diff options
Diffstat (limited to 'packages/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff')
| -rw-r--r-- | packages/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff | 512 |
1 files changed, 512 insertions, 0 deletions
diff --git a/packages/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff b/packages/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff new file mode 100644 index 0000000000..3d2753a01f --- /dev/null +++ b/packages/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff @@ -0,0 +1,512 @@ +Index: configure +=================================================================== +--- flite-1.2-release/configure (.../flite-1.2-orig) (revision 10) ++++ flite-1.2-release/configure (.../release-v1.2) (revision 10) +@@ -1415,16 +1415,16 @@ + echo "$ac_t""no" 1>&6 + fi + +-ac_safe=`echo "sys/asoundlib.h" | sed 'y%./+-%__p_%'` +-echo $ac_n "checking for sys/asoundlib.h""... $ac_c" 1>&6 +-echo "configure:1421: checking for sys/asoundlib.h" >&5 ++ac_safe=`echo "alsa/asoundlib.h" | sed 'y%./+-%__p_%'` ++echo $ac_n "checking for alsa/asoundlib.h""... $ac_c" 1>&6 ++echo "configure:1421: checking for alsa/asoundlib.h" >&5 + if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then + echo $ac_n "(cached) $ac_c" 1>&6 + else + cat > conftest.$ac_ext <<EOF + #line 1426 "configure" + #include "confdefs.h" +-#include <sys/asoundlib.h> ++#include <alsa/asoundlib.h> + EOF + ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out" + { (eval echo configure:1431: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; } +@@ -1445,23 +1445,24 @@ + echo "$ac_t""yes" 1>&6 + AUDIODRIVER="alsa" + AUDIODEFS=-DCST_AUDIO_ALSA ++ AUDIOLIBS=-lasound + else + echo "$ac_t""no" 1>&6 + fi + + ac_safe=`echo "mmsystem.h" | sed 'y%./+-%__p_%'` + echo $ac_n "checking for mmsystem.h""... $ac_c" 1>&6 +-echo "configure:1455: checking for mmsystem.h" >&5 ++echo "configure:1456: checking for mmsystem.h" >&5 + if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then + echo $ac_n "(cached) $ac_c" 1>&6 + else + cat > conftest.$ac_ext <<EOF +-#line 1460 "configure" ++#line 1461 "configure" + #include "confdefs.h" + #include <mmsystem.h> + EOF + ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out" +-{ (eval echo configure:1465: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; } ++{ (eval echo configure:1466: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; } + ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"` + if test -z "$ac_err"; then + rm -rf conftest* +Index: include/cst_sts.h +=================================================================== +--- flite-1.2-release/include/cst_sts.h (.../flite-1.2-orig) (revision 10) ++++ flite-1.2-release/include/cst_sts.h (.../release-v1.2) (revision 10) +@@ -47,9 +47,9 @@ + /* else where, this information plus the indexes in the Unit relation */ + /* allow reconstruction of the signal itself */ + struct cst_sts_struct { +- const unsigned short *frame; +- const int size; /* in samples */ +- const unsigned char *residual; ++ unsigned short *frame; ++ int size; /* in samples */ ++ unsigned char *residual; + }; + typedef struct cst_sts_struct cst_sts; + +Index: configure.in +=================================================================== +--- flite-1.2-release/configure.in (.../flite-1.2-orig) (revision 10) ++++ flite-1.2-release/configure.in (.../release-v1.2) (revision 10) +@@ -131,9 +131,10 @@ + AC_CHECK_HEADER(sys/audioio.h, + [AUDIODRIVER="sun" + AUDIODEFS=-DCST_AUDIO_SUNOS]) +-AC_CHECK_HEADER(sys/asoundlib.h, ++AC_CHECK_HEADER(alsa/asoundlib.h, + [AUDIODRIVER="alsa" +- AUDIODEFS=-DCST_AUDIO_ALSA]) ++ AUDIODEFS=-DCST_AUDIO_ALSA ++ AUDIOLIBS=-lasound]) + AC_CHECK_HEADER(mmsystem.h, + [AUDIODRIVER="wince" + AUDIODEFS=-DCST_AUDIO_WINCE +Index: src/audio/au_alsa.c +=================================================================== +--- flite-1.2-release/src/audio/au_alsa.c (.../flite-1.2-orig) (revision 10) ++++ flite-1.2-release/src/audio/au_alsa.c (.../release-v1.2) (revision 10) +@@ -2,7 +2,7 @@ + /* */ + /* Language Technologies Institute */ + /* Carnegie Mellon University */ +-/* Copyright (c) 2001 */ ++/* Copyright (c) 2000 */ + /* All Rights Reserved. */ + /* */ + /* Permission is hereby granted, free of charge, to use and distribute */ +@@ -29,158 +29,283 @@ + /* ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF */ + /* THIS SOFTWARE. */ + /* */ ++/*********************************************************************** */ ++/* Author: Lukas Loehrer ( */ ++/* Date: January 2005 */ + /*************************************************************************/ +-/* Author: Geoff Harrison (mandrake@cepstral.com) */ +-/* Date: Sepetember 2001 */ +-/*************************************************************************/ + /* */ +-/* Access to ALSA audio devices */ +-/* */ ++/* Native access to alsa audio devices on Linux */ ++/* Tested with libasound version 1.0.10 */ + /*************************************************************************/ + +-#include <stdio.h> + #include <stdlib.h> + #include <unistd.h> + #include <sys/types.h> ++#include <assert.h> ++#include <errno.h> ++ + #include "cst_string.h" + #include "cst_wave.h" + #include "cst_audio.h" + +-#include <sys/asoundlib.h> ++#include <alsa/asoundlib.h> + +-#include <sys/stat.h> +-#include <fcntl.h> + +-static int alsa_card = 0, alsa_device = 0; ++/*static char *pcm_dev_name = "hw:0,0"; */ ++static char *pcm_dev_name ="default"; + ++static inline void print_pcm_state(snd_pcm_t *handle, char *msg) ++{ ++ fprintf(stderr, "PCM state at %s = %s\n", msg, ++ snd_pcm_state_name(snd_pcm_state(handle))); ++} ++ + cst_audiodev *audio_open_alsa(int sps, int channels, cst_audiofmt fmt) + { +- snd_pcm_channel_info_t pinfo; +- snd_pcm_channel_params_t params; +- snd_pcm_channel_setup_t setup; +- snd_pcm_t *pcm; +- cst_audiodev *ad; +- int err; ++ cst_audiodev *ad; ++ unsigned int real_rate; ++ int err; + +-#ifdef __QNXNTO__ +- if (snd_pcm_open_preferred(&pcm,&alsa_card,&alsa_device,SND_PCM_OPEN_PLAYBACK) < 0) +- { +- cst_errmsg("alsa_audio: failed to open audio device\n"); +- cst_error(); +- } +- if (snd_pcm_plugin_set_disable(pcm,PLUGIN_DISABLE_MMAP) < 0) +- { +- cst_errmsg("alsa_audio: failed to disable mmap\n"); +- snd_pcm_close(pcm); +- cst_error(); +- } +-#else +- if (snd_pcm_open(&pcm,alsa_card,alsa_device,SND_PCM_OPEN_PLAYBACK) < 0) +- { +- cst_errmsg("alsa_audio: failed to open audio device\n"); +- cst_error(); +- } +-#endif ++ /* alsa specific stuff */ ++ snd_pcm_t *pcm_handle; ++ snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; ++ snd_pcm_hw_params_t *hwparams; ++ snd_pcm_format_t format; ++ snd_pcm_access_t access = SND_PCM_ACCESS_RW_INTERLEAVED; + ++ /* Allocate the snd_pcm_hw_params_t structure on the stack. */ ++ snd_pcm_hw_params_alloca(&hwparams); + +- memset(&pinfo, 0, sizeof(pinfo)); +- memset(¶ms, 0, sizeof(params)); +- memset(&setup, 0, sizeof(setup)); ++ /* Open pcm device */ ++ err = snd_pcm_open(&pcm_handle, pcm_dev_name, stream, 0); ++ if (err < 0) ++ { ++ cst_errmsg("audio_open_alsa: failed to open audio device %s. %s\n", ++ pcm_dev_name, snd_strerror(err)); ++ return NULL; ++ } + +- pinfo.channel = SND_PCM_CHANNEL_PLAYBACK; +- snd_pcm_plugin_info(pcm,&pinfo); ++ /* Init hwparams with full configuration space */ ++ err = snd_pcm_hw_params_any(pcm_handle, hwparams); ++ if (err < 0) ++ { ++ snd_pcm_close(pcm_handle); ++ cst_errmsg("audio_open_alsa: failed to get hardware parameters from audio device. %s\n", snd_strerror(err)); ++ return NULL; ++ } + +- params.mode = SND_PCM_MODE_BLOCK; +- params.channel = SND_PCM_CHANNEL_PLAYBACK; +- params.start_mode = SND_PCM_START_DATA; +- params.stop_mode = SND_PCM_STOP_STOP; ++ /* Set access mode */ ++ err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, access); ++ if (err < 0) ++ { ++ snd_pcm_close(pcm_handle); ++ cst_errmsg("audio_open_alsa: failed to set access mode. %s.\n", snd_strerror(err)); ++ return NULL; ++ } + +- params.buf.block.frag_size = pinfo.max_fragment_size; +- params.buf.block.frags_max = 1; +- params.buf.block.frags_min = 1; +- +- params.format.interleave = 1; +- params.format.rate = sps; +- params.format.voices = channels; +- +- switch (fmt) +- { +- case CST_AUDIO_LINEAR16: ++ /* Determine matching alsa sample format */ ++ /* This could be implemented in a more */ ++ /* flexible way (byte order conversion). */ ++ switch (fmt) ++ { ++ case CST_AUDIO_LINEAR16: + if (CST_LITTLE_ENDIAN) +- params.format.format = SND_PCM_SFMT_S16_LE; ++ format = SND_PCM_FORMAT_S16_LE; + else +- params.format.format = SND_PCM_SFMT_S16_BE; ++ format = SND_PCM_FORMAT_S16_BE; + break; +- case CST_AUDIO_LINEAR8: +- params.format.format = SND_PCM_SFMT_U8; ++ case CST_AUDIO_LINEAR8: ++ format = SND_PCM_FORMAT_U8; + break; +- case CST_AUDIO_MULAW: +- params.format.format = SND_PCM_SFMT_MU_LAW; ++ case CST_AUDIO_MULAW: ++ format = SND_PCM_FORMAT_MU_LAW; + break; +- } ++ default: ++ snd_pcm_close(pcm_handle); ++ cst_errmsg("audio_open_alsa: failed to find suitable format.\n"); ++ return NULL; ++ break; ++ } + +- if((err = snd_pcm_plugin_params(pcm,¶ms)) < 0) +- { +- cst_errmsg("alsa_audio params setting failed: %s\n",snd_strerror(err)); +- snd_pcm_close(pcm); +- cst_error(); +- } +- if((err = snd_pcm_plugin_setup(pcm,SND_PCM_CHANNEL_PLAYBACK)) > 0) { +- cst_errmsg("alsa_audio sound prepare setting failed: %s\n",snd_strerror(err)); +- snd_pcm_close(pcm); +- cst_error(); +- } +- if((err = snd_pcm_plugin_prepare(pcm,SND_PCM_CHANNEL_PLAYBACK)) > 0) { +- cst_errmsg("alsa_audio sound prepare setting failed: %s\n",snd_strerror(err)); +- snd_pcm_close(pcm); +- cst_error(); +- } ++ /* Set samble format */ ++ err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, format); ++ if (err <0) ++ { ++ snd_pcm_close(pcm_handle); ++ cst_errmsg("audio_open_alsa: failed to set format. %s.\n", snd_strerror(err)); ++ return NULL; ++ } + +- pinfo.channel = SND_PCM_CHANNEL_PLAYBACK; +- snd_pcm_plugin_info(pcm,&pinfo); ++ /* Set sample rate near the disired rate */ ++ real_rate = sps; ++ err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &real_rate, 0); ++ if (err < 0) ++ { ++ snd_pcm_close(pcm_handle); ++ cst_errmsg("audio_open_alsa: failed to set sample rate near %d. %s.\n", sps, snd_strerror(err)); ++ return NULL; ++ } ++ /*FIXME: This is probably too strict */ ++ assert(sps == real_rate); + +- ad = cst_alloc(cst_audiodev, 1); +- ad->platform_data = pcm; +- ad->sps = ad->real_sps = sps; +- ad->channels = ad->real_channels = channels; +- ad->fmt = ad->real_fmt = fmt; ++ /* Set number of channels */ ++ assert(channels >0); ++ err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, channels); ++ if (err < 0) ++ { ++ snd_pcm_close(pcm_handle); ++ cst_errmsg("audio_open_alsa: failed to set number of channels to %d. %s.