Jack) -+ o Speaker - Speaker -+ o Pre - Special PRE widget (exec before all others) -+ o Post - Special POST widget (exec after all others) -+ -+(Widgets are defined in include/sound/soc-dapm.h) -+ -+Widgets are usually added in the codec driver and the machine driver. There are -+convience macros defined in soc-dapm.h that can be used to quickly build a -+list of widgets of the codecs and machines DAPM widgets. -+ -+Most widgets have a name, register, shift and invert. Some widgets have extra -+parameters for stream name and kcontrols. -+ -+ -+2.1 Stream Domain Widgets -+------------------------- -+ -+Stream Widgets relate to the stream power domain and only consist of ADC's -+(analog to digital converters) and DAC's (digital to analog converters). -+ -+Stream widgets have the following format:- -+ -+SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), -+ -+NOTE: the stream name must match the corresponding stream name in your codecs -+snd_soc_codec_dai. -+ -+e.g. stream widgets for HiFi playback and capture -+ -+SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), -+SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), -+ -+ -+2.2 Path Domain Widgets -+----------------------- -+ -+Path domain widgets have a ability to control or effect the audio signal or -+audio paths within the audio subsystem. They have the following form:- -+ -+SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) -+ -+Any widget kcontrols can be set using the controls and num_controls members. -+ -+e.g. Mixer widget (the kcontrols are declared first) -+ -+/* Output Mixer */ -+static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = { -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), -+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0), -+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), -+}; -+ -+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, -+ ARRAY_SIZE(wm8731_output_mixer_controls)), -+ -+ -+2.3 Platform/Machine domain Widgets -+----------------------------------- -+ -+Machine widgets are different from codec widgets in that they don't have a -+codec register bit associated with them. A machine widget is assigned to each -+machine audio component (non codec) that can be independently powered. e.g. -+ -+ o Speaker Amp -+ o Microphone Bias -+ o Jack connectors -+ -+A machine widget can have an optional call back. -+ -+e.g. Jack connector widget for an external Mic that enables Mic Bias -+when the Mic is inserted:- -+ -+static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) -+{ -+ if(SND_SOC_DAPM_EVENT_ON(event)) -+ set_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS); -+ else -+ reset_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS); -+ -+ return 0; -+} -+ -+SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), -+ -+ -+2.4 Codec Domain -+---------------- -+ -+The Codec power domain has no widgets and is handled by the codecs DAPM event -+handler. This handler is called when the codec powerstate is changed wrt to any -+stream event or by kernel PM events. -+ -+ -+2.5 Virtual Widgets -+------------------- -+ -+Sometimes widgets exist in the codec or machine audio map that don't have any -+corresponding register bit for power control. In this case it's necessary to -+create a virtual widget - a widget with no control bits e.g. -+ -+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), -+ -+This can be used to merge to signal paths together in software. -+ -+After all the widgets have been defined, they can then be added to the DAPM -+subsystem individually with a call to snd_soc_dapm_new_control(). -+ -+ -+3. Codec Widget Interconnections -+================================ -+ -+Widgets are connected to each other within the codec and machine by audio -+paths (called interconnections). Each interconnection must be defined in order -+to create a map of all audio paths between widgets. -+This is easiest with a diagram of the codec (and schematic of the machine audio -+system), as it requires joining widgets together via their audio signal paths. -+ -+i.e. from the WM8731 codec's output mixer (wm8731.c) -+ -+The WM8731 output mixer has 3 inputs (sources) -+ -+ 1. Line Bypass Input -+ 2. DAC (HiFi playback) -+ 3. Mic Sidetone Input -+ -+Each input in this example has a kcontrol associated with it (defined in example -+above) and is connected to the output mixer via it's kcontrol name. We can now -+connect the destination widget (wrt audio signal) with it's source widgets. -+ -+ /* output mixer */ -+ {"Output Mixer", "Line Bypass Switch", "Line Input"}, -+ {"Output Mixer", "HiFi Playback Switch", "DAC"}, -+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, -+ -+So we have :- -+ -+ Destination Widget <=== Path Name <=== Source Widget -+ -+Or:- -+ -+ Sink, Path, Source -+ -+Or :- -+ -+ "Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch". -+ -+When there is no path name connecting widgets (e.g. a direct connection) we -+pass NULL for the path name. -+ -+Interconnections are created with a call to:- -+ -+snd_soc_dapm_connect_input(codec, sink, path, source); -+ -+Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and -+interconnections have been registered with the core. This causes the core to -+scan the codec and machine so that the internal DAPM state matches the -+physical state of the machine. -+ -+ -+3.1 Machine Widget Interconnections -+----------------------------------- -+Machine widget interconnections are created in the same way as codec ones and -+directly connect the codec pins to machine level widgets. -+ -+e.g. connects the speaker out codec pins to the internal speaker. -+ -+ /* ext speaker connected to codec pins LOUT2, ROUT2 */ -+ {"Ext Spk", NULL , "ROUT2"}, -+ {"Ext Spk", NULL , "LOUT2"}, -+ -+This allows the DAPM to power on and off pins that are connected (and in use) -+and pins that are NC respectively. -+ -+ -+4 Endpoint Widgets -+=================== -+An endpoint is a start or end point (widget) of an audio signal within the -+machine and includes the codec. e.g. -+ -+ o Headphone Jack -+ o Internal Speaker -+ o Internal Mic -+ o Mic Jack -+ o Codec Pins -+ -+When a codec pin is NC it can be marked as not used with a call to -+ -+snd_soc_dapm_set_endpoint(codec, "Widget Name", 0); -+ -+The last argument is 0 for inactive and 1 for active. This way the pin and its -+input widget will never be powered up and consume power. -+ -+This also applies to machine widgets. e.g. if a headphone is connected to a -+jack then the jack can be marked active. If the headphone is removed, then -+the headphone jack can be marked inactive. -+ -+ -+5 DAPM Widget Events -+==================== -+ -+Some widgets can register their interest with the DAPM core in PM events. -+e.g. A Speaker with an amplifier registers a widget so the amplifier can be -+powered only when the spk is in use. -+ -+/* turn speaker amplifier on/off depending on use */ -+static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) -+{ -+ if (SND_SOC_DAPM_EVENT_ON(event)) -+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); -+ else -+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); -+ -+ return 0; -+} -+ -+/* corgi machine dapm widgets */ -+static const struct snd_soc_dapm_widget wm8731_dapm_widgets = -+ SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); -+ -+Please see soc-dapm.h for all other widgets that support events. -+ -+ -+5.1 Event types -+--------------- -+ -+The following event types are supported by event widgets. -+ -+/* dapm event types */ -+#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ -+#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ -+#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ -+#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ -+#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ -+#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ -Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/machine.txt -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/machine.txt -@@ -0,0 +1,114 @@ -+ASoC Machine Driver -+=================== -+ -+The ASoC machine (or board) driver is the code that glues together the platform -+and codec drivers. -+ -+The machine driver can contain codec and platform specific code. It registers -+the audio subsystem with the kernel as a platform device and is represented by -+the following struct:- -+ -+/* SoC machine */ -+struct snd_soc_machine { -+ char *name; -+ -+ int (*probe)(struct platform_device *pdev); -+ int (*remove)(struct platform_device *pdev); -+ -+ /* the pre and post PM functions are used to do any PM work before and -+ * after the codec and DAI's do any PM work. */ -+ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); -+ int (*suspend_post)(struct platform_device *pdev, pm_message_t state); -+ int (*resume_pre)(struct platform_device *pdev); -+ int (*resume_post)(struct platform_device *pdev); -+ -+ /* machine stream operations */ -+ struct snd_soc_ops *ops; -+ -+ /* CPU <--> Codec DAI links */ -+ struct snd_soc_dai_link *dai_link; -+ int num_links; -+}; -+ -+probe()/remove() -+---------------- -+probe/remove are optional. Do any machine specific probe here. -+ -+ -+suspend()/resume() -+------------------ -+The machine driver has pre and post versions of suspend and resume to take care -+of any machine audio tasks that have to be done before or after the codec, DAI's -+and DMA is suspended and resumed. Optional. -+ -+ -+Machine operations -+------------------ -+The machine specific audio operations can be set here. Again this is optional. -+ -+ -+Machine DAI Configuration -+------------------------- -+The machine DAI configuration glues all the codec and CPU DAI's together. It can -+also be used to set up the DAI system clock and for any machine related DAI -+initialisation e.g. the machine audio map can be connected to the codec audio -+map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c -+for examples. -+ -+struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. -+ -+/* corgi digital audio interface glue - connects codec <--> CPU */ -+static struct snd_soc_dai_link corgi_dai = { -+ .name = "WM8731", -+ .stream_name = "WM8731", -+ .cpu_dai = &pxa_i2s_dai, -+ .codec_dai = &wm8731_dai, -+ .init = corgi_wm8731_init, -+ .config_sysclk = corgi_config_sysclk, -+}; -+ -+struct snd_soc_machine then sets up the machine with it's DAI's. e.g. -+ -+/* corgi audio machine driver */ -+static struct snd_soc_machine snd_soc_machine_corgi = { -+ .name = "Corgi", -+ .dai_link = &corgi_dai, -+ .num_links = 1, -+ .ops = &corgi_ops, -+}; -+ -+ -+Machine Audio Subsystem -+----------------------- -+ -+The machine soc device glues the platform, machine and codec driver together. -+Private data can also be set here. e.g. -+ -+/* corgi audio private data */ -+static struct wm8731_setup_data corgi_wm8731_setup = { -+ .i2c_address = 0x1b, -+}; -+ -+/* corgi audio subsystem */ -+static struct snd_soc_device corgi_snd_devdata = { -+ .machine = &snd_soc_machine_corgi, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8731, -+ .codec_data = &corgi_wm8731_setup, -+}; -+ -+ -+Machine Power Map -+----------------- -+ -+The machine driver can optionally extend the codec power map and to become an -+audio power map of the audio subsystem. This allows for automatic power up/down -+of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack -+sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for -+details. -+ -+ -+Machine Controls -+---------------- -+ -+Machine specific audio mixer controls can be added in the dai init function. -\ No newline at end of file -Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/overview.txt -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/overview.txt -@@ -0,0 +1,83 @@ -+ALSA SoC Layer -+============== -+ -+The overall project goal of the ALSA System on Chip (ASoC) layer is to provide -+better ALSA support for embedded system on chip procesors (e.g. pxa2xx, au1x00, -+iMX, etc) and portable audio codecs. Currently there is some support in the -+kernel for SoC audio, however it has some limitations:- -+ -+ * Currently, codec drivers are often tightly coupled to the underlying SoC -+ cpu. This is not ideal and leads to code duplication i.e. Linux now has 4 -+ different wm8731 drivers for 4 different SoC platforms. -+ -+ * There is no standard method to signal user initiated audio events. -+ e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion -+ event. These are quite common events on portable devices and ofter require -+ machine specific code to re route audio, enable amps etc after such an event. -+ -+ * Current drivers tend to power up the entire codec when playing -+ (or recording) audio. This is fine for a PC, but tends to waste a lot of -+ power on portable devices. There is also no support for saving power via -+ changing codec oversampling rates, bias currents, etc. -+ -+ -+ASoC Design -+=========== -+ -+The ASoC layer is designed to address these issues and provide the following -+features :- -+ -+ * Codec independence. Allows reuse of codec drivers on other platforms -+ and machines. -+ -+ * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface -+ and codec registers it's audio interface capabilities with the core and are -+ subsequently matched and configured when the application hw params are known. -+ -+ * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to -+ it's minimum power state at all times. This includes powering up/down -+ internal power blocks depending on the internal codec audio routing and any -+ active streams. -+ -+ * Pop and click reduction. Pops and clicks can be reduced by powering the -+ codec up/down in the correct sequence (including using digital mute). ASoC -+ signals the codec when to change power states. -+ -+ * Machine specific controls: Allow machines to add controls to the sound card -+ e.g. volume control for speaker amp. -+ -+To achieve all this, ASoC basically splits an embedded audio system into 3 -+components :- -+ -+ * Codec driver: The codec driver is platform independent and contains audio -+ controls, audio interface capabilities, codec dapm definition and codec IO -+ functions. -+ -+ * Platform driver: The platform driver contains the audio dma engine and audio -+ interface drivers (e.g. I2S, AC97, PCM) for that platform. -+ -+ * Machine driver: The machine driver handles any machine specific controls and -+ audio events. i.e. turing on an amp at start of playback. -+ -+ -+Documentation -+============= -+ -+The documentation is spilt into the following sections:- -+ -+overview.txt: This file. -+ -+codec.txt: Codec driver internals. -+ -+DAI.txt: Description of Digital Audio Interface standards and how to configure -+a DAI within your codec and CPU DAI drivers. -+ -+dapm.txt: Dynamic Audio Power Management -+ -+platform.txt: Platform audio DMA and DAI. -+ -+machine.txt: Machine driver internals. -+ -+pop_clicks.txt: How to minimise audio artifacts. -+ -+clocking.txt: ASoC clocking for best power performance. -\ No newline at end of file -Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/platform.txt -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/platform.txt -@@ -0,0 +1,58 @@ -+ASoC Platform Driver -+==================== -+ -+An ASoC platform driver can be divided into audio DMA and SoC DAI configuration -+and control. The platform drivers only target the SoC CPU and must have no board -+specific code. -+ -+Audio DMA -+========= -+ -+The platform DMA driver optionally supports the following alsa operations:- -+ -+/* SoC audio ops */ -+struct snd_soc_ops { -+ int (*startup)(snd_pcm_substream_t *); -+ void (*shutdown)(snd_pcm_substream_t *); -+ int (*hw_params)(snd_pcm_substream_t *, snd_pcm_hw_params_t *); -+ int (*hw_free)(snd_pcm_substream_t *); -+ int (*prepare)(snd_pcm_substream_t *); -+ int (*trigger)(snd_pcm_substream_t *, int); -+}; -+ -+The platform driver exports it's DMA functionailty via struct snd_soc_platform:- -+ -+struct snd_soc_platform { -+ char *name; -+ -+ int (*probe)(struct platform_device *pdev); -+ int (*remove)(struct platform_device *pdev); -+ int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); -+ int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); -+ -+ /* pcm creation and destruction */ -+ int (*pcm_new)(snd_card_t *, struct snd_soc_codec_dai *, snd_pcm_t *); -+ void (*pcm_free)(snd_pcm_t *); -+ -+ /* platform stream ops */ -+ snd_pcm_ops_t *pcm_ops; -+}; -+ -+Please refer to the alsa driver documentation for details of audio DMA. -+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm -+ -+An example DMA driver is soc/pxa/pxa2xx-pcm.c -+ -+ -+SoC DAI Drivers -+=============== -+ -+Each SoC DAI driver must provide the following features:- -+ -+ 1) Digital audio interface (DAI) description -+ 2) Digital audio interface configuration -+ 3) PCM's description -+ 4) Sysclk configuration -+ 5) Suspend and resume (optional) -+ -+Please see codec.txt for a description of items 1 - 4. -Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/pops_clicks.txt -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/pops_clicks.txt -@@ -0,0 +1,52 @@ -+Audio Pops and Clicks -+===================== -+ -+Pops and clicks are unwanted audio artifacts caused by the powering up and down -+of components within the audio subsystem. This is noticable on PC's when an audio -+module is either loaded or unloaded (at module load time the sound card is -+powered up and causes a popping noise on the speakers). -+ -+Pops and clicks can be more frequent on portable systems with DAPM. This is because -+the components within the subsystem are being dynamically powered depending on -+the audio usage and this can subsequently cause a small pop or click every time a -+component power state is changed. -+ -+ -+Minimising Playback Pops and Clicks -+=================================== -+ -+Playback pops in portable audio subsystems cannot be completely eliminated atm, -+however future audio codec hardware will have better pop and click supression. -+Pops can be reduced within playback by powering the audio components in a -+specific order. This order is different for startup and shutdown and follows -+some basic rules:- -+ -+ Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute -+ -+ Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC -+ -+This assumes that the codec PCM output path from the DAC is via a mixer and then -+a PGA (programmable gain amplifier) before being output to the speakers. -+ -+ -+Minimising Capture Pops and Clicks -+================================== -+ -+Capture artifacts are somewhat easier to get rid as we can delay activating the -+ADC until all the pops have occured. This follows similar power rules to -+playback in that components are powered in a sequence depending upon stream -+startup or shutdown. -+ -+ Startup Order - Input PGA --> Mixers --> ADC -+ -+ Shutdown Order - ADC --> Mixers --> Input PGA -+ -+ -+Zipper Noise -+============ -+An unwanted zipper noise can occur within the audio playback or capture stream -+when a volume control is changed near its maximum gain value. The zipper noise -+is heard when the gain increase or decrease changes the mean audio signal -+amplitude too quickly. It can be minimised by enabling the zero cross setting -+for each volume control. The ZC forces the gain change to occur when the signal -+crosses the zero amplitude line. -Index: linux-2.6-pxa-new/include/sound/ac97_codec.h -=================================================================== ---- linux-2.6-pxa-new.orig/include/sound/ac97_codec.h -+++ linux-2.6-pxa-new/include/sound/ac97_codec.h -@@ -425,6 +425,7 @@ struct snd_ac97_build_ops { - - struct snd_ac97_bus_ops { - void (*reset) (struct snd_ac97 *ac97); -+ void (*warm_reset)(struct snd_ac97 *ac97); - void (*write) (struct snd_ac97 *ac97, unsigned short reg, unsigned short val); - unsigned short (*read) (struct snd_ac97 *ac97, unsigned short reg); - void (*wait) (struct snd_ac97 *ac97); -Index: linux-2.6-pxa-new/include/sound/soc-dapm.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/include/sound/soc-dapm.h -@@ -0,0 +1,286 @@ -+/* -+ * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management -+ * -+ * Author: Liam Girdwood -+ * Created: Aug 11th 2005 -+ * Copyright: Wolfson Microelectronics. PLC. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef __LINUX_SND_SOC_DAPM_H -+#define __LINUX_SND_SOC_DAPM_H -+ -+#include <linux/device.h> -+#include <linux/types.h> -+#include <sound/control.h> -+#include <sound/soc.h> -+ -+/* widget has no PM register bit */ -+#define SND_SOC_NOPM -1 -+ -+/* -+ * SoC dynamic audio power managment -+ * -+ * We can have upto 4 power domains -+ * 1. Codec domain - VREF, VMID -+ * Usually controlled at codec probe/remove, although can be set -+ * at stream time if power is not needed for sidetone, etc. -+ * 2. Platform/Machine domain - physically connected inputs and outputs -+ * Is platform/machine and user action specific, is set in the machine -+ * driver and by userspace e.g when HP are inserted -+ * 3. Path domain - Internal codec path mixers -+ * Are automatically set when mixer and mux settings are -+ * changed by the user. -+ * 4. Stream domain - DAC's and ADC's. -+ * Enabled when stream playback/capture is started. -+ */ -+ -+/* codec domain */ -+#define SND_SOC_DAPM_VMID(wname) \ -+{ .id = snd_soc_dapm_vmid, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0} -+ -+/* platform domain */ -+#define SND_SOC_DAPM_INPUT(wname) \ -+{ .id = snd_soc_dapm_input, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0} -+#define SND_SOC_DAPM_OUTPUT(wname) \ -+{ .id = snd_soc_dapm_output, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0} -+#define SND_SOC_DAPM_MIC(wname, wevent) \ -+{ .id = snd_soc_dapm_mic, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0, .event = wevent, \ -+ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} -+#define SND_SOC_DAPM_HP(wname, wevent) \ -+{ .id = snd_soc_dapm_hp, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0, .event = wevent, \ -+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} -+#define SND_SOC_DAPM_SPK(wname, wevent) \ -+{ .id = snd_soc_dapm_spk, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0, .event = wevent, \ -+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} -+#define SND_SOC_DAPM_LINE(wname, wevent) \ -+{ .id = snd_soc_dapm_line, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0, .event = wevent, \ -+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} -+ -+/* path domain */ -+#define SND_SOC_DAPM_PGA(wname, wreg, wshift, winvert,\ -+ wcontrols, wncontrols) \ -+{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} -+#define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \ -+ wcontrols, wncontrols)\ -+{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} -+#define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \ -+{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0} -+#define SND_SOC_DAPM_SWITCH(wname, wreg, wshift, winvert, wcontrols) \ -+{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1} -+#define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \ -+{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1} -+ -+/* path domain with event - event handler must return 0 for success */ -+#define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \ -+ wncontrols, wevent, wflags) \ -+{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ -+ .event = wevent, .event_flags = wflags} -+#define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \ -+ wncontrols, wevent, wflags) \ -+{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ -+ .event = wevent, .event_flags = wflags} -+#define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \ -+{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0, \ -+ .event = wevent, .event_flags = wflags} -+#define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \ -+ wevent, wflags) \ -+{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1 \ -+ .event = wevent, .event_flags = wflags} -+#define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \ -+ wevent, wflags) \ -+{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \ -+ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ -+ .event = wevent, .event_flags = wflags} -+ -+/* events that are pre and post DAPM */ -+#define SND_SOC_DAPM_PRE(wname, wevent) \ -+{ .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0, .event = wevent, \ -+ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD} -+#define SND_SOC_DAPM_POST(wname, wevent) \ -+{ .id = snd_soc_dapm_post, .name = wname, .kcontrols = NULL, \ -+ .num_kcontrols = 0, .event = wevent, \ -+ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} -+ -+/* stream domain */ -+#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ -+{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ -+ .shift = wshift, .invert = winvert} -+#define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \ -+{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ -+ .shift = wshift, .invert = winvert} -+ -+/* dapm kcontrol types */ -+#define SOC_DAPM_SINGLE(xname, reg, shift, mask, invert) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -+ .info = snd_soc_info_volsw, \ -+ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ -+ .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } -+#define SOC_DAPM_DOUBLE(xname, reg, shift_left, shift_right, mask, invert, \ -+ power) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ -+ .info = snd_soc_info_volsw, \ -+ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ -+ .private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\ -+ ((mask) << 16) | ((invert) << 24) } -+#define SOC_DAPM_ENUM(xname, xenum) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -+ .info = snd_soc_info_enum_double, \ -+ .get = snd_soc_dapm_get_enum_double, \ -+ .put = snd_soc_dapm_put_enum_double, \ -+ .private_value = (unsigned long)&xenum } -+ -+/* dapm stream operations */ -+#define SND_SOC_DAPM_STREAM_NOP 0x0 -+#define SND_SOC_DAPM_STREAM_START 0x1 -+#define SND_SOC_DAPM_STREAM_STOP 0x2 -+#define SND_SOC_DAPM_STREAM_SUSPEND 0x4 -+#define SND_SOC_DAPM_STREAM_RESUME 0x8 -+#define SND_SOC_DAPM_STREAM_PAUSE_PUSH 0x10 -+#define SND_SOC_DAPM_STREAM_PAUSE_RELEASE 0x20 -+ -+/* dapm event types */ -+#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ -+#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ -+#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ -+#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ -+#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ -+#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ -+ -+/* convenience event type detection */ -+#define SND_SOC_DAPM_EVENT_ON(e) \ -+ (e & (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU)) -+#define SND_SOC_DAPM_EVENT_OFF(e) \ -+ (e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)) -+ -+struct snd_soc_dapm_widget; -+enum snd_soc_dapm_type; -+struct snd_soc_dapm_path; -+struct snd_soc_dapm_pin; -+ -+/* dapm controls */ -+int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_dapm_new_control(struct snd_soc_codec *codec, -+ const struct snd_soc_dapm_widget *widget); -+ -+/* dapm path setup */ -+int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, -+ const char *sink_name, const char *control_name, const char *src_name); -+int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); -+void snd_soc_dapm_free(struct snd_soc_device *socdev); -+ -+/* dapm events */ -+int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, -+ int event); -+ -+/* dapm sys fs - used by the core */ -+int snd_soc_dapm_sys_add(struct device *dev); -+ -+/* dapm audio endpoint control */ -+int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, -+ char *pin, int status); -+int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec); -+ -+/* dapm widget types */ -+enum snd_soc_dapm_type { -+ snd_soc_dapm_input = 0, /* input pin */ -+ snd_soc_dapm_output, /* output pin */ -+ snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ -+ snd_soc_dapm_mixer, /* mixes several analog signals together */ -+ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ -+ snd_soc_dapm_adc, /* analog to digital converter */ -+ snd_soc_dapm_dac, /* digital to analog converter */ -+ snd_soc_dapm_micbias, /* microphone bias (power) */ -+ snd_soc_dapm_mic, /* microphone */ -+ snd_soc_dapm_hp, /* headphones */ -+ snd_soc_dapm_spk, /* speaker */ -+ snd_soc_dapm_line, /* line input/output */ -+ snd_soc_dapm_switch, /* analog switch */ -+ snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */ -+ snd_soc_dapm_pre, /* machine specific pre widget - exec first */ -+ snd_soc_dapm_post, /* machine specific post widget - exec last */ -+}; -+ -+/* dapm audio path between two widgets */ -+struct snd_soc_dapm_path { -+ char *name; -+ char *long_name; -+ -+ /* source (input) and sink (output) widgets */ -+ struct snd_soc_dapm_widget *source; -+ struct snd_soc_dapm_widget *sink; -+ struct snd_kcontrol *kcontrol; -+ -+ /* status */ -+ u32 connect:1; /* source and sink widgets are connected */ -+ u32 walked:1; /* path has been walked */ -+ -+ struct list_head list_source; -+ struct list_head list_sink; -+ struct list_head list; -+}; -+ -+/* dapm widget */ -+struct snd_soc_dapm_widget { -+ enum snd_soc_dapm_type id; -+ char *name; /* widget name */ -+ char *sname; /* stream name */ -+ struct snd_soc_codec *codec; -+ struct list_head list; -+ -+ /* dapm control */ -+ short reg; /* negative reg = no direct dapm */ -+ unsigned char shift; /* bits to shift */ -+ unsigned int saved_value; /* widget saved value */ -+ unsigned int value; /* widget current value */ -+ unsigned char power:1; /* block power status */ -+ unsigned char invert:1; /* invert the power bit */ -+ unsigned char active:1; /* active stream on DAC, ADC's */ -+ unsigned char connected:1; /* connected codec pin */ -+ unsigned char new:1; /* cnew complete */ -+ unsigned char ext:1; /* has external widgets */ -+ unsigned char muted:1; /* muted for pop reduction */ -+ unsigned char suspend:1; /* was active before suspend */ -+ unsigned char pmdown:1; /* waiting for timeout */ -+ -+ /* external events */ -+ unsigned short event_flags; /* flags to specify event types */ -+ int (*event)(struct snd_soc_dapm_widget*, int); -+ -+ /* kcontrols that relate to this widget */ -+ int num_kcontrols; -+ const struct snd_kcontrol_new *kcontrols; -+ -+ /* widget input and outputs */ -+ struct list_head sources; -+ struct list_head sinks; -+}; -+ -+#endif -Index: linux-2.6-pxa-new/include/sound/soc.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/include/sound/soc.h -@@ -0,0 +1,487 @@ -+/* -+ * linux/sound/soc.h -- ALSA SoC Layer -+ * -+ * Author: Liam Girdwood -+ * Created: Aug 11th 2005 -+ * Copyright: Wolfson Microelectronics. PLC. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef __LINUX_SND_SOC_H -+#define __LINUX_SND_SOC_H -+ -+#include <linux/platform_device.h> -+#include <linux/types.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/control.h> -+#include <sound/ac97_codec.h> -+ -+#define SND_SOC_VERSION "0.12.4" -+ -+/* -+ * Convenience kcontrol builders -+ */ -+#define SOC_SINGLE_VALUE(reg,shift,mask,invert) ((reg) | ((shift) << 8) |\ -+ ((shift) << 12) | ((mask) << 16) | ((invert) << 24)) -+#define SOC_SINGLE_VALUE_EXT(reg,mask,invert) ((reg) | ((mask) << 16) |\ -+ ((invert) << 31)) -+#define SOC_SINGLE(xname, reg, shift, mask, invert) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -+ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ -+ .put = snd_soc_put_volsw, \ -+ .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } -+#define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ -+ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ -+ .put = snd_soc_put_volsw, \ -+ .private_value = (reg) | ((shift_left) << 8) | \ -+ ((shift_right) << 12) | ((mask) << 16) | ((invert) << 24) } -+#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ -+ .info = snd_soc_info_volsw_2r, \ -+ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ -+ .private_value = (reg_left) | ((shift) << 8) | \ -+ ((mask) << 12) | ((invert) << 20) | ((reg_right) << 24) } -+#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ -+{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ -+ .mask = xmask, .texts = xtexts } -+#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) \ -+ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xtexts) -+#define SOC_ENUM_SINGLE_EXT(xmask, xtexts) \ -+{ .mask = xmask, .texts = xtexts } -+#define SOC_ENUM(xname, xenum) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ -+ .info = snd_soc_info_enum_double, \ -+ .get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \ -+ .private_value = (unsigned long)&xenum } -+#define SOC_SINGLE_EXT(xname, xreg, xmask, xinvert,\ -+ xhandler_get, xhandler_put) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -+ .info = snd_soc_info_volsw_ext, \ -+ .get = xhandler_get, .put = xhandler_put, \ -+ .private_value = SOC_SINGLE_VALUE_EXT(xreg, xmask, xinvert) } -+#define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -+ .info = snd_soc_info_bool_ext, \ -+ .get = xhandler_get, .put = xhandler_put, \ -+ .private_value = xdata } -+#define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ -+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -+ .info = snd_soc_info_enum_ext, \ -+ .get = xhandler_get, .put = xhandler_put, \ -+ .private_value = (unsigned long)&xenum } -+ -+/* -+ * Digital Audio Interface (DAI) types -+ */ -+#define SND_SOC_DAI_AC97 0x1 -+#define SND_SOC_DAI_I2S 0x2 -+#define SND_SOC_DAI_PCM 0x4 -+ -+/* -+ * DAI hardware audio formats -+ */ -+#define SND_SOC_DAIFMT_I2S (1 << 0) /* I2S mode */ -+#define SND_SOC_DAIFMT_RIGHT_J (1 << 1) /* Right justified mode */ -+#define SND_SOC_DAIFMT_LEFT_J (1 << 2) /* Left Justified mode */ -+#define SND_SOC_DAIFMT_DSP_A (1 << 3) /* L data msb after FRM or LRC */ -+#define SND_SOC_DAIFMT_DSP_B (1 << 4) /* L data msb during FRM or LRC */ -+#define SND_SOC_DAIFMT_AC97 (1 << 5) /* AC97 */ -+ -+/* -+ * DAI hardware signal inversions -+ */ -+#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */ -+#define SND_SOC_DAIFMT_NB_IF (1 << 9) /* normal bclk + inv frm */ -+#define SND_SOC_DAIFMT_IB_NF (1 << 10) /* invert bclk + nor frm */ -+#define SND_SOC_DAIFMT_IB_IF (1 << 11) /* invert bclk + frm */ -+ -+/* -+ * DAI hardware clock masters -+ * This is wrt the codec, the inverse is true for the interface -+ * i.e. if the codec is clk and frm master then the interface is -+ * clk and frame slave. -+ */ -+#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & frm master */ -+#define SND_SOC_DAIFMT_CBS_CFM (1 << 13) /* codec clk slave & frm master */ -+#define SND_SOC_DAIFMT_CBM_CFS (1 << 14) /* codec clk master & frame slave */ -+#define SND_SOC_DAIFMT_CBS_CFS (1 << 15) /* codec clk & frm slave */ -+ -+#define SND_SOC_DAIFMT_FORMAT_MASK 0x00ff -+#define SND_SOC_DAIFMT_INV_MASK 0x0f00 -+#define SND_SOC_DAIFMT_CLOCK_MASK 0xf000 -+ -+/* -+ * DAI hardware audio direction -+ */ -+#define SND_SOC_DAIDIR_PLAYBACK 0x1 -+#define SND_SOC_DAIDIR_CAPTURE 0x2 -+ -+/* -+ * DAI hardware Time Division Multiplexing (TDM) Slots -+ * Left and Right data word positions -+ * This is measured in words (sample size) and not bits. -+ */ -+#define SND_SOC_DAITDM_LRDW(l,r) ((l << 8) | r) -+ -+/* -+ * DAI hardware clock ratios -+ * bit clock can either be a generated by dividing mclk or -+ * by multiplying sample rate, hence there are 2 definitions below -+ * depending on codec type. -+ */ -+/* ratio of sample rate to mclk/sysclk */ -+#define SND_SOC_FS_ALL 0xffff /* all mclk supported */ -+ -+/* bit clock dividers */ -+#define SND_SOC_FSBD(x) (1 << (x - 1)) /* ratio mclk:bclk */ -+#define SND_SOC_FSBD_REAL(x) (ffs(x)) -+ -+/* bit clock ratio to (sample rate * channels * word size) */ -+#define SND_SOC_FSBW(x) (1 << (x - 1)) -+#define SND_SOC_FSBW_REAL(x) (ffs(x)) -+/* all bclk ratios supported */ -+#define SND_SOC_FSB_ALL ~0ULL -+ -+/* -+ * DAI hardware flags -+ */ -+/* use bfs mclk divider mode (BCLK = MCLK / x) */ -+#define SND_SOC_DAI_BFS_DIV 0x1 -+/* use bfs rate mulitplier (BCLK = RATE * x)*/ -+#define SND_SOC_DAI_BFS_RATE 0x2 -+/* use bfs rcw multiplier (BCLK = RATE * CHN * WORD SIZE) */ -+#define SND_SOC_DAI_BFS_RCW 0x4 -+/* capture and playback can use different clocks */ -+#define SND_SOC_DAI_ASYNC 0x8 -+/* can use gated BCLK */ -+#define SND_SOC_DAI_GATED 0x10 -+ -+/* -+ * AC97 codec ID's bitmask -+ */ -+#define SND_SOC_DAI_AC97_ID0 (1 << 0) -+#define SND_SOC_DAI_AC97_ID1 (1 << 1) -+#define SND_SOC_DAI_AC97_ID2 (1 << 2) -+#define SND_SOC_DAI_AC97_ID3 (1 << 3) -+ -+struct snd_soc_device; -+struct snd_soc_pcm_stream; -+struct snd_soc_ops; -+struct snd_soc_dai_mode; -+struct snd_soc_pcm_runtime; -+struct snd_soc_codec_dai; -+struct snd_soc_cpu_dai; -+struct snd_soc_codec; -+struct snd_soc_machine_config; -+struct soc_enum; -+struct snd_soc_ac97_ops; -+struct snd_soc_clock_info; -+ -+typedef int (*hw_write_t)(void *,const char* ,int); -+typedef int (*hw_read_t)(void *,char* ,int); -+ -+extern struct snd_ac97_bus_ops soc_ac97_ops; -+ -+/* pcm <-> DAI connect */ -+void snd_soc_free_pcms(struct snd_soc_device *socdev); -+int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); -+int snd_soc_register_card(struct snd_soc_device *socdev); -+ -+/* set runtime hw params */ -+int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, -+ const struct snd_pcm_hardware *hw); -+int snd_soc_get_rate(int rate); -+ -+/* codec IO */ -+#define snd_soc_read(codec, reg) codec->read(codec, reg) -+#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value) -+ -+/* codec register bit access */ -+int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, -+ unsigned short mask, unsigned short value); -+int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, -+ unsigned short mask, unsigned short value); -+ -+int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, -+ struct snd_ac97_bus_ops *ops, int num); -+void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); -+ -+/* -+ *Controls -+ */ -+struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, -+ void *data, char *long_name); -+int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo); -+int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo); -+int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo); -+int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo); -+int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo); -+int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo); -+int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol); -+ -+/* SoC PCM stream information */ -+struct snd_soc_pcm_stream { -+ char *stream_name; -+ unsigned int rate_min; /* min rate */ -+ unsigned int rate_max; /* max rate */ -+ unsigned int channels_min; /* min channels */ -+ unsigned int channels_max; /* max channels */ -+ unsigned int active:1; /* stream is in use */ -+}; -+ -+/* SoC audio ops */ -+struct snd_soc_ops { -+ int (*startup)(struct snd_pcm_substream *); -+ void (*shutdown)(struct snd_pcm_substream *); -+ int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); -+ int (*hw_free)(struct snd_pcm_substream *); -+ int (*prepare)(struct snd_pcm_substream *); -+ int (*trigger)(struct snd_pcm_substream *, int); -+}; -+ -+/* SoC DAI hardware mode */ -+struct snd_soc_dai_mode { -+ u16 fmt; /* SND_SOC_DAIFMT_* */ -+ u16 tdm; /* SND_SOC_HWTDM_* */ -+ u64 pcmfmt; /* SNDRV_PCM_FMTBIT_* */ -+ u16 pcmrate; /* SND_SOC_HWRATE_* */ -+ u16 pcmdir:2; /* SND_SOC_HWDIR_* */ -+ u16 flags:8; /* hw flags */ -+ u16 fs; /* mclk to rate divider */ -+ u64 bfs; /* mclk to bclk dividers */ -+ unsigned long priv; /* private mode data */ -+}; -+ -+/* DAI capabilities */ -+struct snd_soc_dai_cap { -+ int num_modes; /* number of DAI modes */ -+ struct snd_soc_dai_mode *mode; /* array of supported DAI modes */ -+}; -+ -+/* SoC Codec DAI */ -+struct snd_soc_codec_dai { -+ char *name; -+ int id; -+ -+ /* DAI capabilities */ -+ struct snd_soc_pcm_stream playback; -+ struct snd_soc_pcm_stream capture; -+ struct snd_soc_dai_cap caps; -+ -+ /* DAI runtime info */ -+ struct snd_soc_dai_mode dai_runtime; -+ struct snd_soc_ops ops; -+ unsigned int (*config_sysclk)(struct snd_soc_codec_dai*, -+ struct snd_soc_clock_info *info, unsigned int clk); -+ int (*digital_mute)(struct snd_soc_codec *, -+ struct snd_soc_codec_dai*, int); -+ unsigned int mclk; /* the audio master clock */ -+ unsigned int pll_in; /* the PLL input clock */ -+ unsigned int pll_out; /* the PLL output clock */ -+ unsigned int clk_div; /* internal clock divider << 1 (for fractions) */ -+ unsigned int active; -+ unsigned char pop_wait:1; -+ -+ /* DAI private data */ -+ void *private_data; -+}; -+ -+/* SoC CPU DAI */ -+struct snd_soc_cpu_dai { -+ -+ /* DAI description */ -+ char *name; -+ unsigned int id; -+ unsigned char type; -+ -+ /* DAI callbacks */ -+ int (*probe)(struct platform_device *pdev); -+ void (*remove)(struct platform_device *pdev); -+ int (*suspend)(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *cpu_dai); -+ int (*resume)(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *cpu_dai); -+ unsigned int (*config_sysclk)(struct snd_soc_cpu_dai *cpu_dai, -+ struct snd_soc_clock_info *info, unsigned int clk); -+ -+ /* DAI capabilities */ -+ struct snd_soc_pcm_stream capture; -+ struct snd_soc_pcm_stream playback; -+ struct snd_soc_dai_cap caps; -+ -+ /* DAI runtime info */ -+ struct snd_soc_dai_mode dai_runtime; -+ struct snd_soc_ops ops; -+ struct snd_pcm_runtime *runtime; -+ unsigned char active:1; -+ unsigned int mclk; -+ void *dma_data; -+ -+ /* DAI private data */ -+ void *private_data; -+}; -+ -+/* SoC Audio Codec */ -+struct snd_soc_codec { -+ char *name; -+ struct module *owner; -+ struct mutex mutex; -+ -+ /* callbacks */ -+ int (*dapm_event)(struct snd_soc_codec *codec, int event); -+ -+ /* runtime */ -+ struct snd_card *card; -+ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ -+ unsigned int active; -+ unsigned int pcm_devs; -+ void *private_data; -+ -+ /* codec IO */ -+ void *control_data; /* codec control (i2c/3wire) data */ -+ unsigned int (*read)(struct snd_soc_codec *, unsigned int); -+ int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); -+ hw_write_t hw_write; -+ hw_read_t hw_read; -+ void *reg_cache; -+ short reg_cache_size; -+ short reg_cache_step; -+ -+ /* dapm */ -+ struct list_head dapm_widgets; -+ struct list_head dapm_paths; -+ unsigned int dapm_state; -+ unsigned int suspend_dapm_state; -+ -+ /* codec DAI's */ -+ struct snd_soc_codec_dai *dai; -+ unsigned int num_dai; -+}; -+ -+/* codec device */ -+struct snd_soc_codec_device { -+ int (*probe)(struct platform_device *pdev); -+ int (*remove)(struct platform_device *pdev); -+ int (*suspend)(struct platform_device *pdev, pm_message_t state); -+ int (*resume)(struct platform_device *pdev); -+}; -+ -+/* SoC platform interface */ -+struct snd_soc_platform { -+ char *name; -+ -+ int (*probe)(struct platform_device *pdev); -+ int (*remove)(struct platform_device *pdev); -+ int (*suspend)(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *cpu_dai); -+ int (*resume)(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *cpu_dai); -+ -+ /* pcm creation and destruction */ -+ int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, -+ struct snd_pcm *); -+ void (*pcm_free)(struct snd_pcm *); -+ -+ /* platform stream ops */ -+ struct snd_pcm_ops *pcm_ops; -+}; -+ -+/* SoC machine DAI configuration, glues a codec and cpu DAI together */ -+struct snd_soc_dai_link { -+ char *name; /* Codec name */ -+ char *stream_name; /* Stream name */ -+ -+ /* DAI */ -+ struct snd_soc_codec_dai *codec_dai; -+ struct snd_soc_cpu_dai *cpu_dai; -+ u32 flags; /* DAI config preference flags */ -+ -+ /* codec/machine specific init - e.g. add machine controls */ -+ int (*init)(struct snd_soc_codec *codec); -+ -+ /* audio sysclock configuration */ -+ unsigned int (*config_sysclk)(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info); -+}; -+ -+/* SoC machine */ -+struct snd_soc_machine { -+ char *name; -+ -+ int (*probe)(struct platform_device *pdev); -+ int (*remove)(struct platform_device *pdev); -+ -+ /* the pre and post PM functions are used to do any PM work before and -+ * after the codec and DAI's do any PM work. */ -+ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); -+ int (*suspend_post)(struct platform_device *pdev, pm_message_t state); -+ int (*resume_pre)(struct platform_device *pdev); -+ int (*resume_post)(struct platform_device *pdev); -+ -+ /* machine stream operations */ -+ struct snd_soc_ops *ops; -+ -+ /* CPU <--> Codec DAI links */ -+ struct snd_soc_dai_link *dai_link; -+ int num_links; -+}; -+ -+/* SoC Device - the audio subsystem */ -+struct snd_soc_device { -+ struct device *dev; -+ struct snd_soc_machine *machine; -+ struct snd_soc_platform *platform; -+ struct snd_soc_codec *codec; -+ struct snd_soc_codec_device *codec_dev; -+ void *codec_data; -+}; -+ -+/* runtime channel data */ -+struct snd_soc_pcm_runtime { -+ struct snd_soc_codec_dai *codec_dai; -+ struct snd_soc_cpu_dai *cpu_dai; -+ struct snd_soc_device *socdev; -+}; -+ -+/* enumerated kcontrol */ -+struct soc_enum { -+ unsigned short reg; -+ unsigned short reg2; -+ unsigned char shift_l; -+ unsigned char shift_r; -+ unsigned int mask; -+ const char **texts; -+ void *dapm; -+}; -+ -+/* clocking configuration data */ -+struct snd_soc_clock_info { -+ unsigned int rate; -+ unsigned int fs; -+ unsigned int bclk_master; -+}; -+ -+#endif -Index: linux-2.6-pxa-new/sound/Kconfig -=================================================================== ---- linux-2.6-pxa-new.orig/sound/Kconfig -+++ linux-2.6-pxa-new/sound/Kconfig -@@ -76,6 +76,8 @@ source "sound/sparc/Kconfig" - - source "sound/parisc/Kconfig" - -+source "sound/soc/Kconfig" -+ - endmenu - - menu "Open Sound System" -Index: linux-2.6-pxa-new/sound/soc/Kconfig -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/Kconfig -@@ -0,0 +1,37 @@ -+# -+# SoC audio configuration -+# -+ -+menu "SoC audio support" -+ depends on SND!=n -+ -+config SND_SOC_AC97_BUS -+ bool -+ -+config SND_SOC -+ tristate "SoC audio support" -+ ---help--- -+ -+ If you want SoC support, you should say Y here and also to the -+ specific driver for your SoC below. You will also need to select the -+ specific codec(s) attached to the SoC -+ -+ This SoC audio support can also be built as a module. If so, the module -+ will be called snd-soc-core. -+ -+# All the supported Soc's -+menu "Soc Platforms" -+depends on SND_SOC -+source "sound/soc/pxa/Kconfig" -+source "sound/soc/at91/Kconfig" -+source "sound/soc/imx/Kconfig" -+source "sound/soc/s3c24xx/Kconfig" -+endmenu -+ -+# Supported codecs -+menu "Soc Codecs" -+depends on SND_SOC -+source "sound/soc/codecs/Kconfig" -+endmenu -+ -+endmenu -Index: linux-2.6-pxa-new/sound/soc/Makefile -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/Makefile -@@ -0,0 +1,4 @@ -+snd-soc-core-objs := soc-core.o soc-dapm.o -+ -+obj-$(CONFIG_SND_SOC) += snd-soc-core.o -+obj-$(CONFIG_SND_SOC) += pxa/ at91/ imx/ s3c24xx/ codecs/ -Index: linux-2.6-pxa-new/sound/soc/codecs/Kconfig -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/Kconfig -@@ -0,0 +1,90 @@ -+config SND_SOC_AC97_CODEC -+ tristate "SoC generic AC97 support" -+ depends SND_SOC -+ help -+ Say Y or M if you want generic AC97 support. This is not required -+ for the AC97 codecs listed below. -+ -+config SND_SOC_WM8711 -+ tristate "SoC driver for the WM8711 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8711 codec. -+ -+config SND_SOC_WM8510 -+ tristate "SoC driver for the WM8510 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8711 codec. -+ -+config SND_SOC_WM8731 -+ tristate "SoC driver for the WM8731 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8731 codec. -+ -+config SND_SOC_WM8750 -+ tristate "SoC driver for the WM8750 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8750 codec. -+ -+config SND_SOC_WM8753 -+ tristate "SoC driver for the WM8753 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8753 codec. -+ -+config SND_SOC_WM8772 -+ tristate "SoC driver for the WM8772 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8772 codec. -+ -+config SND_SOC_WM8971 -+ tristate "SoC driver for the WM8971 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8971 codec. -+ -+config SND_SOC_WM8976 -+ tristate "SoC driver for the WM8976 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8976 codec. -+ -+config SND_SOC_WM8974 -+ tristate "SoC driver for the WM8974 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8974 codec. -+ -+config SND_SOC_WM8980 -+ tristate "SoC driver for the WM8980 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM8980 codec. -+ -+config SND_SOC_WM9713 -+ tristate "SoC driver for the WM9713 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM9713 codec. -+ -+config SND_SOC_WM9712 -+ tristate "SoC driver for the WM9712 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the WM9712 codec. -+ -+config SND_SOC_UDA1380 -+ tristate "SoC driver for the UDA1380 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the UDA1380 codec. -+ -+config SND_SOC_AK4535 -+ tristate "SoC driver for the AK4535 codec" -+ depends SND_SOC -+ help -+ Say Y or M if you want to support the AK4535 codec. -Index: linux-2.6-pxa-new/sound/soc/codecs/Makefile -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/Makefile -@@ -0,0 +1,31 @@ -+snd-soc-ac97-objs := ac97.o -+snd-soc-wm8711-objs := wm8711.o -+snd-soc-wm8510-objs := wm8510.o -+snd-soc-wm8731-objs := wm8731.o -+snd-soc-wm8750-objs := wm8750.o -+snd-soc-wm8753-objs := wm8753.o -+snd-soc-wm8772-objs := wm8772.o -+snd-soc-wm8971-objs := wm8971.o -+snd-soc-wm8974-objs := wm8974.o -+snd-soc-wm8976-objs := wm8976.o -+snd-soc-wm8980-objs := wm8980.o -+snd-soc-uda1380-objs := uda1380.o -+snd-soc-ak4535-objs := ak4535.o -+snd-soc-wm9713-objs := wm9713.o -+snd-soc-wm9712-objs := wm9712.o -+ -+obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o -+obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o -+obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o -+obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o -+obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o -+obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o -+obj-$(CONFIG_SND_SOC_WM8772) += snd-soc-wm8772.o -+obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o -+obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o -+obj-$(CONFIG_SND_SOC_WM8976) += snd-soc-wm8976.o -+obj-$(CONFIG_SND_SOC_WM8980) += snd-soc-wm8980.o -+obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o -+obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o -+obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o -+obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o -Index: linux-2.6-pxa-new/sound/soc/codecs/ac97.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/ac97.c -@@ -0,0 +1,167 @@ -+/* -+ * ac97.c -- ALSA Soc AC97 codec support -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 17th Oct 2005 Initial version. -+ * -+ * Generic AC97 support. -+ */ -+ -+#include <linux/init.h> -+#include <linux/kernel.h> -+#include <linux/device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/ac97_codec.h> -+#include <sound/initval.h> -+#include <sound/soc.h> -+ -+#define AC97_VERSION "0.5" -+ -+#define AC97_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define AC97_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+/* may need to expand this */ -+static struct snd_soc_dai_mode soc_ac97[] = { -+ {0, 0, SNDRV_PCM_FMTBIT_S16_LE, AC97_RATES}, -+ {0, 0, SNDRV_PCM_FMTBIT_S18_3LE, AC97_RATES}, -+ {0, 0, SNDRV_PCM_FMTBIT_S20_3LE, AC97_RATES}, -+}; -+ -+static int ac97_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? -+ AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; -+ return snd_ac97_set_rate(codec->ac97, reg, runtime->rate); -+} -+ -+static struct snd_soc_codec_dai ac97_dai = { -+ .name = "AC97 HiFi", -+ .playback = { -+ .stream_name = "AC97 Playback", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .stream_name = "AC97 Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .prepare = ac97_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(soc_ac97), -+ .mode = soc_ac97,}, -+}; -+ -+static unsigned int ac97_read(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ return soc_ac97_ops.read(codec->ac97, reg); -+} -+ -+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int val) -+{ -+ soc_ac97_ops.write(codec->ac97, reg, val); -+ return 0; -+} -+ -+static int ac97_soc_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec; -+ struct snd_ac97_bus *ac97_bus; -+ struct snd_ac97_template ac97_template; -+ int ret = 0; -+ -+ printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); -+ -+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (socdev->codec == NULL) -+ return -ENOMEM; -+ codec = socdev->codec; -+ mutex_init(&codec->mutex); -+ -+ codec->name = "AC97"; -+ codec->owner = THIS_MODULE; -+ codec->dai = &ac97_dai; -+ codec->num_dai = 1; -+ codec->write = ac97_write; -+ codec->read = ac97_read; -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if(ret < 0) -+ goto err; -+ -+ /* add codec as bus device for standard ac97 */ -+ ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus); -+ if(ret < 0) -+ goto bus_err; -+ -+ memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); -+ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97); -+ if(ret < 0) -+ goto bus_err; -+ -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) -+ goto bus_err; -+ return 0; -+ -+bus_err: -+ snd_soc_free_pcms(socdev); -+ -+err: -+ kfree(socdev->codec->reg_cache); -+ kfree(socdev->codec); -+ socdev->codec = NULL; -+ return ret; -+} -+ -+static int ac97_soc_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if(codec == NULL) -+ return 0; -+ -+ snd_soc_free_pcms(socdev); -+ kfree(socdev->codec->reg_cache); -+ kfree(socdev->codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_ac97= { -+ .probe = ac97_soc_probe, -+ .remove = ac97_soc_remove, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_ac97); -+ -+MODULE_DESCRIPTION("Soc Generic AC97 driver"); -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/ac97.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/ac97.h -@@ -0,0 +1,18 @@ -+/* -+ * linux/sound/codecs/ac97.h -- ALSA SoC Layer -+ * -+ * Author: Liam Girdwood -+ * Created: Dec 1st 2005 -+ * Copyright: Wolfson Microelectronics. PLC. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef __LINUX_SND_SOC_AC97_H -+#define __LINUX_SND_SOC_AC97_H -+ -+extern struct snd_soc_codec_device soc_codec_dev_ac97; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/ak4535.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/ak4535.c -@@ -0,0 +1,701 @@ -+/* -+ * ak4535.c -- AK4535 ALSA Soc Audio driver -+ * -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Author: Richard Purdie <richard@openedhand.com> -+ * -+ * Based on wm8753.c by Liam Girdwood -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "ak4535.h" -+ -+#define AUDIO_NAME "ak4535" -+#define AK4535_VERSION "0.3" -+ -+struct snd_soc_codec_device soc_codec_dev_ak4535; -+ -+/* -+ * ak4535 register cache -+ */ -+static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { -+ 0x0000, 0x0080, 0x0000, 0x0003, -+ 0x0002, 0x0000, 0x0011, 0x0001, -+ 0x0000, 0x0040, 0x0036, 0x0010, -+ 0x0000, 0x0000, 0x0057, 0x0000, -+}; -+ -+#define AK4535_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | \ -+ SND_SOC_DAIFMT_NB_NF) -+ -+#define AK4535_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define AK4535_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+static struct snd_soc_dai_mode ak4535_modes[] = { -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = AK4535_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = AK4535_RATES, -+ .pcmdir = AK4535_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ { -+ .fmt = AK4535_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = AK4535_RATES, -+ .pcmdir = AK4535_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 32, -+ }, -+}; -+ -+/* -+ * read ak4535 register cache -+ */ -+static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= AK4535_CACHEREGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write ak4535 register cache -+ */ -+static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, -+ u16 reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= AK4535_CACHEREGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+/* -+ * write to the AK4535 register space -+ */ -+static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D8 AK4535 register offset -+ * D7...D0 register data -+ */ -+ data[0] = reg & 0xff; -+ data[1] = value & 0xff; -+ -+ ak4535_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -EIO; -+} -+ -+static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; -+static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; -+static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; -+static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"}; -+static const char *ak4535_mic_select[] = {"Internal", "External"}; -+ -+static const struct soc_enum ak4535_enum[] = { -+ SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain), -+ SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out), -+ SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out), -+ SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp), -+ SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select), -+}; -+ -+static const struct snd_kcontrol_new ak4535_snd_controls[] = { -+ SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0), -+ SOC_ENUM("Mono 1 Output", ak4535_enum[1]), -+ SOC_ENUM("Mono 1 Gain", ak4535_enum[0]), -+ SOC_ENUM("Headphone Output", ak4535_enum[2]), -+ SOC_ENUM("Playback Deemphasis", ak4535_enum[3]), -+ SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0), -+ SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0), -+ SOC_ENUM("Mic Select", ak4535_enum[4]), -+ SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0), -+ SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0), -+ SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0), -+ SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0), -+ SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0), -+ SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0), -+ SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0), -+ SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1), -+ SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1), -+ SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0), -+ SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), -+}; -+ -+/* add non dapm controls */ -+static int ak4535_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&ak4535_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* Mono 1 Mixer */ -+static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { -+ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), -+ SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0), -+}; -+ -+/* Stereo Mixer */ -+static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = { -+ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0), -+ SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0), -+ SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0), -+}; -+ -+/* Input Mixer */ -+static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = { -+ SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0), -+ SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0), -+}; -+ -+/* Input mux */ -+static const struct snd_kcontrol_new ak4535_input_mux_control = -+ SOC_DAPM_ENUM("Input Select", ak4535_enum[0]); -+ -+/* HP L switch */ -+static const struct snd_kcontrol_new ak4535_hpl_control = -+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1); -+ -+/* HP R switch */ -+static const struct snd_kcontrol_new ak4535_hpr_control = -+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1); -+ -+/* Speaker switch */ -+static const struct snd_kcontrol_new ak4535_spk_control = -+ SOC_DAPM_SINGLE("Switch", AK4535_MODE2, 0, 0, 0); -+ -+/* mono 2 switch */ -+static const struct snd_kcontrol_new ak4535_mono2_control = -+ SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0); -+ -+/* Line out switch */ -+static const struct snd_kcontrol_new ak4535_line_control = -+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0); -+ -+/* ak4535 dapm widgets */ -+static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { -+ SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, -+ &ak4535_stereo_mixer_controls[0], -+ ARRAY_SIZE(ak4535_stereo_mixer_controls)), -+ SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, -+ &ak4535_mono1_mixer_controls[0], -+ ARRAY_SIZE(ak4535_mono1_mixer_controls)), -+ SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, -+ &ak4535_input_mixer_controls[0], -+ ARRAY_SIZE(ak4535_mono1_mixer_controls)), -+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, -+ &ak4535_input_mux_control), -+ SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0), -+ SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, -+ &ak4535_mono2_control), -+ SND_SOC_DAPM_SWITCH("Speaker Enable", SND_SOC_NOPM, 0, 0, -+ &ak4535_spk_control), -+ SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, -+ &ak4535_line_control), -+ SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0, -+ &ak4535_hpl_control), -+ SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0, -+ &ak4535_hpr_control), -+ SND_SOC_DAPM_OUTPUT("LOUT"), -+ SND_SOC_DAPM_OUTPUT("HPL"), -+ SND_SOC_DAPM_OUTPUT("ROUT"), -+ SND_SOC_DAPM_OUTPUT("HPR"), -+ SND_SOC_DAPM_OUTPUT("SPP"), -+ SND_SOC_DAPM_OUTPUT("SPN"), -+ SND_SOC_DAPM_OUTPUT("MOUT1"), -+ SND_SOC_DAPM_OUTPUT("MOUT2"), -+ SND_SOC_DAPM_OUTPUT("MICOUT"), -+ SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 1), -+ SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0), -+ -+ SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0), -+ SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0), -+ SND_SOC_DAPM_INPUT("MICIN"), -+ SND_SOC_DAPM_INPUT("MICEXT"), -+ SND_SOC_DAPM_INPUT("AUX"), -+ SND_SOC_DAPM_INPUT("MIN"), -+ SND_SOC_DAPM_INPUT("AIN"), -+}; -+ -+static const char *audio_map[][3] = { -+ /*stereo mixer */ -+ {"Stereo Mixer", "Playback Switch", "DAC"}, -+ {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, -+ {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, -+ -+ /* mono1 mixer */ -+ {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"}, -+ {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, -+ -+ /* mono2 mixer */ -+ {"Mono2 Mixer", "Mono Playback Switch", "Stereo Mixer"}, -+ -+ /* Mic */ -+ {"AIN", NULL, "Mic"}, -+ {"Input Mux", "Internal", "Mic Int Bias"}, -+ {"Input Mux", "External", "Mic Ext Bias"}, -+ {"Mic Int Bias", NULL, "MICIN"}, -+ {"Mic Ext Bias", NULL, "MICEXT"}, -+ {"MICOUT", NULL, "Input Mux"}, -+ -+ /* line out */ -+ {"LOUT", "Switch", "Line"}, -+ {"ROUT", "Switch", "Line Out Enable"}, -+ {"Line Out Enable", NULL, "Line Out"}, -+ {"Line Out", NULL, "Stereo Mixer"}, -+ -+ /* mono1 out */ -+ {"MOUT1", NULL, "Mono Out"}, -+ {"Mono Out", NULL, "Mono Mixer"}, -+ -+ /* left HP */ -+ {"HPL", "Switch", "Left HP Enable"}, -+ {"Left HP Enable", NULL, "HP L Amp"}, -+ {"HP L Amp", NULL, "Stereo Mixer"}, -+ -+ /* right HP */ -+ {"HPR", "Switch", "Right HP Enable"}, -+ {"Right HP Enable", NULL, "HP R Amp"}, -+ {"HP R Amp", NULL, "Stereo Mixer"}, -+ -+ /* speaker */ -+ {"SPP", "Switch", "Speaker Enable"}, -+ {"SPN", "Switch", "Speaker Enable"}, -+ {"Speaker Enable", NULL, "Spk Amp"}, -+ {"Spk Amp", NULL, "MIN"}, -+ -+ /* mono 2 */ -+ {"MOUT2", "Switch", "Mono 2 Enable"}, -+ {"Mono 2 Enable", NULL, "Stereo Mixer"}, -+ -+ /* Aux In */ -+ {"Aux In", NULL, "AUX"}, -+ -+ /* ADC */ -+ {"ADC", NULL, "Input Mixer"}, -+ {"Input Mixer", "Mic Capture Switch", "Mic"}, -+ {"Input Mixer", "Aux Capture Switch", "Aux In"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int ak4535_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(ak4535_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &ak4535_dapm_widgets[i]); -+ } -+ -+ /* set up audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+static int ak4535_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u8 mode = 0, mode2; -+ int bfs; -+ -+ mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2); -+ bfs = SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs); -+ snd_assert(bfs, return -ENODEV); -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ mode = 0x0002; -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ mode = 0x0001; -+ break; -+ } -+ -+ /* set fs */ -+ switch (rtd->codec_dai->dai_runtime.fs) { -+ case 1024: -+ mode2 |= (0x3 << 5); -+ break; -+ case 512: -+ mode2 |= (0x2 << 5); -+ break; -+ case 256: -+ mode2 |= (0x1 << 5); -+ break; -+ } -+ -+ /* bfs */ -+ if (bfs == 64) -+ mode |= 0x4; -+ -+ /* set rate */ -+ ak4535_write(codec, AK4535_MODE1, mode); -+ ak4535_write(codec, AK4535_MODE2, mode2); -+ -+ return 0; -+} -+ -+static unsigned int ak4535_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ if (info->fs != 256) -+ return 0; -+ -+ /* we only support 256 FS atm */ -+ if (info->rate * info->fs == clk) { -+ dai->mclk = clk; -+ return clk; -+ } -+ -+ return 0; -+} -+ -+static int ak4535_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; -+ if (mute) -+ ak4535_write(codec, AK4535_DAC, mute_reg); -+ else -+ ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); -+ return 0; -+} -+ -+static int ak4535_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, clk and osc on, dac unmute, active */ -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, dac mute, inactive */ -+ ak4535_write(codec, AK4535_PM1, 0x80); -+ ak4535_write(codec, AK4535_PM2, 0x0); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, inactive */ -+ ak4535_write(codec, AK4535_PM1, 0x0); -+ ak4535_write(codec, AK4535_PM2, 0x80); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai ak4535_dai = { -+ .name = "AK4535", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = ak4535_config_sysclk, -+ .digital_mute = ak4535_mute, -+ .ops = { -+ .prepare = ak4535_pcm_prepare, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(ak4535_modes), -+ .mode = ak4535_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(ak4535_dai); -+ -+static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int ak4535_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(ak4535_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ ak4535_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the AK4535 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int ak4535_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int ret = 0; -+ -+ codec->name = "AK4535"; -+ codec->owner = THIS_MODULE; -+ codec->read = ak4535_read_reg_cache; -+ codec->write = ak4535_write; -+ codec->dapm_event = ak4535_dapm_event; -+ codec->dai = &ak4535_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(ak4535_reg); -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(ak4535_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, ak4535_reg, -+ sizeof(u16) * ARRAY_SIZE(ak4535_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ak4535_reg); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if (ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ ak4535_add_controls(codec); -+ ak4535_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *ak4535_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver ak4535_i2c_driver; -+static struct i2c_client client_template; -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = ak4535_socdev; -+ struct ak4535_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL){ -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ i2c_set_clientdata(i2c, codec); -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if (ret < 0) { -+ printk(KERN_ERR "failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = ak4535_init(socdev); -+ if (ret < 0) { -+ printk(KERN_ERR "failed to initialise AK4535\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+} -+ -+static int ak4535_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec* codec = i2c_get_clientdata(client); -+ i2c_detach_client(client); -+ kfree(codec->reg_cache); -+ kfree(client); -+ -+ return 0; -+} -+ -+static int ak4535_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, ak4535_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver ak4535_i2c_driver = { -+ .driver = { -+ .name = "AK4535 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_AK4535, -+ .attach_adapter = ak4535_i2c_attach, -+ .detach_client = ak4535_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "AK4535", -+ .driver = &ak4535_i2c_driver, -+}; -+#endif -+ -+static int ak4535_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct ak4535_setup_data *setup; -+ struct snd_soc_codec* codec; -+ int ret = 0; -+ -+ printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ ak4535_socdev = socdev; -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&ak4535_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int ak4535_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec* codec = socdev->codec; -+ -+ if (codec->control_data) -+ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&ak4535_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_ak4535 = { -+ .probe = ak4535_probe, -+ .remove = ak4535_remove, -+ .suspend = ak4535_suspend, -+ .resume = ak4535_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); -+ -+MODULE_DESCRIPTION("Soc AK4535 driver"); -+MODULE_AUTHOR("Richard Purdie"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/ak4535.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/ak4535.h -@@ -0,0 +1,46 @@ -+/* -+ * ak4535.h -- AK4535 Soc Audio driver -+ * -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Author: Richard Purdie <richard@openedhand.com> -+ * -+ * Based on wm8753.h -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _AK4535_H -+#define _AK4535_H -+ -+/* AK4535 register space */ -+ -+#define AK4535_PM1 0x0 -+#define AK4535_PM2 0x1 -+#define AK4535_SIG1 0x2 -+#define AK4535_SIG2 0x3 -+#define AK4535_MODE1 0x4 -+#define AK4535_MODE2 0x5 -+#define AK4535_DAC 0x6 -+#define AK4535_MIC 0x7 -+#define AK4535_TIMER 0x8 -+#define AK4535_ALC1 0x9 -+#define AK4535_ALC2 0xa -+#define AK4535_PGA 0xb -+#define AK4535_LATT 0xc -+#define AK4535_RATT 0xd -+#define AK4535_VOL 0xe -+#define AK4535_STATUS 0xf -+ -+#define AK4535_CACHEREGNUM 0x10 -+ -+struct ak4535_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai ak4535_dai; -+extern struct snd_soc_codec_device soc_codec_dev_ak4535; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/uda1380.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/uda1380.c -@@ -0,0 +1,582 @@ -+/* -+ * uda1380.c - Philips UDA1380 ALSA SoC audio driver -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ * -+ * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC -+ * codec model. -+ * -+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org> -+ * Copyright 2005 Openedhand Ltd. -+ */ -+ -+#include <linux/module.h> -+#include <linux/init.h> -+#include <linux/types.h> -+#include <linux/string.h> -+#include <linux/slab.h> -+#include <linux/errno.h> -+#include <linux/ioctl.h> -+#include <linux/delay.h> -+#include <linux/i2c.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/control.h> -+#include <sound/initval.h> -+#include <sound/info.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include "uda1380.h" -+ -+#define UDA1380_VERSION "0.4" -+ -+/* -+ * uda1380 register cache -+ */ -+static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { -+ 0x0502, 0x0000, 0x0000, 0x3f3f, -+ 0x0202, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0xff00, 0x0000, 0x4800, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x8000, 0x0002, 0x0000, -+}; -+ -+#define UDA1380_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | \ -+ SND_SOC_DAIFMT_NB_NF) -+ -+#define UDA1380_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define UDA1380_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+static struct snd_soc_dai_mode uda1380_modes[] = { -+ /* slave rates capture & playback */ -+ { -+ .fmt = UDA1380_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = UDA1380_RATES, -+ .pcmdir = UDA1380_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ -+ /* slave rates playback */ -+ { -+ .fmt = UDA1380_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+}; -+ -+/* -+ * read uda1380 register cache -+ */ -+static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg == UDA1380_RESET) -+ return 0; -+ if (reg >= UDA1380_CACHEREGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write uda1380 register cache -+ */ -+static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, -+ u16 reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= UDA1380_CACHEREGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+/* -+ * write to the UDA1380 register space -+ */ -+static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[3]; -+ -+ /* data is -+ * data[0] is register offset -+ * data[1] is MS byte -+ * data[2] is LS byte -+ */ -+ data[0] = reg; -+ data[1] = (value & 0xff00) >> 8; -+ data[2] = value & 0x00ff; -+ -+ uda1380_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 3) == 3) -+ return 0; -+ else -+ return -EIO; -+} -+ -+#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) -+ -+/* declarations of ALSA reg_elem_REAL controls */ -+static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz", -+ "96kHz"}; -+static const char *uda1380_input_sel[] = {"Line", "Mic"}; -+ -+static const struct soc_enum uda1380_enum[] = { -+ SOC_ENUM_DOUBLE(UDA1380_DEEMP, 0, 8, 5, uda1380_deemp), -+ SOC_ENUM_SINGLE(UDA1380_ADC, 3, 2, uda1380_input_sel), -+}; -+ -+static const struct snd_kcontrol_new uda1380_snd_controls[] = { -+ SOC_DOUBLE("Playback Volume", UDA1380_MVOL, 0, 8, 127, 0), -+ SOC_DOUBLE("Treble Volume", UDA1380_MODE, 4, 12, 3, 0), -+ SOC_DOUBLE("Bass Volume", UDA1380_MODE, 0, 8, 15, 0), -+ SOC_ENUM("Playback De-emphasis", uda1380_enum[0]), -+ SOC_DOUBLE("Capture Volume", UDA1380_DEC, 0, 8, 127, 0), -+ SOC_DOUBLE("Line Capture Volume", UDA1380_PGA, 0, 8, 15, 0), -+ SOC_SINGLE("Mic Capture Volume", UDA1380_PGA, 8, 11, 0), -+ SOC_DOUBLE("Playback Switch", UDA1380_DEEMP, 3, 11, 1, 0), -+ SOC_SINGLE("Capture Switch", UDA1380_PGA, 15, 1, 0), -+ SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), -+ SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), -+ SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), -+}; -+ -+/* add non dapm controls */ -+static int uda1380_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* Input mux */ -+static const struct snd_kcontrol_new uda1380_input_mux_control = -+ SOC_DAPM_ENUM("Input Select", uda1380_enum[1]); -+ -+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { -+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, -+ &uda1380_input_mux_control), -+ SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0), -+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0), -+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0), -+ SND_SOC_DAPM_INPUT("VINM"), -+ SND_SOC_DAPM_INPUT("VINL"), -+ SND_SOC_DAPM_INPUT("VINR"), -+ SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0), -+ SND_SOC_DAPM_OUTPUT("VOUTLHP"), -+ SND_SOC_DAPM_OUTPUT("VOUTRHP"), -+ SND_SOC_DAPM_OUTPUT("VOUTL"), -+ SND_SOC_DAPM_OUTPUT("VOUTR"), -+ SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0), -+ SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), -+}; -+ -+static const char *audio_map[][3] = { -+ -+ /* analog mixer setup is different from diagram for dapm */ -+ {"HeadPhone Driver", NULL, "Analog Mixer"}, -+ {"VOUTR", NULL, "Analog Mixer"}, -+ {"VOUTL", NULL, "Analog Mixer"}, -+ {"Analog Mixer", NULL, "VINR"}, -+ {"Analog Mixer", NULL, "VINL"}, -+ {"Analog Mixer", NULL, "DAC"}, -+ -+ /* headphone driver */ -+ {"VOUTLHP", NULL, "HeadPhone Driver"}, -+ {"VOUTRHP", NULL, "HeadPhone Driver"}, -+ -+ /* input mux */ -+ {"Left ADC", NULL, "Input Mux"}, -+ {"Input Mux", "Mic", "Mic LNA"}, -+ {"Input Mux", "Line", "Left PGA"}, -+ -+ /* right input */ -+ {"Right ADC", NULL, "Right PGA"}, -+ -+ /* inputs */ -+ {"Mic LNA", NULL, "VINM"}, -+ {"Left PGA", NULL, "VINL"}, -+ {"Right PGA", NULL, "VINR"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int uda1380_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]); -+ } -+ -+ /* set up audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ uda1380_write(codec, UDA1380_CLK, R00_EN_DAC | R00_EN_INT | clk); -+ else -+ uda1380_write(codec, UDA1380_CLK, R00_EN_ADC | R00_EN_DEC | clk); -+ -+ return 0; -+} -+ -+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ uda1380_write(codec, UDA1380_CLK, ~(R00_EN_DAC | R00_EN_INT) & clk); -+ else -+ uda1380_write(codec, UDA1380_CLK, ~(R00_EN_ADC | R00_EN_DEC) & clk); -+} -+ -+static unsigned int uda1380_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ if(info->fs != 256) -+ return 0; -+ -+ /* we only support 256 FS atm */ -+ if(info->rate * info->fs == clk) { -+ dai->mclk = clk; -+ return clk; -+ } -+ -+ return 0; -+} -+ -+static int uda1380_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & 0xbfff; -+ if(mute) -+ uda1380_write(codec, UDA1380_DEEMP, mute_reg | 0x4000); -+ else -+ uda1380_write(codec, UDA1380_DEEMP, mute_reg); -+ return 0; -+} -+ -+static int uda1380_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except internal bias */ -+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, inactive */ -+ uda1380_write(codec, UDA1380_PM, 0x0); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai uda1380_dai = { -+ .name = "UDA1380", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = uda1380_config_sysclk, -+ .digital_mute = uda1380_mute, -+ .ops = { -+ .prepare = uda1380_pcm_prepare, -+ .shutdown = uda1380_pcm_shutdown, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(uda1380_modes), -+ .mode = uda1380_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(uda1380_dai); -+ -+static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int uda1380_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ uda1380_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the UDA1380 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int uda1380_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int ret = 0; -+ -+ codec->name = "UDA1380"; -+ codec->owner = THIS_MODULE; -+ codec->read = uda1380_read_reg_cache; -+ codec->write = uda1380_write; -+ codec->dapm_event = uda1380_dapm_event; -+ codec->dai = &uda1380_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(uda1380_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, uda1380_reg, -+ sizeof(u16) * ARRAY_SIZE(uda1380_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(uda1380_reg); -+ uda1380_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if(ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ uda1380_write(codec, UDA1380_CLK, 0); -+ -+ /* uda1380 init */ -+ uda1380_add_controls(codec); -+ uda1380_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if(ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *uda1380_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver uda1380_i2c_driver; -+static struct i2c_client client_template; -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+ -+static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = uda1380_socdev; -+ struct uda1380_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL){ -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ i2c_set_clientdata(i2c, codec); -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if(ret < 0) { -+ printk(KERN_ERR "failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = uda1380_init(socdev); -+ if(ret < 0) { -+ printk(KERN_ERR "failed to initialise UDA1380\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+} -+ -+static int uda1380_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec* codec = i2c_get_clientdata(client); -+ i2c_detach_client(client); -+ kfree(codec->reg_cache); -+ kfree(client); -+ return 0; -+} -+ -+static int uda1380_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, uda1380_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver uda1380_i2c_driver = { -+ .driver = { -+ .name = "UDA1380 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_UDA1380, -+ .attach_adapter = uda1380_i2c_attach, -+ .detach_client = uda1380_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "UDA1380", -+ .driver = &uda1380_i2c_driver, -+}; -+#endif -+ -+static int uda1380_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct uda1380_setup_data *setup; -+ struct snd_soc_codec* codec; -+ int ret = 0; -+ -+ printk(KERN_INFO "UDA1380 Audio Codec %s", UDA1380_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ uda1380_socdev = socdev; -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&uda1380_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int uda1380_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec* codec = socdev->codec; -+ -+ if (codec->control_data) -+ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&uda1380_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_uda1380 = { -+ .probe = uda1380_probe, -+ .remove = uda1380_remove, -+ .suspend = uda1380_suspend, -+ .resume = uda1380_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); -+ -+MODULE_AUTHOR("Giorgio Padrin"); -+MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/uda1380.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/uda1380.h -@@ -0,0 +1,56 @@ -+/* -+ * Audio support for Philips UDA1380 -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ * -+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org> -+ */ -+ -+#define UDA1380_CLK 0x00 -+#define UDA1380_IFACE 0x01 -+#define UDA1380_PM 0x02 -+#define UDA1380_AMIX 0x03 -+#define UDA1380_HP 0x04 -+#define UDA1380_MVOL 0x10 -+#define UDA1380_MIXVOL 0x11 -+#define UDA1380_MODE 0x12 -+#define UDA1380_DEEMP 0x13 -+#define UDA1380_MIXER 0x14 -+#define UDA1380_INTSTAT 0x18 -+#define UDA1380_DEC 0x20 -+#define UDA1380_PGA 0x21 -+#define UDA1380_ADC 0x22 -+#define UDA1380_AGC 0x23 -+#define UDA1380_DECSTAT 0x28 -+#define UDA1380_RESET 0x7f -+ -+#define UDA1380_CACHEREGNUM 0x24 -+ -+/* Register flags */ -+#define R00_EN_ADC 0x0800 -+#define R00_EN_DEC 0x0400 -+#define R00_EN_DAC 0x0200 -+#define R00_EN_INT 0x0100 -+#define R02_PON_HP 0x2000 -+#define R02_PON_DAC 0x0400 -+#define R02_PON_BIAS 0x0100 -+#define R02_PON_LNA 0x0010 -+#define R02_PON_PGAL 0x0008 -+#define R02_PON_ADCL 0x0004 -+#define R02_PON_PGAR 0x0002 -+#define R02_PON_ADCR 0x0001 -+#define R13_MTM 0x4000 -+#define R21_MT_ADC 0x8000 -+#define R22_SEL_LNA 0x0008 -+#define R22_SEL_MIC 0x0004 -+#define R22_SKIP_DCFIL 0x0002 -+#define R23_AGC_EN 0x0001 -+ -+struct uda1380_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai uda1380_dai; -+extern struct snd_soc_codec_device soc_codec_dev_uda1380; -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8731.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8731.c -@@ -0,0 +1,886 @@ -+/* -+ * wm8731.c -- WM8731 ALSA SoC Audio driver -+ * -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Author: Richard Purdie <richard@openedhand.com> -+ * -+ * Based on wm8753.c by Liam Girdwood -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8731.h" -+ -+#define AUDIO_NAME "wm8731" -+#define WM8731_VERSION "0.12" -+ -+/* -+ * Debug -+ */ -+ -+#define WM8731_DEBUG 0 -+ -+#ifdef WM8731_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+struct snd_soc_codec_device soc_codec_dev_wm8731; -+ -+/* -+ * wm8731 register cache -+ * We can't read the WM8731 register space when we are -+ * using 2 wire for device control, so we cache them instead. -+ * There is no point in caching the reset register -+ */ -+static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { -+ 0x0097, 0x0097, 0x0079, 0x0079, -+ 0x000a, 0x0008, 0x009f, 0x000a, -+ 0x0000, 0x0000 -+}; -+ -+#define WM8731_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ -+ SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8731_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8731_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -+ -+#define WM8731_HIFI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+static struct snd_soc_dai_mode wm8731_modes[] = { -+ /* codec frame and clock master modes */ -+ /* 8k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 1536, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 2304, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 1408, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 2112, -+ .bfs = 64, -+ }, -+ -+ /* 32k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 576, -+ .bfs = 64, -+ }, -+ -+ /* 44.1k & 48k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 64, -+ }, -+ -+ /* 88.2 & 96k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 128, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 192, -+ .bfs = 64, -+ }, -+ -+ /* USB codec frame and clock master modes */ -+ /* 8k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1500, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* 44.1k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 272, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* 48k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 250, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* 88.2k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 136, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* 96k */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 125, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8731_HIFI_BITS, -+ .pcmrate = WM8731_RATES, -+ .pcmdir = WM8731_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8731 register cache -+ */ -+static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg == WM8731_RESET) -+ return 0; -+ if (reg >= WM8731_CACHEREGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write wm8731 register cache -+ */ -+static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec, -+ u16 reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= WM8731_CACHEREGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+/* -+ * write to the WM8731 register space -+ */ -+static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8731 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8731_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -EIO; -+} -+ -+#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0) -+ -+static const char *wm8731_input_select[] = {"Line In", "Mic"}; -+static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -+ -+static const struct soc_enum wm8731_enum[] = { -+ SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select), -+ SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph), -+}; -+ -+static const struct snd_kcontrol_new wm8731_snd_controls[] = { -+ -+SOC_DOUBLE_R("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V, -+ 0, 127, 0), -+SOC_DOUBLE_R("Master Playback ZC Switch", WM8731_LOUT1V, WM8731_ROUT1V, -+ 7, 1, 0), -+ -+SOC_DOUBLE_R("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0), -+SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1), -+ -+SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0), -+SOC_SINGLE("Capture Mic Switch", WM8731_APANA, 1, 1, 1), -+ -+SOC_SINGLE("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1), -+ -+SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1), -+SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0), -+ -+SOC_ENUM("Playback De-emphasis", wm8731_enum[1]), -+}; -+ -+/* add non dapm controls */ -+static int wm8731_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) { -+ if ((err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8731_snd_controls[i],codec, NULL))) < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* Output Mixer */ -+static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = { -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), -+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0), -+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), -+}; -+ -+/* Input mux */ -+static const struct snd_kcontrol_new wm8731_input_mux_controls = -+SOC_DAPM_ENUM("Input Select", wm8731_enum[0]); -+ -+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { -+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, -+ &wm8731_output_mixer_controls[0], -+ ARRAY_SIZE(wm8731_output_mixer_controls)), -+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8731_PWR, 3, 1), -+SND_SOC_DAPM_OUTPUT("LOUT"), -+SND_SOC_DAPM_OUTPUT("LHPOUT"), -+SND_SOC_DAPM_OUTPUT("ROUT"), -+SND_SOC_DAPM_OUTPUT("RHPOUT"), -+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8731_PWR, 2, 1), -+SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &wm8731_input_mux_controls), -+SND_SOC_DAPM_PGA("Line Input", WM8731_PWR, 0, 1, NULL, 0), -+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8731_PWR, 1, 1), -+SND_SOC_DAPM_INPUT("MICIN"), -+SND_SOC_DAPM_INPUT("RLINEIN"), -+SND_SOC_DAPM_INPUT("LLINEIN"), -+}; -+ -+static const char *intercon[][3] = { -+ /* output mixer */ -+ {"Output Mixer", "Line Bypass Switch", "Line Input"}, -+ {"Output Mixer", "HiFi Playback Switch", "DAC"}, -+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, -+ -+ /* outputs */ -+ {"RHPOUT", NULL, "Output Mixer"}, -+ {"ROUT", NULL, "Output Mixer"}, -+ {"LHPOUT", NULL, "Output Mixer"}, -+ {"LOUT", NULL, "Output Mixer"}, -+ -+ /* input mux */ -+ {"Input Mux", "Line In", "Line Input"}, -+ {"Input Mux", "Mic", "Mic Bias"}, -+ {"ADC", NULL, "Input Mux"}, -+ -+ /* inputs */ -+ {"Line Input", NULL, "LLINEIN"}, -+ {"Line Input", NULL, "RLINEIN"}, -+ {"Mic Bias", NULL, "MICIN"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8731_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); -+ } -+ -+ /* set up audio path interconnects */ -+ for(i = 0; intercon[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, intercon[i][0], -+ intercon[i][1], intercon[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+struct _coeff_div { -+ u32 mclk; -+ u32 rate; -+ u16 fs; -+ u8 sr:4; -+ u8 bosr:1; -+ u8 usb:1; -+}; -+ -+/* codec mclk clock divider coefficients */ -+static const struct _coeff_div coeff_div[] = { -+ /* 48k */ -+ {12288000, 48000, 256, 0x0, 0x0, 0x0}, -+ {18432000, 48000, 384, 0x0, 0x1, 0x0}, -+ {12000000, 48000, 250, 0x0, 0x0, 0x1}, -+ -+ /* 32k */ -+ {12288000, 32000, 384, 0x6, 0x0, 0x0}, -+ {18432000, 32000, 576, 0x6, 0x1, 0x0}, -+ -+ /* 8k */ -+ {12288000, 8000, 1536, 0x3, 0x0, 0x0}, -+ {18432000, 8000, 2304, 0x3, 0x1, 0x0}, -+ {11289600, 8000, 1408, 0xb, 0x0, 0x0}, -+ {16934400, 8000, 2112, 0xb, 0x1, 0x0}, -+ {12000000, 8000, 1500, 0x3, 0x0, 0x1}, -+ -+ /* 96k */ -+ {12288000, 96000, 128, 0x7, 0x0, 0x0}, -+ {18432000, 96000, 192, 0x7, 0x1, 0x0}, -+ {12000000, 96000, 125, 0x7, 0x0, 0x1}, -+ -+ /* 44.1k */ -+ {11289600, 44100, 256, 0x8, 0x0, 0x0}, -+ {16934400, 44100, 384, 0x8, 0x1, 0x0}, -+ {12000000, 44100, 272, 0x8, 0x1, 0x1}, -+ -+ /* 88.2k */ -+ {11289600, 88200, 128, 0xf, 0x0, 0x0}, -+ {16934400, 88200, 192, 0xf, 0x1, 0x0}, -+ {12000000, 88200, 136, 0xf, 0x1, 0x1}, -+}; -+ -+static inline int get_coeff(int mclk, int rate) -+{ -+ int i; -+ -+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { -+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) -+ return i; -+ } -+ return 0; -+} -+ -+/* WM8731 supports numerous clocks per sample rate */ -+static unsigned int wm8731_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ dai->mclk = 0; -+ -+ /* check that the calculated FS and rate actually match a clock from -+ * the machine driver */ -+ if (info->fs * info->rate == clk) -+ dai->mclk = clk; -+ -+ return dai->mclk; -+} -+ -+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 iface = 0, srate; -+ int i = get_coeff(rtd->codec_dai->mclk, -+ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate)); -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ iface |= 0x0040; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ srate = (coeff_div[i].sr << 2) | -+ (coeff_div[i].bosr << 1) | coeff_div[i].usb; -+ wm8731_write(codec, WM8731_SRATE, srate); -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ iface |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ iface |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ iface |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ iface |= 0x0013; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ iface |= 0x0004; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ iface |= 0x0008; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ iface |= 0x000c; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_IF: -+ iface |= 0x0090; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ iface |= 0x0080; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ iface |= 0x0010; -+ break; -+ } -+ -+ /* set iface */ -+ wm8731_write(codec, WM8731_IFACE, iface); -+ -+ /* set active */ -+ wm8731_write(codec, WM8731_ACTIVE, 0x0001); -+ return 0; -+} -+ -+static void wm8731_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ /* deactivate */ -+ if (!codec->active) { -+ udelay(50); -+ wm8731_write(codec, WM8731_ACTIVE, 0x0); -+ } -+} -+ -+static int wm8731_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; -+ if (mute) -+ wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8); -+ else -+ wm8731_write(codec, WM8731_APDIGI, mute_reg); -+ return 0; -+} -+ -+static int wm8731_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, osc on, dac unmute */ -+ wm8731_write(codec, WM8731_PWR, reg); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, */ -+ wm8731_write(codec, WM8731_PWR, reg | 0x0040); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, dac mute, inactive */ -+ wm8731_write(codec, WM8731_ACTIVE, 0x0); -+ wm8731_write(codec, WM8731_PWR, 0xffff); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8731_dai = { -+ .name = "WM8731", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = wm8731_config_sysclk, -+ .digital_mute = wm8731_mute, -+ .ops = { -+ .prepare = wm8731_pcm_prepare, -+ .shutdown = wm8731_shutdown, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8731_modes), -+ .mode = wm8731_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8731_dai); -+ -+static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8731_write(codec, WM8731_ACTIVE, 0x0); -+ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8731_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8731_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the WM8731 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8731_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int reg, ret = 0; -+ -+ codec->name = "WM8731"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8731_read_reg_cache; -+ codec->write = wm8731_write; -+ codec->dapm_event = wm8731_dapm_event; -+ codec->dai = &wm8731_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); -+ -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8731_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, -+ wm8731_reg, sizeof(u16) * ARRAY_SIZE(wm8731_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8731_reg); -+ -+ wm8731_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if (ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ /* set the update bits */ -+ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); -+ wm8731_write(codec, WM8731_LOUT1V, reg | 0x0100); -+ reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); -+ wm8731_write(codec, WM8731_ROUT1V, reg | 0x0100); -+ reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); -+ wm8731_write(codec, WM8731_LINVOL, reg | 0x0100); -+ reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); -+ wm8731_write(codec, WM8731_RINVOL, reg | 0x0100); -+ -+ wm8731_add_controls(codec); -+ wm8731_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *wm8731_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8731 2 wire address is determined by GPIO5 -+ * state during powerup. -+ * low = 0x1a -+ * high = 0x1b -+ */ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8731_i2c_driver; -+static struct i2c_client client_template; -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+ -+static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8731_socdev; -+ struct wm8731_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL) { -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ i2c_set_clientdata(i2c, codec); -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if (ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8731_init(socdev); -+ if (ret < 0) { -+ err("failed to initialise WM8731\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+} -+ -+static int wm8731_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec* codec = i2c_get_clientdata(client); -+ i2c_detach_client(client); -+ kfree(codec->reg_cache); -+ kfree(client); -+ return 0; -+} -+ -+static int wm8731_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8731_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8731_i2c_driver = { -+ .driver = { -+ .name = "WM8731 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8731, -+ .attach_adapter = wm8731_i2c_attach, -+ .detach_client = wm8731_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8731", -+ .driver = &wm8731_i2c_driver, -+}; -+#endif -+ -+static int wm8731_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8731_setup_data *setup; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ info("WM8731 Audio Codec %s", WM8731_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ wm8731_socdev = socdev; -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8731_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8731_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8731_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8731 = { -+ .probe = wm8731_probe, -+ .remove = wm8731_remove, -+ .suspend = wm8731_suspend, -+ .resume = wm8731_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); -+ -+MODULE_DESCRIPTION("ASoC WM8731 driver"); -+MODULE_AUTHOR("Richard Purdie"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8731.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8731.h -@@ -0,0 +1,41 @@ -+/* -+ * wm8731.h -- WM8731 Soc Audio driver -+ * -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Author: Richard Purdie <richard@openedhand.com> -+ * -+ * Based on wm8753.h -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _WM8731_H -+#define _WM8731_H -+ -+/* WM8731 register space */ -+ -+#define WM8731_LINVOL 0x00 -+#define WM8731_RINVOL 0x01 -+#define WM8731_LOUT1V 0x02 -+#define WM8731_ROUT1V 0x03 -+#define WM8731_APANA 0x04 -+#define WM8731_APDIGI 0x05 -+#define WM8731_PWR 0x06 -+#define WM8731_IFACE 0x07 -+#define WM8731_SRATE 0x08 -+#define WM8731_ACTIVE 0x09 -+#define WM8731_RESET 0x0f -+ -+#define WM8731_CACHEREGNUM 10 -+ -+struct wm8731_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai wm8731_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8731; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8750.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8750.c -@@ -0,0 +1,1282 @@ -+/* -+ * wm8750.c -- WM8750 ALSA SoC audio driver -+ * -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Author: Richard Purdie <richard@openedhand.com> -+ * -+ * Based on WM8753.c -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8750.h" -+ -+#define AUDIO_NAME "WM8750" -+#define WM8750_VERSION "0.11" -+ -+/* -+ * Debug -+ */ -+ -+#define WM8750_DEBUG 0 -+ -+#ifdef WM8750_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+static struct workqueue_struct *wm8750_workq = NULL; -+static struct work_struct wm8750_dapm_work; -+ -+/* -+ * wm8750 register cache -+ * We can't read the WM8750 register space when we -+ * are using 2 wire for device control, so we cache them instead. -+ */ -+static const u16 wm8750_reg[] = { -+ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ -+ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ -+ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ -+ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ -+ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ -+ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ -+ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ -+ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ -+ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ -+ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ -+ 0x0079, 0x0079, 0x0079, /* 40 */ -+}; -+ -+#define WM8750_HIFI_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ -+ SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8750_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8750_HIFI_FSB \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ -+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) -+ -+#define WM8750_HIFI_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -+ -+#define WM8750_HIFI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+static struct snd_soc_dai_mode wm8750_modes[] = { -+ /* common codec frame and clock master modes */ -+ /* 8k */ -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1536, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1408, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 2304, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 2112, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1500, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ -+ /* 11.025k */ -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1024, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1536, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1088, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ -+ /* 16k */ -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 768, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1152, -+ .bfs = WM8750_HIFI_FSB -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 750, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ -+ /* 22.05k */ -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 512, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 768, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 544, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ -+ /* 32k */ -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 384, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 576, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 375, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ -+ /* 44.1k & 48k */ -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 384, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 272, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 250, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ -+ /* 88.2k & 96k */ -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 128, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 192, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 136, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 125, -+ .bfs = WM8750_HIFI_FSB, -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8750_HIFI_BITS, -+ .pcmrate = WM8750_HIFI_RATES, -+ .pcmdir = WM8750_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8750 register cache -+ */ -+static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg > WM8750_CACHE_REGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write wm8750 register cache -+ */ -+static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg > WM8750_CACHE_REGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8753 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8750_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -EIO; -+} -+ -+#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0) -+ -+/* -+ * WM8750 Controls -+ */ -+static const char *wm8750_bass[] = {"Linear Control", "Adaptive Boost"}; -+static const char *wm8750_bass_filter[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" }; -+static const char *wm8750_treble[] = {"8kHz", "4kHz"}; -+static const char *wm8750_3d_lc[] = {"200Hz", "500Hz"}; -+static const char *wm8750_3d_uc[] = {"2.2kHz", "1.5kHz"}; -+static const char *wm8750_3d_func[] = {"Capture", "Playback"}; -+static const char *wm8750_alc_func[] = {"Off", "Right", "Left", "Stereo"}; -+static const char *wm8750_ng_type[] = {"Constant PGA Gain", -+ "Mute ADC Output"}; -+static const char *wm8750_line_mux[] = {"Line 1", "Line 2", "Line 3", "PGA", -+ "Differential"}; -+static const char *wm8750_pga_sel[] = {"Line 1", "Line 2", "Line 3", -+ "Differential"}; -+static const char *wm8750_out3[] = {"VREF", "ROUT1 + Vol", "MonoOut", -+ "ROUT1"}; -+static const char *wm8750_diff_sel[] = {"Line 1", "Line 2"}; -+static const char *wm8750_adcpol[] = {"Normal", "L Invert", "R Invert", -+ "L + R Invert"}; -+static const char *wm8750_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -+static const char *wm8750_mono_mux[] = {"Stereo", "Mono (Left)", -+ "Mono (Right)", "Digital Mono"}; -+ -+static const struct soc_enum wm8750_enum[] = { -+SOC_ENUM_SINGLE(WM8750_BASS, 7, 2, wm8750_bass), -+SOC_ENUM_SINGLE(WM8750_BASS, 6, 2, wm8750_bass_filter), -+SOC_ENUM_SINGLE(WM8750_TREBLE, 6, 2, wm8750_treble), -+SOC_ENUM_SINGLE(WM8750_3D, 5, 2, wm8750_3d_lc), -+SOC_ENUM_SINGLE(WM8750_3D, 6, 2, wm8750_3d_uc), -+SOC_ENUM_SINGLE(WM8750_3D, 7, 2, wm8750_3d_func), -+SOC_ENUM_SINGLE(WM8750_ALC1, 7, 4, wm8750_alc_func), -+SOC_ENUM_SINGLE(WM8750_NGATE, 1, 2, wm8750_ng_type), -+SOC_ENUM_SINGLE(WM8750_LOUTM1, 0, 5, wm8750_line_mux), -+SOC_ENUM_SINGLE(WM8750_ROUTM1, 0, 5, wm8750_line_mux), -+SOC_ENUM_SINGLE(WM8750_LADCIN, 6, 4, wm8750_pga_sel), /* 10 */ -+SOC_ENUM_SINGLE(WM8750_RADCIN, 6, 4, wm8750_pga_sel), -+SOC_ENUM_SINGLE(WM8750_ADCTL2, 7, 4, wm8750_out3), -+SOC_ENUM_SINGLE(WM8750_ADCIN, 8, 2, wm8750_diff_sel), -+SOC_ENUM_SINGLE(WM8750_ADCDAC, 5, 4, wm8750_adcpol), -+SOC_ENUM_SINGLE(WM8750_ADCDAC, 1, 4, wm8750_deemph), -+SOC_ENUM_SINGLE(WM8750_ADCIN, 6, 4, wm8750_mono_mux), /* 16 */ -+ -+}; -+ -+static const struct snd_kcontrol_new wm8750_snd_controls[] = { -+ -+SOC_DOUBLE_R("Capture Volume", WM8750_LINVOL, WM8750_RINVOL, 0, 63, 0), -+SOC_DOUBLE_R("Capture ZC Switch", WM8750_LINVOL, WM8750_RINVOL, 6, 1, 0), -+SOC_DOUBLE_R("Capture Switch", WM8750_LINVOL, WM8750_RINVOL, 7, 1, 1), -+ -+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8750_LOUT1V, -+ WM8750_ROUT1V, 7, 1, 0), -+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8750_LOUT2V, -+ WM8750_ROUT2V, 7, 1, 0), -+ -+SOC_ENUM("Playback De-emphasis", wm8750_enum[15]), -+ -+SOC_ENUM("Capture Polarity", wm8750_enum[14]), -+SOC_SINGLE("Playback 6dB Attenuate", WM8750_ADCDAC, 7, 1, 0), -+SOC_SINGLE("Capture 6dB Attenuate", WM8750_ADCDAC, 8, 1, 0), -+ -+SOC_DOUBLE_R("PCM Volume", WM8750_LDAC, WM8750_RDAC, 0, 255, 0), -+ -+SOC_ENUM("Bass Boost", wm8750_enum[0]), -+SOC_ENUM("Bass Filter", wm8750_enum[1]), -+SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1), -+ -+SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 0), -+SOC_ENUM("Treble Cut-off", wm8750_enum[2]), -+ -+SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0), -+SOC_SINGLE("3D Volume", WM8750_3D, 1, 15, 0), -+SOC_ENUM("3D Lower Cut-off", wm8750_enum[3]), -+SOC_ENUM("3D Upper Cut-off", wm8750_enum[4]), -+SOC_ENUM("3D Mode", wm8750_enum[5]), -+ -+SOC_SINGLE("ALC Capture Target Volume", WM8750_ALC1, 0, 7, 0), -+SOC_SINGLE("ALC Capture Max Volume", WM8750_ALC1, 4, 7, 0), -+SOC_ENUM("ALC Capture Function", wm8750_enum[6]), -+SOC_SINGLE("ALC Capture ZC Switch", WM8750_ALC2, 7, 1, 0), -+SOC_SINGLE("ALC Capture Hold Time", WM8750_ALC2, 0, 15, 0), -+SOC_SINGLE("ALC Capture Decay Time", WM8750_ALC3, 4, 15, 0), -+SOC_SINGLE("ALC Capture Attack Time", WM8750_ALC3, 0, 15, 0), -+SOC_SINGLE("ALC Capture NG Threshold", WM8750_NGATE, 3, 31, 0), -+SOC_ENUM("ALC Capture NG Type", wm8750_enum[4]), -+SOC_SINGLE("ALC Capture NG Switch", WM8750_NGATE, 0, 1, 0), -+ -+SOC_SINGLE("Left ADC Capture Volume", WM8750_LADC, 0, 255, 0), -+SOC_SINGLE("Right ADC Capture Volume", WM8750_RADC, 0, 255, 0), -+ -+SOC_SINGLE("ZC Timeout Switch", WM8750_ADCTL1, 0, 1, 0), -+SOC_SINGLE("Playback Invert Switch", WM8750_ADCTL1, 1, 1, 0), -+ -+SOC_SINGLE("Right Speaker Playback Invert Switch", WM8750_ADCTL2, 4, 1, 0), -+ -+/* Unimplemented */ -+/* ADCDAC Bit 0 - ADCHPD */ -+/* ADCDAC Bit 4 - HPOR */ -+/* ADCTL1 Bit 2,3 - DATSEL */ -+/* ADCTL1 Bit 4,5 - DMONOMIX */ -+/* ADCTL1 Bit 6,7 - VSEL */ -+/* ADCTL2 Bit 2 - LRCM */ -+/* ADCTL2 Bit 3 - TRI */ -+/* ADCTL3 Bit 5 - HPFLREN */ -+/* ADCTL3 Bit 6 - VROI */ -+/* ADCTL3 Bit 7,8 - ADCLRM */ -+/* ADCIN Bit 4 - LDCM */ -+/* ADCIN Bit 5 - RDCM */ -+ -+SOC_DOUBLE_R("Mic Boost", WM8750_LADCIN, WM8750_RADCIN, 4, 3, 0), -+ -+SOC_DOUBLE_R("Bypass Left Playback Volume", WM8750_LOUTM1, -+ WM8750_LOUTM2, 4, 7, 1), -+SOC_DOUBLE_R("Bypass Right Playback Volume", WM8750_ROUTM1, -+ WM8750_ROUTM2, 4, 7, 1), -+SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8750_MOUTM1, -+ WM8750_MOUTM2, 4, 7, 1), -+ -+SOC_SINGLE("Mono Playback ZC Switch", WM8750_MOUTV, 7, 1, 0), -+ -+SOC_DOUBLE_R("Headphone Playback Volume", WM8750_LOUT1V, WM8750_ROUT1V, -+ 0, 127, 0), -+SOC_DOUBLE_R("Speaker Playback Volume", WM8750_LOUT2V, WM8750_ROUT2V, -+ 0, 127, 0), -+ -+SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0), -+ -+}; -+ -+/* add non dapm controls */ -+static int wm8750_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8750_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ return 0; -+} -+ -+/* -+ * DAPM Controls -+ */ -+ -+/* Left Mixer */ -+static const struct snd_kcontrol_new wm8750_left_mixer_controls[] = { -+SOC_DAPM_SINGLE("Playback Switch", WM8750_LOUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_LOUTM1, 7, 1, 0), -+SOC_DAPM_SINGLE("Right Playback Switch", WM8750_LOUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_LOUTM2, 7, 1, 0), -+}; -+ -+/* Right Mixer */ -+static const struct snd_kcontrol_new wm8750_right_mixer_controls[] = { -+SOC_DAPM_SINGLE("Left Playback Switch", WM8750_ROUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_ROUTM1, 7, 1, 0), -+SOC_DAPM_SINGLE("Playback Switch", WM8750_ROUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_ROUTM2, 7, 1, 0), -+}; -+ -+/* Mono Mixer */ -+static const struct snd_kcontrol_new wm8750_mono_mixer_controls[] = { -+SOC_DAPM_SINGLE("Left Playback Switch", WM8750_MOUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_MOUTM1, 7, 1, 0), -+SOC_DAPM_SINGLE("Right Playback Switch", WM8750_MOUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_MOUTM2, 7, 1, 0), -+}; -+ -+/* Left Line Mux */ -+static const struct snd_kcontrol_new wm8750_left_line_controls = -+SOC_DAPM_ENUM("Route", wm8750_enum[8]); -+ -+/* Right Line Mux */ -+static const struct snd_kcontrol_new wm8750_right_line_controls = -+SOC_DAPM_ENUM("Route", wm8750_enum[9]); -+ -+/* Left PGA Mux */ -+static const struct snd_kcontrol_new wm8750_left_pga_controls = -+SOC_DAPM_ENUM("Route", wm8750_enum[10]); -+ -+/* Right PGA Mux */ -+static const struct snd_kcontrol_new wm8750_right_pga_controls = -+SOC_DAPM_ENUM("Route", wm8750_enum[11]); -+ -+/* Out 3 Mux */ -+static const struct snd_kcontrol_new wm8750_out3_controls = -+SOC_DAPM_ENUM("Route", wm8750_enum[12]); -+ -+/* Differential Mux */ -+static const struct snd_kcontrol_new wm8750_diffmux_controls = -+SOC_DAPM_ENUM("Route", wm8750_enum[13]); -+ -+/* Mono ADC Mux */ -+static const struct snd_kcontrol_new wm8750_monomux_controls = -+SOC_DAPM_ENUM("Route", wm8750_enum[16]); -+ -+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { -+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, -+ &wm8750_left_mixer_controls[0], -+ ARRAY_SIZE(wm8750_left_mixer_controls)), -+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, -+ &wm8750_right_mixer_controls[0], -+ ARRAY_SIZE(wm8750_right_mixer_controls)), -+ SND_SOC_DAPM_MIXER("Mono Mixer", WM8750_PWR2, 2, 0, -+ &wm8750_mono_mixer_controls[0], -+ ARRAY_SIZE(wm8750_mono_mixer_controls)), -+ -+ SND_SOC_DAPM_PGA("Right Out 2", WM8750_PWR2, 3, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Left Out 2", WM8750_PWR2, 4, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Right Out 1", WM8750_PWR2, 5, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Left Out 1", WM8750_PWR2, 6, 0, NULL, 0), -+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8750_PWR2, 7, 0), -+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8750_PWR2, 8, 0), -+ -+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8750_PWR1, 1, 0), -+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8750_PWR1, 2, 0), -+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8750_PWR1, 3, 0), -+ -+ SND_SOC_DAPM_MUX("Left PGA Mux", WM8750_PWR1, 5, 0, -+ &wm8750_left_pga_controls), -+ SND_SOC_DAPM_MUX("Right PGA Mux", WM8750_PWR1, 4, 0, -+ &wm8750_right_pga_controls), -+ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, -+ &wm8750_left_line_controls), -+ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, -+ &wm8750_right_line_controls), -+ -+ SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8750_out3_controls), -+ SND_SOC_DAPM_PGA("Out 3", WM8750_PWR2, 1, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Mono Out 1", WM8750_PWR2, 2, 0, NULL, 0), -+ -+ SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, -+ &wm8750_diffmux_controls), -+ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, -+ &wm8750_monomux_controls), -+ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, -+ &wm8750_monomux_controls), -+ -+ SND_SOC_DAPM_OUTPUT("LOUT1"), -+ SND_SOC_DAPM_OUTPUT("ROUT1"), -+ SND_SOC_DAPM_OUTPUT("LOUT2"), -+ SND_SOC_DAPM_OUTPUT("ROUT2"), -+ SND_SOC_DAPM_OUTPUT("MONO"), -+ SND_SOC_DAPM_OUTPUT("OUT3"), -+ -+ SND_SOC_DAPM_INPUT("LINPUT1"), -+ SND_SOC_DAPM_INPUT("LINPUT2"), -+ SND_SOC_DAPM_INPUT("LINPUT3"), -+ SND_SOC_DAPM_INPUT("RINPUT1"), -+ SND_SOC_DAPM_INPUT("RINPUT2"), -+ SND_SOC_DAPM_INPUT("RINPUT3"), -+}; -+ -+static const char *audio_map[][3] = { -+ /* left mixer */ -+ {"Left Mixer", "Playback Switch", "Left DAC"}, -+ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, -+ {"Left Mixer", "Right Playback Switch", "Right DAC"}, -+ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"}, -+ -+ /* right mixer */ -+ {"Right Mixer", "Left Playback Switch", "Left DAC"}, -+ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"}, -+ {"Right Mixer", "Playback Switch", "Right DAC"}, -+ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"}, -+ -+ /* left out 1 */ -+ {"Left Out 1", NULL, "Left Mixer"}, -+ {"LOUT1", NULL, "Left Out 1"}, -+ -+ /* left out 2 */ -+ {"Left Out 2", NULL, "Left Mixer"}, -+ {"LOUT2", NULL, "Left Out 2"}, -+ -+ /* right out 1 */ -+ {"Right Out 1", NULL, "Right Mixer"}, -+ {"ROUT1", NULL, "Right Out 1"}, -+ -+ /* right out 2 */ -+ {"Right Out 2", NULL, "Right Mixer"}, -+ {"ROUT2", NULL, "Right Out 2"}, -+ -+ /* mono mixer */ -+ {"Mono Mixer", "Left Playback Switch", "Left DAC"}, -+ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"}, -+ {"Mono Mixer", "Right Playback Switch", "Right DAC"}, -+ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"}, -+ -+ /* mono out */ -+ {"Mono Out 1", NULL, "Mono Mixer"}, -+ {"MONO1", NULL, "Mono Out 1"}, -+ -+ /* out 3 */ -+ {"Out3 Mux", "VREF", "VREF"}, -+ {"Out3 Mux", "ROUT1 + Vol", "ROUT1"}, -+ {"Out3 Mux", "ROUT1", "Right Mixer"}, -+ {"Out3 Mux", "MonoOut", "MONO1"}, -+ {"Out 3", NULL, "Out3 Mux"}, -+ {"OUT3", NULL, "Out 3"}, -+ -+ /* Left Line Mux */ -+ {"Left Line Mux", "Line 1", "LINPUT1"}, -+ {"Left Line Mux", "Line 2", "LINPUT2"}, -+ {"Left Line Mux", "Line 3", "LINPUT3"}, -+ {"Left Line Mux", "PGA", "Left PGA Mux"}, -+ {"Left Line Mux", "Differential", "Differential Mux"}, -+ -+ /* Right Line Mux */ -+ {"Right Line Mux", "Line 1", "RINPUT1"}, -+ {"Right Line Mux", "Line 2", "RINPUT2"}, -+ {"Right Line Mux", "Line 3", "RINPUT3"}, -+ {"Right Line Mux", "PGA", "Right PGA Mux"}, -+ {"Right Line Mux", "Differential", "Differential Mux"}, -+ -+ /* Left PGA Mux */ -+ {"Left PGA Mux", "Line 1", "LINPUT1"}, -+ {"Left PGA Mux", "Line 2", "LINPUT2"}, -+ {"Left PGA Mux", "Line 3", "LINPUT3"}, -+ {"Left PGA Mux", "Differential", "Differential Mux"}, -+ -+ /* Right PGA Mux */ -+ {"Right PGA Mux", "Line 1", "RINPUT1"}, -+ {"Right PGA Mux", "Line 2", "RINPUT2"}, -+ {"Right PGA Mux", "Line 3", "RINPUT3"}, -+ {"Right PGA Mux", "Differential", "Differential Mux"}, -+ -+ /* Differential Mux */ -+ {"Differential Mux", "Line 1", "LINPUT1"}, -+ {"Differential Mux", "Line 1", "RINPUT1"}, -+ {"Differential Mux", "Line 2", "LINPUT2"}, -+ {"Differential Mux", "Line 2", "RINPUT2"}, -+ -+ /* Left ADC Mux */ -+ {"Left ADC Mux", "Stereo", "Left PGA Mux"}, -+ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"}, -+ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"}, -+ -+ /* Right ADC Mux */ -+ {"Right ADC Mux", "Stereo", "Right PGA Mux"}, -+ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"}, -+ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"}, -+ -+ /* ADC */ -+ {"Left ADC", NULL, "Left ADC Mux"}, -+ {"Right ADC", NULL, "Right ADC Mux"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8750_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); -+ } -+ -+ /* set up audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+struct _coeff_div { -+ u32 mclk; -+ u32 rate; -+ u16 fs; -+ u8 sr:5; -+ u8 usb:1; -+}; -+ -+/* codec hifi mclk clock divider coefficients */ -+static const struct _coeff_div coeff_div[] = { -+ /* 8k */ -+ {12288000, 8000, 1536, 0x6, 0x0}, -+ {11289600, 8000, 1408, 0x16, 0x0}, -+ {18432000, 8000, 2304, 0x7, 0x0}, -+ {16934400, 8000, 2112, 0x17, 0x0}, -+ {12000000, 8000, 1500, 0x6, 0x1}, -+ -+ /* 11.025k */ -+ {11289600, 11025, 1024, 0x18, 0x0}, -+ {16934400, 11025, 1536, 0x19, 0x0}, -+ {12000000, 11025, 1088, 0x19, 0x1}, -+ -+ /* 16k */ -+ {12288000, 16000, 768, 0xa, 0x0}, -+ {18432000, 16000, 1152, 0xb, 0x0}, -+ {12000000, 16000, 750, 0xa, 0x1}, -+ -+ /* 22.05k */ -+ {11289600, 22050, 512, 0x1a, 0x0}, -+ {16934400, 22050, 768, 0x1b, 0x0}, -+ {12000000, 22050, 544, 0x1b, 0x1}, -+ -+ /* 32k */ -+ {12288000, 32000, 384, 0xc, 0x0}, -+ {18432000, 32000, 576, 0xd, 0x0}, -+ {12000000, 32000, 375, 0xa, 0x1}, -+ -+ /* 44.1k */ -+ {11289600, 44100, 256, 0x10, 0x0}, -+ {16934400, 44100, 384, 0x11, 0x0}, -+ {12000000, 44100, 272, 0x11, 0x1}, -+ -+ /* 48k */ -+ {12288000, 48000, 256, 0x0, 0x0}, -+ {18432000, 48000, 384, 0x1, 0x0}, -+ {12000000, 48000, 250, 0x0, 0x1}, -+ -+ /* 88.2k */ -+ {11289600, 88200, 128, 0x1e, 0x0}, -+ {16934400, 88200, 192, 0x1f, 0x0}, -+ {12000000, 88200, 136, 0x1f, 0x1}, -+ -+ /* 96k */ -+ {12288000, 96000, 128, 0xe, 0x0}, -+ {18432000, 96000, 192, 0xf, 0x0}, -+ {12000000, 96000, 125, 0xe, 0x1}, -+}; -+ -+static inline int get_coeff(int mclk, int rate) -+{ -+ int i; -+ -+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { -+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) -+ return i; -+ } -+ -+ printk(KERN_ERR "wm8750: could not get coeff for mclk %d @ rate %d\n", -+ mclk, rate); -+ return -EINVAL; -+} -+ -+/* WM8750 supports numerous input clocks per sample rate */ -+static unsigned int wm8750_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ dai->mclk = clk; -+ return dai->mclk; -+} -+ -+static int wm8750_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 iface = 0, bfs, srate = 0; -+ int i = get_coeff(rtd->codec_dai->mclk, -+ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate)); -+ -+ /* is coefficient valid ? */ -+ if (i < 0) -+ return i; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ iface = 0x0040; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ iface |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ iface |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ iface |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ iface |= 0x0013; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ iface |= 0x0004; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ iface |= 0x0008; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ iface |= 0x000c; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_IF: -+ iface |= 0x0090; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ iface |= 0x0080; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ iface |= 0x0010; -+ break; -+ } -+ -+ /* set bclk divisor rate */ -+ switch (bfs) { -+ case 1: -+ break; -+ case 4: -+ srate |= (0x1 << 7); -+ break; -+ case 8: -+ srate |= (0x2 << 7); -+ break; -+ case 16: -+ srate |= (0x3 << 7); -+ break; -+ } -+ -+ /* set iface & srate */ -+ wm8750_write(codec, WM8750_IFACE, iface); -+ wm8750_write(codec, WM8750_SRATE, srate | -+ (coeff_div[i].sr << 1) | coeff_div[i].usb); -+ -+ return 0; -+} -+ -+static int wm8750_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; -+ if (mute) -+ wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8); -+ else -+ wm8750_write(codec, WM8750_ADCDAC, mute_reg); -+ return 0; -+} -+ -+static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* set vmid to 50k and unmute dac */ -+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ /* set vmid to 5k for quick power up */ -+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* mute dac and set vmid to 500k, enable VREF */ -+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ wm8750_write(codec, WM8750_PWR1, 0x0001); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8750_dai = { -+ .name = "WM8750", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = wm8750_config_sysclk, -+ .digital_mute = wm8750_mute, -+ .ops = { -+ .prepare = wm8750_pcm_prepare, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8750_modes), -+ .mode = wm8750_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8750_dai); -+ -+static void wm8750_work(void *data) -+{ -+ struct snd_soc_codec *codec = (struct snd_soc_codec *)data; -+ wm8750_dapm_event(codec, codec->dapm_state); -+} -+ -+static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8750_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) { -+ if (i == WM8750_RESET) -+ continue; -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ -+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ /* charge wm8750 caps */ -+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { -+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); -+ codec->dapm_state = SNDRV_CTL_POWER_D0; -+ queue_delayed_work(wm8750_workq, &wm8750_dapm_work, -+ msecs_to_jiffies(1000)); -+ } -+ -+ return 0; -+} -+ -+/* -+ * initialise the WM8750 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8750_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int reg, ret = 0; -+ -+ codec->name = "WM8750"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8750_read_reg_cache; -+ codec->write = wm8750_write; -+ codec->dapm_event = wm8750_dapm_event; -+ codec->dai = &wm8750_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); -+ -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8750_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8750_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8750_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8750_reg); -+ -+ wm8750_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if (ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* charge output caps */ -+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); -+ codec->dapm_state = SNDRV_CTL_POWER_D3hot; -+ queue_delayed_work(wm8750_workq, &wm8750_dapm_work, -+ msecs_to_jiffies(1000)); -+ -+ /* set the update bits */ -+ reg = wm8750_read_reg_cache(codec, WM8750_LDAC); -+ wm8750_write(codec, WM8750_LDAC, reg | 0x0100); -+ reg = wm8750_read_reg_cache(codec, WM8750_RDAC); -+ wm8750_write(codec, WM8750_RDAC, reg | 0x0100); -+ reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V); -+ wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100); -+ reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V); -+ wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100); -+ reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V); -+ wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100); -+ reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V); -+ wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100); -+ reg = wm8750_read_reg_cache(codec, WM8750_LINVOL); -+ wm8750_write(codec, WM8750_LINVOL, reg | 0x0100); -+ reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); -+ wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); -+ -+ wm8750_add_controls(codec); -+ wm8750_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+static struct snd_soc_device *wm8750_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8731 2 wire address is determined by GPIO5 -+ * state during powerup. -+ * low = 0x1a -+ * high = 0x1b -+ */ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8750_i2c_driver; -+static struct i2c_client client_template; -+ -+static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8750_socdev; -+ struct wm8750_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL) { -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ i2c_set_clientdata(i2c, codec); -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if (ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8750_init(socdev); -+ if (ret < 0) { -+ err("failed to initialise WM8750\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+} -+ -+static int wm8750_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec *codec = i2c_get_clientdata(client); -+ i2c_detach_client(client); -+ kfree(codec->reg_cache); -+ kfree(client); -+ return 0; -+} -+ -+static int wm8750_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8750_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8750_i2c_driver = { -+ .driver = { -+ .name = "WM8750 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8750, -+ .attach_adapter = wm8750_i2c_attach, -+ .detach_client = wm8750_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8750", -+ .driver = &wm8750_i2c_driver, -+}; -+#endif -+ -+static int wm8750_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8750_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ info("WM8750 Audio Codec %s", WM8750_VERSION); -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ wm8750_socdev = socdev; -+ INIT_WORK(&wm8750_dapm_work, wm8750_work, codec); -+ wm8750_workq = create_workqueue("wm8750"); -+ if (wm8750_workq == NULL) { -+ kfree(codec); -+ return -ENOMEM; -+ } -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8750_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8750_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ if (wm8750_workq) -+ destroy_workqueue(wm8750_workq); -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8750_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8750 = { -+ .probe = wm8750_probe, -+ .remove = wm8750_remove, -+ .suspend = wm8750_suspend, -+ .resume = wm8750_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); -+ -+MODULE_DESCRIPTION("ASoC WM8750 driver"); -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8750.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8750.h -@@ -0,0 +1,66 @@ -+/* -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Author: Richard Purdie <richard@openedhand.com> -+ * -+ * Based on WM8753.h -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ * -+ */ -+ -+#ifndef _WM8750_H -+#define _WM8750_H -+ -+/* WM8750 register space */ -+ -+#define WM8750_LINVOL 0x00 -+#define WM8750_RINVOL 0x01 -+#define WM8750_LOUT1V 0x02 -+#define WM8750_ROUT1V 0x03 -+#define WM8750_ADCDAC 0x05 -+#define WM8750_IFACE 0x07 -+#define WM8750_SRATE 0x08 -+#define WM8750_LDAC 0x0a -+#define WM8750_RDAC 0x0b -+#define WM8750_BASS 0x0c -+#define WM8750_TREBLE 0x0d -+#define WM8750_RESET 0x0f -+#define WM8750_3D 0x10 -+#define WM8750_ALC1 0x11 -+#define WM8750_ALC2 0x12 -+#define WM8750_ALC3 0x13 -+#define WM8750_NGATE 0x14 -+#define WM8750_LADC 0x15 -+#define WM8750_RADC 0x16 -+#define WM8750_ADCTL1 0x17 -+#define WM8750_ADCTL2 0x18 -+#define WM8750_PWR1 0x19 -+#define WM8750_PWR2 0x1a -+#define WM8750_ADCTL3 0x1b -+#define WM8750_ADCIN 0x1f -+#define WM8750_LADCIN 0x20 -+#define WM8750_RADCIN 0x21 -+#define WM8750_LOUTM1 0x22 -+#define WM8750_LOUTM2 0x23 -+#define WM8750_ROUTM1 0x24 -+#define WM8750_ROUTM2 0x25 -+#define WM8750_MOUTM1 0x26 -+#define WM8750_MOUTM2 0x27 -+#define WM8750_LOUT2V 0x28 -+#define WM8750_ROUT2V 0x29 -+#define WM8750_MOUTV 0x2a -+ -+#define WM8750_CACHE_REGNUM 0x2a -+ -+struct wm8750_setup_data { -+ unsigned short i2c_address; -+ unsigned int mclk; -+}; -+ -+extern struct snd_soc_codec_dai wm8750_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8750; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8753.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8753.c -@@ -0,0 +1,2128 @@ -+/* -+ * wm8753.c -- WM8753 ALSA Soc Audio driver -+ * -+ * Copyright 2003 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Notes: -+ * The WM8753 is a low power, high quality stereo codec with integrated PCM -+ * codec designed for portable digital telephony applications. -+ * -+ * Dual DAI:- -+ * -+ * This driver support 2 DAI PCM's. This makes the default PCM available for -+ * HiFi audio (e.g. MP3, ogg) playback/capture and the other PCM available for -+ * voice. -+ * -+ * Please note that the voice PCM can be connected directly to a Bluetooth -+ * codec or GSM modem and thus cannot be read or written to, although it is -+ * available to be configured with snd_hw_params(), etc and kcontrols in the -+ * normal alsa manner. -+ * -+ * Fast DAI switching:- -+ * -+ * The driver can now fast switch between the DAI configurations via a -+ * an alsa kcontrol. This allows the PCM to remain open. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/version.h> -+#include <linux/kernel.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8753.h" -+ -+#define AUDIO_NAME "wm8753" -+#define WM8753_VERSION "0.16" -+ -+/* -+ * Debug -+ */ -+ -+#define WM8753_DEBUG 0 -+ -+#ifdef WM8753_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+static int caps_charge = 2000; -+module_param(caps_charge, int, 0); -+MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); -+ -+static struct workqueue_struct *wm8753_workq = NULL; -+static struct work_struct wm8753_dapm_work; -+static void wm8753_set_dai_mode(struct snd_soc_codec *codec, -+ unsigned int mode); -+ -+/* -+ * wm8753 register cache -+ * We can't read the WM8753 register space when we -+ * are using 2 wire for device control, so we cache them instead. -+ */ -+static const u16 wm8753_reg[] = { -+ 0x0008, 0x0000, 0x000a, 0x000a, -+ 0x0033, 0x0000, 0x0007, 0x00ff, -+ 0x00ff, 0x000f, 0x000f, 0x007b, -+ 0x0000, 0x0032, 0x0000, 0x00c3, -+ 0x00c3, 0x00c0, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0055, -+ 0x0005, 0x0050, 0x0055, 0x0050, -+ 0x0055, 0x0050, 0x0055, 0x0079, -+ 0x0079, 0x0079, 0x0079, 0x0079, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0097, 0x0097, 0x0000, 0x0004, -+ 0x0000, 0x0083, 0x0024, 0x01ba, -+ 0x0000, 0x0083, 0x0024, 0x01ba, -+ 0x0000, 0x0000 -+}; -+ -+#define WM8753_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \ -+ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8753_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8753_HIFI_FSB \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ -+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) -+ -+#define WM8753_HIFI_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -+ -+#define WM8753_HIFI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+/* -+ * HiFi modes -+ */ -+static struct snd_soc_dai_mode wm8753_hifi_modes[] = { -+ /* codec frame and clock master modes */ -+ /* 8k */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1536, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1408, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 2304, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 2112, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1500, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ -+ /* 11.025k */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1024, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1536, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1088, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ -+ /* 16k */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 768, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt= WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1152, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 750, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ -+ /* 22.05k */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 512, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 768, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 544, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ -+ /* 32k */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 384, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 576, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 375, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ -+ /* 44.1k & 48k */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 384, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 250, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 272, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ -+ /* 88.2k & 96k */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 128, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 192, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 136, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 125, -+ .bfs = WM8753_HIFI_FSB, -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = WM8753_HIFI_RATES, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+#define WM8753_VOICE_FSB \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ -+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) -+ -+#define WM8753_VOICE_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+#define WM8753_VOICE_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+/* -+ * Voice modes -+ */ -+static struct snd_soc_dai_mode wm8753_voice_modes[] = { -+ -+ /* master modes */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_VOICE_BITS, -+ .pcmrate = WM8753_VOICE_RATES, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = WM8753_VOICE_FSB, -+ }, -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM8753_VOICE_BITS, -+ .pcmrate = WM8753_VOICE_RATES, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 384, -+ .bfs = WM8753_VOICE_FSB, -+ }, -+ -+ /* slave modes */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8753_VOICE_BITS, -+ .pcmrate = WM8753_VOICE_RATES, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+ -+/* -+ * Mode 4 -+ */ -+static struct snd_soc_dai_mode wm8753_mixed_modes[] = { -+ /* slave modes */ -+ { -+ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8753_HIFI_BITS, -+ .pcmrate = WM8753_HIFI_RATES, -+ .pcmdir = WM8753_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8753 register cache -+ */ -+static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1)) -+ return -1; -+ return cache[reg - 1]; -+} -+ -+/* -+ * write wm8753 register cache -+ */ -+static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg < 1 || reg > 0x3f) -+ return; -+ cache[reg - 1] = value; -+} -+ -+/* -+ * write to the WM8753 register space -+ */ -+static int wm8753_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8753 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8753_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -EIO; -+} -+ -+#define wm8753_reset(c) wm8753_write(c, WM8753_RESET, 0) -+ -+/* -+ * WM8753 Controls -+ */ -+static const char *wm8753_base[] = {"Linear Control", "Adaptive Boost"}; -+static const char *wm8753_base_filter[] = -+ {"130Hz @ 48kHz", "200Hz @ 48kHz", "100Hz @ 16kHz", "400Hz @ 48kHz", -+ "100Hz @ 8kHz", "200Hz @ 8kHz"}; -+static const char *wm8753_treble[] = {"8kHz", "4kHz"}; -+static const char *wm8753_alc_func[] = {"Off", "Right", "Left", "Stereo"}; -+static const char *wm8753_ng_type[] = {"Constant PGA Gain", "Mute ADC Output"}; -+static const char *wm8753_3d_func[] = {"Capture", "Playback"}; -+static const char *wm8753_3d_uc[] = {"2.2kHz", "1.5kHz"}; -+static const char *wm8753_3d_lc[] = {"200Hz", "500Hz"}; -+static const char *wm8753_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"}; -+static const char *wm8753_mono_mix[] = {"Stereo", "Left", "Right", "Mono"}; -+static const char *wm8753_dac_phase[] = {"Non Inverted", "Inverted"}; -+static const char *wm8753_line_mix[] = {"Line 1 + 2", "Line 1 - 2", -+ "Line 1", "Line 2"}; -+static const char *wm8753_mono_mux[] = {"Line Mix", "Rx Mix"}; -+static const char *wm8753_right_mux[] = {"Line 2", "Rx Mix"}; -+static const char *wm8753_left_mux[] = {"Line 1", "Rx Mix"}; -+static const char *wm8753_rxmsel[] = {"RXP - RXN", "RXP + RXN", "RXP", "RXN"}; -+static const char *wm8753_sidetone_mux[] = {"Left PGA", "Mic 1", "Mic 2", -+ "Right PGA"}; -+static const char *wm8753_mono2_src[] = {"Inverted Mono 1", "Left", "Right", -+ "Left + Right"}; -+static const char *wm8753_out3[] = {"VREF", "ROUT2", "Left + Right"}; -+static const char *wm8753_out4[] = {"VREF", "Capture ST", "LOUT2"}; -+static const char *wm8753_radcsel[] = {"PGA", "Line or RXP-RXN", "Sidetone"}; -+static const char *wm8753_ladcsel[] = {"PGA", "Line or RXP-RXN", "Line"}; -+static const char *wm8753_mono_adc[] = {"Stereo", "Analogue Mix Left", -+ "Analogue Mix Right", "Digital Mono Mix"}; -+static const char *wm8753_adc_hp[] = {"3.4Hz @ 48kHz", "82Hz @ 16k", -+ "82Hz @ 8kHz", "170Hz @ 8kHz"}; -+static const char *wm8753_adc_filter[] = {"HiFi", "Voice"}; -+static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"}; -+static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"}; -+ -+static const struct soc_enum wm8753_enum[] = { -+SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), // 0 -+SOC_ENUM_SINGLE(WM8753_BASS, 4, 6, wm8753_base_filter), // 1 -+SOC_ENUM_SINGLE(WM8753_TREBLE, 6, 2, wm8753_treble), // 2 -+SOC_ENUM_SINGLE(WM8753_ALC1, 7, 4, wm8753_alc_func), // 3 -+SOC_ENUM_SINGLE(WM8753_NGATE, 1, 2, wm8753_ng_type), // 4 -+SOC_ENUM_SINGLE(WM8753_3D, 7, 2, wm8753_3d_func), // 5 -+SOC_ENUM_SINGLE(WM8753_3D, 6, 2, wm8753_3d_uc), // 6 -+SOC_ENUM_SINGLE(WM8753_3D, 5, 2, wm8753_3d_lc), // 7 -+SOC_ENUM_SINGLE(WM8753_DAC, 1, 4, wm8753_deemp), // 8 -+SOC_ENUM_SINGLE(WM8753_DAC, 4, 4, wm8753_mono_mix), // 9 -+SOC_ENUM_SINGLE(WM8753_DAC, 6, 2, wm8753_dac_phase), // 10 -+SOC_ENUM_SINGLE(WM8753_INCTL1, 3, 4, wm8753_line_mix), // 11 -+SOC_ENUM_SINGLE(WM8753_INCTL1, 2, 2, wm8753_mono_mux), // 12 -+SOC_ENUM_SINGLE(WM8753_INCTL1, 1, 2, wm8753_right_mux), // 13 -+SOC_ENUM_SINGLE(WM8753_INCTL1, 0, 2, wm8753_left_mux), // 14 -+SOC_ENUM_SINGLE(WM8753_INCTL2, 6, 4, wm8753_rxmsel), // 15 -+SOC_ENUM_SINGLE(WM8753_INCTL2, 4, 4, wm8753_sidetone_mux),// 16 -+SOC_ENUM_SINGLE(WM8753_OUTCTL, 7, 4, wm8753_mono2_src), // 17 -+SOC_ENUM_SINGLE(WM8753_OUTCTL, 0, 3, wm8753_out3), // 18 -+SOC_ENUM_SINGLE(WM8753_ADCTL2, 7, 3, wm8753_out4), // 19 -+SOC_ENUM_SINGLE(WM8753_ADCIN, 2, 3, wm8753_radcsel), // 20 -+SOC_ENUM_SINGLE(WM8753_ADCIN, 0, 3, wm8753_ladcsel), // 21 -+SOC_ENUM_SINGLE(WM8753_ADCIN, 4, 4, wm8753_mono_adc), // 22 -+SOC_ENUM_SINGLE(WM8753_ADC, 2, 4, wm8753_adc_hp), // 23 -+SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), // 24 -+SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), // 25 -+SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), // 26 -+}; -+ -+ -+static int wm8753_get_dai(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL); -+ -+ ucontrol->value.integer.value[0] = (mode & 0xc) >> 2; -+ return 0; -+} -+ -+static int wm8753_set_dai(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL); -+ -+ if (((mode &0xc) >> 2) == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ mode &= 0xfff3; -+ mode |= (ucontrol->value.integer.value[0] << 2); -+ -+ wm8753_write(codec, WM8753_IOCTL, mode); -+ wm8753_set_dai_mode(codec, ucontrol->value.integer.value[0]); -+ return 1; -+} -+ -+static const struct snd_kcontrol_new wm8753_snd_controls[] = { -+SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0), -+ -+SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 63, 0), -+SOC_DOUBLE_R("ADC Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 0), -+SOC_DOUBLE_R("ADC Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), -+ -+SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0), -+SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0), -+ -+SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0), -+ -+SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1), -+SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1), -+SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1), -+ -+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0), -+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0), -+ -+SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), -+SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), -+SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1), -+SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0), -+ -+SOC_ENUM("Bass Boost", wm8753_enum[0]), -+SOC_ENUM("Bass Filter", wm8753_enum[1]), -+SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 7, 1), -+ -+SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 7, 0), -+SOC_ENUM("Treble Cut-off", wm8753_enum[2]), -+ -+SOC_DOUBLE("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1), -+SOC_SINGLE("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1), -+ -+SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0), -+SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), -+SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 0), -+ -+SOC_ENUM("Capture Filter Select", wm8753_enum[23]), -+SOC_ENUM("Capture Filter Cut-off", wm8753_enum[24]), -+SOC_SINGLE("Capture Filter Switch", WM8753_ADC, 0, 1, 1), -+ -+SOC_SINGLE("ALC Capture Target Volume", WM8753_ALC1, 0, 7, 0), -+SOC_SINGLE("ALC Capture Max Volume", WM8753_ALC1, 4, 7, 0), -+SOC_ENUM("ALC Capture Function", wm8753_enum[3]), -+SOC_SINGLE("ALC Capture ZC Switch", WM8753_ALC2, 8, 1, 0), -+SOC_SINGLE("ALC Capture Hold Time", WM8753_ALC2, 0, 15, 1), -+SOC_SINGLE("ALC Capture Decay Time", WM8753_ALC3, 4, 15, 1), -+SOC_SINGLE("ALC Capture Attack Time", WM8753_ALC3, 0, 15, 0), -+SOC_SINGLE("ALC Capture NG Threshold", WM8753_NGATE, 3, 31, 0), -+SOC_ENUM("ALC Capture NG Type", wm8753_enum[4]), -+SOC_SINGLE("ALC Capture NG Switch", WM8753_NGATE, 0, 1, 0), -+ -+SOC_ENUM("3D Function", wm8753_enum[5]), -+SOC_ENUM("3D Upper Cut-off", wm8753_enum[6]), -+SOC_ENUM("3D Lower Cut-off", wm8753_enum[7]), -+SOC_SINGLE("3D Volume", WM8753_3D, 1, 15, 0), -+SOC_SINGLE("3D Switch", WM8753_3D, 0, 1, 0), -+ -+SOC_SINGLE("Capture 6dB Attenuate", WM8753_ADCTL1, 2, 1, 0), -+SOC_SINGLE("Playback 6dB Attenuate", WM8753_ADCTL1, 1, 1, 0), -+ -+SOC_ENUM("De-emphasis", wm8753_enum[8]), -+SOC_ENUM("Playback Mono Mix", wm8753_enum[9]), -+SOC_ENUM("Playback Phase", wm8753_enum[10]), -+ -+SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0), -+SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), -+ -+SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), -+}; -+ -+/* add non dapm controls */ -+static int wm8753_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8753_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ return 0; -+} -+ -+/* -+ * _DAPM_ Controls -+ */ -+ -+/* Left Mixer */ -+static const struct snd_kcontrol_new wm8753_left_mixer_controls[] = { -+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_LOUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_LOUTM2, 7, 1, 0), -+SOC_DAPM_SINGLE("Left Playback Switch", WM8753_LOUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_LOUTM1, 7, 1, 0), -+}; -+ -+/* Right mixer */ -+static const struct snd_kcontrol_new wm8753_right_mixer_controls[] = { -+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_ROUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_ROUTM2, 7, 1, 0), -+SOC_DAPM_SINGLE("Right Playback Switch", WM8753_ROUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_ROUTM1, 7, 1, 0), -+}; -+ -+/* Mono mixer */ -+static const struct snd_kcontrol_new wm8753_mono_mixer_controls[] = { -+SOC_DAPM_SINGLE("Left Playback Switch", WM8753_MOUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Right Playback Switch", WM8753_MOUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_MOUTM2, 3, 1, 0), -+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_MOUTM2, 7, 1, 0), -+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_MOUTM1, 7, 1, 0), -+}; -+ -+/* Mono 2 Mux */ -+static const struct snd_kcontrol_new wm8753_mono2_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[17]); -+ -+/* Out 3 Mux */ -+static const struct snd_kcontrol_new wm8753_out3_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[18]); -+ -+/* Out 4 Mux */ -+static const struct snd_kcontrol_new wm8753_out4_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[19]); -+ -+/* ADC Mono Mix */ -+static const struct snd_kcontrol_new wm8753_adc_mono_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[22]); -+ -+/* Record mixer */ -+static const struct snd_kcontrol_new wm8753_record_mixer_controls[] = { -+SOC_DAPM_SINGLE("Voice Capture Switch", WM8753_RECMIX2, 3, 1, 0), -+SOC_DAPM_SINGLE("Left Capture Switch", WM8753_RECMIX1, 3, 1, 0), -+SOC_DAPM_SINGLE("Right Capture Switch", WM8753_RECMIX1, 7, 1, 0), -+}; -+ -+/* Left ADC mux */ -+static const struct snd_kcontrol_new wm8753_adc_left_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[21]); -+ -+/* Right ADC mux */ -+static const struct snd_kcontrol_new wm8753_adc_right_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[20]); -+ -+/* MIC mux */ -+static const struct snd_kcontrol_new wm8753_mic_mux_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[16]); -+ -+/* ALC mixer */ -+static const struct snd_kcontrol_new wm8753_alc_mixer_controls[] = { -+SOC_DAPM_SINGLE("Line Capture Switch", WM8753_INCTL2, 3, 1, 0), -+SOC_DAPM_SINGLE("Mic2 Capture Switch", WM8753_INCTL2, 2, 1, 0), -+SOC_DAPM_SINGLE("Mic1 Capture Switch", WM8753_INCTL2, 1, 1, 0), -+SOC_DAPM_SINGLE("Rx Capture Switch", WM8753_INCTL2, 0, 1, 0), -+}; -+ -+/* Left Line mux */ -+static const struct snd_kcontrol_new wm8753_line_left_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[14]); -+ -+/* Right Line mux */ -+static const struct snd_kcontrol_new wm8753_line_right_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[13]); -+ -+/* Mono Line mux */ -+static const struct snd_kcontrol_new wm8753_line_mono_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[12]); -+ -+/* Line mux and mixer */ -+static const struct snd_kcontrol_new wm8753_line_mux_mix_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[11]); -+ -+/* Rx mux and mixer */ -+static const struct snd_kcontrol_new wm8753_rx_mux_mix_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[15]); -+ -+/* Mic Selector Mux */ -+static const struct snd_kcontrol_new wm8753_mic_sel_mux_controls = -+SOC_DAPM_ENUM("Route", wm8753_enum[25]); -+ -+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { -+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8753_PWR1, 5, 0), -+SND_SOC_DAPM_MIXER("Left Mixer", WM8753_PWR4, 0, 0, -+ &wm8753_left_mixer_controls[0], ARRAY_SIZE(wm8753_left_mixer_controls)), -+SND_SOC_DAPM_PGA("Left Out 1", WM8753_PWR3, 8, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Left Out 2", WM8753_PWR3, 6, 0, NULL, 0), -+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", WM8753_PWR1, 3, 0), -+SND_SOC_DAPM_OUTPUT("LOUT1"), -+SND_SOC_DAPM_OUTPUT("LOUT2"), -+SND_SOC_DAPM_MIXER("Right Mixer", WM8753_PWR4, 1, 0, -+ &wm8753_right_mixer_controls[0], ARRAY_SIZE(wm8753_right_mixer_controls)), -+SND_SOC_DAPM_PGA("Right Out 1", WM8753_PWR3, 7, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Right Out 2", WM8753_PWR3, 5, 0, NULL, 0), -+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", WM8753_PWR1, 2, 0), -+SND_SOC_DAPM_OUTPUT("ROUT1"), -+SND_SOC_DAPM_OUTPUT("ROUT2"), -+SND_SOC_DAPM_MIXER("Mono Mixer", WM8753_PWR4, 2, 0, -+ &wm8753_mono_mixer_controls[0], ARRAY_SIZE(wm8753_mono_mixer_controls)), -+SND_SOC_DAPM_PGA("Mono Out 1", WM8753_PWR3, 2, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mono Out 2", WM8753_PWR3, 1, 0, NULL, 0), -+SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0), -+SND_SOC_DAPM_OUTPUT("MONO1"), -+SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls), -+SND_SOC_DAPM_OUTPUT("MONO2"), -+SND_SOC_DAPM_MIXER("Out3 Left + Right", -1, 0, 0, NULL, 0), -+SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls), -+SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0), -+SND_SOC_DAPM_OUTPUT("OUT3"), -+SND_SOC_DAPM_MUX("Out4 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out4_controls), -+SND_SOC_DAPM_PGA("Out 4", WM8753_PWR3, 3, 0, NULL, 0), -+SND_SOC_DAPM_OUTPUT("OUT4"), -+SND_SOC_DAPM_MIXER("Playback Mixer", WM8753_PWR4, 3, 0, -+ &wm8753_record_mixer_controls[0], -+ ARRAY_SIZE(wm8753_record_mixer_controls)), -+SND_SOC_DAPM_ADC("Left ADC", "Left Voice Capture", WM8753_PWR2, 3, 0), -+SND_SOC_DAPM_ADC("Right ADC", "Right Voice Capture", WM8753_PWR2, 2, 0), -+SND_SOC_DAPM_MUX("Capture Left Mixer", SND_SOC_NOPM, 0, 0, -+ &wm8753_adc_mono_controls), -+SND_SOC_DAPM_MUX("Capture Right Mixer", SND_SOC_NOPM, 0, 0, -+ &wm8753_adc_mono_controls), -+SND_SOC_DAPM_MUX("Capture Left Mux", SND_SOC_NOPM, 0, 0, -+ &wm8753_adc_left_controls), -+SND_SOC_DAPM_MUX("Capture Right Mux", SND_SOC_NOPM, 0, 0, -+ &wm8753_adc_right_controls), -+SND_SOC_DAPM_MUX("Mic Sidetone Mux", SND_SOC_NOPM, 0, 0, -+ &wm8753_mic_mux_controls), -+SND_SOC_DAPM_PGA("Left Capture Volume", WM8753_PWR2, 5, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Right Capture Volume", WM8753_PWR2, 4, 0, NULL, 0), -+SND_SOC_DAPM_MIXER("ALC Mixer", WM8753_PWR2, 6, 0, -+ &wm8753_alc_mixer_controls[0], ARRAY_SIZE(wm8753_alc_mixer_controls)), -+SND_SOC_DAPM_MUX("Line Left Mux", SND_SOC_NOPM, 0, 0, -+ &wm8753_line_left_controls), -+SND_SOC_DAPM_MUX("Line Right Mux", SND_SOC_NOPM, 0, 0, -+ &wm8753_line_right_controls), -+SND_SOC_DAPM_MUX("Line Mono Mux", SND_SOC_NOPM, 0, 0, -+ &wm8753_line_mono_controls), -+SND_SOC_DAPM_MUX("Line Mixer", SND_SOC_NOPM, 0, 0, -+ &wm8753_line_mux_mix_controls), -+SND_SOC_DAPM_MUX("Rx Mixer", SND_SOC_NOPM, 0, 0, -+ &wm8753_rx_mux_mix_controls), -+SND_SOC_DAPM_PGA("Mic 1 Volume", WM8753_PWR2, 8, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mic 2 Volume", WM8753_PWR2, 7, 0, NULL, 0), -+SND_SOC_DAPM_MUX("Mic Selection Mux", SND_SOC_NOPM, 0, 0, -+ &wm8753_mic_sel_mux_controls), -+SND_SOC_DAPM_INPUT("LINE1"), -+SND_SOC_DAPM_INPUT("LINE2"), -+SND_SOC_DAPM_INPUT("RXP"), -+SND_SOC_DAPM_INPUT("RXN"), -+SND_SOC_DAPM_INPUT("ACIN"), -+SND_SOC_DAPM_INPUT("ACOP"), -+SND_SOC_DAPM_INPUT("MIC1N"), -+SND_SOC_DAPM_INPUT("MIC1"), -+SND_SOC_DAPM_INPUT("MIC2N"), -+SND_SOC_DAPM_INPUT("MIC2"), -+SND_SOC_DAPM_VMID("VREF"), -+}; -+ -+static const char *audio_map[][3] = { -+ /* left mixer */ -+ {"Left Mixer", "Left Playback Switch", "Left DAC"}, -+ {"Left Mixer", "Voice Playback Switch", "Voice DAC"}, -+ {"Left Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"}, -+ {"Left Mixer", "Bypass Playback Switch", "Line Left Mux"}, -+ -+ /* right mixer */ -+ {"Right Mixer", "Right Playback Switch", "Right DAC"}, -+ {"Right Mixer", "Voice Playback Switch", "Voice DAC"}, -+ {"Right Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"}, -+ {"Right Mixer", "Bypass Playback Switch", "Line Right Mux"}, -+ -+ /* mono mixer */ -+ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, -+ {"Mono Mixer", "Left Playback Switch", "Left DAC"}, -+ {"Mono Mixer", "Right Playback Switch", "Right DAC"}, -+ {"Mono Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"}, -+ {"Mono Mixer", "Bypass Playback Switch", "Line Mono Mux"}, -+ -+ /* left out */ -+ {"Left Out 1", NULL, "Left Mixer"}, -+ {"Left Out 2", NULL, "Left Mixer"}, -+ {"LOUT1", NULL, "Left Out 1"}, -+ {"LOUT2", NULL, "Left Out 2"}, -+ -+ /* right out */ -+ {"Right Out 1", NULL, "Right Mixer"}, -+ {"Right Out 2", NULL, "Right Mixer"}, -+ {"ROUT1", NULL, "Right Out 1"}, -+ {"ROUT2", NULL, "Right Out 2"}, -+ -+ /* mono 1 out */ -+ {"Mono Out 1", NULL, "Mono Mixer"}, -+ {"MONO1", NULL, "Mono Out 1"}, -+ -+ /* mono 2 out */ -+ {"Mono 2 Mux", "Left + Right", "Out3 Left + Right"}, -+ {"Mono 2 Mux", "Inverted Mono 1", "MONO1"}, -+ {"Mono 2 Mux", "Left", "Left Mixer"}, -+ {"Mono 2 Mux", "Right", "Right Mixer"}, -+ {"Mono Out 2", NULL, "Mono 2 Mux"}, -+ {"MONO2", NULL, "Mono Out 2"}, -+ -+ /* out 3 */ -+ {"Out3 Left + Right", NULL, "Left Mixer"}, -+ {"Out3 Left + Right", NULL, "Right Mixer"}, -+ {"Out3 Mux", "VREF", "VREF"}, -+ {"Out3 Mux", "Left + Right", "Out3 Left + Right"}, -+ {"Out3 Mux", "ROUT2", "ROUT2"}, -+ {"Out 3", NULL, "Out3 Mux"}, -+ {"OUT3", NULL, "Out 3"}, -+ -+ /* out 4 */ -+ {"Out4 Mux", "VREF", "VREF"}, -+ {"Out4 Mux", "Capture ST", "Capture ST Mixer"}, -+ {"Out4 Mux", "LOUT2", "LOUT2"}, -+ {"Out 4", NULL, "Out4 Mux"}, -+ {"OUT4", NULL, "Out 4"}, -+ -+ /* record mixer */ -+ {"Playback Mixer", "Left Capture Switch", "Left Mixer"}, -+ {"Playback Mixer", "Voice Capture Switch", "Mono Mixer"}, -+ {"Playback Mixer", "Right Capture Switch", "Right Mixer"}, -+ -+ /* Mic/SideTone Mux */ -+ {"Mic Sidetone Mux", "Left PGA", "Left Capture Volume"}, -+ {"Mic Sidetone Mux", "Right PGA", "Right Capture Volume"}, -+ {"Mic Sidetone Mux", "Mic 1", "Mic 1 Volume"}, -+ {"Mic Sidetone Mux", "Mic 2", "Mic 2 Volume"}, -+ -+ /* Capture Left Mux */ -+ {"Capture Left Mux", "PGA", "Left Capture Volume"}, -+ {"Capture Left Mux", "Line or RXP-RXN", "Line Left Mux"}, -+ {"Capture Left Mux", "Line", "LINE1"}, -+ -+ /* Capture Right Mux */ -+ {"Capture Right Mux", "PGA", "Right Capture Volume"}, -+ {"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"}, -+ {"Capture Right Mux", "Sidetone", "Capture ST Mixer"}, -+ -+ /* Mono Capture mixer-mux */ -+ {"Capture Right Mixer", "Stereo", "Capture Right Mux"}, -+ {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"}, -+ {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"}, -+ {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"}, -+ {"Capture Right Mixer", "Analogue Mix Right", "Capture Right Mux"}, -+ {"Capture Left Mixer", "Digital Mono Mix", "Capture Left Mux"}, -+ {"Capture Left Mixer", "Digital Mono Mix", "Capture Right Mux"}, -+ {"Capture Right Mixer", "Digital Mono Mix", "Capture Left Mux"}, -+ {"Capture Right Mixer", "Digital Mono Mix", "Capture Right Mux"}, -+ -+ /* ADC */ -+ {"Left ADC", NULL, "Capture Left Mixer"}, -+ {"Right ADC", NULL, "Capture Right Mixer"}, -+ -+ /* Left Capture Volume */ -+ {"Left Capture Volume", NULL, "ACIN"}, -+ -+ /* Right Capture Volume */ -+ {"Right Capture Volume", NULL, "Mic 2 Volume"}, -+ -+ /* ALC Mixer */ -+ {"ALC Mixer", "Line Capture Switch", "Line Mixer"}, -+ {"ALC Mixer", "Mic2 Capture Switch", "Mic 2 Volume"}, -+ {"ALC Mixer", "Mic1 Capture Switch", "Mic 1 Volume"}, -+ {"ALC Mixer", "Rx Capture Switch", "Rx Mixer"}, -+ -+ /* Line Left Mux */ -+ {"Line Left Mux", "Line 1", "LINE1"}, -+ {"Line Left Mux", "Rx Mix", "Rx Mixer"}, -+ -+ /* Line Right Mux */ -+ {"Line Right Mux", "Line 2", "LINE2"}, -+ {"Line Right Mux", "Rx Mix", "Rx Mixer"}, -+ -+ /* Line Mono Mux */ -+ {"Line Mono Mux", "Line Mix", "Line Mixer"}, -+ {"Line Mono Mux", "Rx Mix", "Rx Mixer"}, -+ -+ /* Line Mixer/Mux */ -+ {"Line Mixer", "Line 1 + 2", "LINE1"}, -+ {"Line Mixer", "Line 1 - 2", "LINE1"}, -+ {"Line Mixer", "Line 1 + 2", "LINE2"}, -+ {"Line Mixer", "Line 1 - 2", "LINE2"}, -+ {"Line Mixer", "Line 1", "LINE1"}, -+ {"Line Mixer", "Line 2", "LINE2"}, -+ -+ /* Rx Mixer/Mux */ -+ {"Rx Mixer", "RXP - RXN", "RXP"}, -+ {"Rx Mixer", "RXP + RXN", "RXP"}, -+ {"Rx Mixer", "RXP - RXN", "RXN"}, -+ {"Rx Mixer", "RXP + RXN", "RXN"}, -+ {"Rx Mixer", "RXP", "RXP"}, -+ {"Rx Mixer", "RXN", "RXN"}, -+ -+ /* Mic 1 Volume */ -+ {"Mic 1 Volume", NULL, "MIC1N"}, -+ {"Mic 1 Volume", NULL, "Mic Selection Mux"}, -+ -+ /* Mic 2 Volume */ -+ {"Mic 2 Volume", NULL, "MIC2N"}, -+ {"Mic 2 Volume", NULL, "MIC2"}, -+ -+ /* Mic Selector Mux */ -+ {"Mic Selection Mux", "Mic 1", "MIC1"}, -+ {"Mic Selection Mux", "Mic 2", "MIC2N"}, -+ {"Mic Selection Mux", "Mic 3", "MIC2"}, -+ -+ /* ACOP */ -+ {"ACOP", NULL, "ALC Mixer"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8753_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); -+ } -+ -+ /* set up the WM8753 audio map */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+/* PLL divisors */ -+struct _pll_div { -+ u32 pll_in; /* ext clock input */ -+ u32 pll_out; /* pll out freq */ -+ u32 div2:1; -+ u32 n:4; -+ u32 k:24; -+}; -+ -+/* -+ * PLL divisors - -+ */ -+static const struct _pll_div pll_div[] = { -+ {13000000, 12288000, 0, 0x7, 0x23F54A}, -+ {13000000, 11289600, 0, 0x6, 0x3CA2F5}, -+ {12000000, 12288000, 0, 0x8, 0x0C49BA}, -+ {12000000, 11289600, 0, 0x7, 0x21B08A}, -+ {24000000, 12288000, 1, 0x8, 0x0C49BA}, -+ {24000000, 11289600, 1, 0x7, 0x21B08A}, -+ {12288000, 11289600, 0, 0x7, 0x166667}, -+ {26000000, 11289600, 1, 0x6, 0x3CA2F5}, -+ {26000000, 12288000, 1, 0x7, 0x23F54A}, -+}; -+ -+static u32 wm8753_config_pll(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int pll) -+{ -+ u16 reg; -+ int found = 0; -+ -+ if (pll == 1) { -+ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xffef; -+ if (!dai->pll_in || !dai->mclk) { -+ /* disable PLL1 */ -+ wm8753_write(codec, WM8753_PLL1CTL1, 0x0026); -+ wm8753_write(codec, WM8753_CLOCK, reg); -+ return 0; -+ } else { -+ u16 value = 0; -+ int i = 0; -+ -+ /* if we cant match, then use good values for N and K */ -+ for (;i < ARRAY_SIZE(pll_div); i++) { -+ if (pll_div[i].pll_out == dai->pll_out && -+ pll_div[i].pll_in == dai->pll_in) { -+ found = 1; -+ break; -+ } -+ } -+ -+ if (!found) -+ goto err; -+ -+ /* set up N and K PLL divisor ratios */ -+ /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */ -+ value = (pll_div[i].n << 5) + ((pll_div[i].k & 0x3c0000) >> 18); -+ wm8753_write(codec, WM8753_PLL1CTL2, value); -+ -+ /* bits 8:0 = PLL_K[17:9] */ -+ value = (pll_div[i].k & 0x03fe00) >> 9; -+ wm8753_write(codec, WM8753_PLL1CTL3, value); -+ -+ /* bits 8:0 = PLL_K[8:0] */ -+ value = pll_div[i].k & 0x0001ff; -+ wm8753_write(codec, WM8753_PLL1CTL4, value); -+ -+ /* set PLL1 as input and enable */ -+ wm8753_write(codec, WM8753_PLL1CTL1, 0x0027 | -+ (pll_div[i].div2 << 3)); -+ wm8753_write(codec, WM8753_CLOCK, reg | 0x0010); -+ } -+ } else { -+ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfff7; -+ if (!dai->pll_in || !dai->mclk) { -+ /* disable PLL2 */ -+ wm8753_write(codec, WM8753_PLL2CTL1, 0x0026); -+ wm8753_write(codec, WM8753_CLOCK, reg); -+ return 0; -+ } else { -+ u16 value = 0; -+ int i = 0; -+ -+ /* if we cant match, then use good values for N and K */ -+ for (;i < ARRAY_SIZE(pll_div); i++) { -+ if (pll_div[i].pll_out == dai->pll_out && -+ pll_div[i].pll_in == dai->pll_in) { -+ found = 1; -+ break; -+ } -+ } -+ -+ if (!found) -+ goto err; -+ -+ /* set up N and K PLL divisor ratios */ -+ /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */ -+ value = (pll_div[i].n << 5) + ((pll_div[i].k & 0x3c0000) >> 18); -+ wm8753_write(codec, WM8753_PLL2CTL2, value); -+ -+ /* bits 8:0 = PLL_K[17:9] */ -+ value = (pll_div[i].k & 0x03fe00) >> 9; -+ wm8753_write(codec, WM8753_PLL2CTL3, value); -+ -+ /* bits 8:0 = PLL_K[8:0] */ -+ value = pll_div[i].k & 0x0001ff; -+ wm8753_write(codec, WM8753_PLL2CTL4, value); -+ -+ /* set PLL1 as input and enable */ -+ wm8753_write(codec, WM8753_PLL2CTL1, 0x0027 | -+ (pll_div[i].div2 << 3)); -+ wm8753_write(codec, WM8753_CLOCK, reg | 0x0008); -+ } -+ } -+ -+ return dai->pll_in; -+err: -+ return 0; -+} -+ -+struct _coeff_div { -+ u32 mclk; -+ u32 rate; -+ u16 fs; -+ u8 sr:5; -+ u8 usb:1; -+}; -+ -+/* codec hifi mclk (after PLL) clock divider coefficients */ -+static const struct _coeff_div coeff_div[] = { -+ /* 8k */ -+ {12288000, 8000, 1536, 0x6, 0x0}, -+ {11289600, 8000, 1408, 0x16, 0x0}, -+ {18432000, 8000, 2304, 0x7, 0x0}, -+ {16934400, 8000, 2112, 0x17, 0x0}, -+ {12000000, 8000, 1500, 0x6, 0x1}, -+ -+ /* 11.025k */ -+ {11289600, 11025, 1024, 0x18, 0x0}, -+ {16934400, 11025, 1536, 0x19, 0x0}, -+ {12000000, 11025, 1088, 0x19, 0x1}, -+ -+ /* 16k */ -+ {12288000, 16000, 768, 0xa, 0x0}, -+ {18432000, 16000, 1152, 0xb, 0x0}, -+ {12000000, 16000, 750, 0xa, 0x1}, -+ -+ /* 22.05k */ -+ {11289600, 22050, 512, 0x1a, 0x0}, -+ {16934400, 22050, 768, 0x1b, 0x0}, -+ {12000000, 22050, 544, 0x1b, 0x1}, -+ -+ /* 32k */ -+ {12288000, 32000, 384, 0xc, 0x0}, -+ {18432000, 32000, 576, 0xd, 0x0}, -+ {12000000, 32000, 375, 0xa, 0x1}, -+ -+ /* 44.1k */ -+ {11289600, 44100, 256, 0x10, 0x0}, -+ {16934400, 44100, 384, 0x11, 0x0}, -+ {12000000, 44100, 272, 0x11, 0x1}, -+ -+ /* 48k */ -+ {12288000, 48000, 256, 0x0, 0x0}, -+ {18432000, 48000, 384, 0x1, 0x0}, -+ {12000000, 48000, 250, 0x0, 0x1}, -+ -+ /* 88.2k */ -+ {11289600, 88200, 128, 0x1e, 0x0}, -+ {16934400, 88200, 192, 0x1f, 0x0}, -+ {12000000, 88200, 136, 0x1f, 0x1}, -+ -+ /* 96k */ -+ {12288000, 96000, 128, 0xe, 0x0}, -+ {18432000, 96000, 192, 0xf, 0x0}, -+ {12000000, 96000, 125, 0xe, 0x1}, -+}; -+ -+static int get_coeff(int mclk, int rate) -+{ -+ int i; -+ -+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { -+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) -+ return i; -+ } -+ return -EINVAL; -+} -+ -+/* supported HiFi input clocks (that don't use PLL) */ -+const static int hifi_clks[] = {11289600, 12000000, 12288000, -+ 16934400, 18432000}; -+ -+/* The HiFi interface can be clocked in one of two ways:- -+ * o No PLL - MCLK is used directly. -+ * o PLL - PLL is used to generate audio MCLK from input clock. -+ * -+ * We use the direct method if we can as it saves power. -+ */ -+static unsigned int wm8753_config_i2s_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ int i, pll_out; -+ -+ /* is clk supported without the PLL */ -+ for(i = 0; i < ARRAY_SIZE(hifi_clks); i++) { -+ if (clk == hifi_clks[i]) { -+ dai->mclk = clk; -+ dai->pll_in = dai->pll_out = 0; -+ dai->clk_div = 1; -+ return clk; -+ } -+ } -+ -+ /* determine best PLL output speed */ -+ if (info->bclk_master & -+ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) { -+ pll_out = info->fs * info->rate; -+ } else { -+ /* calc slave clock */ -+ switch (info->rate){ -+ case 11025: -+ case 22050: -+ case 44100: -+ case 88200: -+ pll_out = 11289600; -+ break; -+ default: -+ pll_out = 12288000; -+ break; -+ } -+ } -+ -+ /* are input & output clocks supported by PLL */ -+ for (i = 0;i < ARRAY_SIZE(pll_div); i++) { -+ if (pll_div[i].pll_in == clk && pll_div[i].pll_out == pll_out) { -+ dai->pll_in = clk; -+ dai->pll_out = dai->mclk = pll_out; -+ return pll_out; -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+} -+ -+/* valid PCM clock dividers * 2 */ -+static int pcm_divs[] = {2, 6, 11, 4, 8, 12, 16}; -+ -+/* The Voice interface can be clocked in one of four ways:- -+ * o No PLL - MCLK is used directly. -+ * o Div - MCLK is directly divided. -+ * o PLL - PLL is used to generate audio MCLK from input clock. -+ * o PLL & Div - PLL and post divider are used. -+ * -+ * We use the non PLL methods if we can, as it saves power. -+ */ -+ -+static unsigned int wm8753_config_pcm_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ int i, j, best_clk = info->fs * info->rate; -+ -+ /* can we run at this clk without the PLL ? */ -+ for (i = 0; i < ARRAY_SIZE(pcm_divs); i++) { -+ if ((best_clk >> 1) * pcm_divs[i] == clk) { -+ dai->pll_in = 0; -+ dai->clk_div = pcm_divs[i]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ -+ /* now check for PLL support */ -+ for (i = 0; i < ARRAY_SIZE(pll_div); i++) { -+ if (pll_div[i].pll_in == clk) { -+ for (j = 0; j < ARRAY_SIZE(pcm_divs); j++) { -+ if (pll_div[i].pll_out == pcm_divs[j] * (best_clk >> 1)) { -+ dai->pll_in = clk; -+ dai->pll_out = pll_div[i].pll_out; -+ dai->clk_div = pcm_divs[j]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+} -+ -+/* set the format and bit size for ADC and Voice DAC */ -+static void wm8753_adc_vdac_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01e0; -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ voice |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ voice |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ voice |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ voice |= 0x0013; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ voice |= 0x0004; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ voice |= 0x0008; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ voice |= 0x000c; -+ break; -+ } -+ -+ wm8753_write(codec, WM8753_PCM, voice); -+} -+ -+/* configure PCM DAI */ -+static int wm8753_pcm_dai_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 voice, ioctl, srate, srate2, fs, bfs, clock; -+ unsigned int rate; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ fs = rtd->codec_dai->dai_runtime.fs; -+ rate = snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate); -+ voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x001f; -+ -+ /* set master/slave audio interface */ -+ ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x01fd; -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ ioctl |= 0x0002; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ voice |= 0x0040; -+ break; -+ } -+ -+ /* do we need to enable the PLL */ -+ if (rtd->codec_dai->pll_in) { -+ if (wm8753_config_pll(codec, rtd->codec_dai, 2) != -+ rtd->codec_dai->pll_in) { -+ err("could not set pll to %d --> %d", -+ rtd->codec_dai->pll_in, rtd->codec_dai->pll_out); -+ return -ENODEV; -+ } -+ } -+ -+ /* set up PCM divider */ -+ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0x003f; -+ switch (rtd->codec_dai->clk_div) { -+ case 2: /* 1 */ -+ break; -+ case 6: /* 3 */ -+ clock |= (0x2 << 6); -+ break; -+ case 11: /* 5.5 */ -+ clock |= (0x3 << 6); -+ break; -+ case 4: /* 2 */ -+ clock |= (0x4 << 6); -+ break; -+ case 8: /* 4 */ -+ clock |= (0x5 << 6); -+ break; -+ case 12: /* 6 */ -+ clock |= (0x6 << 6); -+ break; -+ case 16: /* 8 */ -+ clock |= (0x7 << 6); -+ break; -+ default: -+ printk(KERN_ERR "wm8753: invalid PCM clk divider %d\n", -+ rtd->codec_dai->clk_div); -+ break; -+ } -+ wm8753_write(codec, WM8753_CLOCK, clock); -+ -+ /* set bclk divisor rate */ -+ srate2 = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x003f; -+ switch (bfs) { -+ case 1: -+ break; -+ case 2: -+ srate2 |= (0x1 << 6); -+ break; -+ case 4: -+ srate2 |= (0x2 << 6); -+ break; -+ case 8: -+ srate2 |= (0x3 << 6); -+ break; -+ case 16: -+ srate2 |= (0x4 << 6); -+ break; -+ } -+ wm8753_write(codec, WM8753_SRATE2, srate2); -+ -+ srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f; -+ if (rtd->codec_dai->dai_runtime.fs == 384) -+ srate |= 0x80; -+ wm8753_write(codec, WM8753_SRATE1, srate); -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_IB_IF: -+ voice |= 0x0090; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ voice |= 0x0080; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ voice |= 0x0010; -+ break; -+ } -+ //printk("voice %x %x ioctl %x %x srate2 %x %x srate1 %x %x\n", -+ //WM8753_PCM, voice, WM8753_IOCTL, ioctl, WM8753_SRATE2, -+ //srate2, WM8753_SRATE1, srate); -+ -+ wm8753_write(codec, WM8753_IOCTL, ioctl); -+ wm8753_write(codec, WM8753_PCM, voice); -+ return 0; -+} -+ -+/* configure hifi DAC wordlength and format */ -+static void wm8753_hdac_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01e0; -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ hifi |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ hifi |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ hifi |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ hifi |= 0x0013; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ hifi |= 0x0004; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ hifi |= 0x0008; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ hifi |= 0x000c; -+ break; -+ } -+ -+ wm8753_write(codec, WM8753_HIFI, hifi); -+} -+ -+/* configure i2s (hifi) DAI clocking */ -+static int wm8753_i2s_dai_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 srate, bfs, hifi, ioctl; -+ unsigned int rate; -+ int i = 0; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ rate = snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate); -+ hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x001f; -+ -+ /* is coefficient valid ? */ -+ if ((i = get_coeff(rtd->codec_dai->mclk, rate)) < 0) -+ return i; -+ -+ srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0; -+ wm8753_write(codec, WM8753_SRATE1, srate | (coeff_div[i].sr << 1) | -+ coeff_div[i].usb); -+ -+ /* do we need to enable the PLL */ -+ if (rtd->codec_dai->pll_in) { -+ if (wm8753_config_pll(codec, rtd->codec_dai, 1) != -+ rtd->codec_dai->pll_in) { -+ err("could not set pll to %d --> %d", -+ rtd->codec_dai->pll_in, rtd->codec_dai->pll_out); -+ return -ENODEV; -+ } -+ } -+ -+ /* set bclk divisor rate */ -+ srate = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x01c7; -+ switch (bfs) { -+ case 1: -+ break; -+ case 2: -+ srate |= (0x1 << 3); -+ break; -+ case 4: -+ srate |= (0x2 << 3); -+ break; -+ case 8: -+ srate |= (0x3 << 3); -+ break; -+ case 16: -+ srate |= (0x4 << 3); -+ break; -+ } -+ wm8753_write(codec, WM8753_SRATE2, srate); -+ -+ /* set master/slave audio interface */ -+ ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x00fe; -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ ioctl |= 0x0001; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ hifi |= 0x0040; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_IB_IF: -+ hifi |= 0x0090; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ hifi |= 0x0080; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ hifi |= 0x0010; -+ break; -+ } -+ wm8753_write(codec, WM8753_IOCTL, ioctl); -+ wm8753_write(codec, WM8753_HIFI, hifi); -+ return 0; -+} -+ -+static int wm8753_mode1v_prepare (struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 clock; -+ -+ /* set clk source as pcmclk */ -+ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb; -+ wm8753_write(codec, WM8753_CLOCK, clock); -+ -+ wm8753_adc_vdac_prepare(substream); -+ return wm8753_pcm_dai_prepare(substream); -+} -+ -+static int wm8753_mode1h_prepare (struct snd_pcm_substream *substream) -+{ -+ wm8753_hdac_prepare(substream); -+ return wm8753_i2s_dai_prepare(substream); -+} -+ -+static int wm8753_mode2_prepare (struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 clock; -+ -+ /* set clk source as pcmclk */ -+ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb; -+ wm8753_write(codec, WM8753_CLOCK, clock); -+ -+ wm8753_adc_vdac_prepare(substream); -+ return wm8753_i2s_dai_prepare(substream); -+} -+ -+static int wm8753_mode3_prepare (struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 clock; -+ -+ /* set clk source as mclk */ -+ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb; -+ wm8753_write(codec, WM8753_CLOCK, clock | 0x4); -+ -+ wm8753_hdac_prepare(substream); -+ wm8753_adc_vdac_prepare(substream); -+ return wm8753_i2s_dai_prepare(substream); -+} -+ -+static int wm8753_mode4_prepare (struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 clock; -+ -+ /* set clk source as mclk */ -+ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb; -+ wm8753_write(codec, WM8753_CLOCK, clock | 0x4); -+ -+ wm8753_hdac_prepare(substream); -+ wm8753_adc_vdac_prepare(substream); -+ return wm8753_i2s_dai_prepare(substream); -+} -+ -+static int wm8753_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7; -+ -+ /* the digital mute covers the HiFi and Voice DAC's on the WM8753. -+ * make sure we check if they are not both active when we mute */ -+ if (mute && dai->id == 1) { -+ if (!wm8753_dai[WM8753_DAI_VOICE].playback.active || -+ !wm8753_dai[WM8753_DAI_HIFI].playback.active) -+ wm8753_write(codec, WM8753_DAC, mute_reg | 0x8); -+ } else { -+ if (mute) -+ wm8753_write(codec, WM8753_DAC, mute_reg | 0x8); -+ else -+ wm8753_write(codec, WM8753_DAC, mute_reg); -+ } -+ -+ return 0; -+} -+ -+static int wm8753_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e; -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* set vmid to 50k and unmute dac */ -+ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ /* set vmid to 5k for quick power up */ -+ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1); -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* mute dac and set vmid to 500k, enable VREF */ -+ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ wm8753_write(codec, WM8753_PWR1, 0x0001); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+/* -+ * The WM8753 supports upto 4 different and mutually exclusive DAI -+ * configurations. This gives 2 PCM's available for use, hifi and voice. -+ * NOTE: The Voice PCM cannot play or caputure audio to the CPU as it's DAI -+ * is connected between the wm8753 and a BT codec or GSM modem. -+ * -+ * 1. Voice over PCM DAI - HIFI DAC over HIFI DAI -+ * 2. Voice over HIFI DAI - HIFI disabled -+ * 3. Voice disabled - HIFI over HIFI -+ * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture -+ */ -+static const struct snd_soc_codec_dai wm8753_all_dai[] = { -+/* DAI HiFi mode 1 */ -+{ .name = "WM8753 HiFi", -+ .id = 1, -+ .playback = { -+ .stream_name = "HiFi Playback", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { /* dummy for fast DAI switching */ -+ .stream_name = "HiFi Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .config_sysclk = wm8753_config_i2s_sysclk, -+ .digital_mute = wm8753_mute, -+ .ops = { -+ .prepare = wm8753_mode1h_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8753_hifi_modes), -+ .mode = wm8753_hifi_modes,}, -+}, -+/* DAI Voice mode 1 */ -+{ .name = "WM8753 Voice", -+ .id = 1, -+ .playback = { -+ .stream_name = "Voice Playback", -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .capture = { -+ .stream_name = "Voice Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .config_sysclk = wm8753_config_pcm_sysclk, -+ .digital_mute = wm8753_mute, -+ .ops = { -+ .prepare = wm8753_mode1v_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8753_voice_modes), -+ .mode = wm8753_voice_modes,}, -+}, -+/* DAI HiFi mode 2 - dummy */ -+{ .name = "WM8753 HiFi", -+ .id = 2, -+}, -+/* DAI Voice mode 2 */ -+{ .name = "WM8753 Voice", -+ .id = 2, -+ .playback = { -+ .stream_name = "Voice Playback", -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .capture = { -+ .stream_name = "Voice Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .config_sysclk = wm8753_config_i2s_sysclk, -+ .digital_mute = wm8753_mute, -+ .ops = { -+ .prepare = wm8753_mode2_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8753_voice_modes), -+ .mode = wm8753_voice_modes,}, -+}, -+/* DAI HiFi mode 3 */ -+{ .name = "WM8753 HiFi", -+ .id = 3, -+ .playback = { -+ .stream_name = "HiFi Playback", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .stream_name = "HiFi Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .config_sysclk = wm8753_config_i2s_sysclk, -+ .digital_mute = wm8753_mute, -+ .ops = { -+ .prepare = wm8753_mode3_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8753_hifi_modes), -+ .mode = wm8753_hifi_modes,}, -+}, -+/* DAI Voice mode 3 - dummy */ -+{ .name = "WM8753 Voice", -+ .id = 3, -+}, -+/* DAI HiFi mode 4 */ -+{ .name = "WM8753 HiFi", -+ .id = 4, -+ .playback = { -+ .stream_name = "HiFi Playback", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .stream_name = "HiFi Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .config_sysclk = wm8753_config_i2s_sysclk, -+ .digital_mute = wm8753_mute, -+ .ops = { -+ .prepare = wm8753_mode4_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8753_mixed_modes), -+ .mode = wm8753_mixed_modes,}, -+}, -+/* DAI Voice mode 4 - dummy */ -+{ .name = "WM8753 Voice", -+ .id = 4, -+}, -+}; -+ -+struct snd_soc_codec_dai wm8753_dai[2]; -+EXPORT_SYMBOL_GPL(wm8753_dai); -+ -+static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) -+{ -+ if (mode < 4) { -+ wm8753_dai[0] = wm8753_all_dai[mode << 1]; -+ wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1]; -+ } -+} -+ -+static void wm8753_work(void *data) -+{ -+ struct snd_soc_codec *codec = (struct snd_soc_codec *)data; -+ wm8753_dapm_event(codec, codec->dapm_state); -+} -+ -+static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8753_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) { -+ if (i + 1 == WM8753_RESET) -+ continue; -+ data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ -+ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ /* charge wm8753 caps */ -+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { -+ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); -+ codec->dapm_state = SNDRV_CTL_POWER_D0; -+ queue_delayed_work(wm8753_workq, &wm8753_dapm_work, -+ msecs_to_jiffies(caps_charge)); -+ } -+ -+ return 0; -+} -+ -+/* -+ * initialise the WM8753 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8753_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int reg, ret = 0; -+ -+ codec->name = "WM8753"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8753_read_reg_cache; -+ codec->write = wm8753_write; -+ codec->dapm_event = wm8753_dapm_event; -+ codec->dai = wm8753_dai; -+ codec->num_dai = 2; -+ codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); -+ -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8753_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8753_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8753_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8753_reg); -+ wm8753_set_dai_mode(codec, 0); -+ -+ wm8753_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if (ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* charge output caps */ -+ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); -+ codec->dapm_state = SNDRV_CTL_POWER_D3hot; -+ queue_delayed_work(wm8753_workq, -+ &wm8753_dapm_work, msecs_to_jiffies(caps_charge)); -+ -+ /* set the update bits */ -+ reg = wm8753_read_reg_cache(codec, WM8753_LDAC); -+ wm8753_write(codec, WM8753_LDAC, reg | 0x0100); -+ reg = wm8753_read_reg_cache(codec, WM8753_RDAC); -+ wm8753_write(codec, WM8753_RDAC, reg | 0x0100); -+ reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V); -+ wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100); -+ reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V); -+ wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100); -+ reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V); -+ wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100); -+ reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V); -+ wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100); -+ reg = wm8753_read_reg_cache(codec, WM8753_LINVOL); -+ wm8753_write(codec, WM8753_LINVOL, reg | 0x0100); -+ reg = wm8753_read_reg_cache(codec, WM8753_RINVOL); -+ wm8753_write(codec, WM8753_RINVOL, reg | 0x0100); -+ -+ wm8753_add_controls(codec); -+ wm8753_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+static struct snd_soc_device *wm8753_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8753 2 wire address is determined by GPIO5 -+ * state during powerup. -+ * low = 0x1a -+ * high = 0x1b -+ */ -+#define I2C_DRIVERID_WM8753 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8753_i2c_driver; -+static struct i2c_client client_template; -+ -+static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8753_socdev; -+ struct wm8753_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL){ -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ i2c_set_clientdata(i2c, codec); -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if (ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8753_init(socdev); -+ if (ret < 0) { -+ err("failed to initialise WM8753\n"); -+ goto err; -+ } -+ -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+} -+ -+static int wm8753_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec *codec = i2c_get_clientdata(client); -+ i2c_detach_client(client); -+ kfree(codec->reg_cache); -+ kfree(client); -+ return 0; -+} -+ -+static int wm8753_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8753_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8753_i2c_driver = { -+ .driver = { -+ .name = "WM8753 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8753, -+ .attach_adapter = wm8753_i2c_attach, -+ .detach_client = wm8753_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8753", -+ .driver = &wm8753_i2c_driver, -+}; -+#endif -+ -+static int wm8753_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8753_setup_data *setup; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ info("WM8753 Audio Codec %s", WM8753_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ wm8753_socdev = socdev; -+ INIT_WORK(&wm8753_dapm_work, wm8753_work, codec); -+ wm8753_workq = create_workqueue("wm8753"); -+ if (wm8753_workq == NULL) { -+ kfree(codec); -+ return -ENOMEM; -+ } -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8753_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8753_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ if (wm8753_workq) -+ destroy_workqueue(wm8753_workq); -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8753_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8753 = { -+ .probe = wm8753_probe, -+ .remove = wm8753_remove, -+ .suspend = wm8753_suspend, -+ .resume = wm8753_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); -+ -+MODULE_DESCRIPTION("ASoC WM8753 driver"); -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8753.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8753.h -@@ -0,0 +1,91 @@ -+/* -+ * wm8753.h -- audio driver for WM8753 -+ * -+ * Copyright 2003 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ */ -+ -+#ifndef _WM8753_H -+#define _WM8753_H -+ -+/* WM8753 register space */ -+ -+#define WM8753_DAC 0x01 -+#define WM8753_ADC 0x02 -+#define WM8753_PCM 0x03 -+#define WM8753_HIFI 0x04 -+#define WM8753_IOCTL 0x05 -+#define WM8753_SRATE1 0x06 -+#define WM8753_SRATE2 0x07 -+#define WM8753_LDAC 0x08 -+#define WM8753_RDAC 0x09 -+#define WM8753_BASS 0x0a -+#define WM8753_TREBLE 0x0b -+#define WM8753_ALC1 0x0c -+#define WM8753_ALC2 0x0d -+#define WM8753_ALC3 0x0e -+#define WM8753_NGATE 0x0f -+#define WM8753_LADC 0x10 -+#define WM8753_RADC 0x11 -+#define WM8753_ADCTL1 0x12 -+#define WM8753_3D 0x13 -+#define WM8753_PWR1 0x14 -+#define WM8753_PWR2 0x15 -+#define WM8753_PWR3 0x16 -+#define WM8753_PWR4 0x17 -+#define WM8753_ID 0x18 -+#define WM8753_INTPOL 0x19 -+#define WM8753_INTEN 0x1a -+#define WM8753_GPIO1 0x1b -+#define WM8753_GPIO2 0x1c -+#define WM8753_RESET 0x1f -+#define WM8753_RECMIX1 0x20 -+#define WM8753_RECMIX2 0x21 -+#define WM8753_LOUTM1 0x22 -+#define WM8753_LOUTM2 0x23 -+#define WM8753_ROUTM1 0x24 -+#define WM8753_ROUTM2 0x25 -+#define WM8753_MOUTM1 0x26 -+#define WM8753_MOUTM2 0x27 -+#define WM8753_LOUT1V 0x28 -+#define WM8753_ROUT1V 0x29 -+#define WM8753_LOUT2V 0x2a -+#define WM8753_ROUT2V 0x2b -+#define WM8753_MOUTV 0x2c -+#define WM8753_OUTCTL 0x2d -+#define WM8753_ADCIN 0x2e -+#define WM8753_INCTL1 0x2f -+#define WM8753_INCTL2 0x30 -+#define WM8753_LINVOL 0x31 -+#define WM8753_RINVOL 0x32 -+#define WM8753_MICBIAS 0x33 -+#define WM8753_CLOCK 0x34 -+#define WM8753_PLL1CTL1 0x35 -+#define WM8753_PLL1CTL2 0x36 -+#define WM8753_PLL1CTL3 0x37 -+#define WM8753_PLL1CTL4 0x38 -+#define WM8753_PLL2CTL1 0x39 -+#define WM8753_PLL2CTL2 0x3a -+#define WM8753_PLL2CTL3 0x3b -+#define WM8753_PLL2CTL4 0x3c -+#define WM8753_BIASCTL 0x3d -+#define WM8753_ADCTL2 0x3f -+ -+struct wm8753_setup_data { -+ unsigned short i2c_address; -+}; -+ -+#define WM8753_DAI_HIFI 0 -+#define WM8753_DAI_VOICE 1 -+ -+extern struct snd_soc_codec_dai wm8753_dai[2]; -+extern struct snd_soc_codec_device soc_codec_dev_wm8753; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8772.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8772.c -@@ -0,0 +1,806 @@ -+/* -+ * wm8772.c -- WM8772 ALSA Soc Audio driver -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/version.h> -+#include <linux/kernel.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8772.h" -+ -+#define AUDIO_NAME "WM8772" -+#define WM8772_VERSION "0.3" -+ -+/* -+ * wm8772 register cache -+ * We can't read the WM8772 register space when we -+ * are using 2 wire for device control, so we cache them instead. -+ */ -+static const u16 wm8772_reg[] = { -+ 0x00ff, 0x00ff, 0x0120, 0x0000, /* 0 */ -+ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /* 4 */ -+ 0x00ff, 0x0000, 0x0080, 0x0040, /* 8 */ -+ 0x0000 -+}; -+ -+#define WM8772_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_IB_NF) -+ -+#define WM8772_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8772_PRATES \ -+ (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ -+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) -+ -+#define WM8772_CRATES \ -+ (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ -+ SNDRV_PCM_RATE_96000) -+ -+static struct snd_soc_dai_mode wm8772_modes[] = { -+ /* common codec frame and clock master modes */ -+ /* 32k */ -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 768, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 512, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 192, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 128, -+ .bfs = 64, -+ }, -+ -+ /* 44.1k */ -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 768, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 512, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 192, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 128, -+ .bfs = 64, -+ }, -+ -+ /* 48k */ -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 768, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 512, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 192, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 128, -+ .bfs = 64, -+ }, -+ -+ /* 96k */ -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8772_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 192, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8772_DIR, -+ .pcmrate = SND_SOC_DAI_BFS_RATE, -+ .fs = 128, -+ .bfs = 64, -+ }, -+ -+ /* 192k */ -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_192000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 192, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_192000, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 128, -+ .bfs = 64, -+ }, -+ -+ /* slave mode */ -+ { -+ .fmt = WM8772_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = WM8772_PRATES, -+ .pcmdir = SND_SOC_DAIDIR_PLAYBACK, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+ { -+ .fmt = WM8772_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = WM8772_CRATES, -+ .pcmdir = SND_SOC_DAIDIR_CAPTURE, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8772 register cache -+ */ -+static inline unsigned int wm8772_read_reg_cache(struct snd_soc_codec * codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg > WM8772_CACHE_REGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write wm8772 register cache -+ */ -+static inline void wm8772_write_reg_cache(struct snd_soc_codec * codec, -+ unsigned int reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg > WM8772_CACHE_REGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+static int wm8772_write(struct snd_soc_codec * codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8772 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8772_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -1; -+} -+ -+#define wm8772_reset(c) wm8772_write(c, WM8772_RESET, 0) -+ -+/* -+ * WM8772 Controls -+ */ -+static const char *wm8772_zero_flag[] = {"All Ch", "Ch 1", "Ch 2", "Ch3"}; -+ -+static const struct soc_enum wm8772_enum[] = { -+SOC_ENUM_SINGLE(WM8772_DACCTRL, 0, 4, wm8772_zero_flag), -+}; -+ -+static const struct snd_kcontrol_new wm8772_snd_controls[] = { -+ -+SOC_SINGLE("Left1 Playback Volume", WM8772_LDAC1VOL, 0, 255, 0), -+SOC_SINGLE("Left2 Playback Volume", WM8772_LDAC2VOL, 0, 255, 0), -+SOC_SINGLE("Left3 Playback Volume", WM8772_LDAC3VOL, 0, 255, 0), -+SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC1VOL, 0, 255, 0), -+SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC2VOL, 0, 255, 0), -+SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC3VOL, 0, 255, 0), -+SOC_SINGLE("Master Playback Volume", WM8772_MDACVOL, 0, 255, 0), -+ -+SOC_SINGLE("Playback Switch", WM8772_DACCH, 0, 1, 0), -+SOC_SINGLE("Capture Switch", WM8772_ADCCTRL, 2, 1, 0), -+ -+SOC_SINGLE("Demp1 Playback Switch", WM8772_DACCTRL, 6, 1, 0), -+SOC_SINGLE("Demp2 Playback Switch", WM8772_DACCTRL, 7, 1, 0), -+SOC_SINGLE("Demp3 Playback Switch", WM8772_DACCTRL, 8, 1, 0), -+ -+SOC_SINGLE("Phase Invert 1 Switch", WM8772_IFACE, 6, 1, 0), -+SOC_SINGLE("Phase Invert 2 Switch", WM8772_IFACE, 7, 1, 0), -+SOC_SINGLE("Phase Invert 3 Switch", WM8772_IFACE, 8, 1, 0), -+ -+SOC_SINGLE("Playback ZC Switch", WM8772_DACCTRL, 0, 1, 0), -+ -+SOC_SINGLE("Capture High Pass Switch", WM8772_ADCCTRL, 3, 1, 0), -+}; -+ -+/* add non dapm controls */ -+static int wm8772_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8772_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8772_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ return 0; -+} -+ -+/* valid wm8772 mclk frequencies */ -+static const int freq_table[5][6] = { -+ {4096000, 6144000, 8192000, 12288000, 16384000, 24576000}, -+ {5644800, 8467000, 11289600, 16934000, 22579200, 33868800}, -+ {6144000, 9216000, 12288000, 18432000, 24576000, 36864000}, -+ {12288000, 18432000, 24576000, 36864000, 0, 0}, -+ {24576000, 36864000, 0, 0, 0}, -+}; -+ -+static unsigned int check_freq(int rate, unsigned int freq) -+{ -+ int i; -+ -+ for(i = 0; i < 6; i++) { -+ if(freq == freq_table[i][rate]) -+ return freq; -+ } -+ return 0; -+} -+ -+static unsigned int wm8772_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ switch (info->rate){ -+ case 32000: -+ dai->mclk = check_freq(0, clk); -+ break; -+ case 44100: -+ dai->mclk = check_freq(1, clk); -+ break; -+ case 48000: -+ dai->mclk = check_freq(2, clk); -+ break; -+ case 96000: -+ dai->mclk = check_freq(3, clk); -+ break; -+ case 192000: -+ dai->mclk = check_freq(4, clk); -+ break; -+ default: -+ dai->mclk = 0; -+ } -+ return dai->mclk; -+} -+ -+static int wm8772_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 diface = wm8772_read_reg_cache(codec, WM8772_IFACE) & 0xffc0; -+ u16 diface_ctrl = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0xfe1f; -+ u16 aiface = 0; -+ u16 aiface_ctrl = wm8772_read_reg_cache(codec, WM8772_ADCCTRL) & 0xfcff; -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ diface_ctrl |= 0x0010; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ diface |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ diface |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ diface |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ diface |= 0x0007; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FORMAT_S20_3LE: -+ diface |= 0x0010; -+ break; -+ case SNDRV_PCM_FORMAT_S24_3LE: -+ diface |= 0x0020; -+ break; -+ case SNDRV_PCM_FORMAT_S32_LE: -+ diface |= 0x0030; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ diface |= 0x0008; -+ break; -+ } -+ -+ /* set rate */ -+ switch (rtd->codec_dai->dai_runtime.fs) { -+ case 768: -+ diface_ctrl |= (0x5 << 6); -+ break; -+ case 512: -+ diface_ctrl |= (0x4 << 6); -+ break; -+ case 384: -+ diface_ctrl |= (0x3 << 6); -+ break; -+ case 256: -+ diface_ctrl |= (0x2 << 6); -+ break; -+ case 192: -+ diface_ctrl |= (0x1 << 6); -+ break; -+ } -+ -+ wm8772_write(codec, WM8772_DACRATE, diface_ctrl); -+ wm8772_write(codec, WM8772_IFACE, diface); -+ -+ } else { -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ aiface |= 0x0010; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ aiface |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ aiface |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ aiface |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ aiface |= 0x0003; -+ aiface_ctrl |= 0x0010; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ aiface |= 0x0004; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ aiface |= 0x0008; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ aiface |= 0x000c; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ aiface_ctrl |= 0x0020; -+ break; -+ } -+ -+ /* set rate */ -+ switch (rtd->codec_dai->dai_runtime.fs) { -+ case 768: -+ aiface |= (0x5 << 5); -+ break; -+ case 512: -+ aiface |= (0x4 << 5); -+ break; -+ case 384: -+ aiface |= (0x3 << 5); -+ break; -+ case 256: -+ aiface |= (0x2 << 5); -+ break; -+ } -+ -+ wm8772_write(codec, WM8772_ADCCTRL, aiface_ctrl); -+ wm8772_write(codec, WM8772_ADCRATE, aiface); -+ } -+ -+ return 0; -+} -+ -+static int wm8772_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ u16 master = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0xffe0; -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, clk and osc on, dac unmute, active */ -+ wm8772_write(codec, WM8772_DACRATE, master); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, dac mute, inactive */ -+ wm8772_write(codec, WM8772_DACRATE, master | 0x0f); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, dac mute, inactive */ -+ wm8772_write(codec, WM8772_DACRATE, master | 0x1f); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8772_dai = { -+ .name = "WM8772", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 6, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = wm8772_config_sysclk, -+ .ops = { -+ .prepare = wm8772_pcm_prepare, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8772_modes), -+ .mode = wm8772_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8772_dai); -+ -+static int wm8772_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8772_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8772_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8772_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the WM8772 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8772_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int reg, ret = 0; -+ -+ codec->name = "WM8772"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8772_read_reg_cache; -+ codec->write = wm8772_write; -+ codec->dapm_event = wm8772_dapm_event; -+ codec->dai = &wm8772_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(wm8772_reg); -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8772_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8772_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8772_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8772_reg); -+ -+ wm8772_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if(ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ /* set the update bits */ -+ reg = wm8772_read_reg_cache(codec, WM8772_MDACVOL); -+ wm8772_write(codec, WM8772_MDACVOL, reg | 0x0100); -+ reg = wm8772_read_reg_cache(codec, WM8772_LDAC1VOL); -+ wm8772_write(codec, WM8772_LDAC1VOL, reg | 0x0100); -+ reg = wm8772_read_reg_cache(codec, WM8772_LDAC2VOL); -+ wm8772_write(codec, WM8772_LDAC2VOL, reg | 0x0100); -+ reg = wm8772_read_reg_cache(codec, WM8772_LDAC3VOL); -+ wm8772_write(codec, WM8772_LDAC3VOL, reg | 0x0100); -+ reg = wm8772_read_reg_cache(codec, WM8772_RDAC1VOL); -+ wm8772_write(codec, WM8772_RDAC1VOL, reg | 0x0100); -+ reg = wm8772_read_reg_cache(codec, WM8772_RDAC2VOL); -+ wm8772_write(codec, WM8772_RDAC2VOL, reg | 0x0100); -+ reg = wm8772_read_reg_cache(codec, WM8772_RDAC3VOL); -+ wm8772_write(codec, WM8772_RDAC3VOL, reg | 0x0100); -+ -+ wm8772_add_controls(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *wm8772_socdev; -+ -+static int wm8772_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8772_setup_data *setup; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ printk(KERN_INFO "WM8772 Audio Codec %s", WM8772_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ wm8772_socdev = socdev; -+ -+ /* Add other interfaces here */ -+#warning do SPI device probe here and then call wm8772_init() -+ -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8772_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8772 = { -+ .probe = wm8772_probe, -+ .remove = wm8772_remove, -+ .suspend = wm8772_suspend, -+ .resume = wm8772_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8772); -+ -+MODULE_DESCRIPTION("ASoC WM8772 driver"); -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8772.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8772.h -@@ -0,0 +1,40 @@ -+/* -+ * wm8772.h -- audio driver for WM8772 -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ */ -+ -+#ifndef _WM8772_H -+#define _WM8772_H -+ -+/* WM8772 register space */ -+ -+#define WM8772_LDAC1VOL 0x00 -+#define WM8772_RDAC1VOL 0x01 -+#define WM8772_DACCH 0x02 -+#define WM8772_IFACE 0x03 -+#define WM8772_LDAC2VOL 0x04 -+#define WM8772_RDAC2VOL 0x05 -+#define WM8772_LDAC3VOL 0x06 -+#define WM8772_RDAC3VOL 0x07 -+#define WM8772_MDACVOL 0x08 -+#define WM8772_DACCTRL 0x09 -+#define WM8772_DACRATE 0x0a -+#define WM8772_ADCRATE 0x0b -+#define WM8772_ADCCTRL 0x0c -+#define WM8772_RESET 0x1f -+ -+#define WM8772_CACHE_REGNUM 10 -+ -+extern struct snd_soc_codec_dai wm8772_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8772; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8971.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8971.c -@@ -0,0 +1,1214 @@ -+/* -+ * wm8971.c -- WM8971 ALSA SoC Audio driver -+ * -+ * Copyright 2005 Lab126, Inc. -+ * -+ * Author: Kenneth Kiraly <kiraly@lab126.com> -+ * -+ * Based on wm8753.c by Liam Girdwood -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8971.h" -+ -+#define AUDIO_NAME "wm8971" -+#define WM8971_VERSION "0.8" -+ -+#undef WM8971_DEBUG -+ -+#ifdef WM8971_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+#define WM8971_REG_COUNT 43 -+ -+static struct workqueue_struct *wm8971_workq = NULL; -+static struct work_struct wm8971_dapm_work; -+ -+/* -+ * wm8971 register cache -+ * We can't read the WM8971 register space when we -+ * are using 2 wire for device control, so we cache them instead. -+ */ -+static const u16 wm8971_reg[] = { -+ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ -+ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ -+ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ -+ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ -+ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ -+ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ -+ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ -+ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ -+ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ -+ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ -+ 0x0079, 0x0079, 0x0079, /* 40 */ -+}; -+ -+#define WM8971_HIFI_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ -+ SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8971_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8971_HIFI_FSB \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ -+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) -+ -+#define WM8971_HIFI_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -+ -+#define WM8971_HIFI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+static struct snd_soc_dai_mode wm8971_modes[] = { -+ /* common codec frame and clock master modes */ -+ /* 8k */ -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1536, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1408, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 2304, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 2112, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1500, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ -+ /* 11.025k */ -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1024, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1536, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1088, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ -+ /* 16k */ -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 768, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1152, -+ .bfs = WM8971_HIFI_FSB -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 750, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ -+ /* 22.05k */ -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 512, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 768, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 544, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ -+ /* 32k */ -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 384, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 576, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 375, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ -+ /* 44.1k & 48k */ -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 384, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 272, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 250, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ -+ /* 88.2k & 96k */ -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 128, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 192, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 136, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 125, -+ .bfs = WM8971_HIFI_FSB, -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8971_HIFI_BITS, -+ .pcmrate = WM8971_HIFI_RATES, -+ .pcmdir = WM8971_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg < WM8971_REG_COUNT) -+ return cache[reg]; -+ -+ return -1; -+} -+ -+static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg < WM8971_REG_COUNT) -+ cache[reg] = value; -+} -+ -+static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8753 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8971_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -EIO; -+} -+ -+#define wm8971_reset(c) wm8971_write(c, WM8971_RESET, 0) -+ -+/* WM8971 Controls */ -+static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; -+static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz", -+ "200Hz @ 48kHz" }; -+static const char *wm8971_treble[] = { "8kHz", "4kHz" }; -+static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" }; -+static const char *wm8971_ng_type[] = { "Constant PGA Gain", -+ "Mute ADC Output" }; -+static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; -+static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", -+ "Mono (Right)", "Digital Mono"}; -+static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" }; -+static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA", -+ "Differential"}; -+static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA", -+ "Differential"}; -+static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"}; -+static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"}; -+static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert", -+ "L + R Invert"}; -+ -+static const struct soc_enum wm8971_enum[] = { -+ SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */ -+ SOC_ENUM_SINGLE(WM8971_BASS, 6, 2, wm8971_bass_filter), -+ SOC_ENUM_SINGLE(WM8971_TREBLE, 6, 2, wm8971_treble), -+ SOC_ENUM_SINGLE(WM8971_ALC1, 7, 4, wm8971_alc_func), -+ SOC_ENUM_SINGLE(WM8971_NGATE, 1, 2, wm8971_ng_type), /* 4 */ -+ SOC_ENUM_SINGLE(WM8971_ADCDAC, 1, 4, wm8971_deemp), -+ SOC_ENUM_SINGLE(WM8971_ADCTL1, 4, 4, wm8971_mono_mux), -+ SOC_ENUM_SINGLE(WM8971_ADCTL1, 1, 2, wm8971_dac_phase), -+ SOC_ENUM_SINGLE(WM8971_LOUTM1, 0, 5, wm8971_lline_mux), /* 8 */ -+ SOC_ENUM_SINGLE(WM8971_ROUTM1, 0, 5, wm8971_rline_mux), -+ SOC_ENUM_SINGLE(WM8971_LADCIN, 6, 4, wm8971_lpga_sel), -+ SOC_ENUM_SINGLE(WM8971_RADCIN, 6, 4, wm8971_rpga_sel), -+ SOC_ENUM_SINGLE(WM8971_ADCDAC, 5, 4, wm8971_adcpol), /* 12 */ -+ SOC_ENUM_SINGLE(WM8971_ADCIN, 6, 4, wm8971_mono_mux), -+}; -+ -+static const struct snd_kcontrol_new wm8971_snd_controls[] = { -+ SOC_DOUBLE_R("Capture Volume", WM8971_LINVOL, WM8971_RINVOL, 0, 63, 0), -+ SOC_DOUBLE_R("Capture ZC Switch", WM8971_LINVOL, WM8971_RINVOL, 6, 1, 0), -+ SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1), -+ -+ SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V, -+ WM8971_ROUT1V, 7, 1, 0), -+ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V, -+ WM8971_ROUT2V, 7, 1, 0), -+ SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0), -+ -+ SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0), -+ -+ SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1, -+ WM8971_LOUTM2, 4, 7, 1), -+ SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1, -+ WM8971_ROUTM2, 4, 7, 1), -+ SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1, -+ WM8971_MOUTM2, 4, 7, 1), -+ -+ SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V, -+ WM8971_ROUT1V, 0, 127, 0), -+ SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V, -+ WM8971_ROUT2V, 0, 127, 0), -+ -+ SOC_ENUM("Bass Boost", wm8971_enum[0]), -+ SOC_ENUM("Bass Filter", wm8971_enum[1]), -+ SOC_SINGLE("Bass Volume", WM8971_BASS, 0, 7, 1), -+ -+ SOC_SINGLE("Treble Volume", WM8971_TREBLE, 0, 7, 0), -+ SOC_ENUM("Treble Cut-off", wm8971_enum[2]), -+ -+ SOC_SINGLE("Capture Filter Switch", WM8971_ADCDAC, 0, 1, 1), -+ -+ SOC_SINGLE("ALC Target Volume", WM8971_ALC1, 0, 7, 0), -+ SOC_SINGLE("ALC Max Volume", WM8971_ALC1, 4, 7, 0), -+ -+ SOC_SINGLE("ALC Capture Target Volume", WM8971_ALC1, 0, 7, 0), -+ SOC_SINGLE("ALC Capture Max Volume", WM8971_ALC1, 4, 7, 0), -+ SOC_ENUM("ALC Capture Function", wm8971_enum[3]), -+ SOC_SINGLE("ALC Capture ZC Switch", WM8971_ALC2, 7, 1, 0), -+ SOC_SINGLE("ALC Capture Hold Time", WM8971_ALC2, 0, 15, 0), -+ SOC_SINGLE("ALC Capture Decay Time", WM8971_ALC3, 4, 15, 0), -+ SOC_SINGLE("ALC Capture Attack Time", WM8971_ALC3, 0, 15, 0), -+ SOC_SINGLE("ALC Capture NG Threshold", WM8971_NGATE, 3, 31, 0), -+ SOC_ENUM("ALC Capture NG Type", wm8971_enum[4]), -+ SOC_SINGLE("ALC Capture NG Switch", WM8971_NGATE, 0, 1, 0), -+ -+ SOC_SINGLE("Capture 6dB Attenuate", WM8971_ADCDAC, 8, 1, 0), -+ SOC_SINGLE("Playback 6dB Attenuate", WM8971_ADCDAC, 7, 1, 0), -+ -+ SOC_ENUM("Playback De-emphasis", wm8971_enum[5]), -+ SOC_ENUM("Playback Function", wm8971_enum[6]), -+ SOC_ENUM("Playback Phase", wm8971_enum[7]), -+ -+ SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0), -+}; -+ -+/* add non-DAPM controls */ -+static int wm8971_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8971_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* -+ * DAPM Controls -+ */ -+ -+/* Left Mixer */ -+static const struct snd_kcontrol_new wm8971_left_mixer_controls[] = { -+SOC_DAPM_SINGLE("Playback Switch", WM8971_LOUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_LOUTM1, 7, 1, 0), -+SOC_DAPM_SINGLE("Right Playback Switch", WM8971_LOUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_LOUTM2, 7, 1, 0), -+}; -+ -+/* Right Mixer */ -+static const struct snd_kcontrol_new wm8971_right_mixer_controls[] = { -+SOC_DAPM_SINGLE("Left Playback Switch", WM8971_ROUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_ROUTM1, 7, 1, 0), -+SOC_DAPM_SINGLE("Playback Switch", WM8971_ROUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_ROUTM2, 7, 1, 0), -+}; -+ -+/* Mono Mixer */ -+static const struct snd_kcontrol_new wm8971_mono_mixer_controls[] = { -+SOC_DAPM_SINGLE("Left Playback Switch", WM8971_MOUTM1, 8, 1, 0), -+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_MOUTM1, 7, 1, 0), -+SOC_DAPM_SINGLE("Right Playback Switch", WM8971_MOUTM2, 8, 1, 0), -+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_MOUTM2, 7, 1, 0), -+}; -+ -+/* Left Line Mux */ -+static const struct snd_kcontrol_new wm8971_left_line_controls = -+SOC_DAPM_ENUM("Route", wm8971_enum[8]); -+ -+/* Right Line Mux */ -+static const struct snd_kcontrol_new wm8971_right_line_controls = -+SOC_DAPM_ENUM("Route", wm8971_enum[9]); -+ -+/* Left PGA Mux */ -+static const struct snd_kcontrol_new wm8971_left_pga_controls = -+SOC_DAPM_ENUM("Route", wm8971_enum[10]); -+ -+/* Right PGA Mux */ -+static const struct snd_kcontrol_new wm8971_right_pga_controls = -+SOC_DAPM_ENUM("Route", wm8971_enum[11]); -+ -+/* Mono ADC Mux */ -+static const struct snd_kcontrol_new wm8971_monomux_controls = -+SOC_DAPM_ENUM("Route", wm8971_enum[13]); -+ -+static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { -+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, -+ &wm8971_left_mixer_controls[0], -+ ARRAY_SIZE(wm8971_left_mixer_controls)), -+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, -+ &wm8971_right_mixer_controls[0], -+ ARRAY_SIZE(wm8971_right_mixer_controls)), -+ SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0, -+ &wm8971_mono_mixer_controls[0], -+ ARRAY_SIZE(wm8971_mono_mixer_controls)), -+ -+ SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Right Out 1", WM8971_PWR2, 5, 0, NULL, 0), -+ SND_SOC_DAPM_PGA("Left Out 1", WM8971_PWR2, 6, 0, NULL, 0), -+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8971_PWR2, 7, 0), -+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8971_PWR2, 8, 0), -+ SND_SOC_DAPM_PGA("Mono Out 1", WM8971_PWR2, 2, 0, NULL, 0), -+ -+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8971_PWR1, 1, 0), -+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8971_PWR1, 2, 0), -+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), -+ -+ SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0, -+ &wm8971_left_pga_controls), -+ SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0, -+ &wm8971_right_pga_controls), -+ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, -+ &wm8971_left_line_controls), -+ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, -+ &wm8971_right_line_controls), -+ -+ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, -+ &wm8971_monomux_controls), -+ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, -+ &wm8971_monomux_controls), -+ -+ SND_SOC_DAPM_OUTPUT("LOUT1"), -+ SND_SOC_DAPM_OUTPUT("ROUT1"), -+ SND_SOC_DAPM_OUTPUT("LOUT2"), -+ SND_SOC_DAPM_OUTPUT("ROUT2"), -+ SND_SOC_DAPM_OUTPUT("MONO"), -+ -+ SND_SOC_DAPM_INPUT("LINPUT1"), -+ SND_SOC_DAPM_INPUT("RINPUT1"), -+ SND_SOC_DAPM_INPUT("MIC"), -+}; -+ -+static const char *audio_map[][3] = { -+ /* left mixer */ -+ {"Left Mixer", "Playback Switch", "Left DAC"}, -+ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, -+ {"Left Mixer", "Right Playback Switch", "Right DAC"}, -+ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"}, -+ -+ /* right mixer */ -+ {"Right Mixer", "Left Playback Switch", "Left DAC"}, -+ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"}, -+ {"Right Mixer", "Playback Switch", "Right DAC"}, -+ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"}, -+ -+ /* left out 1 */ -+ {"Left Out 1", NULL, "Left Mixer"}, -+ {"LOUT1", NULL, "Left Out 1"}, -+ -+ /* left out 2 */ -+ {"Left Out 2", NULL, "Left Mixer"}, -+ {"LOUT2", NULL, "Left Out 2"}, -+ -+ /* right out 1 */ -+ {"Right Out 1", NULL, "Right Mixer"}, -+ {"ROUT1", NULL, "Right Out 1"}, -+ -+ /* right out 2 */ -+ {"Right Out 2", NULL, "Right Mixer"}, -+ {"ROUT2", NULL, "Right Out 2"}, -+ -+ /* mono mixer */ -+ {"Mono Mixer", "Left Playback Switch", "Left DAC"}, -+ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"}, -+ {"Mono Mixer", "Right Playback Switch", "Right DAC"}, -+ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"}, -+ -+ /* mono out */ -+ {"Mono Out", NULL, "Mono Mixer"}, -+ {"MONO1", NULL, "Mono Out"}, -+ -+ /* Left Line Mux */ -+ {"Left Line Mux", "Line", "LINPUT1"}, -+ {"Left Line Mux", "PGA", "Left PGA Mux"}, -+ {"Left Line Mux", "Differential", "Differential Mux"}, -+ -+ /* Right Line Mux */ -+ {"Right Line Mux", "Line", "RINPUT1"}, -+ {"Right Line Mux", "Mic", "MIC"}, -+ {"Right Line Mux", "PGA", "Right PGA Mux"}, -+ {"Right Line Mux", "Differential", "Differential Mux"}, -+ -+ /* Left PGA Mux */ -+ {"Left PGA Mux", "Line", "LINPUT1"}, -+ {"Left PGA Mux", "Differential", "Differential Mux"}, -+ -+ /* Right PGA Mux */ -+ {"Right PGA Mux", "Line", "RINPUT1"}, -+ {"Right PGA Mux", "Differential", "Differential Mux"}, -+ -+ /* Differential Mux */ -+ {"Differential Mux", "Line", "LINPUT1"}, -+ {"Differential Mux", "Line", "RINPUT1"}, -+ -+ /* Left ADC Mux */ -+ {"Left ADC Mux", "Stereo", "Left PGA Mux"}, -+ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"}, -+ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"}, -+ -+ /* Right ADC Mux */ -+ {"Right ADC Mux", "Stereo", "Right PGA Mux"}, -+ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"}, -+ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"}, -+ -+ /* ADC */ -+ {"Left ADC", NULL, "Left ADC Mux"}, -+ {"Right ADC", NULL, "Right ADC Mux"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8971_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8971_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8971_dapm_widgets[i]); -+ } -+ -+ /* set up audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+struct _coeff_div { -+ u32 mclk; -+ u32 rate; -+ u16 fs; -+ u8 sr:5; -+ u8 usb:1; -+}; -+ -+/* codec hifi mclk clock divider coefficients */ -+static const struct _coeff_div coeff_div[] = { -+ /* 8k */ -+ {12288000, 8000, 1536, 0x6, 0x0}, -+ {11289600, 8000, 1408, 0x16, 0x0}, -+ {18432000, 8000, 2304, 0x7, 0x0}, -+ {16934400, 8000, 2112, 0x17, 0x0}, -+ {12000000, 8000, 1500, 0x6, 0x1}, -+ -+ /* 11.025k */ -+ {11289600, 11025, 1024, 0x18, 0x0}, -+ {16934400, 11025, 1536, 0x19, 0x0}, -+ {12000000, 11025, 1088, 0x19, 0x1}, -+ -+ /* 16k */ -+ {12288000, 16000, 768, 0xa, 0x0}, -+ {18432000, 16000, 1152, 0xb, 0x0}, -+ {12000000, 16000, 750, 0xa, 0x1}, -+ -+ /* 22.05k */ -+ {11289600, 22050, 512, 0x1a, 0x0}, -+ {16934400, 22050, 768, 0x1b, 0x0}, -+ {12000000, 22050, 544, 0x1b, 0x1}, -+ -+ /* 32k */ -+ {12288000, 32000, 384, 0xc, 0x0}, -+ {18432000, 32000, 576, 0xd, 0x0}, -+ {12000000, 32000, 375, 0xa, 0x1}, -+ -+ /* 44.1k */ -+ {11289600, 44100, 256, 0x10, 0x0}, -+ {16934400, 44100, 384, 0x11, 0x0}, -+ {12000000, 44100, 272, 0x11, 0x1}, -+ -+ /* 48k */ -+ {12288000, 48000, 256, 0x0, 0x0}, -+ {18432000, 48000, 384, 0x1, 0x0}, -+ {12000000, 48000, 250, 0x0, 0x1}, -+ -+ /* 88.2k */ -+ {11289600, 88200, 128, 0x1e, 0x0}, -+ {16934400, 88200, 192, 0x1f, 0x0}, -+ {12000000, 88200, 136, 0x1f, 0x1}, -+ -+ /* 96k */ -+ {12288000, 96000, 128, 0xe, 0x0}, -+ {18432000, 96000, 192, 0xf, 0x0}, -+ {12000000, 96000, 125, 0xe, 0x1}, -+}; -+ -+static int get_coeff(int mclk, int rate) -+{ -+ int i; -+ -+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { -+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) -+ return i; -+ } -+ return -EINVAL; -+} -+ -+/* WM8971 supports numerous input clocks per sample rate */ -+static unsigned int wm8971_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ dai->mclk = 0; -+ -+ /* check that the calculated FS and rate actually match a clock from -+ * the machine driver */ -+ if (info->fs * info->rate == clk) -+ dai->mclk = clk; -+ -+ return dai->mclk; -+} -+ -+static int wm8971_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 iface = 0, bfs, srate = 0; -+ int i = get_coeff(rtd->codec_dai->mclk, -+ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate)); -+ -+ /* is coefficient valid ? */ -+ if (i < 0) -+ return i; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ iface |= 0x0040; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ iface |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ iface |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ iface |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ iface |= 0x0013; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ iface |= 0x0004; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ iface |= 0x0008; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ iface |= 0x000c; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_IF: -+ iface |= 0x0090; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ iface |= 0x0080; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ iface |= 0x0010; -+ break; -+ } -+ -+ /* set bclk divisor rate */ -+ switch (bfs) { -+ case 1: -+ break; -+ case 4: -+ srate |= (0x1 << 7); -+ break; -+ case 8: -+ srate |= (0x2 << 7); -+ break; -+ case 16: -+ srate |= (0x3 << 7); -+ break; -+ } -+ -+ /* set iface & srate */ -+ wm8971_write(codec, WM8971_AUDIO, iface); -+ wm8971_write(codec, WM8971_SRATE, srate | -+ (coeff_div[i].sr << 1) | coeff_div[i].usb); -+ return 0; -+} -+ -+static int wm8971_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7; -+ if (mute) -+ wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8); -+ else -+ wm8971_write(codec, WM8971_ADCDAC, mute_reg); -+ return 0; -+} -+ -+static int wm8971_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* set vmid to 50k and unmute dac */ -+ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ /* set vmid to 5k for quick power up */ -+ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x01c0); -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* mute dac and set vmid to 500k, enable VREF */ -+ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ wm8971_write(codec, WM8971_PWR1, 0x0001); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8971_dai = { -+ .name = "WM8971", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = wm8971_config_sysclk, -+ .digital_mute = wm8971_mute, -+ .ops = { -+ .prepare = wm8971_pcm_prepare, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8971_modes), -+ .mode = wm8971_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8971_dai); -+ -+static void wm8971_work(void *data) -+{ -+ struct snd_soc_codec *codec = (struct snd_soc_codec *)data; -+ wm8971_dapm_event(codec, codec->dapm_state); -+} -+ -+static int wm8971_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8971_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) { -+ if (i + 1 == WM8971_RESET) -+ continue; -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ -+ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ /* charge wm8971 caps */ -+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { -+ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D2); -+ codec->dapm_state = SNDRV_CTL_POWER_D0; -+ queue_delayed_work(wm8971_workq, &wm8971_dapm_work, -+ msecs_to_jiffies(1000)); -+ } -+ -+ return 0; -+} -+ -+static int wm8971_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int reg, ret = 0; -+ -+ codec->name = "WM8971"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8971_read_reg_cache; -+ codec->write = wm8971_write; -+ codec->dapm_event = wm8971_dapm_event; -+ codec->dai = &wm8971_dai; -+ codec->reg_cache_size = ARRAY_SIZE(wm8971_reg); -+ codec->num_dai = 1; -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8971_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8971_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8971_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8971_reg); -+ -+ wm8971_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if (ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* charge output caps */ -+ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D2); -+ codec->dapm_state = SNDRV_CTL_POWER_D3hot; -+ queue_delayed_work(wm8971_workq, &wm8971_dapm_work, -+ msecs_to_jiffies(1000)); -+ -+ /* set the update bits */ -+ reg = wm8971_read_reg_cache(codec, WM8971_LDAC); -+ wm8971_write(codec, WM8971_LDAC, reg | 0x0100); -+ reg = wm8971_read_reg_cache(codec, WM8971_RDAC); -+ wm8971_write(codec, WM8971_RDAC, reg | 0x0100); -+ -+ reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V); -+ wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100); -+ reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V); -+ wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100); -+ -+ reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V); -+ wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100); -+ reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V); -+ wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100); -+ -+ reg = wm8971_read_reg_cache(codec, WM8971_LINVOL); -+ wm8971_write(codec, WM8971_LINVOL, reg | 0x0100); -+ reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); -+ wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); -+ -+ wm8971_add_controls(codec); -+ wm8971_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+static struct snd_soc_device *wm8971_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8731 2 wire address is determined by GPIO5 -+ * state during powerup. -+ * low = 0x1a -+ * high = 0x1b -+ */ -+#define I2C_DRIVERID_WM8971 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8971_i2c_driver; -+static struct i2c_client client_template; -+ -+static int wm8971_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8971_socdev; -+ struct wm8971_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL) { -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ -+ i2c_set_clientdata(i2c, codec); -+ -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if (ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8971_init(socdev); -+ if (ret < 0) { -+ err("failed to initialise WM8971\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+} -+ -+static int wm8971_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec* codec = i2c_get_clientdata(client); -+ i2c_detach_client(client); -+ kfree(codec->reg_cache); -+ kfree(client); -+ return 0; -+} -+ -+static int wm8971_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8971_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8971_i2c_driver = { -+ .driver = { -+ .name = "WM8971 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8971, -+ .attach_adapter = wm8971_i2c_attach, -+ .detach_client = wm8971_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8971", -+ .driver = &wm8971_i2c_driver, -+}; -+#endif -+ -+static int wm8971_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8971_setup_data *setup; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ info("WM8971 Audio Codec %s", WM8971_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ wm8971_socdev = socdev; -+ -+ INIT_WORK(&wm8971_dapm_work, wm8971_work, codec); -+ wm8971_workq = create_workqueue("wm8971"); -+ if (wm8971_workq == NULL) { -+ kfree(codec); -+ return -ENOMEM; -+ } -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8971_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8971_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ if (wm8971_workq) -+ destroy_workqueue(wm8971_workq); -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8971_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8971 = { -+ .probe = wm8971_probe, -+ .remove = wm8971_remove, -+ .suspend = wm8971_suspend, -+ .resume = wm8971_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971); -+ -+MODULE_DESCRIPTION("ASoC WM8971 driver"); -+MODULE_AUTHOR("Lab126"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8971.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8971.h -@@ -0,0 +1,61 @@ -+/* -+ * wm8971.h -- audio driver for WM8971 -+ * -+ * Copyright 2005 Lab126, Inc. -+ * -+ * Author: Kenneth Kiraly <kiraly@lab126.com> -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ */ -+ -+#ifndef _WM8971_H -+#define _WM8971_H -+ -+#define WM8971_LINVOL 0x00 -+#define WM8971_RINVOL 0x01 -+#define WM8971_LOUT1V 0x02 -+#define WM8971_ROUT1V 0x03 -+#define WM8971_ADCDAC 0x05 -+#define WM8971_AUDIO 0x07 -+#define WM8971_SRATE 0x08 -+#define WM8971_LDAC 0x0a -+#define WM8971_RDAC 0x0b -+#define WM8971_BASS 0x0c -+#define WM8971_TREBLE 0x0d -+#define WM8971_RESET 0x0f -+#define WM8971_ALC1 0x11 -+#define WM8971_ALC2 0x12 -+#define WM8971_ALC3 0x13 -+#define WM8971_NGATE 0x14 -+#define WM8971_LADC 0x15 -+#define WM8971_RADC 0x16 -+#define WM8971_ADCTL1 0x17 -+#define WM8971_ADCTL2 0x18 -+#define WM8971_PWR1 0x19 -+#define WM8971_PWR2 0x1a -+#define WM8971_ADCTL3 0x1b -+#define WM8971_ADCIN 0x1f -+#define WM8971_LADCIN 0x20 -+#define WM8971_RADCIN 0x21 -+#define WM8971_LOUTM1 0x22 -+#define WM8971_LOUTM2 0x23 -+#define WM8971_ROUTM1 0x24 -+#define WM8971_ROUTM2 0x25 -+#define WM8971_MOUTM1 0x26 -+#define WM8971_MOUTM2 0x27 -+#define WM8971_LOUT2V 0x28 -+#define WM8971_ROUT2V 0x29 -+#define WM8971_MOUTV 0x2A -+ -+struct wm8971_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai wm8971_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8971; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8974.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8974.c -@@ -0,0 +1,935 @@ -+/* -+ * wm8974.c -- WM8974 ALSA Soc Audio driver -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * -+ * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/version.h> -+#include <linux/kernel.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8974.h" -+ -+#define AUDIO_NAME "wm8974" -+#define WM8974_VERSION "0.5" -+ -+/* -+ * Debug -+ */ -+ -+#define WM8974_DEBUG 0 -+ -+#ifdef WM8974_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+struct snd_soc_codec_device soc_codec_dev_wm8974; -+ -+/* -+ * wm8974 register cache -+ * We can't read the WM8974 register space when we are -+ * using 2 wire for device control, so we cache them instead. -+ */ -+static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0050, 0x0000, 0x0140, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x00ff, -+ 0x0000, 0x0000, 0x0100, 0x00ff, -+ 0x0000, 0x0000, 0x012c, 0x002c, -+ 0x002c, 0x002c, 0x002c, 0x0000, -+ 0x0032, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0038, 0x000b, 0x0032, 0x0000, -+ 0x0008, 0x000c, 0x0093, 0x00e9, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0003, 0x0010, 0x0000, 0x0000, -+ 0x0000, 0x0002, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0039, 0x0000, -+ 0x0000, -+}; -+ -+#define WM8974_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ -+ SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8974_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8974_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+#define WM8794_BCLK \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\ -+ SND_SOC_FSBD(16) | SND_SOC_FSBD(32)) -+ -+#define WM8794_HIFI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+static struct snd_soc_dai_mode wm8974_modes[] = { -+ /* codec frame and clock master modes */ -+ { -+ .fmt = WM8974_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8794_HIFI_BITS, -+ .pcmrate = WM8974_RATES, -+ .pcmdir = WM8974_DIR, -+ .fs = 256, -+ .bfs = WM8794_BCLK, -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8974_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8794_HIFI_BITS, -+ .pcmrate = WM8974_RATES, -+ .pcmdir = WM8974_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8974 register cache -+ */ -+static inline unsigned int wm8974_read_reg_cache(struct snd_soc_codec * codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg == WM8974_RESET) -+ return 0; -+ if (reg >= WM8974_CACHEREGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write wm8974 register cache -+ */ -+static inline void wm8974_write_reg_cache(struct snd_soc_codec *codec, -+ u16 reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= WM8974_CACHEREGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+/* -+ * write to the WM8974 register space -+ */ -+static int wm8974_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8974 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8974_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -EIO; -+} -+ -+#define wm8974_reset(c) wm8974_write(c, WM8974_RESET, 0) -+ -+static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" }; -+static const char *wm8974_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; -+static const char *wm8974_eqmode[] = {"Capture", "Playback" }; -+static const char *wm8974_bw[] = {"Narrow", "Wide" }; -+static const char *wm8974_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" }; -+static const char *wm8974_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" }; -+static const char *wm8974_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" }; -+static const char *wm8974_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; -+static const char *wm8974_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; -+static const char *wm8974_alc[] = {"ALC", "Limiter" }; -+ -+static const struct soc_enum wm8974_enum[] = { -+ SOC_ENUM_SINGLE(WM8974_COMP, 1, 4, wm8974_companding), /* adc */ -+ SOC_ENUM_SINGLE(WM8974_COMP, 3, 4, wm8974_companding), /* dac */ -+ SOC_ENUM_SINGLE(WM8974_DAC, 4, 4, wm8974_deemp), -+ SOC_ENUM_SINGLE(WM8974_EQ1, 8, 2, wm8974_eqmode), -+ -+ SOC_ENUM_SINGLE(WM8974_EQ1, 5, 4, wm8974_eq1), -+ SOC_ENUM_SINGLE(WM8974_EQ2, 8, 2, wm8974_bw), -+ SOC_ENUM_SINGLE(WM8974_EQ2, 5, 4, wm8974_eq2), -+ SOC_ENUM_SINGLE(WM8974_EQ3, 8, 2, wm8974_bw), -+ -+ SOC_ENUM_SINGLE(WM8974_EQ3, 5, 4, wm8974_eq3), -+ SOC_ENUM_SINGLE(WM8974_EQ4, 8, 2, wm8974_bw), -+ SOC_ENUM_SINGLE(WM8974_EQ4, 5, 4, wm8974_eq4), -+ SOC_ENUM_SINGLE(WM8974_EQ5, 8, 2, wm8974_bw), -+ -+ SOC_ENUM_SINGLE(WM8974_EQ5, 5, 4, wm8974_eq5), -+ SOC_ENUM_SINGLE(WM8974_ALC3, 8, 2, wm8974_alc), -+}; -+ -+static const struct snd_kcontrol_new wm8974_snd_controls[] = { -+ -+SOC_SINGLE("Digital Loopback Switch", WM8974_COMP, 0, 1, 0), -+ -+SOC_ENUM("DAC Companding", wm8974_enum[1]), -+SOC_ENUM("ADC Companding", wm8974_enum[0]), -+ -+SOC_ENUM("Playback De-emphasis", wm8974_enum[2]), -+SOC_SINGLE("DAC Inversion Switch", WM8974_DAC, 0, 1, 0), -+ -+SOC_SINGLE("PCM Volume", WM8974_DACVOL, 0, 127, 0), -+ -+SOC_SINGLE("High Pass Filter Switch", WM8974_ADC, 8, 1, 0), -+SOC_SINGLE("High Pass Cut Off", WM8974_ADC, 4, 7, 0), -+SOC_SINGLE("ADC Inversion Switch", WM8974_COMP, 0, 1, 0), -+ -+SOC_SINGLE("Capture Volume", WM8974_ADCVOL, 0, 127, 0), -+ -+SOC_ENUM("Equaliser Function", wm8974_enum[3]), -+SOC_ENUM("EQ1 Cut Off", wm8974_enum[4]), -+SOC_SINGLE("EQ1 Volume", WM8974_EQ1, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ2 Bandwith", wm8974_enum[5]), -+SOC_ENUM("EQ2 Cut Off", wm8974_enum[6]), -+SOC_SINGLE("EQ2 Volume", WM8974_EQ2, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ3 Bandwith", wm8974_enum[7]), -+SOC_ENUM("EQ3 Cut Off", wm8974_enum[8]), -+SOC_SINGLE("EQ3 Volume", WM8974_EQ3, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ4 Bandwith", wm8974_enum[9]), -+SOC_ENUM("EQ4 Cut Off", wm8974_enum[10]), -+SOC_SINGLE("EQ4 Volume", WM8974_EQ4, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ5 Bandwith", wm8974_enum[11]), -+SOC_ENUM("EQ5 Cut Off", wm8974_enum[12]), -+SOC_SINGLE("EQ5 Volume", WM8974_EQ5, 0, 31, 1), -+ -+SOC_SINGLE("DAC Playback Limiter Switch", WM8974_DACLIM1, 8, 1, 0), -+SOC_SINGLE("DAC Playback Limiter Decay", WM8974_DACLIM1, 4, 15, 0), -+SOC_SINGLE("DAC Playback Limiter Attack", WM8974_DACLIM1, 0, 15, 0), -+ -+SOC_SINGLE("DAC Playback Limiter Threshold", WM8974_DACLIM2, 4, 7, 0), -+SOC_SINGLE("DAC Playback Limiter Boost", WM8974_DACLIM2, 0, 15, 0), -+ -+SOC_SINGLE("ALC Enable Switch", WM8974_ALC1, 8, 1, 0), -+SOC_SINGLE("ALC Capture Max Gain", WM8974_ALC1, 3, 7, 0), -+SOC_SINGLE("ALC Capture Min Gain", WM8974_ALC1, 0, 7, 0), -+ -+SOC_SINGLE("ALC Capture ZC Switch", WM8974_ALC2, 8, 1, 0), -+SOC_SINGLE("ALC Capture Hold", WM8974_ALC2, 4, 7, 0), -+SOC_SINGLE("ALC Capture Target", WM8974_ALC2, 0, 15, 0), -+ -+SOC_ENUM("ALC Capture Mode", wm8974_enum[13]), -+SOC_SINGLE("ALC Capture Decay", WM8974_ALC3, 4, 15, 0), -+SOC_SINGLE("ALC Capture Attack", WM8974_ALC3, 0, 15, 0), -+ -+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8974_NGATE, 3, 1, 0), -+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8974_NGATE, 0, 7, 0), -+ -+SOC_SINGLE("Capture PGA ZC Switch", WM8974_INPPGA, 7, 1, 0), -+SOC_SINGLE("Capture PGA Volume", WM8974_INPPGA, 0, 63, 0), -+ -+SOC_SINGLE("Speaker Playback ZC Switch", WM8974_SPKVOL, 7, 1, 0), -+SOC_SINGLE("Speaker Playback Switch", WM8974_SPKVOL, 6, 1, 1), -+SOC_SINGLE("Speaker Playback Volume", WM8974_SPKVOL, 0, 63, 0), -+ -+SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), -+SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 0), -+}; -+ -+/* add non dapm controls */ -+static int wm8974_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8974_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8974_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* Speaker Output Mixer */ -+static const struct snd_kcontrol_new wm8974_speaker_mixer_controls[] = { -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_SPKMIX, 1, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_SPKMIX, 5, 1, 0), -+SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 1), -+}; -+ -+/* Mono Output Mixer */ -+static const struct snd_kcontrol_new wm8974_mono_mixer_controls[] = { -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_MONOMIX, 1, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_MONOMIX, 2, 1, 0), -+SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 1), -+}; -+ -+/* AUX Input boost vol */ -+static const struct snd_kcontrol_new wm8974_aux_boost_controls = -+SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0); -+ -+/* Mic Input boost vol */ -+static const struct snd_kcontrol_new wm8974_mic_boost_controls = -+SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0); -+ -+/* Capture boost switch */ -+static const struct snd_kcontrol_new wm8974_capture_boost_controls = -+SOC_DAPM_SINGLE("Capture Boost Switch", WM8974_INPPGA, 6, 1, 0); -+ -+/* Aux In to PGA */ -+static const struct snd_kcontrol_new wm8974_aux_capture_boost_controls = -+SOC_DAPM_SINGLE("Aux Capture Boost Switch", WM8974_INPPGA, 2, 1, 0); -+ -+/* Mic P In to PGA */ -+static const struct snd_kcontrol_new wm8974_micp_capture_boost_controls = -+SOC_DAPM_SINGLE("Mic P Capture Boost Switch", WM8974_INPPGA, 0, 1, 0); -+ -+/* Mic N In to PGA */ -+static const struct snd_kcontrol_new wm8974_micn_capture_boost_controls = -+SOC_DAPM_SINGLE("Mic N Capture Boost Switch", WM8974_INPPGA, 1, 1, 0); -+ -+static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = { -+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0, -+ &wm8974_speaker_mixer_controls[0], -+ ARRAY_SIZE(wm8974_speaker_mixer_controls)), -+SND_SOC_DAPM_MIXER("Mono Mixer", WM8974_POWER3, 3, 0, -+ &wm8974_mono_mixer_controls[0], -+ ARRAY_SIZE(wm8974_mono_mixer_controls)), -+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8974_POWER3, 0, 0), -+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8974_POWER3, 0, 0), -+SND_SOC_DAPM_PGA("Aux Input", WM8974_POWER1, 6, 0, NULL, 0), -+SND_SOC_DAPM_PGA("SpkN Out", WM8974_POWER3, 5, 0, NULL, 0), -+SND_SOC_DAPM_PGA("SpkP Out", WM8974_POWER3, 6, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mono Out", WM8974_POWER3, 7, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mic PGA", WM8974_POWER2, 2, 0, NULL, 0), -+ -+SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, -+ &wm8974_aux_boost_controls, 1), -+SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, -+ &wm8974_mic_boost_controls, 1), -+SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, -+ &wm8974_capture_boost_controls), -+ -+SND_SOC_DAPM_MIXER("Boost Mixer", WM8974_POWER2, 4, 0, NULL, 0), -+ -+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8974_POWER1, 4, 0), -+ -+SND_SOC_DAPM_INPUT("MICN"), -+SND_SOC_DAPM_INPUT("MICP"), -+SND_SOC_DAPM_INPUT("AUX"), -+SND_SOC_DAPM_OUTPUT("MONOOUT"), -+SND_SOC_DAPM_OUTPUT("SPKOUTP"), -+SND_SOC_DAPM_OUTPUT("SPKOUTN"), -+}; -+ -+static const char *audio_map[][3] = { -+ /* Mono output mixer */ -+ {"Mono Mixer", "PCM Playback Switch", "DAC"}, -+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, -+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, -+ -+ /* Speaker output mixer */ -+ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, -+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, -+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, -+ -+ /* Outputs */ -+ {"Mono Out", NULL, "Mono Mixer"}, -+ {"MONOOUT", NULL, "Mono Out"}, -+ {"SpkN Out", NULL, "Speaker Mixer"}, -+ {"SpkP Out", NULL, "Speaker Mixer"}, -+ {"SPKOUTN", NULL, "SpkN Out"}, -+ {"SPKOUTP", NULL, "SpkP Out"}, -+ -+ /* Boost Mixer */ -+ {"Boost Mixer", NULL, "ADC"}, -+ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, -+ {"Aux Boost", "Aux Volume", "Boost Mixer"}, -+ {"Capture Boost", "Capture Switch", "Boost Mixer"}, -+ {"Mic Boost", "Mic Volume", "Boost Mixer"}, -+ -+ /* Inputs */ -+ {"MICP", NULL, "Mic Boost"}, -+ {"MICN", NULL, "Mic PGA"}, -+ {"Mic PGA", NULL, "Capture Boost"}, -+ {"AUX", NULL, "Aux Input"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8974_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8974_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8974_dapm_widgets[i]); -+ } -+ -+ /* set up audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+struct pll_ { -+ unsigned int in_hz, out_hz; -+ unsigned int pre:4; /* prescale - 1 */ -+ unsigned int n:4; -+ unsigned int k; -+}; -+ -+struct pll_ pll[] = { -+ {12000000, 11289600, 0, 7, 0x86c220}, -+ {12000000, 12288000, 0, 8, 0x3126e8}, -+ {13000000, 11289600, 0, 6, 0xf28bd4}, -+ {13000000, 12288000, 0, 7, 0x8fd525}, -+ {12288000, 11289600, 0, 7, 0x59999a}, -+ {11289600, 12288000, 0, 8, 0x80dee9}, -+ /* liam - add more entries */ -+}; -+ -+static int set_pll(struct snd_soc_codec *codec, unsigned int in, -+ unsigned int out) -+{ -+ int i; -+ u16 reg; -+ -+ if(out == 0) { -+ reg = wm8974_read_reg_cache(codec, WM8974_POWER1); -+ wm8974_write(codec, WM8974_POWER1, reg & 0x1df); -+ return 0; -+ } -+ -+ for(i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (in == pll[i].in_hz && out == pll[i].out_hz) { -+ wm8974_write(codec, WM8974_PLLN, (pll[i].pre << 4) | pll[i].n); -+ wm8974_write(codec, WM8974_PLLK1, pll[i].k >> 18); -+ wm8974_write(codec, WM8974_PLLK1, (pll[i].k >> 9) && 0x1ff); -+ wm8974_write(codec, WM8974_PLLK1, pll[i].k && 0x1ff); -+ reg = wm8974_read_reg_cache(codec, WM8974_POWER1); -+ wm8974_write(codec, WM8974_POWER1, reg | 0x020); -+ return 0; -+ } -+ } -+ return -EINVAL; -+} -+ -+/* mclk dividers * 2 */ -+static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24}; -+ -+/* we need 256FS to drive the DAC's and ADC's */ -+static unsigned int wm8974_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ int i, j, best_clk = info->fs * info->rate; -+ -+ /* can we run at this clk without the PLL ? */ -+ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) { -+ if ((best_clk >> 1) * mclk_div[i] == clk) { -+ dai->pll_in = 0; -+ dai->clk_div = mclk_div[i]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ -+ /* now check for PLL support */ -+ for (i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (pll[i].in_hz == clk) { -+ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) { -+ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) { -+ dai->pll_in = clk; -+ dai->pll_out = pll[i].out_hz; -+ dai->clk_div = mclk_div[j]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+} -+ -+static int wm8974_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_codec_dai *dai = rtd->codec_dai; -+ u16 iface = 0, bfs, clk = 0, adn; -+ int fs = 48000 << 7, i; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ switch (bfs) { -+ case 2: -+ clk |= 0x1 << 2; -+ break; -+ case 4: -+ clk |= 0x2 << 2; -+ break; -+ case 8: -+ clk |= 0x3 << 2; -+ break; -+ case 16: -+ clk |= 0x4 << 2; -+ break; -+ case 32: -+ clk |= 0x5 << 2; -+ break; -+ } -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ clk |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ iface |= 0x0010; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ iface |= 0x0008; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ iface |= 0x00018; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ iface |= 0x0020; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ iface |= 0x0040; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ iface |= 0x0060; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_IF: -+ iface |= 0x0180; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ iface |= 0x0100; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ iface |= 0x0080; -+ break; -+ } -+ -+ /* filter coefficient */ -+ adn = wm8974_read_reg_cache(codec, WM8974_ADD) & 0x1f1; -+ switch (rtd->codec_dai->dai_runtime.pcmrate) { -+ case SNDRV_PCM_RATE_8000: -+ adn |= 0x5 << 1; -+ fs = 8000 << 7; -+ break; -+ case SNDRV_PCM_RATE_11025: -+ adn |= 0x4 << 1; -+ fs = 11025 << 7; -+ break; -+ case SNDRV_PCM_RATE_16000: -+ adn |= 0x3 << 1; -+ fs = 16000 << 7; -+ break; -+ case SNDRV_PCM_RATE_22050: -+ adn |= 0x2 << 1; -+ fs = 22050 << 7; -+ break; -+ case SNDRV_PCM_RATE_32000: -+ adn |= 0x1 << 1; -+ fs = 32000 << 7; -+ break; -+ case SNDRV_PCM_RATE_44100: -+ fs = 44100 << 7; -+ break; -+ } -+ -+ /* do we need to enable the PLL */ -+ if(dai->pll_in) -+ set_pll(codec, dai->pll_in, dai->pll_out); -+ -+ /* divide the clock to 256 fs */ -+ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) { -+ if (dai->clk_div == mclk_div[i]) { -+ clk |= i << 5; -+ clk &= 0xff; -+ goto set; -+ } -+ } -+ -+set: -+ /* set iface */ -+ wm8974_write(codec, WM8974_IFACE, iface); -+ wm8974_write(codec, WM8974_CLOCK, clk); -+ -+ return 0; -+} -+ -+static int wm8974_hw_free(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ set_pll(codec, 0, 0); -+ return 0; -+} -+ -+static int wm8974_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; -+ if(mute) -+ wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); -+ else -+ wm8974_write(codec, WM8974_DAC, mute_reg); -+ return 0; -+} -+ -+/* liam need to make this lower power with dapm */ -+static int wm8974_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, clk and osc on, dac unmute, active */ -+ wm8974_write(codec, WM8974_POWER1, 0x1ff); -+ wm8974_write(codec, WM8974_POWER2, 0x1ff); -+ wm8974_write(codec, WM8974_POWER3, 0x1ff); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, dac mute, inactive */ -+ -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, dac mute, inactive */ -+ wm8974_write(codec, WM8974_POWER1, 0x0); -+ wm8974_write(codec, WM8974_POWER2, 0x0); -+ wm8974_write(codec, WM8974_POWER3, 0x0); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8974_dai = { -+ .name = "WM8974 HiFi", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 1, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 1, -+ }, -+ .config_sysclk = wm8974_config_sysclk, -+ .digital_mute = wm8974_mute, -+ .ops = { -+ .prepare = wm8974_pcm_prepare, -+ .hw_free = wm8974_hw_free, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8974_modes), -+ .mode = wm8974_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8974_dai); -+ -+static int wm8974_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8974_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8974_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the WM8974 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8974_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int ret = 0; -+ -+ codec->name = "WM8974"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8974_read_reg_cache; -+ codec->write = wm8974_write; -+ codec->dapm_event = wm8974_dapm_event; -+ codec->dai = &wm8974_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(wm8974_reg); -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8974_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8974_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8974_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8974_reg); -+ -+ wm8974_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if(ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8974_add_controls(codec); -+ wm8974_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if(ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *wm8974_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8974 2 wire address is 0x1a -+ */ -+#define I2C_DRIVERID_WM8974 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8974_i2c_driver; -+static struct i2c_client client_template; -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+ -+static int wm8974_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8974_socdev; -+ struct wm8974_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL) { -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ i2c_set_clientdata(i2c, codec); -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if(ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8974_init(socdev); -+ if(ret < 0) { -+ err("failed to initialise WM8974\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+} -+ -+static int wm8974_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec *codec = i2c_get_clientdata(client); -+ i2c_detach_client(client); -+ kfree(codec->reg_cache); -+ kfree(client); -+ return 0; -+} -+ -+static int wm8974_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8974_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8974_i2c_driver = { -+ .driver = { -+ .name = "WM8974 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8974, -+ .attach_adapter = wm8974_i2c_attach, -+ .detach_client = wm8974_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8974", -+ .driver = &wm8974_i2c_driver, -+}; -+#endif -+ -+static int wm8974_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8974_setup_data *setup; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ info("WM8974 Audio Codec %s", WM8974_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ wm8974_socdev = socdev; -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8974_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8974_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8974_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8974 = { -+ .probe = wm8974_probe, -+ .remove = wm8974_remove, -+ .suspend = wm8974_suspend, -+ .resume = wm8974_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8974); -+ -+MODULE_DESCRIPTION("ASoC WM8974 driver"); -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8974.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8974.h -@@ -0,0 +1,64 @@ -+/* -+ * wm8974.h -- WM8974 Soc Audio driver -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _WM8974_H -+#define _WM8974_H -+ -+/* WM8974 register space */ -+ -+#define WM8974_RESET 0x0 -+#define WM8974_POWER1 0x1 -+#define WM8974_POWER2 0x2 -+#define WM8974_POWER3 0x3 -+#define WM8974_IFACE 0x4 -+#define WM8974_COMP 0x5 -+#define WM8974_CLOCK 0x6 -+#define WM8974_ADD 0x7 -+#define WM8974_GPIO 0x8 -+#define WM8974_DAC 0xa -+#define WM8974_DACVOL 0xb -+#define WM8974_ADC 0xe -+#define WM8974_ADCVOL 0xf -+#define WM8974_EQ1 0x12 -+#define WM8974_EQ2 0x13 -+#define WM8974_EQ3 0x14 -+#define WM8974_EQ4 0x15 -+#define WM8974_EQ5 0x16 -+#define WM8974_DACLIM1 0x18 -+#define WM8974_DACLIM2 0x19 -+#define WM8974_NOTCH1 0x1b -+#define WM8974_NOTCH2 0x1c -+#define WM8974_NOTCH3 0x1d -+#define WM8974_NOTCH4 0x1e -+#define WM8974_ALC1 0x20 -+#define WM8974_ALC2 0x21 -+#define WM8974_ALC3 0x22 -+#define WM8974_NGATE 0x23 -+#define WM8974_PLLN 0x24 -+#define WM8974_PLLK1 0x25 -+#define WM8974_PLLK2 0x26 -+#define WM8974_PLLK3 0x27 -+#define WM8974_ATTEN 0x28 -+#define WM8974_INPUT 0x2c -+#define WM8974_INPPGA 0x2d -+#define WM8974_ADCBOOST 0x2f -+#define WM8974_OUTPUT 0x31 -+#define WM8974_SPKMIX 0x32 -+#define WM8974_SPKVOL 0x36 -+#define WM8974_MONOMIX 0x38 -+ -+#define WM8974_CACHEREGNUM 57 -+ -+struct wm8974_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai wm8974_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8974; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/wm9712.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm9712.c -@@ -0,0 +1,781 @@ -+/* -+ * wm9712.c -- ALSA Soc WM9712 codec support -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 4th Feb 2006 Initial version. -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/version.h> -+#include <linux/kernel.h> -+#include <linux/device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/ac97_codec.h> -+#include <sound/initval.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#define WM9712_VERSION "0.4" -+ -+static unsigned int ac97_read(struct snd_soc_codec *codec, -+ unsigned int reg); -+static int ac97_write(struct snd_soc_codec *codec, -+ unsigned int reg, unsigned int val); -+ -+#define AC97_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define AC97_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+/* may need to expand this */ -+static struct snd_soc_dai_mode ac97_modes[] = { -+ { -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE, -+ .pcmrate = AC97_RATES, -+ .pcmdir = AC97_DIR, -+ }, -+}; -+ -+/* -+ * WM9712 register cache -+ */ -+static const u16 wm9712_reg[] = { -+ 0x6174, 0x8000, 0x8000, 0x8000, // 6 -+ 0xf0f0, 0xaaa0, 0xc008, 0x6808, // e -+ 0xe808, 0xaaa0, 0xad00, 0x8000, // 16 -+ 0xe808, 0x3000, 0x8000, 0x0000, // 1e -+ 0x0000, 0x0000, 0x0000, 0x000f, // 26 -+ 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e -+ 0x0000, 0xbb80, 0x0000, 0x0000, // 36 -+ 0x0000, 0x2000, 0x0000, 0x0000, // 3e -+ 0x0000, 0x0000, 0x0000, 0x0000, // 46 -+ 0x0000, 0x0000, 0xf83e, 0xffff, // 4e -+ 0x0000, 0x0000, 0x0000, 0xf83e, // 56 -+ 0x0008, 0x0000, 0x0000, 0x0000, // 5e -+ 0xb032, 0x3e00, 0x0000, 0x0000, // 66 -+ 0x0000, 0x0000, 0x0000, 0x0000, // 6e -+ 0x0000, 0x0000, 0x0000, 0x0006, // 76 -+ 0x0001, 0x0000, 0x574d, 0x4c12, // 7e -+ 0x0000, 0x0000 // virtual hp mixers -+}; -+ -+/* virtual HP mixers regs */ -+#define HPL_MIXER 0x80 -+#define HPR_MIXER 0x82 -+ -+static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"}; -+static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"}; -+static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right", -+ "Mono"}; -+static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"}; -+static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"}; -+static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"}; -+static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; -+static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2", -+ "Stereo"}; -+static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer", -+ "Line", "Headphone Mixer", "Phone Mixer", "Phone"}; -+static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"}; -+static const char *wm9712_diff_sel[] = {"Mic", "Line"}; -+ -+static const struct soc_enum wm9712_enum[] = { -+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select), -+SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux), -+SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src), -+SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src), -+SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc), -+SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base), -+SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain), -+SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic), -+SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel), -+SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel), -+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type), -+SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel), -+}; -+ -+static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = { -+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), -+SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1), -+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), -+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1), -+ -+SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), -+SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0), -+SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0), -+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0), -+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0), -+ -+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), -+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), -+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0), -+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), -+SOC_ENUM("ALC Function", wm9712_enum[0]), -+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), -+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1), -+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), -+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), -+SOC_ENUM("ALC NG Type", wm9712_enum[10]), -+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1), -+ -+SOC_SINGLE("Mic Headphone Volume", AC97_VIDEO, 12, 7, 1), -+SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1), -+ -+SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1), -+SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1), -+SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1), -+ -+SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1), -+SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1), -+SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1), -+ -+SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1), -+SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1), -+SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1), -+ -+SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 0), -+SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1), -+ -+SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0), -+SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1), -+ -+SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1), -+SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1), -+SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0), -+ -+SOC_ENUM("Bass Control", wm9712_enum[5]), -+SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), -+SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), -+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), -+SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0), -+SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0), -+ -+SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), -+SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), -+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), -+SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), -+ -+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), -+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), -+SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), -+}; -+ -+/* add non dapm controls */ -+static int wm9712_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ return 0; -+} -+ -+/* We have to create a fake left and right HP mixers because -+ * the codec only has a single control that is shared by both channels. -+ * This makes it impossible to determine the audio path. -+ */ -+static int mixer_event (struct snd_soc_dapm_widget *w, int event) -+{ -+ u16 l, r, beep, line, phone, mic, pcm, aux; -+ -+ l = ac97_read(w->codec, HPL_MIXER); -+ r = ac97_read(w->codec, HPR_MIXER); -+ beep = ac97_read(w->codec, AC97_PC_BEEP); -+ mic = ac97_read(w->codec, AC97_VIDEO); -+ phone = ac97_read(w->codec, AC97_PHONE); -+ line = ac97_read(w->codec, AC97_LINE); -+ pcm = ac97_read(w->codec, AC97_PCM); -+ aux = ac97_read(w->codec, AC97_CD); -+ -+ if (l & 0x1 || r & 0x1) -+ ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_VIDEO, mic | 0x8000); -+ -+ if (l & 0x2 || r & 0x2) -+ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_PCM, pcm | 0x8000); -+ -+ if (l & 0x4 || r & 0x4) -+ ac97_write(w->codec, AC97_LINE, line & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_LINE, line | 0x8000); -+ -+ if (l & 0x8 || r & 0x8) -+ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_PHONE, phone | 0x8000); -+ -+ if (l & 0x10 || r & 0x10) -+ ac97_write(w->codec, AC97_CD, aux & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_CD, aux | 0x8000); -+ -+ if (l & 0x20 || r & 0x20) -+ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); -+ -+ return 0; -+} -+ -+/* Left Headphone Mixers */ -+static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = { -+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0), -+ SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0), -+ SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0), -+ SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0), -+ SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0), -+ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0), -+}; -+ -+/* Right Headphone Mixers */ -+static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = { -+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0), -+ SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0), -+ SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0), -+ SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0), -+ SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0), -+ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0), -+}; -+ -+/* Speaker Mixer */ -+static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = { -+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1), -+ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1), -+ SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1), -+ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1), -+ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1), -+}; -+ -+/* Phone Mixer */ -+static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = { -+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1), -+ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1), -+ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1), -+ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1), -+ SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1), -+ SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1), -+}; -+ -+/* ALC headphone mux */ -+static const struct snd_kcontrol_new wm9712_alc_mux_controls = -+SOC_DAPM_ENUM("Route", wm9712_enum[1]); -+ -+/* out 3 mux */ -+static const struct snd_kcontrol_new wm9712_out3_mux_controls = -+SOC_DAPM_ENUM("Route", wm9712_enum[2]); -+ -+/* spk mux */ -+static const struct snd_kcontrol_new wm9712_spk_mux_controls = -+SOC_DAPM_ENUM("Route", wm9712_enum[3]); -+ -+/* Capture to Phone mux */ -+static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls = -+SOC_DAPM_ENUM("Route", wm9712_enum[4]); -+ -+/* Capture left select */ -+static const struct snd_kcontrol_new wm9712_capture_selectl_controls = -+SOC_DAPM_ENUM("Route", wm9712_enum[8]); -+ -+/* Capture right select */ -+static const struct snd_kcontrol_new wm9712_capture_selectr_controls = -+SOC_DAPM_ENUM("Route", wm9712_enum[9]); -+ -+/* Mic select */ -+static const struct snd_kcontrol_new wm9712_mic_src_controls = -+SOC_DAPM_ENUM("Route", wm9712_enum[7]); -+ -+/* diff select */ -+static const struct snd_kcontrol_new wm9712_diff_sel_controls = -+SOC_DAPM_ENUM("Route", wm9712_enum[11]); -+ -+static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = { -+SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0, -+ &wm9712_alc_mux_controls), -+SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, -+ &wm9712_out3_mux_controls), -+SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0, -+ &wm9712_spk_mux_controls), -+SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0, -+ &wm9712_capture_phone_mux_controls), -+SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, -+ &wm9712_capture_selectl_controls), -+SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, -+ &wm9712_capture_selectr_controls), -+SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, -+ &wm9712_mic_src_controls), -+SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, -+ &wm9712_diff_sel_controls), -+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1, -+ &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls), -+ mixer_event, SND_SOC_DAPM_POST_REG), -+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1, -+ &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls), -+ mixer_event, SND_SOC_DAPM_POST_REG), -+SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1, -+ &wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)), -+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1, -+ &wm9712_speaker_mixer_controls[0], -+ ARRAY_SIZE(wm9712_speaker_mixer_controls)), -+SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1), -+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1), -+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0), -+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1), -+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1), -+SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), -+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), -+SND_SOC_DAPM_OUTPUT("MONOOUT"), -+SND_SOC_DAPM_OUTPUT("HPOUTL"), -+SND_SOC_DAPM_OUTPUT("HPOUTR"), -+SND_SOC_DAPM_OUTPUT("LOUT2"), -+SND_SOC_DAPM_OUTPUT("ROUT2"), -+SND_SOC_DAPM_OUTPUT("OUT3"), -+SND_SOC_DAPM_INPUT("LINEINL"), -+SND_SOC_DAPM_INPUT("LINEINR"), -+SND_SOC_DAPM_INPUT("PHONE"), -+SND_SOC_DAPM_INPUT("PCBEEP"), -+SND_SOC_DAPM_INPUT("MIC1"), -+SND_SOC_DAPM_INPUT("MIC2"), -+}; -+ -+static const char *audio_map[][3] = { -+ /* virtual mixer - mixes left & right channels for spk and mono */ -+ {"AC97 Mixer", NULL, "Left DAC"}, -+ {"AC97 Mixer", NULL, "Right DAC"}, -+ -+ /* Left HP mixer */ -+ {"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"}, -+ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, -+ {"Left HP Mixer", "Phone Bypass Switch", "Phone PGA"}, -+ {"Left HP Mixer", "Line Bypass Switch", "Line PGA"}, -+ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, -+ {"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"}, -+ {"Left HP Mixer", NULL, "ALC Sidetone Mux"}, -+ //{"Right HP Mixer", NULL, "HP Mixer"}, -+ -+ /* Right HP mixer */ -+ {"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"}, -+ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, -+ {"Right HP Mixer", "Phone Bypass Switch", "Phone PGA"}, -+ {"Right HP Mixer", "Line Bypass Switch", "Line PGA"}, -+ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"}, -+ {"Right HP Mixer", "Mic Sidetone Switch", "Mic PGA"}, -+ {"Right HP Mixer", NULL, "ALC Sidetone Mux"}, -+ -+ /* speaker mixer */ -+ {"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"}, -+ {"Speaker Mixer", "Line Bypass Switch", "Line PGA"}, -+ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"}, -+ {"Speaker Mixer", "Phone Bypass Switch", "Phone PGA"}, -+ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, -+ -+ /* Phone mixer */ -+ {"Phone Mixer", "PCBeep Bypass Switch", "PCBEEP"}, -+ {"Phone Mixer", "Line Bypass Switch", "Line PGA"}, -+ {"Phone Mixer", "Aux Playback Switch", "Aux DAC"}, -+ {"Phone Mixer", "PCM Playback Switch", "AC97 Mixer"}, -+ {"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"}, -+ {"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"}, -+ -+ /* inputs */ -+ {"Line PGA", NULL, "LINEINL"}, -+ {"Line PGA", NULL, "LINEINR"}, -+ {"Phone PGA", NULL, "PHONE"}, -+ {"Mic PGA", NULL, "MIC1"}, -+ {"Mic PGA", NULL, "MIC2"}, -+ -+ /* left capture selector */ -+ {"Left Capture Select", "Mic", "MIC1"}, -+ {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, -+ {"Left Capture Select", "Line", "LINEINL"}, -+ {"Left Capture Select", "Headphone Mixer", "Left HP Mixer"}, -+ {"Left Capture Select", "Phone Mixer", "Phone Mixer"}, -+ {"Left Capture Select", "Phone", "PHONE"}, -+ -+ /* right capture selector */ -+ {"Right Capture Select", "Mic", "MIC2"}, -+ {"Right Capture Select", "Speaker Mixer", "Speaker Mixer"}, -+ {"Right Capture Select", "Line", "LINEINR"}, -+ {"Right Capture Select", "Headphone Mixer", "Right HP Mixer"}, -+ {"Right Capture Select", "Phone Mixer", "Phone Mixer"}, -+ {"Right Capture Select", "Phone", "PHONE"}, -+ -+ /* ALC Sidetone */ -+ {"ALC Sidetone Mux", "Stereo", "Left Capture Select"}, -+ {"ALC Sidetone Mux", "Stereo", "Right Capture Select"}, -+ {"ALC Sidetone Mux", "Left", "Left Capture Select"}, -+ {"ALC Sidetone Mux", "Right", "Right Capture Select"}, -+ -+ /* ADC's */ -+ {"Left ADC", NULL, "Left Capture Select"}, -+ {"Right ADC", NULL, "Right Capture Select"}, -+ -+ /* outputs */ -+ {"MONOOUT", NULL, "Phone Mixer"}, -+ {"HPOUTL", NULL, "Headphone PGA"}, -+ {"Headphone PGA", NULL, "Left HP Mixer"}, -+ {"HPOUTR", NULL, "Headphone PGA"}, -+ {"Headphone PGA", NULL, "Right HP Mixer"}, -+ -+ /* mono hp mixer */ -+ {"Mono HP Mixer", NULL, "Left HP Mixer"}, -+ {"Mono HP Mixer", NULL, "Right HP Mixer"}, -+ -+ /* Out3 Mux */ -+ {"Out3 Mux", "Left", "Left HP Mixer"}, -+ {"Out3 Mux", "Mono", "Phone Mixer"}, -+ {"Out3 Mux", "Left + Right", "Mono HP Mixer"}, -+ {"Out 3 PGA", NULL, "Out3 Mux"}, -+ {"OUT3", NULL, "Out 3 PGA"}, -+ -+ /* speaker Mux */ -+ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"}, -+ {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"}, -+ {"Speaker PGA", NULL, "Speaker Mux"}, -+ {"LOUT2", NULL, "Speaker PGA"}, -+ {"ROUT2", NULL, "Speaker PGA"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm9712_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]); -+ } -+ -+ /* set up audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+static unsigned int ac97_read(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ -+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || -+ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || -+ reg == AC97_REC_GAIN) -+ return soc_ac97_ops.read(codec->ac97, reg); -+ else { -+ reg = reg >> 1; -+ -+ if (reg > (ARRAY_SIZE(wm9712_reg))) -+ return -EIO; -+ -+ return cache[reg]; -+ } -+} -+ -+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int val) -+{ -+ u16 *cache = codec->reg_cache; -+ -+ soc_ac97_ops.write(codec->ac97, reg, val); -+ reg = reg >> 1; -+ if (reg <= (ARRAY_SIZE(wm9712_reg))) -+ cache[reg] = val; -+ -+ return 0; -+} -+ -+static int ac97_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ int reg; -+ u16 vra; -+ -+ vra = ac97_read(codec, AC97_EXTENDED_STATUS); -+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ reg = AC97_PCM_FRONT_DAC_RATE; -+ else -+ reg = AC97_PCM_LR_ADC_RATE; -+ -+ return ac97_write(codec, reg, runtime->rate); -+} -+ -+static int ac97_aux_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 vra, xsle; -+ -+ vra = ac97_read(codec, AC97_EXTENDED_STATUS); -+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); -+ xsle = ac97_read(codec, AC97_PCI_SID); -+ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); -+ -+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) -+ return -ENODEV; -+ -+ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); -+} -+ -+struct snd_soc_codec_dai wm9712_dai[] = { -+{ -+ .name = "AC97 HiFi", -+ .playback = { -+ .stream_name = "HiFi Playback", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .stream_name = "HiFi Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .prepare = ac97_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(ac97_modes), -+ .mode = ac97_modes,}, -+ }, -+ { -+ .name = "AC97 Aux", -+ .playback = { -+ .stream_name = "Aux Playback", -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .ops = { -+ .prepare = ac97_aux_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(ac97_modes), -+ .mode = ac97_modes,}, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm9712_dai); -+ -+static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ u16 reg; -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* liam - maybe enable thermal shutdown */ -+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff; -+ ac97_write(codec, AC97_EXTENDED_MID, reg); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* enable master bias and vmid */ -+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff; -+ ac97_write(codec, AC97_EXTENDED_MID, reg); -+ ac97_write(codec, AC97_POWERDOWN, 0x0000); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* disable everything including AC link */ -+ ac97_write(codec, AC97_EXTENDED_MID, 0xffff); -+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); -+ ac97_write(codec, AC97_POWERDOWN, 0xffff); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) -+{ -+ if (try_warm && soc_ac97_ops.warm_reset) { -+ soc_ac97_ops.warm_reset(codec->ac97); -+ if (!(ac97_read(codec, 0) & 0x8000)) -+ return 1; -+ } -+ -+ soc_ac97_ops.reset(codec->ac97); -+ if (ac97_read(codec, 0) & 0x8000) -+ goto err; -+ return 0; -+ -+err: -+ printk(KERN_ERR "WM9712 AC97 reset failed\n"); -+ return -EIO; -+} -+ -+static int wm9712_soc_suspend(struct platform_device *pdev, -+ pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm9712_soc_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i, ret; -+ u16 *cache = codec->reg_cache; -+ -+ ret = wm9712_reset(codec, 1); -+ if (ret < 0){ -+ printk(KERN_ERR "could not reset AC97 codec\n"); -+ return ret; -+ } -+ -+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ if (ret == 0) { -+ /* Sync reg_cache with the hardware after cold reset */ -+ for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) { -+ if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || -+ (i > 0x58 && i != 0x5c)) -+ continue; -+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); -+ } -+ } -+ -+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) -+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0); -+ -+ return ret; -+} -+ -+static int wm9712_soc_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION); -+ -+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (socdev->codec == NULL) -+ return -ENOMEM; -+ codec = socdev->codec; -+ mutex_init(&codec->mutex); -+ -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm9712_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) { -+ kfree(codec->ac97); -+ kfree(socdev->codec); -+ socdev->codec = NULL; -+ return -ENOMEM; -+ } -+ memcpy(codec->reg_cache, wm9712_reg, sizeof(u16) * ARRAY_SIZE(wm9712_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9712_reg); -+ codec->reg_cache_step = 2; -+ -+ codec->name = "WM9712"; -+ codec->owner = THIS_MODULE; -+ codec->dai = wm9712_dai; -+ codec->num_dai = ARRAY_SIZE(wm9712_dai); -+ codec->write = ac97_write; -+ codec->read = ac97_read; -+ codec->dapm_event = wm9712_dapm_event; -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); -+ if (ret < 0) -+ goto err; -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if (ret < 0) -+ goto pcm_err; -+ -+ ret = wm9712_reset(codec, 0); -+ if (ret < 0) { -+ printk(KERN_ERR "AC97 link error\n"); -+ goto reset_err; -+ } -+ -+ /* set alc mux to none */ -+ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); -+ -+ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm9712_add_controls(codec); -+ wm9712_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) -+ goto reset_err; -+ -+ return 0; -+ -+reset_err: -+ snd_soc_free_pcms(socdev); -+ -+pcm_err: -+ snd_soc_free_ac97_codec(codec); -+ -+err: -+ kfree(socdev->codec->reg_cache); -+ kfree(socdev->codec); -+ socdev->codec = NULL; -+ return ret; -+} -+ -+static int wm9712_soc_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec == NULL) -+ return 0; -+ -+ snd_soc_dapm_free(socdev); -+ snd_soc_free_pcms(socdev); -+ snd_soc_free_ac97_codec(codec); -+ kfree(codec->reg_cache); -+ kfree(codec); -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm9712 = { -+ .probe = wm9712_soc_probe, -+ .remove = wm9712_soc_remove, -+ .suspend = wm9712_soc_suspend, -+ .resume = wm9712_soc_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712); -+ -+MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm9712.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm9712.h -@@ -0,0 +1,14 @@ -+/* -+ * wm9712.h -- WM9712 Soc Audio driver -+ */ -+ -+#ifndef _WM9712_H -+#define _WM9712_H -+ -+#define WM9712_DAI_AC97_HIFI 0 -+#define WM9712_DAI_AC97_AUX 1 -+ -+extern struct snd_soc_codec_dai wm9712_dai[2]; -+extern struct snd_soc_codec_device soc_codec_dev_wm9712; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/wm9713.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm9713.c -@@ -0,0 +1,1313 @@ -+/* -+ * wm9713.c -- ALSA Soc WM9713 codec support -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 4th Feb 2006 Initial version. -+ * -+ * Features:- -+ * -+ * o Support for AC97 Codec, Voice DAC and Aux DAC -+ * o Support for DAPM -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/ac97_codec.h> -+#include <sound/initval.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#define WM9713_VERSION "0.12" -+ -+struct wm9713 { -+ u32 pll; /* current PLL frequency */ -+ u32 pll_resume; /* PLL resume frequency */ -+}; -+ -+static unsigned int ac97_read(struct snd_soc_codec *codec, -+ unsigned int reg); -+static int ac97_write(struct snd_soc_codec *codec, -+ unsigned int reg, unsigned int val); -+ -+#define AC97_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define AC97_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ -+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ -+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+/* may need to expand this */ -+static struct snd_soc_dai_mode ac97_modes[] = { -+ { -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE, -+ .pcmrate = AC97_RATES, -+ }, -+}; -+ -+#define WM9713_VOICE_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \ -+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_DSP_A | \ -+ SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \ -+ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ -+ SND_SOC_DAIFMT_IB_IF) -+ -+#define WM9713_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM9713_VOICE_FSB \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ -+ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) -+ -+#define WM9713_VOICE_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 | \ -+ SNDRV_PCM_RATE_96000) -+ -+#define WM9713_HIFI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+/* -+ * Voice modes -+ */ -+static struct snd_soc_dai_mode wm9713_voice_modes[] = { -+ /* master modes */ -+ { -+ .fmt = WM9713_VOICE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | \ -+ SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = WM9713_HIFI_BITS, -+ .pcmrate = WM9713_VOICE_RATES, -+ .pcmdir = WM9713_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = WM9713_VOICE_FSB, -+ }, -+ -+ /* slave modes */ -+ { -+ .fmt = WM9713_VOICE_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM9713_HIFI_BITS, -+ .pcmrate = WM9713_VOICE_RATES, -+ .pcmdir = WM9713_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * WM9713 register cache -+ * Reg 0x3c bit 15 is used by touch driver. -+ */ -+static const u16 wm9713_reg[] = { -+ 0x6174, 0x8080, 0x8080, 0x8080, // 6 -+ 0xc880, 0xe808, 0xe808, 0x0808, // e -+ 0x00da, 0x8000, 0xd600, 0xaaa0, // 16 -+ 0xaaa0, 0xaaa0, 0x0000, 0x0000, // 1e -+ 0x0f0f, 0x0040, 0x0000, 0x7f00, // 26 -+ 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e -+ 0x0000, 0xbb80, 0x0000, 0x4523, // 36 -+ 0x0000, 0x2000, 0x7eff, 0xffff, // 3e -+ 0x0000, 0x0000, 0x0080, 0x0000, // 46 -+ 0x0000, 0x0000, 0xfffe, 0xffff, // 4e -+ 0x0000, 0x0000, 0x0000, 0xfffe, // 56 -+ 0x4000, 0x0000, 0x0000, 0x0000, // 5e -+ 0xb032, 0x3e00, 0x0000, 0x0000, // 66 -+ 0x0000, 0x0000, 0x0000, 0x0000, // 6e -+ 0x0000, 0x0000, 0x0000, 0x0006, // 76 -+ 0x0001, 0x0000, 0x574d, 0x4c13, // 7e -+ 0x0000, 0x0000, 0x0000 // virtual hp & mic mixers -+}; -+ -+/* virtual HP mixers regs */ -+#define HPL_MIXER 0x80 -+#define HPR_MIXER 0x82 -+#define MICB_MUX 0x82 -+ -+static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; -+static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; -+static const char *wm9713_rec_src[] = -+ {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker", -+ "Mono Out", "Zh"}; -+static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; -+static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; -+static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", -+ "Mono Vmid", "Inv Vmid"}; -+static const char *wm9713_spk_pga[] = -+ {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", -+ "Speaker Vmid", "Inv Vmid"}; -+static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone", -+ "Headphone Vmid"}; -+static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"}; -+static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"}; -+static const char *wm9713_dac_inv[] = -+ {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone", -+ "Headphone Mono", "NC", "Vmid"}; -+static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"}; -+static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"}; -+static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"}; -+static const char *wm9713_micb_select[] = {"MPB", "MPA"}; -+ -+static const struct soc_enum wm9713_enum[] = { -+SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */ -+SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */ -+SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */ -+SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */ -+SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/ -+SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */ -+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/ -+SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */ -+SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */ -+SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */ -+SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */ -+SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */ -+SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */ -+SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */ -+SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */ -+SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */ -+SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */ -+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */ -+SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */ -+SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ -+}; -+ -+static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = { -+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), -+SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1), -+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), -+SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE,15, 7, 1, 1), -+SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), -+SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), -+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), -+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), -+ -+SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), -+SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), -+ -+SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1), -+SOC_ENUM("Capture Volume Steps", wm9713_enum[5]), -+SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 63, 0), -+SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), -+ -+SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), -+SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), -+SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), -+ -+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), -+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), -+SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), -+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), -+SOC_ENUM("ALC Function", wm9713_enum[6]), -+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), -+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0), -+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), -+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), -+SOC_ENUM("ALC NG Type", wm9713_enum[17]), -+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0), -+ -+SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0), -+SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), -+ -+SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1), -+SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0), -+SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1), -+ -+SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1), -+SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0), -+SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1), -+ -+SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1), -+SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), -+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), -+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), -+ -+SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), -+SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), -+SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), -+ -+SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), -+SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), -+SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1), -+ -+SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1), -+SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1), -+SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1), -+ -+SOC_ENUM("Bass Control", wm9713_enum[16]), -+SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), -+SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1), -+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0), -+SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1), -+SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1), -+ -+SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0), -+SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), -+SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), -+}; -+ -+/* add non dapm controls */ -+static int wm9713_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm9713_snd_ac97_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ return 0; -+} -+ -+/* We have to create a fake left and right HP mixers because -+ * the codec only has a single control that is shared by both channels. -+ * This makes it impossible to determine the audio path using the current -+ * register map, thus we add a new (virtual) register to help determine the -+ * audio route within the device. -+ */ -+static int mixer_event (struct snd_soc_dapm_widget *w, int event) -+{ -+ u16 l, r, beep, tone, phone, rec, pcm, aux; -+ -+ l = ac97_read(w->codec, HPL_MIXER); -+ r = ac97_read(w->codec, HPR_MIXER); -+ beep = ac97_read(w->codec, AC97_PC_BEEP); -+ tone = ac97_read(w->codec, AC97_MASTER_TONE); -+ phone = ac97_read(w->codec, AC97_PHONE); -+ rec = ac97_read(w->codec, AC97_REC_SEL); -+ pcm = ac97_read(w->codec, AC97_PCM); -+ aux = ac97_read(w->codec, AC97_AUX); -+ -+ if (event & SND_SOC_DAPM_PRE_REG) -+ return 0; -+ if (l & 0x1 || r & 0x1) -+ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); -+ -+ if (l & 0x2 || r & 0x2) -+ ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000); -+ -+ if (l & 0x4 || r & 0x4) -+ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_PHONE, phone | 0x8000); -+ -+ if (l & 0x8 || r & 0x8) -+ ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000); -+ -+ if (l & 0x10 || r & 0x10) -+ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_PCM, pcm | 0x8000); -+ -+ if (l & 0x20 || r & 0x20) -+ ac97_write(w->codec, AC97_AUX, aux & 0x7fff); -+ else -+ ac97_write(w->codec, AC97_AUX, aux | 0x8000); -+ -+ return 0; -+} -+ -+/* Left Headphone Mixers */ -+static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { -+SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), -+SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), -+SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), -+SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0), -+SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), -+}; -+ -+/* Right Headphone Mixers */ -+static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { -+SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), -+SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), -+SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), -+SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0), -+SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0), -+}; -+ -+/* headphone capture mux */ -+static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[1]); -+ -+/* headphone mic mux */ -+static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[0]); -+ -+/* Speaker Mixer */ -+static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { -+SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), -+SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), -+SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), -+SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), -+SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1), -+SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), -+}; -+ -+/* Mono Mixer */ -+static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { -+SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), -+SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), -+SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), -+SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), -+SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1), -+SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1), -+SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1), -+SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1), -+}; -+ -+/* mono mic mux */ -+static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[2]); -+ -+/* mono output mux */ -+static const struct snd_kcontrol_new wm9713_mono_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[7]); -+ -+/* speaker left output mux */ -+static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[8]); -+ -+/* speaker right output mux */ -+static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[9]); -+ -+/* headphone left output mux */ -+static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[10]); -+ -+/* headphone right output mux */ -+static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[11]); -+ -+/* Out3 mux */ -+static const struct snd_kcontrol_new wm9713_out3_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[12]); -+ -+/* Out4 mux */ -+static const struct snd_kcontrol_new wm9713_out4_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[13]); -+ -+/* DAC inv mux 1 */ -+static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[14]); -+ -+/* DAC inv mux 2 */ -+static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[15]); -+ -+/* Capture source left */ -+static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[3]); -+ -+/* Capture source right */ -+static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[4]); -+ -+/* mic source */ -+static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[18]); -+ -+/* mic source B virtual control */ -+static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls = -+SOC_DAPM_ENUM("Route", wm9713_enum[19]); -+ -+static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = { -+SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_hp_rec_mux_controls), -+SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_hp_mic_mux_controls), -+SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_mono_mic_mux_controls), -+SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_mono_mux_controls), -+SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_hp_spkl_mux_controls), -+SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_hp_spkr_mux_controls), -+SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_hpl_out_mux_controls), -+SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_hpr_out_mux_controls), -+SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_out3_mux_controls), -+SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0, -+ &wm9713_out4_mux_controls), -+SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0, -+ &wm9713_dac_inv1_mux_controls), -+SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0, -+ &wm9713_dac_inv2_mux_controls), -+SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, -+ &wm9713_rec_srcl_mux_controls), -+SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, -+ &wm9713_rec_srcr_mux_controls), -+SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0, -+ &wm9713_mic_sel_mux_controls ), -+SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0, -+ &wm9713_micb_sel_mux_controls ), -+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1, -+ &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls), -+ mixer_event, SND_SOC_DAPM_POST_REG), -+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1, -+ &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls), -+ mixer_event, SND_SOC_DAPM_POST_REG), -+SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1, -+ &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)), -+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1, -+ &wm9713_speaker_mixer_controls[0], -+ ARRAY_SIZE(wm9713_speaker_mixer_controls)), -+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1), -+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1), -+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -+SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -+SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -+SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), -+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), -+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1), -+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1), -+SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0), -+SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0), -+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1), -+SND_SOC_DAPM_OUTPUT("MONO"), -+SND_SOC_DAPM_OUTPUT("HPL"), -+SND_SOC_DAPM_OUTPUT("HPR"), -+SND_SOC_DAPM_OUTPUT("SPKL"), -+SND_SOC_DAPM_OUTPUT("SPKR"), -+SND_SOC_DAPM_OUTPUT("OUT3"), -+SND_SOC_DAPM_OUTPUT("OUT4"), -+SND_SOC_DAPM_INPUT("LINEL"), -+SND_SOC_DAPM_INPUT("LINER"), -+SND_SOC_DAPM_INPUT("MONOIN"), -+SND_SOC_DAPM_INPUT("PCBEEP"), -+SND_SOC_DAPM_INPUT("MIC1"), -+SND_SOC_DAPM_INPUT("MIC2A"), -+SND_SOC_DAPM_INPUT("MIC2B"), -+SND_SOC_DAPM_VMID("VMID"), -+}; -+ -+static const char *audio_map[][3] = { -+ /* left HP mixer */ -+ {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, -+ {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, -+ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, -+ {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, -+ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, -+ {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"}, -+ {"Left HP Mixer", NULL, "Capture Headphone Mux"}, -+ -+ /* right HP mixer */ -+ {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, -+ {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, -+ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, -+ {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, -+ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"}, -+ {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"}, -+ {"Right HP Mixer", NULL, "Capture Headphone Mux"}, -+ -+ /* virtual mixer - mixes left & right channels for spk and mono */ -+ {"AC97 Mixer", NULL, "Left DAC"}, -+ {"AC97 Mixer", NULL, "Right DAC"}, -+ {"Line Mixer", NULL, "Right Line In"}, -+ {"Line Mixer", NULL, "Left Line In"}, -+ {"HP Mixer", NULL, "Left HP Mixer"}, -+ {"HP Mixer", NULL, "Right HP Mixer"}, -+ {"Capture Mixer", NULL, "Left Capture Source"}, -+ {"Capture Mixer", NULL, "Right Capture Source"}, -+ -+ /* speaker mixer */ -+ {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, -+ {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, -+ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, -+ {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, -+ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"}, -+ {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, -+ -+ /* mono mixer */ -+ {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, -+ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, -+ {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, -+ {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, -+ {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, -+ {"Mono Mixer", NULL, "Capture Mono Mux"}, -+ -+ /* DAC inv mux 1 */ -+ {"DAC Inv Mux 1", "Mono", "Mono Mixer"}, -+ {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"}, -+ {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"}, -+ {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"}, -+ {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"}, -+ -+ /* DAC inv mux 2 */ -+ {"DAC Inv Mux 2", "Mono", "Mono Mixer"}, -+ {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"}, -+ {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"}, -+ {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"}, -+ {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"}, -+ -+ /* headphone left mux */ -+ {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"}, -+ -+ /* headphone right mux */ -+ {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"}, -+ -+ /* speaker left mux */ -+ {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"}, -+ {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"}, -+ {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"}, -+ -+ /* speaker right mux */ -+ {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"}, -+ {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"}, -+ {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"}, -+ -+ /* mono mux */ -+ {"Mono Out Mux", "Mono", "Mono Mixer"}, -+ {"Mono Out Mux", "Inv", "DAC Inv Mux 1"}, -+ -+ /* out 3 mux */ -+ {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"}, -+ -+ /* out 4 mux */ -+ {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"}, -+ -+ /* output pga */ -+ {"HPL", NULL, "Left Headphone"}, -+ {"Left Headphone", NULL, "Left Headphone Out Mux"}, -+ {"HPR", NULL, "Right Headphone"}, -+ {"Right Headphone", NULL, "Right Headphone Out Mux"}, -+ {"OUT3", NULL, "Out 3"}, -+ {"Out 3", NULL, "Out 3 Mux"}, -+ {"OUT4", NULL, "Out 4"}, -+ {"Out 4", NULL, "Out 4 Mux"}, -+ {"SPKL", NULL, "Left Speaker"}, -+ {"Left Speaker", NULL, "Left Speaker Out Mux"}, -+ {"SPKR", NULL, "Right Speaker"}, -+ {"Right Speaker", NULL, "Right Speaker Out Mux"}, -+ {"MONO", NULL, "Mono Out"}, -+ {"Mono Out", NULL, "Mono Out Mux"}, -+ -+ /* input pga */ -+ {"Left Line In", NULL, "LINEL"}, -+ {"Right Line In", NULL, "LINER"}, -+ {"Mono In", NULL, "MONOIN"}, -+ {"Mic A PGA", NULL, "Mic A Pre Amp"}, -+ {"Mic B PGA", NULL, "Mic B Pre Amp"}, -+ -+ /* left capture select */ -+ {"Left Capture Source", "Mic 1", "Mic A Pre Amp"}, -+ {"Left Capture Source", "Mic 2", "Mic B Pre Amp"}, -+ {"Left Capture Source", "Line", "LINEL"}, -+ {"Left Capture Source", "Mono In", "MONOIN"}, -+ {"Left Capture Source", "Headphone", "Left HP Mixer"}, -+ {"Left Capture Source", "Speaker", "Speaker Mixer"}, -+ {"Left Capture Source", "Mono Out", "Mono Mixer"}, -+ -+ /* right capture select */ -+ {"Right Capture Source", "Mic 1", "Mic A Pre Amp"}, -+ {"Right Capture Source", "Mic 2", "Mic B Pre Amp"}, -+ {"Right Capture Source", "Line", "LINER"}, -+ {"Right Capture Source", "Mono In", "MONOIN"}, -+ {"Right Capture Source", "Headphone", "Right HP Mixer"}, -+ {"Right Capture Source", "Speaker", "Speaker Mixer"}, -+ {"Right Capture Source", "Mono Out", "Mono Mixer"}, -+ -+ /* left ADC */ -+ {"Left ADC", NULL, "Left Capture Source"}, -+ -+ /* right ADC */ -+ {"Right ADC", NULL, "Right Capture Source"}, -+ -+ /* mic */ -+ {"Mic A Pre Amp", NULL, "Mic A Source"}, -+ {"Mic A Source", "Mic 1", "MIC1"}, -+ {"Mic A Source", "Mic 2 A", "MIC2A"}, -+ {"Mic A Source", "Mic 2 B", "Mic B Source"}, -+ {"Mic B Pre Amp", "MPB", "Mic B Source"}, -+ {"Mic B Source", NULL, "MIC2B"}, -+ -+ /* headphone capture */ -+ {"Capture Headphone Mux", "Stereo", "Capture Mixer"}, -+ {"Capture Headphone Mux", "Left", "Left Capture Source"}, -+ {"Capture Headphone Mux", "Right", "Right Capture Source"}, -+ -+ /* mono capture */ -+ {"Capture Mono Mux", "Stereo", "Capture Mixer"}, -+ {"Capture Mono Mux", "Left", "Left Capture Source"}, -+ {"Capture Mono Mux", "Right", "Right Capture Source"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm9713_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); -+ } -+ -+ /* set up audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+static unsigned int ac97_read(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ -+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || -+ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || -+ reg == AC97_CD) -+ return soc_ac97_ops.read(codec->ac97, reg); -+ else { -+ reg = reg >> 1; -+ -+ if (reg > (ARRAY_SIZE(wm9713_reg))) -+ return -EIO; -+ -+ return cache[reg]; -+ } -+} -+ -+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int val) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg < 0x7c) -+ soc_ac97_ops.write(codec->ac97, reg, val); -+ reg = reg >> 1; -+ if (reg <= (ARRAY_SIZE(wm9713_reg))) -+ cache[reg] = val; -+ -+ return 0; -+} -+ -+struct pll_ { -+ unsigned int in_hz; -+ unsigned int lf:1; /* allows low frequency use */ -+ unsigned int sdm:1; /* allows fraction n div */ -+ unsigned int divsel:1; /* enables input clock div */ -+ unsigned int divctl:1; /* input clock divider */ -+ unsigned int n:4; -+ unsigned int k; -+}; -+ -+struct pll_ pll[] = { -+ {13000000, 0, 1, 0, 0, 7, 0x23f488}, -+ {2048000, 1, 0, 0, 0, 12, 0x0}, -+ {4096000, 1, 0, 0, 0, 6, 0x0}, -+ {12288000, 0, 0, 0, 0, 8, 0x0}, -+ /* liam - add more entries */ -+}; -+ -+/* we must have either 24.576MHz or a PLL freq */ -+static unsigned int wm9713_config_ac97sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ int i; -+ dai->mclk = 0; -+ -+ /* first check if we can get away witout burning any PLL power */ -+ if (24576000 == clk) { -+ /* standard AC97 clock */ -+ dai->mclk = clk; -+ goto out; -+ } -+ -+ /* ok no standard clock, so we must now try the PLL */ -+ for(i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (clk == pll[i].in_hz) { -+ dai->mclk = clk; /* clock out */ -+ goto out; -+ } -+ } -+ -+out: -+ return dai->mclk; -+} -+ -+/* The WM9713 voice DAC can only run at 256FS. This interface and DAC are -+ * clocked by the main AC97 clock divided down to 256 FS. -+ */ -+static unsigned int wm9713_config_vsysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ -+ int i, j, best_clk = info->fs * info->rate; -+ -+ /* can we run at this clk without the PLL ? */ -+ for (i = 1; i <= 16; i++) { -+ if (best_clk * i == clk) { -+ dai->pll_in = 0; -+ dai->clk_div = i << 1; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ -+ /* now check for PLL support */ -+ for (i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (pll[i].in_hz == clk) { -+ for (j = 1; j <= 16; j++) { -+ if (24576000 == j * best_clk) { -+ dai->pll_in = clk; -+ dai->pll_out = 24576000; -+ dai->clk_div = j << 1; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+} -+ -+u32 wm9713_set_pll(struct snd_soc_codec *codec, u32 in) -+{ -+ struct wm9713 *wm = (struct wm9713*)codec->private_data; -+ int i; -+ u16 reg, reg2; -+ -+ /* turn PLL off ? */ -+ if (in == 0) { -+ /* disable PLL power and select ext source */ -+ reg = ac97_read(codec, AC97_HANDSET_RATE); -+ ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); -+ reg = ac97_read(codec, AC97_EXTENDED_MID); -+ ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); -+ wm->pll = 0; -+ return 0; -+ } -+ -+ for (i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (pll[i].in_hz == in) -+ goto found; -+ } -+ return -EINVAL; -+ -+found: -+ if (pll[i].sdm == 0) { -+ reg = (pll[i].n << 12) | (pll[i].lf << 11) | -+ (pll[i].divsel << 9) | (pll[i].divctl << 8); -+ ac97_write(codec, AC97_LINE1_LEVEL, reg); -+ } else { -+ /* write the fractional k to the reg 0x46 pages */ -+ reg2 = (pll[i].n << 12) | (pll[i].lf << 11) | (pll[i].sdm << 10) | -+ (pll[i].divsel << 9) | (pll[i].divctl << 8); -+ -+ reg = reg2 | (0x5 << 4) | (pll[i].k >> 20); /* K [21:20] */ -+ ac97_write(codec, AC97_LINE1_LEVEL, reg); -+ -+ reg = reg2 | (0x4 << 4) | ((pll[i].k >> 16) & 0xf); /* K [19:16] */ -+ ac97_write(codec, AC97_LINE1_LEVEL, reg); -+ -+ reg = reg2 | (0x3 << 4) | ((pll[i].k >> 12) & 0xf); /* K [15:12] */ -+ ac97_write(codec, AC97_LINE1_LEVEL, reg); -+ -+ reg = reg2 | (0x2 << 4) | ((pll[i].k >> 8) & 0xf); /* K [11:8] */ -+ ac97_write(codec, AC97_LINE1_LEVEL, reg); -+ -+ reg = reg2 | (0x1 << 4) | ((pll[i].k >> 4) & 0xf); /* K [7:4] */ -+ ac97_write(codec, AC97_LINE1_LEVEL, reg); -+ -+ reg = reg2 | (0x0 << 4) | (pll[i].k & 0xf); /* K [3:0] */ -+ ac97_write(codec, AC97_LINE1_LEVEL, reg); -+ } -+ -+ /* turn PLL on and select as source */ -+ reg = ac97_read(codec, AC97_EXTENDED_MID); -+ ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); -+ reg = ac97_read(codec, AC97_HANDSET_RATE); -+ ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); -+ /* wait 10ms AC97 link frames for the link to stabilise */ -+ schedule_timeout_interruptible(msecs_to_jiffies(10)); -+ wm->pll = in; -+ return 0; -+} -+EXPORT_SYMBOL_GPL(wm9713_set_pll); -+ -+static int wm9713_voice_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 reg = 0x8000, bfs, div, gpio; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffe2; -+ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ -+ case SND_SOC_DAIFMT_CBM_CFM: -+ reg |= 0x4000; -+ gpio |= 0x0008; -+ break; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ reg |= 0x6000; -+ gpio |= 0x000c; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ reg |= 0x0200; -+ gpio |= 0x000d; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFM: -+ gpio |= 0x0009; -+ break; -+ } -+ ac97_write(codec, AC97_GPIO_CFG, gpio); -+ -+ /* enable PLL if needed */ -+ if (rtd->codec_dai->pll_in) -+ wm9713_set_pll(codec, rtd->codec_dai->pll_in); -+ -+ /* set the PCM divider */ -+ div = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; -+ ac97_write(codec, AC97_HANDSET_RATE, div | -+ ((rtd->codec_dai->clk_div >> 1) -1) << 8); -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_IB_IF: -+ reg |= 0x00c0; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ reg |= 0x0080; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ reg |= 0x0040; -+ break; -+ } -+ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ reg |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ reg |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ reg |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ reg |= 0x0043; -+ break; -+ } -+ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ reg |= 0x0004; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ reg |= 0x0008; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ reg |= 0x000c; -+ break; -+ } -+ -+ switch (bfs) { -+ case 2: -+ reg |= (0x1 << 9); -+ break; -+ case 4: -+ reg |= (0x2 << 9); -+ break; -+ case 8: -+ reg |= (0x3 << 9); -+ break; -+ case 16: -+ reg |= (0x4 << 9); -+ break; -+ } -+ -+ /* enable PCM interface in master mode */ -+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); -+ return 0; -+} -+ -+static void wm9713_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (!codec->active) -+ wm9713_set_pll(codec, 0); -+} -+ -+static void wm9713_voiceshutdown(snd_pcm_substream_t *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 status; -+ -+ wm9713_shutdown(substream); -+ -+ /* Gracefully shut down the voice interface. */ -+ status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; -+ ac97_write(codec,AC97_HANDSET_RATE,0x0280); -+ schedule_timeout_interruptible(msecs_to_jiffies(1)); -+ ac97_write(codec,AC97_HANDSET_RATE,0x0F80); -+ ac97_write(codec,AC97_EXTENDED_MID,status); -+} -+ -+static int ac97_hifi_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ int reg; -+ u16 vra; -+ -+ /* we need a 24576000Hz clock to run at the correct speed */ -+ if (rtd->codec_dai->mclk != 24576000) -+ wm9713_set_pll(codec, rtd->codec_dai->mclk); -+ -+ vra = ac97_read(codec, AC97_EXTENDED_STATUS); -+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ reg = AC97_PCM_FRONT_DAC_RATE; -+ else -+ reg = AC97_PCM_LR_ADC_RATE; -+ -+ return ac97_write(codec, reg, runtime->rate); -+} -+ -+static int ac97_aux_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 vra, xsle; -+ -+ /* we need a 24576000Hz clock to run at the correct speed */ -+ if (rtd->codec_dai->mclk != 24576000) -+ wm9713_set_pll(codec, rtd->codec_dai->mclk); -+ -+ vra = ac97_read(codec, AC97_EXTENDED_STATUS); -+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); -+ xsle = ac97_read(codec, AC97_PCI_SID); -+ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); -+ -+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) -+ return -ENODEV; -+ -+ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); -+} -+ -+struct snd_soc_codec_dai wm9713_dai[] = { -+{ -+ .name = "AC97 HiFi", -+ .playback = { -+ .stream_name = "HiFi Playback", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .stream_name = "HiFi Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .config_sysclk = wm9713_config_ac97sysclk, -+ .ops = { -+ .shutdown = wm9713_shutdown, -+ .prepare = ac97_hifi_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(ac97_modes), -+ .mode = ac97_modes,},}, -+ { -+ .name = "AC97 Aux", -+ .playback = { -+ .stream_name = "Aux Playback", -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .config_sysclk = wm9713_config_ac97sysclk, -+ .ops = { -+ .shutdown = wm9713_shutdown, -+ .prepare = ac97_aux_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(ac97_modes), -+ .mode = ac97_modes,} -+ }, -+ { -+ .name = "WM9713 Voice", -+ .playback = { -+ .stream_name = "Voice Playback", -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .capture = { -+ .stream_name = "Voice Capture", -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .config_sysclk = wm9713_config_vsysclk, -+ .ops = { -+ .prepare = wm9713_voice_prepare, -+ .shutdown = wm9713_voiceshutdown,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm9713_voice_modes), -+ .mode = wm9713_voice_modes,}, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm9713_dai); -+ -+int wm9713_reset(struct snd_soc_codec *codec, int try_warm) -+{ -+ if (try_warm && soc_ac97_ops.warm_reset) { -+ soc_ac97_ops.warm_reset(codec->ac97); -+ if (!(ac97_read(codec, 0) & 0x8000)) -+ return 1; -+ } -+ -+ soc_ac97_ops.reset(codec->ac97); -+ if (ac97_read(codec, 0) & 0x8000) -+ return -EIO; -+ return 0; -+} -+EXPORT_SYMBOL_GPL(wm9713_reset); -+ -+static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ u16 reg; -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* enable thermal shutdown */ -+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; -+ ac97_write(codec, AC97_EXTENDED_MID, reg); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* enable master bias and vmid */ -+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; -+ ac97_write(codec, AC97_EXTENDED_MID, reg); -+ ac97_write(codec, AC97_POWERDOWN, 0x0000); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* disable everything including AC link */ -+ ac97_write(codec, AC97_EXTENDED_MID, 0xffff); -+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); -+ ac97_write(codec, AC97_POWERDOWN, 0xffff); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+static int wm9713_soc_suspend(struct platform_device *pdev, -+ pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ struct wm9713 *wm = (struct wm9713*)codec->private_data; -+ -+ if (wm->pll) { -+ wm->pll_resume = wm->pll; -+ wm9713_set_pll(codec, 0); -+ } -+ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm9713_soc_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ struct wm9713 *wm = (struct wm9713*)codec->private_data; -+ int i, ret; -+ u16 *cache = codec->reg_cache; -+ -+ if ((ret = wm9713_reset(codec, 1)) < 0){ -+ printk(KERN_ERR "could not reset AC97 codec\n"); -+ return ret; -+ } -+ -+ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ /* only synchronise the codec if warm reset failed */ -+ if (ret == 0) { -+ for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i+=2) { -+ if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || -+ i == AC97_EXTENDED_MSTATUS || i > 0x66) -+ continue; -+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); -+ } -+ } -+ -+ if (wm->pll_resume) { -+ wm9713_set_pll(codec, wm->pll_resume); -+ wm->pll_resume = 0; -+ } -+ -+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) -+ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); -+ -+ return ret; -+} -+ -+static int wm9713_soc_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec; -+ int ret = 0, reg; -+ -+ printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); -+ -+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (socdev->codec == NULL) -+ return -ENOMEM; -+ codec = socdev->codec; -+ mutex_init(&codec->mutex); -+ -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm9713_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL){ -+ kfree(socdev->codec); -+ socdev->codec = NULL; -+ return -ENOMEM; -+ } -+ memcpy(codec->reg_cache, wm9713_reg, -+ sizeof(u16) * ARRAY_SIZE(wm9713_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9713_reg); -+ codec->reg_cache_step = 2; -+ -+ codec->private_data = kzalloc(sizeof(struct wm9713), GFP_KERNEL); -+ if (codec->private_data == NULL) { -+ kfree(codec->reg_cache); -+ kfree(socdev->codec); -+ socdev->codec = NULL; -+ return -ENOMEM; -+ } -+ -+ codec->name = "WM9713"; -+ codec->owner = THIS_MODULE; -+ codec->dai = wm9713_dai; -+ codec->num_dai = ARRAY_SIZE(wm9713_dai); -+ codec->write = ac97_write; -+ codec->read = ac97_read; -+ codec->dapm_event = wm9713_dapm_event; -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); -+ if (ret < 0) -+ goto err; -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if (ret < 0) -+ goto pcm_err; -+ -+ /* do a cold reset for the controller and then try -+ * a warm reset followed by an optional cold reset for codec */ -+ wm9713_reset(codec, 0); -+ ret = wm9713_reset(codec, 1); -+ if (ret < 0) { -+ printk(KERN_ERR "AC97 link error\n"); -+ goto reset_err; -+ } -+ -+ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ /* unmute the adc - move to kcontrol */ -+ reg = ac97_read(codec, AC97_CD) & 0x7fff; -+ ac97_write(codec, AC97_CD, reg); -+ -+ wm9713_add_controls(codec); -+ wm9713_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) -+ goto reset_err; -+ return 0; -+ -+reset_err: -+ snd_soc_free_pcms(socdev); -+ -+pcm_err: -+ snd_soc_free_ac97_codec(codec); -+ -+err: -+ kfree(socdev->codec->private_data); -+ kfree(socdev->codec->reg_cache); -+ kfree(socdev->codec); -+ socdev->codec = NULL; -+ return ret; -+} -+ -+static int wm9713_soc_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec == NULL) -+ return 0; -+ -+ snd_soc_dapm_free(socdev); -+ snd_soc_free_pcms(socdev); -+ snd_soc_free_ac97_codec(codec); -+ kfree(codec->private_data); -+ kfree(codec->reg_cache); -+ kfree(codec); -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm9713= { -+ .probe = wm9713_soc_probe, -+ .remove = wm9713_soc_remove, -+ .suspend = wm9713_soc_suspend, -+ .resume = wm9713_soc_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713); -+ -+MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm9713.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm9713.h -@@ -0,0 +1,18 @@ -+/* -+ * wm9713.h -- WM9713 Soc Audio driver -+ */ -+ -+#ifndef _WM9713_H -+#define _WM9713_H -+ -+#define WM9713_DAI_AC97_HIFI 0 -+#define WM9713_DAI_AC97_AUX 1 -+#define WM9713_DAI_PCM_VOICE 2 -+ -+extern struct snd_soc_codec_device soc_codec_dev_wm9713; -+extern struct snd_soc_codec_dai wm9713_dai[3]; -+ -+u32 wm9713_set_pll(struct snd_soc_codec *codec, u32 in); -+int wm9713_reset(struct snd_soc_codec *codec, int try_warm); -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/pxa/Kconfig -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/Kconfig -@@ -0,0 +1,125 @@ -+menu "SoC Audio for the Intel PXA2xx" -+ -+config SND_PXA2xx_SOC -+ tristate "SoC Audio for the Intel PXA2xx chip" -+ depends on ARCH_PXA && SND -+ select SND_PCM -+ help -+ Say Y or M if you want to add support for codecs attached to -+ the PXA2xx AC97, I2S or SSP interface. You will also need -+ to select the audio interfaces to support below. -+ -+config SND_PXA2xx_AC97 -+ tristate -+ select SND_AC97_CODEC -+ -+config SND_PXA2xx_SOC_AC97 -+ tristate -+ select SND_AC97_BUS -+ select SND_SOC_AC97_BUS -+ -+config SND_PXA2xx_SOC_I2S -+ tristate -+ -+config SND_PXA2xx_SOC_SSP -+ tristate -+ select PXA_SSP -+ -+config SND_PXA2xx_SOC_MAINSTONE -+ tristate "SoC AC97 Audio support for Intel Mainstone" -+ depends on SND_PXA2xx_SOC && MACH_MAINSTONE -+ select SND_PXA2xx_AC97 -+ help -+ Say Y if you want to add support for generic AC97 SoC audio on Mainstone. -+ -+config SND_PXA2xx_SOC_MAINSTONE_WM8731 -+ tristate "SoC I2S Audio support for Intel Mainstone - WM8731" -+ depends on SND_PXA2xx_SOC && MACH_MAINSTONE -+ select SND_PXA2xx_SOC_I2S -+ help -+ Say Y if you want to add support for SoC audio on Mainstone -+ with the WM8731. -+ -+config SND_PXA2xx_SOC_MAINSTONE_WM8753 -+ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM8753" -+ depends on SND_PXA2xx_SOC && MACH_MAINSTONE -+ select SND_PXA2xx_SOC_I2S -+ select SND_PXA2xx_SOC_SSP -+ help -+ Say Y if you want to add support for SoC audio on Mainstone -+ with the WM8753. -+ -+config SND_PXA2xx_SOC_MAINSTONE_WM8974 -+ tristate "SoC I2S Audio support for Intel Mainstone - WM8974" -+ depends on SND_PXA2xx_SOC && MACH_MAINSTONE -+ select SND_PXA2xx_SOC_I2S -+ help -+ Say Y if you want to add support for SoC audio on Mainstone -+ with the WM8974. -+ -+config SND_PXA2xx_SOC_MAINSTONE_WM9713 -+ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM9713" -+ depends on SND_PXA2xx_SOC && MACH_MAINSTONE -+ select SND_PXA2xx_SOC_AC97 -+ select SND_PXA2xx_SOC_SSP -+ help -+ Say Y if you want to add support for SoC voice audio on Mainstone -+ with the WM9713. -+ -+config SND_MAINSTONE_BASEBAND -+ tristate "Example SoC Baseband Audio support for Intel Mainstone" -+ depends on SND_PXA2xx_SOC && MACH_MAINSTONE -+ select SND_PXA2xx_SOC_AC97 -+ help -+ Say Y if you want to add support for SoC baseband on Mainstone -+ with the WM9713 and example Baseband modem. -+ -+config SND_MAINSTONE_BLUETOOTH -+ tristate "Example SoC Bluetooth Audio support for Intel Mainstone" -+ depends on SND_PXA2xx_SOC && MACH_MAINSTONE -+ select SND_PXA2xx_SOC_I2S -+ help -+ Say Y if you want to add support for SoC bluetooth on Mainstone -+ with the WM8753 and example Bluetooth codec. -+ -+config SND_PXA2xx_SOC_MAINSTONE_WM9712 -+ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM9712" -+ depends on SND_PXA2xx_SOC && MACH_MAINSTONE -+ select SND_PXA2xx_SOC_AC97 -+ help -+ Say Y if you want to add support for SoC voice audio on Mainstone -+ with the WM9712. -+ -+config SND_PXA2xx_SOC_CORGI -+ tristate "SoC Audio support for Sharp Zaurus SL-C7x0" -+ depends on SND_PXA2xx_SOC && PXA_SHARP_C7xx -+ select SND_PXA2xx_SOC_I2S -+ help -+ Say Y if you want to add support for SoC audio on Sharp -+ Zaurus SL-C7x0 models (Corgi, Shepherd, Husky). -+ -+config SND_PXA2xx_SOC_SPITZ -+ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" -+ depends on SND_PXA2xx_SOC && PXA_SHARP_Cxx00 -+ select SND_PXA2xx_SOC_I2S -+ help -+ Say Y if you want to add support for SoC audio on Sharp -+ Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita). -+ -+config SND_PXA2xx_SOC_POODLE -+ tristate "SoC Audio support for Poodle" -+ depends on SND_PXA2xx_SOC && MACH_POODLE -+ select SND_PXA2xx_SOC_I2S -+ help -+ Say Y if you want to add support for SoC audio on Sharp -+ Zaurus SL-5600 model (Poodle). -+ -+config SND_PXA2xx_SOC_TOSA -+ tristate "SoC AC97 Audio support for Tosa" -+ depends on SND_PXA2xx_SOC && MACH_TOSA -+ select SND_PXA2xx_SOC_AC97 -+ help -+ Say Y if you want to add support for SoC audio on Sharp -+ Zaurus SL-C6000x models (Tosa). -+ -+endmenu -Index: linux-2.6-pxa-new/sound/soc/pxa/Makefile -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/Makefile -@@ -0,0 +1,36 @@ -+# PXA Platform Support -+snd-soc-pxa2xx-objs := pxa2xx-pcm.o -+snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o -+snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o -+snd-soc-pxa2xx-ssp-objs := pxa2xx-ssp.o -+ -+obj-$(CONFIG_SND_PXA2xx_SOC) += snd-soc-pxa2xx.o -+obj-$(CONFIG_SND_PXA2xx_SOC_AC97) += snd-soc-pxa2xx-ac97.o -+obj-$(CONFIG_SND_PXA2xx_SOC_I2S) += snd-soc-pxa2xx-i2s.o -+obj-$(CONFIG_SND_PXA2xx_SOC_SSP) += snd-soc-pxa2xx-ssp.o -+ -+# PXA Machine Support -+snd-soc-corgi-objs := corgi.o -+snd-soc-mainstone-wm8731-objs := mainstone_wm8731.o -+snd-soc-mainstone-wm8753-objs := mainstone_wm8753.o -+snd-soc-mainstone-wm8974-objs := mainstone_wm8974.o -+snd-soc-mainstone-wm9713-objs := mainstone_wm9713.o -+snd-soc-mainstone-wm9712-objs := mainstone_wm9712.o -+snd-soc-mainstone-baseband-objs := mainstone_baseband.o -+snd-soc-mainstone-bluetooth-objs := mainstone_bluetooth.o -+snd-soc-poodle-objs := poodle.o -+snd-soc-tosa-objs := tosa.o -+snd-soc-spitz-objs := spitz.o -+ -+obj-$(CONFIG_SND_PXA2xx_SOC_CORGI) += snd-soc-corgi.o -+obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8731) += snd-soc-mainstone-wm8731.o -+obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8753) += snd-soc-mainstone-wm8753.o -+obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8974) += snd-soc-mainstone-wm8974.o -+obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM9713) += snd-soc-mainstone-wm9713.o -+obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM9712) += snd-soc-mainstone-wm9712.o -+obj-$(CONFIG_SND_MAINSTONE_BASEBAND) += snd-soc-mainstone-baseband.o -+obj-$(CONFIG_SND_MAINSTONE_BLUETOOTH) += snd-soc-mainstone-bluetooth.o -+obj-$(CONFIG_SND_PXA2xx_SOC_POODLE) += snd-soc-poodle.o -+obj-$(CONFIG_SND_PXA2xx_SOC_TOSA) += snd-soc-tosa.o -+obj-$(CONFIG_SND_PXA2xx_SOC_SPITZ) += snd-soc-spitz.o -+ -Index: linux-2.6-pxa-new/sound/soc/pxa/corgi.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/corgi.c -@@ -0,0 +1,361 @@ -+/* -+ * corgi.c -- SoC audio for Corgi -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> -+ * Richard Purdie <richard@openedhand.com> -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 30th Nov 2005 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/timer.h> -+#include <linux/interrupt.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/mach-types.h> -+#include <asm/hardware/scoop.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/hardware.h> -+#include <asm/arch/corgi.h> -+#include <asm/arch/audio.h> -+ -+#include "../codecs/wm8731.h" -+#include "pxa2xx-pcm.h" -+ -+#define CORGI_HP 0 -+#define CORGI_MIC 1 -+#define CORGI_LINE 2 -+#define CORGI_HEADSET 3 -+#define CORGI_HP_OFF 4 -+#define CORGI_SPK_ON 0 -+#define CORGI_SPK_OFF 1 -+ -+ /* audio clock in Hz - rounded from 12.235MHz */ -+#define CORGI_AUDIO_CLOCK 12288000 -+ -+static int corgi_jack_func; -+static int corgi_spk_func; -+ -+static void corgi_ext_control(struct snd_soc_codec *codec) -+{ -+ int spk = 0, mic = 0, line = 0, hp = 0, hs = 0; -+ -+ /* set up jack connection */ -+ switch (corgi_jack_func) { -+ case CORGI_HP: -+ hp = 1; -+ /* set = unmute headphone */ -+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); -+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); -+ break; -+ case CORGI_MIC: -+ mic = 1; -+ /* reset = mute headphone */ -+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); -+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); -+ break; -+ case CORGI_LINE: -+ line = 1; -+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); -+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); -+ break; -+ case CORGI_HEADSET: -+ hs = 1; -+ mic = 1; -+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); -+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); -+ break; -+ } -+ -+ if (corgi_spk_func == CORGI_SPK_ON) -+ spk = 1; -+ -+ /* set the enpoints to their new connetion states */ -+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); -+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic); -+ snd_soc_dapm_set_endpoint(codec, "Line Jack", line); -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); -+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); -+ -+ /* signal a DAPM event */ -+ snd_soc_dapm_sync_endpoints(codec); -+} -+ -+static int corgi_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_codec *codec = rtd->socdev->codec; -+ -+ /* check the jack status at stream startup */ -+ corgi_ext_control(codec); -+ return 0; -+} -+ -+/* we need to unmute the HP at shutdown as the mute burns power on corgi */ -+static int corgi_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_codec *codec = rtd->socdev->codec; -+ -+ /* set = unmute headphone */ -+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); -+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); -+ return 0; -+} -+ -+static struct snd_soc_ops corgi_ops = { -+ .startup = corgi_startup, -+ .shutdown = corgi_shutdown, -+}; -+ -+static int corgi_get_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = corgi_jack_func; -+ return 0; -+} -+ -+static int corgi_set_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ if (corgi_jack_func == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ corgi_jack_func = ucontrol->value.integer.value[0]; -+ corgi_ext_control(codec); -+ return 1; -+} -+ -+static int corgi_get_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = corgi_spk_func; -+ return 0; -+} -+ -+static int corgi_set_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ if (corgi_spk_func == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ corgi_spk_func = ucontrol->value.integer.value[0]; -+ corgi_ext_control(codec); -+ return 1; -+} -+ -+static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) -+{ -+ if (SND_SOC_DAPM_EVENT_ON(event)) -+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); -+ else -+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); -+ -+ return 0; -+} -+ -+static int corgi_mic_event(struct snd_soc_dapm_widget *w, int event) -+{ -+ if (SND_SOC_DAPM_EVENT_ON(event)) -+ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); -+ else -+ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); -+ -+ return 0; -+} -+ -+/* corgi machine dapm widgets */ -+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { -+SND_SOC_DAPM_HP("Headphone Jack", NULL), -+SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event), -+SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event), -+SND_SOC_DAPM_LINE("Line Jack", NULL), -+SND_SOC_DAPM_HP("Headset Jack", NULL), -+}; -+ -+/* Corgi machine audio map (connections to the codec pins) */ -+static const char *audio_map[][3] = { -+ -+ /* headset Jack - in = micin, out = LHPOUT*/ -+ {"Headset Jack", NULL, "LHPOUT"}, -+ -+ /* headphone connected to LHPOUT1, RHPOUT1 */ -+ {"Headphone Jack", NULL, "LHPOUT"}, -+ {"Headphone Jack", NULL, "RHPOUT"}, -+ -+ /* speaker connected to LOUT, ROUT */ -+ {"Ext Spk", NULL, "ROUT"}, -+ {"Ext Spk", NULL, "LOUT"}, -+ -+ /* mic is connected to MICIN (via right channel of headphone jack) */ -+ {"MICIN", NULL, "Mic Jack"}, -+ -+ /* Same as the above but no mic bias for line signals */ -+ {"MICIN", NULL, "Line Jack"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", -+ "Off"}; -+static const char *spk_function[] = {"On", "Off"}; -+static const struct soc_enum corgi_enum[] = { -+ SOC_ENUM_SINGLE_EXT(5, jack_function), -+ SOC_ENUM_SINGLE_EXT(2, spk_function), -+}; -+ -+static const struct snd_kcontrol_new wm8731_corgi_controls[] = { -+ SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack, -+ corgi_set_jack), -+ SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk, -+ corgi_set_spk), -+}; -+ -+/* -+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device -+ */ -+static int corgi_wm8731_init(struct snd_soc_codec *codec) -+{ -+ int i, err; -+ -+ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); -+ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); -+ -+ /* Add corgi specific controls */ -+ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8731_corgi_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ /* Add corgi specific widgets */ -+ for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); -+ } -+ -+ /* Set up corgi specific audio path audio_map */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ return 0; -+} -+ -+static unsigned int corgi_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) { -+ /* pxa2xx is i2s master */ -+ switch (info->rate) { -+ case 44100: -+ case 88200: -+ /* configure codec digital filters for 44.1, 88.2 */ -+ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ 11289600); -+ break; -+ default: -+ /* configure codec digital filters for all other rates */ -+ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ CORGI_AUDIO_CLOCK); -+ break; -+ } -+ /* config pxa i2s as master */ -+ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, -+ CORGI_AUDIO_CLOCK); -+ } else { -+ /* codec is i2s master - -+ * only configure codec DAI clock and filters */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ CORGI_AUDIO_CLOCK); -+ } -+} -+ -+/* corgi digital audio interface glue - connects codec <--> CPU */ -+static struct snd_soc_dai_link corgi_dai = { -+ .name = "WM8731", -+ .stream_name = "WM8731", -+ .cpu_dai = &pxa_i2s_dai, -+ .codec_dai = &wm8731_dai, -+ .init = corgi_wm8731_init, -+ .config_sysclk = corgi_config_sysclk, -+}; -+ -+/* corgi audio machine driver */ -+static struct snd_soc_machine snd_soc_machine_corgi = { -+ .name = "Corgi", -+ .dai_link = &corgi_dai, -+ .num_links = 1, -+ .ops = &corgi_ops, -+}; -+ -+/* corgi audio private data */ -+static struct wm8731_setup_data corgi_wm8731_setup = { -+ .i2c_address = 0x1b, -+}; -+ -+/* corgi audio subsystem */ -+static struct snd_soc_device corgi_snd_devdata = { -+ .machine = &snd_soc_machine_corgi, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8731, -+ .codec_data = &corgi_wm8731_setup, -+}; -+ -+static struct platform_device *corgi_snd_device; -+ -+static int __init corgi_init(void) -+{ -+ int ret; -+ -+ if (!(machine_is_corgi() || machine_is_shepherd() || machine_is_husky())) -+ return -ENODEV; -+ -+ corgi_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!corgi_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(corgi_snd_device, &corgi_snd_devdata); -+ corgi_snd_devdata.dev = &corgi_snd_device->dev; -+ ret = platform_device_add(corgi_snd_device); -+ -+ if (ret) -+ platform_device_put(corgi_snd_device); -+ -+ return ret; -+} -+ -+static void __exit corgi_exit(void) -+{ -+ platform_device_unregister(corgi_snd_device); -+} -+ -+module_init(corgi_init); -+module_exit(corgi_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Richard Purdie"); -+MODULE_DESCRIPTION("ALSA SoC Corgi"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/mainstone.c -@@ -0,0 +1,126 @@ -+/* -+ * mainstone.c -- SoC audio for Mainstone -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c -+ * Copyright: MontaVista Software Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 30th Oct 2005 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/device.h> -+#include <linux/i2c.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/mainstone.h> -+#include <asm/arch/audio.h> -+ -+#include "../codecs/ac97.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine mainstone; -+static long mst_audio_suspend_mask; -+ -+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ mst_audio_suspend_mask = MST_MSCWR2; -+ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_resume(struct platform_device *pdev) -+{ -+ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_probe(struct platform_device *pdev) -+{ -+ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_remove(struct platform_device *pdev) -+{ -+ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static struct snd_soc_machine_config codecs[] = { -+{ -+ .name = "AC97", -+ .sname = "AC97 HiFi", -+ .iface = &pxa_ac97_interface[0], -+}, -+{ -+ .name = "AC97 Aux", -+ .sname = "AC97 Aux", -+ .iface = &pxa_ac97_interface[1], -+}, -+}; -+ -+static struct snd_soc_machine mainstone = { -+ .name = "Mainstone", -+ .probe = mainstone_probe, -+ .remove = mainstone_remove, -+ .suspend_pre = mainstone_suspend, -+ .resume_post = mainstone_resume, -+ .config = codecs, -+ .nconfigs = ARRAY_SIZE(codecs), -+}; -+ -+static struct snd_soc_device mainstone_snd_devdata = { -+ .machine = &mainstone, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_ac97, -+}; -+ -+static struct platform_device *mainstone_snd_device; -+ -+static int __init mainstone_init(void) -+{ -+ int ret; -+ -+ mainstone_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!mainstone_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata); -+ mainstone_snd_devdata.dev = &mainstone_snd_device->dev; -+ ret = platform_device_add(mainstone_snd_device); -+ -+ if (ret) -+ platform_device_put(mainstone_snd_device); -+ -+ return ret; -+} -+ -+static void __exit mainstone_exit(void) -+{ -+ platform_device_unregister(mainstone_snd_device); -+} -+ -+module_init(mainstone_init); -+module_exit(mainstone_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("ALSA SoC Mainstone"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_baseband.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_baseband.c -@@ -0,0 +1,249 @@ -+/* -+ * mainstone_baseband.c -+ * Mainstone Example Baseband modem -- ALSA Soc Audio Layer -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 15th Apr 2006 Initial version. -+ * -+ * This is example code to demonstrate connecting a baseband modem to the PCM -+ * DAI on the WM9713 codec on the Intel Mainstone platform. It is by no means -+ * complete as it requires code to control the modem. -+ * -+ * The architecture consists of the WM9713 AC97 DAI connected to the PXA27x -+ * AC97 controller and the WM9713 PCM DAI connected to the basebands DAI. The -+ * baseband is controlled via a serial port. Audio is routed between the PXA27x -+ * and the baseband via internal WM9713 analog paths. -+ * -+ * This driver is not the baseband modem driver. This driver only calls -+ * functions from the Baseband driver to set up it's PCM DAI. -+ * -+ * It's intended to use this driver as follows:- -+ * -+ * 1. open() WM9713 PCM audio device. -+ * 2. open() serial device (for AT commands). -+ * 3. configure PCM audio device (rate etc) - sets up WM9713 PCM DAI, -+ * this will also set up the baseband PCM DAI (via calling baseband driver). -+ * 4. send any further AT commands to set up baseband. -+ * 5. configure codec audio mixer paths. -+ * 6. open(), configure and read/write AC97 audio device - to Tx/Rx voice -+ * -+ * The PCM audio device is opened but IO is never performed on it as the IO is -+ * directly between the codec and the baseband (and not the CPU). -+ * -+ * TODO: -+ * o Implement callbacks -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/platform_device.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/hardware.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/audio.h> -+#include <asm/arch/ssp.h> -+ -+#include "../codecs/wm9713.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine mainstone; -+ -+#define BASEBAND_XXX_DAIFMT \ -+ (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS |\ -+ SND_SOC_DAIFMT_NB_NF) -+ -+#define BASEBAND_XXX_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+/* -+ * PCM modes - 8k 16bit mono baseband modem is master -+ */ -+static struct snd_soc_dai_mode mainstone_example_modes[] = { -+ /* port master clk & frame modes */ -+ {BASEBAND_XXX_DAIFMT, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE, -+ SNDRV_PCM_RATE_8000, BASEBAND_XXX_DIR, SND_SOC_DAI_BFS_RATE, 256, 64}, -+}; -+ -+/* Do specific baseband PCM voice startup here */ -+static int mainstone_baseband_startup(struct snd_pcm_substream *substream) -+{ -+ return 0; -+} -+ -+/* Do specific baseband PCM voice shutdown here */ -+static void mainstone_baseband_shutdown (struct snd_pcm_substream *substream) -+{ -+} -+ -+/* Do specific baseband modem PCM voice hw params init here */ -+static int mainstone_baseband_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ return 0; -+} -+ -+/* Do specific baseband modem PCM voice hw params free here */ -+static int mainstone_baseband_hw_free(struct snd_pcm_substream *substream) -+{ -+ return 0; -+} -+ -+static struct snd_soc_cpu_dai mainstone_example_dai[] = { -+ { .name = "Baseband", -+ .id = 0, -+ .type = SND_SOC_DAI_PCM, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .ops = { -+ .startup = mainstone_baseband_startup, -+ .shutdown = mainstone_baseband_shutdown, -+ .hw_params = mainstone_baseband_hw_params, -+ .hw_free = mainstone_baseband_hw_free, -+ }, -+ .caps = { -+ .mode = mainstone_example_modes, -+ .num_modes = ARRAY_SIZE(mainstone_example_modes),}, -+ }, -+}; -+ -+/* do we need to do any thing on the mainstone when the stream is -+ * started and stopped -+ */ -+static int mainstone_startup(struct snd_pcm_substream *substream) -+{ -+ return 0; -+} -+ -+static void mainstone_shutdown(struct snd_pcm_substream *substream) -+{ -+} -+ -+static struct snd_soc_ops mainstone_ops = { -+ .startup = mainstone_startup, -+ .shutdown = mainstone_shutdown, -+}; -+ -+/* PM */ -+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ return 0; -+} -+ -+static int mainstone_resume(struct platform_device *pdev) -+{ -+ return 0; -+} -+ -+static int mainstone_probe(struct platform_device *pdev) -+{ -+ return 0; -+} -+ -+static int mainstone_remove(struct platform_device *pdev) -+{ -+ return 0; -+} -+ -+static int mainstone_wm9713_init(struct snd_soc_codec *codec) -+{ -+ return 0; -+} -+ -+unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ /* wm8753 has pll that generates mclk from 13MHz xtal */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000); -+} -+ -+/* the physical audio connections between the WM9713, Baseband and pxa2xx */ -+static struct snd_soc_dai_link mainstone_dai[] = { -+{ -+ .name = "AC97", -+ .stream_name = "AC97 HiFi", -+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], -+ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], -+ .init = mainstone_wm9713_init, -+}, -+{ -+ .name = "AC97 Aux", -+ .stream_name = "AC97 Aux", -+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], -+ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], -+}, -+{ -+ .name = "Baseband", -+ .stream_name = "Voice", -+ .cpu_dai = mainstone_example_dai, -+ .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], -+ .config_sysclk = mainstone_config_sysclk, -+}, -+}; -+ -+static struct snd_soc_machine mainstone = { -+ .name = "Mainstone", -+ .probe = mainstone_probe, -+ .remove = mainstone_remove, -+ .suspend_pre = mainstone_suspend, -+ .resume_post = mainstone_resume, -+ .ops = &mainstone_ops, -+ .dai_link = mainstone_dai, -+ .num_links = ARRAY_SIZE(mainstone_dai), -+}; -+ -+static struct snd_soc_device mainstone_snd_ac97_devdata = { -+ .machine = &mainstone, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm9713, -+}; -+ -+static struct platform_device *mainstone_snd_ac97_device; -+ -+static int __init mainstone_init(void) -+{ -+ int ret; -+ -+ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1); -+ if (!mainstone_snd_ac97_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata); -+ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev; -+ -+ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0) -+ platform_device_put(mainstone_snd_ac97_device); -+ -+ return ret; -+} -+ -+static void __exit mainstone_exit(void) -+{ -+ platform_device_unregister(mainstone_snd_ac97_device); -+} -+ -+module_init(mainstone_init); -+module_exit(mainstone_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("Mainstone Example Baseband PCM Interface"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_bluetooth.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_bluetooth.c -@@ -0,0 +1,399 @@ -+/* -+ * mainstone_bluetooth.c -+ * Mainstone Example Bluetooth -- ALSA Soc Audio Layer -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 15th May 2006 Initial version. -+ * -+ * This is example code to demonstrate connecting a bluetooth codec to the PCM -+ * DAI on the WM8753 codec on the Intel Mainstone platform. It is by no means -+ * complete as it requires code to control the BT codec. -+ * -+ * The architecture consists of the WM8753 HIFI DAI connected to the PXA27x -+ * I2S controller and the WM8753 PCM DAI connected to the bluetooth DAI. The -+ * bluetooth codec and wm8753 are controlled via I2C. Audio is routed between -+ * the PXA27x and the bluetooth via internal WM8753 analog paths. -+ * -+ * This example supports the following audio input/outputs. -+ * -+ * o Board mounted Mic and Speaker (spk has amplifier) -+ * o Headphones via jack socket -+ * o BT source and sink -+ * -+ * This driver is not the bluetooth codec driver. This driver only calls -+ * functions from the Bluetooth driver to set up it's PCM DAI. -+ * -+ * It's intended to use the driver as follows:- -+ * -+ * 1. open() WM8753 PCM audio device. -+ * 2. configure PCM audio device (rate etc) - sets up WM8753 PCM DAI, -+ * this should also set up the BT codec DAI (via calling bt driver). -+ * 3. configure codec audio mixer paths. -+ * 4. open(), configure and read/write HIFI audio device - to Tx/Rx voice -+ * -+ * The PCM audio device is opened but IO is never performed on it as the IO is -+ * directly between the codec and the BT codec (and not the CPU). -+ * -+ * TODO: -+ * o Implement callbacks -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/platform_device.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/hardware.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/audio.h> -+#include <asm/arch/ssp.h> -+ -+#include "../codecs/wm8753.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine mainstone; -+ -+#define BLUETOOTH_DAIFMT \ -+ (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS |\ -+ SND_SOC_DAIFMT_NB_NF) -+ -+#define BLUETOOTH_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+/* -+ * PCM modes - 8k 16bit mono BT codec is master -+ */ -+static struct snd_soc_dai_mode mainstone_bt_modes[] = { -+ /* port master clk & frame modes */ -+ {BLUETOOTH_DAIFMT, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE, -+ SNDRV_PCM_RATE_8000, BLUETOOTH_DIR, SND_SOC_DAI_BFS_RATE, 256, 64}, -+}; -+ -+/* Do specific bluetooth PCM startup here */ -+static int mainstone_bt_startup(struct snd_pcm_substream *substream) -+{ -+ return 0; -+} -+ -+/* Do specific bluetooth PCM shutdown here */ -+static void mainstone_bt_shutdown (struct snd_pcm_substream *substream) -+{ -+} -+ -+/* Do pecific bluetooth PCM hw params init here */ -+static int mainstone_bt_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ return 0; -+} -+ -+/* Do specific bluetooth PCM hw params free here */ -+static int mainstone_bt_hw_free(struct snd_pcm_substream *substream) -+{ -+ return 0; -+} -+ -+static struct snd_soc_cpu_dai mainstone_bt_dai[] = { -+ { .name = "Bluetooth", -+ .id = 0, -+ .type = SND_SOC_DAI_PCM, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .ops = { -+ .startup = mainstone_bt_startup, -+ .shutdown = mainstone_bt_shutdown, -+ .hw_params = mainstone_bt_hw_params, -+ .hw_free = mainstone_bt_hw_free, -+ }, -+ .caps = { -+ .mode = mainstone_bt_modes, -+ .num_modes = ARRAY_SIZE(mainstone_bt_modes),}, -+ }, -+}; -+ -+/* do we need to do any thing on the mainstone when the stream is -+ * started and stopped -+ */ -+static int mainstone_startup(struct snd_pcm_substream *substream) -+{ -+ return 0; -+} -+ -+static void mainstone_shutdown(struct snd_pcm_substream *substream) -+{ -+} -+ -+static struct snd_soc_ops mainstone_ops = { -+ .startup = mainstone_startup, -+ .shutdown = mainstone_shutdown, -+}; -+ -+/* PM */ -+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ return 0; -+} -+ -+static int mainstone_resume(struct platform_device *pdev) -+{ -+ return 0; -+} -+ -+static int mainstone_probe(struct platform_device *pdev) -+{ -+ return 0; -+} -+ -+static int mainstone_remove(struct platform_device *pdev) -+{ -+ return 0; -+} -+ -+/* -+ * Machine audio functions. -+ * -+ * The machine now has 3 extra audio controls. -+ * -+ * Jack function: Sets function (device plugged into Jack) to nothing (Off) -+ * or Headphones. -+ * -+ * Mic function: Set the on board Mic to On or Off -+ * Spk function: Set the on board Spk to On or Off -+ * -+ * example: BT playback (of far end) and capture (of near end) -+ * Set Mic and Speaker to On, open BT alsa interface as above and set up -+ * internal audio paths. -+ */ -+ -+static int machine_jack_func = 0; -+static int machine_spk_func = 0; -+static int machine_mic_func = 0; -+ -+static int machine_get_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = machine_jack_func; -+ return 0; -+} -+ -+static int machine_set_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ machine_jack_func = ucontrol->value.integer.value[0]; -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", machine_jack_func); -+ return 0; -+} -+ -+static int machine_get_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = machine_spk_func; -+ return 0; -+} -+ -+static int machine_set_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ machine_spk_func = ucontrol->value.integer.value[0]; -+ snd_soc_dapm_set_endpoint(codec, "Spk", machine_spk_func); -+ return 0; -+} -+ -+static int machine_get_mic(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = machine_spk_func; -+ return 0; -+} -+ -+static int machine_set_mic(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ machine_spk_func = ucontrol->value.integer.value[0]; -+ snd_soc_dapm_set_endpoint(codec, "Mic", machine_mic_func); -+ return 0; -+} -+ -+/* turns on board speaker amp on/off */ -+static int machine_amp_event(struct snd_soc_dapm_widget *w, int event) -+{ -+#if 0 -+ if (SND_SOC_DAPM_EVENT_ON(event)) -+ /* on */ -+ else -+ /* off */ -+#endif -+ return 0; -+} -+ -+/* machine dapm widgets */ -+static const struct snd_soc_dapm_widget machine_dapm_widgets[] = { -+SND_SOC_DAPM_HP("Headphone Jack", NULL), -+SND_SOC_DAPM_SPK("Spk", machine_amp_event), -+SND_SOC_DAPM_MIC("Mic", NULL), -+}; -+ -+/* machine connections to the codec pins */ -+static const char* audio_map[][3] = { -+ -+ /* headphone connected to LOUT1, ROUT1 */ -+ {"Headphone Jack", NULL, "LOUT"}, -+ {"Headphone Jack", NULL, "ROUT"}, -+ -+ /* speaker connected to LOUT2, ROUT2 */ -+ {"Spk", NULL, "ROUT2"}, -+ {"Spk", NULL, "LOUT2"}, -+ -+ /* mic is connected to MIC1 (via Mic Bias) */ -+ {"MIC1", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Mic"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+static const char* jack_function[] = {"Off", "Headphone"}; -+static const char* spk_function[] = {"Off", "On"}; -+static const char* mic_function[] = {"Off", "On"}; -+static const struct soc_enum machine_ctl_enum[] = { -+ SOC_ENUM_SINGLE_EXT(2, jack_function), -+ SOC_ENUM_SINGLE_EXT(2, spk_function), -+ SOC_ENUM_SINGLE_EXT(2, mic_function), -+}; -+ -+static const struct snd_kcontrol_new wm8753_machine_controls[] = { -+ SOC_ENUM_EXT("Jack Function", machine_ctl_enum[0], machine_get_jack, machine_set_jack), -+ SOC_ENUM_EXT("Speaker Function", machine_ctl_enum[1], machine_get_spk, machine_set_spk), -+ SOC_ENUM_EXT("Mic Function", machine_ctl_enum[2], machine_get_mic, machine_set_mic), -+}; -+ -+static int mainstone_wm8753_init(struct snd_soc_codec *codec) -+{ -+ int i, err; -+ -+ /* not used on this machine - e.g. will never be powered up */ -+ snd_soc_dapm_set_endpoint(codec, "OUT3", 0); -+ snd_soc_dapm_set_endpoint(codec, "OUT4", 0); -+ snd_soc_dapm_set_endpoint(codec, "MONO2", 0); -+ snd_soc_dapm_set_endpoint(codec, "MONO1", 0); -+ snd_soc_dapm_set_endpoint(codec, "LINE1", 0); -+ snd_soc_dapm_set_endpoint(codec, "LINE2", 0); -+ snd_soc_dapm_set_endpoint(codec, "RXP", 0); -+ snd_soc_dapm_set_endpoint(codec, "RXN", 0); -+ snd_soc_dapm_set_endpoint(codec, "MIC2", 0); -+ -+ /* Add machine specific controls */ -+ for (i = 0; i < ARRAY_SIZE(wm8753_machine_controls); i++) { -+ if ((err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8753_machine_controls[i],codec, NULL))) < 0) -+ return err; -+ } -+ -+ /* Add machine specific widgets */ -+ for(i = 0; i < ARRAY_SIZE(machine_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &machine_dapm_widgets[i]); -+ } -+ -+ /* Set up machine specific audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ return 0; -+} -+ -+/* this configures the clocking between the WM8753 and the BT codec */ -+unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ /* wm8753 has pll that generates mclk from 13MHz xtal */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000); -+} -+ -+static struct snd_soc_dai_link mainstone_dai[] = { -+{ /* Hifi Playback - for similatious use with voice below */ -+ .name = "WM8753", -+ .stream_name = "WM8753 HiFi", -+ .cpu_dai = &pxa_i2s_dai, -+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], -+ .init = mainstone_wm8753_init, -+ .config_sysclk = mainstone_config_sysclk, -+}, -+{ /* Voice via BT */ -+ .name = "Bluetooth", -+ .stream_name = "Voice", -+ .cpu_dai = mainstone_bt_dai, -+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], -+ .config_sysclk = mainstone_config_sysclk, -+}, -+}; -+ -+static struct snd_soc_machine mainstone = { -+ .name = "Mainstone", -+ .probe = mainstone_probe, -+ .remove = mainstone_remove, -+ .suspend_pre = mainstone_suspend, -+ .resume_post = mainstone_resume, -+ .ops = &mainstone_ops, -+ .dai_link = mainstone_dai, -+ .num_links = ARRAY_SIZE(mainstone_dai), -+}; -+ -+static struct snd_soc_device mainstone_snd_wm8753_devdata = { -+ .machine = &mainstone, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8753, -+}; -+ -+static struct platform_device *mainstone_snd_wm8753_device; -+ -+static int __init mainstone_init(void) -+{ -+ int ret; -+ -+ mainstone_snd_wm8753_device = platform_device_alloc("soc-audio", -1); -+ if (!mainstone_snd_wm8753_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(mainstone_snd_wm8753_device, &mainstone_snd_wm8753_devdata); -+ mainstone_snd_wm8753_devdata.dev = &mainstone_snd_wm8753_device->dev; -+ -+ if((ret = platform_device_add(mainstone_snd_wm8753_device)) != 0) -+ platform_device_put(mainstone_snd_wm8753_device); -+ -+ return ret; -+} -+ -+static void __exit mainstone_exit(void) -+{ -+ platform_device_unregister(mainstone_snd_wm8753_device); -+} -+ -+module_init(mainstone_init); -+module_exit(mainstone_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("Mainstone Example Bluetooth PCM Interface"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8731.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8731.c -@@ -0,0 +1,156 @@ -+/* -+ * mainstone.c -- SoC audio for Mainstone -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c -+ * Copyright: MontaVista Software Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 5th June 2006 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/device.h> -+#include <linux/i2c.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/mainstone.h> -+#include <asm/arch/audio.h> -+ -+#include "../codecs/wm8731.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine mainstone; -+ -+ -+static const struct snd_soc_dapm_widget dapm_widgets[] = { -+ SND_SOC_DAPM_MIC("Int Mic", NULL), -+ SND_SOC_DAPM_SPK("Ext Spk", NULL), -+}; -+ -+static const char* intercon[][3] = { -+ -+ /* speaker connected to LHPOUT */ -+ {"Ext Spk", NULL, "LHPOUT"}, -+ -+ /* mic is connected to Mic Jack, with WM8731 Mic Bias */ -+ {"MICIN", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Int Mic"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+/* -+ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. -+ */ -+static int mainstone_wm8731_init(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ -+ /* Add specific widgets */ -+ for(i = 0; i < ARRAY_SIZE(dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &dapm_widgets[i]); -+ } -+ -+ /* Set up specific audio path interconnects */ -+ for(i = 0; intercon[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]); -+ } -+ -+ /* not connected */ -+ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); -+ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); -+ -+ /* always connected */ -+ snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); -+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ -+ return 0; -+} -+ -+unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ /* we have a 12.288MHz crystal */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12288000); -+} -+ -+static struct snd_soc_dai_link mainstone_dai[] = { -+{ -+ .name = "WM8731", -+ .stream_name = "WM8731 HiFi", -+ .cpu_dai = &pxa_i2s_dai, -+ .codec_dai = &wm8731_dai, -+ .init = mainstone_wm8731_init, -+ .config_sysclk = mainstone_config_sysclk, -+}, -+}; -+ -+static struct snd_soc_machine mainstone = { -+ .name = "Mainstone", -+ .dai_link = mainstone_dai, -+ .num_links = ARRAY_SIZE(mainstone_dai), -+}; -+ -+static struct wm8731_setup_data corgi_wm8731_setup = { -+ .i2c_address = 0x1b, -+}; -+ -+static struct snd_soc_device mainstone_snd_devdata = { -+ .machine = &mainstone, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8731, -+ .codec_data = &corgi_wm8731_setup, -+}; -+ -+static struct platform_device *mainstone_snd_device; -+ -+static int __init mainstone_init(void) -+{ -+ int ret; -+ -+ mainstone_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!mainstone_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata); -+ mainstone_snd_devdata.dev = &mainstone_snd_device->dev; -+ ret = platform_device_add(mainstone_snd_device); -+ -+ if (ret) -+ platform_device_put(mainstone_snd_device); -+ -+ return ret; -+} -+ -+static void __exit mainstone_exit(void) -+{ -+ platform_device_unregister(mainstone_snd_device); -+} -+ -+module_init(mainstone_init); -+module_exit(mainstone_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("ALSA SoC WM8731 Mainstone"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8753.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8753.c -@@ -0,0 +1,226 @@ -+/* -+ * mainstone.c -- SoC audio for Mainstone -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c -+ * Copyright: MontaVista Software Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 30th Oct 2005 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/device.h> -+#include <linux/i2c.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/mainstone.h> -+#include <asm/arch/audio.h> -+ -+#include "../codecs/wm8753.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine mainstone; -+ -+static int mainstone_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) { -+ /* enable USB on the go MUX so we can use SSPFRM2 */ -+ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_SEL; -+ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_RST; -+ } -+ return 0; -+} -+ -+static void mainstone_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) { -+ /* disable USB on the go MUX so we can use ttyS0 */ -+ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_SEL; -+ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_RST; -+ } -+} -+ -+static struct snd_soc_ops mainstone_ops = { -+ .startup = mainstone_startup, -+ .shutdown = mainstone_shutdown, -+}; -+ -+static long mst_audio_suspend_mask; -+ -+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ mst_audio_suspend_mask = MST_MSCWR2; -+ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_resume(struct platform_device *pdev) -+{ -+ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_probe(struct platform_device *pdev) -+{ -+ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_remove(struct platform_device *pdev) -+{ -+ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+/* example machine audio_mapnections */ -+static const char* audio_map[][3] = { -+ -+ /* mic is connected to mic1 - with bias */ -+ {"MIC1", NULL, "Mic Bias"}, -+ {"MIC1N", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Mic1 Jack"}, -+ {"Mic Bias", NULL, "Mic1 Jack"}, -+ -+ {"ACIN", NULL, "ACOP"}, -+ {NULL, NULL, NULL}, -+}; -+ -+/* headphone detect support on my board */ -+static const char * hp_pol[] = {"Headphone", "Speaker"}; -+static const struct soc_enum wm8753_enum = -+ SOC_ENUM_SINGLE(WM8753_OUTCTL, 1, 2, hp_pol); -+ -+static const struct snd_kcontrol_new wm8753_mainstone_controls[] = { -+ SOC_SINGLE("Headphone Detect Switch", WM8753_OUTCTL, 6, 1, 0), -+ SOC_ENUM("Headphone Detect Polarity", wm8753_enum), -+}; -+ -+/* -+ * This is an example machine initialisation for a wm8753 connected to a -+ * Mainstone II. It is missing logic to detect hp/mic insertions and logic -+ * to re-route the audio in such an event. -+ */ -+static int mainstone_wm8753_init(struct snd_soc_codec *codec) -+{ -+ int i, err; -+ -+ /* set up mainstone codec pins */ -+ snd_soc_dapm_set_endpoint(codec, "RXP", 0); -+ snd_soc_dapm_set_endpoint(codec, "RXN", 0); -+ snd_soc_dapm_set_endpoint(codec, "MIC2", 0); -+ -+ /* add mainstone specific controls */ -+ for (i = 0; i < ARRAY_SIZE(wm8753_mainstone_controls); i++) { -+ if ((err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8753_mainstone_controls[i],codec, NULL))) < 0) -+ return err; -+ } -+ -+ /* set up mainstone specific audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ return 0; -+} -+ -+unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ /* wm8753 has pll that generates mclk from 13MHz xtal */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000); -+} -+ -+static struct snd_soc_dai_link mainstone_dai[] = { -+{ /* Hifi Playback - for similatious use with voice below */ -+ .name = "WM8753", -+ .stream_name = "WM8753 HiFi", -+ .cpu_dai = &pxa_i2s_dai, -+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], -+ .init = mainstone_wm8753_init, -+ .config_sysclk = mainstone_config_sysclk, -+}, -+{ /* Voice via BT */ -+ .name = "Bluetooth", -+ .stream_name = "Voice", -+ .cpu_dai = &pxa_ssp_dai[1], -+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], -+ .config_sysclk = mainstone_config_sysclk, -+}, -+}; -+ -+static struct snd_soc_machine mainstone = { -+ .name = "Mainstone", -+ .probe = mainstone_probe, -+ .remove = mainstone_remove, -+ .suspend_pre = mainstone_suspend, -+ .resume_post = mainstone_resume, -+ .ops = &mainstone_ops, -+ .dai_link = mainstone_dai, -+ .num_links = ARRAY_SIZE(mainstone_dai), -+}; -+ -+static struct wm8753_setup_data mainstone_wm8753_setup = { -+ .i2c_address = 0x1a, -+}; -+ -+static struct snd_soc_device mainstone_snd_devdata = { -+ .machine = &mainstone, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8753, -+ .codec_data = &mainstone_wm8753_setup, -+}; -+ -+static struct platform_device *mainstone_snd_device; -+ -+static int __init mainstone_init(void) -+{ -+ int ret; -+ -+ mainstone_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!mainstone_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata); -+ mainstone_snd_devdata.dev = &mainstone_snd_device->dev; -+ ret = platform_device_add(mainstone_snd_device); -+ -+ if (ret) -+ platform_device_put(mainstone_snd_device); -+ -+ return ret; -+} -+ -+static void __exit mainstone_exit(void) -+{ -+ platform_device_unregister(mainstone_snd_device); -+} -+ -+module_init(mainstone_init); -+module_exit(mainstone_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("ALSA SoC WM8753 Mainstone"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8974.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8974.c -@@ -0,0 +1,112 @@ -+/* -+ * mainstone.c -- SoC audio for Mainstone -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c -+ * Copyright: MontaVista Software Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 30th Oct 2005 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/device.h> -+#include <linux/i2c.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/mainstone.h> -+#include <asm/arch/audio.h> -+ -+#include "../codecs/wm8974.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine mainstone; -+ -+static int mainstone_wm8974_init(struct snd_soc_codec *codec) -+{ -+ return 0; -+} -+ -+unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ /* we have a PLL */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12288000); -+ -+} -+ -+static struct snd_soc_dai_link mainstone_dai[] = { -+{ -+ .name = "WM8974", -+ .stream_name = "WM8974 HiFi", -+ .cpu_dai = &pxa_i2s_dai, -+ .codec_dai = &wm8974_dai, -+ .init = mainstone_wm8974_init, -+ .config_sysclk = mainstone_config_sysclk, -+}, -+}; -+ -+static struct snd_soc_machine mainstone = { -+ .name = "Mainstone", -+ .dai_link = mainstone_dai, -+ .num_links = ARRAY_SIZE(mainstone_dai), -+}; -+ -+static struct wm8974_setup_data mainstone_wm8974_setup = { -+ .i2c_address = 0x1a, -+}; -+ -+static struct snd_soc_device mainstone_snd_devdata = { -+ .machine = &mainstone, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8974, -+ .codec_data = &mainstone_wm8974_setup, -+}; -+ -+static struct platform_device *mainstone_snd_device; -+ -+static int __init mainstone_init(void) -+{ -+ int ret; -+ -+ mainstone_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!mainstone_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata); -+ mainstone_snd_devdata.dev = &mainstone_snd_device->dev; -+ ret = platform_device_add(mainstone_snd_device); -+ -+ if (ret) -+ platform_device_put(mainstone_snd_device); -+ -+ return ret; -+} -+ -+static void __exit mainstone_exit(void) -+{ -+ platform_device_unregister(mainstone_snd_device); -+} -+ -+module_init(mainstone_init); -+module_exit(mainstone_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("ALSA SoC Mainstone"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9712.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9712.c -@@ -0,0 +1,171 @@ -+/* -+ * mainstone.c -- SoC audio for Mainstone -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c -+ * Copyright: MontaVista Software Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 29th Jan 2006 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/device.h> -+#include <linux/i2c.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/mainstone.h> -+#include <asm/arch/audio.h> -+ -+#include "../codecs/wm9712.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine mainstone; -+static long mst_audio_suspend_mask; -+ -+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ mst_audio_suspend_mask = MST_MSCWR2; -+ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_resume(struct platform_device *pdev) -+{ -+ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_probe(struct platform_device *pdev) -+{ -+ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_remove(struct platform_device *pdev) -+{ -+ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+/* mainstone machine dapm widgets */ -+static const struct snd_soc_dapm_widget mainstone_dapm_widgets[] = { -+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL), -+}; -+ -+/* example machine interconnections */ -+static const char* intercon[][3] = { -+ -+ /* mic is connected to mic1 - with bias */ -+ {"MIC1", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Mic (Internal)"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+/* -+ * This is an example machine initialisation for a wm8753 connected to a -+ * Mainstone II. It is missing logic to detect hp/mic insertions and logic -+ * to re-route the audio in such an event. -+ */ -+static int mainstone_wm9712_init(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ /* set up mainstone codec pins */ -+ snd_soc_dapm_set_endpoint(codec, "RXP", 0); -+ snd_soc_dapm_set_endpoint(codec, "RXN", 0); -+ //snd_soc_dapm_set_endpoint(codec, "MIC2", 0); -+ -+ /* Add mainstone specific widgets */ -+ for(i = 0; i < ARRAY_SIZE(mainstone_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &mainstone_dapm_widgets[i]); -+ } -+ -+ /* set up mainstone specific audio path interconnects */ -+ for(i = 0; intercon[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]); -+ } -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ return 0; -+} -+ -+static struct snd_soc_dai_link mainstone_dai[] = { -+{ -+ .name = "AC97", -+ .stream_name = "AC97 HiFi", -+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], -+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], -+ .init = mainstone_wm9712_init, -+}, -+{ -+ .name = "AC97 Aux", -+ .stream_name = "AC97 Aux", -+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], -+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], -+}, -+}; -+ -+static struct snd_soc_machine mainstone = { -+ .name = "Mainstone", -+ .probe = mainstone_probe, -+ .remove = mainstone_remove, -+ .suspend_pre = mainstone_suspend, -+ .resume_post = mainstone_resume, -+ .dai_link = mainstone_dai, -+ .num_links = ARRAY_SIZE(mainstone_dai), -+}; -+ -+static struct snd_soc_device mainstone_snd_ac97_devdata = { -+ .machine = &mainstone, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm9712, -+}; -+ -+static struct platform_device *mainstone_snd_ac97_device; -+ -+static int __init mainstone_init(void) -+{ -+ int ret; -+ -+ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1); -+ if (!mainstone_snd_ac97_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata); -+ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev; -+ -+ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0) -+ platform_device_put(mainstone_snd_ac97_device); -+ -+ return ret; -+} -+ -+static void __exit mainstone_exit(void) -+{ -+ platform_device_unregister(mainstone_snd_ac97_device); -+} -+ -+module_init(mainstone_init); -+module_exit(mainstone_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("ALSA SoC WM9712 Mainstone"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9713.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9713.c -@@ -0,0 +1,263 @@ -+/* -+ * mainstone.c -- SoC audio for Mainstone -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c -+ * Copyright: MontaVista Software Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 29th Jan 2006 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/device.h> -+#include <linux/i2c.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/mainstone.h> -+#include <asm/arch/audio.h> -+ -+#include "../codecs/wm9713.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine mainstone; -+ -+static int mainstone_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) { -+ /* enable USB on the go MUX so we can use SSPFRM2 */ -+ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_SEL; -+ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_RST; -+ } -+ return 0; -+} -+ -+static void mainstone_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) { -+ /* disable USB on the go MUX so we can use ttyS0 */ -+ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_SEL; -+ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_RST; -+ } -+} -+ -+static struct snd_soc_ops mainstone_ops = { -+ .startup = mainstone_startup, -+ .shutdown = mainstone_shutdown, -+}; -+ -+static int test = 0; -+static int get_test(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = test; -+ return 0; -+} -+ -+static int set_test(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ test = ucontrol->value.integer.value[0]; -+ if(test) { -+ -+ } else { -+ -+ } -+ return 0; -+} -+ -+static long mst_audio_suspend_mask; -+ -+static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ mst_audio_suspend_mask = MST_MSCWR2; -+ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_resume(struct platform_device *pdev) -+{ -+ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_probe(struct platform_device *pdev) -+{ -+ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static int mainstone_remove(struct platform_device *pdev) -+{ -+ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; -+ return 0; -+} -+ -+static const char* test_function[] = {"Off", "On"}; -+static const struct soc_enum mainstone_enum[] = { -+ SOC_ENUM_SINGLE_EXT(2, test_function), -+}; -+ -+static const struct snd_kcontrol_new mainstone_controls[] = { -+ SOC_ENUM_EXT("ATest Function", mainstone_enum[0], get_test, set_test), -+}; -+ -+/* mainstone machine dapm widgets */ -+static const struct snd_soc_dapm_widget mainstone_dapm_widgets[] = { -+ SND_SOC_DAPM_MIC("Mic 1", NULL), -+ SND_SOC_DAPM_MIC("Mic 2", NULL), -+ SND_SOC_DAPM_MIC("Mic 3", NULL), -+}; -+ -+/* example machine audio_mapnections */ -+static const char* audio_map[][3] = { -+ -+ /* mic is connected to mic1 - with bias */ -+ {"MIC1", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Mic 1"}, -+ /* mic is connected to mic2A - with bias */ -+ {"MIC2A", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Mic 2"}, -+ /* mic is connected to mic2B - with bias */ -+ {"MIC2B", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Mic 3"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+/* -+ * This is an example machine initialisation for a wm9713 connected to a -+ * Mainstone II. It is missing logic to detect hp/mic insertions and logic -+ * to re-route the audio in such an event. -+ */ -+static int mainstone_wm9713_init(struct snd_soc_codec *codec) -+{ -+ int i, err; -+ -+ /* set up mainstone codec pins */ -+ snd_soc_dapm_set_endpoint(codec, "RXP", 0); -+ snd_soc_dapm_set_endpoint(codec, "RXN", 0); -+ //snd_soc_dapm_set_endpoint(codec, "MIC2", 0); -+ -+ /* Add test specific controls */ -+ for (i = 0; i < ARRAY_SIZE(mainstone_controls); i++) { -+ if ((err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&mainstone_controls[i],codec, NULL))) < 0) -+ return err; -+ } -+ -+ /* Add mainstone specific widgets */ -+ for(i = 0; i < ARRAY_SIZE(mainstone_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &mainstone_dapm_widgets[i]); -+ } -+ -+ /* set up mainstone specific audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ return 0; -+} -+ -+/* configure the system audio clock */ -+unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 24576000); -+} -+ -+static struct snd_soc_dai_link mainstone_dai[] = { -+{ -+ .name = "AC97", -+ .stream_name = "AC97 HiFi", -+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], -+ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], -+ .init = mainstone_wm9713_init, -+ .config_sysclk = mainstone_config_sysclk, -+}, -+{ -+ .name = "AC97 Aux", -+ .stream_name = "AC97 Aux", -+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], -+ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], -+ .config_sysclk = mainstone_config_sysclk, -+}, -+{ -+ .name = "WM9713", -+ .stream_name = "WM9713 Voice", -+ .cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2], -+ .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], -+ .config_sysclk = mainstone_config_sysclk, -+}, -+}; -+ -+static struct snd_soc_machine mainstone = { -+ .name = "Mainstone", -+ .probe = mainstone_probe, -+ .remove = mainstone_remove, -+ .suspend_pre = mainstone_suspend, -+ .resume_post = mainstone_resume, -+ .ops = &mainstone_ops, -+ .dai_link = mainstone_dai, -+ .num_links = ARRAY_SIZE(mainstone_dai), -+}; -+ -+static struct snd_soc_device mainstone_snd_ac97_devdata = { -+ .machine = &mainstone, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm9713, -+}; -+ -+static struct platform_device *mainstone_snd_ac97_device; -+ -+static int __init mainstone_init(void) -+{ -+ int ret; -+ -+ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1); -+ if (!mainstone_snd_ac97_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata); -+ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev; -+ -+ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0) -+ platform_device_put(mainstone_snd_ac97_device); -+ -+ return ret; -+} -+ -+static void __exit mainstone_exit(void) -+{ -+ platform_device_unregister(mainstone_snd_ac97_device); -+} -+ -+module_init(mainstone_init); -+module_exit(mainstone_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("ALSA SoC WM9713 Mainstone"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/poodle.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/poodle.c -@@ -0,0 +1,329 @@ -+/* -+ * poodle.c -- SoC audio for Poodle -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> -+ * Richard Purdie <richard@openedhand.com> -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/timer.h> -+#include <linux/interrupt.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/mach-types.h> -+#include <asm/hardware/locomo.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/hardware.h> -+#include <asm/arch/poodle.h> -+#include <asm/arch/audio.h> -+ -+#include "../codecs/wm8731.h" -+#include "pxa2xx-pcm.h" -+ -+#define POODLE_HP 1 -+#define POODLE_HP_OFF 0 -+#define POODLE_SPK_ON 1 -+#define POODLE_SPK_OFF 0 -+ -+ /* audio clock in Hz - rounded from 12.235MHz */ -+#define POODLE_AUDIO_CLOCK 12288000 -+ -+static int poodle_jack_func; -+static int poodle_spk_func; -+ -+static void poodle_ext_control(struct snd_soc_codec *codec) -+{ -+ int spk = 0; -+ -+ /* set up jack connection */ -+ if (poodle_jack_func == POODLE_HP) { -+ /* set = unmute headphone */ -+ locomo_gpio_write(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_MUTE_L, 1); -+ locomo_gpio_write(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_MUTE_R, 1); -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); -+ } else { -+ locomo_gpio_write(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_MUTE_L, 0); -+ locomo_gpio_write(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_MUTE_R, 0); -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); -+ } -+ -+ if (poodle_spk_func == POODLE_SPK_ON) -+ spk = 1; -+ -+ /* set the enpoints to their new connetion states */ -+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); -+ -+ /* signal a DAPM event */ -+ snd_soc_dapm_sync_endpoints(codec); -+} -+ -+static int poodle_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_codec *codec = rtd->socdev->codec; -+ -+ /* check the jack status at stream startup */ -+ poodle_ext_control(codec); -+ return 0; -+} -+ -+/* we need to unmute the HP at shutdown as the mute burns power on poodle */ -+static int poodle_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_codec *codec = rtd->socdev->codec; -+ -+ /* set = unmute headphone */ -+ locomo_gpio_write(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_MUTE_L, 1); -+ locomo_gpio_write(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_MUTE_R, 1); -+ return 0; -+} -+ -+static struct snd_soc_ops poodle_ops = { -+ .startup = poodle_startup, -+ .shutdown = poodle_shutdown, -+}; -+ -+static int poodle_get_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = poodle_jack_func; -+ return 0; -+} -+ -+static int poodle_set_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ if (poodle_jack_func == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ poodle_jack_func = ucontrol->value.integer.value[0]; -+ poodle_ext_control(codec); -+ return 1; -+} -+ -+static int poodle_get_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = poodle_spk_func; -+ return 0; -+} -+ -+static int poodle_set_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ if (poodle_spk_func == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ poodle_spk_func = ucontrol->value.integer.value[0]; -+ poodle_ext_control(codec); -+ return 1; -+} -+ -+static int poodle_amp_event(struct snd_soc_dapm_widget *w, int event) -+{ -+ if (SND_SOC_DAPM_EVENT_ON(event)) -+ locomo_gpio_write(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_AMP_ON, 0); -+ else -+ locomo_gpio_write(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_AMP_ON, 1); -+ -+ return 0; -+} -+ -+/* poodle machine dapm widgets */ -+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { -+SND_SOC_DAPM_HP("Headphone Jack", NULL), -+SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event), -+}; -+ -+/* Corgi machine audio_mapnections to the codec pins */ -+static const char *audio_map[][3] = { -+ -+ /* headphone connected to LHPOUT1, RHPOUT1 */ -+ {"Headphone Jack", NULL, "LHPOUT"}, -+ {"Headphone Jack", NULL, "RHPOUT"}, -+ -+ /* speaker connected to LOUT, ROUT */ -+ {"Ext Spk", NULL, "ROUT"}, -+ {"Ext Spk", NULL, "LOUT"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+static const char *jack_function[] = {"Off", "Headphone"}; -+static const char *spk_function[] = {"Off", "On"}; -+static const struct soc_enum poodle_enum[] = { -+ SOC_ENUM_SINGLE_EXT(2, jack_function), -+ SOC_ENUM_SINGLE_EXT(2, spk_function), -+}; -+ -+static const snd_kcontrol_new_t wm8731_poodle_controls[] = { -+ SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack, -+ poodle_set_jack), -+ SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk, -+ poodle_set_spk), -+}; -+ -+/* -+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device -+ */ -+static int poodle_wm8731_init(struct snd_soc_codec *codec) -+{ -+ int i, err; -+ -+ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); -+ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); -+ snd_soc_dapm_set_endpoint(codec, "MICIN", 1); -+ -+ /* Add poodle specific controls */ -+ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8731_poodle_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ /* Add poodle specific widgets */ -+ for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); -+ } -+ -+ /* Set up poodle specific audio path audio_map */ -+ for (i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ return 0; -+} -+ -+static unsigned int poodle_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) { -+ /* pxa2xx is i2s master */ -+ switch (info->rate) { -+ case 44100: -+ case 88200: -+ /* configure codec digital filters for 44.1, 88.2 */ -+ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ 11289600); -+ break; -+ default: -+ /* configure codec digital filters for all other rates */ -+ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ POODLE_AUDIO_CLOCK); -+ break; -+ } -+ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, -+ POODLE_AUDIO_CLOCK); -+ } else { -+ /* codec is i2s master - -+ * only configure codec DAI clock and filters */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ POODLE_AUDIO_CLOCK); -+ } -+} -+ -+/* poodle digital audio interface glue - connects codec <--> CPU */ -+static struct snd_soc_dai_link poodle_dai = { -+ .name = "WM8731", -+ .stream_name = "WM8731", -+ .cpu_dai = &pxa_i2s_dai, -+ .codec_dai = &wm8731_dai, -+ .init = poodle_wm8731_init, -+ .config_sysclk = poodle_config_sysclk, -+}; -+ -+/* poodle audio machine driver */ -+static struct snd_soc_machine snd_soc_machine_poodle = { -+ .name = "Poodle", -+ .dai_link = &poodle_dai, -+ .num_links = 1, -+ .ops = &poodle_ops, -+}; -+ -+/* poodle audio private data */ -+static struct wm8731_setup_data poodle_wm8731_setup = { -+ .i2c_address = 0x1b, -+}; -+ -+/* poodle audio subsystem */ -+static struct snd_soc_device poodle_snd_devdata = { -+ .machine = &snd_soc_machine_poodle, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8731, -+ .codec_data = &poodle_wm8731_setup, -+}; -+ -+static struct platform_device *poodle_snd_device; -+ -+static int __init poodle_init(void) -+{ -+ int ret; -+ -+ if (!machine_is_poodle()) -+ return -ENODEV; -+ -+ locomo_gpio_set_dir(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_AMP_ON, 0); -+ /* should we mute HP at startup - burning power ?*/ -+ locomo_gpio_set_dir(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_MUTE_L, 0); -+ locomo_gpio_set_dir(&poodle_locomo_device.dev, -+ POODLE_LOCOMO_GPIO_MUTE_R, 0); -+ -+ poodle_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!poodle_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(poodle_snd_device, &poodle_snd_devdata); -+ poodle_snd_devdata.dev = &poodle_snd_device->dev; -+ ret = platform_device_add(poodle_snd_device); -+ -+ if (ret) -+ platform_device_put(poodle_snd_device); -+ -+ return ret; -+} -+ -+static void __exit poodle_exit(void) -+{ -+ platform_device_unregister(poodle_snd_device); -+} -+ -+module_init(poodle_init); -+module_exit(poodle_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Richard Purdie"); -+MODULE_DESCRIPTION("ALSA SoC Poodle"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ac97.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ac97.c -@@ -0,0 +1,437 @@ -+/* -+ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip. -+ * -+ * Author: Nicolas Pitre -+ * Created: Dec 02, 2004 -+ * Copyright: MontaVista Software Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/platform_device.h> -+#include <linux/interrupt.h> -+#include <linux/wait.h> -+#include <linux/delay.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/ac97_codec.h> -+#include <sound/initval.h> -+#include <sound/soc.h> -+ -+#include <asm/irq.h> -+#include <linux/mutex.h> -+#include <asm/hardware.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/audio.h> -+ -+#include "pxa2xx-pcm.h" -+ -+static DEFINE_MUTEX(car_mutex); -+static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); -+static volatile long gsr_bits; -+ -+#define AC97_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define AC97_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -+ -+/* may need to expand this */ -+static struct snd_soc_dai_mode pxa2xx_ac97_modes[] = { -+ { -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = AC97_RATES, -+ .pcmdir = AC97_DIR, -+ }, -+}; -+ -+/* -+ * Beware PXA27x bugs: -+ * -+ * o Slot 12 read from modem space will hang controller. -+ * o CDONE, SDONE interrupt fails after any slot 12 IO. -+ * -+ * We therefore have an hybrid approach for waiting on SDONE (interrupt or -+ * 1 jiffy timeout if interrupt never comes). -+ */ -+ -+static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, -+ unsigned short reg) -+{ -+ unsigned short val = -1; -+ volatile u32 *reg_addr; -+ -+ mutex_lock(&car_mutex); -+ -+ /* set up primary or secondary codec/modem space */ -+#ifdef CONFIG_PXA27x -+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; -+#else -+ if (reg == AC97_GPIO_STATUS) -+ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; -+ else -+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; -+#endif -+ reg_addr += (reg >> 1); -+ -+#ifndef CONFIG_PXA27x -+ if (reg == AC97_GPIO_STATUS) { -+ /* read from controller cache */ -+ val = *reg_addr; -+ goto out; -+ } -+#endif -+ -+ /* start read access across the ac97 link */ -+ GSR = GSR_CDONE | GSR_SDONE; -+ gsr_bits = 0; -+ val = *reg_addr; -+ -+ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); -+ if (!((GSR | gsr_bits) & GSR_SDONE)) { -+ printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n", -+ __FUNCTION__, reg, GSR | gsr_bits); -+ val = -1; -+ goto out; -+ } -+ -+ /* valid data now */ -+ GSR = GSR_CDONE | GSR_SDONE; -+ gsr_bits = 0; -+ val = *reg_addr; -+ /* but we've just started another cycle... */ -+ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); -+ -+out: mutex_unlock(&car_mutex); -+ return val; -+} -+ -+static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, -+ unsigned short val) -+{ -+ volatile u32 *reg_addr; -+ -+ mutex_lock(&car_mutex); -+ -+ /* set up primary or secondary codec/modem space */ -+#ifdef CONFIG_PXA27x -+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; -+#else -+ if (reg == AC97_GPIO_STATUS) -+ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; -+ else -+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; -+#endif -+ reg_addr += (reg >> 1); -+ -+ GSR = GSR_CDONE | GSR_SDONE; -+ gsr_bits = 0; -+ *reg_addr = val; -+ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1); -+ if (!((GSR | gsr_bits) & GSR_CDONE)) -+ printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n", -+ __FUNCTION__, reg, GSR | gsr_bits); -+ -+ mutex_unlock(&car_mutex); -+} -+ -+static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) -+{ -+ gsr_bits = 0; -+ -+#ifdef CONFIG_PXA27x -+ /* warm reset broken on Bulverde, -+ so manually keep AC97 reset high */ -+ pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); -+ udelay(10); -+ GCR |= GCR_WARM_RST; -+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); -+ udelay(500); -+#else -+ GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; -+ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); -+#endif -+ -+ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) -+ printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", -+ __FUNCTION__, gsr_bits); -+ -+ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); -+ GCR |= GCR_SDONE_IE|GCR_CDONE_IE; -+} -+ -+static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) -+{ -+ GCR &= GCR_COLD_RST; /* clear everything but nCRST */ -+ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ -+ -+ gsr_bits = 0; -+#ifdef CONFIG_PXA27x -+ /* PXA27x Developers Manual section 13.5.2.2.1 */ -+ pxa_set_cken(1 << 31, 1); -+ udelay(5); -+ pxa_set_cken(1 << 31, 0); -+ GCR = GCR_COLD_RST; -+ udelay(50); -+#else -+ GCR = GCR_COLD_RST; -+ GCR |= GCR_CDONE_IE|GCR_SDONE_IE; -+ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); -+#endif -+ -+ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) -+ printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", -+ __FUNCTION__, gsr_bits); -+ -+ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); -+ GCR |= GCR_SDONE_IE|GCR_CDONE_IE; -+} -+ -+static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) -+{ -+ long status; -+ -+ status = GSR; -+ if (status) { -+ GSR = status; -+ gsr_bits |= status; -+ wake_up(&gsr_wq); -+ -+#ifdef CONFIG_PXA27x -+ /* Although we don't use those we still need to clear them -+ since they tend to spuriously trigger when MMC is used -+ (hardware bug? go figure)... */ -+ MISR = MISR_EOC; -+ PISR = PISR_EOC; -+ MCSR = MCSR_EOC; -+#endif -+ -+ return IRQ_HANDLED; -+ } -+ -+ return IRQ_NONE; -+} -+ -+struct snd_ac97_bus_ops soc_ac97_ops = { -+ .read = pxa2xx_ac97_read, -+ .write = pxa2xx_ac97_write, -+ .warm_reset = pxa2xx_ac97_warm_reset, -+ .reset = pxa2xx_ac97_cold_reset, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { -+ .name = "AC97 PCM Stereo out", -+ .dev_addr = __PREG(PCDR), -+ .drcmr = &DRCMRTXPCDR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST32 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { -+ .name = "AC97 PCM Stereo in", -+ .dev_addr = __PREG(PCDR), -+ .drcmr = &DRCMRRXPCDR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST32 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { -+ .name = "AC97 Aux PCM (Slot 5) Mono out", -+ .dev_addr = __PREG(MODR), -+ .drcmr = &DRCMRTXMODR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { -+ .name = "AC97 Aux PCM (Slot 5) Mono in", -+ .dev_addr = __PREG(MODR), -+ .drcmr = &DRCMRRXMODR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { -+ .name = "AC97 Mic PCM (Slot 6) Mono in", -+ .dev_addr = __PREG(MCDR), -+ .drcmr = &DRCMRRXMCDR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+#ifdef CONFIG_PM -+static int pxa2xx_ac97_suspend(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ GCR |= GCR_ACLINK_OFF; -+ pxa_set_cken(CKEN2_AC97, 0); -+ return 0; -+} -+ -+static int pxa2xx_ac97_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ pxa_gpio_mode(GPIO31_SYNC_AC97_MD); -+ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); -+ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); -+ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); -+#ifdef CONFIG_PXA27x -+ /* Use GPIO 113 as AC97 Reset on Bulverde */ -+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); -+#endif -+ pxa_set_cken(CKEN2_AC97, 1); -+ return 0; -+} -+ -+#else -+#define pxa2xx_ac97_suspend NULL -+#define pxa2xx_ac97_resume NULL -+#endif -+ -+static int pxa2xx_ac97_probe(struct platform_device *pdev) -+{ -+ int ret; -+ -+ ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); -+ if (ret < 0) -+ goto err; -+ -+ pxa_gpio_mode(GPIO31_SYNC_AC97_MD); -+ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); -+ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); -+ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); -+#ifdef CONFIG_PXA27x -+ /* Use GPIO 113 as AC97 Reset on Bulverde */ -+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); -+#endif -+ pxa_set_cken(CKEN2_AC97, 1); -+ return 0; -+ -+ err: -+ if (CKEN & CKEN2_AC97) { -+ GCR |= GCR_ACLINK_OFF; -+ free_irq(IRQ_AC97, NULL); -+ pxa_set_cken(CKEN2_AC97, 0); -+ } -+ return ret; -+} -+ -+static void pxa2xx_ac97_remove(struct platform_device *pdev) -+{ -+ GCR |= GCR_ACLINK_OFF; -+ free_irq(IRQ_AC97, NULL); -+ pxa_set_cken(CKEN2_AC97, 0); -+} -+ -+static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; -+ else -+ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in; -+ -+ return 0; -+} -+ -+static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; -+ else -+ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in; -+ -+ return 0; -+} -+ -+static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ return -ENODEV; -+ else -+ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in; -+ -+ return 0; -+} -+ -+/* -+ * There is only 1 physical AC97 interface for pxa2xx, but it -+ * has extra fifo's that can be used for aux DACs and ADCs. -+ */ -+struct snd_soc_cpu_dai pxa_ac97_dai[] = { -+{ -+ .name = "pxa2xx-ac97", -+ .id = 0, -+ .type = SND_SOC_DAI_AC97, -+ .probe = pxa2xx_ac97_probe, -+ .remove = pxa2xx_ac97_remove, -+ .suspend = pxa2xx_ac97_suspend, -+ .resume = pxa2xx_ac97_resume, -+ .playback = { -+ .stream_name = "AC97 Playback", -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .capture = { -+ .stream_name = "AC97 Capture", -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .ops = { -+ .hw_params = pxa2xx_ac97_hw_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes), -+ .mode = pxa2xx_ac97_modes,}, -+}, -+{ -+ .name = "pxa2xx-ac97-aux", -+ .id = 1, -+ .type = SND_SOC_DAI_AC97, -+ .playback = { -+ .stream_name = "AC97 Aux Playback", -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .capture = { -+ .stream_name = "AC97 Aux Capture", -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .ops = { -+ .hw_params = pxa2xx_ac97_hw_aux_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes), -+ .mode = pxa2xx_ac97_modes,}, -+}, -+{ -+ .name = "pxa2xx-ac97-mic", -+ .id = 2, -+ .type = SND_SOC_DAI_AC97, -+ .capture = { -+ .stream_name = "AC97 Mic Capture", -+ .channels_min = 1, -+ .channels_max = 1,}, -+ .ops = { -+ .hw_params = pxa2xx_ac97_hw_mic_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes), -+ .mode = pxa2xx_ac97_modes,},}, -+}; -+ -+EXPORT_SYMBOL_GPL(pxa_ac97_dai); -+EXPORT_SYMBOL_GPL(soc_ac97_ops); -+ -+MODULE_AUTHOR("Nicolas Pitre"); -+MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-i2s.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-i2s.c -@@ -0,0 +1,354 @@ -+/* -+ * pxa2xx-i2s.c -- ALSA Soc Audio Layer -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 12th Aug 2005 Initial version. -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/device.h> -+#include <linux/delay.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/initval.h> -+#include <sound/soc.h> -+ -+#include <asm/hardware.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/audio.h> -+ -+#include "pxa2xx-pcm.h" -+ -+/* used to disable sysclk if external crystal is used */ -+static int extclk; -+module_param(extclk, int, 0); -+MODULE_PARM_DESC(extclk, "set to 1 to disable pxa2xx i2s sysclk"); -+ -+struct pxa_i2s_port { -+ u32 sadiv; -+ u32 sacr0; -+ u32 sacr1; -+ u32 saimr; -+ int master; -+}; -+static struct pxa_i2s_port pxa_i2s; -+ -+#define PXA_I2S_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF) -+ -+#define PXA_I2S_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define PXA_I2S_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -+ -+/* priv is divider */ -+static struct snd_soc_dai_mode pxa2xx_i2s_modes[] = { -+ /* pxa2xx I2S frame and clock master modes */ -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0x48, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0x34, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0x24, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0x1a, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0xd, -+ }, -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = PXA_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = SND_SOC_FSBD(4), -+ .priv = 0xc, -+ }, -+ -+ /* pxa2xx I2S frame master and clock slave mode */ -+ { -+ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = PXA_I2S_RATES, -+ .pcmdir = PXA_I2S_DIR, -+ .fs = SND_SOC_FS_ALL, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .bfs = 64, -+ .priv = 0x48, -+ }, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { -+ .name = "I2S PCM Stereo out", -+ .dev_addr = __PREG(SADR), -+ .drcmr = &DRCMRTXSADR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST32 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { -+ .name = "I2S PCM Stereo in", -+ .dev_addr = __PREG(SADR), -+ .drcmr = &DRCMRRXSADR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST32 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_gpio gpio_bus[] = { -+ { /* I2S SoC Slave */ -+ .rx = GPIO29_SDATA_IN_I2S_MD, -+ .tx = GPIO30_SDATA_OUT_I2S_MD, -+ .clk = GPIO28_BITCLK_IN_I2S_MD, -+ .frm = GPIO31_SYNC_I2S_MD, -+ }, -+ { /* I2S SoC Master */ -+#ifdef CONFIG_PXA27x -+ .sys = GPIO113_I2S_SYSCLK_MD, -+#else -+ .sys = GPIO32_SYSCLK_I2S_MD, -+#endif -+ .rx = GPIO29_SDATA_IN_I2S_MD, -+ .tx = GPIO30_SDATA_OUT_I2S_MD, -+ .clk = GPIO28_BITCLK_OUT_I2S_MD, -+ .frm = GPIO31_SYNC_I2S_MD, -+ }, -+}; -+ -+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if (!rtd->cpu_dai->active) { -+ SACR0 |= SACR0_RST; -+ SACR0 = 0; -+ } -+ -+ return 0; -+} -+ -+/* wait for I2S controller to be ready */ -+static int pxa_i2s_wait(void) -+{ -+ int i; -+ -+ /* flush the Rx FIFO */ -+ for(i = 0; i < 16; i++) -+ SADR; -+ return 0; -+} -+ -+static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ pxa_i2s.master = 0; -+ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CBS_CFS) -+ pxa_i2s.master = 1; -+ -+ if (pxa_i2s.master && !extclk) -+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys); -+ -+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx); -+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); -+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm); -+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk); -+ pxa_set_cken(CKEN8_I2S, 1); -+ pxa_i2s_wait(); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ rtd->cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out; -+ else -+ rtd->cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in; -+ -+ /* is port used by another stream */ -+ if (!(SACR0 & SACR0_ENB)) { -+ -+ SACR0 = 0; -+ SACR1 = 0; -+ if (pxa_i2s.master) -+ SACR0 |= SACR0_BCKD; -+ -+ SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1); -+ -+ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_LEFT_J) -+ SACR1 |= SACR1_AMSL; -+ } -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SAIMR |= SAIMR_TFS; -+ else -+ SAIMR |= SAIMR_RFS; -+ -+ SADIV = rtd->cpu_dai->dai_runtime.priv; -+ return 0; -+} -+ -+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ SACR0 |= SACR0_ENB; -+ break; -+ case SNDRV_PCM_TRIGGER_RESUME: -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ case SNDRV_PCM_TRIGGER_STOP: -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ break; -+ default: -+ ret = -EINVAL; -+ } -+ -+ return ret; -+} -+ -+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) -+{ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ SACR1 |= SACR1_DRPL; -+ SAIMR &= ~SAIMR_TFS; -+ } else { -+ SACR1 |= SACR1_DREC; -+ SAIMR &= ~SAIMR_RFS; -+ } -+ -+ if (SACR1 & (SACR1_DREC | SACR1_DRPL)) { -+ SACR0 &= ~SACR0_ENB; -+ pxa_i2s_wait(); -+ pxa_set_cken(CKEN8_I2S, 0); -+ } -+} -+ -+#ifdef CONFIG_PM -+static int pxa2xx_i2s_suspend(struct platform_device *dev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if (!dai->active) -+ return 0; -+ -+ /* store registers */ -+ pxa_i2s.sacr0 = SACR0; -+ pxa_i2s.sacr1 = SACR1; -+ pxa_i2s.saimr = SAIMR; -+ pxa_i2s.sadiv = SADIV; -+ -+ /* deactivate link */ -+ SACR0 &= ~SACR0_ENB; -+ pxa_i2s_wait(); -+ return 0; -+} -+ -+static int pxa2xx_i2s_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if (!dai->active) -+ return 0; -+ -+ pxa_i2s_wait(); -+ -+ SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; -+ SACR1 = pxa_i2s.sacr1; -+ SAIMR = pxa_i2s.saimr; -+ SADIV = pxa_i2s.sadiv; -+ SACR0 |= SACR0_ENB; -+ -+ return 0; -+} -+ -+#else -+#define pxa2xx_i2s_suspend NULL -+#define pxa2xx_i2s_resume NULL -+#endif -+ -+/* pxa2xx I2S sysclock is always 256 FS */ -+static unsigned int pxa_i2s_config_sysclk(struct snd_soc_cpu_dai *iface, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ return info->rate << 8; -+} -+ -+struct snd_soc_cpu_dai pxa_i2s_dai = { -+ .name = "pxa2xx-i2s", -+ .id = 0, -+ .type = SND_SOC_DAI_I2S, -+ .suspend = pxa2xx_i2s_suspend, -+ .resume = pxa2xx_i2s_resume, -+ .config_sysclk = pxa_i2s_config_sysclk, -+ .playback = { -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = pxa2xx_i2s_startup, -+ .shutdown = pxa2xx_i2s_shutdown, -+ .trigger = pxa2xx_i2s_trigger, -+ .hw_params = pxa2xx_i2s_hw_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(pxa2xx_i2s_modes), -+ .mode = pxa2xx_i2s_modes,}, -+}; -+ -+EXPORT_SYMBOL_GPL(pxa_i2s_dai); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.c -@@ -0,0 +1,363 @@ -+/* -+ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip -+ * -+ * Author: Nicolas Pitre -+ * Created: Nov 30, 2004 -+ * Copyright: (C) 2004 MontaVista Software, Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/init.h> -+#include <linux/platform_device.h> -+#include <linux/slab.h> -+#include <linux/dma-mapping.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+ -+#include <asm/dma.h> -+#include <asm/hardware.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/audio.h> -+ -+#include "pxa2xx-pcm.h" -+ -+static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { -+ .info = SNDRV_PCM_INFO_MMAP | -+ SNDRV_PCM_INFO_MMAP_VALID | -+ SNDRV_PCM_INFO_INTERLEAVED | -+ SNDRV_PCM_INFO_PAUSE | -+ SNDRV_PCM_INFO_RESUME, -+ .formats = SNDRV_PCM_FMTBIT_S16_LE | -+ SNDRV_PCM_FMTBIT_S24_LE | -+ SNDRV_PCM_FMTBIT_S32_LE, -+ .period_bytes_min = 32, -+ .period_bytes_max = 8192 - 32, -+ .periods_min = 1, -+ .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc), -+ .buffer_bytes_max = 128 * 1024, -+ .fifo_size = 32, -+}; -+ -+struct pxa2xx_runtime_data { -+ int dma_ch; -+ struct pxa2xx_pcm_dma_params *params; -+ pxa_dma_desc *dma_desc_array; -+ dma_addr_t dma_desc_array_phys; -+}; -+ -+static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) -+{ -+ struct snd_pcm_substream *substream = dev_id; -+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; -+ int dcsr; -+ -+ dcsr = DCSR(dma_ch); -+ DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN; -+ -+ if (dcsr & DCSR_ENDINTR) { -+ snd_pcm_period_elapsed(substream); -+ } else { -+ printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", -+ prtd->params->name, dma_ch, dcsr ); -+ } -+} -+ -+static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct pxa2xx_runtime_data *prtd = runtime->private_data; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct pxa2xx_pcm_dma_params *dma = rtd->cpu_dai->dma_data; -+ size_t totsize = params_buffer_bytes(params); -+ size_t period = params_period_bytes(params); -+ pxa_dma_desc *dma_desc; -+ dma_addr_t dma_buff_phys, next_desc_phys; -+ int ret; -+ -+ /* this may get called several times by oss emulation -+ * with different params */ -+ if (prtd->params == NULL) { -+ prtd->params = dma; -+ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW, -+ pxa2xx_pcm_dma_irq, substream); -+ if (ret < 0) -+ return ret; -+ prtd->dma_ch = ret; -+ } else if (prtd->params != dma) { -+ pxa_free_dma(prtd->dma_ch); -+ prtd->params = dma; -+ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW, -+ pxa2xx_pcm_dma_irq, substream); -+ if (ret < 0) -+ return ret; -+ prtd->dma_ch = ret; -+ } -+ -+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); -+ runtime->dma_bytes = totsize; -+ -+ dma_desc = prtd->dma_desc_array; -+ next_desc_phys = prtd->dma_desc_array_phys; -+ dma_buff_phys = runtime->dma_addr; -+ do { -+ next_desc_phys += sizeof(pxa_dma_desc); -+ dma_desc->ddadr = next_desc_phys; -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ dma_desc->dsadr = dma_buff_phys; -+ dma_desc->dtadr = prtd->params->dev_addr; -+ } else { -+ dma_desc->dsadr = prtd->params->dev_addr; -+ dma_desc->dtadr = dma_buff_phys; -+ } -+ if (period > totsize) -+ period = totsize; -+ dma_desc->dcmd = prtd->params->dcmd | period | DCMD_ENDIRQEN; -+ dma_desc++; -+ dma_buff_phys += period; -+ } while (totsize -= period); -+ dma_desc[-1].ddadr = prtd->dma_desc_array_phys; -+ -+ return 0; -+} -+ -+static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) -+{ -+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; -+ -+ if (prtd && prtd->params) -+ *prtd->params->drcmr = 0; -+ -+ if (prtd->dma_ch) { -+ snd_pcm_set_runtime_buffer(substream, NULL); -+ pxa_free_dma(prtd->dma_ch); -+ prtd->dma_ch = 0; -+ } -+ -+ return 0; -+} -+ -+static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; -+ -+ DCSR(prtd->dma_ch) &= ~DCSR_RUN; -+ DCSR(prtd->dma_ch) = 0; -+ DCMD(prtd->dma_ch) = 0; -+ *prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD; -+ -+ return 0; -+} -+ -+static int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; -+ DCSR(prtd->dma_ch) = DCSR_RUN; -+ break; -+ -+ case SNDRV_PCM_TRIGGER_STOP: -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ DCSR(prtd->dma_ch) &= ~DCSR_RUN; -+ break; -+ -+ case SNDRV_PCM_TRIGGER_RESUME: -+ DCSR(prtd->dma_ch) |= DCSR_RUN; -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; -+ DCSR(prtd->dma_ch) |= DCSR_RUN; -+ break; -+ -+ default: -+ ret = -EINVAL; -+ } -+ -+ return ret; -+} -+ -+static snd_pcm_uframes_t -+pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct pxa2xx_runtime_data *prtd = runtime->private_data; -+ -+ dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? -+ DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch); -+ snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr); -+ -+ if (x == runtime->buffer_size) -+ x = 0; -+ return x; -+} -+ -+static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct pxa2xx_runtime_data *prtd; -+ int ret; -+ -+ snd_soc_set_runtime_hwparams(substream, &pxa2xx_pcm_hardware); -+ -+ /* -+ * For mysterious reasons (and despite what the manual says) -+ * playback samples are lost if the DMA count is not a multiple -+ * of the DMA burst size. Let's add a rule to enforce that. -+ */ -+ ret = snd_pcm_hw_constraint_step(runtime, 0, -+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); -+ if (ret) -+ goto out; -+ -+ ret = snd_pcm_hw_constraint_step(runtime, 0, -+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); -+ if (ret) -+ goto out; -+ -+ prtd = kzalloc(sizeof(struct pxa2xx_runtime_data), GFP_KERNEL); -+ if (prtd == NULL) { -+ ret = -ENOMEM; -+ goto out; -+ } -+ -+ prtd->dma_desc_array = -+ dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE, -+ &prtd->dma_desc_array_phys, GFP_KERNEL); -+ if (!prtd->dma_desc_array) { -+ ret = -ENOMEM; -+ goto err1; -+ } -+ -+ runtime->private_data = prtd; -+ return 0; -+ -+ err1: -+ kfree(prtd); -+ out: -+ return ret; -+} -+ -+static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct pxa2xx_runtime_data *prtd = runtime->private_data; -+ -+ dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, -+ prtd->dma_desc_array, prtd->dma_desc_array_phys); -+ kfree(prtd); -+ return 0; -+} -+ -+static int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, -+ struct vm_area_struct *vma) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ return dma_mmap_writecombine(substream->pcm->card->dev, vma, -+ runtime->dma_area, -+ runtime->dma_addr, -+ runtime->dma_bytes); -+} -+ -+struct snd_pcm_ops pxa2xx_pcm_ops = { -+ .open = pxa2xx_pcm_open, -+ .close = pxa2xx_pcm_close, -+ .ioctl = snd_pcm_lib_ioctl, -+ .hw_params = pxa2xx_pcm_hw_params, -+ .hw_free = pxa2xx_pcm_hw_free, -+ .prepare = pxa2xx_pcm_prepare, -+ .trigger = pxa2xx_pcm_trigger, -+ .pointer = pxa2xx_pcm_pointer, -+ .mmap = pxa2xx_pcm_mmap, -+}; -+ -+static int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -+{ -+ struct snd_pcm_substream *substream = pcm->streams[stream].substream; -+ struct snd_dma_buffer *buf = &substream->dma_buffer; -+ size_t size = pxa2xx_pcm_hardware.buffer_bytes_max; -+ buf->dev.type = SNDRV_DMA_TYPE_DEV; -+ buf->dev.dev = pcm->card->dev; -+ buf->private_data = NULL; -+ buf->area = dma_alloc_writecombine(pcm->card->dev, size, -+ &buf->addr, GFP_KERNEL); -+ if (!buf->area) -+ return -ENOMEM; -+ buf->bytes = size; -+ return 0; -+} -+ -+static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -+{ -+ struct snd_pcm_substream *substream; -+ struct snd_dma_buffer *buf; -+ int stream; -+ -+ for (stream = 0; stream < 2; stream++) { -+ substream = pcm->streams[stream].substream; -+ if (!substream) -+ continue; -+ -+ buf = &substream->dma_buffer; -+ if (!buf->area) -+ continue; -+ -+ dma_free_writecombine(pcm->card->dev, buf->bytes, -+ buf->area, buf->addr); -+ buf->area = NULL; -+ } -+} -+ -+static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; -+ -+int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, -+ struct snd_pcm *pcm) -+{ -+ int ret = 0; -+ -+ if (!card->dev->dma_mask) -+ card->dev->dma_mask = &pxa2xx_pcm_dmamask; -+ if (!card->dev->coherent_dma_mask) -+ card->dev->coherent_dma_mask = DMA_32BIT_MASK; -+ -+ if (dai->playback.channels_min) { -+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, -+ SNDRV_PCM_STREAM_PLAYBACK); -+ if (ret) -+ goto out; -+ } -+ -+ if (dai->capture.channels_min) { -+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, -+ SNDRV_PCM_STREAM_CAPTURE); -+ if (ret) -+ goto out; -+ } -+ out: -+ return ret; -+} -+ -+struct snd_soc_platform pxa2xx_soc_platform = { -+ .name = "pxa2xx-audio", -+ .pcm_ops = &pxa2xx_pcm_ops, -+ .pcm_new = pxa2xx_pcm_new, -+ .pcm_free = pxa2xx_pcm_free_dma_buffers, -+}; -+ -+EXPORT_SYMBOL_GPL(pxa2xx_soc_platform); -+ -+MODULE_AUTHOR("Nicolas Pitre"); -+MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.h -@@ -0,0 +1,48 @@ -+/* -+ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip -+ * -+ * Author: Nicolas Pitre -+ * Created: Nov 30, 2004 -+ * Copyright: MontaVista Software, Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _PXA2XX_PCM_H -+#define _PXA2XX_PCM_H -+ -+struct pxa2xx_pcm_dma_params { -+ char *name; /* stream identifier */ -+ u32 dcmd; /* DMA descriptor dcmd field */ -+ volatile u32 *drcmr; /* the DMA request channel to use */ -+ u32 dev_addr; /* device physical address for DMA */ -+}; -+ -+struct pxa2xx_gpio { -+ u32 sys; -+ u32 rx; -+ u32 tx; -+ u32 clk; -+ u32 frm; -+}; -+ -+/* pxa2xx DAI ID's */ -+#define PXA2XX_DAI_AC97_HIFI 0 -+#define PXA2XX_DAI_AC97_AUX 1 -+#define PXA2XX_DAI_AC97_MIC 2 -+#define PXA2XX_DAI_I2S 0 -+#define PXA2XX_DAI_SSP1 0 -+#define PXA2XX_DAI_SSP2 1 -+#define PXA2XX_DAI_SSP3 2 -+ -+extern struct snd_soc_cpu_dai pxa_ac97_dai[3]; -+extern struct snd_soc_cpu_dai pxa_i2s_dai; -+extern struct snd_soc_cpu_dai pxa_ssp_dai[3]; -+ -+/* platform data */ -+extern struct snd_soc_platform pxa2xx_soc_platform; -+extern struct snd_ac97_bus_ops pxa2xx_ac97_ops; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ssp.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ssp.c -@@ -0,0 +1,767 @@ -+/* -+ * pxa2xx-ssp.c -- ALSA Soc Audio Layer -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 12th Aug 2005 Initial version. -+ * -+ * TODO: -+ * o Fix master mode (bug) -+ * o Fix resume (bug) -+ * o Add support for other clocks -+ * o Test network mode for > 16bit sample size -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/platform_device.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/initval.h> -+#include <sound/soc.h> -+ -+#include <asm/hardware.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/audio.h> -+#include <asm/arch/ssp.h> -+ -+#include "pxa2xx-pcm.h" -+ -+/* -+ * SSP sysclock frequency in Hz -+ * Neither default pxa2xx PLL clocks are good for audio, hence pxa27x -+ * has audio clock. I would recommend using the pxa27x audio clock or an -+ * external clock or making the codec master to gurantee better sample rates. -+ */ -+#ifdef CONFIG_PXA27x -+static int sysclk[3] = {13000000, 13000000, 13000000}; -+#else -+static int sysclk[3] = {1843200, 1843200, 1843200}; -+#endif -+module_param_array(sysclk, int, NULL, 0); -+MODULE_PARM_DESC(sysclk, "sysclk frequency in Hz"); -+ -+/* -+ * SSP sysclock source. -+ * sysclk is ignored if audio clock is used -+ */ -+#ifdef CONFIG_PXA27x -+static int clksrc[3] = {0, 0, 0}; -+#else -+static int clksrc[3] = {0, 0, 0}; -+#endif -+module_param_array(clksrc, int, NULL, 0); -+MODULE_PARM_DESC(clksrc, -+ "sysclk source, 0 = internal PLL, 1 = ext, 2 = network, 3 = audio clock"); -+ -+/* -+ * SSP GPIO's -+ */ -+#define GPIO26_SSP1RX_MD (26 | GPIO_ALT_FN_1_IN) -+#define GPIO25_SSP1TX_MD (25 | GPIO_ALT_FN_2_OUT) -+#define GPIO23_SSP1CLKS_MD (23 | GPIO_ALT_FN_2_IN) -+#define GPIO24_SSP1FRMS_MD (24 | GPIO_ALT_FN_2_IN) -+#define GPIO23_SSP1CLKM_MD (23 | GPIO_ALT_FN_2_OUT) -+#define GPIO24_SSP1FRMM_MD (24 | GPIO_ALT_FN_2_OUT) -+ -+#define GPIO11_SSP2RX_MD (11 | GPIO_ALT_FN_2_IN) -+#define GPIO13_SSP2TX_MD (13 | GPIO_ALT_FN_1_OUT) -+#define GPIO22_SSP2CLKS_MD (22 | GPIO_ALT_FN_3_IN) -+#define GPIO88_SSP2FRMS_MD (88 | GPIO_ALT_FN_3_IN) -+#define GPIO22_SSP2CLKM_MD (22 | GPIO_ALT_FN_3_OUT) -+#define GPIO88_SSP2FRMM_MD (88 | GPIO_ALT_FN_3_OUT) -+ -+#define GPIO82_SSP3RX_MD (82 | GPIO_ALT_FN_1_IN) -+#define GPIO81_SSP3TX_MD (81 | GPIO_ALT_FN_1_OUT) -+#define GPIO84_SSP3CLKS_MD (84 | GPIO_ALT_FN_1_IN) -+#define GPIO83_SSP3FRMS_MD (83 | GPIO_ALT_FN_1_IN) -+#define GPIO84_SSP3CLKM_MD (84 | GPIO_ALT_FN_1_OUT) -+#define GPIO83_SSP3FRMM_MD (83 | GPIO_ALT_FN_1_OUT) -+ -+#define PXA_SSP_MDAIFMT \ -+ (SND_SOC_DAIFMT_DSP_B |SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_CBM_CFS | \ -+ SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF) -+ -+#define PXA_SSP_SDAIFMT \ -+ (SND_SOC_DAIFMT_DSP_B |SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS | \ -+ SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF) -+ -+#define PXA_SSP_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define PXA_SSP_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ -+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) -+ -+#define PXA_SSP_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+/* -+ * SSP modes -+ */ -+static struct snd_soc_dai_mode pxa2xx_ssp_modes[] = { -+ /* port slave clk & frame modes */ -+ { -+ .fmt = PXA_SSP_SDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = PXA_SSP_RATES, -+ .pcmdir = PXA_SSP_DIR, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+ -+ /* port master clk & frame modes */ -+#ifdef CONFIG_PXA27x -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 256, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_11025, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 256, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 256, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_22050, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 256, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 256, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 256, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 256, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 128, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+ { -+ .fmt = PXA_SSP_MDAIFMT, -+ .pcmfmt = PXA_SSP_BITS, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = PXA_SSP_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = 128, -+ .bfs = SND_SOC_FSBW(1), -+ }, -+#endif -+}; -+ -+static struct ssp_dev ssp[3]; -+#ifdef CONFIG_PM -+static struct ssp_state ssp_state[3]; -+#endif -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_out = { -+ .name = "SSP1 PCM Mono out", -+ .dev_addr = __PREG(SSDR_P1), -+ .drcmr = &DRCMRTXSSDR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_in = { -+ .name = "SSP1 PCM Mono in", -+ .dev_addr = __PREG(SSDR_P1), -+ .drcmr = &DRCMRRXSSDR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_out = { -+ .name = "SSP1 PCM Stereo out", -+ .dev_addr = __PREG(SSDR_P1), -+ .drcmr = &DRCMRTXSSDR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST16 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_in = { -+ .name = "SSP1 PCM Stereo in", -+ .dev_addr = __PREG(SSDR_P1), -+ .drcmr = &DRCMRRXSSDR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST16 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_out = { -+ .name = "SSP2 PCM Mono out", -+ .dev_addr = __PREG(SSDR_P2), -+ .drcmr = &DRCMRTXSS2DR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_in = { -+ .name = "SSP2 PCM Mono in", -+ .dev_addr = __PREG(SSDR_P2), -+ .drcmr = &DRCMRRXSS2DR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_out = { -+ .name = "SSP2 PCM Stereo out", -+ .dev_addr = __PREG(SSDR_P2), -+ .drcmr = &DRCMRTXSS2DR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST16 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_in = { -+ .name = "SSP2 PCM Stereo in", -+ .dev_addr = __PREG(SSDR_P2), -+ .drcmr = &DRCMRRXSS2DR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST16 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_out = { -+ .name = "SSP3 PCM Mono out", -+ .dev_addr = __PREG(SSDR_P3), -+ .drcmr = &DRCMRTXSS3DR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_in = { -+ .name = "SSP3 PCM Mono in", -+ .dev_addr = __PREG(SSDR_P3), -+ .drcmr = &DRCMRRXSS3DR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST16 | DCMD_WIDTH2, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_out = { -+ .name = "SSP3 PCM Stereo out", -+ .dev_addr = __PREG(SSDR_P3), -+ .drcmr = &DRCMRTXSS3DR, -+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | -+ DCMD_BURST16 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_in = { -+ .name = "SSP3 PCM Stereo in", -+ .dev_addr = __PREG(SSDR_P3), -+ .drcmr = &DRCMRRXSS3DR, -+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | -+ DCMD_BURST16 | DCMD_WIDTH4, -+}; -+ -+static struct pxa2xx_pcm_dma_params *ssp_dma_params[3][4] = { -+ {&pxa2xx_ssp1_pcm_mono_out, &pxa2xx_ssp1_pcm_mono_in, -+ &pxa2xx_ssp1_pcm_stereo_out,&pxa2xx_ssp1_pcm_stereo_in,}, -+ {&pxa2xx_ssp2_pcm_mono_out, &pxa2xx_ssp2_pcm_mono_in, -+ &pxa2xx_ssp2_pcm_stereo_out, &pxa2xx_ssp2_pcm_stereo_in,}, -+ {&pxa2xx_ssp3_pcm_mono_out, &pxa2xx_ssp3_pcm_mono_in, -+ &pxa2xx_ssp3_pcm_stereo_out,&pxa2xx_ssp3_pcm_stereo_in,}, -+}; -+ -+static struct pxa2xx_gpio ssp_gpios[3][4] = { -+ {{ /* SSP1 SND_SOC_DAIFMT_CBM_CFM */ -+ .rx = GPIO26_SSP1RX_MD, -+ .tx = GPIO25_SSP1TX_MD, -+ .clk = (23 | GPIO_ALT_FN_2_IN), -+ .frm = (24 | GPIO_ALT_FN_2_IN), -+ }, -+ { /* SSP1 SND_SOC_DAIFMT_CBS_CFS */ -+ .rx = GPIO26_SSP1RX_MD, -+ .tx = GPIO25_SSP1TX_MD, -+ .clk = (23 | GPIO_ALT_FN_2_OUT), -+ .frm = (24 | GPIO_ALT_FN_2_OUT), -+ }, -+ { /* SSP1 SND_SOC_DAIFMT_CBS_CFM */ -+ .rx = GPIO26_SSP1RX_MD, -+ .tx = GPIO25_SSP1TX_MD, -+ .clk = (23 | GPIO_ALT_FN_2_OUT), -+ .frm = (24 | GPIO_ALT_FN_2_IN), -+ }, -+ { /* SSP1 SND_SOC_DAIFMT_CBM_CFS */ -+ .rx = GPIO26_SSP1RX_MD, -+ .tx = GPIO25_SSP1TX_MD, -+ .clk = (23 | GPIO_ALT_FN_2_IN), -+ .frm = (24 | GPIO_ALT_FN_2_OUT), -+ }}, -+ {{ /* SSP2 SND_SOC_DAIFMT_CBM_CFM */ -+ .rx = GPIO11_SSP2RX_MD, -+ .tx = GPIO13_SSP2TX_MD, -+ .clk = (22 | GPIO_ALT_FN_3_IN), -+ .frm = (88 | GPIO_ALT_FN_3_IN), -+ }, -+ { /* SSP2 SND_SOC_DAIFMT_CBS_CFS */ -+ .rx = GPIO11_SSP2RX_MD, -+ .tx = GPIO13_SSP2TX_MD, -+ .clk = (22 | GPIO_ALT_FN_3_OUT), -+ .frm = (88 | GPIO_ALT_FN_3_OUT), -+ }, -+ { /* SSP2 SND_SOC_DAIFMT_CBS_CFM */ -+ .rx = GPIO11_SSP2RX_MD, -+ .tx = GPIO13_SSP2TX_MD, -+ .clk = (22 | GPIO_ALT_FN_3_OUT), -+ .frm = (88 | GPIO_ALT_FN_3_IN), -+ }, -+ { /* SSP2 SND_SOC_DAIFMT_CBM_CFS */ -+ .rx = GPIO11_SSP2RX_MD, -+ .tx = GPIO13_SSP2TX_MD, -+ .clk = (22 | GPIO_ALT_FN_3_IN), -+ .frm = (88 | GPIO_ALT_FN_3_OUT), -+ }}, -+ {{ /* SSP3 SND_SOC_DAIFMT_CBM_CFM */ -+ .rx = GPIO82_SSP3RX_MD, -+ .tx = GPIO81_SSP3TX_MD, -+ .clk = (84 | GPIO_ALT_FN_3_IN), -+ .frm = (83 | GPIO_ALT_FN_3_IN), -+ }, -+ { /* SSP3 SND_SOC_DAIFMT_CBS_CFS */ -+ .rx = GPIO82_SSP3RX_MD, -+ .tx = GPIO81_SSP3TX_MD, -+ .clk = (84 | GPIO_ALT_FN_3_OUT), -+ .frm = (83 | GPIO_ALT_FN_3_OUT), -+ }, -+ { /* SSP3 SND_SOC_DAIFMT_CBS_CFM */ -+ .rx = GPIO82_SSP3RX_MD, -+ .tx = GPIO81_SSP3TX_MD, -+ .clk = (84 | GPIO_ALT_FN_3_OUT), -+ .frm = (83 | GPIO_ALT_FN_3_IN), -+ }, -+ { /* SSP3 SND_SOC_DAIFMT_CBM_CFS */ -+ .rx = GPIO82_SSP3RX_MD, -+ .tx = GPIO81_SSP3TX_MD, -+ .clk = (84 | GPIO_ALT_FN_3_IN), -+ .frm = (83 | GPIO_ALT_FN_3_OUT), -+ }}, -+}; -+ -+static int pxa2xx_ssp_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ int ret = 0; -+ -+ if (!rtd->cpu_dai->active) { -+ ret = ssp_init (&ssp[rtd->cpu_dai->id], rtd->cpu_dai->id + 1, -+ SSP_NO_IRQ); -+ if (ret < 0) -+ return ret; -+ ssp_disable(&ssp[rtd->cpu_dai->id]); -+ } -+ return ret; -+} -+ -+static void pxa2xx_ssp_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if (!rtd->cpu_dai->active) { -+ ssp_disable(&ssp[rtd->cpu_dai->id]); -+ ssp_exit(&ssp[rtd->cpu_dai->id]); -+ } -+} -+ -+#ifdef CONFIG_PM -+ -+#if defined (CONFIG_PXA27x) -+static int cken[3] = {CKEN23_SSP1, CKEN3_SSP2, CKEN4_SSP3}; -+#else -+static int cken[3] = {CKEN3_SSP, CKEN9_NSSP, CKEN10_ASSP}; -+#endif -+ -+static int pxa2xx_ssp_suspend(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if (!dai->active) -+ return 0; -+ -+ ssp_save_state(&ssp[dai->id], &ssp_state[dai->id]); -+ pxa_set_cken(cken[dai->id], 0); -+ return 0; -+} -+ -+static int pxa2xx_ssp_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if (!dai->active) -+ return 0; -+ -+ pxa_set_cken(cken[dai->id], 1); -+ ssp_restore_state(&ssp[dai->id], &ssp_state[dai->id]); -+ ssp_enable(&ssp[dai->id]); -+ -+ return 0; -+} -+ -+#else -+#define pxa2xx_ssp_suspend NULL -+#define pxa2xx_ssp_resume NULL -+#endif -+ -+/* todo - check clk source and PLL before returning clock rate */ -+static unsigned int pxa_ssp_config_sysclk(struct snd_soc_cpu_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ /* audio clock ? (divide by 1) */ -+ if (clksrc[dai->id] == 3) { -+ switch(info->rate){ -+ case 8000: -+ case 16000: -+ case 32000: -+ case 48000: -+ case 96000: -+ return 12288000; -+ break; -+ case 11025: -+ case 22050: -+ case 44100: -+ case 88200: -+ return 11289600; -+ break; -+ } -+ } -+ -+ /* pll */ -+ return sysclk[dai->id]; -+} -+ -+#ifdef CONFIG_PXA27x -+static u32 pxa27x_set_audio_clk(unsigned int rate, unsigned int fs) -+{ -+ u32 aclk = 0, div = 0; -+ -+ if (rate == 0 || fs == 0) -+ return 0; -+ -+ switch(rate){ -+ case 8000: -+ case 16000: -+ case 32000: -+ case 48000: -+ case 96000: -+ aclk = 0x2 << 4; -+ div = 12288000 / (rate * fs); -+ break; -+ case 11025: -+ case 22050: -+ case 44100: -+ case 88200: -+ aclk = 0x1 << 4; -+ div = 11289600 / (rate * fs); -+ break; -+ } -+ -+ aclk |= ffs(div) - 1; -+ return aclk; -+} -+#endif -+ -+static inline int get_scr(int srate, int id) -+{ -+ if (srate == 0) -+ return 0; -+ return (sysclk[id] / srate) - 1; -+} -+ -+static int pxa2xx_ssp_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ int fmt = 0, dma = 0, fs, chn = params_channels(params); -+ u32 ssp_mode = 0, ssp_setup = 0, psp_mode = 0, rate = 0; -+ -+ fs = rtd->cpu_dai->dai_runtime.fs; -+ -+ /* select correct DMA params */ -+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) -+ dma = 1; -+ if (chn == 2 || rtd->cpu_dai->dai_runtime.pcmfmt != PXA_SSP_BITS) -+ dma += 2; -+ rtd->cpu_dai->dma_data = ssp_dma_params[rtd->cpu_dai->id][dma]; -+ -+ /* is port used by another stream */ -+ if (SSCR0 & SSCR0_SSE) -+ return 0; -+ -+ /* bit size */ -+ switch(rtd->cpu_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ ssp_mode |=SSCR0_DataSize(16); -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ ssp_mode |=(SSCR0_EDSS | SSCR0_DataSize(8)); -+ /* use network mode for stereo samples > 16 bits */ -+ if (chn == 2) { -+ ssp_mode |= (SSCR0_MOD | SSCR0_SlotsPerFrm(2) << 24); -+ /* active slots 0,1 */ -+ SSTSA_P(rtd->cpu_dai->id +1) = 0x3; -+ SSRSA_P(rtd->cpu_dai->id +1) = 0x3; -+ } -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ ssp_mode |= (SSCR0_EDSS | SSCR0_DataSize(16)); -+ /* use network mode for stereo samples > 16 bits */ -+ if (chn == 2) { -+ ssp_mode |= (SSCR0_MOD | SSCR0_SlotsPerFrm(2) << 24); -+ /* active slots 0,1 */ -+ SSTSA_P(rtd->cpu_dai->id +1) = 0x3; -+ SSRSA_P(rtd->cpu_dai->id +1) = 0x3; -+ } -+ break; -+ } -+ -+ ssp_mode |= SSCR0_PSP; -+ ssp_setup = SSCR1_RxTresh(14) | SSCR1_TxTresh(1) | -+ SSCR1_TRAIL | SSCR1_RWOT; -+ -+ switch(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ ssp_setup |= (SSCR1_SCLKDIR | SSCR1_SFRMDIR); -+ break; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ ssp_setup |= SSCR1_SCLKDIR; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFM: -+ ssp_setup |= SSCR1_SFRMDIR; -+ break; -+ } -+ -+ switch(rtd->cpu_dai->dai_runtime.fmt) { -+ case SND_SOC_DAIFMT_CBS_CFS: -+ fmt = 1; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFM: -+ fmt = 2; -+ break; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ fmt = 3; -+ break; -+ } -+ -+ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].rx); -+ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].tx); -+ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].frm); -+ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].clk); -+ -+ switch (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ psp_mode |= SSPSP_SFRMP | SSPSP_FSRT; -+ break; -+ } -+ -+ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_DSP_A) -+ psp_mode |= SSPSP_SCMODE(2); -+ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_DSP_B) -+ psp_mode |= SSPSP_SCMODE(3); -+ -+ switch(clksrc[rtd->cpu_dai->id]) { -+ case 2: /* network clock */ -+ ssp_mode |= SSCR0_NCS | SSCR0_MOD; -+ case 1: /* external clock */ -+ ssp_mode |= SSCR0_ECS; -+ case 0: /* internal clock */ -+ rate = get_scr(snd_soc_get_rate(rtd->cpu_dai->dai_runtime.pcmrate), -+ rtd->cpu_dai->id); -+ break; -+#ifdef CONFIG_PXA27x -+ case 3: /* audio clock */ -+ ssp_mode |= (1 << 30); -+ SSACD_P(rtd->cpu_dai->id) = (0x1 << 3) | -+ pxa27x_set_audio_clk( -+ snd_soc_get_rate(rtd->cpu_dai->dai_runtime.pcmrate), fs); -+ break; -+#endif -+ } -+ -+ ssp_config(&ssp[rtd->cpu_dai->id], ssp_mode, ssp_setup, psp_mode, -+ SSCR0_SerClkDiv(rate)); -+#if 0 -+ printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x\n", -+ SSCR0_P(rtd->cpu_dai->id+1), SSCR1_P(rtd->cpu_dai->id+1), -+ SSTO_P(rtd->cpu_dai->id+1), SSPSP_P(rtd->cpu_dai->id+1), -+ SSSR_P(rtd->cpu_dai->id+1)); -+#endif -+ return 0; -+} -+ -+static int pxa2xx_ssp_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_RESUME: -+ ssp_enable(&ssp[rtd->cpu_dai->id]); -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_TSRE; -+ else -+ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_RSRE; -+ SSSR_P(rtd->cpu_dai->id+1) |= SSSR_P(rtd->cpu_dai->id+1); -+ break; -+ case SNDRV_PCM_TRIGGER_START: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_TSRE; -+ else -+ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_RSRE; -+ ssp_enable(&ssp[rtd->cpu_dai->id]); -+ break; -+ case SNDRV_PCM_TRIGGER_STOP: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_TSRE; -+ else -+ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_RSRE; -+ break; -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ ssp_disable(&ssp[rtd->cpu_dai->id]); -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_TSRE; -+ else -+ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_RSRE; -+ break; -+ -+ default: -+ ret = -EINVAL; -+ } -+#if 0 -+ printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x\n", -+ SSCR0_P(rtd->cpu_dai->id+1), SSCR1_P(rtd->cpu_dai->id+1), -+ SSTO_P(rtd->cpu_dai->id+1), SSPSP_P(rtd->cpu_dai->id+1), -+ SSSR_P(rtd->cpu_dai->id+1)); -+#endif -+ return ret; -+} -+ -+struct snd_soc_cpu_dai pxa_ssp_dai[] = { -+ { .name = "pxa2xx-ssp1", -+ .id = 0, -+ .type = SND_SOC_DAI_PCM, -+ .suspend = pxa2xx_ssp_suspend, -+ .resume = pxa2xx_ssp_resume, -+ .config_sysclk = pxa_ssp_config_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = pxa2xx_ssp_startup, -+ .shutdown = pxa2xx_ssp_shutdown, -+ .trigger = pxa2xx_ssp_trigger, -+ .hw_params = pxa2xx_ssp_hw_params,}, -+ .caps = { -+ .mode = pxa2xx_ssp_modes, -+ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),}, -+ }, -+ { .name = "pxa2xx-ssp2", -+ .id = 1, -+ .type = SND_SOC_DAI_PCM, -+ .suspend = pxa2xx_ssp_suspend, -+ .resume = pxa2xx_ssp_resume, -+ .config_sysclk = pxa_ssp_config_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = pxa2xx_ssp_startup, -+ .shutdown = pxa2xx_ssp_shutdown, -+ .trigger = pxa2xx_ssp_trigger, -+ .hw_params = pxa2xx_ssp_hw_params,}, -+ .caps = { -+ .mode = pxa2xx_ssp_modes, -+ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),}, -+ }, -+ { .name = "pxa2xx-ssp3", -+ .id = 2, -+ .type = SND_SOC_DAI_PCM, -+ .suspend = pxa2xx_ssp_suspend, -+ .resume = pxa2xx_ssp_resume, -+ .config_sysclk = pxa_ssp_config_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = pxa2xx_ssp_startup, -+ .shutdown = pxa2xx_ssp_shutdown, -+ .trigger = pxa2xx_ssp_trigger, -+ .hw_params = pxa2xx_ssp_hw_params,}, -+ .caps = { -+ .mode = pxa2xx_ssp_modes, -+ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),}, -+ }, -+}; -+ -+EXPORT_SYMBOL_GPL(pxa_ssp_dai); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("pxa2xx SSP/PCM SoC Interface"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/spitz.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/spitz.c -@@ -0,0 +1,374 @@ -+/* -+ * spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> -+ * Richard Purdie <richard@openedhand.com> -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 30th Nov 2005 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/timer.h> -+#include <linux/interrupt.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/mach-types.h> -+#include <asm/hardware/scoop.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/hardware.h> -+#include <asm/arch/akita.h> -+#include <asm/arch/spitz.h> -+#include <asm/mach-types.h> -+#include "../codecs/wm8750.h" -+#include "pxa2xx-pcm.h" -+ -+#define SPITZ_HP 0 -+#define SPITZ_MIC 1 -+#define SPITZ_LINE 2 -+#define SPITZ_HEADSET 3 -+#define SPITZ_HP_OFF 4 -+#define SPITZ_SPK_ON 0 -+#define SPITZ_SPK_OFF 1 -+ -+ /* audio clock in Hz - rounded from 12.235MHz */ -+#define SPITZ_AUDIO_CLOCK 12288000 -+ -+static int spitz_jack_func; -+static int spitz_spk_func; -+ -+static void spitz_ext_control(struct snd_soc_codec *codec) -+{ -+ if (spitz_spk_func == SPITZ_SPK_ON) -+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); -+ else -+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0); -+ -+ /* set up jack connection */ -+ switch (spitz_jack_func) { -+ case SPITZ_HP: -+ /* enable and unmute hp jack, disable mic bias */ -+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); -+ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); -+ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); -+ break; -+ case SPITZ_MIC: -+ /* enable mic jack and bias, mute hp */ -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); -+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); -+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); -+ break; -+ case SPITZ_LINE: -+ /* enable line jack, disable mic bias and mute hp */ -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 1); -+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); -+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); -+ break; -+ case SPITZ_HEADSET: -+ /* enable and unmute headset jack enable mic bias, mute L hp */ -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); -+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1); -+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); -+ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); -+ break; -+ case SPITZ_HP_OFF: -+ -+ /* jack removed, everything off */ -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); -+ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); -+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); -+ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); -+ break; -+ } -+ snd_soc_dapm_sync_endpoints(codec); -+} -+ -+static int spitz_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_codec *codec = rtd->socdev->codec; -+ -+ /* check the jack status at stream startup */ -+ spitz_ext_control(codec); -+ return 0; -+} -+ -+static struct snd_soc_ops spitz_ops = { -+ .startup = spitz_startup, -+}; -+ -+static int spitz_get_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = spitz_jack_func; -+ return 0; -+} -+ -+static int spitz_set_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ if (spitz_jack_func == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ spitz_jack_func = ucontrol->value.integer.value[0]; -+ spitz_ext_control(codec); -+ return 1; -+} -+ -+static int spitz_get_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = spitz_spk_func; -+ return 0; -+} -+ -+static int spitz_set_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ if (spitz_spk_func == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ spitz_spk_func = ucontrol->value.integer.value[0]; -+ spitz_ext_control(codec); -+ return 1; -+} -+ -+static int spitz_mic_bias(struct snd_soc_dapm_widget *w, int event) -+{ -+ if (machine_is_borzoi() || machine_is_spitz()) { -+ if (SND_SOC_DAPM_EVENT_ON(event)) -+ set_scoop_gpio(&spitzscoop2_device.dev, -+ SPITZ_SCP2_MIC_BIAS); -+ else -+ reset_scoop_gpio(&spitzscoop2_device.dev, -+ SPITZ_SCP2_MIC_BIAS); -+ } -+ -+ if (machine_is_akita()) { -+ if (SND_SOC_DAPM_EVENT_ON(event)) -+ akita_set_ioexp(&akitaioexp_device.dev, -+ AKITA_IOEXP_MIC_BIAS); -+ else -+ akita_reset_ioexp(&akitaioexp_device.dev, -+ AKITA_IOEXP_MIC_BIAS); -+ } -+ return 0; -+} -+ -+/* spitz machine dapm widgets */ -+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { -+ SND_SOC_DAPM_HP("Headphone Jack", NULL), -+ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), -+ SND_SOC_DAPM_SPK("Ext Spk", NULL), -+ SND_SOC_DAPM_LINE("Line Jack", NULL), -+ -+ /* headset is a mic and mono headphone */ -+ SND_SOC_DAPM_HP("Headset Jack", NULL), -+}; -+ -+/* Spitz machine audio_map */ -+static const char *audio_map[][3] = { -+ -+ /* headphone connected to LOUT1, ROUT1 */ -+ {"Headphone Jack", NULL, "LOUT1"}, -+ {"Headphone Jack", NULL, "ROUT1"}, -+ -+ /* headset connected to ROUT1 and LINPUT1 with bias (def below) */ -+ {"Headset Jack", NULL, "ROUT1"}, -+ -+ /* ext speaker connected to LOUT2, ROUT2 */ -+ {"Ext Spk", NULL , "ROUT2"}, -+ {"Ext Spk", NULL , "LOUT2"}, -+ -+ /* mic is connected to input 1 - with bias */ -+ {"LINPUT1", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Mic Jack"}, -+ -+ /* line is connected to input 1 - no bias */ -+ {"LINPUT1", NULL, "Line Jack"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", -+ "Off"}; -+static const char *spk_function[] = {"On", "Off"}; -+static const struct soc_enum spitz_enum[] = { -+ SOC_ENUM_SINGLE_EXT(5, jack_function), -+ SOC_ENUM_SINGLE_EXT(2, spk_function), -+}; -+ -+static const struct snd_kcontrol_new wm8750_spitz_controls[] = { -+ SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack, -+ spitz_set_jack), -+ SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk, -+ spitz_set_spk), -+}; -+ -+/* -+ * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device -+ */ -+static int spitz_wm8750_init(struct snd_soc_codec *codec) -+{ -+ int i, err; -+ -+ /* NC codec pins */ -+ snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); -+ snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0); -+ snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0); -+ snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0); -+ snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0); -+ snd_soc_dapm_set_endpoint(codec, "OUT3", 0); -+ snd_soc_dapm_set_endpoint(codec, "MONO", 0); -+ -+ /* Add spitz specific controls */ -+ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ /* Add spitz specific widgets */ -+ for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); -+ } -+ -+ /* Set up spitz specific audio path audio_map */ -+ for (i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ return 0; -+} -+ -+static unsigned int spitz_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) { -+ /* pxa2xx is i2s master */ -+ switch (info->rate) { -+ case 11025: -+ case 22050: -+ case 44100: -+ case 88200: -+ /* configure codec digital filters -+ * for 11.025, 22.05, 44.1, 88.2 */ -+ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ 11289600); -+ break; -+ default: -+ /* configure codec digital filters for all other rates */ -+ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ SPITZ_AUDIO_CLOCK); -+ break; -+ } -+ /* configure pxa2xx i2s interface clocks as master */ -+ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, -+ SPITZ_AUDIO_CLOCK); -+ } else { -+ /* codec is i2s master - only configure codec DAI clock */ -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, -+ SPITZ_AUDIO_CLOCK); -+ } -+} -+ -+/* spitz digital audio interface glue - connects codec <--> CPU */ -+static struct snd_soc_dai_link spitz_dai = { -+ .name = "wm8750", -+ .stream_name = "WM8750", -+ .cpu_dai = &pxa_i2s_dai, -+ .codec_dai = &wm8750_dai, -+ .init = spitz_wm8750_init, -+ .config_sysclk = spitz_config_sysclk, -+}; -+ -+/* spitz audio machine driver */ -+static struct snd_soc_machine snd_soc_machine_spitz = { -+ .name = "Spitz", -+ .dai_link = &spitz_dai, -+ .num_links = 1, -+ .ops = &spitz_ops, -+}; -+ -+/* spitz audio private data */ -+static struct wm8750_setup_data spitz_wm8750_setup = { -+ .i2c_address = 0x1b, -+}; -+ -+/* spitz audio subsystem */ -+static struct snd_soc_device spitz_snd_devdata = { -+ .machine = &snd_soc_machine_spitz, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8750, -+ .codec_data = &spitz_wm8750_setup, -+}; -+ -+static struct platform_device *spitz_snd_device; -+ -+static int __init spitz_init(void) -+{ -+ int ret; -+ -+ if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita())) -+ return -ENODEV; -+ -+ spitz_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!spitz_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(spitz_snd_device, &spitz_snd_devdata); -+ spitz_snd_devdata.dev = &spitz_snd_device->dev; -+ ret = platform_device_add(spitz_snd_device); -+ -+ if (ret) -+ platform_device_put(spitz_snd_device); -+ -+ return ret; -+} -+ -+static void __exit spitz_exit(void) -+{ -+ platform_device_unregister(spitz_snd_device); -+} -+ -+module_init(spitz_init); -+module_exit(spitz_exit); -+ -+MODULE_AUTHOR("Richard Purdie"); -+MODULE_DESCRIPTION("ALSA SoC Spitz"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/pxa/tosa.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/pxa/tosa.c -@@ -0,0 +1,287 @@ -+/* -+ * tosa.c -- SoC audio for Tosa -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> -+ * Richard Purdie <richard@openedhand.com> -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 30th Nov 2005 Initial version. -+ * -+ * GPIO's -+ * 1 - Jack Insertion -+ * 5 - Hookswitch (headset answer/hang up switch) -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/device.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/mach-types.h> -+#include <asm/hardware/tmio.h> -+#include <asm/arch/pxa-regs.h> -+#include <asm/arch/hardware.h> -+#include <asm/arch/audio.h> -+#include <asm/arch/tosa.h> -+ -+#include "../codecs/wm9712.h" -+#include "pxa2xx-pcm.h" -+ -+static struct snd_soc_machine tosa; -+ -+#define TOSA_HP 0 -+#define TOSA_MIC_INT 1 -+#define TOSA_HEADSET 2 -+#define TOSA_HP_OFF 3 -+#define TOSA_SPK_ON 0 -+#define TOSA_SPK_OFF 1 -+ -+static int tosa_jack_func; -+static int tosa_spk_func; -+ -+static void tosa_ext_control(struct snd_soc_codec *codec) -+{ -+ int spk = 0, mic_int = 0, hp = 0, hs = 0; -+ -+ /* set up jack connection */ -+ switch (tosa_jack_func) { -+ case TOSA_HP: -+ hp = 1; -+ break; -+ case TOSA_MIC_INT: -+ mic_int = 1; -+ break; -+ case TOSA_HEADSET: -+ hs = 1; -+ break; -+ } -+ -+ if (tosa_spk_func == TOSA_SPK_ON) -+ spk = 1; -+ -+ snd_soc_dapm_set_endpoint(codec, "Speaker", spk); -+ snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int); -+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); -+ snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); -+ snd_soc_dapm_sync_endpoints(codec); -+} -+ -+static int tosa_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_codec *codec = rtd->socdev->codec; -+ -+ /* check the jack status at stream startup */ -+ tosa_ext_control(codec); -+ return 0; -+} -+ -+static struct snd_soc_ops tosa_ops = { -+ .startup = tosa_startup, -+}; -+ -+static int tosa_get_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = tosa_jack_func; -+ return 0; -+} -+ -+static int tosa_set_jack(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ if (tosa_jack_func == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ tosa_jack_func = ucontrol->value.integer.value[0]; -+ tosa_ext_control(codec); -+ return 1; -+} -+ -+static int tosa_get_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ ucontrol->value.integer.value[0] = tosa_spk_func; -+ return 0; -+} -+ -+static int tosa_set_spk(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ -+ if (tosa_spk_func == ucontrol->value.integer.value[0]) -+ return 0; -+ -+ tosa_spk_func = ucontrol->value.integer.value[0]; -+ tosa_ext_control(codec); -+ return 1; -+} -+ -+/* tosa dapm event handlers */ -+static int tosa_hp_event(struct snd_soc_dapm_widget *w, int event) -+{ -+ if (SND_SOC_DAPM_EVENT_ON(event)) -+ set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE); -+ else -+ reset_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE); -+ return 0; -+} -+ -+/* tosa machine dapm widgets */ -+static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = { -+SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event), -+SND_SOC_DAPM_HP("Headset Jack", NULL), -+SND_SOC_DAPM_MIC("Mic (Internal)", NULL), -+SND_SOC_DAPM_SPK("Speaker", NULL), -+}; -+ -+/* tosa audio map */ -+static const char *audio_map[][3] = { -+ -+ /* headphone connected to HPOUTL, HPOUTR */ -+ {"Headphone Jack", NULL, "HPOUTL"}, -+ {"Headphone Jack", NULL, "HPOUTR"}, -+ -+ /* ext speaker connected to LOUT2, ROUT2 */ -+ {"Speaker", NULL, "LOUT2"}, -+ {"Speaker", NULL, "ROUT2"}, -+ -+ /* internal mic is connected to mic1, mic2 differential - with bias */ -+ {"MIC1", NULL, "Mic Bias"}, -+ {"MIC2", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Mic (Internal)"}, -+ -+ /* headset is connected to HPOUTR, and LINEINR with bias */ -+ {"Headset Jack", NULL, "HPOUTR"}, -+ {"LINEINR", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Headset Jack"}, -+ -+ {NULL, NULL, NULL}, -+}; -+ -+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", -+ "Off"}; -+static const char *spk_function[] = {"On", "Off"}; -+static const struct soc_enum tosa_enum[] = { -+ SOC_ENUM_SINGLE_EXT(5, jack_function), -+ SOC_ENUM_SINGLE_EXT(2, spk_function), -+}; -+ -+static const struct snd_kcontrol_new tosa_controls[] = { -+ SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack, -+ tosa_set_jack), -+ SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk, -+ tosa_set_spk), -+}; -+ -+static int tosa_ac97_init(struct snd_soc_codec *codec) -+{ -+ int i, err; -+ -+ snd_soc_dapm_set_endpoint(codec, "OUT3", 0); -+ snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); -+ -+ /* add tosa specific controls */ -+ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&tosa_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ /* add tosa specific widgets */ -+ for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]); -+ } -+ -+ /* set up tosa specific audio path audio_map */ -+ for (i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ return 0; -+} -+ -+static struct snd_soc_dai_link tosa_dai[] = { -+{ -+ .name = "AC97", -+ .stream_name = "AC97 HiFi", -+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], -+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], -+ .init = tosa_ac97_init, -+}, -+{ -+ .name = "AC97 Aux", -+ .stream_name = "AC97 Aux", -+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], -+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], -+}, -+}; -+ -+static struct snd_soc_machine tosa = { -+ .name = "Tosa", -+ .dai_link = tosa_dai, -+ .num_links = ARRAY_SIZE(tosa_dai), -+ .ops = &tosa_ops, -+}; -+ -+static struct snd_soc_device tosa_snd_devdata = { -+ .machine = &tosa, -+ .platform = &pxa2xx_soc_platform, -+ .codec_dev = &soc_codec_dev_wm9712, -+}; -+ -+static struct platform_device *tosa_snd_device; -+ -+static int __init tosa_init(void) -+{ -+ int ret; -+ -+ if (!machine_is_tosa()) -+ return -ENODEV; -+ -+ tosa_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!tosa_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(tosa_snd_device, &tosa_snd_devdata); -+ tosa_snd_devdata.dev = &tosa_snd_device->dev; -+ ret = platform_device_add(tosa_snd_device); -+ -+ if (ret) -+ platform_device_put(tosa_snd_device); -+ -+ return ret; -+} -+ -+static void __exit tosa_exit(void) -+{ -+ platform_device_unregister(tosa_snd_device); -+} -+ -+module_init(tosa_init); -+module_exit(tosa_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Richard Purdie"); -+MODULE_DESCRIPTION("ALSA SoC Tosa"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/soc-dapm.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/soc-dapm.c -@@ -0,0 +1,1327 @@ -+/* -+ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 12th Aug 2005 Initial version. -+ * 25th Oct 2005 Implemented path power domain. -+ * 18th Dec 2005 Implemented machine and stream level power domain. -+ * -+ * Features: -+ * o Changes power status of internal codec blocks depending on the -+ * dynamic configuration of codec internal audio paths and active -+ * DAC's/ADC's. -+ * o Platform power domain - can support external components i.e. amps and -+ * mic/meadphone insertion events. -+ * o Automatic Mic Bias support -+ * o Jack insertion power event initiation - e.g. hp insertion will enable -+ * sinks, dacs, etc -+ * o Delayed powerdown of audio susbsytem to reduce pops between a quick -+ * device reopen. -+ * -+ * Todo: -+ * o DAPM power change sequencing - allow for configurable per -+ * codec sequences. -+ * o Support for analogue bias optimisation. -+ * o Support for reduced codec oversampling rates. -+ * o Support for reduced codec bias currents. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/bitops.h> -+#include <linux/platform_device.h> -+#include <linux/jiffies.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+/* debug */ -+#define DAPM_DEBUG 0 -+#if DAPM_DEBUG -+#define dump_dapm(codec, action) dbg_dump_dapm(codec, action) -+#define dbg(format, arg...) printk(format, ## arg) -+#else -+#define dump_dapm(codec, action) -+#define dbg(format, arg...) -+#endif -+ -+#define POP_DEBUG 0 -+#if POP_DEBUG -+#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */ -+#define pop_wait(time) schedule_timeout_interruptible(msecs_to_jiffies(time)) -+#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME) -+#else -+#define pop_dbg(format, arg...) -+#define pop_wait(time) -+#endif -+ -+/* dapm power sequences - make this per codec in the future */ -+static int dapm_up_seq[] = { -+ snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic, -+ snd_soc_dapm_mux, snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_pga, -+ snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post -+}; -+static int dapm_down_seq[] = { -+ snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, -+ snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic, -+ snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_post -+}; -+ -+static int dapm_status = 1; -+module_param(dapm_status, int, 0); -+MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); -+ -+/* create a new dapm widget */ -+static struct snd_soc_dapm_widget *dapm_cnew_widget( -+ const struct snd_soc_dapm_widget *_widget) -+{ -+ struct snd_soc_dapm_widget* widget; -+ widget = kmalloc(sizeof(struct snd_soc_dapm_widget), GFP_KERNEL); -+ if (!widget) -+ return NULL; -+ -+ memcpy(widget, _widget, sizeof(struct snd_soc_dapm_widget)); -+ return widget; -+} -+ -+/* set up initial codec paths */ -+static void dapm_set_path_status(struct snd_soc_dapm_widget *w, -+ struct snd_soc_dapm_path *p, int i) -+{ -+ switch (w->id) { -+ case snd_soc_dapm_switch: -+ case snd_soc_dapm_mixer: { -+ int val; -+ int reg = w->kcontrols[i].private_value & 0xff; -+ int shift = (w->kcontrols[i].private_value >> 8) & 0x0f; -+ int mask = (w->kcontrols[i].private_value >> 16) & 0xff; -+ int invert = (w->kcontrols[i].private_value >> 24) & 0x01; -+ -+ val = snd_soc_read(w->codec, reg); -+ val = (val >> shift) & mask; -+ -+ if ((invert && !val) || (!invert && val)) -+ p->connect = 1; -+ else -+ p->connect = 0; -+ } -+ break; -+ case snd_soc_dapm_mux: { -+ struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; -+ int val, item, bitmask; -+ -+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) -+ ; -+ val = snd_soc_read(w->codec, e->reg); -+ item = (val >> e->shift_l) & (bitmask - 1); -+ -+ p->connect = 0; -+ for (i = 0; i < e->mask; i++) { -+ if (!(strcmp(p->name, e->texts[i])) && item == i) -+ p->connect = 1; -+ } -+ } -+ break; -+ /* does not effect routing - always connected */ -+ case snd_soc_dapm_pga: -+ case snd_soc_dapm_output: -+ case snd_soc_dapm_adc: -+ case snd_soc_dapm_input: -+ case snd_soc_dapm_dac: -+ case snd_soc_dapm_micbias: -+ case snd_soc_dapm_vmid: -+ p->connect = 1; -+ break; -+ /* does effect routing - dynamically connected */ -+ case snd_soc_dapm_hp: -+ case snd_soc_dapm_mic: -+ case snd_soc_dapm_spk: -+ case snd_soc_dapm_line: -+ case snd_soc_dapm_pre: -+ case snd_soc_dapm_post: -+ p->connect = 0; -+ break; -+ } -+} -+ -+/* connect mux widget to it's interconnecting audio paths */ -+static int dapm_connect_mux(struct snd_soc_codec *codec, -+ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, -+ struct snd_soc_dapm_path *path, const char *control_name, -+ const struct snd_kcontrol_new *kcontrol) -+{ -+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; -+ int i; -+ -+ for (i = 0; i < e->mask; i++) { -+ if (!(strcmp(control_name, e->texts[i]))) { -+ list_add(&path->list, &codec->dapm_paths); -+ list_add(&path->list_sink, &dest->sources); -+ list_add(&path->list_source, &src->sinks); -+ path->name = (char*)e->texts[i]; -+ dapm_set_path_status(dest, path, 0); -+ return 0; -+ } -+ } -+ -+ return -ENODEV; -+} -+ -+/* connect mixer widget to it's interconnecting audio paths */ -+static int dapm_connect_mixer(struct snd_soc_codec *codec, -+ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, -+ struct snd_soc_dapm_path *path, const char *control_name) -+{ -+ int i; -+ -+ /* search for mixer kcontrol */ -+ for (i = 0; i < dest->num_kcontrols; i++) { -+ if (!strcmp(control_name, dest->kcontrols[i].name)) { -+ list_add(&path->list, &codec->dapm_paths); -+ list_add(&path->list_sink, &dest->sources); -+ list_add(&path->list_source, &src->sinks); -+ path->name = dest->kcontrols[i].name; -+ dapm_set_path_status(dest, path, i); -+ return 0; -+ } -+ } -+ return -ENODEV; -+} -+ -+/* update dapm codec register bits */ -+static int dapm_update_bits(struct snd_soc_dapm_widget *widget) -+{ -+ int change, power; -+ unsigned short old, new; -+ struct snd_soc_codec *codec = widget->codec; -+ -+ /* check for valid widgets */ -+ if (widget->reg < 0 || widget->id == snd_soc_dapm_input || -+ widget->id == snd_soc_dapm_output || -+ widget->id == snd_soc_dapm_hp || -+ widget->id == snd_soc_dapm_mic || -+ widget->id == snd_soc_dapm_line || -+ widget->id == snd_soc_dapm_spk) -+ return 0; -+ -+ power = widget->power; -+ if (widget->invert) -+ power = (power ? 0:1); -+ -+ old = snd_soc_read(codec, widget->reg); -+ new = (old & ~(0x1 << widget->shift)) | (power << widget->shift); -+ -+ change = old != new; -+ if (change) { -+ pop_dbg("pop test %s : %s in %d ms\n", widget->name, -+ widget->power ? "on" : "off", POP_TIME); -+ snd_soc_write(codec, widget->reg, new); -+ pop_wait(POP_TIME); -+ } -+ dbg("reg old %x new %x change %d\n", old, new, change); -+ return change; -+} -+ -+/* ramps the volume up or down to minimise pops before or after a -+ * DAPM power event */ -+static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) -+{ -+ const struct snd_kcontrol_new *k = widget->kcontrols; -+ -+ if (widget->muted && !power) -+ return 0; -+ if (!widget->muted && power) -+ return 0; -+ -+ if (widget->num_kcontrols && k) { -+ int reg = k->private_value & 0xff; -+ int shift = (k->private_value >> 8) & 0x0f; -+ int mask = (k->private_value >> 16) & 0xff; -+ int invert = (k->private_value >> 24) & 0x01; -+ -+ if (power) { -+ int i; -+ /* power up has happended, increase volume to last level */ -+ if (invert) { -+ for (i = mask; i > widget->saved_value; i--) -+ snd_soc_update_bits(widget->codec, reg, mask, i); -+ } else { -+ for (i = 0; i < widget->saved_value; i++) -+ snd_soc_update_bits(widget->codec, reg, mask, i); -+ } -+ widget->muted = 0; -+ } else { -+ /* power down is about to occur, decrease volume to mute */ -+ int val = snd_soc_read(widget->codec, reg); -+ int i = widget->saved_value = (val >> shift) & mask; -+ if (invert) { -+ for (; i < mask; i++) -+ snd_soc_update_bits(widget->codec, reg, mask, i); -+ } else { -+ for (; i > 0; i--) -+ snd_soc_update_bits(widget->codec, reg, mask, i); -+ } -+ widget->muted = 1; -+ } -+ } -+ return 0; -+} -+ -+/* create new dapm mixer control */ -+static int dapm_new_mixer(struct snd_soc_codec *codec, -+ struct snd_soc_dapm_widget *w) -+{ -+ int i, ret = 0; -+ char name[32]; -+ struct snd_soc_dapm_path *path; -+ -+ /* add kcontrol */ -+ for (i = 0; i < w->num_kcontrols; i++) { -+ -+ /* match name */ -+ list_for_each_entry(path, &w->sources, list_sink) { -+ -+ /* mixer/mux paths name must match control name */ -+ if (path->name != (char*)w->kcontrols[i].name) -+ continue; -+ -+ /* add dapm control with long name */ -+ snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name); -+ path->long_name = kstrdup (name, GFP_KERNEL); -+ if (path->long_name == NULL) -+ return -ENOMEM; -+ -+ path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, -+ path->long_name); -+ ret = snd_ctl_add(codec->card, path->kcontrol); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n", -+ path->long_name); -+ kfree(path->long_name); -+ path->long_name = NULL; -+ return ret; -+ } -+ } -+ } -+ return ret; -+} -+ -+/* create new dapm mux control */ -+static int dapm_new_mux(struct snd_soc_codec *codec, -+ struct snd_soc_dapm_widget *w) -+{ -+ struct snd_soc_dapm_path *path = NULL; -+ struct snd_kcontrol *kcontrol; -+ int ret = 0; -+ -+ if (!w->num_kcontrols) { -+ printk(KERN_ERR "asoc: mux %s has no controls\n", w->name); -+ return -EINVAL; -+ } -+ -+ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); -+ ret = snd_ctl_add(codec->card, kcontrol); -+ if (ret < 0) -+ goto err; -+ -+ list_for_each_entry(path, &w->sources, list_sink) -+ path->kcontrol = kcontrol; -+ -+ return ret; -+ -+err: -+ printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name); -+ return ret; -+} -+ -+/* create new dapm volume control */ -+static int dapm_new_pga(struct snd_soc_codec *codec, -+ struct snd_soc_dapm_widget *w) -+{ -+ struct snd_kcontrol *kcontrol; -+ int ret = 0; -+ -+ if (!w->num_kcontrols) -+ return -EINVAL; -+ -+ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); -+ ret = snd_ctl_add(codec->card, kcontrol); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name); -+ return ret; -+ } -+ -+ return ret; -+} -+ -+/* reset 'walked' bit for each dapm path */ -+static inline void dapm_clear_walk(struct snd_soc_codec *codec) -+{ -+ struct snd_soc_dapm_path *p; -+ -+ list_for_each_entry(p, &codec->dapm_paths, list) -+ p->walked = 0; -+} -+ -+/* -+ * Recursively check for a completed path to an active or physically connected -+ * output widget. Returns number of complete paths. -+ */ -+static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) -+{ -+ struct snd_soc_dapm_path *path; -+ int con = 0; -+ -+ if (widget->id == snd_soc_dapm_adc && widget->active) -+ return 1; -+ -+ if (widget->connected) { -+ /* connected pin ? */ -+ if (widget->id == snd_soc_dapm_output && !widget->ext) -+ return 1; -+ -+ /* connected jack or spk ? */ -+ if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk || -+ widget->id == snd_soc_dapm_line) -+ return 1; -+ } -+ -+ list_for_each_entry(path, &widget->sinks, list_source) { -+ if (path->walked) -+ continue; -+ -+ if (path->sink && path->connect) { -+ path->walked = 1; -+ con += is_connected_output_ep(path->sink); -+ } -+ } -+ -+ return con; -+} -+ -+/* -+ * Recursively check for a completed path to an active or physically connected -+ * input widget. Returns number of complete paths. -+ */ -+static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) -+{ -+ struct snd_soc_dapm_path *path; -+ int con = 0; -+ -+ /* active stream ? */ -+ if (widget->id == snd_soc_dapm_dac && widget->active) -+ return 1; -+ -+ if (widget->connected) { -+ /* connected pin ? */ -+ if (widget->id == snd_soc_dapm_input && !widget->ext) -+ return 1; -+ -+ /* connected VMID/Bias for lower pops */ -+ if (widget->id == snd_soc_dapm_vmid) -+ return 1; -+ -+ /* connected jack ? */ -+ if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line) -+ return 1; -+ } -+ -+ list_for_each_entry(path, &widget->sources, list_sink) { -+ if (path->walked) -+ continue; -+ -+ if (path->source && path->connect) { -+ path->walked = 1; -+ con += is_connected_input_ep(path->source); -+ } -+ } -+ -+ return con; -+} -+ -+/* -+ * Scan each dapm widget for complete audio path. -+ * A complete path is a route that has valid endpoints i.e.:- -+ * -+ * o DAC to output pin. -+ * o Input Pin to ADC. -+ * o Input pin to Output pin (bypass, sidetone) -+ * o DAC to ADC (loopback). -+ */ -+int dapm_power_widgets(struct snd_soc_codec *codec, int event) -+{ -+ struct snd_soc_dapm_widget *w; -+ int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power; -+ -+ /* do we have a sequenced stream event */ -+ if (event == SND_SOC_DAPM_STREAM_START) { -+ c = ARRAY_SIZE(dapm_up_seq); -+ seq = dapm_up_seq; -+ } else if (event == SND_SOC_DAPM_STREAM_STOP) { -+ c = ARRAY_SIZE(dapm_down_seq); -+ seq = dapm_down_seq; -+ } -+ -+ for(i = 0; i < c; i++) { -+ list_for_each_entry(w, &codec->dapm_widgets, list) { -+ -+ /* is widget in stream order */ -+ if (seq && seq[i] && w->id != seq[i]) -+ continue; -+ -+ /* vmid - no action */ -+ if (w->id == snd_soc_dapm_vmid) -+ continue; -+ -+ /* active ADC */ -+ if (w->id == snd_soc_dapm_adc && w->active) { -+ in = is_connected_input_ep(w); -+ dapm_clear_walk(w->codec); -+ w->power = (in != 0) ? 1 : 0; -+ dapm_update_bits(w); -+ continue; -+ } -+ -+ /* active DAC */ -+ if (w->id == snd_soc_dapm_dac && w->active) { -+ out = is_connected_output_ep(w); -+ dapm_clear_walk(w->codec); -+ w->power = (out != 0) ? 1 : 0; -+ dapm_update_bits(w); -+ continue; -+ } -+ -+ /* programmable gain/attenuation */ -+ if (w->id == snd_soc_dapm_pga) { -+ int on; -+ in = is_connected_input_ep(w); -+ dapm_clear_walk(w->codec); -+ out = is_connected_output_ep(w); -+ dapm_clear_walk(w->codec); -+ w->power = on = (out != 0 && in != 0) ? 1 : 0; -+ -+ if (!on) -+ dapm_set_pga(w, on); /* lower volume to reduce pops */ -+ dapm_update_bits(w); -+ if (on) -+ dapm_set_pga(w, on); /* restore volume from zero */ -+ -+ continue; -+ } -+ -+ /* pre and post event widgets */ -+ if (w->id == snd_soc_dapm_pre) { -+ if (!w->event) -+ continue; -+ -+ if (event == SND_SOC_DAPM_STREAM_START) { -+ ret = w->event(w, SND_SOC_DAPM_PRE_PMU); -+ if (ret < 0) -+ return ret; -+ } else if (event == SND_SOC_DAPM_STREAM_STOP) { -+ ret = w->event(w, SND_SOC_DAPM_PRE_PMD); -+ if (ret < 0) -+ return ret; -+ } -+ continue; -+ } -+ if (w->id == snd_soc_dapm_post) { -+ if (!w->event) -+ continue; -+ -+ if (event == SND_SOC_DAPM_STREAM_START) { -+ ret = w->event(w, SND_SOC_DAPM_POST_PMU); -+ if (ret < 0) -+ return ret; -+ } else if (event == SND_SOC_DAPM_STREAM_STOP) { -+ ret = w->event(w, SND_SOC_DAPM_POST_PMD); -+ if (ret < 0) -+ return ret; -+ } -+ continue; -+ } -+ -+ /* all other widgets */ -+ in = is_connected_input_ep(w); -+ dapm_clear_walk(w->codec); -+ out = is_connected_output_ep(w); -+ dapm_clear_walk(w->codec); -+ power = (out != 0 && in != 0) ? 1 : 0; -+ power_change = (w->power == power) ? 0: 1; -+ w->power = power; -+ -+ /* call any power change event handlers */ -+ if (power_change) { -+ if (w->event) { -+ dbg("power %s event for %s flags %x\n", -+ w->power ? "on" : "off", w->name, w->event_flags); -+ if (power) { -+ /* power up event */ -+ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { -+ ret = w->event(w, SND_SOC_DAPM_PRE_PMU); -+ if (ret < 0) -+ return ret; -+ } -+ dapm_update_bits(w); -+ if (w->event_flags & SND_SOC_DAPM_POST_PMU){ -+ ret = w->event(w, SND_SOC_DAPM_POST_PMU); -+ if (ret < 0) -+ return ret; -+ } -+ } else { -+ /* power down event */ -+ if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { -+ ret = w->event(w, SND_SOC_DAPM_PRE_PMD); -+ if (ret < 0) -+ return ret; -+ } -+ dapm_update_bits(w); -+ if (w->event_flags & SND_SOC_DAPM_POST_PMD) { -+ ret = w->event(w, SND_SOC_DAPM_POST_PMD); -+ if (ret < 0) -+ return ret; -+ } -+ } -+ } else -+ /* no event handler */ -+ dapm_update_bits(w); -+ } -+ } -+ } -+ -+ return ret; -+} -+ -+#if DAPM_DEBUG -+static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) -+{ -+ struct snd_soc_dapm_widget *w; -+ struct snd_soc_dapm_path *p = NULL; -+ int in, out; -+ -+ printk("DAPM %s %s\n", codec->name, action); -+ -+ list_for_each_entry(w, &codec->dapm_widgets, list) { -+ -+ /* only display widgets that effect routing */ -+ switch (w->id) { -+ case snd_soc_dapm_pre: -+ case snd_soc_dapm_post: -+ case snd_soc_dapm_vmid: -+ continue; -+ case snd_soc_dapm_mux: -+ case snd_soc_dapm_output: -+ case snd_soc_dapm_input: -+ case snd_soc_dapm_switch: -+ case snd_soc_dapm_hp: -+ case snd_soc_dapm_mic: -+ case snd_soc_dapm_spk: -+ case snd_soc_dapm_line: -+ case snd_soc_dapm_micbias: -+ case snd_soc_dapm_dac: -+ case snd_soc_dapm_adc: -+ case snd_soc_dapm_pga: -+ case snd_soc_dapm_mixer: -+ if (w->name) { -+ in = is_connected_input_ep(w); -+ dapm_clear_walk(w->codec); -+ out = is_connected_output_ep(w); -+ dapm_clear_walk(w->codec); -+ printk("%s: %s in %d out %d\n", w->name, -+ w->power ? "On":"Off",in, out); -+ -+ list_for_each_entry(p, &w->sources, list_sink) { -+ if (p->connect) -+ printk(" in %s %s\n", p->name ? p->name : "static", -+ p->source->name); -+ } -+ list_for_each_entry(p, &w->sinks, list_source) { -+ p = list_entry(lp, struct snd_soc_dapm_path, list_source); -+ if (p->connect) -+ printk(" out %s %s\n", p->name ? p->name : "static", -+ p->sink->name); -+ } -+ } -+ break; -+ } -+ } -+} -+#endif -+ -+/* test and update the power status of a mux widget */ -+int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, -+ struct snd_kcontrol *kcontrol, int mask, int val, struct soc_enum* e) -+{ -+ struct snd_soc_dapm_path *path; -+ int found = 0; -+ -+ if (widget->id != snd_soc_dapm_mux) -+ return -ENODEV; -+ -+ if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) -+ return 0; -+ -+ /* find dapm widget path assoc with kcontrol */ -+ list_for_each_entry(path, &widget->codec->dapm_paths, list) { -+ if (path->kcontrol != kcontrol) -+ continue; -+ -+ if (!path->name || ! e->texts[val]) -+ continue; -+ -+ found = 1; -+ /* we now need to match the string in the enum to the path */ -+ if (!(strcmp(path->name, e->texts[val]))) -+ path->connect = 1; /* new connection */ -+ else -+ path->connect = 0; /* old connection must be powered down */ -+ } -+ -+ if (found) -+ dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); -+ -+ return 0; -+} -+EXPORT_SYMBOL_GPL(dapm_mux_update_power); -+ -+/* test and update the power status of a mixer widget */ -+int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, -+ struct snd_kcontrol *kcontrol, int reg, int val_mask, int val, int invert) -+{ -+ struct snd_soc_dapm_path *path; -+ int found = 0; -+ -+ if (widget->id != snd_soc_dapm_mixer) -+ return -ENODEV; -+ -+ if (!snd_soc_test_bits(widget->codec, reg, val_mask, val)) -+ return 0; -+ -+ /* find dapm widget path assoc with kcontrol */ -+ list_for_each_entry(path, &widget->codec->dapm_paths, list) { -+ if (path->kcontrol != kcontrol) -+ continue; -+ -+ /* found, now check type */ -+ found = 1; -+ if (val) -+ /* new connection */ -+ path->connect = invert ? 0:1; -+ else -+ /* old connection must be powered down */ -+ path->connect = invert ? 1:0; -+ break; -+ } -+ -+ if (found) -+ dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); -+ -+ return 0; -+} -+EXPORT_SYMBOL_GPL(dapm_mixer_update_power); -+ -+/* show dapm widget status in sys fs */ -+static ssize_t dapm_widget_show(struct device *dev, -+ struct device_attribute *attr, char *buf) -+{ -+ struct snd_soc_device *devdata = dev_get_drvdata(dev); -+ struct snd_soc_codec *codec = devdata->codec; -+ struct snd_soc_dapm_widget *w; -+ int count = 0; -+ char *state = "not set"; -+ -+ list_for_each_entry(w, &codec->dapm_widgets, list) { -+ -+ /* only display widgets that burnm power */ -+ switch (w->id) { -+ case snd_soc_dapm_hp: -+ case snd_soc_dapm_mic: -+ case snd_soc_dapm_spk: -+ case snd_soc_dapm_line: -+ case snd_soc_dapm_micbias: -+ case snd_soc_dapm_dac: -+ case snd_soc_dapm_adc: -+ case snd_soc_dapm_pga: -+ case snd_soc_dapm_mixer: -+ if (w->name) -+ count += sprintf(buf + count, "%s: %s\n", -+ w->name, w->power ? "On":"Off"); -+ break; -+ default: -+ break; -+ } -+ } -+ -+ switch(codec->dapm_state){ -+ case SNDRV_CTL_POWER_D0: -+ state = "D0"; -+ break; -+ case SNDRV_CTL_POWER_D1: -+ state = "D1"; -+ break; -+ case SNDRV_CTL_POWER_D2: -+ state = "D2"; -+ break; -+ case SNDRV_CTL_POWER_D3hot: -+ state = "D3hot"; -+ break; -+ case SNDRV_CTL_POWER_D3cold: -+ state = "D3cold"; -+ break; -+ } -+ count += sprintf(buf + count, "PM State: %s\n", state); -+ -+ return count; -+} -+ -+static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -+ -+int snd_soc_dapm_sys_add(struct device *dev) -+{ -+ int ret = 0; -+ -+ if (dapm_status) -+ ret = device_create_file(dev, &dev_attr_dapm_widget); -+ -+ return ret; -+} -+ -+static void snd_soc_dapm_sys_remove(struct device *dev) -+{ -+ if (dapm_status) -+ device_remove_file(dev, &dev_attr_dapm_widget); -+} -+ -+/* free all dapm widgets and resources */ -+void dapm_free_widgets(struct snd_soc_codec *codec) -+{ -+ struct snd_soc_dapm_widget *w, *next_w; -+ struct snd_soc_dapm_path *p, *next_p; -+ -+ list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) { -+ list_del(&w->list); -+ kfree(w); -+ } -+ -+ list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) { -+ list_del(&p->list); -+ kfree(p->long_name); -+ kfree(p); -+ } -+} -+ -+/** -+ * snd_soc_dapm_sync_endpoints - scan and power dapm paths -+ * @codec: audio codec -+ * -+ * Walks all dapm audio paths and powers widgets according to their -+ * stream or path usage. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) -+{ -+ return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); -+ -+/** -+ * snd_soc_dapm_connect_input - connect dapm widgets -+ * @codec: audio codec -+ * @sink: name of target widget -+ * @control: mixer control name -+ * @source: name of source name -+ * -+ * Connects 2 dapm widgets together via a named audio path. The sink is -+ * the widget receiving the audio signal, whilst the source is the sender -+ * of the audio signal. -+ * -+ * Returns 0 for success else error. -+ */ -+int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, -+ const char * control, const char *source) -+{ -+ struct snd_soc_dapm_path *path; -+ struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; -+ int ret = 0; -+ -+ /* find src and dest widgets */ -+ list_for_each_entry(w, &codec->dapm_widgets, list) { -+ -+ if (!wsink && !(strcmp(w->name, sink))) { -+ wsink = w; -+ continue; -+ } -+ if (!wsource && !(strcmp(w->name, source))) { -+ wsource = w; -+ } -+ } -+ -+ if (wsource == NULL || wsink == NULL) -+ return -ENODEV; -+ -+ path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL); -+ if (!path) -+ return -ENOMEM; -+ -+ path->source = wsource; -+ path->sink = wsink; -+ INIT_LIST_HEAD(&path->list); -+ INIT_LIST_HEAD(&path->list_source); -+ INIT_LIST_HEAD(&path->list_sink); -+ -+ /* check for external widgets */ -+ if (wsink->id == snd_soc_dapm_input) { -+ if (wsource->id == snd_soc_dapm_micbias || -+ wsource->id == snd_soc_dapm_mic || -+ wsink->id == snd_soc_dapm_line) -+ wsink->ext = 1; -+ } -+ if (wsource->id == snd_soc_dapm_output) { -+ if (wsink->id == snd_soc_dapm_spk || -+ wsink->id == snd_soc_dapm_hp || -+ wsink->id == snd_soc_dapm_line) -+ wsource->ext = 1; -+ } -+ -+ /* connect static paths */ -+ if (control == NULL) { -+ list_add(&path->list, &codec->dapm_paths); -+ list_add(&path->list_sink, &wsink->sources); -+ list_add(&path->list_source, &wsource->sinks); -+ path->connect = 1; -+ return 0; -+ } -+ -+ /* connect dynamic paths */ -+ switch(wsink->id) { -+ case snd_soc_dapm_adc: -+ case snd_soc_dapm_dac: -+ case snd_soc_dapm_pga: -+ case snd_soc_dapm_input: -+ case snd_soc_dapm_output: -+ case snd_soc_dapm_micbias: -+ case snd_soc_dapm_vmid: -+ case snd_soc_dapm_pre: -+ case snd_soc_dapm_post: -+ list_add(&path->list, &codec->dapm_paths); -+ list_add(&path->list_sink, &wsink->sources); -+ list_add(&path->list_source, &wsource->sinks); -+ path->connect = 1; -+ return 0; -+ case snd_soc_dapm_mux: -+ ret = dapm_connect_mux(codec, wsource, wsink, path, control, -+ &wsink->kcontrols[0]); -+ if (ret != 0) -+ goto err; -+ break; -+ case snd_soc_dapm_switch: -+ case snd_soc_dapm_mixer: -+ ret = dapm_connect_mixer(codec, wsource, wsink, path, control); -+ if (ret != 0) -+ goto err; -+ break; -+ case snd_soc_dapm_hp: -+ case snd_soc_dapm_mic: -+ case snd_soc_dapm_line: -+ case snd_soc_dapm_spk: -+ list_add(&path->list, &codec->dapm_paths); -+ list_add(&path->list_sink, &wsink->sources); -+ list_add(&path->list_source, &wsource->sinks); -+ path->connect = 0; -+ return 0; -+ } -+ return 0; -+ -+err: -+ printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source, -+ control, sink); -+ kfree(path); -+ return ret; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input); -+ -+/** -+ * snd_soc_dapm_new_widgets - add new dapm widgets -+ * @codec: audio codec -+ * -+ * Checks the codec for any new dapm widgets and creates them if found. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) -+{ -+ struct snd_soc_dapm_widget *w; -+ -+ mutex_lock(&codec->mutex); -+ list_for_each_entry(w, &codec->dapm_widgets, list) -+ { -+ if (w->new) -+ continue; -+ -+ switch(w->id) { -+ case snd_soc_dapm_switch: -+ case snd_soc_dapm_mixer: -+ dapm_new_mixer(codec, w); -+ break; -+ case snd_soc_dapm_mux: -+ dapm_new_mux(codec, w); -+ break; -+ case snd_soc_dapm_adc: -+ case snd_soc_dapm_dac: -+ case snd_soc_dapm_pga: -+ dapm_new_pga(codec, w); -+ break; -+ case snd_soc_dapm_input: -+ case snd_soc_dapm_output: -+ case snd_soc_dapm_micbias: -+ case snd_soc_dapm_spk: -+ case snd_soc_dapm_hp: -+ case snd_soc_dapm_mic: -+ case snd_soc_dapm_line: -+ case snd_soc_dapm_vmid: -+ case snd_soc_dapm_pre: -+ case snd_soc_dapm_post: -+ break; -+ } -+ w->new = 1; -+ } -+ -+ dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); -+ mutex_unlock(&codec->mutex); -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); -+ -+/** -+ * snd_soc_dapm_get_volsw - dapm mixer get callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to get the value of a dapm mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); -+ int reg = kcontrol->private_value & 0xff; -+ int shift = (kcontrol->private_value >> 8) & 0x0f; -+ int rshift = (kcontrol->private_value >> 12) & 0x0f; -+ int mask = (kcontrol->private_value >> 16) & 0xff; -+ int invert = (kcontrol->private_value >> 24) & 0x01; -+ -+ /* return the saved value if we are powered down */ -+ if (widget->id == snd_soc_dapm_pga && !widget->power) { -+ ucontrol->value.integer.value[0] = widget->saved_value; -+ return 0; -+ } -+ -+ ucontrol->value.integer.value[0] = -+ (snd_soc_read(widget->codec, reg) >> shift) & mask; -+ if (shift != rshift) -+ ucontrol->value.integer.value[1] = -+ (snd_soc_read(widget->codec, reg) >> rshift) & mask; -+ if (invert) { -+ ucontrol->value.integer.value[0] = -+ mask - ucontrol->value.integer.value[0]; -+ if (shift != rshift) -+ ucontrol->value.integer.value[1] = -+ mask - ucontrol->value.integer.value[1]; -+ } -+ -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); -+ -+/** -+ * snd_soc_dapm_put_volsw - dapm mixer set callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to set the value of a dapm mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); -+ int reg = kcontrol->private_value & 0xff; -+ int shift = (kcontrol->private_value >> 8) & 0x0f; -+ int rshift = (kcontrol->private_value >> 12) & 0x0f; -+ int mask = (kcontrol->private_value >> 16) & 0xff; -+ int invert = (kcontrol->private_value >> 24) & 0x01; -+ unsigned short val, val2, val_mask; -+ int ret; -+ -+ val = (ucontrol->value.integer.value[0] & mask); -+ -+ if (invert) -+ val = mask - val; -+ val_mask = mask << shift; -+ val = val << shift; -+ if (shift != rshift) { -+ val2 = (ucontrol->value.integer.value[1] & mask); -+ if (invert) -+ val2 = mask - val2; -+ val_mask |= mask << rshift; -+ val |= val2 << rshift; -+ } -+ -+ mutex_lock(&widget->codec->mutex); -+ widget->value = val; -+ -+ /* save volume value if the widget is powered down */ -+ if (widget->id == snd_soc_dapm_pga && !widget->power) { -+ widget->saved_value = val; -+ mutex_unlock(&widget->codec->mutex); -+ return 1; -+ } -+ -+ dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert); -+ if (widget->event) { -+ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { -+ ret = widget->event(widget, SND_SOC_DAPM_PRE_REG); -+ if (ret < 0) -+ goto out; -+ } -+ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); -+ if (widget->event_flags & SND_SOC_DAPM_POST_REG) -+ ret = widget->event(widget, SND_SOC_DAPM_POST_REG); -+ } else -+ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); -+ -+out: -+ mutex_unlock(&widget->codec->mutex); -+ return ret; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); -+ -+/** -+ * snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to get the value of a dapm enumerated double mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); -+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; -+ unsigned short val, bitmask; -+ -+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) -+ ; -+ val = snd_soc_read(widget->codec, e->reg); -+ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); -+ if (e->shift_l != e->shift_r) -+ ucontrol->value.enumerated.item[1] = -+ (val >> e->shift_r) & (bitmask - 1); -+ -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); -+ -+/** -+ * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to set the value of a dapm enumerated double mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); -+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; -+ unsigned short val, mux; -+ unsigned short mask, bitmask; -+ int ret = 0; -+ -+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) -+ ; -+ if (ucontrol->value.enumerated.item[0] > e->mask - 1) -+ return -EINVAL; -+ mux = ucontrol->value.enumerated.item[0]; -+ val = mux << e->shift_l; -+ mask = (bitmask - 1) << e->shift_l; -+ if (e->shift_l != e->shift_r) { -+ if (ucontrol->value.enumerated.item[1] > e->mask - 1) -+ return -EINVAL; -+ val |= ucontrol->value.enumerated.item[1] << e->shift_r; -+ mask |= (bitmask - 1) << e->shift_r; -+ } -+ -+ mutex_lock(&widget->codec->mutex); -+ widget->value = val; -+ dapm_mux_update_power(widget, kcontrol, mask, mux, e); -+ if (widget->event) { -+ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { -+ ret = widget->event(widget, SND_SOC_DAPM_PRE_REG); -+ if (ret < 0) -+ goto out; -+ } -+ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); -+ if (widget->event_flags & SND_SOC_DAPM_POST_REG) -+ ret = widget->event(widget, SND_SOC_DAPM_POST_REG); -+ } else -+ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); -+ -+out: -+ mutex_unlock(&widget->codec->mutex); -+ return ret; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); -+ -+/** -+ * snd_soc_dapm_new_control - create new dapm control -+ * @codec: audio codec -+ * @widget: widget template -+ * -+ * Creates a new dapm control based upon the template. -+ * -+ * Returns 0 for success else error. -+ */ -+int snd_soc_dapm_new_control(struct snd_soc_codec *codec, -+ const struct snd_soc_dapm_widget *widget) -+{ -+ struct snd_soc_dapm_widget *w; -+ -+ if ((w = dapm_cnew_widget(widget)) == NULL) -+ return -ENOMEM; -+ -+ w->codec = codec; -+ INIT_LIST_HEAD(&w->sources); -+ INIT_LIST_HEAD(&w->sinks); -+ INIT_LIST_HEAD(&w->list); -+ list_add(&w->list, &codec->dapm_widgets); -+ -+ /* machine layer set ups unconnected pins and insertions */ -+ w->connected = 1; -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); -+ -+/** -+ * snd_soc_dapm_stream_event - send a stream event to the dapm core -+ * @codec: audio codec -+ * @stream: stream name -+ * @event: stream event -+ * -+ * Sends a stream event to the dapm core. The core then makes any -+ * necessary widget power changes. -+ * -+ * Returns 0 for success else error. -+ */ -+int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, -+ char *stream, int event) -+{ -+ struct snd_soc_dapm_widget *w; -+ -+ mutex_lock(&codec->mutex); -+ list_for_each_entry(w, &codec->dapm_widgets, list) -+ { -+ if (!w->sname) -+ continue; -+ dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname, -+ stream, event); -+ if (strstr(w->sname, stream)) { -+ switch(event) { -+ case SND_SOC_DAPM_STREAM_START: -+ w->active = 1; -+ break; -+ case SND_SOC_DAPM_STREAM_STOP: -+ w->active = 0; -+ break; -+ case SND_SOC_DAPM_STREAM_SUSPEND: -+ if (w->active) -+ w->suspend = 1; -+ w->active = 0; -+ break; -+ case SND_SOC_DAPM_STREAM_RESUME: -+ if (w->suspend) { -+ w->active = 1; -+ w->suspend = 0; -+ } -+ break; -+ case SND_SOC_DAPM_STREAM_PAUSE_PUSH: -+ break; -+ case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: -+ break; -+ } -+ } -+ } -+ mutex_unlock(&codec->mutex); -+ -+ dapm_power_widgets(codec, event); -+ dump_dapm(codec, __FUNCTION__); -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); -+ -+/** -+ * snd_soc_dapm_set_endpoint - set audio endpoint status -+ * @codec: audio codec -+ * @endpoint: audio signal endpoint (or start point) -+ * @status: point status -+ * -+ * Set audio endpoint status - connected or disconnected. -+ * -+ * Returns 0 for success else error. -+ */ -+int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, -+ char *endpoint, int status) -+{ -+ struct snd_soc_dapm_widget *w; -+ -+ list_for_each_entry(w, &codec->dapm_widgets, list) { -+ if (!strcmp(w->name, endpoint)) { -+ w->connected = status; -+ } -+ } -+ -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); -+ -+/** -+ * snd_soc_dapm_free - free dapm resources -+ * @socdev: SoC device -+ * -+ * Free all dapm widgets and resources. -+ */ -+void snd_soc_dapm_free(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ snd_soc_dapm_sys_remove(socdev->dev); -+ dapm_free_widgets(codec); -+} -+EXPORT_SYMBOL_GPL(snd_soc_dapm_free); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/soc-core.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/soc-core.c -@@ -0,0 +1,2063 @@ -+/* -+ * soc-core.c -- ALSA SoC Audio Layer -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 12th Aug 2005 Initial version. -+ * 25th Oct 2005 Working Codec, Interface and Platform registration. -+ * -+ * TODO: -+ * o Add hw rules to enforce rates, etc. -+ * o More testing with other codecs/machines. -+ * o Add more codecs and platforms to ensure good API coverage. -+ * o Support TDM on PCM and I2S -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/bitops.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+/* debug */ -+#define SOC_DEBUG 1 -+#if SOC_DEBUG -+#define dbg(format, arg...) printk(format, ## arg) -+#else -+#define dbg(format, arg...) -+#endif -+/* debug DAI capabilities matching */ -+#define SOC_DEBUG_DAI 1 -+#if SOC_DEBUG_DAI -+#define dbgc(format, arg...) printk(format, ## arg) -+#else -+#define dbgc(format, arg...) -+#endif -+ -+#define CODEC_CPU(codec, cpu) ((codec << 4) | cpu) -+ -+static DEFINE_MUTEX(pcm_mutex); -+static DEFINE_MUTEX(io_mutex); -+static struct workqueue_struct *soc_workq; -+static struct work_struct soc_stream_work; -+static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); -+ -+/* supported sample rates */ -+/* ATTENTION: these values depend on the definition in pcm.h! */ -+static const unsigned int rates[] = { -+ 5512, 8000, 11025, 16000, 22050, 32000, 44100, -+ 48000, 64000, 88200, 96000, 176400, 192000 -+}; -+ -+/* -+ * This is a timeout to do a DAPM powerdown after a stream is closed(). -+ * It can be used to eliminate pops between different playback streams, e.g. -+ * between two audio tracks. -+ */ -+static int pmdown_time = 5000; -+module_param(pmdown_time, int, 0); -+MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); -+ -+#ifdef CONFIG_SND_SOC_AC97_BUS -+/* unregister ac97 codec */ -+static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) -+{ -+ if (codec->ac97->dev.bus) -+ device_unregister(&codec->ac97->dev); -+ return 0; -+} -+ -+/* stop no dev release warning */ -+static void soc_ac97_device_release(struct device *dev){} -+ -+/* register ac97 codec to bus */ -+static int soc_ac97_dev_register(struct snd_soc_codec *codec) -+{ -+ int err; -+ -+ codec->ac97->dev.bus = &ac97_bus_type; -+ codec->ac97->dev.parent = NULL; -+ codec->ac97->dev.release = soc_ac97_device_release; -+ -+ snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", -+ codec->card->number, 0, codec->name); -+ err = device_register(&codec->ac97->dev); -+ if (err < 0) { -+ snd_printk(KERN_ERR "Can't register ac97 bus\n"); -+ codec->ac97->dev.bus = NULL; -+ return err; -+ } -+ return 0; -+} -+#endif -+ -+static inline const char* get_dai_name(int type) -+{ -+ switch(type) { -+ case SND_SOC_DAI_AC97: -+ return "AC97"; -+ case SND_SOC_DAI_I2S: -+ return "I2S"; -+ case SND_SOC_DAI_PCM: -+ return "PCM"; -+ } -+ return NULL; -+} -+ -+/* get rate format from rate */ -+static inline int soc_get_rate_format(int rate) -+{ -+ int i; -+ -+ for (i = 0; i < ARRAY_SIZE(rates); i++) { -+ if (rates[i] == rate) -+ return 1 << i; -+ } -+ return 0; -+} -+ -+/* gets the audio system mclk/sysclk for the given parameters */ -+static unsigned inline int soc_get_mclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_machine *machine = socdev->machine; -+ int i; -+ -+ /* find the matching machine config and get it's mclk for the given -+ * sample rate and hardware format */ -+ for(i = 0; i < machine->num_links; i++) { -+ if (machine->dai_link[i].cpu_dai == rtd->cpu_dai && -+ machine->dai_link[i].config_sysclk) -+ return machine->dai_link[i].config_sysclk(rtd, info); -+ } -+ return 0; -+} -+ -+/* changes a bitclk multiplier mask to a divider mask */ -+static u64 soc_bfs_rcw_to_div(u64 bfs, int rate, unsigned int mclk, -+ unsigned int pcmfmt, unsigned int chn) -+{ -+ int i, j; -+ u64 bfs_ = 0; -+ int size = snd_pcm_format_physical_width(pcmfmt), min = 0; -+ -+ if (size <= 0) -+ return 0; -+ -+ /* the minimum bit clock that has enough bandwidth */ -+ min = size * rate * chn; -+ dbgc("rcw --> div min bclk %d with mclk %d\n", min, mclk); -+ -+ for (i = 0; i < 64; i++) { -+ if ((bfs >> i) & 0x1) { -+ j = min * (i + 1); -+ bfs_ |= SND_SOC_FSBD(mclk/j); -+ dbgc("rcw --> div support mult %d\n", -+ SND_SOC_FSBD_REAL(1<<i)); -+ } -+ } -+ -+ return bfs_; -+} -+ -+/* changes a bitclk divider mask to a multiplier mask */ -+static u64 soc_bfs_div_to_rcw(u64 bfs, int rate, unsigned int mclk, -+ unsigned int pcmfmt, unsigned int chn) -+{ -+ int i, j; -+ u64 bfs_ = 0; -+ -+ int size = snd_pcm_format_physical_width(pcmfmt), min = 0; -+ -+ if (size <= 0) -+ return 0; -+ -+ /* the minimum bit clock that has enough bandwidth */ -+ min = size * rate * chn; -+ dbgc("div to rcw min bclk %d with mclk %d\n", min, mclk); -+ -+ for (i = 0; i < 64; i++) { -+ if ((bfs >> i) & 0x1) { -+ j = mclk / (i + 1); -+ if (j >= min) { -+ bfs_ |= SND_SOC_FSBW(j/min); -+ dbgc("div --> rcw support div %d\n", -+ SND_SOC_FSBW_REAL(1<<i)); -+ } -+ } -+ } -+ -+ return bfs_; -+} -+ -+/* changes a constant bitclk to a multiplier mask */ -+static u64 soc_bfs_rate_to_rcw(u64 bfs, int rate, unsigned int mclk, -+ unsigned int pcmfmt, unsigned int chn) -+{ -+ unsigned int bfs_ = rate * bfs; -+ int size = snd_pcm_format_physical_width(pcmfmt), min = 0; -+ -+ if (size <= 0) -+ return 0; -+ -+ /* the minimum bit clock that has enough bandwidth */ -+ min = size * rate * chn; -+ dbgc("rate --> rcw min bclk %d with mclk %d\n", min, mclk); -+ -+ if (bfs_ < min) -+ return 0; -+ else { -+ bfs_ = SND_SOC_FSBW(bfs_/min); -+ dbgc("rate --> rcw support div %d\n", SND_SOC_FSBW_REAL(bfs_)); -+ return bfs_; -+ } -+} -+ -+/* changes a bitclk multiplier mask to a divider mask */ -+static u64 soc_bfs_rate_to_div(u64 bfs, int rate, unsigned int mclk, -+ unsigned int pcmfmt, unsigned int chn) -+{ -+ unsigned int bfs_ = rate * bfs; -+ int size = snd_pcm_format_physical_width(pcmfmt), min = 0; -+ -+ if (size <= 0) -+ return 0; -+ -+ /* the minimum bit clock that has enough bandwidth */ -+ min = size * rate * chn; -+ dbgc("rate --> div min bclk %d with mclk %d\n", min, mclk); -+ -+ if (bfs_ < min) -+ return 0; -+ else { -+ bfs_ = SND_SOC_FSBW(mclk/bfs_); -+ dbgc("rate --> div support div %d\n", SND_SOC_FSBD_REAL(bfs_)); -+ return bfs_; -+ } -+} -+ -+/* Matches codec DAI and SoC CPU DAI hardware parameters */ -+static int soc_hw_match_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_dai_mode *codec_dai_mode = NULL; -+ struct snd_soc_dai_mode *cpu_dai_mode = NULL; -+ struct snd_soc_clock_info clk_info; -+ unsigned int fs, mclk, rate = params_rate(params), -+ chn, j, k, cpu_bclk, codec_bclk, pcmrate; -+ u16 fmt = 0; -+ u64 codec_bfs, cpu_bfs; -+ -+ dbg("asoc: match version %s\n", SND_SOC_VERSION); -+ clk_info.rate = rate; -+ pcmrate = soc_get_rate_format(rate); -+ -+ /* try and find a match from the codec and cpu DAI capabilities */ -+ for (j = 0; j < rtd->codec_dai->caps.num_modes; j++) { -+ for (k = 0; k < rtd->cpu_dai->caps.num_modes; k++) { -+ codec_dai_mode = &rtd->codec_dai->caps.mode[j]; -+ cpu_dai_mode = &rtd->cpu_dai->caps.mode[k]; -+ -+ if (!(codec_dai_mode->pcmrate & cpu_dai_mode->pcmrate & -+ pcmrate)) { -+ dbgc("asoc: DAI[%d:%d] failed to match rate\n", j, k); -+ continue; -+ } -+ -+ fmt = codec_dai_mode->fmt & cpu_dai_mode->fmt; -+ if (!(fmt & SND_SOC_DAIFMT_FORMAT_MASK)) { -+ dbgc("asoc: DAI[%d:%d] failed to match format\n", j, k); -+ continue; -+ } -+ -+ if (!(fmt & SND_SOC_DAIFMT_CLOCK_MASK)) { -+ dbgc("asoc: DAI[%d:%d] failed to match clock masters\n", -+ j, k); -+ continue; -+ } -+ -+ if (!(fmt & SND_SOC_DAIFMT_INV_MASK)) { -+ dbgc("asoc: DAI[%d:%d] failed to match invert\n", j, k); -+ continue; -+ } -+ -+ if (!(codec_dai_mode->pcmfmt & cpu_dai_mode->pcmfmt)) { -+ dbgc("asoc: DAI[%d:%d] failed to match pcm format\n", j, k); -+ continue; -+ } -+ -+ if (!(codec_dai_mode->pcmdir & cpu_dai_mode->pcmdir)) { -+ dbgc("asoc: DAI[%d:%d] failed to match direction\n", j, k); -+ continue; -+ } -+ -+ /* todo - still need to add tdm selection */ -+ rtd->cpu_dai->dai_runtime.fmt = -+ rtd->codec_dai->dai_runtime.fmt = -+ 1 << (ffs(fmt & SND_SOC_DAIFMT_FORMAT_MASK) -1) | -+ 1 << (ffs(fmt & SND_SOC_DAIFMT_CLOCK_MASK) - 1) | -+ 1 << (ffs(fmt & SND_SOC_DAIFMT_INV_MASK) - 1); -+ clk_info.bclk_master = -+ rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK; -+ -+ /* make sure the ratio between rate and master -+ * clock is acceptable*/ -+ fs = (cpu_dai_mode->fs & codec_dai_mode->fs); -+ if (fs == 0) { -+ dbgc("asoc: DAI[%d:%d] failed to match FS\n", j, k); -+ continue; -+ } -+ clk_info.fs = rtd->cpu_dai->dai_runtime.fs = -+ rtd->codec_dai->dai_runtime.fs = fs; -+ -+ /* calculate audio system clocking using slowest clocks possible*/ -+ mclk = soc_get_mclk(rtd, &clk_info); -+ if (mclk == 0) { -+ dbgc("asoc: DAI[%d:%d] configuration not clockable\n", j, k); -+ dbgc("asoc: rate %d fs %d master %x\n", rate, fs, -+ clk_info.bclk_master); -+ continue; -+ } -+ -+ /* calculate word size (per channel) and frame size */ -+ rtd->codec_dai->dai_runtime.pcmfmt = -+ rtd->cpu_dai->dai_runtime.pcmfmt = -+ 1 << params_format(params); -+ -+ chn = params_channels(params); -+ /* i2s always has left and right */ -+ if (params_channels(params) == 1 && -+ rtd->cpu_dai->dai_runtime.fmt & (SND_SOC_DAIFMT_I2S | -+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_LEFT_J)) -+ chn <<= 1; -+ -+ /* Calculate bfs - the ratio between bitclock and the sample rate -+ * We must take into consideration the dividers and multipliers -+ * used in the codec and cpu DAI modes. We always choose the -+ * lowest possible clocks to reduce power. -+ */ -+ switch (CODEC_CPU(codec_dai_mode->flags, cpu_dai_mode->flags)) { -+ case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_DIV): -+ /* cpu & codec bfs dividers */ -+ rtd->cpu_dai->dai_runtime.bfs = -+ rtd->codec_dai->dai_runtime.bfs = -+ 1 << (fls(codec_dai_mode->bfs & cpu_dai_mode->bfs) - 1); -+ break; -+ case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_RCW): -+ /* normalise bfs codec divider & cpu rcw mult */ -+ codec_bfs = soc_bfs_div_to_rcw(codec_dai_mode->bfs, rate, -+ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); -+ rtd->cpu_dai->dai_runtime.bfs = -+ 1 << (ffs(codec_bfs & cpu_dai_mode->bfs) - 1); -+ cpu_bfs = soc_bfs_rcw_to_div(cpu_dai_mode->bfs, rate, mclk, -+ rtd->codec_dai->dai_runtime.pcmfmt, chn); -+ rtd->codec_dai->dai_runtime.bfs = -+ 1 << (fls(codec_dai_mode->bfs & cpu_bfs) - 1); -+ break; -+ case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_DIV): -+ /* normalise bfs codec rcw mult & cpu divider */ -+ codec_bfs = soc_bfs_rcw_to_div(codec_dai_mode->bfs, rate, -+ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); -+ rtd->cpu_dai->dai_runtime.bfs = -+ 1 << (fls(codec_bfs & cpu_dai_mode->bfs) -1); -+ cpu_bfs = soc_bfs_div_to_rcw(cpu_dai_mode->bfs, rate, mclk, -+ rtd->codec_dai->dai_runtime.pcmfmt, chn); -+ rtd->codec_dai->dai_runtime.bfs = -+ 1 << (ffs(codec_dai_mode->bfs & cpu_bfs) -1); -+ break; -+ case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_RCW): -+ /* codec & cpu bfs rate rcw multipliers */ -+ rtd->cpu_dai->dai_runtime.bfs = -+ rtd->codec_dai->dai_runtime.bfs = -+ 1 << (ffs(codec_dai_mode->bfs & cpu_dai_mode->bfs) -1); -+ break; -+ case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_RATE): -+ /* normalise cpu bfs rate const multiplier & codec div */ -+ cpu_bfs = soc_bfs_rate_to_div(cpu_dai_mode->bfs, rate, -+ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); -+ if(codec_dai_mode->bfs & cpu_bfs) { -+ rtd->codec_dai->dai_runtime.bfs = cpu_bfs; -+ rtd->cpu_dai->dai_runtime.bfs = cpu_dai_mode->bfs; -+ } else -+ rtd->cpu_dai->dai_runtime.bfs = 0; -+ break; -+ case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_RATE): -+ /* normalise cpu bfs rate const multiplier & codec rcw mult */ -+ cpu_bfs = soc_bfs_rate_to_rcw(cpu_dai_mode->bfs, rate, -+ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); -+ if(codec_dai_mode->bfs & cpu_bfs) { -+ rtd->codec_dai->dai_runtime.bfs = cpu_bfs; -+ rtd->cpu_dai->dai_runtime.bfs = cpu_dai_mode->bfs; -+ } else -+ rtd->cpu_dai->dai_runtime.bfs = 0; -+ break; -+ case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_RCW): -+ /* normalise cpu bfs rate rcw multiplier & codec const mult */ -+ codec_bfs = soc_bfs_rate_to_rcw(codec_dai_mode->bfs, rate, -+ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); -+ if(cpu_dai_mode->bfs & codec_bfs) { -+ rtd->cpu_dai->dai_runtime.bfs = codec_bfs; -+ rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs; -+ } else -+ rtd->cpu_dai->dai_runtime.bfs = 0; -+ break; -+ case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_DIV): -+ /* normalise cpu bfs div & codec const mult */ -+ codec_bfs = soc_bfs_rate_to_div(codec_dai_mode->bfs, rate, -+ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); -+ if(codec_dai_mode->bfs & codec_bfs) { -+ rtd->cpu_dai->dai_runtime.bfs = codec_bfs; -+ rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs; -+ } else -+ rtd->cpu_dai->dai_runtime.bfs = 0; -+ break; -+ case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_RATE): -+ /* cpu & codec constant mult */ -+ if(codec_dai_mode->bfs == cpu_dai_mode->bfs) -+ rtd->cpu_dai->dai_runtime.bfs = -+ rtd->codec_dai->dai_runtime.bfs = -+ codec_dai_mode->bfs; -+ else -+ rtd->cpu_dai->dai_runtime.bfs = -+ rtd->codec_dai->dai_runtime.bfs = 0; -+ break; -+ default: -+ if(codec_dai_mode->flags == 0) -+ printk(KERN_ERR "asoc: error missing codec DAI flags\n"); -+ else -+ printk(KERN_ERR "asoc: error missing CPU DAI flags\n"); -+ break; -+ } -+ -+ /* make sure the bit clock speed is acceptable */ -+ if (!rtd->cpu_dai->dai_runtime.bfs || -+ !rtd->codec_dai->dai_runtime.bfs) { -+ dbgc("asoc: DAI[%d:%d] failed to match BFS\n", j, k); -+ dbgc("asoc: cpu_dai %llu codec %llu\n", -+ rtd->cpu_dai->dai_runtime.bfs, -+ rtd->codec_dai->dai_runtime.bfs); -+ dbgc("asoc: mclk %d hwfmt %x\n", mclk, fmt); -+ continue; -+ } -+ -+ goto found; -+ } -+ } -+ printk(KERN_ERR "asoc: no matching DAI found between codec and CPU\n"); -+ return -EINVAL; -+ -+found: -+ /* we have matching DAI's, so complete the runtime info */ -+ rtd->codec_dai->dai_runtime.pcmrate = -+ rtd->cpu_dai->dai_runtime.pcmrate = -+ soc_get_rate_format(rate); -+ -+ rtd->codec_dai->dai_runtime.priv = codec_dai_mode->priv; -+ rtd->cpu_dai->dai_runtime.priv = cpu_dai_mode->priv; -+ rtd->codec_dai->dai_runtime.flags = codec_dai_mode->flags; -+ rtd->cpu_dai->dai_runtime.flags = cpu_dai_mode->flags; -+ -+ /* for debug atm */ -+ dbg("asoc: DAI[%d:%d] Match OK\n", j, k); -+ if (rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) { -+ codec_bclk = (rtd->codec_dai->dai_runtime.fs * params_rate(params)) / -+ SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ dbg("asoc: codec fs %d mclk %d bfs div %d bclk %d\n", -+ rtd->codec_dai->dai_runtime.fs, mclk, -+ SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk); -+ } else if(rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RATE) { -+ codec_bclk = params_rate(params) * rtd->codec_dai->dai_runtime.bfs; -+ dbg("asoc: codec fs %d mclk %d bfs rate mult %llu bclk %d\n", -+ rtd->codec_dai->dai_runtime.fs, mclk, -+ rtd->codec_dai->dai_runtime.bfs, codec_bclk); -+ } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RCW) { -+ codec_bclk = params_rate(params) * params_channels(params) * -+ snd_pcm_format_physical_width(rtd->codec_dai->dai_runtime.pcmfmt) * -+ SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs); -+ dbg("asoc: codec fs %d mclk %d bfs rcw mult %d bclk %d\n", -+ rtd->codec_dai->dai_runtime.fs, mclk, -+ SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk); -+ } else -+ codec_bclk = 0; -+ -+ if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) { -+ cpu_bclk = (rtd->cpu_dai->dai_runtime.fs * params_rate(params)) / -+ SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); -+ dbg("asoc: cpu fs %d mclk %d bfs div %d bclk %d\n", -+ rtd->cpu_dai->dai_runtime.fs, mclk, -+ SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk); -+ } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RATE) { -+ cpu_bclk = params_rate(params) * rtd->cpu_dai->dai_runtime.bfs; -+ dbg("asoc: cpu fs %d mclk %d bfs rate mult %llu bclk %d\n", -+ rtd->cpu_dai->dai_runtime.fs, mclk, -+ rtd->cpu_dai->dai_runtime.bfs, cpu_bclk); -+ } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RCW) { -+ cpu_bclk = params_rate(params) * params_channels(params) * -+ snd_pcm_format_physical_width(rtd->cpu_dai->dai_runtime.pcmfmt) * -+ SND_SOC_FSBW_REAL(rtd->cpu_dai->dai_runtime.bfs); -+ dbg("asoc: cpu fs %d mclk %d bfs mult rcw %d bclk %d\n", -+ rtd->cpu_dai->dai_runtime.fs, mclk, -+ SND_SOC_FSBW_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk); -+ } else -+ cpu_bclk = 0; -+ -+ /* -+ * Check we have matching bitclocks. If we don't then it means the -+ * sysclock returned by either the codec or cpu DAI (selected by the -+ * machine sysclock function) is wrong compared with the supported DAI -+ * modes for the codec or cpu DAI. -+ */ -+ if (cpu_bclk != codec_bclk && cpu_bclk){ -+ printk(KERN_ERR -+ "asoc: codec and cpu bitclocks differ, audio may be wrong speed\n" -+ ); -+ printk(KERN_ERR "asoc: codec %d != cpu %d\n", codec_bclk, cpu_bclk); -+ } -+ -+ switch(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ dbg("asoc: DAI codec BCLK master, LRC master\n"); -+ break; -+ case SND_SOC_DAIFMT_CBS_CFM: -+ dbg("asoc: DAI codec BCLK slave, LRC master\n"); -+ break; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ dbg("asoc: DAI codec BCLK master, LRC slave\n"); -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ dbg("asoc: DAI codec BCLK slave, LRC slave\n"); -+ break; -+ } -+ dbg("asoc: mode %x, invert %x\n", -+ rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK, -+ rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK); -+ dbg("asoc: audio rate %d chn %d fmt %x\n", params_rate(params), -+ params_channels(params), params_format(params)); -+ -+ return 0; -+} -+ -+static inline u32 get_rates(struct snd_soc_dai_mode *modes, int nmodes) -+{ -+ int i; -+ u32 rates = 0; -+ -+ for(i = 0; i < nmodes; i++) -+ rates |= modes[i].pcmrate; -+ -+ return rates; -+} -+ -+static inline u64 get_formats(struct snd_soc_dai_mode *modes, int nmodes) -+{ -+ int i; -+ u64 formats = 0; -+ -+ for(i = 0; i < nmodes; i++) -+ formats |= modes[i].pcmfmt; -+ -+ return formats; -+} -+ -+/* -+ * Called by ALSA when a PCM substream is opened, the runtime->hw record is -+ * then initialized and any private data can be allocated. This also calls -+ * startup for the cpu DAI, platform, machine and codec DAI. -+ */ -+static int soc_pcm_open(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct snd_soc_machine *machine = socdev->machine; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_codec_dai *codec_dai = rtd->codec_dai; -+ struct snd_soc_cpu_dai *cpu_dai = rtd->cpu_dai; -+ int ret = 0; -+ -+ mutex_lock(&pcm_mutex); -+ -+ /* startup the audio subsystem */ -+ if (rtd->cpu_dai->ops.startup) { -+ ret = rtd->cpu_dai->ops.startup(substream); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: can't open interface %s\n", -+ rtd->cpu_dai->name); -+ goto out; -+ } -+ } -+ -+ if (platform->pcm_ops->open) { -+ ret = platform->pcm_ops->open(substream); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); -+ goto platform_err; -+ } -+ } -+ -+ if (machine->ops && machine->ops->startup) { -+ ret = machine->ops->startup(substream); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: %s startup failed\n", machine->name); -+ goto machine_err; -+ } -+ } -+ -+ if (rtd->codec_dai->ops.startup) { -+ ret = rtd->codec_dai->ops.startup(substream); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: can't open codec %s\n", -+ rtd->codec_dai->name); -+ goto codec_dai_err; -+ } -+ } -+ -+ /* create runtime params from DMA, codec and cpu DAI */ -+ if (runtime->hw.rates) -+ runtime->hw.rates &= -+ get_rates(codec_dai->caps.mode, codec_dai->caps.num_modes) & -+ get_rates(cpu_dai->caps.mode, cpu_dai->caps.num_modes); -+ else -+ runtime->hw.rates = -+ get_rates(codec_dai->caps.mode, codec_dai->caps.num_modes) & -+ get_rates(cpu_dai->caps.mode, cpu_dai->caps.num_modes); -+ if (runtime->hw.formats) -+ runtime->hw.formats &= -+ get_formats(codec_dai->caps.mode, codec_dai->caps.num_modes) & -+ get_formats(cpu_dai->caps.mode, cpu_dai->caps.num_modes); -+ else -+ runtime->hw.formats = -+ get_formats(codec_dai->caps.mode, codec_dai->caps.num_modes) & -+ get_formats(cpu_dai->caps.mode, cpu_dai->caps.num_modes); -+ -+ /* Check that the codec and cpu DAI's are compatible */ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ runtime->hw.rate_min = -+ max(rtd->codec_dai->playback.rate_min, -+ rtd->cpu_dai->playback.rate_min); -+ runtime->hw.rate_max = -+ min(rtd->codec_dai->playback.rate_max, -+ rtd->cpu_dai->playback.rate_max); -+ runtime->hw.channels_min = -+ max(rtd->codec_dai->playback.channels_min, -+ rtd->cpu_dai->playback.channels_min); -+ runtime->hw.channels_max = -+ min(rtd->codec_dai->playback.channels_max, -+ rtd->cpu_dai->playback.channels_max); -+ } else { -+ runtime->hw.rate_min = -+ max(rtd->codec_dai->capture.rate_min, -+ rtd->cpu_dai->capture.rate_min); -+ runtime->hw.rate_max = -+ min(rtd->codec_dai->capture.rate_max, -+ rtd->cpu_dai->capture.rate_max); -+ runtime->hw.channels_min = -+ max(rtd->codec_dai->capture.channels_min, -+ rtd->cpu_dai->capture.channels_min); -+ runtime->hw.channels_max = -+ min(rtd->codec_dai->capture.channels_max, -+ rtd->cpu_dai->capture.channels_max); -+ } -+ -+ snd_pcm_limit_hw_rates(runtime); -+ if (!runtime->hw.rates) { -+ printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", -+ rtd->codec_dai->name, rtd->cpu_dai->name); -+ goto codec_dai_err; -+ } -+ if (!runtime->hw.formats) { -+ printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", -+ rtd->codec_dai->name, rtd->cpu_dai->name); -+ goto codec_dai_err; -+ } -+ if (!runtime->hw.channels_min || !runtime->hw.channels_max) { -+ printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", -+ rtd->codec_dai->name, rtd->cpu_dai->name); -+ goto codec_dai_err; -+ } -+ -+ dbg("asoc: %s <-> %s info:\n", rtd->codec_dai->name, rtd->cpu_dai->name); -+ dbg("asoc: rate mask 0x%x\n", runtime->hw.rates); -+ dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, -+ runtime->hw.channels_max); -+ dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, -+ runtime->hw.rate_max); -+ -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ rtd->cpu_dai->playback.active = rtd->codec_dai->playback.active = 1; -+ else -+ rtd->cpu_dai->capture.active = rtd->codec_dai->capture.active = 1; -+ rtd->cpu_dai->active = rtd->codec_dai->active = 1; -+ rtd->cpu_dai->runtime = runtime; -+ socdev->codec->active++; -+ mutex_unlock(&pcm_mutex); -+ return 0; -+ -+codec_dai_err: -+ if (machine->ops && machine->ops->shutdown) -+ machine->ops->shutdown(substream); -+ -+machine_err: -+ if (platform->pcm_ops->close) -+ platform->pcm_ops->close(substream); -+ -+platform_err: -+ if (rtd->cpu_dai->ops.shutdown) -+ rtd->cpu_dai->ops.shutdown(substream); -+out: -+ mutex_unlock(&pcm_mutex); -+ return ret; -+} -+ -+/* -+ * Power down the audio subsytem pmdown_time msecs after close is called. -+ * This is to ensure there are no pops or clicks in between any music tracks -+ * due to DAPM power cycling. -+ */ -+static void close_delayed_work(void *data) -+{ -+ struct snd_soc_device *socdev = data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_codec_dai *codec_dai; -+ int i; -+ -+ mutex_lock(&pcm_mutex); -+ for(i = 0; i < codec->num_dai; i++) { -+ codec_dai = &codec->dai[i]; -+ -+ dbg("pop wq checking: %s status: %s waiting: %s\n", -+ codec_dai->playback.stream_name, -+ codec_dai->playback.active ? "active" : "inactive", -+ codec_dai->pop_wait ? "yes" : "no"); -+ -+ /* are we waiting on this codec DAI stream */ -+ if (codec_dai->pop_wait == 1) { -+ -+ codec_dai->pop_wait = 0; -+ snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, -+ SND_SOC_DAPM_STREAM_STOP); -+ -+ /* power down the codec power domain if no longer active */ -+ if (codec->active == 0) { -+ dbg("pop wq D3 %s %s\n", codec->name, -+ codec_dai->playback.stream_name); -+ if (codec->dapm_event) -+ codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ } -+ } -+ } -+ mutex_unlock(&pcm_mutex); -+} -+ -+/* -+ * Called by ALSA when a PCM substream is closed. Private data can be -+ * freed here. The cpu DAI, codec DAI, machine and platform are also -+ * shutdown. -+ */ -+static int soc_codec_close(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_machine *machine = socdev->machine; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ mutex_lock(&pcm_mutex); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ rtd->cpu_dai->playback.active = rtd->codec_dai->playback.active = 0; -+ else -+ rtd->cpu_dai->capture.active = rtd->codec_dai->capture.active = 0; -+ -+ if (rtd->codec_dai->playback.active == 0 && -+ rtd->codec_dai->capture.active == 0) { -+ rtd->cpu_dai->active = rtd->codec_dai->active = 0; -+ } -+ codec->active--; -+ -+ if (rtd->cpu_dai->ops.shutdown) -+ rtd->cpu_dai->ops.shutdown(substream); -+ -+ if (rtd->codec_dai->ops.shutdown) -+ rtd->codec_dai->ops.shutdown(substream); -+ -+ if (machine->ops && machine->ops->shutdown) -+ machine->ops->shutdown(substream); -+ -+ if (platform->pcm_ops->close) -+ platform->pcm_ops->close(substream); -+ rtd->cpu_dai->runtime = NULL; -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ /* start delayed pop wq here for playback streams */ -+ rtd->codec_dai->pop_wait = 1; -+ queue_delayed_work(soc_workq, &soc_stream_work, -+ msecs_to_jiffies(pmdown_time)); -+ } else { -+ /* capture streams can be powered down now */ -+ snd_soc_dapm_stream_event(codec, rtd->codec_dai->capture.stream_name, -+ SND_SOC_DAPM_STREAM_STOP); -+ -+ if (codec->active == 0 && rtd->codec_dai->pop_wait == 0){ -+ if (codec->dapm_event) -+ codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ } -+ } -+ -+ mutex_unlock(&pcm_mutex); -+ return 0; -+} -+ -+/* -+ * Called by ALSA when the PCM substream is prepared, can set format, sample -+ * rate, etc. This function is non atomic and can be called multiple times, -+ * it can refer to the runtime info. -+ */ -+static int soc_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_codec *codec = socdev->codec; -+ int ret = 0; -+ -+ mutex_lock(&pcm_mutex); -+ if (platform->pcm_ops->prepare) { -+ ret = platform->pcm_ops->prepare(substream); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: platform prepare error\n"); -+ goto out; -+ } -+ } -+ -+ if (rtd->codec_dai->ops.prepare) { -+ ret = rtd->codec_dai->ops.prepare(substream); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: codec DAI prepare error\n"); -+ goto out; -+ } -+ } -+ -+ if (rtd->cpu_dai->ops.prepare) -+ ret = rtd->cpu_dai->ops.prepare(substream); -+ -+ /* we only want to start a DAPM playback stream if we are not waiting -+ * on an existing one stopping */ -+ if (rtd->codec_dai->pop_wait) { -+ /* we are waiting for the delayed work to start */ -+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) -+ snd_soc_dapm_stream_event(codec, -+ rtd->codec_dai->capture.stream_name, -+ SND_SOC_DAPM_STREAM_START); -+ else { -+ rtd->codec_dai->pop_wait = 0; -+ cancel_delayed_work(&soc_stream_work); -+ if (rtd->codec_dai->digital_mute) -+ rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0); -+ } -+ } else { -+ /* no delayed work - do we need to power up codec */ -+ if (codec->dapm_state != SNDRV_CTL_POWER_D0) { -+ -+ if (codec->dapm_event) -+ codec->dapm_event(codec, SNDRV_CTL_POWER_D1); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ snd_soc_dapm_stream_event(codec, -+ rtd->codec_dai->playback.stream_name, -+ SND_SOC_DAPM_STREAM_START); -+ else -+ snd_soc_dapm_stream_event(codec, -+ rtd->codec_dai->capture.stream_name, -+ SND_SOC_DAPM_STREAM_START); -+ -+ if (codec->dapm_event) -+ codec->dapm_event(codec, SNDRV_CTL_POWER_D0); -+ if (rtd->codec_dai->digital_mute) -+ rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0); -+ -+ } else { -+ /* codec already powered - power on widgets */ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ snd_soc_dapm_stream_event(codec, -+ rtd->codec_dai->playback.stream_name, -+ SND_SOC_DAPM_STREAM_START); -+ else -+ snd_soc_dapm_stream_event(codec, -+ rtd->codec_dai->capture.stream_name, -+ SND_SOC_DAPM_STREAM_START); -+ if (rtd->codec_dai->digital_mute) -+ rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0); -+ } -+ } -+ -+out: -+ mutex_unlock(&pcm_mutex); -+ return ret; -+} -+ -+/* -+ * Called by ALSA when the hardware params are set by application. This -+ * function can also be called multiple times and can allocate buffers -+ * (using snd_pcm_lib_* ). It's non-atomic. -+ */ -+static int soc_pcm_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_machine *machine = socdev->machine; -+ int ret = 0; -+ -+ mutex_lock(&pcm_mutex); -+ -+ /* we don't need to match any AC97 params */ -+ if (rtd->cpu_dai->type != SND_SOC_DAI_AC97) { -+ ret = soc_hw_match_params(substream, params); -+ if (ret < 0) -+ goto out; -+ } else { -+ struct snd_soc_clock_info clk_info; -+ clk_info.rate = params_rate(params); -+ ret = soc_get_mclk(rtd, &clk_info); -+ if (ret < 0) -+ goto out; -+ } -+ -+ if (rtd->codec_dai->ops.hw_params) { -+ ret = rtd->codec_dai->ops.hw_params(substream, params); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: can't set codec %s hw params\n", -+ rtd->codec_dai->name); -+ goto out; -+ } -+ } -+ -+ if (rtd->cpu_dai->ops.hw_params) { -+ ret = rtd->cpu_dai->ops.hw_params(substream, params); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: can't set interface %s hw params\n", -+ rtd->cpu_dai->name); -+ goto interface_err; -+ } -+ } -+ -+ if (platform->pcm_ops->hw_params) { -+ ret = platform->pcm_ops->hw_params(substream, params); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: can't set platform %s hw params\n", -+ platform->name); -+ goto platform_err; -+ } -+ } -+ -+ if (machine->ops && machine->ops->hw_params) { -+ ret = machine->ops->hw_params(substream, params); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: machine hw_params failed\n"); -+ goto machine_err; -+ } -+ } -+ -+out: -+ mutex_unlock(&pcm_mutex); -+ return ret; -+ -+machine_err: -+ if (platform->pcm_ops->hw_free) -+ platform->pcm_ops->hw_free(substream); -+ -+platform_err: -+ if (rtd->cpu_dai->ops.hw_free) -+ rtd->cpu_dai->ops.hw_free(substream); -+ -+interface_err: -+ if (rtd->codec_dai->ops.hw_free) -+ rtd->codec_dai->ops.hw_free(substream); -+ -+ mutex_unlock(&pcm_mutex); -+ return ret; -+} -+ -+/* -+ * Free's resources allocated by hw_params, can be called multiple times -+ */ -+static int soc_pcm_hw_free(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_machine *machine = socdev->machine; -+ -+ mutex_lock(&pcm_mutex); -+ -+ /* apply codec digital mute */ -+ if (!codec->active && rtd->codec_dai->digital_mute) -+ rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 1); -+ -+ /* free any machine hw params */ -+ if (machine->ops && machine->ops->hw_free) -+ machine->ops->hw_free(substream); -+ -+ /* free any DMA resources */ -+ if (platform->pcm_ops->hw_free) -+ platform->pcm_ops->hw_free(substream); -+ -+ /* now free hw params for the DAI's */ -+ if (rtd->codec_dai->ops.hw_free) -+ rtd->codec_dai->ops.hw_free(substream); -+ -+ if (rtd->cpu_dai->ops.hw_free) -+ rtd->cpu_dai->ops.hw_free(substream); -+ -+ mutex_unlock(&pcm_mutex); -+ return 0; -+} -+ -+static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_platform *platform = socdev->platform; -+ int ret; -+ -+ if (rtd->codec_dai->ops.trigger) { -+ ret = rtd->codec_dai->ops.trigger(substream, cmd); -+ if (ret < 0) -+ return ret; -+ } -+ -+ if (platform->pcm_ops->trigger) { -+ ret = platform->pcm_ops->trigger(substream, cmd); -+ if (ret < 0) -+ return ret; -+ } -+ -+ if (rtd->cpu_dai->ops.trigger) { -+ ret = rtd->cpu_dai->ops.trigger(substream, cmd); -+ if (ret < 0) -+ return ret; -+ } -+ return 0; -+} -+ -+/* ASoC PCM operations */ -+static struct snd_pcm_ops soc_pcm_ops = { -+ .open = soc_pcm_open, -+ .close = soc_codec_close, -+ .hw_params = soc_pcm_hw_params, -+ .hw_free = soc_pcm_hw_free, -+ .prepare = soc_pcm_prepare, -+ .trigger = soc_pcm_trigger, -+}; -+ -+#ifdef CONFIG_PM -+/* powers down audio subsystem for suspend */ -+static int soc_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_machine *machine = socdev->machine; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev; -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ -+ /* mute any active DAC's */ -+ for(i = 0; i < machine->num_links; i++) { -+ struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; -+ if (dai->digital_mute && dai->playback.active) -+ dai->digital_mute(codec, dai, 1); -+ } -+ -+ if (machine->suspend_pre) -+ machine->suspend_pre(pdev, state); -+ -+ for(i = 0; i < machine->num_links; i++) { -+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; -+ if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) -+ cpu_dai->suspend(pdev, cpu_dai); -+ if (platform->suspend) -+ platform->suspend(pdev, cpu_dai); -+ } -+ -+ /* close any waiting streams and save state */ -+ flush_workqueue(soc_workq); -+ codec->suspend_dapm_state = codec->dapm_state; -+ -+ for(i = 0; i < codec->num_dai; i++) { -+ char *stream = codec->dai[i].playback.stream_name; -+ if (stream != NULL) -+ snd_soc_dapm_stream_event(codec, stream, -+ SND_SOC_DAPM_STREAM_SUSPEND); -+ stream = codec->dai[i].capture.stream_name; -+ if (stream != NULL) -+ snd_soc_dapm_stream_event(codec, stream, -+ SND_SOC_DAPM_STREAM_SUSPEND); -+ } -+ -+ if (codec_dev->suspend) -+ codec_dev->suspend(pdev, state); -+ -+ for(i = 0; i < machine->num_links; i++) { -+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; -+ if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) -+ cpu_dai->suspend(pdev, cpu_dai); -+ } -+ -+ if (machine->suspend_post) -+ machine->suspend_post(pdev, state); -+ -+ return 0; -+} -+ -+/* powers up audio subsystem after a suspend */ -+static int soc_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_machine *machine = socdev->machine; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev; -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ -+ if (machine->resume_pre) -+ machine->resume_pre(pdev); -+ -+ for(i = 0; i < machine->num_links; i++) { -+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; -+ if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) -+ cpu_dai->resume(pdev, cpu_dai); -+ } -+ -+ if (codec_dev->resume) -+ codec_dev->resume(pdev); -+ -+ for(i = 0; i < codec->num_dai; i++) { -+ char* stream = codec->dai[i].playback.stream_name; -+ if (stream != NULL) -+ snd_soc_dapm_stream_event(codec, stream, -+ SND_SOC_DAPM_STREAM_RESUME); -+ stream = codec->dai[i].capture.stream_name; -+ if (stream != NULL) -+ snd_soc_dapm_stream_event(codec, stream, -+ SND_SOC_DAPM_STREAM_RESUME); -+ } -+ -+ /* unmute any active DAC's */ -+ for(i = 0; i < machine->num_links; i++) { -+ struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; -+ if (dai->digital_mute && dai->playback.active) -+ dai->digital_mute(codec, dai, 0); -+ } -+ -+ for(i = 0; i < machine->num_links; i++) { -+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; -+ if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) -+ cpu_dai->resume(pdev, cpu_dai); -+ if (platform->resume) -+ platform->resume(pdev, cpu_dai); -+ } -+ -+ if (machine->resume_post) -+ machine->resume_post(pdev); -+ -+ return 0; -+} -+ -+#else -+#define soc_suspend NULL -+#define soc_resume NULL -+#endif -+ -+/* probes a new socdev */ -+static int soc_probe(struct platform_device *pdev) -+{ -+ int ret = 0, i; -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_machine *machine = socdev->machine; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev; -+ -+ if (machine->probe) { -+ ret = machine->probe(pdev); -+ if(ret < 0) -+ return ret; -+ } -+ -+ for (i = 0; i < machine->num_links; i++) { -+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; -+ if (cpu_dai->probe) { -+ ret = cpu_dai->probe(pdev); -+ if(ret < 0) -+ goto cpu_dai_err; -+ } -+ } -+ -+ if (codec_dev->probe) { -+ ret = codec_dev->probe(pdev); -+ if(ret < 0) -+ goto cpu_dai_err; -+ } -+ -+ if (platform->probe) { -+ ret = platform->probe(pdev); -+ if(ret < 0) -+ goto platform_err; -+ } -+ -+ /* DAPM stream work */ -+ soc_workq = create_workqueue("kdapm"); -+ if (soc_workq == NULL) -+ goto work_err; -+ INIT_WORK(&soc_stream_work, close_delayed_work, socdev); -+ return 0; -+ -+work_err: -+ if (platform->remove) -+ platform->remove(pdev); -+ -+platform_err: -+ if (codec_dev->remove) -+ codec_dev->remove(pdev); -+ -+cpu_dai_err: -+ for (i--; i > 0; i--) { -+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; -+ if (cpu_dai->remove) -+ cpu_dai->remove(pdev); -+ } -+ -+ if (machine->remove) -+ machine->remove(pdev); -+ -+ return ret; -+} -+ -+/* removes a socdev */ -+static int soc_remove(struct platform_device *pdev) -+{ -+ int i; -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_machine *machine = socdev->machine; -+ struct snd_soc_platform *platform = socdev->platform; -+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev; -+ -+ if (soc_workq) -+ destroy_workqueue(soc_workq); -+ -+ if (platform->remove) -+ platform->remove(pdev); -+ -+ if (codec_dev->remove) -+ codec_dev->remove(pdev); -+ -+ for (i = 0; i < machine->num_links; i++) { -+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; -+ if (cpu_dai->remove) -+ cpu_dai->remove(pdev); -+ } -+ -+ if (machine->remove) -+ machine->remove(pdev); -+ -+ return 0; -+} -+ -+/* ASoC platform driver */ -+static struct platform_driver soc_driver = { -+ .driver = { -+ .name = "soc-audio", -+ }, -+ .probe = soc_probe, -+ .remove = soc_remove, -+ .suspend = soc_suspend, -+ .resume = soc_resume, -+}; -+ -+/* create a new pcm */ -+static int soc_new_pcm(struct snd_soc_device *socdev, -+ struct snd_soc_dai_link *dai_link, int num) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai; -+ struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai; -+ struct snd_soc_pcm_runtime *rtd; -+ struct snd_pcm *pcm; -+ char new_name[64]; -+ int ret = 0, playback = 0, capture = 0; -+ -+ rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); -+ if (rtd == NULL) -+ return -ENOMEM; -+ rtd->cpu_dai = cpu_dai; -+ rtd->codec_dai = codec_dai; -+ rtd->socdev = socdev; -+ -+ /* check client and interface hw capabilities */ -+ sprintf(new_name, "%s %s-%s-%d",dai_link->stream_name, codec_dai->name, -+ get_dai_name(cpu_dai->type), num); -+ -+ if (codec_dai->playback.channels_min) -+ playback = 1; -+ if (codec_dai->capture.channels_min) -+ capture = 1; -+ -+ ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, -+ capture, &pcm); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); -+ kfree(rtd); -+ return ret; -+ } -+ -+ pcm->private_data = rtd; -+ soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap; -+ soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer; -+ soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl; -+ soc_pcm_ops.copy = socdev->platform->pcm_ops->copy; -+ soc_pcm_ops.silence = socdev->platform->pcm_ops->silence; -+ soc_pcm_ops.ack = socdev->platform->pcm_ops->ack; -+ soc_pcm_ops.page = socdev->platform->pcm_ops->page; -+ -+ if (playback) -+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); -+ -+ if (capture) -+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); -+ -+ ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: platform pcm constructor failed\n"); -+ kfree(rtd); -+ return ret; -+ } -+ -+ pcm->private_free = socdev->platform->pcm_free; -+ printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, -+ cpu_dai->name); -+ return ret; -+} -+ -+/* codec register dump */ -+static ssize_t codec_reg_show(struct device *dev, -+ struct device_attribute *attr, char *buf) -+{ -+ struct snd_soc_device *devdata = dev_get_drvdata(dev); -+ struct snd_soc_codec *codec = devdata->codec; -+ int i, step = 1, count = 0; -+ -+ if (!codec->reg_cache_size) -+ return 0; -+ -+ if (codec->reg_cache_step) -+ step = codec->reg_cache_step; -+ -+ count += sprintf(buf, "%s registers\n", codec->name); -+ for(i = 0; i < codec->reg_cache_size; i += step) -+ count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i)); -+ -+ return count; -+} -+static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); -+ -+/** -+ * snd_soc_new_ac97_codec - initailise AC97 device -+ * @codec: audio codec -+ * @ops: AC97 bus operations -+ * @num: AC97 codec number -+ * -+ * Initialises AC97 codec resources for use by ad-hoc devices only. -+ */ -+int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, -+ struct snd_ac97_bus_ops *ops, int num) -+{ -+ mutex_lock(&codec->mutex); -+ -+ codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); -+ if (codec->ac97 == NULL) { -+ mutex_unlock(&codec->mutex); -+ return -ENOMEM; -+ } -+ -+ codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); -+ if (codec->ac97->bus == NULL) { -+ kfree(codec->ac97); -+ codec->ac97 = NULL; -+ mutex_unlock(&codec->mutex); -+ return -ENOMEM; -+ } -+ -+ codec->ac97->bus->ops = ops; -+ codec->ac97->num = num; -+ mutex_unlock(&codec->mutex); -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); -+ -+/** -+ * snd_soc_free_ac97_codec - free AC97 codec device -+ * @codec: audio codec -+ * -+ * Frees AC97 codec device resources. -+ */ -+void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) -+{ -+ mutex_lock(&codec->mutex); -+ kfree(codec->ac97->bus); -+ kfree(codec->ac97); -+ codec->ac97 = NULL; -+ mutex_unlock(&codec->mutex); -+} -+EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); -+ -+/** -+ * snd_soc_update_bits - update codec register bits -+ * @codec: audio codec -+ * @reg: codec register -+ * @mask: register mask -+ * @value: new value -+ * -+ * Writes new register value. -+ * -+ * Returns 1 for change else 0. -+ */ -+int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, -+ unsigned short mask, unsigned short value) -+{ -+ int change; -+ unsigned short old, new; -+ -+ mutex_lock(&io_mutex); -+ old = snd_soc_read(codec, reg); -+ new = (old & ~mask) | value; -+ change = old != new; -+ if (change) -+ snd_soc_write(codec, reg, new); -+ -+ mutex_unlock(&io_mutex); -+ return change; -+} -+EXPORT_SYMBOL_GPL(snd_soc_update_bits); -+ -+/** -+ * snd_soc_test_bits - test register for change -+ * @codec: audio codec -+ * @reg: codec register -+ * @mask: register mask -+ * @value: new value -+ * -+ * Tests a register with a new value and checks if the new value is -+ * different from the old value. -+ * -+ * Returns 1 for change else 0. -+ */ -+int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, -+ unsigned short mask, unsigned short value) -+{ -+ int change; -+ unsigned short old, new; -+ -+ mutex_lock(&io_mutex); -+ old = snd_soc_read(codec, reg); -+ new = (old & ~mask) | value; -+ change = old != new; -+ mutex_unlock(&io_mutex); -+ -+ return change; -+} -+EXPORT_SYMBOL_GPL(snd_soc_test_bits); -+ -+/** -+ * snd_soc_get_rate - get int sample rate -+ * @hwpcmrate: the hardware pcm rate -+ * -+ * Returns the audio rate integaer value, else 0. -+ */ -+int snd_soc_get_rate(int hwpcmrate) -+{ -+ int rate = ffs(hwpcmrate) - 1; -+ -+ if (rate > ARRAY_SIZE(rates)) -+ return 0; -+ return rates[rate]; -+} -+EXPORT_SYMBOL_GPL(snd_soc_get_rate); -+ -+/** -+ * snd_soc_new_pcms - create new sound card and pcms -+ * @socdev: the SoC audio device -+ * -+ * Create a new sound card based upon the codec and interface pcms. -+ * -+ * Returns 0 for success, else error. -+ */ -+int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char * xid) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_machine *machine = socdev->machine; -+ int ret = 0, i; -+ -+ mutex_lock(&codec->mutex); -+ -+ /* register a sound card */ -+ codec->card = snd_card_new(idx, xid, codec->owner, 0); -+ if (!codec->card) { -+ printk(KERN_ERR "asoc: can't create sound card for codec %s\n", -+ codec->name); -+ mutex_unlock(&codec->mutex); -+ return -ENODEV; -+ } -+ -+ codec->card->dev = socdev->dev; -+ codec->card->private_data = codec; -+ strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); -+ -+ /* create the pcms */ -+ for(i = 0; i < machine->num_links; i++) { -+ ret = soc_new_pcm(socdev, &machine->dai_link[i], i); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: can't create pcm %s\n", -+ machine->dai_link[i].stream_name); -+ mutex_unlock(&codec->mutex); -+ return ret; -+ } -+ } -+ -+ mutex_unlock(&codec->mutex); -+ return ret; -+} -+EXPORT_SYMBOL_GPL(snd_soc_new_pcms); -+ -+/** -+ * snd_soc_register_card - register sound card -+ * @socdev: the SoC audio device -+ * -+ * Register a SoC sound card. Also registers an AC97 device if the -+ * codec is AC97 for ad hoc devices. -+ * -+ * Returns 0 for success, else error. -+ */ -+int snd_soc_register_card(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_machine *machine = socdev->machine; -+ int ret = 0, i, ac97 = 0, err = 0; -+ -+ mutex_lock(&codec->mutex); -+ for(i = 0; i < machine->num_links; i++) { -+ if (socdev->machine->dai_link[i].init) { -+ err = socdev->machine->dai_link[i].init(codec); -+ if (err < 0) { -+ printk(KERN_ERR "asoc: failed to init %s\n", -+ socdev->machine->dai_link[i].stream_name); -+ continue; -+ } -+ } -+ if (socdev->machine->dai_link[i].cpu_dai->type == SND_SOC_DAI_AC97) -+ ac97 = 1; -+ } -+ snprintf(codec->card->shortname, sizeof(codec->card->shortname), -+ "%s", machine->name); -+ snprintf(codec->card->longname, sizeof(codec->card->longname), -+ "%s (%s)", machine->name, codec->name); -+ -+ ret = snd_card_register(codec->card); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n", -+ codec->name); -+ goto out; -+ } -+ -+#ifdef CONFIG_SND_SOC_AC97_BUS -+ if (ac97) { -+ ret = soc_ac97_dev_register(codec); -+ if (ret < 0) { -+ printk(KERN_ERR "asoc: AC97 device register failed\n"); -+ snd_card_free(codec->card); -+ goto out; -+ } -+ } -+#endif -+ -+ err = snd_soc_dapm_sys_add(socdev->dev); -+ if (err < 0) -+ printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); -+ -+ err = device_create_file(socdev->dev, &dev_attr_codec_reg); -+ if (err < 0) -+ printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n"); -+out: -+ mutex_unlock(&codec->mutex); -+ return ret; -+} -+EXPORT_SYMBOL_GPL(snd_soc_register_card); -+ -+/** -+ * snd_soc_free_pcms - free sound card and pcms -+ * @socdev: the SoC audio device -+ * -+ * Frees sound card and pcms associated with the socdev. -+ * Also unregister the codec if it is an AC97 device. -+ */ -+void snd_soc_free_pcms(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ mutex_lock(&codec->mutex); -+#ifdef CONFIG_SND_SOC_AC97_BUS -+ if (codec->ac97) -+ soc_ac97_dev_unregister(codec); -+#endif -+ -+ if (codec->card) -+ snd_card_free(codec->card); -+ device_remove_file(socdev->dev, &dev_attr_codec_reg); -+ mutex_unlock(&codec->mutex); -+} -+EXPORT_SYMBOL_GPL(snd_soc_free_pcms); -+ -+/** -+ * snd_soc_set_runtime_hwparams - set the runtime hardware parameters -+ * @substream: the pcm substream -+ * @hw: the hardware parameters -+ * -+ * Sets the substream runtime hardware parameters. -+ */ -+int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, -+ const struct snd_pcm_hardware *hw) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ runtime->hw.info = hw->info; -+ runtime->hw.formats = hw->formats; -+ runtime->hw.period_bytes_min = hw->period_bytes_min; -+ runtime->hw.period_bytes_max = hw->period_bytes_max; -+ runtime->hw.periods_min = hw->periods_min; -+ runtime->hw.periods_max = hw->periods_max; -+ runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; -+ runtime->hw.fifo_size = hw->fifo_size; -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); -+ -+/** -+ * snd_soc_cnew - create new control -+ * @_template: control template -+ * @data: control private data -+ * @lnng_name: control long name -+ * -+ * Create a new mixer control from a template control. -+ * -+ * Returns 0 for success, else error. -+ */ -+struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, -+ void *data, char *long_name) -+{ -+ struct snd_kcontrol_new template; -+ -+ memcpy(&template, _template, sizeof(template)); -+ if (long_name) -+ template.name = long_name; -+ template.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; -+ template.index = 0; -+ -+ return snd_ctl_new1(&template, data); -+} -+EXPORT_SYMBOL_GPL(snd_soc_cnew); -+ -+/** -+ * snd_soc_info_enum_double - enumerated double mixer info callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to provide information about a double enumerated -+ * mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo) -+{ -+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; -+ -+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; -+ uinfo->count = e->shift_l == e->shift_r ? 1 : 2; -+ uinfo->value.enumerated.items = e->mask; -+ -+ if (uinfo->value.enumerated.item > e->mask - 1) -+ uinfo->value.enumerated.item = e->mask - 1; -+ strcpy(uinfo->value.enumerated.name, -+ e->texts[uinfo->value.enumerated.item]); -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); -+ -+/** -+ * snd_soc_get_enum_double - enumerated double mixer get callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to get the value of a double enumerated mixer. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; -+ unsigned short val, bitmask; -+ -+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) -+ ; -+ val = snd_soc_read(codec, e->reg); -+ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); -+ if (e->shift_l != e->shift_r) -+ ucontrol->value.enumerated.item[1] = -+ (val >> e->shift_r) & (bitmask - 1); -+ -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); -+ -+/** -+ * snd_soc_put_enum_double - enumerated double mixer put callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to set the value of a double enumerated mixer. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; -+ unsigned short val; -+ unsigned short mask, bitmask; -+ -+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) -+ ; -+ if (ucontrol->value.enumerated.item[0] > e->mask - 1) -+ return -EINVAL; -+ val = ucontrol->value.enumerated.item[0] << e->shift_l; -+ mask = (bitmask - 1) << e->shift_l; -+ if (e->shift_l != e->shift_r) { -+ if (ucontrol->value.enumerated.item[1] > e->mask - 1) -+ return -EINVAL; -+ val |= ucontrol->value.enumerated.item[1] << e->shift_r; -+ mask |= (bitmask - 1) << e->shift_r; -+ } -+ -+ return snd_soc_update_bits(codec, e->reg, mask, val); -+} -+EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); -+ -+/** -+ * snd_soc_info_enum_ext - external enumerated single mixer info callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to provide information about an external enumerated -+ * single mixer. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo) -+{ -+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; -+ -+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; -+ uinfo->count = 1; -+ uinfo->value.enumerated.items = e->mask; -+ -+ if (uinfo->value.enumerated.item > e->mask - 1) -+ uinfo->value.enumerated.item = e->mask - 1; -+ strcpy(uinfo->value.enumerated.name, -+ e->texts[uinfo->value.enumerated.item]); -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); -+ -+/** -+ * snd_soc_info_volsw_ext - external single mixer info callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to provide information about a single external mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo) -+{ -+ int mask = kcontrol->private_value; -+ -+ uinfo->type = -+ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; -+ uinfo->count = 1; -+ uinfo->value.integer.min = 0; -+ uinfo->value.integer.max = mask; -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); -+ -+/** -+ * snd_soc_info_bool_ext - external single boolean mixer info callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to provide information about a single boolean external mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo) -+{ -+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; -+ uinfo->count = 1; -+ uinfo->value.integer.min = 0; -+ uinfo->value.integer.max = 1; -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_info_bool_ext); -+ -+/** -+ * snd_soc_info_volsw - single mixer info callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to provide information about a single mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo) -+{ -+ int mask = (kcontrol->private_value >> 16) & 0xff; -+ int shift = (kcontrol->private_value >> 8) & 0x0f; -+ int rshift = (kcontrol->private_value >> 12) & 0x0f; -+ -+ uinfo->type = -+ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; -+ uinfo->count = shift == rshift ? 1 : 2; -+ uinfo->value.integer.min = 0; -+ uinfo->value.integer.max = mask; -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_info_volsw); -+ -+/** -+ * snd_soc_get_volsw - single mixer get callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to get the value of a single mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ int reg = kcontrol->private_value & 0xff; -+ int shift = (kcontrol->private_value >> 8) & 0x0f; -+ int rshift = (kcontrol->private_value >> 12) & 0x0f; -+ int mask = (kcontrol->private_value >> 16) & 0xff; -+ int invert = (kcontrol->private_value >> 24) & 0x01; -+ -+ ucontrol->value.integer.value[0] = -+ (snd_soc_read(codec, reg) >> shift) & mask; -+ if (shift != rshift) -+ ucontrol->value.integer.value[1] = -+ (snd_soc_read(codec, reg) >> rshift) & mask; -+ if (invert) { -+ ucontrol->value.integer.value[0] = -+ mask - ucontrol->value.integer.value[0]; -+ if (shift != rshift) -+ ucontrol->value.integer.value[1] = -+ mask - ucontrol->value.integer.value[1]; -+ } -+ -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_get_volsw); -+ -+/** -+ * snd_soc_put_volsw - single mixer put callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to set the value of a single mixer control. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ int reg = kcontrol->private_value & 0xff; -+ int shift = (kcontrol->private_value >> 8) & 0x0f; -+ int rshift = (kcontrol->private_value >> 12) & 0x0f; -+ int mask = (kcontrol->private_value >> 16) & 0xff; -+ int invert = (kcontrol->private_value >> 24) & 0x01; -+ int err; -+ unsigned short val, val2, val_mask; -+ -+ val = (ucontrol->value.integer.value[0] & mask); -+ if (invert) -+ val = mask - val; -+ val_mask = mask << shift; -+ val = val << shift; -+ if (shift != rshift) { -+ val2 = (ucontrol->value.integer.value[1] & mask); -+ if (invert) -+ val2 = mask - val2; -+ val_mask |= mask << rshift; -+ val |= val2 << rshift; -+ } -+ err = snd_soc_update_bits(codec, reg, val_mask, val); -+ return err; -+} -+EXPORT_SYMBOL_GPL(snd_soc_put_volsw); -+ -+/** -+ * snd_soc_info_volsw_2r - double mixer info callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to provide information about a double mixer control that -+ * spans 2 codec registers. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_info *uinfo) -+{ -+ int mask = (kcontrol->private_value >> 12) & 0xff; -+ -+ uinfo->type = -+ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; -+ uinfo->count = 2; -+ uinfo->value.integer.min = 0; -+ uinfo->value.integer.max = mask; -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); -+ -+/** -+ * snd_soc_get_volsw_2r - double mixer get callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to get the value of a double mixer control that spans 2 registers. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ int reg = kcontrol->private_value & 0xff; -+ int reg2 = (kcontrol->private_value >> 24) & 0xff; -+ int shift = (kcontrol->private_value >> 8) & 0x0f; -+ int mask = (kcontrol->private_value >> 12) & 0xff; -+ int invert = (kcontrol->private_value >> 20) & 0x01; -+ -+ ucontrol->value.integer.value[0] = -+ (snd_soc_read(codec, reg) >> shift) & mask; -+ ucontrol->value.integer.value[1] = -+ (snd_soc_read(codec, reg2) >> shift) & mask; -+ if (invert) { -+ ucontrol->value.integer.value[0] = -+ mask - ucontrol->value.integer.value[0]; -+ ucontrol->value.integer.value[1] = -+ mask - ucontrol->value.integer.value[1]; -+ } -+ -+ return 0; -+} -+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); -+ -+/** -+ * snd_soc_put_volsw_2r - double mixer set callback -+ * @kcontrol: mixer control -+ * @uinfo: control element information -+ * -+ * Callback to set the value of a double mixer control that spans 2 registers. -+ * -+ * Returns 0 for success. -+ */ -+int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, -+ struct snd_ctl_elem_value *ucontrol) -+{ -+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); -+ int reg = kcontrol->private_value & 0xff; -+ int reg2 = (kcontrol->private_value >> 24) & 0xff; -+ int shift = (kcontrol->private_value >> 8) & 0x0f; -+ int mask = (kcontrol->private_value >> 12) & 0xff; -+ int invert = (kcontrol->private_value >> 20) & 0x01; -+ int err; -+ unsigned short val, val2, val_mask; -+ -+ val_mask = mask << shift; -+ val = (ucontrol->value.integer.value[0] & mask); -+ val2 = (ucontrol->value.integer.value[1] & mask); -+ -+ if (invert) { -+ val = mask - val; -+ val2 = mask - val2; -+ } -+ -+ val = val << shift; -+ val2 = val2 << shift; -+ -+ if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0) -+ return err; -+ -+ err = snd_soc_update_bits(codec, reg2, val_mask, val2); -+ return err; -+} -+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); -+ -+static int __devinit snd_soc_init(void) -+{ -+ printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); -+ return platform_driver_register(&soc_driver); -+} -+ -+static void snd_soc_exit(void) -+{ -+ platform_driver_unregister(&soc_driver); -+} -+ -+module_init(snd_soc_init); -+module_exit(snd_soc_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("ALSA SoC Core"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/at91/Kconfig -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/at91/Kconfig -@@ -0,0 +1,24 @@ -+menu "SoC Audio for the Atmel AT91" -+ -+config SND_AT91_SOC -+ tristate "SoC Audio for the Atmel AT91 System-on-Chip" -+ depends on ARCH_AT91 && SND -+ select SND_PCM -+ help -+ Say Y or M if you want to add support for codecs attached to -+ the AT91 SSC interface. You will also need -+ to select the audio interfaces to support below. -+ -+config SND_AT91_SOC_I2S -+ tristate -+ -+config SND_AT91_SOC_ETI_B1_WM8731 -+ tristate "SoC I2S Audio support for Endrelia ETI-B1 board" -+ depends on SND_AT91_SOC && MACH_ETI_B1 -+ select SND_AT91_SOC_I2S -+ select SND_SOC_WM8731 -+ help -+ Say Y if you want to add support for SoC audio on Endrelia -+ ETI-B1 board. -+ -+endmenu -Index: linux-2.6-pxa-new/sound/soc/at91/Makefile -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/at91/Makefile -@@ -0,0 +1,11 @@ -+# AT91 Platform Support -+snd-soc-at91-objs := at91rm9200-pcm.o -+snd-soc-at91-i2s-objs := at91rm9200-i2s.o -+ -+obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o -+obj-$(CONFIG_SND_AT91_SOC_I2S) += snd-soc-at91-i2s.o -+ -+# AT91 Machine Support -+snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o -+ -+obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o -Index: linux-2.6-pxa-new/sound/soc/at91/at91rm9200-i2s.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/at91/at91rm9200-i2s.c -@@ -0,0 +1,715 @@ -+/* -+ * at91rm9200-i2s.c -- ALSA Soc Audio Layer Platform driver and DMA engine -+ * -+ * Author: Frank Mandarino <fmandarino@endrelia.com> -+ * Endrelia Technologies Inc. -+ * -+ * Based on pxa2xx Platform drivers by -+ * Liam Girdwood <liam.girdwood@wolfsonmicro.com> -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 3rd Mar 2006 Initial version. -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/interrupt.h> -+#include <linux/device.h> -+#include <linux/delay.h> -+#include <linux/clk.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/initval.h> -+#include <sound/soc.h> -+ -+#include <asm/arch/at91rm9200.h> -+#include <asm/arch/at91rm9200_ssc.h> -+#include <asm/arch/at91rm9200_pdc.h> -+#include <asm/arch/hardware.h> -+ -+#include "at91rm9200-pcm.h" -+ -+#if 0 -+#define DBG(x...) printk(KERN_DEBUG "at91rm9200-i2s:" x) -+#else -+#define DBG(x...) -+#endif -+ -+#define AT91RM9200_I2S_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF) -+ -+#define AT91RM9200_I2S_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+/* priv is (SSC_CMR.DIV << 16 | SSC_TCMR.PERIOD ) */ -+static struct snd_soc_dai_mode at91rm9200_i2s[] = { -+ -+ /* 8k: BCLK = (MCLK/10) = (60MHz/50) = 1.2MHz */ -+ { -+ .fmt = AT91RM9200_I2S_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = AT91RM9200_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1500, -+ .bfs = SND_SOC_FSBD(10), -+ .priv = (25 << 16 | 74), -+ }, -+ -+ /* 16k: BCLK = (MCLK/3) ~= (60MHz/14) = 4.285714MHz */ -+ { -+ .fmt = AT91RM9200_I2S_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_16000, -+ .pcmdir = AT91RM9200_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 750, -+ .bfs = SND_SOC_FSBD(3), -+ .priv = (7 << 16 | 133), -+ }, -+ -+ /* 32k: BCLK = (MCLK/3) ~= (60MHz/14) = 4.285714MHz */ -+ { -+ .fmt = AT91RM9200_I2S_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = AT91RM9200_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 375, -+ .bfs = SND_SOC_FSBD(3), -+ .priv = (7 << 16 | 66), -+ }, -+ -+ /* 48k: BCLK = (MCLK/5) ~= (60MHz/26) = 2.3076923MHz */ -+ { -+ .fmt = AT91RM9200_I2S_DAIFMT, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = AT91RM9200_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 250, -+ .bfs SND_SOC_FSBD(5), -+ .priv = (13 << 16 | 23), -+ }, -+}; -+ -+ -+/* -+ * SSC registers required by the PCM DMA engine. -+ */ -+static struct at91rm9200_ssc_regs ssc_reg[3] = { -+ { -+ .cr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_SSC_CR), -+ .ier = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_SSC_IER), -+ .idr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_SSC_IDR), -+ }, -+ { -+ .cr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_SSC_CR), -+ .ier = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_SSC_IER), -+ .idr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_SSC_IDR), -+ }, -+ { -+ .cr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_SSC_CR), -+ .ier = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_SSC_IER), -+ .idr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_SSC_IDR), -+ }, -+}; -+ -+static struct at91rm9200_pdc_regs pdc_tx_reg[3] = { -+ { -+ .xpr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_TPR), -+ .xcr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_TCR), -+ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_TNPR), -+ .xncr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_TNCR), -+ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_PTCR), -+ }, -+ { -+ .xpr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_TPR), -+ .xcr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_TCR), -+ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_TNPR), -+ .xncr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_TNCR), -+ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_PTCR), -+ }, -+ { -+ .xpr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_TPR), -+ .xcr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_TCR), -+ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_TNPR), -+ .xncr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_TNCR), -+ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_PTCR), -+ }, -+}; -+ -+static struct at91rm9200_pdc_regs pdc_rx_reg[3] = { -+ { -+ .xpr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_RPR), -+ .xcr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_RCR), -+ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_RNPR), -+ .xncr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_RNCR), -+ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_PTCR), -+ }, -+ { -+ .xpr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_RPR), -+ .xcr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_RCR), -+ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_RNPR), -+ .xncr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_RNCR), -+ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_PTCR), -+ }, -+ { -+ .xpr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_RPR), -+ .xcr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_RCR), -+ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_RNPR), -+ .xncr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_RNCR), -+ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_PTCR), -+ }, -+}; -+ -+/* -+ * SSC & PDC status bits for transmit and receive. -+ */ -+static struct at91rm9200_ssc_mask ssc_tx_mask = { -+ .ssc_enable = AT91_SSC_TXEN, -+ .ssc_disable = AT91_SSC_TXDIS, -+ .ssc_endx = AT91_SSC_ENDTX, -+ .ssc_endbuf = AT91_SSC_TXBUFE, -+ .pdc_enable = AT91_PDC_TXTEN, -+ .pdc_disable = AT91_PDC_TXTDIS, -+}; -+ -+static struct at91rm9200_ssc_mask ssc_rx_mask = { -+ .ssc_enable = AT91_SSC_RXEN, -+ .ssc_disable = AT91_SSC_RXDIS, -+ .ssc_endx = AT91_SSC_ENDRX, -+ .ssc_endbuf = AT91_SSC_RXBUFF, -+ .pdc_enable = AT91_PDC_RXTEN, -+ .pdc_disable = AT91_PDC_RXTDIS, -+}; -+ -+/* -+ * A MUTEX is used to protect an SSC initialzed flag which allows -+ * the substream hw_params() call to initialize the SSC only if -+ * there are no other substreams open. If there are other -+ * substreams open, the hw_param() call can only check that -+ * it is using the same format and rate. -+ */ -+static DECLARE_MUTEX(ssc0_mutex); -+static DECLARE_MUTEX(ssc1_mutex); -+static DECLARE_MUTEX(ssc2_mutex); -+ -+/* -+ * DMA parameters. -+ */ -+static at91rm9200_pcm_dma_params_t ssc_dma_params[3][2] = { -+ {{ -+ .name = "SSC0/I2S PCM Stereo out", -+ .ssc = &ssc_reg[0], -+ .pdc = &pdc_tx_reg[0], -+ .mask = &ssc_tx_mask, -+ }, -+ { -+ .name = "SSC0/I2S PCM Stereo in", -+ .ssc = &ssc_reg[0], -+ .pdc = &pdc_rx_reg[0], -+ .mask = &ssc_rx_mask, -+ }}, -+ {{ -+ .name = "SSC1/I2S PCM Stereo out", -+ .ssc = &ssc_reg[1], -+ .pdc = &pdc_tx_reg[1], -+ .mask = &ssc_tx_mask, -+ }, -+ { -+ .name = "SSC1/I2S PCM Stereo in", -+ .ssc = &ssc_reg[1], -+ .pdc = &pdc_rx_reg[1], -+ .mask = &ssc_rx_mask, -+ }}, -+ {{ -+ .name = "SSC2/I2S PCM Stereo out", -+ .ssc = &ssc_reg[2], -+ .pdc = &pdc_tx_reg[2], -+ .mask = &ssc_tx_mask, -+ }, -+ { -+ .name = "SSC1/I2S PCM Stereo in", -+ .ssc = &ssc_reg[2], -+ .pdc = &pdc_rx_reg[2], -+ .mask = &ssc_rx_mask, -+ }}, -+}; -+ -+ -+struct at91rm9200_ssc_state { -+ u32 ssc_cmr; -+ u32 ssc_rcmr; -+ u32 ssc_rfmr; -+ u32 ssc_tcmr; -+ u32 ssc_tfmr; -+ u32 ssc_sr; -+ u32 ssc_imr; -+}; -+ -+static struct at91rm9200_ssc_info { -+ char *name; -+ void __iomem *ssc_base; -+ u32 pid; -+ spinlock_t lock; /* lock for dir_mask */ -+ int dir_mask; /* 0=unused, 1=playback, 2=capture */ -+ struct semaphore *mutex; -+ int initialized; -+ int pcmfmt; -+ int rate; -+ at91rm9200_pcm_dma_params_t *dma_params[2]; -+ struct at91rm9200_ssc_state ssc_state; -+ -+} ssc_info[3] = { -+ { -+ .name = "ssc0", -+ .ssc_base = (void __iomem *) AT91_VA_BASE_SSC0, -+ .pid = AT91_ID_SSC0, -+ .lock = SPIN_LOCK_UNLOCKED, -+ .dir_mask = 0, -+ .mutex = &ssc0_mutex, -+ .initialized = 0, -+ }, -+ { -+ .name = "ssc1", -+ .ssc_base = (void __iomem *) AT91_VA_BASE_SSC1, -+ .pid = AT91_ID_SSC1, -+ .lock = SPIN_LOCK_UNLOCKED, -+ .dir_mask = 0, -+ .mutex = &ssc1_mutex, -+ .initialized = 0, -+ }, -+ { -+ .name = "ssc2", -+ .ssc_base = (void __iomem *) AT91_VA_BASE_SSC2, -+ .pid = AT91_ID_SSC2, -+ .lock = SPIN_LOCK_UNLOCKED, -+ .dir_mask = 0, -+ .mutex = &ssc2_mutex, -+ .initialized = 0, -+ }, -+}; -+ -+ -+static irqreturn_t at91rm9200_i2s_interrupt(int irq, void *dev_id) -+{ -+ struct at91rm9200_ssc_info *ssc_p = dev_id; -+ at91rm9200_pcm_dma_params_t *dma_params; -+ u32 ssc_sr; -+ int i; -+ -+ ssc_sr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_SR) -+ & at91_ssc_read(ssc_p->ssc_base + AT91_SSC_IMR); -+ -+ /* -+ * Loop through the substreams attached to this SSC. If -+ * a DMA-related interrupt occurred on that substream, call -+ * the DMA interrupt handler function, if one has been -+ * registered in the dma_params structure by the PCM driver. -+ */ -+ for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { -+ dma_params = ssc_p->dma_params[i]; -+ -+ if (dma_params != NULL && dma_params->dma_intr_handler != NULL && -+ (ssc_sr & -+ (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) -+ -+ dma_params->dma_intr_handler(ssc_sr, dma_params->substream); -+ } -+ -+ return IRQ_HANDLED; -+} -+ -+static int at91rm9200_i2s_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct at91rm9200_ssc_info *ssc_p = &ssc_info[rtd->cpu_dai->id]; -+ int dir_mask; -+ -+ DBG("i2s_startup: SSC_SR=0x%08lx\n", -+ at91_ssc_read(ssc_p->ssc_base + AT91_SSC_SR)); -+ dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; -+ -+ spin_lock_irq(&ssc_p->lock); -+ if (ssc_p->dir_mask & dir_mask) { -+ spin_unlock_irq(&ssc_p->lock); -+ return -EBUSY; -+ } -+ ssc_p->dir_mask |= dir_mask; -+ spin_unlock_irq(&ssc_p->lock); -+ -+ return 0; -+} -+ -+static void at91rm9200_i2s_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct at91rm9200_ssc_info *ssc_p = &ssc_info[rtd->cpu_dai->id]; -+ at91rm9200_pcm_dma_params_t *dma_params = rtd->cpu_dai->dma_data; -+ int dir, dir_mask; -+ -+ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; -+ -+ if (dma_params != NULL) { -+ at91_ssc_write(dma_params->ssc->cr, dma_params->mask->ssc_disable); -+ DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), -+ at91_ssc_read(ssc_p->ssc_base + AT91_SSC_SR)); -+ -+ dma_params->substream = NULL; -+ ssc_p->dma_params[dir] = NULL; -+ } -+ -+ dir_mask = 1 << dir; -+ -+ spin_lock_irq(&ssc_p->lock); -+ ssc_p->dir_mask &= ~dir_mask; -+ if (!ssc_p->dir_mask) { -+ /* Shutdown the SSC clock. */ -+ DBG("Stopping pid %d clock\n", ssc_p->pid); -+ at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->pid); -+ -+ if (ssc_p->initialized) -+ free_irq(ssc_p->pid, ssc_p); -+ -+ /* Reset the SSC */ -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CR, AT91_SSC_SWRST); -+ -+ /* Force a re-init on the next hw_params() call. */ -+ ssc_p->initialized = 0; -+ } -+ spin_unlock_irq(&ssc_p->lock); -+} -+ -+#ifdef CONFIG_PM -+static int at91rm9200_i2s_suspend(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ struct at91rm9200_ssc_info *ssc_p; -+ -+ if(!dai->active) -+ return 0; -+ -+ ssc_p = &ssc_info[dai->id]; -+ -+ /* Save the status register before disabling transmit and receive. */ -+ ssc_p->state->ssc_sr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_SR); -+ at91_ssc_write(ssc_p->ssc_base + -+ AT91_SSC_CR, AT91_SSC_TXDIS | AT91_SSC_RXDIS); -+ -+ /* Save the current interrupt mask, then disable unmasked interrupts. */ -+ ssc_p->state->ssc_imr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_IMR); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_IDR, ssc_p->state->ssc_imr); -+ -+ ssc_p->state->ssc_cmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_CMR); -+ ssc_p->state->ssc_rcmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_RCMR); -+ ssc_p->state->ssc_rfmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_RCMR); -+ ssc_p->state->ssc_tcmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_RCMR); -+ ssc_p->state->ssc_tfmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_RCMR); -+ -+ return 0; -+} -+ -+static int at91rm9200_i2s_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ struct at91rm9200_ssc_info *ssc_p; -+ u32 cr_mask; -+ -+ if(!dai->active) -+ return 0; -+ -+ ssc_p = &ssc_info[dai->id]; -+ -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, ssc_p->state->ssc_tfmr); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, ssc_p->state->ssc_tcmr); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, ssc_p->state->ssc_rfmr); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, ssc_p->state->ssc_rcmr); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CMR, ssc_p->state->ssc_cmr); -+ -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_IER, ssc_p->state->ssc_imr); -+ -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CR, -+ ((ssc_p->state->ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) | -+ ((ssc_p->state->ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0)); -+ -+ return 0; -+} -+ -+#else -+#define at91rm9200_i2s_suspend NULL -+#define at91rm9200_i2s_resume NULL -+#endif -+ -+static unsigned int at91rm9200_i2s_config_sysclk( -+ struct snd_soc_cpu_dai *iface, struct snd_soc_clock_info *info, -+ unsigned int clk) -+{ -+ /* Currently, there is only support for USB (12Mhz) mode */ -+ if (clk != 12000000) -+ return 0; -+ return 12000000; -+} -+ -+static int at91rm9200_i2s_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ int id = rtd->cpu_dai->id; -+ struct at91rm9200_ssc_info *ssc_p = &ssc_info[id]; -+ at91rm9200_pcm_dma_params_t *dma_params; -+ unsigned int pcmfmt, rate; -+ int dir, channels, bits; -+ struct clk *mck_clk; -+ unsigned long bclk; -+ u32 div, period, tfmr, rfmr, tcmr, rcmr; -+ int ret; -+ -+ /* -+ * Currently, there is only one set of dma params for -+ * each direction. If more are added, this code will -+ * have to be changed to select the proper set. -+ */ -+ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; -+ -+ dma_params = &ssc_dma_params[id][dir]; -+ dma_params->substream = substream; -+ -+ ssc_p->dma_params[dir] = dma_params; -+ rtd->cpu_dai->dma_data = dma_params; -+ -+ rate = params_rate(params); -+ channels = params_channels(params); -+ -+ pcmfmt = rtd->cpu_dai->dai_runtime.pcmfmt; -+ switch (pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ /* likely this is all we'll ever support, but ... */ -+ bits = 16; -+ dma_params->pdc_xfer_size = 2; -+ break; -+ default: -+ printk(KERN_WARNING "at91rm9200-i2s: unsupported format %x\n", -+ pcmfmt); -+ return -EINVAL; -+ } -+ -+ /* Don't allow both SSC substreams to initialize at the same time. */ -+ down(ssc_p->mutex); -+ -+ /* -+ * If this SSC is alreadly initialized, then this substream must use -+ * the same format and rate. -+ */ -+ if (ssc_p->initialized) { -+ if (pcmfmt != ssc_p->pcmfmt || rate != ssc_p->rate) { -+ printk(KERN_WARNING "at91rm9200-i2s: " -+ "incompatible substream in other direction\n"); -+ up(ssc_p->mutex); -+ return -EINVAL; -+ } -+ } else { -+ /* Enable PMC peripheral clock for this SSC */ -+ DBG("Starting pid %d clock\n", ssc_p->pid); -+ at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->pid); -+ -+ /* Reset the SSC */ -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CR, AT91_SSC_SWRST); -+ -+ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_RPR, 0); -+ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_RCR, 0); -+ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_RNPR, 0); -+ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_RNCR, 0); -+ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_TPR, 0); -+ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_TCR, 0); -+ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_TNPR, 0); -+ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_TNCR, 0); -+ -+ mck_clk = clk_get(NULL, "mck"); -+ -+ div = rtd->cpu_dai->dai_runtime.priv >> 16; -+ period = rtd->cpu_dai->dai_runtime.priv & 0xffff; -+ bclk = 60000000 / (2 * div); -+ -+ DBG("mck %ld fsbd %d bfs %d bfs_real %d bclk %ld div %d period %d\n", -+ clk_get_rate(mck_clk), -+ SND_SOC_FSBD(6), -+ rtd->cpu_dai->dai_runtime.bfs, -+ SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs), -+ bclk, -+ div, -+ period); -+ -+ clk_put(mck_clk); -+ -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CMR, div); -+ -+ /* -+ * Setup the TFMR and RFMR for the proper data format. -+ */ -+ tfmr = -+ (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) -+ | (( 0 << 23) & AT91_SSC_FSDEN) -+ | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) -+ | (((bits - 1) << 16) & AT91_SSC_FSLEN) -+ | (((channels - 1) << 8) & AT91_SSC_DATNB) -+ | (( 1 << 7) & AT91_SSC_MSBF) -+ | (( 0 << 5) & AT91_SSC_DATDEF) -+ | (((bits - 1) << 0) & AT91_SSC_DATALEN); -+ DBG("SSC_TFMR=0x%08x\n", tfmr); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_TFMR, tfmr); -+ -+ rfmr = -+ (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) -+ | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) -+ | (( 0 << 16) & AT91_SSC_FSLEN) -+ | (((channels - 1) << 8) & AT91_SSC_DATNB) -+ | (( 1 << 7) & AT91_SSC_MSBF) -+ | (( 0 << 5) & AT91_SSC_LOOP) -+ | (((bits - 1) << 0) & AT91_SSC_DATALEN); -+ -+ DBG("SSC_RFMR=0x%08x\n", rfmr); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RFMR, rfmr); -+ -+ /* -+ * Setup the TCMR and RCMR to generate the proper BCLK -+ * and LRC signals. -+ */ -+ tcmr = -+ (( period << 24) & AT91_SSC_PERIOD) -+ | (( 1 << 16) & AT91_SSC_STTDLY) -+ | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) -+ | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) -+ | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) -+ | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); -+ -+ DBG("SSC_TCMR=0x%08x\n", tcmr); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_TCMR, tcmr); -+ -+ rcmr = -+ (( 0 << 24) & AT91_SSC_PERIOD) -+ | (( 1 << 16) & AT91_SSC_STTDLY) -+ | (( AT91_SSC_START_TX_RX ) & AT91_SSC_START) -+ | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) -+ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) -+ | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS); -+ -+ DBG("SSC_RCMR=0x%08x\n", rcmr); -+ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, rcmr); -+ -+ if ((ret = request_irq(ssc_p->pid, at91rm9200_i2s_interrupt, -+ 0, ssc_p->name, ssc_p)) < 0) { -+ printk(KERN_WARNING "at91rm9200-i2s: request_irq failure\n"); -+ return ret; -+ } -+ -+ /* -+ * Save the current substream parameters in order to check -+ * that the substream in the opposite direction uses the -+ * same parameters. -+ */ -+ ssc_p->pcmfmt = pcmfmt; -+ ssc_p->rate = rate; -+ ssc_p->initialized = 1; -+ -+ DBG("hw_params: SSC initialized\n"); -+ } -+ -+ up(ssc_p->mutex); -+ -+ return 0; -+} -+ -+ -+static int at91rm9200_i2s_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ at91rm9200_pcm_dma_params_t *dma_params = rtd->cpu_dai->dma_data; -+ -+ at91_ssc_write(dma_params->ssc->cr, dma_params->mask->ssc_enable); -+ -+ DBG("%s enabled SSC_SR=0x%08lx\n", -+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "transmit" : "receive", -+ at91_ssc_read(ssc_info[rtd->cpu_dai->id].ssc_base + AT91_SSC_SR)); -+ return 0; -+} -+ -+ -+struct snd_soc_cpu_dai at91rm9200_i2s_dai[] = { -+ { .name = "at91rm9200-ssc0/i2s", -+ .id = 0, -+ .type = SND_SOC_DAI_I2S, -+ .suspend = at91rm9200_i2s_suspend, -+ .resume = at91rm9200_i2s_resume, -+ .config_sysclk = at91rm9200_i2s_config_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = at91rm9200_i2s_startup, -+ .shutdown = at91rm9200_i2s_shutdown, -+ .prepare = at91rm9200_i2s_prepare, -+ .hw_params = at91rm9200_i2s_hw_params,}, -+ .caps = { -+ .mode = &at91rm9200_i2s[0], -+ .num_modes = ARRAY_SIZE(at91rm9200_i2s),}, -+ }, -+ { .name = "at91rm9200-ssc1/i2s", -+ .id = 1, -+ .type = SND_SOC_DAI_I2S, -+ .suspend = at91rm9200_i2s_suspend, -+ .resume = at91rm9200_i2s_resume, -+ .config_sysclk = at91rm9200_i2s_config_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = at91rm9200_i2s_startup, -+ .shutdown = at91rm9200_i2s_shutdown, -+ .prepare = at91rm9200_i2s_prepare, -+ .hw_params = at91rm9200_i2s_hw_params,}, -+ .caps = { -+ .mode = &at91rm9200_i2s[0], -+ .num_modes = ARRAY_SIZE(at91rm9200_i2s),}, -+ }, -+ { .name = "at91rm9200-ssc2/i2s", -+ .id = 2, -+ .type = SND_SOC_DAI_I2S, -+ .suspend = at91rm9200_i2s_suspend, -+ .resume = at91rm9200_i2s_resume, -+ .config_sysclk = at91rm9200_i2s_config_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = at91rm9200_i2s_startup, -+ .shutdown = at91rm9200_i2s_shutdown, -+ .prepare = at91rm9200_i2s_prepare, -+ .hw_params = at91rm9200_i2s_hw_params,}, -+ .caps = { -+ .mode = &at91rm9200_i2s[0], -+ .num_modes = ARRAY_SIZE(at91rm9200_i2s),}, -+ }, -+}; -+ -+EXPORT_SYMBOL_GPL(at91rm9200_i2s_dai); -+ -+/* Module information */ -+MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com"); -+MODULE_DESCRIPTION("AT91RM9200 I2S ASoC Interface"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/at91/at91rm9200-pcm.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/at91/at91rm9200-pcm.c -@@ -0,0 +1,428 @@ -+/* -+ * at91rm9200-pcm.c -- ALSA PCM interface for the Atmel AT91RM9200 chip. -+ * -+ * Author: Frank Mandarino <fmandarino@endrelia.com> -+ * Endrelia Technologies Inc. -+ * Created: Mar 3, 2006 -+ * -+ * Based on pxa2xx-pcm.c by: -+ * -+ * Author: Nicolas Pitre -+ * Created: Nov 30, 2004 -+ * Copyright: (C) 2004 MontaVista Software, Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/init.h> -+#include <linux/platform_device.h> -+#include <linux/slab.h> -+#include <linux/dma-mapping.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+ -+#include <asm/arch/at91rm9200.h> -+#include <asm/arch/at91rm9200_ssc.h> -+#include <asm/arch/at91rm9200_pdc.h> -+#include <asm/arch/hardware.h> -+ -+#include "at91rm9200-pcm.h" -+ -+#if 0 -+#define DBG(x...) printk(KERN_INFO "at91rm9200-pcm: " x) -+#else -+#define DBG(x...) -+#endif -+ -+static const snd_pcm_hardware_t at91rm9200_pcm_hardware = { -+ .info = SNDRV_PCM_INFO_MMAP | -+ SNDRV_PCM_INFO_MMAP_VALID | -+ SNDRV_PCM_INFO_INTERLEAVED | -+ SNDRV_PCM_INFO_PAUSE, -+ .formats = SNDRV_PCM_FMTBIT_S16_LE, -+ .period_bytes_min = 32, -+ .period_bytes_max = 8192, -+ .periods_min = 2, -+ .periods_max = 1024, -+ .buffer_bytes_max = 32 * 1024, -+}; -+ -+struct at91rm9200_runtime_data { -+ at91rm9200_pcm_dma_params_t *params; -+ dma_addr_t dma_buffer; /* physical address of dma buffer */ -+ dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ -+ size_t period_size; -+ dma_addr_t period_ptr; /* physical address of next period */ -+ u32 pdc_xpr_save; /* PDC register save */ -+ u32 pdc_xcr_save; -+ u32 pdc_xnpr_save; -+ u32 pdc_xncr_save; -+}; -+ -+static void at91rm9200_pcm_dma_irq(u32 ssc_sr, -+ struct snd_pcm_substream *substream) -+{ -+ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; -+ at91rm9200_pcm_dma_params_t *params = prtd->params; -+ static int count = 0; -+ -+ count++; -+ -+ if (ssc_sr & params->mask->ssc_endbuf) { -+ -+ printk(KERN_WARNING -+ "at91rm9200-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", -+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK -+ ? "underrun" : "overrun", -+ params->name, ssc_sr, count); -+ -+ /* re-start the PDC */ -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); -+ -+ prtd->period_ptr += prtd->period_size; -+ if (prtd->period_ptr >= prtd->dma_buffer_end) { -+ prtd->period_ptr = prtd->dma_buffer; -+ } -+ -+ at91_ssc_write(params->pdc->xpr, prtd->period_ptr); -+ at91_ssc_write(params->pdc->xcr, -+ prtd->period_size / params->pdc_xfer_size); -+ -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_enable); -+ } -+ -+ if (ssc_sr & params->mask->ssc_endx) { -+ -+ /* Load the PDC next pointer and counter registers */ -+ prtd->period_ptr += prtd->period_size; -+ if (prtd->period_ptr >= prtd->dma_buffer_end) { -+ prtd->period_ptr = prtd->dma_buffer; -+ } -+ at91_ssc_write(params->pdc->xnpr, prtd->period_ptr); -+ at91_ssc_write(params->pdc->xncr, -+ prtd->period_size / params->pdc_xfer_size); -+ } -+ -+ snd_pcm_period_elapsed(substream); -+} -+ -+static int at91rm9200_pcm_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ snd_pcm_runtime_t *runtime = substream->runtime; -+ struct at91rm9200_runtime_data *prtd = runtime->private_data; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ /* this may get called several times by oss emulation -+ * with different params */ -+ -+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); -+ runtime->dma_bytes = params_buffer_bytes(params); -+ -+ prtd->params = rtd->cpu_dai->dma_data; -+ prtd->params->dma_intr_handler = at91rm9200_pcm_dma_irq; -+ -+ prtd->dma_buffer = runtime->dma_addr; -+ prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; -+ prtd->period_size = params_period_bytes(params); -+ -+ DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n", -+ prtd->params->name, runtime->dma_bytes, prtd->period_size); -+ return 0; -+} -+ -+static int at91rm9200_pcm_hw_free(struct snd_pcm_substream *substream) -+{ -+ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; -+ at91rm9200_pcm_dma_params_t *params = prtd->params; -+ -+ if (params != NULL) { -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); -+ prtd->params->dma_intr_handler = NULL; -+ } -+ -+ return 0; -+} -+ -+static int at91rm9200_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; -+ at91rm9200_pcm_dma_params_t *params = prtd->params; -+ -+ at91_ssc_write(params->ssc->idr, -+ params->mask->ssc_endx | params->mask->ssc_endbuf); -+ -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); -+ return 0; -+} -+ -+static int at91rm9200_pcm_trigger(struct snd_pcm_substream *substream, -+ int cmd) -+{ -+ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; -+ at91rm9200_pcm_dma_params_t *params = prtd->params; -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ prtd->period_ptr = prtd->dma_buffer; -+ -+ at91_ssc_write(params->pdc->xpr, prtd->period_ptr); -+ at91_ssc_write(params->pdc->xcr, -+ prtd->period_size / params->pdc_xfer_size); -+ -+ prtd->period_ptr += prtd->period_size; -+ at91_ssc_write(params->pdc->xnpr, prtd->period_ptr); -+ at91_ssc_write(params->pdc->xncr, -+ prtd->period_size / params->pdc_xfer_size); -+ -+ DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n", -+ (unsigned long) prtd->period_ptr, -+ at91_ssc_read(params->pdc->xpr), -+ at91_ssc_read(params->pdc->xcr), -+ at91_ssc_read(params->pdc->xnpr), -+ at91_ssc_read(params->pdc->xncr)); -+ -+ at91_ssc_write(params->ssc->ier, -+ params->mask->ssc_endx | params->mask->ssc_endbuf); -+ -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_enable); -+ -+ DBG("sr=%lx imr=%lx\n", at91_ssc_read(params->ssc->ier - 4), -+ at91_ssc_read(params->ssc->ier + 8)); -+ break; -+ -+ case SNDRV_PCM_TRIGGER_STOP: -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); -+ break; -+ -+ case SNDRV_PCM_TRIGGER_RESUME: -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_enable); -+ break; -+ -+ default: -+ ret = -EINVAL; -+ } -+ -+ return ret; -+} -+ -+static snd_pcm_uframes_t at91rm9200_pcm_pointer( -+ struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct at91rm9200_runtime_data *prtd = runtime->private_data; -+ at91rm9200_pcm_dma_params_t *params = prtd->params; -+ dma_addr_t ptr; -+ snd_pcm_uframes_t x; -+ -+ ptr = (dma_addr_t) at91_ssc_read(params->pdc->xpr); -+ x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); -+ -+ if (x == runtime->buffer_size) -+ x = 0; -+ return x; -+} -+ -+static int at91rm9200_pcm_open(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct at91rm9200_runtime_data *prtd; -+ int ret = 0; -+ -+ snd_soc_set_runtime_hwparams(substream, &at91rm9200_pcm_hardware); -+ -+ /* ensure that buffer size is a multiple of period size */ -+ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); -+ if (ret < 0) -+ goto out; -+ -+ prtd = kzalloc(sizeof(struct at91rm9200_runtime_data), GFP_KERNEL); -+ if (prtd == NULL) { -+ ret = -ENOMEM; -+ goto out; -+ } -+ runtime->private_data = prtd; -+ -+ out: -+ return ret; -+} -+ -+static int at91rm9200_pcm_close(struct snd_pcm_substream *substream) -+{ -+ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; -+ -+ kfree(prtd); -+ return 0; -+} -+ -+static int at91rm9200_pcm_mmap(struct snd_pcm_substream *substream, -+ struct vm_area_struct *vma) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ -+ return dma_mmap_writecombine(substream->pcm->card->dev, vma, -+ runtime->dma_area, -+ runtime->dma_addr, -+ runtime->dma_bytes); -+} -+ -+struct snd_pcm_ops at91rm9200_pcm_ops = { -+ .open = at91rm9200_pcm_open, -+ .close = at91rm9200_pcm_close, -+ .ioctl = snd_pcm_lib_ioctl, -+ .hw_params = at91rm9200_pcm_hw_params, -+ .hw_free = at91rm9200_pcm_hw_free, -+ .prepare = at91rm9200_pcm_prepare, -+ .trigger = at91rm9200_pcm_trigger, -+ .pointer = at91rm9200_pcm_pointer, -+ .mmap = at91rm9200_pcm_mmap, -+}; -+ -+static int at91rm9200_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, -+ int stream) -+{ -+ struct snd_pcm_substream *substream = pcm->streams[stream].substream; -+ struct snd_dma_buffer *buf = &substream->dma_buffer; -+ size_t size = at91rm9200_pcm_hardware.buffer_bytes_max; -+ -+ buf->dev.type = SNDRV_DMA_TYPE_DEV; -+ buf->dev.dev = pcm->card->dev; -+ buf->private_data = NULL; -+ buf->area = dma_alloc_writecombine(pcm->card->dev, size, -+ &buf->addr, GFP_KERNEL); -+ -+ DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", -+ (void *) buf->area, -+ (void *) buf->addr, -+ size); -+ -+ if (!buf->area) -+ return -ENOMEM; -+ -+ buf->bytes = size; -+ return 0; -+} -+ -+static u64 at91rm9200_pcm_dmamask = 0xffffffff; -+ -+static int at91rm9200_pcm_new(struct snd_card *card, -+ struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) -+{ -+ int ret = 0; -+ -+ if (!card->dev->dma_mask) -+ card->dev->dma_mask = &at91rm9200_pcm_dmamask; -+ if (!card->dev->coherent_dma_mask) -+ card->dev->coherent_dma_mask = 0xffffffff; -+ -+ if (dai->playback.channels_min) { -+ ret = at91rm9200_pcm_preallocate_dma_buffer(pcm, -+ SNDRV_PCM_STREAM_PLAYBACK); -+ if (ret) -+ goto out; -+ } -+ -+ if (dai->capture.channels_min) { -+ ret = at91rm9200_pcm_preallocate_dma_buffer(pcm, -+ SNDRV_PCM_STREAM_CAPTURE); -+ if (ret) -+ goto out; -+ } -+ out: -+ return ret; -+} -+ -+static void at91rm9200_pcm_free_dma_buffers(struct snd_pcm *pcm) -+{ -+ struct snd_pcm_substream *substream; -+ struct snd_dma_buffer *buf; -+ int stream; -+ -+ for (stream = 0; stream < 2; stream++) { -+ substream = pcm->streams[stream].substream; -+ if (!substream) -+ continue; -+ -+ buf = &substream->dma_buffer; -+ if (!buf->area) -+ continue; -+ -+ dma_free_writecombine(pcm->card->dev, buf->bytes, -+ buf->area, buf->addr); -+ buf->area = NULL; -+ } -+} -+ -+static int at91rm9200_pcm_suspend(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ struct snd_pcm_runtime *runtime = dai->runtime; -+ struct at91rm9200_runtime_data *prtd; -+ at91rm9200_pcm_dma_params_t *params; -+ -+ if (!runtime) -+ return 0; -+ -+ prtd = runtime->private_data; -+ params = prtd->params; -+ -+ /* disable the PDC and save the PDC registers */ -+ -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); -+ -+ prtd->pdc_xpr_save = at91_ssc_read(params->pdc->xpr); -+ prtd->pdc_xcr_save = at91_ssc_read(params->pdc->xcr); -+ prtd->pdc_xnpr_save = at91_ssc_read(params->pdc->xnpr); -+ prtd->pdc_xncr_save = at91_ssc_read(params->pdc->xncr); -+ -+ return 0; -+} -+ -+static int at91rm9200_pcm_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ struct snd_pcm_runtime *runtime = dai->runtime; -+ struct at91rm9200_runtime_data *prtd; -+ at91rm9200_pcm_dma_params_t *params; -+ -+ if (!runtime) -+ return 0; -+ -+ prtd = runtime->private_data; -+ params = prtd->params; -+ -+ /* restore the PDC registers and enable the PDC */ -+ at91_ssc_write(params->pdc->xpr, prtd->pdc_xpr_save); -+ at91_ssc_write(params->pdc->xcr, prtd->pdc_xcr_save); -+ at91_ssc_write(params->pdc->xnpr, prtd->pdc_xnpr_save); -+ at91_ssc_write(params->pdc->xncr, prtd->pdc_xncr_save); -+ -+ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_enable); -+ return 0; -+} -+ -+struct snd_soc_platform at91rm9200_soc_platform = { -+ .name = "at91rm9200-audio", -+ .pcm_ops = &at91rm9200_pcm_ops, -+ .pcm_new = at91rm9200_pcm_new, -+ .pcm_free = at91rm9200_pcm_free_dma_buffers, -+ .suspend = at91rm9200_pcm_suspend, -+ .resume = at91rm9200_pcm_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(at91rm9200_soc_platform); -+ -+MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); -+MODULE_DESCRIPTION("Atmel AT91RM9200 PCM module"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/at91/at91rm9200-pcm.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/at91/at91rm9200-pcm.h -@@ -0,0 +1,75 @@ -+/* -+ * at91rm9200-pcm.h - ALSA PCM interface for the Atmel AT91RM9200 chip -+ * -+ * Author: Frank Mandarino <fmandarino@endrelia.com> -+ * Endrelia Technologies Inc. -+ * Created: Mar 3, 2006 -+ * -+ * Based on pxa2xx-pcm.h by: -+ * -+ * Author: Nicolas Pitre -+ * Created: Nov 30, 2004 -+ * Copyright: MontaVista Software, Inc. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+/* -+ * Registers and status bits that are required by the PCM driver. -+ */ -+struct at91rm9200_ssc_regs { -+ void __iomem *cr; /* SSC control */ -+ void __iomem *ier; /* SSC interrupt enable */ -+ void __iomem *idr; /* SSC interrupt disable */ -+}; -+ -+struct at91rm9200_pdc_regs { -+ void __iomem *xpr; /* PDC recv/trans pointer */ -+ void __iomem *xcr; /* PDC recv/trans counter */ -+ void __iomem *xnpr; /* PDC next recv/trans pointer */ -+ void __iomem *xncr; /* PDC next recv/trans counter */ -+ void __iomem *ptcr; /* PDC transfer control */ -+}; -+ -+struct at91rm9200_ssc_mask { -+ u32 ssc_enable; /* SSC recv/trans enable */ -+ u32 ssc_disable; /* SSC recv/trans disable */ -+ u32 ssc_endx; /* SSC ENDTX or ENDRX */ -+ u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ -+ u32 pdc_enable; /* PDC recv/trans enable */ -+ u32 pdc_disable; /* PDC recv/trans disable */ -+}; -+ -+ -+/* -+ * This structure, shared between the PCM driver and the interface, -+ * contains all information required by the PCM driver to perform the -+ * PDC DMA operation. All fields except dma_intr_handler() are initialized -+ * by the interface. The dms_intr_handler() pointer is set by the PCM -+ * driver and called by the interface SSC interrupt handler if it is -+ * non-NULL. -+ */ -+typedef struct { -+ char *name; /* stream identifier */ -+ int pdc_xfer_size; /* PDC counter increment in bytes */ -+ struct at91rm9200_ssc_regs *ssc; /* SSC register addresses */ -+ struct at91rm9200_pdc_regs *pdc; /* PDC receive/transmit registers */ -+ struct at91rm9200_ssc_mask *mask;/* SSC & PDC status bits */ -+ snd_pcm_substream_t *substream; -+ void (*dma_intr_handler)(u32, snd_pcm_substream_t *); -+} at91rm9200_pcm_dma_params_t; -+ -+extern struct snd_soc_cpu_dai at91rm9200_i2s_dai[3]; -+extern struct snd_soc_platform at91rm9200_soc_platform; -+ -+ -+/* -+ * SSC I/O helpers. -+ * E.g., at91_ssc_write(AT91_SSC(1) + AT91_SSC_CR, AT91_SSC_RXEN); -+ */ -+#define AT91_SSC(x) (((x)==0) ? AT91_VA_BASE_SSC0 :\ -+ ((x)==1) ? AT91_VA_BASE_SSC1 : ((x)==2) ? AT91_VA_BASE_SSC2 : NULL) -+#define at91_ssc_read(a) ((unsigned long) __raw_readl(a)) -+#define at91_ssc_write(a,v) __raw_writel((v),(a)) -Index: linux-2.6-pxa-new/sound/soc/imx/imx-ssi.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/imx-ssi.c -@@ -0,0 +1,452 @@ -+/* -+ * imx-ssi.c -- SSI driver for Freescale IMX -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Based on mxc-alsa-mc13783 (C) 2006 Freescale. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 29th Aug 2006 Initial version. -+ * -+ */ -+ -+#define IMX_DSP_DAIFMT \ -+ ( SND_SOC_DAIFMT_DSP__A |SND_SOC_DAIFMT_DSP_B | \ -+ SND_SOC_DAIFMT_CBS_CFS |SND_SOC_DAIFMT_CBM_CFS | \ -+ SND_SOC_DAIFMT_CBS_CFM |SND_SOC_DAIFMT_NB_NF |\ -+ SND_SOC_DAIFMT_NB_IF) -+ -+#define IMX_DSP_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define IMX_DSP_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ -+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ -+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ -+ SNDRV_PCM_RATE_96000) -+ -+#define IMX_DSP_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE) -+ -+static struct snd_soc_dai_mode imx_dsp_pcm_modes[] = { -+ -+ /* frame master and clock slave mode */ -+ {IMX_DSP_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, -+ SND_SOC_DAITDM_LRDW(0,0), IMX_DSP_BITS, IMX_DSP_RATES, -+ IMX_DSP_DIR, 0, SND_SOC_FS_ALL, -+ SND_SOC_FSB(32) | SND_SOC_FSB(32) | SND_SOC_FSB(16)}, -+ -+}; -+ -+static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_out = { -+ .name = "SSI1 PCM Stereo out", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = emi_2_per, -+ .watermark_level = SDMA_TXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI1_STX0, -+ .event_id = DMA_REQ_SSI1_TX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_in = { -+ .name = "SSI1 PCM Stereo in", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_RXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI1_SRX0, -+ .event_id = DMA_REQ_SSI1_RX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_out = { -+ .name = "SSI2 PCM Stereo out", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_TXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI2_STX0, -+ .event_id = DMA_REQ_SSI2_TX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_in = { -+ .name = "SSI2 PCM Stereo in", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_RXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI2_SRX0, -+ .event_id = DMA_REQ_SSI2_RX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static int imx_dsp_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if (!rtd->cpu_dai->active) { -+ -+ } -+ -+ return 0; -+} -+ -+static int imx_ssi1_hw_tx_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 bfs, div; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); -+ -+ SSI1_STCR = 0; -+ SSI1_STCCR = 0; -+ -+ /* DAI mode */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_DSP_B: -+ SSI1_STCR |= SSI_STCR_TEFS; // data 1 bit after sync -+ case SND_SOC_DAIFMT_DSP_A: -+ SSI1_STCR |= SSI_STCR_TFSL; // frame is 1 bclk long -+ break; -+ } -+ -+ /* DAI clock inversion */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_IB_IF: -+ SSI1_STCR |= SSI_STCR_TFSI | SSI_STCR_TSCKP; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ SSI1_STCR |= SSI_STCR_TSCKP; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ SSI1_STCR |= SSI_STCR_TFSI; -+ break; -+ } -+ -+ /* DAI data (word) size */ -+ switch(rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ SSI1_STCCR |= SSI_STCCR_WL(16); -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ SSI1_STCCR |= SSI_STCCR_WL(20); -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ SSI1_STCCR |= SSI_STCCR_WL(24); -+ break; -+ } -+ -+ /* DAI clock master masks */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ -+ case SND_SOC_DAIFMT_CBM_CFM: -+ SSI1_STCR |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFM: -+ SSI1_STCR |= SSI_STCR_TFDIR; -+ break; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ SSI1_STCR |= SSI_STCR_TXDIR; -+ break; -+ } -+ -+ /* DAI BCLK ratio to SYSCLK / MCLK */ -+ /* prescaler modulus - todo */ -+ switch (bfs) { -+ case 2: -+ break; -+ case 4: -+ break; -+ case 8: -+ break; -+ case 16: -+ break; -+ } -+ -+ /* TDM - todo, only fifo 0 atm */ -+ SSI1_STCR |= SSI_STCR_TFEN0; -+ SSI1_STCCR |= SSI_STCCR_DC(params_channels(params)); -+ -+ return 0; -+} -+ -+static int imx_ssi1_hw_rx_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 bfs, div; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); -+ -+ SSI1_SRCR = 0; -+ SSI1_SRCCR = 0; -+ -+ /* DAI mode */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_DSP_B: -+ SSI1_SRCR |= SSI_SRCR_REFS; // data 1 bit after sync -+ case SND_SOC_DAIFMT_DSP_A: -+ SSI1_SRCR |= SSI_SRCR_RFSL; // frame is 1 bclk long -+ break; -+ } -+ -+ /* DAI clock inversion */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_IB_IF: -+ SSI1_SRCR |= SSI_SRCR_TFSI | SSI_SRCR_TSCKP; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ SSI1_SRCR |= SSI_SRCR_RSCKP; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ SSI1_SRCR |= SSI_SRCR_RFSI; -+ break; -+ } -+ -+ /* DAI data (word) size */ -+ switch(rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ SSI1_SRCCR |= SSI_SRCCR_WL(16); -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ SSI1_SRCCR |= SSI_SRCCR_WL(20); -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ SSI1_SRCCR |= SSI_SRCCR_WL(24); -+ break; -+ } -+ -+ /* DAI clock master masks */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ -+ case SND_SOC_DAIFMT_CBM_CFM: -+ SSI1_SRCR |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFM: -+ SSI1_SRCR |= SSI_SRCR_RFDIR; -+ break; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ SSI1_SRCR |= SSI_SRCR_RXDIR; -+ break; -+ } -+ -+ /* DAI BCLK ratio to SYSCLK / MCLK */ -+ /* prescaler modulus - todo */ -+ switch (bfs) { -+ case 2: -+ break; -+ case 4: -+ break; -+ case 8: -+ break; -+ case 16: -+ break; -+ } -+ -+ /* TDM - todo, only fifo 0 atm */ -+ SSI1_SRCR |= SSI_SRCR_RFEN0; -+ SSI1_SRCCR |= SSI_SRCCR_DC(params_channels(params)); -+ -+ return 0; -+} -+ -+static int imx_ssi_dsp_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ /* clear register if not enabled */ -+ if(!(SSI1_SCR & SSI_SCR_SSIEN)) -+ SSI1_SCR = 0; -+ -+ /* async */ -+ if (rtd->cpu_dai->flags & SND_SOC_DAI_ASYNC) -+ SSI1_SCR |= SSI_SCR_SYN; -+ -+ /* DAI mode */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ case SND_SOC_DAIFMT_LEFT_J: -+ SSI1_SCR |= SSI_SCR_NET; -+ break; -+ } -+ -+ /* TDM - to complete */ -+ -+ /* Tx/Rx config */ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ return imx_ssi1_dsp_hw_tx_params(substream, params); -+ } else { -+ return imx_ssi1_dsp_hw_rx_params(substream, params); -+ } -+} -+ -+ -+ -+static int imx_ssi_dsp_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ SSI1_SCR |= SSI_SCR_TE; -+ SSI1_SIER |= SSI_SIER_TDMAE; -+ } else { -+ SSI1_SCR |= SSI_SCR_RE; -+ SSI1_SIER |= SSI_SIER_RDMAE; -+ } -+ SSI1_SCR |= SSI_SCR_SSIEN; -+ -+ break; -+ case SNDRV_PCM_TRIGGER_RESUME: -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSI1_SCR |= SSI_SCR_TE; -+ else -+ SSI1_SCR |= SSI_SCR_RE; -+ break -+ case SNDRV_PCM_TRIGGER_STOP: -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSI1_SCR &= ~SSI_SCR_TE; -+ else -+ SSI1_SCR &= ~SSI_SCR_RE; -+ break; -+ default: -+ ret = -EINVAL; -+ } -+ -+ return ret; -+} -+ -+static void imx_ssi_dsp_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ /* shutdown SSI */ -+ if (!rtd->cpu_dai->active) { -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR &= ~SSI_SCR_SSIEN; -+ else -+ SSI2_SCR &= ~SSI_SCR_SSIEN; -+ } -+} -+ -+#ifdef CONFIG_PM -+static int imx_ssi_dsp_suspend(struct platform_device *dev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if(!dai->active) -+ return 0; -+ -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR &= ~SSI_SCR_SSIEN; -+ else -+ SSI2_SCR &= ~SSI_SCR_SSIEN; -+ -+ return 0; -+} -+ -+static int imx_ssi_dsp_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if(!dai->active) -+ return 0; -+ -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR |= SSI_SCR_SSIEN; -+ else -+ SSI2_SCR |= SSI_SCR_SSIEN; -+ -+ return 0; -+} -+ -+#else -+#define imx_ssi_dsp_suspend NULL -+#define imx_ssi_dsp_resume NULL -+#endif -+ -+static unsigned int imx_ssi_config_dsp_sysclk(struct snd_soc_cpu_dai *iface, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ return clk; -+} -+ -+struct snd_soc_cpu_dai imx_ssi_dsp_dai = { -+ .name = "imx-dsp-1", -+ .id = 0, -+ .type = SND_SOC_DAI_PCM, -+ .suspend = imx_ssi_dsp_suspend, -+ .resume = imx_ssi_dsp_resume, -+ .config_sysclk = imx_ssi_config_dsp_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = imx_ssi_dsp_startup, -+ .shutdown = imx_ssi_dsp_shutdown, -+ .trigger = imx_ssi_trigger, -+ .hw_params = imx_ssi_dsp_hw_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(imx_dsp_modes), -+ .mode = imx_dsp_modes,}, -+}, -+{ -+ .name = "imx-dsp-2", -+ .id = 1, -+ .type = SND_SOC_DAI_PCM, -+ .suspend = imx_ssi_dsp_suspend, -+ .resume = imx_ssi_dsp_resume, -+ .config_sysclk = imx_ssi_config_dsp_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = imx_dsp_startup, -+ .shutdown = imx_dsp_shutdown, -+ .trigger = imx_ssi1_trigger, -+ .hw_params = imx_ssi1_pcm_hw_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(imx_dsp_modes), -+ .mode = imx_dsp_modes,}, -+}; -+ -+ -+EXPORT_SYMBOL_GPL(imx_ssi_dsp_dai); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("i.MX ASoC SSI driver"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/imx/Kconfig -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/Kconfig -@@ -0,0 +1,31 @@ -+menu "SoC Audio for the Freescale i.MX" -+ -+config SND_MXC_SOC -+ tristate "SoC Audio for the Freescale i.MX CPU" -+ depends on ARCH_MXC && SND -+ select SND_PCM -+ help -+ Say Y or M if you want to add support for codecs attached to -+ the MXC AC97, I2S or SSP interface. You will also need -+ to select the audio interfaces to support below. -+ -+config SND_MXC_AC97 -+ tristate -+ select SND_AC97_CODEC -+ -+config SND_MXC_SOC_AC97 -+ tristate -+ select SND_AC97_BUS -+ -+config SND_MXC_SOC_SSI -+ tristate -+ -+config SND_MXC_SOC_MX3_WM8753 -+ tristate "SoC Audio support for MX31 - WM8753" -+ depends on SND_MXC_SOC && ARCH_MX3 -+ select SND_MXC_SOC_SSI -+ help -+ Say Y if you want to add support for SoC audio on MX31ADS -+ with the WM8753. -+ -+endmenu -Index: linux-2.6-pxa-new/sound/soc/imx/Makefile -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/Makefile -@@ -0,0 +1,18 @@ -+# i.MX Platform Support -+snd-soc-imx21-objs := imx21-pcm.o -+snd-soc-imx31-objs := imx31-pcm.o -+snd-soc-imx-ac97-objs := imx-ac97.o -+snd-soc-imx-i2s-objs := imx-i2s.o -+ -+obj-$(CONFIG_SND_MXC_SOC) += snd-soc-imx.o -+obj-$(CONFIG_SND_MXC_SOC_AC97) += snd-soc-imx-ac97.o -+obj-$(CONFIG_SND_MXC_SOC_I2S) += snd-soc-imx-i2s.o -+ -+# i.MX Machine Support -+snd-soc-mx31ads-wm8753-objs := mx31ads_wm8753.o -+obj-$(CONFIG_SND_SOC_MX31ADS_WM8753) += snd-soc-mx31ads-wm8753.o -+snd-soc-mx21ads-wm8753-objs := mx21ads_wm8753.o -+obj-$(CONFIG_SND_SOC_MX21ADS_WM8753) += snd-soc-mx21ads-wm8753.o -+snd-soc-mx21ads-wm8731-objs := mx21ads_wm8731.o -+obj-$(CONFIG_SND_SOC_MX21ADS_WM8731) += snd-soc-mx21ads-wm8731.o -+ -Index: linux-2.6.17/sound/Makefile -=================================================================== ---- linux-2.6.17.orig/sound/Makefile 2006-06-18 02:49:35.000000000 +0100 -+++ linux-2.6.17/sound/Makefile 2006-07-04 14:04:41.000000000 +0100 -@@ -4,7 +4,7 @@ - obj-$(CONFIG_SOUND) += soundcore.o - obj-$(CONFIG_SOUND_PRIME) += oss/ - obj-$(CONFIG_DMASOUND) += oss/ --obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ -+obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ - - ifeq ($(CONFIG_SND),y) - obj-y += last.o -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8711.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8711.c -@@ -0,0 +1,843 @@ -+/* -+ * wm8711.c -- WM8711 ALSA SoC Audio driver -+ * -+ * Copyright 2006 Wolfson Microelectronics -+ * -+ * Author: Mike Arthur <linux@wolfsonmicro.com> -+ * -+ * Based on wm8711.c by Richard Purdie -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8711.h" -+ -+#define AUDIO_NAME "wm8711" -+#define WM8711_VERSION "0.2" -+ -+/* -+ * Debug -+ */ -+ -+#define WM8711_DEBUG 0 -+ -+#ifdef WM8711_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+struct snd_soc_codec_device soc_codec_dev_wm8711; -+ -+/* -+ * wm8711 register cache -+ * We can't read the WM8711 register space when we are -+ * using 2 wire for device control, so we cache them instead. -+ * There is no point in caching the reset register -+ */ -+static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { -+ 0x0079, 0x0079, 0x000a, 0x0008, -+ 0x009f, 0x000a, 0x0000, 0x0000 -+}; -+ -+#define WM8711_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ -+ SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8711_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK) -+ -+#define WM8711_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -+ -+#define WM8711_HIFI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ -+ SNDRV_PCM_FMTBIT_S32_LE) -+ -+static struct snd_soc_dai_mode wm8711_modes[] = { -+ /* codec frame and clock master modes */ -+ /* 8k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 1536, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 2304, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 1408, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 2112, -+ .bfs = 64, -+ }, -+ -+ /* 32k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_32000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 576, -+ .bfs = 64, -+ }, -+ -+ /* 44.1k & 48k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 256, -+ .bfs = 64, -+ }, -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 64, -+ }, -+ -+ /* 88.2 & 96k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 128, -+ .bfs = 64, -+ -+ }, -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 192, -+ .bfs = 64, -+ }, -+ -+ /* USB codec frame and clock master modes */ -+ /* 8k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_8000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 1500, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* 44.1k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 272, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* 48k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_48000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 250, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* 88.2k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_88200, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 136, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* 96k */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = SNDRV_PCM_RATE_96000, -+ .pcmdir = WM8711_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 125, -+ .bfs = SND_SOC_FSBD(1), -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8711_HIFI_BITS, -+ .pcmrate = WM8711_RATES, -+ .pcmdir = WM8711_DIR, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8711 register cache -+ */ -+static inline unsigned int wm8711_read_reg_cache(struct snd_soc_codec * codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg == WM8711_RESET) -+ return 0; -+ if (reg >= WM8711_CACHEREGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write wm8711 register cache -+ */ -+static inline void wm8711_write_reg_cache(struct snd_soc_codec *codec, -+ u16 reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= WM8711_CACHEREGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+/* -+ * write to the WM8711 register space -+ */ -+static int wm8711_write(struct snd_soc_codec * codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8753 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8711_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -EIO; -+} -+ -+#define wm8711_reset(c) wm8711_write(c, WM8711_RESET, 0) -+ -+static const struct snd_kcontrol_new wm8711_snd_controls[] = { -+ -+SOC_DOUBLE_R("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V, -+ 0, 127, 0), -+SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V, -+ 7, 1, 0), -+ -+}; -+ -+/* add non dapm controls */ -+static int wm8711_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8711_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8711_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* Output Mixer */ -+static const snd_kcontrol_new_t wm8711_output_mixer_controls[] = { -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0), -+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0), -+}; -+ -+static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = { -+SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1, -+ &wm8711_output_mixer_controls[0], -+ ARRAY_SIZE(wm8711_output_mixer_controls)), -+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1), -+SND_SOC_DAPM_OUTPUT("LOUT"), -+SND_SOC_DAPM_OUTPUT("LHPOUT"), -+SND_SOC_DAPM_OUTPUT("ROUT"), -+SND_SOC_DAPM_OUTPUT("RHPOUT"), -+}; -+ -+static const char *intercon[][3] = { -+ /* output mixer */ -+ {"Output Mixer", "Line Bypass Switch", "Line Input"}, -+ {"Output Mixer", "HiFi Playback Switch", "DAC"}, -+ -+ /* outputs */ -+ {"RHPOUT", NULL, "Output Mixer"}, -+ {"ROUT", NULL, "Output Mixer"}, -+ {"LHPOUT", NULL, "Output Mixer"}, -+ {"LOUT", NULL, "Output Mixer"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8711_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8711_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8711_dapm_widgets[i]); -+ } -+ -+ /* set up audio path interconnects */ -+ for(i = 0; intercon[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], -+ intercon[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+struct _coeff_div { -+ u32 mclk; -+ u32 rate; -+ u16 fs; -+ u8 sr:4; -+ u8 bosr:1; -+ u8 usb:1; -+}; -+ -+/* codec mclk clock divider coefficients */ -+static const struct _coeff_div coeff_div[] = { -+ /* 48k */ -+ {12288000, 48000, 256, 0x0, 0x0, 0x0}, -+ {18432000, 48000, 384, 0x0, 0x1, 0x0}, -+ {12000000, 48000, 250, 0x0, 0x0, 0x1}, -+ -+ /* 32k */ -+ {12288000, 32000, 384, 0x6, 0x0, 0x0}, -+ {18432000, 32000, 576, 0x6, 0x1, 0x0}, -+ -+ /* 8k */ -+ {12288000, 8000, 1536, 0x3, 0x0, 0x0}, -+ {18432000, 8000, 2304, 0x3, 0x1, 0x0}, -+ {11289600, 8000, 1408, 0xb, 0x0, 0x0}, -+ {16934400, 8000, 2112, 0xb, 0x1, 0x0}, -+ {12000000, 8000, 1500, 0x3, 0x0, 0x1}, -+ -+ /* 96k */ -+ {12288000, 96000, 128, 0x7, 0x0, 0x0}, -+ {18432000, 96000, 192, 0x7, 0x1, 0x0}, -+ {12000000, 96000, 125, 0x7, 0x0, 0x1}, -+ -+ /* 44.1k */ -+ {11289600, 44100, 256, 0x8, 0x0, 0x0}, -+ {16934400, 44100, 384, 0x8, 0x1, 0x0}, -+ {12000000, 44100, 272, 0x8, 0x1, 0x1}, -+ -+ /* 88.2k */ -+ {11289600, 88200, 128, 0xf, 0x0, 0x0}, -+ {16934400, 88200, 192, 0xf, 0x1, 0x0}, -+ {12000000, 88200, 136, 0xf, 0x1, 0x1}, -+}; -+ -+static inline int get_coeff(int mclk, int rate) -+{ -+ int i; -+ -+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { -+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) -+ return i; -+ } -+ return 0; -+} -+ -+/* WM8711 supports numerous clocks per sample rate */ -+static unsigned int wm8711_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ dai->mclk = 0; -+ -+ /* check that the calculated FS and rate actually match a clock from -+ * the machine driver */ -+ if (info->fs * info->rate == clk) -+ dai->mclk = clk; -+ -+ return dai->mclk; -+} -+ -+static int wm8711_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 iface = 0, srate; -+ int i = get_coeff(rtd->codec_dai->mclk, -+ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate)); -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ iface |= 0x0040; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | -+ coeff_div[i].usb; -+ wm8711_write(codec, WM8711_SRATE, srate); -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ iface |= 0x0002; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ iface |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ iface |= 0x0003; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ iface |= 0x0013; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FORMAT_S16_LE: -+ break; -+ case SNDRV_PCM_FORMAT_S20_3LE: -+ iface |= 0x0004; -+ break; -+ case SNDRV_PCM_FORMAT_S24_LE: -+ iface |= 0x0008; -+ break; -+ case SNDRV_PCM_FORMAT_S32_LE: -+ iface |= 0x000c; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_IF: -+ iface |= 0x0090; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ iface |= 0x0080; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ iface |= 0x0010; -+ break; -+ } -+ -+ /* set iface */ -+ wm8711_write(codec, WM8711_IFACE, iface); -+ -+ /* set active */ -+ wm8711_write(codec, WM8711_ACTIVE, 0x0001); -+ return 0; -+} -+ -+static void wm8711_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ /* deactivate */ -+ if (!codec->active) { -+ udelay(50); -+ wm8711_write(codec, WM8711_ACTIVE, 0x0); -+ } -+} -+ -+static int wm8711_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8711_read_reg_cache(codec, WM8711_APDIGI) & 0xfff7; -+ if (mute) -+ wm8711_write(codec, WM8711_APDIGI, mute_reg | 0x8); -+ else -+ wm8711_write(codec, WM8711_APDIGI, mute_reg); -+ -+ return 0; -+} -+ -+static int wm8711_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ u16 reg = wm8711_read_reg_cache(codec, WM8711_PWR) & 0xff7f; -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, osc on, dac unmute */ -+ wm8711_write(codec, WM8711_PWR, reg); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, */ -+ wm8711_write(codec, WM8711_PWR, reg | 0x0040); -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, dac mute, inactive */ -+ wm8711_write(codec, WM8711_ACTIVE, 0x0); -+ wm8711_write(codec, WM8711_PWR, 0xffff); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8711_dai = { -+ .name = "WM8711", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = wm8711_config_sysclk, -+ .digital_mute = wm8711_mute, -+ .ops = { -+ .prepare = wm8711_pcm_prepare, -+ .shutdown = wm8711_shutdown, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8711_modes), -+ .mode = wm8711_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8711_dai); -+ -+static int wm8711_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8711_write(codec, WM8711_ACTIVE, 0x0); -+ wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8711_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8711_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the WM8711 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8711_init(struct snd_soc_device* socdev) -+{ -+ struct snd_soc_codec* codec = socdev->codec; -+ int reg, ret = 0; -+ -+ codec->name = "WM8711"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8711_read_reg_cache; -+ codec->write = wm8711_write; -+ codec->dapm_event = wm8711_dapm_event; -+ codec->dai = &wm8711_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(wm8711_reg); -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8711_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8711_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8711_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8711_reg); -+ -+ wm8711_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if (ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ -+ /* set the update bits */ -+ reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V); -+ wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100); -+ reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V); -+ wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100); -+ -+ wm8711_add_controls(codec); -+ wm8711_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if (ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *wm8711_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8711 2 wire address is determined by GPIO5 -+ * state during powerup. -+ * low = 0x1a -+ * high = 0x1b -+ */ -+#define I2C_DRIVERID_WM8711 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8711_i2c_driver; -+static struct i2c_client client_template; -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+ -+static int wm8711_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8711_socdev; -+ struct wm8711_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL){ -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ -+ i2c_set_clientdata(i2c, codec); -+ -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if (ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8711_init(socdev); -+ if (ret < 0) { -+ err("failed to initialise WM8711\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+ -+} -+ -+static int wm8711_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec* codec = i2c_get_clientdata(client); -+ -+ i2c_detach_client(client); -+ -+ kfree(codec->reg_cache); -+ kfree(client); -+ -+ return 0; -+} -+ -+static int wm8711_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8711_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8711_i2c_driver = { -+ .driver = { -+ .name = "WM8711 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8711, -+ .attach_adapter = wm8711_i2c_attach, -+ .detach_client = wm8711_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8711", -+ .driver = &wm8711_i2c_driver, -+}; -+#endif -+ -+static int wm8711_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8711_setup_data *setup; -+ struct snd_soc_codec* codec; -+ int ret = 0; -+ -+ info("WM8711 Audio Codec %s", WM8711_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ wm8711_socdev = socdev; -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8711_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8711_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8711_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8711 = { -+ .probe = wm8711_probe, -+ .remove = wm8711_remove, -+ .suspend = wm8711_suspend, -+ .resume = wm8711_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); -+ -+MODULE_DESCRIPTION("ASoC WM8711 driver"); -+MODULE_AUTHOR("Mike Arthur"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8711.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8711.h -@@ -0,0 +1,39 @@ -+/* -+ * wm8711.h -- WM8711 Soc Audio driver -+ * -+ * Copyright 2006 Wolfson Microelectronics -+ * -+ * Author: Mike Arthur <linux@wolfsonmicro.com> -+ * -+ * Based on wm8731.h -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _WM8711_H -+#define _WM8711_H -+ -+/* WM8711 register space */ -+ -+#define WM8711_LOUT1V 0x02 -+#define WM8711_ROUT1V 0x03 -+#define WM8711_APANA 0x04 -+#define WM8711_APDIGI 0x05 -+#define WM8711_PWR 0x06 -+#define WM8711_IFACE 0x07 -+#define WM8711_SRATE 0x08 -+#define WM8711_ACTIVE 0x09 -+#define WM8711_RESET 0x0f -+ -+#define WM8711_CACHEREGNUM 8 -+ -+struct wm8711_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai wm8711_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8711; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8980.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8980.c -@@ -0,0 +1,991 @@ -+/* -+ * wm8980.c -- WM8980 ALSA Soc Audio driver -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * -+ * Authors: -+ * Mike Arthur <linux@wolfsonmicro.com> -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/version.h> -+#include <linux/kernel.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8980.h" -+ -+#define AUDIO_NAME "wm8980" -+#define WM8980_VERSION "0.2" -+ -+/* -+ * Debug -+ */ -+ -+#define WM8980_DEBUG 0 -+ -+#ifdef WM8980_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+struct snd_soc_codec_device soc_codec_dev_wm8980; -+ -+/* -+ * wm8980 register cache -+ * We can't read the WM8980 register space when we are -+ * using 2 wire for device control, so we cache them instead. -+ */ -+static const u16 wm8980_reg[WM8980_CACHEREGNUM] = { -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0050, 0x0000, 0x0140, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x00ff, -+ 0x00ff, 0x0000, 0x0100, 0x00ff, -+ 0x00ff, 0x0000, 0x012c, 0x002c, -+ 0x002c, 0x002c, 0x002c, 0x0000, -+ 0x0032, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0038, 0x000b, 0x0032, 0x0000, -+ 0x0008, 0x000c, 0x0093, 0x00e9, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0033, 0x0010, 0x0010, 0x0100, -+ 0x0100, 0x0002, 0x0001, 0x0001, -+ 0x0039, 0x0039, 0x0039, 0x0039, -+ 0x0001, 0x0001, -+}; -+ -+#define WM8980_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \ -+ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8980_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8980_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+#define WM8980_PCM_FORMATS \ -+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ -+ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | \ -+ SNDRV_PCM_FORMAT_S32_LE) -+ -+#define WM8980_BCLK \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\ -+ SND_SOC_FSBD(16) | SND_SOC_FSBD(32)) -+ -+static struct snd_soc_dai_mode wm8980_modes[] = { -+ /* codec frame and clock master modes */ -+ { -+ .fmt = WM8980_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8980_PCM_FORMATS, -+ .pcmrate = WM8980_RATES, -+ .pcmdir = WM8980_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = WM8980_BCLK, -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8980_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8980_PCM_FORMATS, -+ .pcmrate = WM8980_RATES, -+ .pcmdir = WM8980_DIR, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8980 register cache -+ */ -+static inline unsigned int wm8980_read_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg == WM8980_RESET) -+ return 0; -+ if (reg >= WM8980_CACHEREGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write wm8980 register cache -+ */ -+static inline void wm8980_write_reg_cache(struct snd_soc_codec *codec, -+ u16 reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= WM8980_CACHEREGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+/* -+ * write to the WM8980 register space -+ */ -+static int wm8980_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8980 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8980_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -1; -+} -+ -+#define wm8980_reset(c) wm8980_write(c, WM8980_RESET, 0) -+ -+static const char *wm8980_companding[] = {"Off", "NC", "u-law", "A-law" }; -+static const char *wm8980_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; -+static const char *wm8980_eqmode[] = {"Capture", "Playback" }; -+static const char *wm8980_bw[] = {"Narrow", "Wide" }; -+static const char *wm8980_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" }; -+static const char *wm8980_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" }; -+static const char *wm8980_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" }; -+static const char *wm8980_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; -+static const char *wm8980_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; -+static const char *wm8980_alc[] = -+ {"ALC both on", "ALC left only", "ALC right only", "Limiter" }; -+ -+static const struct soc_enum wm8980_enum[] = { -+ SOC_ENUM_SINGLE(WM8980_COMP, 1, 4, wm8980_companding), /* adc */ -+ SOC_ENUM_SINGLE(WM8980_COMP, 3, 4, wm8980_companding), /* dac */ -+ SOC_ENUM_SINGLE(WM8980_DAC, 4, 4, wm8980_deemp), -+ SOC_ENUM_SINGLE(WM8980_EQ1, 8, 2, wm8980_eqmode), -+ -+ SOC_ENUM_SINGLE(WM8980_EQ1, 5, 4, wm8980_eq1), -+ SOC_ENUM_SINGLE(WM8980_EQ2, 8, 2, wm8980_bw), -+ SOC_ENUM_SINGLE(WM8980_EQ2, 5, 4, wm8980_eq2), -+ SOC_ENUM_SINGLE(WM8980_EQ3, 8, 2, wm8980_bw), -+ -+ SOC_ENUM_SINGLE(WM8980_EQ3, 5, 4, wm8980_eq3), -+ SOC_ENUM_SINGLE(WM8980_EQ4, 8, 2, wm8980_bw), -+ SOC_ENUM_SINGLE(WM8980_EQ4, 5, 4, wm8980_eq4), -+ SOC_ENUM_SINGLE(WM8980_EQ5, 8, 2, wm8980_bw), -+ -+ SOC_ENUM_SINGLE(WM8980_EQ5, 5, 4, wm8980_eq5), -+ SOC_ENUM_SINGLE(WM8980_ALC3, 8, 2, wm8980_alc), -+}; -+ -+static const struct snd_kcontrol_new wm8980_snd_controls[] = { -+SOC_SINGLE("Digital Loopback Switch", WM8980_COMP, 0, 1, 0), -+ -+SOC_ENUM("ADC Companding", wm8980_enum[0]), -+SOC_ENUM("DAC Companding", wm8980_enum[1]), -+ -+SOC_SINGLE("Jack Detection Enable", WM8980_JACK1, 6, 1, 0), -+ -+SOC_SINGLE("DAC Right Inversion Switch", WM8980_DAC, 1, 1, 0), -+SOC_SINGLE("DAC Left Inversion Switch", WM8980_DAC, 0, 1, 0), -+ -+SOC_SINGLE("Left Playback Volume", WM8980_DACVOLL, 0, 127, 0), -+SOC_SINGLE("Right Playback Volume", WM8980_DACVOLR, 0, 127, 0), -+ -+SOC_SINGLE("High Pass Filter Switch", WM8980_ADC, 8, 1, 0), -+SOC_SINGLE("High Pass Filter Switch", WM8980_ADC, 8, 1, 0), -+SOC_SINGLE("High Pass Cut Off", WM8980_ADC, 4, 7, 0), -+SOC_SINGLE("Right ADC Inversion Switch", WM8980_ADC, 1, 1, 0), -+SOC_SINGLE("Left ADC Inversion Switch", WM8980_ADC, 0, 1, 0), -+ -+SOC_SINGLE("Left Capture Volume", WM8980_ADCVOLL, 0, 127, 0), -+SOC_SINGLE("Right Capture Volume", WM8980_ADCVOLR, 0, 127, 0), -+ -+SOC_ENUM("Equaliser Function", wm8980_enum[3]), -+SOC_ENUM("EQ1 Cut Off", wm8980_enum[4]), -+SOC_SINGLE("EQ1 Volume", WM8980_EQ1, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ2 Bandwith", wm8980_enum[5]), -+SOC_ENUM("EQ2 Cut Off", wm8980_enum[6]), -+SOC_SINGLE("EQ2 Volume", WM8980_EQ2, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ3 Bandwith", wm8980_enum[7]), -+SOC_ENUM("EQ3 Cut Off", wm8980_enum[8]), -+SOC_SINGLE("EQ3 Volume", WM8980_EQ3, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ4 Bandwith", wm8980_enum[9]), -+SOC_ENUM("EQ4 Cut Off", wm8980_enum[10]), -+SOC_SINGLE("EQ4 Volume", WM8980_EQ4, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ5 Bandwith", wm8980_enum[11]), -+SOC_ENUM("EQ5 Cut Off", wm8980_enum[12]), -+SOC_SINGLE("EQ5 Volume", WM8980_EQ5, 0, 31, 1), -+ -+SOC_SINGLE("DAC Playback Limiter Switch", WM8980_DACLIM1, 8, 1, 0), -+SOC_SINGLE("DAC Playback Limiter Decay", WM8980_DACLIM1, 4, 15, 0), -+SOC_SINGLE("DAC Playback Limiter Attack", WM8980_DACLIM1, 0, 15, 0), -+ -+SOC_SINGLE("DAC Playback Limiter Threshold", WM8980_DACLIM2, 4, 7, 0), -+SOC_SINGLE("DAC Playback Limiter Boost", WM8980_DACLIM2, 0, 15, 0), -+ -+SOC_SINGLE("ALC Enable Switch", WM8980_ALC1, 8, 1, 0), -+SOC_SINGLE("ALC Capture Max Gain", WM8980_ALC1, 3, 7, 0), -+SOC_SINGLE("ALC Capture Min Gain", WM8980_ALC1, 0, 7, 0), -+ -+SOC_SINGLE("ALC Capture ZC Switch", WM8980_ALC2, 8, 1, 0), -+SOC_SINGLE("ALC Capture Hold", WM8980_ALC2, 4, 7, 0), -+SOC_SINGLE("ALC Capture Target", WM8980_ALC2, 0, 15, 0), -+ -+SOC_ENUM("ALC Capture Mode", wm8980_enum[13]), -+SOC_SINGLE("ALC Capture Decay", WM8980_ALC3, 4, 15, 0), -+SOC_SINGLE("ALC Capture Attack", WM8980_ALC3, 0, 15, 0), -+ -+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8980_NGATE, 3, 1, 0), -+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8980_NGATE, 0, 7, 0), -+ -+SOC_SINGLE("Left Capture PGA ZC Switch", WM8980_INPPGAL, 7, 1, 0), -+SOC_SINGLE("Left Capture PGA Volume", WM8980_INPPGAL, 0, 63, 0), -+ -+SOC_SINGLE("Right Capture PGA ZC Switch", WM8980_INPPGAR, 7, 1, 0), -+SOC_SINGLE("Right Capture PGA Volume", WM8980_INPPGAR, 0, 63, 0), -+ -+SOC_SINGLE("Left Headphone Playback ZC Switch", WM8980_HPVOLL, 7, 1, 0), -+SOC_SINGLE("Left Headphone Playback Switch", WM8980_HPVOLL, 6, 1, 1), -+SOC_SINGLE("Left Headphone Playback Volume", WM8980_HPVOLL, 0, 63, 0), -+ -+SOC_SINGLE("Right Headphone Playback ZC Switch", WM8980_HPVOLR, 7, 1, 0), -+SOC_SINGLE("Right Headphone Playback Switch", WM8980_HPVOLR, 6, 1, 1), -+SOC_SINGLE("Right Headphone Playback Volume", WM8980_HPVOLR, 0, 63, 0), -+ -+SOC_SINGLE("Left Speaker Playback ZC Switch", WM8980_SPKVOLL, 7, 1, 0), -+SOC_SINGLE("Left Speaker Playback Switch", WM8980_SPKVOLL, 6, 1, 1), -+SOC_SINGLE("Left Speaker Playback Volume", WM8980_SPKVOLL, 0, 63, 0), -+ -+SOC_SINGLE("Right Speaker Playback ZC Switch", WM8980_SPKVOLR, 7, 1, 0), -+SOC_SINGLE("Right Speaker Playback Switch", WM8980_SPKVOLR, 6, 1, 1), -+SOC_SINGLE("Right Speaker Playback Volume", WM8980_SPKVOLR, 0, 63, 0), -+ -+SOC_DOUBLE_R("Capture Boost(+20dB)", WM8980_ADCBOOSTL, WM8980_ADCBOOSTR, -+ 8, 1, 0), -+}; -+ -+/* add non dapm controls */ -+static int wm8980_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8980_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8980_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* Left Output Mixer */ -+static const snd_kcontrol_new_t wm8980_left_mixer_controls[] = { -+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8980_OUTPUT, 6, 1, 1), -+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8980_MIXL, 0, 1, 1), -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8980_MIXL, 1, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", WM8980_MIXL, 5, 1, 0), -+}; -+ -+/* Right Output Mixer */ -+static const snd_kcontrol_new_t wm8980_right_mixer_controls[] = { -+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8980_OUTPUT, 5, 1, 1), -+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8980_MIXR, 0, 1, 1), -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8980_MIXR, 1, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", WM8980_MIXR, 5, 1, 0), -+}; -+ -+/* Left AUX Input boost vol */ -+static const snd_kcontrol_new_t wm8980_laux_boost_controls = -+SOC_DAPM_SINGLE("Left Aux Volume", WM8980_ADCBOOSTL, 0, 3, 0); -+ -+/* Right AUX Input boost vol */ -+static const snd_kcontrol_new_t wm8980_raux_boost_controls = -+SOC_DAPM_SINGLE("Right Aux Volume", WM8980_ADCBOOSTR, 0, 3, 0); -+ -+/* Left Input boost vol */ -+static const snd_kcontrol_new_t wm8980_lmic_boost_controls = -+SOC_DAPM_SINGLE("Left Input Volume", WM8980_ADCBOOSTL, 4, 3, 0); -+ -+/* Right Input boost vol */ -+static const snd_kcontrol_new_t wm8980_rmic_boost_controls = -+SOC_DAPM_SINGLE("Right Input Volume", WM8980_ADCBOOSTR, 4, 3, 0); -+ -+/* Left Aux In to PGA */ -+static const snd_kcontrol_new_t wm8980_laux_capture_boost_controls = -+SOC_DAPM_SINGLE("Left Capture Switch", WM8980_ADCBOOSTL, 8, 1, 0); -+ -+/* Right Aux In to PGA */ -+static const snd_kcontrol_new_t wm8980_raux_capture_boost_controls = -+SOC_DAPM_SINGLE("Right Capture Switch", WM8980_ADCBOOSTR, 8, 1, 0); -+ -+/* Left Input P In to PGA */ -+static const snd_kcontrol_new_t wm8980_lmicp_capture_boost_controls = -+SOC_DAPM_SINGLE("Left Input P Capture Boost Switch", WM8980_INPUT, 0, 1, 0); -+ -+/* Right Input P In to PGA */ -+static const snd_kcontrol_new_t wm8980_rmicp_capture_boost_controls = -+SOC_DAPM_SINGLE("Right Input P Capture Boost Switch", WM8980_INPUT, 4, 1, 0); -+ -+/* Left Input N In to PGA */ -+static const snd_kcontrol_new_t wm8980_lmicn_capture_boost_controls = -+SOC_DAPM_SINGLE("Left Input N Capture Boost Switch", WM8980_INPUT, 1, 1, 0); -+ -+/* Right Input N In to PGA */ -+static const snd_kcontrol_new_t wm8980_rmicn_capture_boost_controls = -+SOC_DAPM_SINGLE("Right Input N Capture Boost Switch", WM8980_INPUT, 5, 1, 0); -+ -+// TODO Widgets -+static const struct snd_soc_dapm_widget wm8980_dapm_widgets[] = { -+#if 0 -+//SND_SOC_DAPM_MUTE("Mono Mute", WM8980_MONOMIX, 6, 0), -+//SND_SOC_DAPM_MUTE("Speaker Mute", WM8980_SPKMIX, 6, 0), -+ -+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8980_POWER3, 2, 0, -+ &wm8980_speaker_mixer_controls[0], -+ ARRAY_SIZE(wm8980_speaker_mixer_controls)), -+SND_SOC_DAPM_MIXER("Mono Mixer", WM8980_POWER3, 3, 0, -+ &wm8980_mono_mixer_controls[0], -+ ARRAY_SIZE(wm8980_mono_mixer_controls)), -+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8980_POWER3, 0, 0), -+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8980_POWER3, 0, 0), -+SND_SOC_DAPM_PGA("Aux Input", WM8980_POWER1, 6, 0, NULL, 0), -+SND_SOC_DAPM_PGA("SpkN Out", WM8980_POWER3, 5, 0, NULL, 0), -+SND_SOC_DAPM_PGA("SpkP Out", WM8980_POWER3, 6, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mono Out", WM8980_POWER3, 7, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mic PGA", WM8980_POWER2, 2, 0, NULL, 0), -+ -+SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, -+ &wm8980_aux_boost_controls, 1), -+SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, -+ &wm8980_mic_boost_controls, 1), -+SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, -+ &wm8980_capture_boost_controls), -+ -+SND_SOC_DAPM_MIXER("Boost Mixer", WM8980_POWER2, 4, 0, NULL, 0), -+ -+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8980_POWER1, 4, 0), -+ -+SND_SOC_DAPM_INPUT("MICN"), -+SND_SOC_DAPM_INPUT("MICP"), -+SND_SOC_DAPM_INPUT("AUX"), -+SND_SOC_DAPM_OUTPUT("MONOOUT"), -+SND_SOC_DAPM_OUTPUT("SPKOUTP"), -+SND_SOC_DAPM_OUTPUT("SPKOUTN"), -+#endif -+}; -+ -+static const char *audio_map[][3] = { -+ /* Mono output mixer */ -+ {"Mono Mixer", "PCM Playback Switch", "DAC"}, -+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, -+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, -+ -+ /* Speaker output mixer */ -+ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, -+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, -+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, -+ -+ /* Outputs */ -+ {"Mono Out", NULL, "Mono Mixer"}, -+ {"MONOOUT", NULL, "Mono Out"}, -+ {"SpkN Out", NULL, "Speaker Mixer"}, -+ {"SpkP Out", NULL, "Speaker Mixer"}, -+ {"SPKOUTN", NULL, "SpkN Out"}, -+ {"SPKOUTP", NULL, "SpkP Out"}, -+ -+ /* Boost Mixer */ -+ {"Boost Mixer", NULL, "ADC"}, -+ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, -+ {"Aux Boost", "Aux Volume", "Boost Mixer"}, -+ {"Capture Boost", "Capture Switch", "Boost Mixer"}, -+ {"Mic Boost", "Mic Volume", "Boost Mixer"}, -+ -+ /* Inputs */ -+ {"MICP", NULL, "Mic Boost"}, -+ {"MICN", NULL, "Mic PGA"}, -+ {"Mic PGA", NULL, "Capture Boost"}, -+ {"AUX", NULL, "Aux Input"}, -+ -+ /* */ -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8980_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8980_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8980_dapm_widgets[i]); -+ } -+ -+ /* set up audio path map */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], -+ audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+struct pll_ { -+ unsigned int in_hz, out_hz; -+ unsigned int pre:4; /* prescale - 1 */ -+ unsigned int n:4; -+ unsigned int k; -+}; -+ -+struct pll_ pll[] = { -+ {12000000, 11289600, 0, 7, 0x86c220}, -+ {12000000, 12288000, 0, 8, 0x3126e8}, -+ {13000000, 11289600, 0, 6, 0xf28bd4}, -+ {13000000, 12288000, 0, 7, 0x8fd525}, -+ {12288000, 11289600, 0, 7, 0x59999a}, -+ {11289600, 12288000, 0, 8, 0x80dee9}, -+ /* TODO: liam - add more entries */ -+}; -+ -+static int set_pll(struct snd_soc_codec *codec, unsigned int in, -+ unsigned int out) -+{ -+ int i; -+ u16 reg; -+ -+ if(out == 0) { -+ reg = wm8980_read_reg_cache(codec, WM8980_POWER1); -+ wm8980_write(codec, WM8980_POWER1, reg & 0x1df); -+ return 0; -+ } -+ -+ for(i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (in == pll[i].in_hz && out == pll[i].out_hz) { -+ wm8980_write(codec, WM8980_PLLN, (pll[i].pre << 4) | pll[i].n); -+ wm8980_write(codec, WM8980_PLLK1, pll[i].k >> 18); -+ wm8980_write(codec, WM8980_PLLK1, (pll[i].k >> 9) && 0x1ff); -+ wm8980_write(codec, WM8980_PLLK1, pll[i].k && 0x1ff); -+ reg = wm8980_read_reg_cache(codec, WM8980_POWER1); -+ wm8980_write(codec, WM8980_POWER1, reg | 0x020); -+ return 0; -+ } -+ } -+ return -EINVAL; -+} -+ -+/* mclk dividers * 2 */ -+static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24}; -+ -+/* we need 256FS to drive the DAC's and ADC's */ -+static unsigned int wm8980_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ int i, j, best_clk = info->fs * info->rate; -+ -+ /* can we run at this clk without the PLL ? */ -+ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) { -+ if ((best_clk >> 1) * mclk_div[i] == clk) { -+ dai->pll_in = 0; -+ dai->clk_div = mclk_div[i]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ -+ /* now check for PLL support */ -+ for (i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (pll[i].in_hz == clk) { -+ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) { -+ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) { -+ dai->pll_in = clk; -+ dai->pll_out = pll[i].out_hz; -+ dai->clk_div = mclk_div[j]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+} -+ -+static int wm8980_pcm_prepare(snd_pcm_substream_t *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_codec_dai *dai = rtd->codec_dai; -+ u16 iface = 0, bfs, clk = 0, adn; -+ int fs = 48000 << 7, i; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ switch (bfs) { -+ case 2: -+ clk |= 0x1 << 2; -+ break; -+ case 4: -+ clk |= 0x2 << 2; -+ break; -+ case 8: -+ clk |= 0x3 << 2; -+ break; -+ case 16: -+ clk |= 0x4 << 2; -+ break; -+ case 32: -+ clk |= 0x5 << 2; -+ break; -+ } -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ clk |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ iface |= 0x0010; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ iface |= 0x0008; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ iface |= 0x00018; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ iface |= 0x0020; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ iface |= 0x0040; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ iface |= 0x0060; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_IF: -+ iface |= 0x0180; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ iface |= 0x0100; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ iface |= 0x0080; -+ break; -+ } -+ -+ /* filter coefficient */ -+ adn = wm8980_read_reg_cache(codec, WM8980_ADD) & 0x1f1; -+ switch (rtd->codec_dai->dai_runtime.pcmrate) { -+ case SNDRV_PCM_RATE_8000: -+ adn |= 0x5 << 1; -+ fs = 8000 << 7; -+ break; -+ case SNDRV_PCM_RATE_11025: -+ adn |= 0x4 << 1; -+ fs = 11025 << 7; -+ break; -+ case SNDRV_PCM_RATE_16000: -+ adn |= 0x3 << 1; -+ fs = 16000 << 7; -+ break; -+ case SNDRV_PCM_RATE_22050: -+ adn |= 0x2 << 1; -+ fs = 22050 << 7; -+ break; -+ case SNDRV_PCM_RATE_32000: -+ adn |= 0x1 << 1; -+ fs = 32000 << 7; -+ break; -+ case SNDRV_PCM_RATE_44100: -+ fs = 44100 << 7; -+ break; -+ } -+ -+ /* do we need to enable the PLL */ -+ if(dai->pll_in) -+ set_pll(codec, dai->pll_in, dai->pll_out); -+ -+ /* divide the clock to 256 fs */ -+ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) { -+ if (dai->clk_div == mclk_div[i]) { -+ clk |= i << 5; -+ clk &= 0xff; -+ goto set; -+ } -+ } -+ -+set: -+ /* set iface */ -+ wm8980_write(codec, WM8980_IFACE, iface); -+ wm8980_write(codec, WM8980_CLOCK, clk); -+ -+ return 0; -+} -+ -+static int wm8980_hw_free(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ set_pll(codec, 0, 0); -+ return 0; -+} -+ -+static int wm8980_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8980_read_reg_cache(codec, WM8980_DAC) & 0xffbf; -+ if(mute) -+ wm8980_write(codec, WM8980_DAC, mute_reg | 0x40); -+ else -+ wm8980_write(codec, WM8980_DAC, mute_reg); -+ -+ return 0; -+} -+ -+/* TODO: liam need to make this lower power with dapm */ -+static int wm8980_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, clk and osc on, dac unmute, active */ -+ wm8980_write(codec, WM8980_POWER1, 0x1ff); -+ wm8980_write(codec, WM8980_POWER2, 0x1ff); -+ wm8980_write(codec, WM8980_POWER3, 0x1ff); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, dac mute, inactive */ -+ -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, dac mute, inactive */ -+ wm8980_write(codec, WM8980_POWER1, 0x0); -+ wm8980_write(codec, WM8980_POWER2, 0x0); -+ wm8980_write(codec, WM8980_POWER3, 0x0); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8980_dai = { -+ .name = "WM8980 HiFi", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .config_sysclk = wm8980_config_sysclk, -+ .digital_mute = wm8980_mute, -+ .ops = { -+ .prepare = wm8980_pcm_prepare, -+ .hw_free = wm8980_hw_free, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8980_modes), -+ .mode = wm8980_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8980_dai); -+ -+static int wm8980_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8980_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8980_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8980_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the WM8980 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8980_init(struct snd_soc_device* socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int ret = 0; -+ -+ codec->name = "WM8980"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8980_read_reg_cache; -+ codec->write = wm8980_write; -+ codec->dapm_event = wm8980_dapm_event; -+ codec->dai = &wm8980_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(wm8980_reg); -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8980_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8980_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8980_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8980_reg); -+ -+ wm8980_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if(ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8980_add_controls(codec); -+ wm8980_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if(ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *wm8980_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8980 2 wire address is 0x1a -+ */ -+#define I2C_DRIVERID_WM8980 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8980_i2c_driver; -+static struct i2c_client client_template; -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+ -+static int wm8980_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8980_socdev; -+ struct wm8980_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL){ -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ -+ i2c_set_clientdata(i2c, codec); -+ -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if(ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8980_init(socdev); -+ if(ret < 0) { -+ err("failed to initialise WM8980\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+ -+} -+ -+static int wm8980_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec *codec = i2c_get_clientdata(client); -+ -+ i2c_detach_client(client); -+ -+ kfree(codec->reg_cache); -+ kfree(client); -+ -+ return 0; -+} -+ -+static int wm8980_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8980_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8980_i2c_driver = { -+ .driver = { -+ .name = "WM8980 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8980, -+ .attach_adapter = wm8980_i2c_attach, -+ .detach_client = wm8980_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8980", -+ .driver = &wm8980_i2c_driver, -+}; -+#endif -+ -+static int wm8980_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8980_setup_data *setup; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ info("WM8980 Audio Codec %s", WM8980_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ wm8980_socdev = socdev; -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8980_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8980_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8980_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8980 = { -+ .probe = wm8980_probe, -+ .remove = wm8980_remove, -+ .suspend = wm8980_suspend, -+ .resume = wm8980_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8980); -+ -+MODULE_DESCRIPTION("ASoC WM8980 driver"); -+MODULE_AUTHOR("Mike Arthur"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8980.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8980.h -@@ -0,0 +1,77 @@ -+/* -+ * wm8980.h -- WM8980 Soc Audio driver -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _WM8980_H -+#define _WM8980_H -+ -+/* WM8980 register space */ -+ -+#define WM8980_RESET 0x0 -+#define WM8980_POWER1 0x1 -+#define WM8980_POWER2 0x2 -+#define WM8980_POWER3 0x3 -+#define WM8980_IFACE 0x4 -+#define WM8980_COMP 0x5 -+#define WM8980_CLOCK 0x6 -+#define WM8980_ADD 0x7 -+#define WM8980_GPIO 0x8 -+#define WM8980_JACK1 0x9 -+#define WM8980_DAC 0xa -+#define WM8980_DACVOLL 0xb -+#define WM8980_DACVOLR 0xc -+#define WM8980_JACK2 0xd -+#define WM8980_ADC 0xe -+#define WM8980_ADCVOLL 0xf -+#define WM8980_ADCVOLR 0x10 -+#define WM8980_EQ1 0x12 -+#define WM8980_EQ2 0x13 -+#define WM8980_EQ3 0x14 -+#define WM8980_EQ4 0x15 -+#define WM8980_EQ5 0x16 -+#define WM8980_DACLIM1 0x18 -+#define WM8980_DACLIM2 0x19 -+#define WM8980_NOTCH1 0x1b -+#define WM8980_NOTCH2 0x1c -+#define WM8980_NOTCH3 0x1d -+#define WM8980_NOTCH4 0x1e -+#define WM8980_ALC1 0x20 -+#define WM8980_ALC2 0x21 -+#define WM8980_ALC3 0x22 -+#define WM8980_NGATE 0x23 -+#define WM8980_PLLN 0x24 -+#define WM8980_PLLK1 0x25 -+#define WM8980_PLLK2 0x26 -+#define WM8980_PLLK3 0x27 -+#define WM8980_VIDEO 0x28 -+#define WM8980_3D 0x29 -+#define WM8980_BEEP 0x2b -+#define WM8980_INPUT 0x2c -+#define WM8980_INPPGAL 0x2d -+#define WM8980_INPPGAR 0x2e -+#define WM8980_ADCBOOSTL 0x2f -+#define WM8980_ADCBOOSTR 0x30 -+#define WM8980_OUTPUT 0x31 -+#define WM8980_MIXL 0x32 -+#define WM8980_MIXR 0x33 -+#define WM8980_HPVOLL 0x34 -+#define WM8980_HPVOLR 0x35 -+#define WM8980_SPKVOLL 0x36 -+#define WM8980_SPKVOLR 0x37 -+#define WM8980_OUT3MIX 0x38 -+#define WM8980_MONOMIX 0x39 -+ -+#define WM8980_CACHEREGNUM 58 -+ -+struct wm8980_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai wm8980_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8980; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/at91/eti_b1_wm8731.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/at91/eti_b1_wm8731.c -@@ -0,0 +1,230 @@ -+/* -+ * eti_b1_wm8731 -- SoC audio for Endrelia ETI_B1. -+ * -+ * Author: Frank Mandarino <fmandarino@endrelia.com> -+ * Endrelia Technologies Inc. -+ * Created: Mar 29, 2006 -+ * -+ * Based on corgi.c by: -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Copyright 2005 Openedhand Ltd. -+ * -+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> -+ * Richard Purdie <richard@openedhand.com> -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 30th Nov 2005 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/version.h> -+#include <linux/kernel.h> -+#include <linux/clk.h> -+#include <linux/timer.h> -+#include <linux/interrupt.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+ -+#include <asm/arch/at91rm9200.h> -+#include <asm/arch/gpio.h> -+#include <asm/arch/hardware.h> -+ -+#include "../codecs/wm8731.h" -+#include "at91rm9200-pcm.h" -+ -+#if 0 -+#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731:" x) -+#else -+#define DBG(x...) -+#endif -+ -+static struct clk *pck1_clk; -+static struct clk *pllb_clk; -+ -+static int eti_b1_startup(snd_pcm_substream_t *substream) -+{ -+ /* Start PCK1 clock. */ -+ clk_enable(pck1_clk); -+ DBG("pck1 started\n"); -+ -+ return 0; -+} -+ -+static void eti_b1_shutdown(snd_pcm_substream_t *substream) -+{ -+ /* Stop PCK1 clock. */ -+ clk_disable(pck1_clk); -+ DBG("pck1 stopped\n"); -+} -+ -+static struct snd_soc_ops eti_b1_ops = { -+ .startup = eti_b1_startup, -+ .shutdown = eti_b1_shutdown, -+}; -+ -+ -+static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { -+ SND_SOC_DAPM_MIC("Int Mic", NULL), -+ SND_SOC_DAPM_SPK("Ext Spk", NULL), -+}; -+ -+static const char *intercon[][3] = { -+ -+ /* speaker connected to LHPOUT */ -+ {"Ext Spk", NULL, "LHPOUT"}, -+ -+ /* mic is connected to Mic Jack, with WM8731 Mic Bias */ -+ {"MICIN", NULL, "Mic Bias"}, -+ {"Mic Bias", NULL, "Int Mic"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+/* -+ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. -+ */ -+static int eti_b1_wm8731_init(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ DBG("eti_b1_wm8731_init() called\n"); -+ -+ /* Add specific widgets */ -+ for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]); -+ } -+ -+ /* Set up specific audio path interconnects */ -+ for(i = 0; intercon[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, intercon[i][0], -+ intercon[i][1], intercon[i][2]); -+ } -+ -+ /* not connected */ -+ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); -+ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); -+ -+ /* always connected */ -+ snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); -+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); -+ -+ snd_soc_dapm_sync_endpoints(codec); -+ -+ return 0; -+} -+ -+unsigned int eti_b1_config_sysclk(struct snd_soc_pcm_runtime *rtd, -+ struct snd_soc_clock_info *info) -+{ -+ if(info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) { -+ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12000000); -+ } -+ return 0; -+} -+ -+static struct snd_soc_dai_link eti_b1_dai = { -+ .name = "WM8731", -+ .stream_name = "WM8731", -+ .cpu_dai = &at91rm9200_i2s_dai[1], -+ .codec_dai = &wm8731_dai, -+ .init = eti_b1_wm8731_init, -+ .config_sysclk = eti_b1_config_sysclk, -+}; -+ -+static struct snd_soc_machine snd_soc_machine_eti_b1 = { -+ .name = "ETI_B1", -+ .dai_link = &eti_b1_dai, -+ .num_links = 1, -+ .ops = &eti_b1_ops, -+}; -+ -+static struct wm8731_setup_data eti_b1_wm8731_setup = { -+ .i2c_address = 0x1a, -+}; -+ -+static struct snd_soc_device eti_b1_snd_devdata = { -+ .machine = &snd_soc_machine_eti_b1, -+ .platform = &at91rm9200_soc_platform, -+ .codec_dev = &soc_codec_dev_wm8731, -+ .codec_data = &eti_b1_wm8731_setup, -+}; -+ -+static struct platform_device *eti_b1_snd_device; -+ -+static int __init eti_b1_init(void) -+{ -+ int ret; -+ u32 ssc_pio_lines; -+ -+ eti_b1_snd_device = platform_device_alloc("soc-audio", -1); -+ if (!eti_b1_snd_device) -+ return -ENOMEM; -+ -+ platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); -+ eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; -+ -+ ret = platform_device_add(eti_b1_snd_device); -+ if (ret) { -+ platform_device_put(eti_b1_snd_device); -+ return ret; -+ } -+ -+ ssc_pio_lines = AT91_PB6_TF1 | AT91_PB7_TK1 | AT91_PB8_TD1 -+ | AT91_PB9_RD1 /* | AT91_PB10_RK1 | AT91_PB11_RF1 */; -+ -+ /* Reset all PIO registers and assign lines to peripheral A */ -+ at91_sys_write(AT91_PIOB + PIO_PDR, ssc_pio_lines); -+ at91_sys_write(AT91_PIOB + PIO_ODR, ssc_pio_lines); -+ at91_sys_write(AT91_PIOB + PIO_IFDR, ssc_pio_lines); -+ at91_sys_write(AT91_PIOB + PIO_CODR, ssc_pio_lines); -+ at91_sys_write(AT91_PIOB + PIO_IDR, ssc_pio_lines); -+ at91_sys_write(AT91_PIOB + PIO_MDDR, ssc_pio_lines); -+ at91_sys_write(AT91_PIOB + PIO_PUDR, ssc_pio_lines); -+ at91_sys_write(AT91_PIOB + PIO_ASR, ssc_pio_lines); -+ at91_sys_write(AT91_PIOB + PIO_OWDR, ssc_pio_lines); -+ -+ /* -+ * Set PCK1 parent to PLLB and its rate to 12 Mhz. -+ */ -+ pllb_clk = clk_get(NULL, "pllb"); -+ pck1_clk = clk_get(NULL, "pck1"); -+ -+ clk_set_parent(pck1_clk, pllb_clk); -+ clk_set_rate(pck1_clk, 12000000); -+ -+ DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); -+ -+ /* assign the GPIO pin to PCK1 */ -+ at91_set_B_periph(AT91_PIN_PA24, 0); -+ -+ return ret; -+} -+ -+static void __exit eti_b1_exit(void) -+{ -+ clk_put(pck1_clk); -+ clk_put(pllb_clk); -+ -+ platform_device_unregister(eti_b1_snd_device); -+} -+ -+module_init(eti_b1_init); -+module_exit(eti_b1_exit); -+ -+/* Module information */ -+MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); -+MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8510.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8510.c -@@ -0,0 +1,895 @@ -+/* -+ * wm8510.c -- WM8510 ALSA Soc Audio driver -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * -+ * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/version.h> -+#include <linux/kernel.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8510.h" -+ -+#define AUDIO_NAME "wm8510" -+#define WM8510_VERSION "0.5" -+ -+/* -+ * Debug -+ */ -+ -+#define WM8510_DEBUG 0 -+ -+#ifdef WM8510_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+struct snd_soc_codec_device soc_codec_dev_wm8510; -+ -+/* -+ * wm8510 register cache -+ * We can't read the WM8510 register space when we are -+ * using 2 wire for device control, so we cache them instead. -+ */ -+static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0050, 0x0000, 0x0140, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x00ff, -+ 0x0000, 0x0000, 0x0100, 0x00ff, -+ 0x0000, 0x0000, 0x012c, 0x002c, -+ 0x002c, 0x002c, 0x002c, 0x0000, -+ 0x0032, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0038, 0x000b, 0x0032, 0x0000, -+ 0x0008, 0x000c, 0x0093, 0x00e9, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0003, 0x0010, 0x0000, 0x0000, -+ 0x0000, 0x0002, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0039, 0x0000, -+ 0x0000, -+}; -+ -+#define WM8510_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ -+ SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8510_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8510_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+#define WM8794_BCLK \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\ -+ SND_SOC_FSBD(16) | SND_SOC_FSBD(32)) -+ -+#define WM8794_HIFI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -+ -+static struct snd_soc_dai_mode wm8510_modes[] = { -+ /* codec frame and clock master modes */ -+ { -+ .fmt = WM8510_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8794_HIFI_BITS, -+ .pcmrate = WM8510_RATES, -+ .pcmdir = WM8510_DIR, -+ .fs = 256, -+ .bfs = WM8794_BCLK, -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8510_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8794_HIFI_BITS, -+ .pcmrate = WM8510_RATES, -+ .pcmdir = WM8510_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8510 register cache -+ */ -+static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec * codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg == WM8510_RESET) -+ return 0; -+ if (reg >= WM8510_CACHEREGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write wm8510 register cache -+ */ -+static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, -+ u16 reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= WM8510_CACHEREGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+/* -+ * write to the WM8510 register space -+ */ -+static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8510 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8510_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -EIO; -+} -+ -+#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) -+ -+static const char *wm8510_companding[] = {"Off", "NC", "u-law", "A-law" }; -+static const char *wm8510_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; -+static const char *wm8510_alc[] = {"ALC", "Limiter" }; -+ -+static const struct soc_enum wm8510_enum[] = { -+ SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */ -+ SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */ -+ SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp), -+ SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc), -+}; -+ -+static const struct snd_kcontrol_new wm8510_snd_controls[] = { -+ -+SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0), -+ -+SOC_ENUM("DAC Companding", wm8510_enum[1]), -+SOC_ENUM("ADC Companding", wm8510_enum[0]), -+ -+SOC_ENUM("Playback De-emphasis", wm8510_enum[2]), -+SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0), -+ -+SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0), -+ -+SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0), -+SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0), -+SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0), -+ -+SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0), -+ -+SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0), -+SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0), -+SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0), -+ -+SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0), -+SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0), -+ -+SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0), -+SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0), -+SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0), -+ -+SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0), -+SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0), -+SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0), -+ -+SOC_ENUM("ALC Capture Mode", wm8510_enum[3]), -+SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0), -+SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0), -+ -+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0), -+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0), -+ -+SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0), -+SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0), -+ -+SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0), -+SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1), -+SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0), -+ -+SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), -+SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 0), -+}; -+ -+/* add non dapm controls */ -+static int wm8510_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8510_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* Speaker Output Mixer */ -+static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), -+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 1), -+}; -+ -+/* Mono Output Mixer */ -+static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), -+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 1), -+}; -+ -+/* AUX Input boost vol */ -+static const struct snd_kcontrol_new wm8510_aux_boost_controls = -+SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0); -+ -+/* Mic Input boost vol */ -+static const struct snd_kcontrol_new wm8510_mic_boost_controls = -+SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0); -+ -+/* Capture boost switch */ -+static const struct snd_kcontrol_new wm8510_capture_boost_controls = -+SOC_DAPM_SINGLE("Capture Boost Switch", WM8510_INPPGA, 6, 1, 0); -+ -+/* Aux In to PGA */ -+static const struct snd_kcontrol_new wm8510_aux_capture_boost_controls = -+SOC_DAPM_SINGLE("Aux Capture Boost Switch", WM8510_INPPGA, 2, 1, 0); -+ -+/* Mic P In to PGA */ -+static const struct snd_kcontrol_new wm8510_micp_capture_boost_controls = -+SOC_DAPM_SINGLE("Mic P Capture Boost Switch", WM8510_INPPGA, 0, 1, 0); -+ -+/* Mic N In to PGA */ -+static const struct snd_kcontrol_new wm8510_micn_capture_boost_controls = -+SOC_DAPM_SINGLE("Mic N Capture Boost Switch", WM8510_INPPGA, 1, 1, 0); -+ -+static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = { -+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0, -+ &wm8510_speaker_mixer_controls[0], -+ ARRAY_SIZE(wm8510_speaker_mixer_controls)), -+SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0, -+ &wm8510_mono_mixer_controls[0], -+ ARRAY_SIZE(wm8510_mono_mixer_controls)), -+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), -+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER3, 0, 0), -+SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), -+SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), -+SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, NULL, 0), -+ -+SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, -+ &wm8510_aux_boost_controls, 1), -+SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, -+ &wm8510_mic_boost_controls, 1), -+SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, -+ &wm8510_capture_boost_controls), -+ -+SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, NULL, 0), -+ -+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0), -+ -+SND_SOC_DAPM_INPUT("MICN"), -+SND_SOC_DAPM_INPUT("MICP"), -+SND_SOC_DAPM_INPUT("AUX"), -+SND_SOC_DAPM_OUTPUT("MONOOUT"), -+SND_SOC_DAPM_OUTPUT("SPKOUTP"), -+SND_SOC_DAPM_OUTPUT("SPKOUTN"), -+}; -+ -+static const char *audio_map[][3] = { -+ /* Mono output mixer */ -+ {"Mono Mixer", "PCM Playback Switch", "DAC"}, -+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, -+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, -+ -+ /* Speaker output mixer */ -+ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, -+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, -+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, -+ -+ /* Outputs */ -+ {"Mono Out", NULL, "Mono Mixer"}, -+ {"MONOOUT", NULL, "Mono Out"}, -+ {"SpkN Out", NULL, "Speaker Mixer"}, -+ {"SpkP Out", NULL, "Speaker Mixer"}, -+ {"SPKOUTN", NULL, "SpkN Out"}, -+ {"SPKOUTP", NULL, "SpkP Out"}, -+ -+ /* Boost Mixer */ -+ {"Boost Mixer", NULL, "ADC"}, -+ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, -+ {"Aux Boost", "Aux Volume", "Boost Mixer"}, -+ {"Capture Boost", "Capture Switch", "Boost Mixer"}, -+ {"Mic Boost", "Mic Volume", "Boost Mixer"}, -+ -+ /* Inputs */ -+ {"MICP", NULL, "Mic Boost"}, -+ {"MICN", NULL, "Mic PGA"}, -+ {"Mic PGA", NULL, "Capture Boost"}, -+ {"AUX", NULL, "Aux Input"}, -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8510_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8510_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8510_dapm_widgets[i]); -+ } -+ -+ /* set up audio path audio_mapnects */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], -+ audio_map[i][1], audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+struct pll_ { -+ unsigned int in_hz, out_hz; -+ unsigned int pre:4; /* prescale - 1 */ -+ unsigned int n:4; -+ unsigned int k; -+}; -+ -+struct pll_ pll[] = { -+ {12000000, 11289600, 0, 7, 0x86c220}, -+ {12000000, 12288000, 0, 8, 0x3126e8}, -+ {13000000, 11289600, 0, 6, 0xf28bd4}, -+ {13000000, 12288000, 0, 7, 0x8fd525}, -+ {12288000, 11289600, 0, 7, 0x59999a}, -+ {11289600, 12288000, 0, 8, 0x80dee9}, -+ /* liam - add more entries */ -+}; -+ -+static int set_pll(struct snd_soc_codec *codec, unsigned int in, -+ unsigned int out) -+{ -+ int i; -+ u16 reg; -+ -+ if(out == 0) { -+ reg = wm8510_read_reg_cache(codec, WM8510_POWER1); -+ wm8510_write(codec, WM8510_POWER1, reg & 0x1df); -+ return 0; -+ } -+ -+ for(i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (in == pll[i].in_hz && out == pll[i].out_hz) { -+ wm8510_write(codec, WM8510_PLLN, (pll[i].pre << 4) | pll[i].n); -+ wm8510_write(codec, WM8510_PLLK1, pll[i].k >> 18); -+ wm8510_write(codec, WM8510_PLLK1, (pll[i].k >> 9) && 0x1ff); -+ wm8510_write(codec, WM8510_PLLK1, pll[i].k && 0x1ff); -+ reg = wm8510_read_reg_cache(codec, WM8510_POWER1); -+ wm8510_write(codec, WM8510_POWER1, reg | 0x020); -+ return 0; -+ } -+ } -+ return -EINVAL; -+} -+ -+/* mclk dividers * 2 */ -+static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24}; -+ -+/* we need 256FS to drive the DAC's and ADC's */ -+static unsigned int wm8510_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ int i, j, best_clk = info->fs * info->rate; -+ -+ /* can we run at this clk without the PLL ? */ -+ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) { -+ if ((best_clk >> 1) * mclk_div[i] == clk) { -+ dai->pll_in = 0; -+ dai->clk_div = mclk_div[i]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ -+ /* now check for PLL support */ -+ for (i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (pll[i].in_hz == clk) { -+ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) { -+ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) { -+ dai->pll_in = clk; -+ dai->pll_out = pll[i].out_hz; -+ dai->clk_div = mclk_div[j]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+} -+ -+static int wm8510_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_codec_dai *dai = rtd->codec_dai; -+ u16 iface = 0, bfs, clk = 0, adn; -+ int fs = 48000 << 7, i; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ switch (bfs) { -+ case 2: -+ clk |= 0x1 << 2; -+ break; -+ case 4: -+ clk |= 0x2 << 2; -+ break; -+ case 8: -+ clk |= 0x3 << 2; -+ break; -+ case 16: -+ clk |= 0x4 << 2; -+ break; -+ case 32: -+ clk |= 0x5 << 2; -+ break; -+ } -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ clk |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ iface |= 0x0010; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ iface |= 0x0008; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ iface |= 0x00018; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ iface |= 0x0020; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ iface |= 0x0040; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ iface |= 0x0060; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_IF: -+ iface |= 0x0180; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ iface |= 0x0100; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ iface |= 0x0080; -+ break; -+ } -+ -+ /* filter coefficient */ -+ adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; -+ switch (rtd->codec_dai->dai_runtime.pcmrate) { -+ case SNDRV_PCM_RATE_8000: -+ adn |= 0x5 << 1; -+ fs = 8000 << 7; -+ break; -+ case SNDRV_PCM_RATE_11025: -+ adn |= 0x4 << 1; -+ fs = 11025 << 7; -+ break; -+ case SNDRV_PCM_RATE_16000: -+ adn |= 0x3 << 1; -+ fs = 16000 << 7; -+ break; -+ case SNDRV_PCM_RATE_22050: -+ adn |= 0x2 << 1; -+ fs = 22050 << 7; -+ break; -+ case SNDRV_PCM_RATE_32000: -+ adn |= 0x1 << 1; -+ fs = 32000 << 7; -+ break; -+ case SNDRV_PCM_RATE_44100: -+ fs = 44100 << 7; -+ break; -+ } -+ -+ /* do we need to enable the PLL */ -+ if(dai->pll_in) -+ set_pll(codec, dai->pll_in, dai->pll_out); -+ -+ /* divide the clock to 256 fs */ -+ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) { -+ if (dai->clk_div == mclk_div[i]) { -+ clk |= i << 5; -+ clk &= 0xff; -+ goto set; -+ } -+ } -+ -+set: -+ /* set iface */ -+ wm8510_write(codec, WM8510_IFACE, iface); -+ wm8510_write(codec, WM8510_CLOCK, clk); -+ -+ return 0; -+} -+ -+static int wm8510_hw_free(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ set_pll(codec, 0, 0); -+ return 0; -+} -+ -+static int wm8510_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; -+ if(mute) -+ wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); -+ else -+ wm8510_write(codec, WM8510_DAC, mute_reg); -+ return 0; -+} -+ -+/* liam need to make this lower power with dapm */ -+static int wm8510_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, clk and osc on, dac unmute, active */ -+ wm8510_write(codec, WM8510_POWER1, 0x1ff); -+ wm8510_write(codec, WM8510_POWER2, 0x1ff); -+ wm8510_write(codec, WM8510_POWER3, 0x1ff); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, dac mute, inactive */ -+ -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, dac mute, inactive */ -+ wm8510_write(codec, WM8510_POWER1, 0x0); -+ wm8510_write(codec, WM8510_POWER2, 0x0); -+ wm8510_write(codec, WM8510_POWER3, 0x0); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8510_dai = { -+ .name = "WM8510 HiFi", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 1, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 1, -+ }, -+ .config_sysclk = wm8510_config_sysclk, -+ .digital_mute = wm8510_mute, -+ .ops = { -+ .prepare = wm8510_pcm_prepare, -+ .hw_free = wm8510_hw_free, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8510_modes), -+ .mode = wm8510_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8510_dai); -+ -+static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8510_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8510_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the WM8510 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8510_init(struct snd_soc_device *socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int ret = 0; -+ -+ codec->name = "WM8510"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8510_read_reg_cache; -+ codec->write = wm8510_write; -+ codec->dapm_event = wm8510_dapm_event; -+ codec->dai = &wm8510_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(wm8510_reg); -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8510_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8510_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8510_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8510_reg); -+ -+ wm8510_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if(ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8510_add_controls(codec); -+ wm8510_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if(ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *wm8510_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8510 2 wire address is 0x1a -+ */ -+#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8510_i2c_driver; -+static struct i2c_client client_template; -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+ -+static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8510_socdev; -+ struct wm8510_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL){ -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ i2c_set_clientdata(i2c, codec); -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if(ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8510_init(socdev); -+ if(ret < 0) { -+ err("failed to initialise WM8510\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+} -+ -+static int wm8510_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec *codec = i2c_get_clientdata(client); -+ i2c_detach_client(client); -+ kfree(codec->reg_cache); -+ kfree(client); -+ return 0; -+} -+ -+static int wm8510_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8510_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8510_i2c_driver = { -+ .driver = { -+ .name = "WM8510 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8510, -+ .attach_adapter = wm8510_i2c_attach, -+ .detach_client = wm8510_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8510", -+ .driver = &wm8510_i2c_driver, -+}; -+#endif -+ -+static int wm8510_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8510_setup_data *setup; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ info("WM8510 Audio Codec %s", WM8510_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ wm8510_socdev = socdev; -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8510_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8510_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8510_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8510 = { -+ .probe = wm8510_probe, -+ .remove = wm8510_remove, -+ .suspend = wm8510_suspend, -+ .resume = wm8510_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); -+ -+MODULE_DESCRIPTION("ASoC WM8510 driver"); -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8510.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8510.h -@@ -0,0 +1,64 @@ -+/* -+ * wm8510.h -- WM8510 Soc Audio driver -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _WM8510_H -+#define _WM8510_H -+ -+/* WM8510 register space */ -+ -+#define WM8510_RESET 0x0 -+#define WM8510_POWER1 0x1 -+#define WM8510_POWER2 0x2 -+#define WM8510_POWER3 0x3 -+#define WM8510_IFACE 0x4 -+#define WM8510_COMP 0x5 -+#define WM8510_CLOCK 0x6 -+#define WM8510_ADD 0x7 -+#define WM8510_GPIO 0x8 -+#define WM8510_DAC 0xa -+#define WM8510_DACVOL 0xb -+#define WM8510_ADC 0xe -+#define WM8510_ADCVOL 0xf -+#define WM8510_EQ1 0x12 -+#define WM8510_EQ2 0x13 -+#define WM8510_EQ3 0x14 -+#define WM8510_EQ4 0x15 -+#define WM8510_EQ5 0x16 -+#define WM8510_DACLIM1 0x18 -+#define WM8510_DACLIM2 0x19 -+#define WM8510_NOTCH1 0x1b -+#define WM8510_NOTCH2 0x1c -+#define WM8510_NOTCH3 0x1d -+#define WM8510_NOTCH4 0x1e -+#define WM8510_ALC1 0x20 -+#define WM8510_ALC2 0x21 -+#define WM8510_ALC3 0x22 -+#define WM8510_NGATE 0x23 -+#define WM8510_PLLN 0x24 -+#define WM8510_PLLK1 0x25 -+#define WM8510_PLLK2 0x26 -+#define WM8510_PLLK3 0x27 -+#define WM8510_ATTEN 0x28 -+#define WM8510_INPUT 0x2c -+#define WM8510_INPPGA 0x2d -+#define WM8510_ADCBOOST 0x2f -+#define WM8510_OUTPUT 0x31 -+#define WM8510_SPKMIX 0x32 -+#define WM8510_SPKVOL 0x36 -+#define WM8510_MONOMIX 0x38 -+ -+#define WM8510_CACHEREGNUM 57 -+ -+struct wm8510_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai wm8510_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8510; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/imx/imx-ac97.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/imx-ac97.c -@@ -0,0 +1,281 @@ -+/* -+ * imx-ssi.c -- SSI driver for Freescale IMX -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Based on mxc-alsa-mc13783 (C) 2006 Freescale. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 29th Aug 2006 Initial version. -+ * -+ */ -+ -+#define IMX_AC97_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ -+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+/* may need to expand this */ -+static struct snd_soc_dai_mode imx_ssi_ac97_modes[] = { -+ {0, 0, SNDRV_PCM_FMTBIT_S16_LE, IMX_AC97_RATES}, -+ {0, 0, SNDRV_PCM_FMTBIT_S18_3LE, IMX_AC97_RATES}, -+ {0, 0, SNDRV_PCM_FMTBIT_S20_3LE, IMX_AC97_RATES}, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_out = { -+ .name = "SSI1 PCM Stereo out", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = emi_2_per, -+ .watermark_level = SDMA_TXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI1_STX0, -+ .event_id = DMA_REQ_SSI1_TX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_in = { -+ .name = "SSI1 PCM Stereo in", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_RXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI1_SRX0, -+ .event_id = DMA_REQ_SSI1_RX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_out = { -+ .name = "SSI2 PCM Stereo out", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_TXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI2_STX0, -+ .event_id = DMA_REQ_SSI2_TX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_in = { -+ .name = "SSI2 PCM Stereo in", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_RXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI2_SRX0, -+ .event_id = DMA_REQ_SSI2_RX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg) -+{ -+} -+ -+static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) -+{ -+} -+ -+static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) -+{ -+} -+ -+static void imx_ssi_ac97_cold_reset(struct snd_ac97 *ac97) -+{ -+} -+ -+struct snd_ac97_bus_ops soc_ac97_ops = { -+ .read = imx_ssi_ac97_read, -+ .write = imx_ssi_ac97_write, -+ .warm_reset = imx_ssi_ac97_warm_reset, -+ .reset = imx_ssi_ac97_cold_reset, -+}; -+ -+ -+static intimx_ssi1_ac97_probe(struct platform_device *pdev) -+{ -+ int ret; -+ -+ -+ return ret; -+} -+ -+static void imx_ssi1_ac97_remove(struct platform_device *pdev) -+{ -+ /* shutdown SSI */ -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR &= ~SSI_SCR_SSIEN; -+ else -+ SSI2_SCR &= ~SSI_SCR_SSIEN; -+ } -+ -+} -+ -+static int imx_ssi1_ac97_prepare(struct snd_pcm_substream *substream) -+{ -+ // set vra -+} -+ -+static int imx_ssi_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if (!rtd->cpu_dai->active) { -+ -+ } -+ -+ return 0; -+} -+ -+static int imx_ssi1_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ SSI1_SCR |= SSI_SCR_TE; -+ SSI1_SIER |= SSI_SIER_TDMAE; -+ } else { -+ SSI1_SCR |= SSI_SCR_RE; -+ SSI1_SIER |= SSI_SIER_RDMAE; -+ } -+ SSI1_SCR |= SSI_SCR_SSIEN; -+ -+ break; -+ case SNDRV_PCM_TRIGGER_RESUME: -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSI1_SCR |= SSI_SCR_TE; -+ else -+ SSI1_SCR |= SSI_SCR_RE; -+ break -+ case SNDRV_PCM_TRIGGER_STOP: -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSI1_SCR &= ~SSI_SCR_TE; -+ else -+ SSI1_SCR &= ~SSI_SCR_RE; -+ break; -+ default: -+ ret = -EINVAL; -+ } -+ -+ return ret; -+} -+ -+static void imx_ssi_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ -+} -+ -+#ifdef CONFIG_PM -+static int imx_ssi_suspend(struct platform_device *dev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if(!dai->active) -+ return 0; -+ -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR &= ~SSI_SCR_SSIEN; -+ else -+ SSI2_SCR &= ~SSI_SCR_SSIEN; -+ -+ return 0; -+} -+ -+static int imx_ssi_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if(!dai->active) -+ return 0; -+ -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR |= SSI_SCR_SSIEN; -+ else -+ SSI2_SCR |= SSI_SCR_SSIEN; -+ -+ return 0; -+} -+ -+#else -+#define imx_ssi_suspend NULL -+#define imx_ssi_resume NULL -+#endif -+ -+static unsigned int imx_ssi_config_ac97_sysclk(struct snd_soc_cpu_dai *iface, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ return clk; -+} -+ -+struct snd_soc_cpu_dai imx_ssi_ac97_dai = { -+ .name = "imx-ac97-1", -+ .id = 0, -+ .type = SND_SOC_DAI_AC97, -+ .suspend = imx_ssi_suspend, -+ .resume = imx_ssi_resume, -+ .config_sysclk = imx_ssi_ac97_config_sysclk, -+ .playback = { -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .ops = { -+ .probe = imx_ac97_probe, -+ .remove = imx_ac97_shutdown, -+ .trigger = imx_ssi1_trigger, -+ .prepare = imx_ssi_ac97_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(imx_ssi_ac97_modes), -+ .mode = imx_ssi_ac97_modes,}, -+}, -+{ -+ .name = "imx-ac97-2", -+ .id = 1, -+ .type = SND_SOC_DAI_AC97, -+ .suspend = imx_ssi_suspend, -+ .resume = imx_ssi_resume, -+ .config_sysclk = imx_ssi_ac97_config_sysclk, -+ .playback = { -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .ops = { -+ .probe = imx_ac97_probe, -+ .remove = imx_ac97_shutdown, -+ .trigger = imx_ssi1_trigger, -+ .prepare = imx_ssi_ac97_prepare,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(imx_ssi_ac97_modes), -+ .mode = imx_ssi_ac97_modes,}, -+}; -+ -+EXPORT_SYMBOL_GPL(imx_ssi_ac97_dai); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("i.MX ASoC AC97 driver"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/imx/imx-i2s.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/imx-i2s.c -@@ -0,0 +1,473 @@ -+/* -+ * imx-ssi.c -- SSI driver for Freescale IMX -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Based on mxc-alsa-mc13783 (C) 2006 Freescale. -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 29th Aug 2006 Initial version. -+ * -+ */ -+ -+#define IMX_SSI_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J |\ -+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_DSP__A |\ -+ SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS |\ -+ SND_SOC_DAIFMT_CBM_CFS | SND_SOC_DAIFMT_CBS_CFM |\ -+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF) -+ -+#define IMX_SSI_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define IMX_SSI_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ -+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ -+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ -+ SNDRV_PCM_RATE_96000) -+ -+#define IMX_SSI_BITS \ -+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ -+ SNDRV_PCM_FMTBIT_S24_LE) -+ -+static struct snd_soc_dai_mode imx_ssi_pcm_modes[] = { -+ -+ /* frame master and clock slave mode */ -+ { -+ .fmt = IMX_SSI_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, -+ .tdm = SND_SOC_DAITDM_LRDW(0,0), -+ .pcmfmt = IMX_SSI_BITS, -+ .pcmrate = IMX_SSI_RATES, -+ .pcmdir = IMX_SSI_DIR, -+ .flags = SND_SOC_DAI_BFS_RCW, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSBW(1) | SND_SOC_FSBW(2), -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_out = { -+ .name = "SSI1 PCM Stereo out", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = emi_2_per, -+ .watermark_level = SDMA_TXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI1_STX0, -+ .event_id = DMA_REQ_SSI1_TX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_in = { -+ .name = "SSI1 PCM Stereo in", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_RXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI1_SRX0, -+ .event_id = DMA_REQ_SSI1_RX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_out = { -+ .name = "SSI2 PCM Stereo out", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_TXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI2_STX0, -+ .event_id = DMA_REQ_SSI2_TX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_in = { -+ .name = "SSI2 PCM Stereo in", -+ .params = { -+ .bd_number = 1, -+ .transfer_type = per_2_emi, -+ .watermark_level = SDMA_RXFIFO_WATERMARK, -+ .word_size = TRANSFER_16BIT, // maybe add this in setup func -+ .per_address = SSI2_SRX0, -+ .event_id = DMA_REQ_SSI2_RX1, -+ .peripheral_type = SSI, -+ }, -+}; -+ -+ -+static int imx_ssi_startup(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ if (!rtd->cpu_dai->active) { -+ -+ } -+ -+ return 0; -+} -+ -+static int imx_ssi1_hw_tx_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 bfs, div; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); -+ -+ SSI1_STCR = 0; -+ SSI1_STCCR = 0; -+ -+ /* DAI mode */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ SSI1_STCR |= SSI_STCR_TSCKP | SSI_STCR_TFSI | -+ SSI_STCR_TEFS | SSI_STCR_TXBIT0; -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ SSI1_STCR |= SSI_STCR_TSCKP | SSI_STCR_TFSI | SSI_STCR_TXBIT0; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ SSI1_STCR |= SSI_STCR_TEFS; // data 1 bit after sync -+ case SND_SOC_DAIFMT_DSP_A: -+ SSI1_STCR |= SSI_STCR_TFSL; // frame is 1 bclk long -+ -+ /* DAI clock inversion */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_IB_IF: -+ SSI1_STCR |= SSI_STCR_TFSI | SSI_STCR_TSCKP; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ SSI1_STCR |= SSI_STCR_TSCKP; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ SSI1_STCR |= SSI_STCR_TFSI; -+ break; -+ } -+ break; -+ } -+ -+ /* DAI data (word) size */ -+ switch(rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ SSI1_STCCR |= SSI_STCCR_WL(16); -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ SSI1_STCCR |= SSI_STCCR_WL(20); -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ SSI1_STCCR |= SSI_STCCR_WL(24); -+ break; -+ } -+ -+ /* DAI clock master masks */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ -+ case SND_SOC_DAIFMT_CBM_CFM: -+ SSI1_STCR |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFM: -+ SSI1_STCR |= SSI_STCR_TFDIR; -+ break; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ SSI1_STCR |= SSI_STCR_TXDIR; -+ break; -+ } -+ -+ /* DAI BCLK ratio to SYSCLK / MCLK */ -+ /* prescaler modulus - todo */ -+ switch (bfs) { -+ case 2: -+ break; -+ case 4: -+ break; -+ case 8: -+ break; -+ case 16: -+ break; -+ } -+ -+ /* TDM - todo, only fifo 0 atm */ -+ SSI1_STCR |= SSI_STCR_TFEN0; -+ SSI1_STCCR |= SSI_STCCR_DC(params_channels(params)); -+ -+ return 0; -+} -+ -+static int imx_ssi1_hw_rx_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ u16 bfs, div; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); -+ -+ SSI1_SRCR = 0; -+ SSI1_SRCCR = 0; -+ -+ /* DAI mode */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ SSI1_SRCR |= SSI_SRCR_RSCKP | SSI_SRCR_RFSI | -+ SSI_STCR_REFS | SSI_STCR_RXBIT0; -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ SSI1_SRCR |= SSI_SRCR_RSCKP | SSI_SRCR_RFSI | SSI_SRCR_RXBIT0; -+ break; -+ case SND_SOC_DAIFMT_DSP_B: -+ SSI1_SRCR |= SSI_SRCR_REFS; // data 1 bit after sync -+ case SND_SOC_DAIFMT_DSP_A: -+ SSI1_SRCR |= SSI_SRCR_RFSL; // frame is 1 bclk long -+ -+ /* DAI clock inversion */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_IB_IF: -+ SSI1_SRCR |= SSI_SRCR_TFSI | SSI_SRCR_TSCKP; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ SSI1_SRCR |= SSI_SRCR_RSCKP; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ SSI1_SRCR |= SSI_SRCR_RFSI; -+ break; -+ } -+ break; -+ } -+ -+ /* DAI data (word) size */ -+ switch(rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ SSI1_SRCCR |= SSI_SRCCR_WL(16); -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ SSI1_SRCCR |= SSI_SRCCR_WL(20); -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ SSI1_SRCCR |= SSI_SRCCR_WL(24); -+ break; -+ } -+ -+ /* DAI clock master masks */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ -+ case SND_SOC_DAIFMT_CBM_CFM: -+ SSI1_SRCR |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFM: -+ SSI1_SRCR |= SSI_SRCR_RFDIR; -+ break; -+ case SND_SOC_DAIFMT_CBM_CFS: -+ SSI1_SRCR |= SSI_SRCR_RXDIR; -+ break; -+ } -+ -+ /* DAI BCLK ratio to SYSCLK / MCLK */ -+ /* prescaler modulus - todo */ -+ switch (bfs) { -+ case 2: -+ break; -+ case 4: -+ break; -+ case 8: -+ break; -+ case 16: -+ break; -+ } -+ -+ /* TDM - todo, only fifo 0 atm */ -+ SSI1_SRCR |= SSI_SRCR_RFEN0; -+ SSI1_SRCCR |= SSI_SRCCR_DC(params_channels(params)); -+ -+ return 0; -+} -+ -+static int imx_ssi1_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ /* clear register if not enabled */ -+ if(!(SSI1_SCR & SSI_SCR_SSIEN)) -+ SSI1_SCR = 0; -+ -+ /* async */ -+ if (rtd->cpu_dai->flags & SND_SOC_DAI_ASYNC) -+ SSI1_SCR |= SSI_SCR_SYN; -+ -+ /* DAI mode */ -+ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ case SND_SOC_DAIFMT_LEFT_J: -+ SSI1_SCR |= SSI_SCR_NET; -+ break; -+ } -+ -+ /* TDM - to complete */ -+ -+ /* Tx/Rx config */ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ return imx_ssi1_hw_tx_params(substream, params); -+ } else { -+ return imx_ssi1_hw_rx_params(substream, params); -+ } -+} -+ -+ -+ -+static int imx_ssi1_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ SSI1_SCR |= SSI_SCR_TE; -+ SSI1_SIER |= SSI_SIER_TDMAE; -+ } else { -+ SSI1_SCR |= SSI_SCR_RE; -+ SSI1_SIER |= SSI_SIER_RDMAE; -+ } -+ SSI1_SCR |= SSI_SCR_SSIEN; -+ -+ break; -+ case SNDRV_PCM_TRIGGER_RESUME: -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSI1_SCR |= SSI_SCR_TE; -+ else -+ SSI1_SCR |= SSI_SCR_RE; -+ break -+ case SNDRV_PCM_TRIGGER_STOP: -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ SSI1_SCR &= ~SSI_SCR_TE; -+ else -+ SSI1_SCR &= ~SSI_SCR_RE; -+ break; -+ default: -+ ret = -EINVAL; -+ } -+ -+ return ret; -+} -+ -+static void imx_ssi_shutdown(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ -+ /* shutdown SSI */ -+ if (!rtd->cpu_dai->active) { -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR &= ~SSI_SCR_SSIEN; -+ else -+ SSI2_SCR &= ~SSI_SCR_SSIEN; -+ } -+} -+ -+#ifdef CONFIG_PM -+static int imx_ssi_suspend(struct platform_device *dev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if(!dai->active) -+ return 0; -+ -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR &= ~SSI_SCR_SSIEN; -+ else -+ SSI2_SCR &= ~SSI_SCR_SSIEN; -+ -+ return 0; -+} -+ -+static int imx_ssi_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+ if(!dai->active) -+ return 0; -+ -+ if(rtd->cpu_dai->id == 0) -+ SSI1_SCR |= SSI_SCR_SSIEN; -+ else -+ SSI2_SCR |= SSI_SCR_SSIEN; -+ -+ return 0; -+} -+ -+#else -+#define imx_ssi_suspend NULL -+#define imx_ssi_resume NULL -+#endif -+ -+static unsigned int imx_ssi_config_pcm_sysclk(struct snd_soc_cpu_dai *iface, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ return clk; -+} -+ -+struct snd_soc_cpu_dai imx_ssi_pcm_dai = { -+ .name = "imx-i2s-1", -+ .id = 0, -+ .type = SND_SOC_DAI_I2S, -+ .suspend = imx_ssi_suspend, -+ .resume = imx_ssi_resume, -+ .config_sysclk = imx_ssi_config_pcm_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = imx_ssi_startup, -+ .shutdown = imx_ssi_shutdown, -+ .trigger = imx_ssi1_trigger, -+ .hw_params = imx_ssi1_pcm_hw_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(imx_ssi_modes), -+ .mode = imx_ssi_modes,}, -+}, -+{ -+ .name = "imx-i2s-2", -+ .id = 1, -+ .type = SND_SOC_DAI_I2S, -+ .suspend = imx_ssi_suspend, -+ .resume = imx_ssi_resume, -+ .config_sysclk = imx_ssi_config_pcm_sysclk, -+ .playback = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 1, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = imx_ssi_startup, -+ .shutdown = imx_ssi_shutdown, -+ .trigger = imx_ssi1_trigger, -+ .hw_params = imx_ssi1_pcm_hw_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(imx_ssi_modes), -+ .mode = imx_ssi_modes,}, -+}; -+ -+ -+EXPORT_SYMBOL_GPL(imx_ssi_i2s_dai); -+ -+/* Module information */ -+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("i.MX ASoC I2S driver"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/include/linux/i2c-id.h -=================================================================== ---- linux-2.6.17/include/linux/i2c-id.h.orig 2006-11-25 00:15:33.571185017 +0100 -+++ linux-2.6.17/include/linux/i2c-id.h 2006-11-25 00:17:31.017877925 +0100 -@@ -113,6 +113,9 @@ - #define I2C_DRIVERID_PCF8563 83 /* Philips PCF8563 RTC */ - #define I2C_DRIVERID_RS5C372 84 /* Ricoh RS5C372 RTC */ - -+#define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ -+#define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ -+ - #define I2C_DRIVERID_I2CDEV 900 - #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ - #define I2C_DRIVERID_ALERT 903 /* SMBus Alert Responder Client */ -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8976.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8976.c -@@ -0,0 +1,953 @@ -+/* -+ * wm8976.c -- WM8976 ALSA Soc Audio driver -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#include <linux/module.h> -+#include <linux/moduleparam.h> -+#include <linux/version.h> -+#include <linux/kernel.h> -+#include <linux/init.h> -+#include <linux/delay.h> -+#include <linux/pm.h> -+#include <linux/i2c.h> -+#include <linux/platform_device.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <sound/soc-dapm.h> -+#include <sound/initval.h> -+ -+#include "wm8976.h" -+ -+#define AUDIO_NAME "wm8976" -+#define WM8976_VERSION "0.2" -+ -+/* -+ * Debug -+ */ -+ -+#define WM8976_DEBUG 0 -+ -+#ifdef WM8976_DEBUG -+#define dbg(format, arg...) \ -+ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -+#else -+#define dbg(format, arg...) do {} while (0) -+#endif -+#define err(format, arg...) \ -+ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -+#define info(format, arg...) \ -+ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -+#define warn(format, arg...) \ -+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) -+ -+struct snd_soc_codec_device soc_codec_dev_wm8976; -+ -+/* -+ * wm8976 register cache -+ * We can't read the WM8976 register space when we are -+ * using 2 wire for device control, so we cache them instead. -+ */ -+static const u16 wm8976_reg[WM8976_CACHEREGNUM] = { -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0050, 0x0000, 0x0140, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x00ff, -+ 0x00ff, 0x0000, 0x0100, 0x00ff, -+ 0x00ff, 0x0000, 0x012c, 0x002c, -+ 0x002c, 0x002c, 0x002c, 0x0000, -+ 0x0032, 0x0000, 0x0000, 0x0000, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0038, 0x000b, 0x0032, 0x0000, -+ 0x0008, 0x000c, 0x0093, 0x00e9, -+ 0x0000, 0x0000, 0x0000, 0x0000, -+ 0x0033, 0x0010, 0x0010, 0x0100, -+ 0x0100, 0x0002, 0x0001, 0x0001, -+ 0x0039, 0x0039, 0x0039, 0x0039, -+ 0x0001, 0x0001, -+}; -+ -+#define WM8976_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ -+ SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \ -+ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF) -+ -+#define WM8976_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define WM8976_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000) -+ -+#define WM8976_PCM_FORMATS \ -+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ -+ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | \ -+ SNDRV_PCM_FORMAT_S32_LE) -+ -+#define WM8976_BCLK \ -+ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\ -+ SND_SOC_FSBD(16) | SND_SOC_FSBD(32)) -+ -+static struct snd_soc_dai_mode wm8976_modes[] = { -+ /* codec frame and clock master modes */ -+ { -+ .fmt = WM8976_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, -+ .pcmfmt = WM8976_PCM_FORMATS, -+ .pcmrate = WM8976_RATES, -+ .pcmdir = WM8976_DIR, -+ .flags = SND_SOC_DAI_BFS_DIV, -+ .fs = 256, -+ .bfs = WM8976_BCLK, -+ }, -+ -+ /* codec frame and clock slave modes */ -+ { -+ .fmt = WM8976_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = WM8976_PCM_FORMATS, -+ .pcmrate = WM8976_RATES, -+ .pcmdir = WM8976_DIR, -+ .fs = SND_SOC_FS_ALL, -+ .bfs = SND_SOC_FSB_ALL, -+ }, -+}; -+ -+/* -+ * read wm8976 register cache -+ */ -+static inline unsigned int wm8976_read_reg_cache(struct snd_soc_codec *codec, -+ unsigned int reg) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg == WM8976_RESET) -+ return 0; -+ if (reg >= WM8976_CACHEREGNUM) -+ return -1; -+ return cache[reg]; -+} -+ -+/* -+ * write wm8976 register cache -+ */ -+static inline void wm8976_write_reg_cache(struct snd_soc_codec *codec, -+ u16 reg, unsigned int value) -+{ -+ u16 *cache = codec->reg_cache; -+ if (reg >= WM8976_CACHEREGNUM) -+ return; -+ cache[reg] = value; -+} -+ -+/* -+ * write to the WM8976 register space -+ */ -+static int wm8976_write(struct snd_soc_codec *codec, unsigned int reg, -+ unsigned int value) -+{ -+ u8 data[2]; -+ -+ /* data is -+ * D15..D9 WM8976 register offset -+ * D8...D0 register data -+ */ -+ data[0] = (reg << 1) | ((value >> 8) & 0x0001); -+ data[1] = value & 0x00ff; -+ -+ wm8976_write_reg_cache (codec, reg, value); -+ if (codec->hw_write(codec->control_data, data, 2) == 2) -+ return 0; -+ else -+ return -1; -+} -+ -+#define wm8976_reset(c) wm8976_write(c, WM8976_RESET, 0) -+ -+static const char *wm8976_companding[] = {"Off", "NC", "u-law", "A-law" }; -+static const char *wm8976_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; -+static const char *wm8976_eqmode[] = {"Capture", "Playback" }; -+static const char *wm8976_bw[] = {"Narrow", "Wide" }; -+static const char *wm8976_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" }; -+static const char *wm8976_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" }; -+static const char *wm8976_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" }; -+static const char *wm8976_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; -+static const char *wm8976_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; -+static const char *wm8976_alc[] = -+ {"ALC both on", "ALC left only", "ALC right only", "Limiter" }; -+ -+static const struct soc_enum wm8976_enum[] = { -+ SOC_ENUM_SINGLE(WM8976_COMP, 1, 4, wm8976_companding), /* adc */ -+ SOC_ENUM_SINGLE(WM8976_COMP, 3, 4, wm8976_companding), /* dac */ -+ SOC_ENUM_SINGLE(WM8976_DAC, 4, 4, wm8976_deemp), -+ SOC_ENUM_SINGLE(WM8976_EQ1, 8, 2, wm8976_eqmode), -+ -+ SOC_ENUM_SINGLE(WM8976_EQ1, 5, 4, wm8976_eq1), -+ SOC_ENUM_SINGLE(WM8976_EQ2, 8, 2, wm8976_bw), -+ SOC_ENUM_SINGLE(WM8976_EQ2, 5, 4, wm8976_eq2), -+ SOC_ENUM_SINGLE(WM8976_EQ3, 8, 2, wm8976_bw), -+ -+ SOC_ENUM_SINGLE(WM8976_EQ3, 5, 4, wm8976_eq3), -+ SOC_ENUM_SINGLE(WM8976_EQ4, 8, 2, wm8976_bw), -+ SOC_ENUM_SINGLE(WM8976_EQ4, 5, 4, wm8976_eq4), -+ SOC_ENUM_SINGLE(WM8976_EQ5, 8, 2, wm8976_bw), -+ -+ SOC_ENUM_SINGLE(WM8976_EQ5, 5, 4, wm8976_eq5), -+ SOC_ENUM_SINGLE(WM8976_ALC3, 8, 2, wm8976_alc), -+}; -+ -+static const struct snd_kcontrol_new wm8976_snd_controls[] = { -+SOC_SINGLE("Digital Loopback Switch", WM8976_COMP, 0, 1, 0), -+ -+SOC_ENUM("ADC Companding", wm8976_enum[0]), -+SOC_ENUM("DAC Companding", wm8976_enum[1]), -+ -+SOC_SINGLE("Jack Detection Enable", WM8976_JACK1, 6, 1, 0), -+ -+SOC_DOUBLE("DAC Inversion Switch", WM8976_DAC, 0, 1, 1, 0), -+ -+SOC_DOUBLE_R("Headphone Playback Volume", WM8976_DACVOLL, WM8976_DACVOLR, 0, 127, 0), -+ -+SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0), -+SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0), -+SOC_SINGLE("High Pass Cut Off", WM8976_ADC, 4, 7, 0), -+ -+SOC_DOUBLE("ADC Inversion Switch", WM8976_ADC, 0, 1, 1, 0), -+ -+SOC_SINGLE("Capture Volume", WM8976_ADCVOL, 0, 127, 0), -+ -+SOC_ENUM("Equaliser Function", wm8976_enum[3]), -+SOC_ENUM("EQ1 Cut Off", wm8976_enum[4]), -+SOC_SINGLE("EQ1 Volume", WM8976_EQ1, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ2 Bandwith", wm8976_enum[5]), -+SOC_ENUM("EQ2 Cut Off", wm8976_enum[6]), -+SOC_SINGLE("EQ2 Volume", WM8976_EQ2, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ3 Bandwith", wm8976_enum[7]), -+SOC_ENUM("EQ3 Cut Off", wm8976_enum[8]), -+SOC_SINGLE("EQ3 Volume", WM8976_EQ3, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ4 Bandwith", wm8976_enum[9]), -+SOC_ENUM("EQ4 Cut Off", wm8976_enum[10]), -+SOC_SINGLE("EQ4 Volume", WM8976_EQ4, 0, 31, 1), -+ -+SOC_ENUM("Equaliser EQ5 Bandwith", wm8976_enum[11]), -+SOC_ENUM("EQ5 Cut Off", wm8976_enum[12]), -+SOC_SINGLE("EQ5 Volume", WM8976_EQ5, 0, 31, 1), -+ -+SOC_SINGLE("DAC Playback Limiter Switch", WM8976_DACLIM1, 8, 1, 0), -+SOC_SINGLE("DAC Playback Limiter Decay", WM8976_DACLIM1, 4, 15, 0), -+SOC_SINGLE("DAC Playback Limiter Attack", WM8976_DACLIM1, 0, 15, 0), -+ -+SOC_SINGLE("DAC Playback Limiter Threshold", WM8976_DACLIM2, 4, 7, 0), -+SOC_SINGLE("DAC Playback Limiter Boost", WM8976_DACLIM2, 0, 15, 0), -+ -+SOC_SINGLE("ALC Enable Switch", WM8976_ALC1, 8, 1, 0), -+SOC_SINGLE("ALC Capture Max Gain", WM8976_ALC1, 3, 7, 0), -+SOC_SINGLE("ALC Capture Min Gain", WM8976_ALC1, 0, 7, 0), -+ -+SOC_SINGLE("ALC Capture ZC Switch", WM8976_ALC2, 8, 1, 0), -+SOC_SINGLE("ALC Capture Hold", WM8976_ALC2, 4, 7, 0), -+SOC_SINGLE("ALC Capture Target", WM8976_ALC2, 0, 15, 0), -+ -+SOC_ENUM("ALC Capture Mode", wm8976_enum[13]), -+SOC_SINGLE("ALC Capture Decay", WM8976_ALC3, 4, 15, 0), -+SOC_SINGLE("ALC Capture Attack", WM8976_ALC3, 0, 15, 0), -+ -+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8976_NGATE, 3, 1, 0), -+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8976_NGATE, 0, 7, 0), -+ -+SOC_SINGLE("Capture PGA ZC Switch", WM8976_INPPGA, 7, 1, 0), -+SOC_SINGLE("Capture PGA Volume", WM8976_INPPGA, 0, 63, 0), -+ -+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8976_HPVOLL, WM8976_HPVOLR, 7, 1, 0), -+SOC_DOUBLE_R("Headphone Playback Switch", WM8976_HPVOLL, WM8976_HPVOLR, 6, 1, 1), -+SOC_DOUBLE_R("Headphone Playback Volume", WM8976_HPVOLL, WM8976_HPVOLR, 0, 63, 0), -+ -+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8976_SPKVOLL, WM8976_SPKVOLR, 7, 1, 0), -+SOC_DOUBLE_R("Speaker Playback Switch", WM8976_SPKVOLL, WM8976_SPKVOLR, 6, 1, 1), -+SOC_DOUBLE_R("Speaker Playback Volume", WM8976_SPKVOLL, WM8976_SPKVOLR, 0, 63, 0), -+ -+SOC_SINGLE("Capture Boost(+20dB)", WM8976_ADCBOOST, 8, 1, 0), -+}; -+ -+/* add non dapm controls */ -+static int wm8976_add_controls(struct snd_soc_codec *codec) -+{ -+ int err, i; -+ -+ for (i = 0; i < ARRAY_SIZE(wm8976_snd_controls); i++) { -+ err = snd_ctl_add(codec->card, -+ snd_soc_cnew(&wm8976_snd_controls[i],codec, NULL)); -+ if (err < 0) -+ return err; -+ } -+ -+ return 0; -+} -+ -+/* Left Output Mixer */ -+static const snd_kcontrol_new_t wm8976_left_mixer_controls[] = { -+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_OUTPUT, 6, 1, 1), -+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_MIXL, 0, 1, 1), -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXL, 1, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXL, 5, 1, 0), -+}; -+ -+/* Right Output Mixer */ -+static const snd_kcontrol_new_t wm8976_right_mixer_controls[] = { -+SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_OUTPUT, 5, 1, 1), -+SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_MIXR, 0, 1, 1), -+SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXR, 1, 1, 0), -+SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXR, 5, 1, 0), -+}; -+ -+/* Left AUX Input boost vol */ -+static const snd_kcontrol_new_t wm8976_laux_boost_controls = -+SOC_DAPM_SINGLE("Aux Volume", WM8976_ADCBOOST, 0, 3, 0); -+ -+/* Left Input boost vol */ -+static const snd_kcontrol_new_t wm8976_lmic_boost_controls = -+SOC_DAPM_SINGLE("Input Volume", WM8976_ADCBOOST, 4, 3, 0); -+ -+/* Left Aux In to PGA */ -+static const snd_kcontrol_new_t wm8976_laux_capture_boost_controls = -+SOC_DAPM_SINGLE("Capture Switch", WM8976_ADCBOOST, 8, 1, 0); -+ -+/* Left Input P In to PGA */ -+static const snd_kcontrol_new_t wm8976_lmicp_capture_boost_controls = -+SOC_DAPM_SINGLE("Input P Capture Boost Switch", WM8976_INPUT, 0, 1, 0); -+ -+/* Left Input N In to PGA */ -+static const snd_kcontrol_new_t wm8976_lmicn_capture_boost_controls = -+SOC_DAPM_SINGLE("Input N Capture Boost Switch", WM8976_INPUT, 1, 1, 0); -+ -+// TODO Widgets -+static const struct snd_soc_dapm_widget wm8976_dapm_widgets[] = { -+#if 0 -+//SND_SOC_DAPM_MUTE("Mono Mute", WM8976_MONOMIX, 6, 0), -+//SND_SOC_DAPM_MUTE("Speaker Mute", WM8976_SPKMIX, 6, 0), -+ -+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8976_POWER3, 2, 0, -+ &wm8976_speaker_mixer_controls[0], -+ ARRAY_SIZE(wm8976_speaker_mixer_controls)), -+SND_SOC_DAPM_MIXER("Mono Mixer", WM8976_POWER3, 3, 0, -+ &wm8976_mono_mixer_controls[0], -+ ARRAY_SIZE(wm8976_mono_mixer_controls)), -+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8976_POWER3, 0, 0), -+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8976_POWER3, 0, 0), -+SND_SOC_DAPM_PGA("Aux Input", WM8976_POWER1, 6, 0, NULL, 0), -+SND_SOC_DAPM_PGA("SpkN Out", WM8976_POWER3, 5, 0, NULL, 0), -+SND_SOC_DAPM_PGA("SpkP Out", WM8976_POWER3, 6, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mono Out", WM8976_POWER3, 7, 0, NULL, 0), -+SND_SOC_DAPM_PGA("Mic PGA", WM8976_POWER2, 2, 0, NULL, 0), -+ -+SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, -+ &wm8976_aux_boost_controls, 1), -+SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, -+ &wm8976_mic_boost_controls, 1), -+SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, -+ &wm8976_capture_boost_controls), -+ -+SND_SOC_DAPM_MIXER("Boost Mixer", WM8976_POWER2, 4, 0, NULL, 0), -+ -+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8976_POWER1, 4, 0), -+ -+SND_SOC_DAPM_INPUT("MICN"), -+SND_SOC_DAPM_INPUT("MICP"), -+SND_SOC_DAPM_INPUT("AUX"), -+SND_SOC_DAPM_OUTPUT("MONOOUT"), -+SND_SOC_DAPM_OUTPUT("SPKOUTP"), -+SND_SOC_DAPM_OUTPUT("SPKOUTN"), -+#endif -+}; -+ -+static const char *audio_map[][3] = { -+ /* Mono output mixer */ -+ {"Mono Mixer", "PCM Playback Switch", "DAC"}, -+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, -+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, -+ -+ /* Speaker output mixer */ -+ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, -+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, -+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, -+ -+ /* Outputs */ -+ {"Mono Out", NULL, "Mono Mixer"}, -+ {"MONOOUT", NULL, "Mono Out"}, -+ {"SpkN Out", NULL, "Speaker Mixer"}, -+ {"SpkP Out", NULL, "Speaker Mixer"}, -+ {"SPKOUTN", NULL, "SpkN Out"}, -+ {"SPKOUTP", NULL, "SpkP Out"}, -+ -+ /* Boost Mixer */ -+ {"Boost Mixer", NULL, "ADC"}, -+ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, -+ {"Aux Boost", "Aux Volume", "Boost Mixer"}, -+ {"Capture Boost", "Capture Switch", "Boost Mixer"}, -+ {"Mic Boost", "Mic Volume", "Boost Mixer"}, -+ -+ /* Inputs */ -+ {"MICP", NULL, "Mic Boost"}, -+ {"MICN", NULL, "Mic PGA"}, -+ {"Mic PGA", NULL, "Capture Boost"}, -+ {"AUX", NULL, "Aux Input"}, -+ -+ /* */ -+ -+ /* terminator */ -+ {NULL, NULL, NULL}, -+}; -+ -+static int wm8976_add_widgets(struct snd_soc_codec *codec) -+{ -+ int i; -+ -+ for(i = 0; i < ARRAY_SIZE(wm8976_dapm_widgets); i++) { -+ snd_soc_dapm_new_control(codec, &wm8976_dapm_widgets[i]); -+ } -+ -+ /* set up audio path map */ -+ for(i = 0; audio_map[i][0] != NULL; i++) { -+ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], -+ audio_map[i][2]); -+ } -+ -+ snd_soc_dapm_new_widgets(codec); -+ return 0; -+} -+ -+struct pll_ { -+ unsigned int in_hz, out_hz; -+ unsigned int pre:4; /* prescale - 1 */ -+ unsigned int n:4; -+ unsigned int k; -+}; -+ -+struct pll_ pll[] = { -+ {12000000, 11289600, 0, 7, 0x86c220}, -+ {12000000, 12288000, 0, 8, 0x3126e8}, -+ {13000000, 11289600, 0, 6, 0xf28bd4}, -+ {13000000, 12288000, 0, 7, 0x8fd525}, -+ {12288000, 11289600, 0, 7, 0x59999a}, -+ {11289600, 12288000, 0, 8, 0x80dee9}, -+ /* TODO: liam - add more entries */ -+}; -+ -+static int set_pll(struct snd_soc_codec *codec, unsigned int in, -+ unsigned int out) -+{ -+ int i; -+ u16 reg; -+ -+ if(out == 0) { -+ reg = wm8976_read_reg_cache(codec, WM8976_POWER1); -+ wm8976_write(codec, WM8976_POWER1, reg & 0x1df); -+ return 0; -+ } -+ -+ for(i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (in == pll[i].in_hz && out == pll[i].out_hz) { -+ wm8976_write(codec, WM8976_PLLN, (pll[i].pre << 4) | pll[i].n); -+ wm8976_write(codec, WM8976_PLLK1, pll[i].k >> 18); -+ wm8976_write(codec, WM8976_PLLK1, (pll[i].k >> 9) && 0x1ff); -+ wm8976_write(codec, WM8976_PLLK1, pll[i].k && 0x1ff); -+ reg = wm8976_read_reg_cache(codec, WM8976_POWER1); -+ wm8976_write(codec, WM8976_POWER1, reg | 0x020); -+ return 0; -+ } -+ } -+ return -EINVAL; -+} -+ -+/* mclk dividers * 2 */ -+static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24}; -+ -+/* we need 256FS to drive the DAC's and ADC's */ -+static unsigned int wm8976_config_sysclk(struct snd_soc_codec_dai *dai, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ int i, j, best_clk = info->fs * info->rate; -+ -+ /* can we run at this clk without the PLL ? */ -+ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) { -+ if ((best_clk >> 1) * mclk_div[i] == clk) { -+ dai->pll_in = 0; -+ dai->clk_div = mclk_div[i]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ -+ /* now check for PLL support */ -+ for (i = 0; i < ARRAY_SIZE(pll); i++) { -+ if (pll[i].in_hz == clk) { -+ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) { -+ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) { -+ dai->pll_in = clk; -+ dai->pll_out = pll[i].out_hz; -+ dai->clk_div = mclk_div[j]; -+ dai->mclk = best_clk; -+ return dai->mclk; -+ } -+ } -+ } -+ } -+ -+ /* this clk is not supported */ -+ return 0; -+} -+ -+static int wm8976_pcm_prepare(snd_pcm_substream_t *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct snd_soc_codec_dai *dai = rtd->codec_dai; -+ u16 iface = 0, bfs, clk = 0, adn; -+ int fs = 48000 << 7, i; -+ -+ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); -+ switch (bfs) { -+ case 2: -+ clk |= 0x1 << 2; -+ break; -+ case 4: -+ clk |= 0x2 << 2; -+ break; -+ case 8: -+ clk |= 0x3 << 2; -+ break; -+ case 16: -+ clk |= 0x4 << 2; -+ break; -+ case 32: -+ clk |= 0x5 << 2; -+ break; -+ } -+ -+ /* set master/slave audio interface */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { -+ case SND_SOC_DAIFMT_CBM_CFM: -+ clk |= 0x0001; -+ break; -+ case SND_SOC_DAIFMT_CBS_CFS: -+ break; -+ } -+ -+ /* interface format */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -+ case SND_SOC_DAIFMT_I2S: -+ iface |= 0x0010; -+ break; -+ case SND_SOC_DAIFMT_RIGHT_J: -+ break; -+ case SND_SOC_DAIFMT_LEFT_J: -+ iface |= 0x0008; -+ break; -+ case SND_SOC_DAIFMT_DSP_A: -+ iface |= 0x00018; -+ break; -+ } -+ -+ /* bit size */ -+ switch (rtd->codec_dai->dai_runtime.pcmfmt) { -+ case SNDRV_PCM_FMTBIT_S16_LE: -+ break; -+ case SNDRV_PCM_FMTBIT_S20_3LE: -+ iface |= 0x0020; -+ break; -+ case SNDRV_PCM_FMTBIT_S24_LE: -+ iface |= 0x0040; -+ break; -+ case SNDRV_PCM_FMTBIT_S32_LE: -+ iface |= 0x0060; -+ break; -+ } -+ -+ /* clock inversion */ -+ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { -+ case SND_SOC_DAIFMT_NB_NF: -+ break; -+ case SND_SOC_DAIFMT_IB_IF: -+ iface |= 0x0180; -+ break; -+ case SND_SOC_DAIFMT_IB_NF: -+ iface |= 0x0100; -+ break; -+ case SND_SOC_DAIFMT_NB_IF: -+ iface |= 0x0080; -+ break; -+ } -+ -+ /* filter coefficient */ -+ adn = wm8976_read_reg_cache(codec, WM8976_ADD) & 0x1f1; -+ switch (rtd->codec_dai->dai_runtime.pcmrate) { -+ case SNDRV_PCM_RATE_8000: -+ adn |= 0x5 << 1; -+ fs = 8000 << 7; -+ break; -+ case SNDRV_PCM_RATE_11025: -+ adn |= 0x4 << 1; -+ fs = 11025 << 7; -+ break; -+ case SNDRV_PCM_RATE_16000: -+ adn |= 0x3 << 1; -+ fs = 16000 << 7; -+ break; -+ case SNDRV_PCM_RATE_22050: -+ adn |= 0x2 << 1; -+ fs = 22050 << 7; -+ break; -+ case SNDRV_PCM_RATE_32000: -+ adn |= 0x1 << 1; -+ fs = 32000 << 7; -+ break; -+ case SNDRV_PCM_RATE_44100: -+ fs = 44100 << 7; -+ break; -+ } -+ -+ /* do we need to enable the PLL */ -+ if(dai->pll_in) -+ set_pll(codec, dai->pll_in, dai->pll_out); -+ -+ /* divide the clock to 256 fs */ -+ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) { -+ if (dai->clk_div == mclk_div[i]) { -+ clk |= i << 5; -+ clk &= 0xff; -+ goto set; -+ } -+ } -+ -+set: -+ /* set iface */ -+ wm8976_write(codec, WM8976_IFACE, iface); -+ wm8976_write(codec, WM8976_CLOCK, clk); -+ -+ return 0; -+} -+ -+static int wm8976_hw_free(struct snd_pcm_substream *substream) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ struct snd_soc_device *socdev = rtd->socdev; -+ struct snd_soc_codec *codec = socdev->codec; -+ set_pll(codec, 0, 0); -+ return 0; -+} -+ -+static int wm8976_mute(struct snd_soc_codec *codec, -+ struct snd_soc_codec_dai *dai, int mute) -+{ -+ u16 mute_reg = wm8976_read_reg_cache(codec, WM8976_DAC) & 0xffbf; -+ if(mute) -+ wm8976_write(codec, WM8976_DAC, mute_reg | 0x40); -+ else -+ wm8976_write(codec, WM8976_DAC, mute_reg); -+ -+ return 0; -+} -+ -+/* TODO: liam need to make this lower power with dapm */ -+static int wm8976_dapm_event(struct snd_soc_codec *codec, int event) -+{ -+ -+ switch (event) { -+ case SNDRV_CTL_POWER_D0: /* full On */ -+ /* vref/mid, clk and osc on, dac unmute, active */ -+ wm8976_write(codec, WM8976_POWER1, 0x1ff); -+ wm8976_write(codec, WM8976_POWER2, 0x1ff); -+ wm8976_write(codec, WM8976_POWER3, 0x1ff); -+ break; -+ case SNDRV_CTL_POWER_D1: /* partial On */ -+ case SNDRV_CTL_POWER_D2: /* partial On */ -+ break; -+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ -+ /* everything off except vref/vmid, dac mute, inactive */ -+ -+ break; -+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ -+ /* everything off, dac mute, inactive */ -+ wm8976_write(codec, WM8976_POWER1, 0x0); -+ wm8976_write(codec, WM8976_POWER2, 0x0); -+ wm8976_write(codec, WM8976_POWER3, 0x0); -+ break; -+ } -+ codec->dapm_state = event; -+ return 0; -+} -+ -+struct snd_soc_codec_dai wm8976_dai = { -+ .name = "WM8976 HiFi", -+ .playback = { -+ .stream_name = "Playback", -+ .channels_min = 1, -+ .channels_max = 2, -+ }, -+ .capture = { -+ .stream_name = "Capture", -+ .channels_min = 1, -+ .channels_max = 1, -+ }, -+ .config_sysclk = wm8976_config_sysclk, -+ .digital_mute = wm8976_mute, -+ .ops = { -+ .prepare = wm8976_pcm_prepare, -+ .hw_free = wm8976_hw_free, -+ }, -+ .caps = { -+ .num_modes = ARRAY_SIZE(wm8976_modes), -+ .mode = wm8976_modes, -+ }, -+}; -+EXPORT_SYMBOL_GPL(wm8976_dai); -+ -+static int wm8976_suspend(struct platform_device *pdev, pm_message_t state) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ return 0; -+} -+ -+static int wm8976_resume(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ int i; -+ u8 data[2]; -+ u16 *cache = codec->reg_cache; -+ -+ /* Sync reg_cache with the hardware */ -+ for (i = 0; i < ARRAY_SIZE(wm8976_reg); i++) { -+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); -+ data[1] = cache[i] & 0x00ff; -+ codec->hw_write(codec->control_data, data, 2); -+ } -+ wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8976_dapm_event(codec, codec->suspend_dapm_state); -+ return 0; -+} -+ -+/* -+ * initialise the WM8976 driver -+ * register the mixer and dsp interfaces with the kernel -+ */ -+static int wm8976_init(struct snd_soc_device* socdev) -+{ -+ struct snd_soc_codec *codec = socdev->codec; -+ int ret = 0; -+ -+ codec->name = "WM8976"; -+ codec->owner = THIS_MODULE; -+ codec->read = wm8976_read_reg_cache; -+ codec->write = wm8976_write; -+ codec->dapm_event = wm8976_dapm_event; -+ codec->dai = &wm8976_dai; -+ codec->num_dai = 1; -+ codec->reg_cache_size = ARRAY_SIZE(wm8976_reg); -+ codec->reg_cache = -+ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8976_reg), GFP_KERNEL); -+ if (codec->reg_cache == NULL) -+ return -ENOMEM; -+ memcpy(codec->reg_cache, wm8976_reg, -+ sizeof(u16) * ARRAY_SIZE(wm8976_reg)); -+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8976_reg); -+ -+ wm8976_reset(codec); -+ -+ /* register pcms */ -+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -+ if(ret < 0) { -+ kfree(codec->reg_cache); -+ return ret; -+ } -+ -+ /* power on device */ -+ wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3hot); -+ wm8976_add_controls(codec); -+ wm8976_add_widgets(codec); -+ ret = snd_soc_register_card(socdev); -+ if(ret < 0) { -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+ } -+ -+ return ret; -+} -+ -+static struct snd_soc_device *wm8976_socdev; -+ -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ -+/* -+ * WM8976 2 wire address is 0x1a -+ */ -+#define I2C_DRIVERID_WM8976 0xfefe /* liam - need a proper id */ -+ -+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -+ -+/* Magic definition of all other variables and things */ -+I2C_CLIENT_INSMOD; -+ -+static struct i2c_driver wm8976_i2c_driver; -+static struct i2c_client client_template; -+ -+/* If the i2c layer weren't so broken, we could pass this kind of data -+ around */ -+ -+static int wm8976_codec_probe(struct i2c_adapter *adap, int addr, int kind) -+{ -+ struct snd_soc_device *socdev = wm8976_socdev; -+ struct wm8976_setup_data *setup = socdev->codec_data; -+ struct snd_soc_codec *codec = socdev->codec; -+ struct i2c_client *i2c; -+ int ret; -+ -+ if (addr != setup->i2c_address) -+ return -ENODEV; -+ -+ client_template.adapter = adap; -+ client_template.addr = addr; -+ -+ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); -+ if (i2c == NULL){ -+ kfree(codec); -+ return -ENOMEM; -+ } -+ memcpy(i2c, &client_template, sizeof(struct i2c_client)); -+ -+ i2c_set_clientdata(i2c, codec); -+ -+ codec->control_data = i2c; -+ -+ ret = i2c_attach_client(i2c); -+ if(ret < 0) { -+ err("failed to attach codec at addr %x\n", addr); -+ goto err; -+ } -+ -+ ret = wm8976_init(socdev); -+ if(ret < 0) { -+ err("failed to initialise WM8976\n"); -+ goto err; -+ } -+ return ret; -+ -+err: -+ kfree(codec); -+ kfree(i2c); -+ return ret; -+ -+} -+ -+static int wm8976_i2c_detach(struct i2c_client *client) -+{ -+ struct snd_soc_codec *codec = i2c_get_clientdata(client); -+ -+ i2c_detach_client(client); -+ -+ kfree(codec->reg_cache); -+ kfree(client); -+ -+ return 0; -+} -+ -+static int wm8976_i2c_attach(struct i2c_adapter *adap) -+{ -+ return i2c_probe(adap, &addr_data, wm8976_codec_probe); -+} -+ -+/* corgi i2c codec control layer */ -+static struct i2c_driver wm8976_i2c_driver = { -+ .driver = { -+ .name = "WM8976 I2C Codec", -+ .owner = THIS_MODULE, -+ }, -+ .id = I2C_DRIVERID_WM8976, -+ .attach_adapter = wm8976_i2c_attach, -+ .detach_client = wm8976_i2c_detach, -+ .command = NULL, -+}; -+ -+static struct i2c_client client_template = { -+ .name = "WM8976", -+ .driver = &wm8976_i2c_driver, -+}; -+#endif -+ -+static int wm8976_probe(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct wm8976_setup_data *setup; -+ struct snd_soc_codec *codec; -+ int ret = 0; -+ -+ info("WM8976 Audio Codec %s", WM8976_VERSION); -+ -+ setup = socdev->codec_data; -+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); -+ if (codec == NULL) -+ return -ENOMEM; -+ -+ socdev->codec = codec; -+ mutex_init(&codec->mutex); -+ INIT_LIST_HEAD(&codec->dapm_widgets); -+ INIT_LIST_HEAD(&codec->dapm_paths); -+ -+ wm8976_socdev = socdev; -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ if (setup->i2c_address) { -+ normal_i2c[0] = setup->i2c_address; -+ codec->hw_write = (hw_write_t)i2c_master_send; -+ ret = i2c_add_driver(&wm8976_i2c_driver); -+ if (ret != 0) -+ printk(KERN_ERR "can't add i2c driver"); -+ } -+#else -+ /* Add other interfaces here */ -+#endif -+ return ret; -+} -+ -+/* power down chip */ -+static int wm8976_remove(struct platform_device *pdev) -+{ -+ struct snd_soc_device *socdev = platform_get_drvdata(pdev); -+ struct snd_soc_codec *codec = socdev->codec; -+ -+ if (codec->control_data) -+ wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3cold); -+ -+ snd_soc_free_pcms(socdev); -+ snd_soc_dapm_free(socdev); -+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) -+ i2c_del_driver(&wm8976_i2c_driver); -+#endif -+ kfree(codec); -+ -+ return 0; -+} -+ -+struct snd_soc_codec_device soc_codec_dev_wm8976 = { -+ .probe = wm8976_probe, -+ .remove = wm8976_remove, -+ .suspend = wm8976_suspend, -+ .resume = wm8976_resume, -+}; -+ -+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8976); -+ -+MODULE_DESCRIPTION("ASoC WM8976 driver"); -+MODULE_AUTHOR("Graeme Gregory"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/codecs/wm8976.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/codecs/wm8976.h -@@ -0,0 +1,73 @@ -+/* -+ * wm8976.h -- WM8976 Soc Audio driver -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _WM8976_H -+#define _WM8976_H -+ -+/* WM8976 register space */ -+ -+#define WM8976_RESET 0x0 -+#define WM8976_POWER1 0x1 -+#define WM8976_POWER2 0x2 -+#define WM8976_POWER3 0x3 -+#define WM8976_IFACE 0x4 -+#define WM8976_COMP 0x5 -+#define WM8976_CLOCK 0x6 -+#define WM8976_ADD 0x7 -+#define WM8976_GPIO 0x8 -+#define WM8976_JACK1 0x9 -+#define WM8976_DAC 0xa -+#define WM8976_DACVOLL 0xb -+#define WM8976_DACVOLR 0xc -+#define WM8976_JACK2 0xd -+#define WM8976_ADC 0xe -+#define WM8976_ADCVOL 0xf -+#define WM8976_EQ1 0x12 -+#define WM8976_EQ2 0x13 -+#define WM8976_EQ3 0x14 -+#define WM8976_EQ4 0x15 -+#define WM8976_EQ5 0x16 -+#define WM8976_DACLIM1 0x18 -+#define WM8976_DACLIM2 0x19 -+#define WM8976_NOTCH1 0x1b -+#define WM8976_NOTCH2 0x1c -+#define WM8976_NOTCH3 0x1d -+#define WM8976_NOTCH4 0x1e -+#define WM8976_ALC1 0x20 -+#define WM8976_ALC2 0x21 -+#define WM8976_ALC3 0x22 -+#define WM8976_NGATE 0x23 -+#define WM8976_PLLN 0x24 -+#define WM8976_PLLK1 0x25 -+#define WM8976_PLLK2 0x26 -+#define WM8976_PLLK3 0x27 -+#define WM8976_3D 0x29 -+#define WM8976_BEEP 0x2b -+#define WM8976_INPUT 0x2c -+#define WM8976_INPPGA 0x2d -+#define WM8976_ADCBOOST 0x2f -+#define WM8976_OUTPUT 0x31 -+#define WM8976_MIXL 0x32 -+#define WM8976_MIXR 0x33 -+#define WM8976_HPVOLL 0x34 -+#define WM8976_HPVOLR 0x35 -+#define WM8976_SPKVOLL 0x36 -+#define WM8976_SPKVOLR 0x37 -+#define WM8976_OUT3MIX 0x38 -+#define WM8976_MONOMIX 0x39 -+ -+#define WM8976_CACHEREGNUM 58 -+ -+struct wm8976_setup_data { -+ unsigned short i2c_address; -+}; -+ -+extern struct snd_soc_codec_dai wm8976_dai; -+extern struct snd_soc_codec_device soc_codec_dev_wm8976; -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/imx/imx21-pcm.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/imx21-pcm.c -@@ -0,0 +1,454 @@ -+/* -+ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Based on pxa2xx-pcm.c by Nicolas Pitre, (C) 2004 MontaVista Software, Inc. -+ * and on mxc-alsa-mc13783 (C) 2006 Freescale. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ * -+ * Revision history -+ * 29th Aug 2006 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/init.h> -+#include <linux/platform_device.h> -+#include <linux/slab.h> -+#include <linux/dma-mapping.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <asm/dma.h> -+#include <asm/hardware.h> -+ -+#include "imx-pcm.h" -+ -+/* debug */ -+#define IMX_DEBUG 0 -+#if IMX_DEBUG -+#define dbg(format, arg...) printk(format, ## arg) -+#else -+#define dbg(format, arg...) -+#endif -+ -+static const struct snd_pcm_hardware mxc_pcm_hardware = { -+ .info = (SNDRV_PCM_INFO_INTERLEAVED | -+ SNDRV_PCM_INFO_BLOCK_TRANSFER | -+ SNDRV_PCM_INFO_MMAP | -+ SNDRV_PCM_INFO_MMAP_VALID | -+ SNDRV_PCM_INFO_PAUSE | -+ SNDRV_PCM_INFO_RESUME), -+ .formats = SNDRV_PCM_FMTBIT_S16_LE | -+ SNDRV_PCM_FMTBIT_S24_LE, -+ .buffer_bytes_max = 32 * 1024, -+ .period_bytes_min = 64, -+ .period_bytes_max = 8 * 1024, -+ .periods_min = 2, -+ .periods_max = 255, -+ .fifo_size = 0, -+}; -+ -+struct mxc_runtime_data { -+ int dma_ch; -+ struct mxc_pcm_dma_param *dma_params; -+}; -+ -+/*! -+ * This function stops the current dma transfert for playback -+ * and clears the dma pointers. -+ * -+ * @param substream pointer to the structure of the current stream. -+ * -+ */ -+static void audio_stop_dma(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size); -+ unsigned int offset dma_size * s->periods; -+ unsigned long flags; -+ -+ spin_lock_irqsave(&prtd->dma_lock, flags); -+ -+ dbg("MXC : audio_stop_dma active = 0\n"); -+ prtd->active = 0; -+ prtd->period = 0; -+ prtd->periods = 0; -+ -+ /* this stops the dma channel and clears the buffer ptrs */ -+ mxc_dma_stop(prtd->dma_wchannel); -+ if(substream == SNDRV_PCM_STREAM_PLAYBACK) -+ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, -+ DMA_TO_DEVICE); -+ else -+ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, -+ DMA_FROM_DEVICE); -+ -+ spin_unlock_irqrestore(&prtd->dma_lock, flags); -+} -+ -+/*! -+ * This function is called whenever a new audio block needs to be -+ * transferred to mc13783. The function receives the address and the size -+ * of the new block and start a new DMA transfer. -+ * -+ * @param substream pointer to the structure of the current stream. -+ * -+ */ -+static int dma_new_period(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ unsigned int dma_size; -+ unsigned int offset; -+ int ret=0; -+ dma_request_t sdma_request; -+ -+ if (prtd->active){ -+ memset(&sdma_request, 0, sizeof(dma_request_t)); -+ dma_size = frames_to_bytes(runtime, runtime->period_size); -+ dbg("s->period (%x) runtime->periods (%d)\n", -+ s->period,runtime->periods); -+ dbg("runtime->period_size (%d) dma_size (%d)\n", -+ (unsigned int)runtime->period_size, -+ runtime->dma_bytes); -+ -+ offset = dma_size * prtd->period; -+ snd_assert(dma_size <= DMA_BUF_SIZE, ); -+ if(substream == SNDRV_PCM_STREAM_PLAYBACK) -+ sdma_request.sourceAddr = (char*)(dma_map_single(NULL, -+ runtime->dma_area + offset, dma_size, DMA_TO_DEVICE)); -+ else -+ sdma_request.destAddr = (char*)(dma_map_single(NULL, -+ runtime->dma_area + offset, dma_size, DMA_FROM_DEVICE)); -+ sdma_request.count = dma_size; -+ -+ dbg("MXC: Start DMA offset (%d) size (%d)\n", offset, -+ runtime->dma_bytes); -+ -+ mxc_dma_set_config(prtd->dma_wchannel, &sdma_request, 0); -+ if((ret = mxc_dma_start(prtd->dma_wchannel)) < 0) { -+ dbg("audio_process_dma: cannot queue DMA buffer\ -+ (%i)\n", ret); -+ return err; -+ } -+ prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ -+ prtd->period++; -+ prtd->period %= runtime->periods; -+ } -+ return ret; -+} -+ -+ -+/*! -+ * This is a callback which will be called -+ * when a TX transfer finishes. The call occurs -+ * in interrupt context. -+ * -+ * @param dat pointer to the structure of the current stream. -+ * -+ */ -+static void audio_dma_irq(void *data) -+{ -+ struct snd_pcm_substream *substream; -+ struct snd_pcm_runtime *runtime; -+ struct mxc_runtime_data *prtd; -+ unsigned int dma_size; -+ unsigned int previous_period; -+ unsigned int offset; -+ -+ substream = data; -+ runtime = substream->runtime; -+ prtd = runtime->private_data; -+ previous_period = prtd->periods; -+ dma_size = frames_to_bytes(runtime, runtime->period_size); -+ offset = dma_size * previous_period; -+ -+ prtd->tx_spin = 0; -+ prtd->periods++; -+ prtd->periods %= runtime->periods; -+ -+ /* -+ * Give back to the CPU the access to the non cached memory -+ */ -+ if(substream == SNDRV_PCM_STREAM_PLAYBACK) -+ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, -+ DMA_TO_DEVICE); -+ else -+ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, -+ DMA_FROM_DEVICE); -+ /* -+ * If we are getting a callback for an active stream then we inform -+ * the PCM middle layer we've finished a period -+ */ -+ if (prtd->active) -+ snd_pcm_period_elapsed(substream); -+ -+ /* -+ * Trig next DMA transfer -+ */ -+ dma_new_period(substream); -+} -+ -+/*! -+ * This function configures the hardware to allow audio -+ * playback operations. It is called by ALSA framework. -+ * -+ * @param substream pointer to the structure of the current stream. -+ * -+ * @return 0 on success, -1 otherwise. -+ */ -+static int -+snd_mxc_prepare(struct snd_pcm_substream *substream) -+{ -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ int ret = 0; -+ prtd->period = 0; -+ prtd->periods = 0; -+ -+ dma_channel_params params; -+ int channel = 0; // passed in ? -+ -+ if ((ret = mxc_request_dma(&channel, "ALSA TX SDMA") < 0)){ -+ dbg("error requesting a write dma channel\n"); -+ return ret; -+ } -+ -+ /* configure DMA params */ -+ memset(¶ms, 0, sizeof(dma_channel_params)); -+ params.bd_number = 1; -+ params.arg = s; -+ params.callback = callback; -+ params.transfer_type = emi_2_per; -+ params.watermark_level = SDMA_TXFIFO_WATERMARK; -+ params.word_size = TRANSFER_16BIT; -+ //dbg(KERN_ERR "activating connection SSI1 - SDMA\n"); -+ params.per_address = SSI1_BASE_ADDR; -+ params.event_id = DMA_REQ_SSI1_TX1; -+ params.peripheral_type = SSI; -+ -+ /* set up chn with params */ -+ mxc_dma_setup_channel(channel, ¶ms); -+ s->dma_wchannel = channel; -+ -+ return ret; -+} -+ -+static int mxc_pcm_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ int ret; -+ -+ if((ret=snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) -+ return ret; -+ runtime->dma_addr = virt_to_phys(runtime->dma_area); -+ -+ return ret; -+} -+ -+static int mxc_pcm_hw_free(struct snd_pcm_substream *substream) -+{ -+ return snd_pcm_lib_free_pages(substream); -+} -+ -+static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ struct mxc_runtime_data *prtd = substream->runtime->private_data; -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ prtd->tx_spin = 0; -+ /* requested stream startup */ -+ prtd->active = 1; -+ ret = dma_new_period(substream); -+ break; -+ case SNDRV_PCM_TRIGGER_STOP: -+ /* requested stream shutdown */ -+ ret = audio_stop_dma(substream); -+ break; -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ prtd->active = 0; -+ prtd->periods = 0; -+ break; -+ case SNDRV_PCM_TRIGGER_RESUME: -+ prtd->active = 1; -+ prtd->tx_spin = 0; -+ ret = dma_new_period(substream); -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ prtd->active = 0; -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ prtd->active = 1; -+ if (prtd->old_offset) { -+ prtd->tx_spin = 0; -+ ret = dma_new_period(substream); -+ } -+ break; -+ default: -+ ret = -EINVAL; -+ break; -+ } -+ -+ return ret; -+} -+ -+static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ unsigned int offset = 0; -+ -+ /* tx_spin value is used here to check if a transfert is active */ -+ if (prtd->tx_spin){ -+ offset = (runtime->period_size * (prtd->periods)) + -+ (runtime->period_size >> 1); -+ if (offset >= runtime->buffer_size) -+ offset = runtime->period_size >> 1; -+ } else { -+ offset = (runtime->period_size * (s->periods)); -+ if (offset >= runtime->buffer_size) -+ offset = 0; -+ } -+ -+ return offset; -+} -+ -+ -+static int mxc_pcm_open(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd; -+ int ret; -+ -+ snd_soc_set_runtime_hwparams(substream, &mxc_pcm_hardware); -+ -+ if ((err = snd_pcm_hw_constraint_integer(runtime, -+ SNDRV_PCM_HW_PARAM_PERIODS)) < 0) -+ return err; -+ if ((err = snd_pcm_hw_constraint_list(runtime, 0, -+ SNDRV_PCM_HW_PARAM_RATE, &hw_playback_rates)) < 0) -+ return err; -+ msleep(10); // liam - why -+ -+ /* setup DMA controller for playback */ -+ if((err = configure_write_channel(&mxc_mc13783->s[SNDRV_PCM_STREAM_PLAYBACK], -+ audio_dma_irq)) < 0 ) -+ return err; -+ -+ if((prtd = kzalloc(sizeof(struct mxc_runtime_data), GFP_KERNEL)) == NULL) { -+ ret = -ENOMEM; -+ goto out; -+ } -+ -+ runtime->private_data = prtd; -+ return 0; -+ -+ err1: -+ kfree(prtd); -+ out: -+ return ret; -+} -+ -+static int mxc_pcm_close(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ -+// mxc_mc13783_t *chip; -+ audio_stream_t *s; -+ device_data_t* device; -+ int ssi; -+ -+ //chip = snd_pcm_substream_chip(substream); -+ s = &chip->s[substream->pstr->stream]; -+ device = &s->stream_device; -+ ssi = device->ssi; -+ -+ //disable_stereodac(); -+ -+ ssi_transmit_enable(ssi, false); -+ ssi_interrupt_disable(ssi, ssi_tx_dma_interrupt_enable); -+ ssi_tx_fifo_enable(ssi, ssi_fifo_0, false); -+ ssi_enable(ssi, false); -+ -+ chip->s[substream->pstr->stream].stream = NULL; -+ -+ return 0; -+} -+ -+static int -+mxc_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ return dma_mmap_writecombine(substream->pcm->card->dev, vma, -+ runtime->dma_area, -+ runtime->dma_addr, -+ runtime->dma_bytes); -+} -+ -+struct snd_pcm_ops mxc_pcm_ops = { -+ .open = mxc_pcm_open, -+ .close = mxc_pcm_close, -+ .ioctl = snd_pcm_lib_ioctl, -+ .hw_params = mxc_pcm_hw_params, -+ .hw_free = mxc_pcm_hw_free, -+ .prepare = mxc_pcm_prepare, -+ .trigger = mxc_pcm_trigger, -+ .pointer = mxc_pcm_pointer, -+ .mmap = mxc_pcm_mmap, -+}; -+ -+static u64 mxc_pcm_dmamask = 0xffffffff; -+ -+int mxc_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, -+ struct snd_pcm *pcm) -+{ -+ int ret = 0; -+ -+ if (!card->dev->dma_mask) -+ card->dev->dma_mask = &mxc_pcm_dmamask; -+ if (!card->dev->coherent_dma_mask) -+ card->dev->coherent_dma_mask = 0xffffffff; -+ -+ if (dai->playback.channels_min) { -+ ret = mxc_pcm_preallocate_dma_buffer(pcm, -+ SNDRV_PCM_STREAM_PLAYBACK); -+ if (ret) -+ goto out; -+ } -+ -+ if (dai->capture.channels_min) { -+ ret = mxc_pcm_preallocate_dma_buffer(pcm, -+ SNDRV_PCM_STREAM_CAPTURE); -+ if (ret) -+ goto out; -+ } -+ out: -+ return ret; -+} -+ -+struct snd_soc_platform mxc_soc_platform = { -+ .name = "mxc-audio", -+ .pcm_ops = &mxc_pcm_ops, -+ .pcm_new = mxc_pcm_new, -+ .pcm_free = mxc_pcm_free_dma_buffers, -+}; -+ -+EXPORT_SYMBOL_GPL(mxc_soc_platform); -+ -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_DESCRIPTION("Freescale i.MX PCM DMA module"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/imx/imx21-pcm.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/imx21-pcm.h -@@ -0,0 +1,237 @@ -+/* -+ * mxc-pcm.h :- ASoC platform header for Freescale i.MX -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _MXC_PCM_H -+#define _MXC_PCM_H -+ -+struct { -+ char *name; /* stream identifier */ -+ dma_channel_params dma_params; -+} mxc_pcm_dma_param; -+ -+extern struct snd_soc_cpu_dai mxc_ssi_dai[3]; -+ -+/* platform data */ -+extern struct snd_soc_platform mxc_soc_platform; -+extern struct snd_ac97_bus_ops mxc_ac97_ops; -+ -+/* temp until imx-regs.h is up2date */ -+#define SSI1_STX0 __REG(IMX_SSI1_BASE + 0x00) -+#define SSI1_STX0_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x00) -+#define SSI1_STX1 __REG(IMX_SSI1_BASE + 0x04) -+#define SSI1_STX1_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x04) -+#define SSI1_SRX0 __REG(IMX_SSI1_BASE + 0x08) -+#define SSI1_SRX0_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x08) -+#define SSI1_SRX1 __REG(IMX_SSI1_BASE + 0x0c) -+#define SSI1_SRX1_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x0c) -+#define SSI1_SCR __REG(IMX_SSI1_BASE + 0x10) -+#define SSI1_SISR __REG(IMX_SSI1_BASE + 0x14) -+#define SSI1_SIER __REG(IMX_SSI1_BASE + 0x18) -+#define SSI1_STCR __REG(IMX_SSI1_BASE + 0x1c) -+#define SSI1_SRCR __REG(IMX_SSI1_BASE + 0x20) -+#define SSI1_STCCR __REG(IMX_SSI1_BASE + 0x24) -+#define SSI1_SRCCR __REG(IMX_SSI1_BASE + 0x28) -+#define SSI1_SFCSR __REG(IMX_SSI1_BASE + 0x2c) -+#define SSI1_STR __REG(IMX_SSI1_BASE + 0x30) -+#define SSI1_SOR __REG(IMX_SSI1_BASE + 0x34) -+#define SSI1_SACNT __REG(IMX_SSI1_BASE + 0x38) -+#define SSI1_SACADD __REG(IMX_SSI1_BASE + 0x3c) -+#define SSI1_SACDAT __REG(IMX_SSI1_BASE + 0x40) -+#define SSI1_SATAG __REG(IMX_SSI1_BASE + 0x44) -+#define SSI1_STMSK __REG(IMX_SSI1_BASE + 0x48) -+#define SSI1_SRMSK __REG(IMX_SSI1_BASE + 0x4c) -+ -+#define SSI2_STX0 __REG(IMX_SSI2_BASE + 0x00) -+#define SSI2_STX0_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x00) -+#define SSI2_STX1 __REG(IMX_SSI2_BASE + 0x04) -+#define SSI2_STX1_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x04) -+#define SSI2_SRX0 __REG(IMX_SSI2_BASE + 0x08) -+#define SSI2_SRX0_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x08) -+#define SSI2_SRX1 __REG(IMX_SSI2_BASE + 0x0c) -+#define SSI2_SRX1_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x0c) -+#define SSI2_SCR __REG(IMX_SSI2_BASE + 0x10) -+#define SSI2_SISR __REG(IMX_SSI2_BASE + 0x14) -+#define SSI2_SIER __REG(IMX_SSI2_BASE + 0x18) -+#define SSI2_STCR __REG(IMX_SSI2_BASE + 0x1c) -+#define SSI2_SRCR __REG(IMX_SSI2_BASE + 0x20) -+#define SSI2_STCCR __REG(IMX_SSI2_BASE + 0x24) -+#define SSI2_SRCCR __REG(IMX_SSI2_BASE + 0x28) -+#define SSI2_SFCSR __REG(IMX_SSI2_BASE + 0x2c) -+#define SSI2_STR __REG(IMX_SSI2_BASE + 0x30) -+#define SSI2_SOR __REG(IMX_SSI2_BASE + 0x34) -+#define SSI2_SACNT __REG(IMX_SSI2_BASE + 0x38) -+#define SSI2_SACADD __REG(IMX_SSI2_BASE + 0x3c) -+#define SSI2_SACDAT __REG(IMX_SSI2_BASE + 0x40) -+#define SSI2_SATAG __REG(IMX_SSI2_BASE + 0x44) -+#define SSI2_STMSK __REG(IMX_SSI2_BASE + 0x48) -+#define SSI2_SRMSK __REG(IMX_SSI2_BASE + 0x4c) -+ -+#define SSI_SCR_CLK_IST (1 << 9) -+#define SSI_SCR_TCH_EN (1 << 8) -+#define SSI_SCR_SYS_CLK_EN (1 << 7) -+#define SSI_SCR_I2S_MODE_NORM (0 << 5) -+#define SSI_SCR_I2S_MODE_MSTR (1 << 5) -+#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) -+#define SSI_SCR_SYN (1 << 4) -+#define SSI_SCR_NET (1 << 3) -+#define SSI_SCR_RE (1 << 2) -+#define SSI_SCR_TE (1 << 1) -+#define SSI_SCR_SSIEN (1 << 0) -+ -+#define SSI_SISR_CMDAU (1 << 18) -+#define SSI_SISR_CMDDU (1 << 17) -+#define SSI_SISR_RXT (1 << 16) -+#define SSI_SISR_RDR1 (1 << 15) -+#define SSI_SISR_RDR0 (1 << 14) -+#define SSI_SISR_TDE1 (1 << 13) -+#define SSI_SISR_TDE0 (1 << 12) -+#define SSI_SISR_ROE1 (1 << 11) -+#define SSI_SISR_ROE0 (1 << 10) -+#define SSI_SISR_TUE1 (1 << 9) -+#define SSI_SISR_TUE0 (1 << 8) -+#define SSI_SISR_TFS (1 << 7) -+#define SSI_SISR_RFS (1 << 6) -+#define SSI_SISR_TLS (1 << 5) -+#define SSI_SISR_RLS (1 << 4) -+#define SSI_SISR_RFF1 (1 << 3) -+#define SSI_SISR_RFF0 (1 << 2) -+#define SSI_SISR_TFE1 (1 << 1) -+#define SSI_SISR_TFE0 (1 << 0) -+ -+#define SSI_SIER_RDMAE (1 << 22) -+#define SSI_SIER_RIE (1 << 21) -+#define SSI_SIER_TDMAE (1 << 20) -+#define SSI_SIER_TIE (1 << 19) -+#define SSI_SIER_CMDAU_EN (1 << 18) -+#define SSI_SIER_CMDDU_EN (1 << 17) -+#define SSI_SIER_RXT_EN (1 << 16) -+#define SSI_SIER_RDR1_EN (1 << 15) -+#define SSI_SIER_RDR0_EN (1 << 14) -+#define SSI_SIER_TDE1_EN (1 << 13) -+#define SSI_SIER_TDE0_EN (1 << 12) -+#define SSI_SIER_ROE1_EN (1 << 11) -+#define SSI_SIER_ROE0_EN (1 << 10) -+#define SSI_SIER_TUE1_EN (1 << 9) -+#define SSI_SIER_TUE0_EN (1 << 8) -+#define SSI_SIER_TFS_EN (1 << 7) -+#define SSI_SIER_RFS_EN (1 << 6) -+#define SSI_SIER_TLS_EN (1 << 5) -+#define SSI_SIER_RLS_EN (1 << 4) -+#define SSI_SIER_RFF1_EN (1 << 3) -+#define SSI_SIER_RFF0_EN (1 << 2) -+#define SSI_SIER_TFE1_EN (1 << 1) -+#define SSI_SIER_TFE0_EN (1 << 0) -+ -+#define SSI_STCR_TXBIT0 (1 << 9) -+#define SSI_STCR_TFEN1 (1 << 8) -+#define SSI_STCR_TFEN0 (1 << 7) -+#define SSI_STCR_TFDIR (1 << 6) -+#define SSI_STCR_TXDIR (1 << 5) -+#define SSI_STCR_TSHFD (1 << 4) -+#define SSI_STCR_TSCKP (1 << 3) -+#define SSI_STCR_TFSI (1 << 2) -+#define SSI_STCR_TFSL (1 << 1) -+#define SSI_STCR_TEFS (1 << 0) -+ -+#define SSI_SRCR_RXBIT0 (1 << 9) -+#define SSI_SRCR_RFEN1 (1 << 8) -+#define SSI_SRCR_RFEN0 (1 << 7) -+#define SSI_SRCR_RFDIR (1 << 6) -+#define SSI_SRCR_RXDIR (1 << 5) -+#define SSI_SRCR_RSHFD (1 << 4) -+#define SSI_SRCR_RSCKP (1 << 3) -+#define SSI_SRCR_RFSI (1 << 2) -+#define SSI_SRCR_RFSL (1 << 1) -+#define SSI_SRCR_REFS (1 << 0) -+ -+#define SSI_STCCR_DIV2 (1 << 18) -+#define SSI_STCCR_PSR (1 << 15) -+#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) -+#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) -+#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) -+ -+#define SSI_SRCCR_DIV2 (1 << 18) -+#define SSI_SRCCR_PSR (1 << 15) -+#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) -+#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) -+#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) -+ -+ -+#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) -+#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) -+#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) -+#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) -+#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) -+#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) -+#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) -+#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) -+ -+#define SSI_STR_TEST (1 << 15) -+#define SSI_STR_RCK2TCK (1 << 14) -+#define SSI_STR_RFS2TFS (1 << 13) -+#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) -+#define SSI_STR_TXD2RXD (1 << 7) -+#define SSI_STR_TCK2RCK (1 << 6) -+#define SSI_STR_TFS2RFS (1 << 5) -+#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) -+ -+#define SSI_SOR_CLKOFF (1 << 6) -+#define SSI_SOR_RX_CLR (1 << 5) -+#define SSI_SOR_TX_CLR (1 << 4) -+#define SSI_SOR_INIT (1 << 3) -+#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) -+#define SSI_SOR_SYNRST (1 << 0) -+ -+#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) -+#define SSI_SACNT_WR (x << 4) -+#define SSI_SACNT_RD (x << 3) -+#define SSI_SACNT_TIF (x << 2) -+#define SSI_SACNT_FV (x << 1) -+#define SSI_SACNT_A97EN (x << 0) -+ -+ -+/* AUDMUX registers */ -+#define AUDMUX_HPCR1 __REG(IMX_AUDMUX_BASE + 0x00) -+#define AUDMUX_HPCR2 __REG(IMX_AUDMUX_BASE + 0x04) -+#define AUDMUX_HPCR3 __REG(IMX_AUDMUX_BASE + 0x08) -+#define AUDMUX_PPCR1 __REG(IMX_AUDMUX_BASE + 0x10) -+#define AUDMUX_PPCR2 __REG(IMX_AUDMUX_BASE + 0x14) -+#define AUDMUX_PPCR3 __REG(IMX_AUDMUX_BASE + 0x18) -+ -+#define AUDMUX_HPCR_TFSDIR (1 << 31) -+#define AUDMUX_HPCR_TCLKDIR (1 << 30) -+#define AUDMUX_HPCR_TFCSEL_TX (0 << 26) -+#define AUDMUX_HPCR_TFCSEL_RX (8 << 26) -+#define AUDMUX_HPCR_TFCSEL(x) (((x) & 0x7) << 26) -+#define AUDMUX_HPCR_RFSDIR (1 << 25) -+#define AUDMUX_HPCR_RCLKDIR (1 << 24) -+#define AUDMUX_HPCR_RFCSEL_TX (0 << 20) -+#define AUDMUX_HPCR_RFCSEL_RX (8 << 20) -+#define AUDMUX_HPCR_RFCSEL(x) (((x) & 0x7) << 20) -+#define AUDMUX_HPCR_RXDSEL(x) (((x) & 0x7) << 13) -+#define AUDMUX_HPCR_SYN (1 << 12) -+#define AUDMUX_HPCR_TXRXEN (1 << 10) -+#define AUDMUX_HPCR_INMEN (1 << 8) -+#define AUDMUX_HPCR_INMMASK(x) (((x) & 0xff) << 0) -+ -+#define AUDMUX_PPCR_TFSDIR (1 << 31) -+#define AUDMUX_PPCR_TCLKDIR (1 << 30) -+#define AUDMUX_PPCR_TFCSEL_TX (0 << 26) -+#define AUDMUX_PPCR_TFCSEL_RX (8 << 26) -+#define AUDMUX_PPCR_TFCSEL(x) (((x) & 0x7) << 26) -+#define AUDMUX_PPCR_RFSDIR (1 << 25) -+#define AUDMUX_PPCR_RCLKDIR (1 << 24) -+#define AUDMUX_PPCR_RFCSEL_TX (0 << 20) -+#define AUDMUX_PPCR_RFCSEL_RX (8 << 20) -+#define AUDMUX_PPCR_RFCSEL(x) (((x) & 0x7) << 20) -+#define AUDMUX_PPCR_RXDSEL(x) (((x) & 0x7) << 13) -+#define AUDMUX_PPCR_SYN (1 << 12) -+#define AUDMUX_PPCR_TXRXEN (1 << 10) -+ -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/imx/imx31-pcm.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/imx31-pcm.c -@@ -0,0 +1,454 @@ -+/* -+ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's -+ * -+ * Copyright 2006 Wolfson Microelectronics PLC. -+ * Author: Liam Girdwood -+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * Based on pxa2xx-pcm.c by Nicolas Pitre, (C) 2004 MontaVista Software, Inc. -+ * and on mxc-alsa-mc13783 (C) 2006 Freescale. -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ * -+ * Revision history -+ * 29th Aug 2006 Initial version. -+ * -+ */ -+ -+#include <linux/module.h> -+#include <linux/init.h> -+#include <linux/platform_device.h> -+#include <linux/slab.h> -+#include <linux/dma-mapping.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+#include <asm/dma.h> -+#include <asm/hardware.h> -+ -+#include "imx-pcm.h" -+ -+/* debug */ -+#define IMX_DEBUG 0 -+#if IMX_DEBUG -+#define dbg(format, arg...) printk(format, ## arg) -+#else -+#define dbg(format, arg...) -+#endif -+ -+static const struct snd_pcm_hardware mxc_pcm_hardware = { -+ .info = (SNDRV_PCM_INFO_INTERLEAVED | -+ SNDRV_PCM_INFO_BLOCK_TRANSFER | -+ SNDRV_PCM_INFO_MMAP | -+ SNDRV_PCM_INFO_MMAP_VALID | -+ SNDRV_PCM_INFO_PAUSE | -+ SNDRV_PCM_INFO_RESUME), -+ .formats = SNDRV_PCM_FMTBIT_S16_LE | -+ SNDRV_PCM_FMTBIT_S24_LE, -+ .buffer_bytes_max = 32 * 1024, -+ .period_bytes_min = 64, -+ .period_bytes_max = 8 * 1024, -+ .periods_min = 2, -+ .periods_max = 255, -+ .fifo_size = 0, -+}; -+ -+struct mxc_runtime_data { -+ int dma_ch; -+ struct mxc_pcm_dma_param *dma_params; -+}; -+ -+/*! -+ * This function stops the current dma transfert for playback -+ * and clears the dma pointers. -+ * -+ * @param substream pointer to the structure of the current stream. -+ * -+ */ -+static void audio_stop_dma(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size); -+ unsigned int offset dma_size * s->periods; -+ unsigned long flags; -+ -+ spin_lock_irqsave(&prtd->dma_lock, flags); -+ -+ dbg("MXC : audio_stop_dma active = 0\n"); -+ prtd->active = 0; -+ prtd->period = 0; -+ prtd->periods = 0; -+ -+ /* this stops the dma channel and clears the buffer ptrs */ -+ mxc_dma_stop(prtd->dma_wchannel); -+ if(substream == SNDRV_PCM_STREAM_PLAYBACK) -+ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, -+ DMA_TO_DEVICE); -+ else -+ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, -+ DMA_FROM_DEVICE); -+ -+ spin_unlock_irqrestore(&prtd->dma_lock, flags); -+} -+ -+/*! -+ * This function is called whenever a new audio block needs to be -+ * transferred to mc13783. The function receives the address and the size -+ * of the new block and start a new DMA transfer. -+ * -+ * @param substream pointer to the structure of the current stream. -+ * -+ */ -+static int dma_new_period(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ unsigned int dma_size; -+ unsigned int offset; -+ int ret=0; -+ dma_request_t sdma_request; -+ -+ if (prtd->active){ -+ memset(&sdma_request, 0, sizeof(dma_request_t)); -+ dma_size = frames_to_bytes(runtime, runtime->period_size); -+ dbg("s->period (%x) runtime->periods (%d)\n", -+ s->period,runtime->periods); -+ dbg("runtime->period_size (%d) dma_size (%d)\n", -+ (unsigned int)runtime->period_size, -+ runtime->dma_bytes); -+ -+ offset = dma_size * prtd->period; -+ snd_assert(dma_size <= DMA_BUF_SIZE, ); -+ if(substream == SNDRV_PCM_STREAM_PLAYBACK) -+ sdma_request.sourceAddr = (char*)(dma_map_single(NULL, -+ runtime->dma_area + offset, dma_size, DMA_TO_DEVICE)); -+ else -+ sdma_request.destAddr = (char*)(dma_map_single(NULL, -+ runtime->dma_area + offset, dma_size, DMA_FROM_DEVICE)); -+ sdma_request.count = dma_size; -+ -+ dbg("MXC: Start DMA offset (%d) size (%d)\n", offset, -+ runtime->dma_bytes); -+ -+ mxc_dma_set_config(prtd->dma_wchannel, &sdma_request, 0); -+ if((ret = mxc_dma_start(prtd->dma_wchannel)) < 0) { -+ dbg("audio_process_dma: cannot queue DMA buffer\ -+ (%i)\n", ret); -+ return err; -+ } -+ prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ -+ prtd->period++; -+ prtd->period %= runtime->periods; -+ } -+ return ret; -+} -+ -+ -+/*! -+ * This is a callback which will be called -+ * when a TX transfer finishes. The call occurs -+ * in interrupt context. -+ * -+ * @param dat pointer to the structure of the current stream. -+ * -+ */ -+static void audio_dma_irq(void *data) -+{ -+ struct snd_pcm_substream *substream; -+ struct snd_pcm_runtime *runtime; -+ struct mxc_runtime_data *prtd; -+ unsigned int dma_size; -+ unsigned int previous_period; -+ unsigned int offset; -+ -+ substream = data; -+ runtime = substream->runtime; -+ prtd = runtime->private_data; -+ previous_period = prtd->periods; -+ dma_size = frames_to_bytes(runtime, runtime->period_size); -+ offset = dma_size * previous_period; -+ -+ prtd->tx_spin = 0; -+ prtd->periods++; -+ prtd->periods %= runtime->periods; -+ -+ /* -+ * Give back to the CPU the access to the non cached memory -+ */ -+ if(substream == SNDRV_PCM_STREAM_PLAYBACK) -+ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, -+ DMA_TO_DEVICE); -+ else -+ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, -+ DMA_FROM_DEVICE); -+ /* -+ * If we are getting a callback for an active stream then we inform -+ * the PCM middle layer we've finished a period -+ */ -+ if (prtd->active) -+ snd_pcm_period_elapsed(substream); -+ -+ /* -+ * Trig next DMA transfer -+ */ -+ dma_new_period(substream); -+} -+ -+/*! -+ * This function configures the hardware to allow audio -+ * playback operations. It is called by ALSA framework. -+ * -+ * @param substream pointer to the structure of the current stream. -+ * -+ * @return 0 on success, -1 otherwise. -+ */ -+static int -+snd_mxc_prepare(struct snd_pcm_substream *substream) -+{ -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ int ret = 0; -+ prtd->period = 0; -+ prtd->periods = 0; -+ -+ dma_channel_params params; -+ int channel = 0; // passed in ? -+ -+ if ((ret = mxc_request_dma(&channel, "ALSA TX SDMA") < 0)){ -+ dbg("error requesting a write dma channel\n"); -+ return ret; -+ } -+ -+ /* configure DMA params */ -+ memset(¶ms, 0, sizeof(dma_channel_params)); -+ params.bd_number = 1; -+ params.arg = s; -+ params.callback = callback; -+ params.transfer_type = emi_2_per; -+ params.watermark_level = SDMA_TXFIFO_WATERMARK; -+ params.word_size = TRANSFER_16BIT; -+ //dbg(KERN_ERR "activating connection SSI1 - SDMA\n"); -+ params.per_address = SSI1_BASE_ADDR; -+ params.event_id = DMA_REQ_SSI1_TX1; -+ params.peripheral_type = SSI; -+ -+ /* set up chn with params */ -+ mxc_dma_setup_channel(channel, ¶ms); -+ s->dma_wchannel = channel; -+ -+ return ret; -+} -+ -+static int mxc_pcm_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ int ret; -+ -+ if((ret=snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) -+ return ret; -+ runtime->dma_addr = virt_to_phys(runtime->dma_area); -+ -+ return ret; -+} -+ -+static int mxc_pcm_hw_free(struct snd_pcm_substream *substream) -+{ -+ return snd_pcm_lib_free_pages(substream); -+} -+ -+static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ struct mxc_runtime_data *prtd = substream->runtime->private_data; -+ int ret = 0; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ prtd->tx_spin = 0; -+ /* requested stream startup */ -+ prtd->active = 1; -+ ret = dma_new_period(substream); -+ break; -+ case SNDRV_PCM_TRIGGER_STOP: -+ /* requested stream shutdown */ -+ ret = audio_stop_dma(substream); -+ break; -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ prtd->active = 0; -+ prtd->periods = 0; -+ break; -+ case SNDRV_PCM_TRIGGER_RESUME: -+ prtd->active = 1; -+ prtd->tx_spin = 0; -+ ret = dma_new_period(substream); -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ prtd->active = 0; -+ break; -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ prtd->active = 1; -+ if (prtd->old_offset) { -+ prtd->tx_spin = 0; -+ ret = dma_new_period(substream); -+ } -+ break; -+ default: -+ ret = -EINVAL; -+ break; -+ } -+ -+ return ret; -+} -+ -+static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ unsigned int offset = 0; -+ -+ /* tx_spin value is used here to check if a transfert is active */ -+ if (prtd->tx_spin){ -+ offset = (runtime->period_size * (prtd->periods)) + -+ (runtime->period_size >> 1); -+ if (offset >= runtime->buffer_size) -+ offset = runtime->period_size >> 1; -+ } else { -+ offset = (runtime->period_size * (s->periods)); -+ if (offset >= runtime->buffer_size) -+ offset = 0; -+ } -+ -+ return offset; -+} -+ -+ -+static int mxc_pcm_open(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd; -+ int ret; -+ -+ snd_soc_set_runtime_hwparams(substream, &mxc_pcm_hardware); -+ -+ if ((err = snd_pcm_hw_constraint_integer(runtime, -+ SNDRV_PCM_HW_PARAM_PERIODS)) < 0) -+ return err; -+ if ((err = snd_pcm_hw_constraint_list(runtime, 0, -+ SNDRV_PCM_HW_PARAM_RATE, &hw_playback_rates)) < 0) -+ return err; -+ msleep(10); // liam - why -+ -+ /* setup DMA controller for playback */ -+ if((err = configure_write_channel(&mxc_mc13783->s[SNDRV_PCM_STREAM_PLAYBACK], -+ audio_dma_irq)) < 0 ) -+ return err; -+ -+ if((prtd = kzalloc(sizeof(struct mxc_runtime_data), GFP_KERNEL)) == NULL) { -+ ret = -ENOMEM; -+ goto out; -+ } -+ -+ runtime->private_data = prtd; -+ return 0; -+ -+ err1: -+ kfree(prtd); -+ out: -+ return ret; -+} -+ -+static int mxc_pcm_close(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct mxc_runtime_data *prtd = runtime->private_data; -+ -+// mxc_mc13783_t *chip; -+ audio_stream_t *s; -+ device_data_t* device; -+ int ssi; -+ -+ //chip = snd_pcm_substream_chip(substream); -+ s = &chip->s[substream->pstr->stream]; -+ device = &s->stream_device; -+ ssi = device->ssi; -+ -+ //disable_stereodac(); -+ -+ ssi_transmit_enable(ssi, false); -+ ssi_interrupt_disable(ssi, ssi_tx_dma_interrupt_enable); -+ ssi_tx_fifo_enable(ssi, ssi_fifo_0, false); -+ ssi_enable(ssi, false); -+ -+ chip->s[substream->pstr->stream].stream = NULL; -+ -+ return 0; -+} -+ -+static int -+mxc_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ return dma_mmap_writecombine(substream->pcm->card->dev, vma, -+ runtime->dma_area, -+ runtime->dma_addr, -+ runtime->dma_bytes); -+} -+ -+struct snd_pcm_ops mxc_pcm_ops = { -+ .open = mxc_pcm_open, -+ .close = mxc_pcm_close, -+ .ioctl = snd_pcm_lib_ioctl, -+ .hw_params = mxc_pcm_hw_params, -+ .hw_free = mxc_pcm_hw_free, -+ .prepare = mxc_pcm_prepare, -+ .trigger = mxc_pcm_trigger, -+ .pointer = mxc_pcm_pointer, -+ .mmap = mxc_pcm_mmap, -+}; -+ -+static u64 mxc_pcm_dmamask = 0xffffffff; -+ -+int mxc_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, -+ struct snd_pcm *pcm) -+{ -+ int ret = 0; -+ -+ if (!card->dev->dma_mask) -+ card->dev->dma_mask = &mxc_pcm_dmamask; -+ if (!card->dev->coherent_dma_mask) -+ card->dev->coherent_dma_mask = 0xffffffff; -+ -+ if (dai->playback.channels_min) { -+ ret = mxc_pcm_preallocate_dma_buffer(pcm, -+ SNDRV_PCM_STREAM_PLAYBACK); -+ if (ret) -+ goto out; -+ } -+ -+ if (dai->capture.channels_min) { -+ ret = mxc_pcm_preallocate_dma_buffer(pcm, -+ SNDRV_PCM_STREAM_CAPTURE); -+ if (ret) -+ goto out; -+ } -+ out: -+ return ret; -+} -+ -+struct snd_soc_platform mxc_soc_platform = { -+ .name = "mxc-audio", -+ .pcm_ops = &mxc_pcm_ops, -+ .pcm_new = mxc_pcm_new, -+ .pcm_free = mxc_pcm_free_dma_buffers, -+}; -+ -+EXPORT_SYMBOL_GPL(mxc_soc_platform); -+ -+MODULE_AUTHOR("Liam Girdwood"); -+MODULE_DESCRIPTION("Freescale i.MX PCM DMA module"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/imx/imx31-pcm.h -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/imx/imx31-pcm.h -@@ -0,0 +1,237 @@ -+/* -+ * mxc-pcm.h :- ASoC platform header for Freescale i.MX -+ * -+ * This program is free software; you can redistribute it and/or modify -+ * it under the terms of the GNU General Public License version 2 as -+ * published by the Free Software Foundation. -+ */ -+ -+#ifndef _MXC_PCM_H -+#define _MXC_PCM_H -+ -+struct { -+ char *name; /* stream identifier */ -+ dma_channel_params dma_params; -+} mxc_pcm_dma_param; -+ -+extern struct snd_soc_cpu_dai mxc_ssi_dai[3]; -+ -+/* platform data */ -+extern struct snd_soc_platform mxc_soc_platform; -+extern struct snd_ac97_bus_ops mxc_ac97_ops; -+ -+/* temp until imx-regs.h is up2date */ -+#define SSI1_STX0 __REG(IMX_SSI1_BASE + 0x00) -+#define SSI1_STX0_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x00) -+#define SSI1_STX1 __REG(IMX_SSI1_BASE + 0x04) -+#define SSI1_STX1_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x04) -+#define SSI1_SRX0 __REG(IMX_SSI1_BASE + 0x08) -+#define SSI1_SRX0_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x08) -+#define SSI1_SRX1 __REG(IMX_SSI1_BASE + 0x0c) -+#define SSI1_SRX1_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x0c) -+#define SSI1_SCR __REG(IMX_SSI1_BASE + 0x10) -+#define SSI1_SISR __REG(IMX_SSI1_BASE + 0x14) -+#define SSI1_SIER __REG(IMX_SSI1_BASE + 0x18) -+#define SSI1_STCR __REG(IMX_SSI1_BASE + 0x1c) -+#define SSI1_SRCR __REG(IMX_SSI1_BASE + 0x20) -+#define SSI1_STCCR __REG(IMX_SSI1_BASE + 0x24) -+#define SSI1_SRCCR __REG(IMX_SSI1_BASE + 0x28) -+#define SSI1_SFCSR __REG(IMX_SSI1_BASE + 0x2c) -+#define SSI1_STR __REG(IMX_SSI1_BASE + 0x30) -+#define SSI1_SOR __REG(IMX_SSI1_BASE + 0x34) -+#define SSI1_SACNT __REG(IMX_SSI1_BASE + 0x38) -+#define SSI1_SACADD __REG(IMX_SSI1_BASE + 0x3c) -+#define SSI1_SACDAT __REG(IMX_SSI1_BASE + 0x40) -+#define SSI1_SATAG __REG(IMX_SSI1_BASE + 0x44) -+#define SSI1_STMSK __REG(IMX_SSI1_BASE + 0x48) -+#define SSI1_SRMSK __REG(IMX_SSI1_BASE + 0x4c) -+ -+#define SSI2_STX0 __REG(IMX_SSI2_BASE + 0x00) -+#define SSI2_STX0_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x00) -+#define SSI2_STX1 __REG(IMX_SSI2_BASE + 0x04) -+#define SSI2_STX1_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x04) -+#define SSI2_SRX0 __REG(IMX_SSI2_BASE + 0x08) -+#define SSI2_SRX0_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x08) -+#define SSI2_SRX1 __REG(IMX_SSI2_BASE + 0x0c) -+#define SSI2_SRX1_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x0c) -+#define SSI2_SCR __REG(IMX_SSI2_BASE + 0x10) -+#define SSI2_SISR __REG(IMX_SSI2_BASE + 0x14) -+#define SSI2_SIER __REG(IMX_SSI2_BASE + 0x18) -+#define SSI2_STCR __REG(IMX_SSI2_BASE + 0x1c) -+#define SSI2_SRCR __REG(IMX_SSI2_BASE + 0x20) -+#define SSI2_STCCR __REG(IMX_SSI2_BASE + 0x24) -+#define SSI2_SRCCR __REG(IMX_SSI2_BASE + 0x28) -+#define SSI2_SFCSR __REG(IMX_SSI2_BASE + 0x2c) -+#define SSI2_STR __REG(IMX_SSI2_BASE + 0x30) -+#define SSI2_SOR __REG(IMX_SSI2_BASE + 0x34) -+#define SSI2_SACNT __REG(IMX_SSI2_BASE + 0x38) -+#define SSI2_SACADD __REG(IMX_SSI2_BASE + 0x3c) -+#define SSI2_SACDAT __REG(IMX_SSI2_BASE + 0x40) -+#define SSI2_SATAG __REG(IMX_SSI2_BASE + 0x44) -+#define SSI2_STMSK __REG(IMX_SSI2_BASE + 0x48) -+#define SSI2_SRMSK __REG(IMX_SSI2_BASE + 0x4c) -+ -+#define SSI_SCR_CLK_IST (1 << 9) -+#define SSI_SCR_TCH_EN (1 << 8) -+#define SSI_SCR_SYS_CLK_EN (1 << 7) -+#define SSI_SCR_I2S_MODE_NORM (0 << 5) -+#define SSI_SCR_I2S_MODE_MSTR (1 << 5) -+#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) -+#define SSI_SCR_SYN (1 << 4) -+#define SSI_SCR_NET (1 << 3) -+#define SSI_SCR_RE (1 << 2) -+#define SSI_SCR_TE (1 << 1) -+#define SSI_SCR_SSIEN (1 << 0) -+ -+#define SSI_SISR_CMDAU (1 << 18) -+#define SSI_SISR_CMDDU (1 << 17) -+#define SSI_SISR_RXT (1 << 16) -+#define SSI_SISR_RDR1 (1 << 15) -+#define SSI_SISR_RDR0 (1 << 14) -+#define SSI_SISR_TDE1 (1 << 13) -+#define SSI_SISR_TDE0 (1 << 12) -+#define SSI_SISR_ROE1 (1 << 11) -+#define SSI_SISR_ROE0 (1 << 10) -+#define SSI_SISR_TUE1 (1 << 9) -+#define SSI_SISR_TUE0 (1 << 8) -+#define SSI_SISR_TFS (1 << 7) -+#define SSI_SISR_RFS (1 << 6) -+#define SSI_SISR_TLS (1 << 5) -+#define SSI_SISR_RLS (1 << 4) -+#define SSI_SISR_RFF1 (1 << 3) -+#define SSI_SISR_RFF0 (1 << 2) -+#define SSI_SISR_TFE1 (1 << 1) -+#define SSI_SISR_TFE0 (1 << 0) -+ -+#define SSI_SIER_RDMAE (1 << 22) -+#define SSI_SIER_RIE (1 << 21) -+#define SSI_SIER_TDMAE (1 << 20) -+#define SSI_SIER_TIE (1 << 19) -+#define SSI_SIER_CMDAU_EN (1 << 18) -+#define SSI_SIER_CMDDU_EN (1 << 17) -+#define SSI_SIER_RXT_EN (1 << 16) -+#define SSI_SIER_RDR1_EN (1 << 15) -+#define SSI_SIER_RDR0_EN (1 << 14) -+#define SSI_SIER_TDE1_EN (1 << 13) -+#define SSI_SIER_TDE0_EN (1 << 12) -+#define SSI_SIER_ROE1_EN (1 << 11) -+#define SSI_SIER_ROE0_EN (1 << 10) -+#define SSI_SIER_TUE1_EN (1 << 9) -+#define SSI_SIER_TUE0_EN (1 << 8) -+#define SSI_SIER_TFS_EN (1 << 7) -+#define SSI_SIER_RFS_EN (1 << 6) -+#define SSI_SIER_TLS_EN (1 << 5) -+#define SSI_SIER_RLS_EN (1 << 4) -+#define SSI_SIER_RFF1_EN (1 << 3) -+#define SSI_SIER_RFF0_EN (1 << 2) -+#define SSI_SIER_TFE1_EN (1 << 1) -+#define SSI_SIER_TFE0_EN (1 << 0) -+ -+#define SSI_STCR_TXBIT0 (1 << 9) -+#define SSI_STCR_TFEN1 (1 << 8) -+#define SSI_STCR_TFEN0 (1 << 7) -+#define SSI_STCR_TFDIR (1 << 6) -+#define SSI_STCR_TXDIR (1 << 5) -+#define SSI_STCR_TSHFD (1 << 4) -+#define SSI_STCR_TSCKP (1 << 3) -+#define SSI_STCR_TFSI (1 << 2) -+#define SSI_STCR_TFSL (1 << 1) -+#define SSI_STCR_TEFS (1 << 0) -+ -+#define SSI_SRCR_RXBIT0 (1 << 9) -+#define SSI_SRCR_RFEN1 (1 << 8) -+#define SSI_SRCR_RFEN0 (1 << 7) -+#define SSI_SRCR_RFDIR (1 << 6) -+#define SSI_SRCR_RXDIR (1 << 5) -+#define SSI_SRCR_RSHFD (1 << 4) -+#define SSI_SRCR_RSCKP (1 << 3) -+#define SSI_SRCR_RFSI (1 << 2) -+#define SSI_SRCR_RFSL (1 << 1) -+#define SSI_SRCR_REFS (1 << 0) -+ -+#define SSI_STCCR_DIV2 (1 << 18) -+#define SSI_STCCR_PSR (1 << 15) -+#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) -+#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) -+#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) -+ -+#define SSI_SRCCR_DIV2 (1 << 18) -+#define SSI_SRCCR_PSR (1 << 15) -+#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) -+#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) -+#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) -+ -+ -+#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) -+#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) -+#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) -+#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) -+#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) -+#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) -+#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) -+#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) -+ -+#define SSI_STR_TEST (1 << 15) -+#define SSI_STR_RCK2TCK (1 << 14) -+#define SSI_STR_RFS2TFS (1 << 13) -+#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) -+#define SSI_STR_TXD2RXD (1 << 7) -+#define SSI_STR_TCK2RCK (1 << 6) -+#define SSI_STR_TFS2RFS (1 << 5) -+#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) -+ -+#define SSI_SOR_CLKOFF (1 << 6) -+#define SSI_SOR_RX_CLR (1 << 5) -+#define SSI_SOR_TX_CLR (1 << 4) -+#define SSI_SOR_INIT (1 << 3) -+#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) -+#define SSI_SOR_SYNRST (1 << 0) -+ -+#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) -+#define SSI_SACNT_WR (x << 4) -+#define SSI_SACNT_RD (x << 3) -+#define SSI_SACNT_TIF (x << 2) -+#define SSI_SACNT_FV (x << 1) -+#define SSI_SACNT_A97EN (x << 0) -+ -+ -+/* AUDMUX registers */ -+#define AUDMUX_HPCR1 __REG(IMX_AUDMUX_BASE + 0x00) -+#define AUDMUX_HPCR2 __REG(IMX_AUDMUX_BASE + 0x04) -+#define AUDMUX_HPCR3 __REG(IMX_AUDMUX_BASE + 0x08) -+#define AUDMUX_PPCR1 __REG(IMX_AUDMUX_BASE + 0x10) -+#define AUDMUX_PPCR2 __REG(IMX_AUDMUX_BASE + 0x14) -+#define AUDMUX_PPCR3 __REG(IMX_AUDMUX_BASE + 0x18) -+ -+#define AUDMUX_HPCR_TFSDIR (1 << 31) -+#define AUDMUX_HPCR_TCLKDIR (1 << 30) -+#define AUDMUX_HPCR_TFCSEL_TX (0 << 26) -+#define AUDMUX_HPCR_TFCSEL_RX (8 << 26) -+#define AUDMUX_HPCR_TFCSEL(x) (((x) & 0x7) << 26) -+#define AUDMUX_HPCR_RFSDIR (1 << 25) -+#define AUDMUX_HPCR_RCLKDIR (1 << 24) -+#define AUDMUX_HPCR_RFCSEL_TX (0 << 20) -+#define AUDMUX_HPCR_RFCSEL_RX (8 << 20) -+#define AUDMUX_HPCR_RFCSEL(x) (((x) & 0x7) << 20) -+#define AUDMUX_HPCR_RXDSEL(x) (((x) & 0x7) << 13) -+#define AUDMUX_HPCR_SYN (1 << 12) -+#define AUDMUX_HPCR_TXRXEN (1 << 10) -+#define AUDMUX_HPCR_INMEN (1 << 8) -+#define AUDMUX_HPCR_INMMASK(x) (((x) & 0xff) << 0) -+ -+#define AUDMUX_PPCR_TFSDIR (1 << 31) -+#define AUDMUX_PPCR_TCLKDIR (1 << 30) -+#define AUDMUX_PPCR_TFCSEL_TX (0 << 26) -+#define AUDMUX_PPCR_TFCSEL_RX (8 << 26) -+#define AUDMUX_PPCR_TFCSEL(x) (((x) & 0x7) << 26) -+#define AUDMUX_PPCR_RFSDIR (1 << 25) -+#define AUDMUX_PPCR_RCLKDIR (1 << 24) -+#define AUDMUX_PPCR_RFCSEL_TX (0 << 20) -+#define AUDMUX_PPCR_RFCSEL_RX (8 << 20) -+#define AUDMUX_PPCR_RFCSEL(x) (((x) & 0x7) << 20) -+#define AUDMUX_PPCR_RXDSEL(x) (((x) & 0x7) << 13) -+#define AUDMUX_PPCR_SYN (1 << 12) -+#define AUDMUX_PPCR_TXRXEN (1 << 10) -+ -+ -+#endif -Index: linux-2.6-pxa-new/sound/soc/s3c24xx/s3c24xx-i2s.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/s3c24xx/s3c24xx-i2s.c -@@ -0,0 +1,271 @@ -+/* -+ * s3c24xx-i2s.c -- ALSA Soc Audio Layer -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Graeme Gregory -+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 10th Nov 2006 Initial version. -+ */ -+ -+#include <linux/init.h> -+#include <linux/module.h> -+#include <linux/device.h> -+#include <linux/delay.h> -+#include <linux/clk.h> -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/initval.h> -+#include <sound/soc.h> -+ -+#include <asm/hardware.h> -+#include <asm/io.h> -+#include <asm/arch/regs-iis.h> -+#include <asm/arch/regs-gpio.h> -+#include <asm/arch/regs-clock.h> -+#include <asm/arch/audio.h> -+#include <asm/dma.h> -+#include <asm/arch/dma.h> -+ -+#include "s3c24xx-pcm.h" -+ -+/* used to disable sysclk if external crystal is used */ -+static int extclk = 0; -+module_param(extclk, int, 0); -+MODULE_PARM_DESC(extclk, "set to 1 to disable s3c24XX i2s sysclk"); -+ -+#define S3C24XX_I2S_DAIFMT \ -+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF) -+ -+#define S3C24XX_I2S_DIR \ -+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) -+ -+#define S3C24XX_I2S_RATES \ -+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ -+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ -+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -+ -+/* priv is divider */ -+static struct snd_soc_dai_mode s3c24xx_i2s_modes[] = -+{ -+ /* s3c24xx I2S frame and clock master modes */ -+ { -+ .fmt = S3C24XX_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, -+ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, -+ .pcmrate = SNDRV_PCM_RATE_44100, -+ .pcmdir = S3C24XX_I2S_DIR, -+ .flags = SND_SOC_DAI_BFS_RATE, -+ .fs = 384, -+ .bfs = 32, -+ .priv = 0x00 -+ }, -+}; -+ -+static struct s3c2410_dma_client s3c24xx_dma_client_out = { -+ .name = "I2S PCM Stereo out" -+}; -+ -+static struct s3c2410_dma_client s3c24xx_dma_client_in = { -+ .name = "I2S PCM Stereo in" -+}; -+ -+static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_out = { -+ .client = &s3c24xx_dma_client_out, -+ .channel = DMACH_I2S_OUT, -+ .dma_addr = S3C2410_PA_IIS+S3C2410_IISFIFO -+}; -+ -+static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_in = { -+ .client = &s3c24xx_dma_client_in, -+ .channel = DMACH_I2S_IN, -+ .dma_addr = S3C2410_PA_IIS+S3C2410_IISFIFO -+}; -+ -+ -+struct s3c24xx_i2s_port { -+ int master; -+}; -+static struct s3c24xx_i2s_port s3c24xx_i2s; -+ -+/* Empty for the s3c24xx platforms */ -+static int s3c24xx_i2s_startup(struct snd_pcm_substream *substream) -+{ -+ return 0; -+} -+ -+static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ unsigned long iiscon; -+ unsigned long iismod; -+ unsigned long iisfcon; -+ -+ s3c24xx_i2s.master = 0; -+ if(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CBS_CFS) -+ s3c24xx_i2s.master = 1; -+ -+ /* Configure the I2S pins in correct mode */ -+ s3c2410_gpio_cfgpin(S3C2410_GPE0,S3C2410_GPE0_I2SLRCK); -+ if (s3c24xx_i2s.master && !extclk){ -+ printk("Setting Clock Output as we are Master\n"); -+ s3c2410_gpio_cfgpin(S3C2410_GPE1,S3C2410_GPE1_I2SSCLK); -+ } -+ s3c2410_gpio_cfgpin(S3C2410_GPE2,S3C2410_GPE2_CDCLK); -+ s3c2410_gpio_cfgpin(S3C2410_GPE3,S3C2410_GPE3_I2SSDI); -+ s3c2410_gpio_cfgpin(S3C2410_GPE4,S3C2410_GPE4_I2SSDO); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ { -+ rtd->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; -+ } -+ else -+ { -+ rtd->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in; -+ } -+ -+ /* Working copies of registers */ -+ iiscon=readl(S3C24XX_VA_IIS+S3C2410_IISCON); -+ iismod=readl(S3C24XX_VA_IIS+S3C2410_IISMOD); -+ iisfcon=readl(S3C24XX_VA_IIS+S3C2410_IISFCON); -+ /* is port used by another stream */ -+ if (!(iiscon & S3C2410_IISCON_IISEN)) { -+ -+ /* Clear the registers */ -+ -+ iismod |= S3C2410_IISMOD_32FS | S3C2410_IISMOD_384FS; -+ -+ if (!s3c24xx_i2s.master) -+ iismod |= S3C2410_IISMOD_SLAVE; -+ -+ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_LEFT_J) -+ iismod |= S3C2410_IISMOD_MSB; -+ } -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ { -+ iismod |= S3C2410_IISMOD_TXMODE; -+ iiscon |= S3C2410_IISCON_TXDMAEN; -+ iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; -+ } -+ else -+ { -+ iismod |= S3C2410_IISMOD_RXMODE; -+ iiscon |= S3C2410_IISCON_RXDMAEN; -+ iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; -+ } -+ -+ writel(iiscon, S3C24XX_VA_IIS+S3C2410_IISCON); -+ writel(iismod, S3C24XX_VA_IIS+S3C2410_IISMOD); -+ writel(iisfcon, S3C24XX_VA_IIS+S3C2410_IISFCON); -+ -+ printk("IISCON: %lx IISMOD: %lx", iiscon, iismod); -+ -+ return 0; -+} -+ -+static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ int ret = 0; -+ unsigned long iiscon; -+ -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ /* Enable the IIS unit */ -+ iiscon = readl(S3C24XX_VA_IIS+S3C2410_IISCON); -+ iiscon |= S3C2410_IISCON_IISEN; -+ writel(iiscon, S3C24XX_VA_IIS+S3C2410_IISCON); -+ break; -+ case SNDRV_PCM_TRIGGER_RESUME: -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ case SNDRV_PCM_TRIGGER_STOP: -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ break; -+ default: -+ ret = -EINVAL; -+ } -+ -+ return ret; -+} -+ -+static void s3c24xx_i2s_shutdown(struct snd_pcm_substream *substream) -+{ -+ unsigned long iismod, iiscon; -+ -+ iismod=readl(S3C24XX_VA_IIS+S3C2410_IISMOD); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -+ iismod &= ~S3C2410_IISMOD_TXMODE; -+ } else { -+ iismod &= ~S3C2410_IISMOD_RXMODE; -+ } -+ -+ writel(iismod,S3C24XX_VA_IIS+S3C2410_IISMOD); -+ -+ iiscon=readl(S3C24XX_VA_IIS+S3C2410_IISCON); -+ -+ if (iismod & ( S3C2410_IISMOD_TXMODE | S3C2410_IISMOD_RXMODE )) { -+ iiscon &= ! S3C2410_IISCON_IISEN; -+ writel(iiscon,S3C24XX_VA_IIS+S3C2410_IISCON); -+ } -+} -+ -+#ifdef CONFIG_PM -+static int s3c24xx_i2s_suspend(struct platform_device *dev, -+ struct snd_soc_cpu_dai *dai) -+{ -+} -+ -+static int s3c24xx_i2s_resume(struct platform_device *pdev, -+ struct snd_soc_cpu_dai *dai) -+{ -+} -+ -+#else -+#define s3c24xx_i2s_suspend NULL -+#define s3c24xx_i2s_resume NULL -+#endif -+ -+/* s3c24xx I2S sysclock is always 384 FS */ -+static unsigned int s3c24xx_i2s_config_sysclk(struct snd_soc_cpu_dai *iface, -+ struct snd_soc_clock_info *info, unsigned int clk) -+{ -+ return info->rate * 384; -+} -+ -+struct snd_soc_cpu_dai s3c24xx_i2s_dai = { -+ .name = "s3c24xx-i2s", -+ .id = 0, -+ .type = SND_SOC_DAI_I2S, -+ .suspend = s3c24xx_i2s_suspend, -+ .resume = s3c24xx_i2s_resume, -+ .config_sysclk = s3c24xx_i2s_config_sysclk, -+ .playback = { -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .capture = { -+ .channels_min = 2, -+ .channels_max = 2,}, -+ .ops = { -+ .startup = s3c24xx_i2s_startup, -+ .shutdown = s3c24xx_i2s_shutdown, -+ .trigger = s3c24xx_i2s_trigger, -+ .hw_params = s3c24xx_i2s_hw_params,}, -+ .caps = { -+ .num_modes = ARRAY_SIZE(s3c24xx_i2s_modes), -+ .mode = s3c24xx_i2s_modes,}, -+}; -+ -+EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); -+ -+/* Module information */ -+MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com"); -+MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/s3c24xx/s3c24xx-pcm.c -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/s3c24xx/s3c24xx-pcm.c -@@ -0,0 +1,362 @@ -+/* -+ * s3c24xx-pcm.c -- ALSA Soc Audio Layer -+ * -+ * Copyright 2005 Wolfson Microelectronics PLC. -+ * Author: Graeme Gregory -+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com -+ * -+ * This program is free software; you can redistribute it and/or modify it -+ * under the terms of the GNU General Public License as published by the -+ * Free Software Foundation; either version 2 of the License, or (at your -+ * option) any later version. -+ * -+ * Revision history -+ * 10th Nov 2006 Initial version. -+ */ -+ -+#include <linux/module.h> -+#include <linux/init.h> -+#include <linux/platform_device.h> -+#include <linux/slab.h> -+#include <linux/dma-mapping.h> -+ -+#include <sound/driver.h> -+#include <sound/core.h> -+#include <sound/pcm.h> -+#include <sound/pcm_params.h> -+#include <sound/soc.h> -+ -+#include <asm/dma.h> -+#include <asm/io.h> -+#include <asm/hardware.h> -+#include <asm/arch/dma.h> -+#include <asm/arch/audio.h> -+ -+#include "s3c24xx-pcm.h" -+ -+static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { -+ .info = SNDRV_PCM_INFO_MMAP | -+ SNDRV_PCM_INFO_MMAP_VALID | -+ SNDRV_PCM_INFO_INTERLEAVED | -+ SNDRV_PCM_INFO_PAUSE | -+ SNDRV_PCM_INFO_RESUME, -+ .formats = SNDRV_PCM_FMTBIT_S16_LE, -+ .period_bytes_min = 32, -+ .period_bytes_max = 8192, -+ .periods_min = 1, -+ .periods_max = 8192, -+ .buffer_bytes_max = 256 * 1024, -+ .fifo_size = 32, -+}; -+ -+struct s3c24xx_runtime_data { -+ dma_addr_t dma_buffer; -+ dma_addr_t dma_buffer_end; -+ size_t period_size; -+ dma_addr_t period_ptr; -+ s3c24xx_pcm_dma_params_t *params; -+}; -+ -+/* Move the pointer onto the next period, dealing with wrap around. -+ */ -+void static next_period(struct s3c24xx_runtime_data *prtd) -+{ -+ prtd->period_ptr+=prtd->period_size; -+ if(prtd->period_ptr>=prtd->dma_buffer_end) -+ { -+ prtd->period_ptr=prtd->dma_buffer; -+ } -+} -+ -+void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, -+ void *dev_id, int size, -+ enum s3c2410_dma_buffresult result) -+{ -+ struct snd_pcm_substream *substream = dev_id; -+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; -+ -+ if(result==S3C2410_RES_OK) -+ { -+ next_period(prtd); -+ s3c2410_dma_enqueue(prtd->params->channel, substream, prtd->period_ptr, prtd->period_size); -+ } -+ snd_pcm_period_elapsed(substream); -+ -+} -+ -+static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, -+ struct snd_pcm_hw_params *params) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct s3c24xx_runtime_data *prtd = runtime->private_data; -+ struct snd_soc_pcm_runtime *rtd = substream->private_data; -+ s3c24xx_pcm_dma_params_t *dma = rtd->cpu_dai->dma_data; -+ int ret; -+ -+ printk("Entered s3c24xx hw_params\n"); -+ -+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); -+ runtime->dma_bytes = params_buffer_bytes(params); -+ -+ prtd->params=dma; -+ if(ret=s3c2410_dma_request(prtd->params->channel, -+ prtd->params->client,NULL)) -+ { -+ printk("Failed to get dma channel %d for %s\n",prtd->params->channel, -+ prtd->params->client->name); -+ return ret; -+ } -+ -+ //s3c2410_dma_setflags(prtd->params->channel,S3C2410_DMAF_AUTOSTART); -+ if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ { -+ s3c2410_dma_devconfig(prtd->params->channel, S3C2410_DMASRC_MEM, -+ S3C2410_DISRCC_INC | S3C2410_DISRCC_APB, -+ prtd->params->dma_addr); -+ } -+ else -+ { -+ s3c2410_dma_devconfig(prtd->params->channel, S3C2410_DMASRC_HW, -+ S3C2410_DISRCC_INC | S3C2410_DISRCC_APB, -+ prtd->params->dma_addr); -+ } -+ -+ s3c2410_dma_config(prtd->params->channel,2,S3C2410_DCON_HANDSHAKE); -+ -+ s3c2410_dma_set_buffdone_fn(prtd->params->channel, s3c24xx_audio_buffdone); -+ -+ prtd->dma_buffer = runtime->dma_addr; -+ prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; -+ prtd->period_size = params_period_bytes(params); -+ -+ return 0; -+} -+ -+static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) -+{ -+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; -+ -+ printk("Entered s3c24xx hw_free\n"); -+ -+ return 0; -+} -+ -+static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) -+{ -+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; -+ -+ printk("Entered s3c24xx prepare\n"); -+ -+ /* Set the period that is to be queued in DMA */ -+ prtd->period_ptr = prtd->dma_buffer; -+ -+ /* queue the first period */ -+ s3c2410_dma_enqueue(prtd->params->channel, substream, prtd->period_ptr, prtd->period_size); -+ -+ /* Move to next period to be queued */ -+ next_period(prtd); -+ -+ /* queue the second buffer */ -+ s3c2410_dma_enqueue(prtd->params->channel, substream, prtd->period_ptr, prtd->period_size); -+ -+ -+ return 0; -+} -+ -+static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -+{ -+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; -+ int ret = 0; -+ -+ printk("Entered s3c24xx trigger\n"); -+ switch (cmd) { -+ case SNDRV_PCM_TRIGGER_START: -+ case SNDRV_PCM_TRIGGER_RESUME: -+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -+ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); -+ break; -+ -+ case SNDRV_PCM_TRIGGER_STOP: -+ case SNDRV_PCM_TRIGGER_SUSPEND: -+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -+ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); -+ break; -+ -+ default: -+ ret = -EINVAL; -+ } -+ -+ return ret; -+} -+ -+static snd_pcm_uframes_t s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct s3c24xx_runtime_data *prtd = runtime->private_data; -+ dma_addr_t dst,src; -+ snd_pcm_uframes_t x; -+ -+ printk("Entered s3c24xx pointer\n"); -+ -+ s3c2410_dma_getposition(prtd->params->channel, &src, &dst); -+ -+ printk("DMA Position: %lx, %lx\n", src, dst); -+ -+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -+ { -+ x = bytes_to_frames(runtime, src - prtd->dma_buffer); -+ } -+ else -+ { -+ x = bytes_to_frames(runtime, dst - prtd->dma_buffer); -+ } -+ -+ if (x == runtime->buffer_size) -+ x=0; -+ return x; -+ -+} -+ -+static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct s3c24xx_runtime_data *prtd; -+ int ret; -+ -+ printk("Entered s3c24xx open\n"); -+ -+ snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); -+ -+ if((prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL)) == NULL) -+ { -+ ret = -ENOMEM; -+ goto out; -+ } -+ -+ runtime->private_data = prtd; -+ return 0; -+ -+out: -+ return ret; -+} -+ -+static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ struct s3c24xx_runtime_data *prtd = runtime->private_data; -+ -+ printk("Entered s3c24xx close\n"); -+ -+ s3c2410_dma_free(prtd->params->channel, prtd->params->client); -+ -+ return 0; -+} -+ -+static int -+s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) -+{ -+ struct snd_pcm_runtime *runtime = substream->runtime; -+ -+ printk("Entered s3c24xx mmap\n"); -+ -+ return dma_mmap_writecombine(substream->pcm->card->dev, vma, -+ runtime->dma_area, -+ runtime->dma_addr, -+ runtime->dma_bytes); -+} -+ -+struct snd_pcm_ops s3c24xx_pcm_ops = { -+ .open = s3c24xx_pcm_open, -+ .close = s3c24xx_pcm_close, -+ .ioctl = snd_pcm_lib_ioctl, -+ .hw_params = s3c24xx_pcm_hw_params, -+ .hw_free = s3c24xx_pcm_hw_free, -+ .prepare = s3c24xx_pcm_prepare, -+ .trigger = s3c24xx_pcm_trigger, -+ .pointer = s3c24xx_pcm_pointer, -+ .mmap = s3c24xx_pcm_mmap, -+}; -+ -+static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -+{ -+ struct snd_pcm_substream *substream = pcm->streams[stream].substream; -+ struct snd_dma_buffer *buf = &substream->dma_buffer; -+ size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; -+ -+ printk("Entered s3c24xx preaccolate_dma_buffer\n"); -+ -+ buf->dev.type = SNDRV_DMA_TYPE_DEV; -+ buf->dev.dev = pcm->card->dev; -+ buf->private_data = NULL; -+ buf->area = dma_alloc_writecombine(pcm->card->dev, size, -+ &buf->addr, GFP_KERNEL); -+ if (!buf->area) -+ return -ENOMEM; -+ buf->bytes = size; -+ return 0; -+} -+ -+static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -+{ -+ struct snd_pcm_substream *substream; -+ struct snd_dma_buffer *buf; -+ int stream; -+ -+ printk("Entered s3c24xx free_dma_buffers\n"); -+ -+ for (stream = 0; stream < 2; stream++) { -+ substream = pcm->streams[stream].substream; -+ if (!substream) -+ continue; -+ -+ buf = &substream->dma_buffer; -+ if (!buf->area) -+ continue; -+ -+ dma_free_writecombine(pcm->card->dev, buf->bytes, -+ buf->area, buf->addr); -+ buf->area = NULL; -+ } -+} -+ -+static u64 s3c24xx_pcm_dmamask = 0xffffffff; -+ -+int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, -+ struct snd_pcm *pcm) -+{ -+ int ret = 0; -+ -+ printk("Entered s3c24xx new\n"); -+ -+ if (!card->dev->dma_mask) -+ card->dev->dma_mask = &s3c24xx_pcm_dmamask; -+ if (!card->dev->coherent_dma_mask) -+ card->dev->coherent_dma_mask = 0xffffffff; -+ -+ if (dai->playback.channels_min) { -+ ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); -+ if (ret) -+ goto out; -+ } -+ -+ if (dai->capture.channels_min) { -+ ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); -+ if (ret) -+ goto out; -+ } -+ out: -+ return ret; -+} -+ -+struct snd_soc_platform s3c24xx_soc_platform = { -+ .name = "s3c24xx-audio", -+ .pcm_ops = &s3c24xx_pcm_ops, -+ .pcm_new = s3c24xx_pcm_new, -+ .pcm_free = s3c24xx_pcm_free_dma_buffers, -+}; -+ -+EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); -+ -+MODULE_AUTHOR("Graeme Gregory"); -+MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module"); -+MODULE_LICENSE("GPL"); -Index: linux-2.6-pxa-new/sound/soc/s3c24xx/Kconfig -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/s3c24xx/Kconfig -@@ -0,0 +1,26 @@ -+menu "SoC Audio for the Atmel AT91" -+ -+config SND_S3C24XX_SOC -+ tristate "SoC Audio for the Samsung S3C24xx System-on-Chip" -+ depends on ARCH_S3C2410 && SND -+ select SND_PCM -+ help -+ Say Y or M if you want to add support for codecs attached to -+ the Samsung S3C24xx. -+ -+config SND_S3C24XX_SOC_I2S -+ tristate -+ -+config SND_S3C24XX_SOC_AC97 -+ tristate -+ -+# graeme - add mach dep -+config SND_S3C24XX_SOC_SMDK2440 -+ tristate "SoC I2S Audio support for SMDK2440" -+ depends on SND_S3C24XX_SOC -+ select SND_S3C24XX_SOC_I2S -+ select SND_SOC_UDA1380 -+ help -+ Say Y if you want to add support for SoC audio on -+ -+endmenu -Index: linux-2.6-pxa-new/sound/soc/s3c24xx/Makefile -=================================================================== ---- /dev/null -+++ linux-2.6-pxa-new/sound/soc/s3c24xx/Makefile -@@ -0,0 +1,11 @@ -+# S3C24xx Platform Support -+snd-soc-s3c24xx-objs := s3c24xx-pcm.o -+snd-soc-at91-i2s-objs := s3c24xx-i2s.o -+ -+obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o -+obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o -+ -+# S3C24xx Machine Support -+snd-soc-smdk2440-uda1380-objs := smdk2440_uda1380.o -+ -+obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2440) += snd-soc-smdk2440-uda1380.o |