\n", channels, snd_strerror(err)); ++ return NULL; ++ } + +- return ad; ++ /* Commit hardware parameters */ ++ err = snd_pcm_hw_params(pcm_handle, hwparams); ++ if (err < 0) ++ { ++ snd_pcm_close(pcm_handle); ++ cst_errmsg("audio_open_alsa: failed to set hw parameters. %s.\n", snd_strerror(err)); ++ return NULL; ++ } ++ ++ /* Make sure the device is ready to accept data */ ++ assert(snd_pcm_state(pcm_handle) == SND_PCM_STATE_PREPARED); ++ ++ /* Write hardware parameters to flite audio device data structure */ ++ ad = cst_alloc(cst_audiodev, 1); ++ assert(ad != NULL); ++ ad->real_sps = ad->sps = sps; ++ ad->real_channels = ad->channels = channels; ++ ad->real_fmt = ad->fmt = fmt; ++ ad->platform_data = (void *) pcm_handle; ++ ++ return ad; + } + + int audio_close_alsa(cst_audiodev *ad) + { +- snd_pcm_t *pcm; ++ int result; ++ snd_pcm_t *pcm_handle; + +- if (ad == NULL) +- return 0; ++ if (ad == NULL) ++ return 0; + +- pcm = ad->platform_data; +- snd_pcm_plugin_flush(pcm,0); +- snd_pcm_close(pcm); +- cst_free(ad); ++ pcm_handle = (snd_pcm_t *) ad->platform_data; ++ result = snd_pcm_close(pcm_handle); ++ if (result < 0) ++ { ++ cst_errmsg("audio_close_alsa: Error: %s.\n", snd_strerror(result)); ++ } ++ cst_free(ad); ++ return result; ++} + +- return 0; ++/* Returns zero if recovery was successful. */ ++static int recover_from_error(snd_pcm_t *pcm_handle, ssize_t res) ++{ ++ if (res == -EPIPE) /* xrun */ ++ { ++ res = snd_pcm_prepare(pcm_handle); ++ if (res < 0) ++ { ++ /* Failed to recover from xrun */ ++ cst_errmsg("recover_from_write_error: failed to recover from xrun. %s\n.", snd_strerror(res)); ++ return res; ++ } ++ } ++ else if (res == -ESTRPIPE) /* Suspend */ ++ { ++ while ((res = snd_pcm_resume(pcm_handle)) == -EAGAIN) ++ { ++ snd_pcm_wait(pcm_handle, 1000); ++ } ++ if (res < 0) ++ { ++ res = snd_pcm_prepare(pcm_handle); ++ if (res <0) ++ { ++ /* Resume failed */ ++ cst_errmsg("audio_recover_from_write_error: failed to resume after suspend. %s\n.", snd_strerror(res)); ++ return res; ++ } ++ } ++ } ++ else if (res < 0) ++ { ++ /* Unknown failure */ ++ cst_errmsg("audio_recover_from_write_error: %s.\n", snd_strerror(res)); ++ return res; ++ } ++ return 0; + } + + int audio_write_alsa(cst_audiodev *ad, void *samples, int num_bytes) + { +- snd_pcm_t *pcm = ad->platform_data; ++ size_t frame_size; ++ ssize_t num_frames, res; ++ snd_pcm_t *pcm_handle; ++ char *buf = (char *) samples; + +- return snd_pcm_plugin_write(pcm,samples,num_bytes); ++ /* Determine frame size in bytes */ ++ frame_size = audio_bps(ad->real_fmt) * ad->real_channels; ++ /* Require that only complete frames are handed in */ ++ assert((num_bytes % frame_size) == 0); ++ num_frames = num_bytes / frame_size; ++ pcm_handle = (snd_pcm_t *) ad->platform_data; ++ ++ while (num_frames > 0) ++ { ++ res = snd_pcm_writei(pcm_handle, buf, num_frames); ++ if (res != num_frames) ++ { ++ if (res == -EAGAIN || (res > 0 && res < num_frames)) ++ { ++ snd_pcm_wait(pcm_handle, 100); ++ } ++ else if (recover_from_error(pcm_handle, res) < 0) ++ { ++ return -1; ++ } ++ } ++ ++ if (res >0) ++ { ++ num_frames -= res; ++ buf += res * frame_size; ++ } ++ } ++ return num_bytes; + } + + int audio_flush_alsa(cst_audiodev *ad) + { +- snd_pcm_t *pcm = ad->platform_data; +- +- return snd_pcm_plugin_flush(pcm,0); ++ int result; ++ result = snd_pcm_drain((snd_pcm_t *) ad->platform_data); ++ if (result < 0) ++ { ++ cst_errmsg("audio_flush_alsa: Error: %s.\n", snd_strerror(result)); ++ } ++ /* Prepare device for more data */ ++ result = snd_pcm_prepare((snd_pcm_t *) ad->platform_data); ++if (result < 0) ++ { ++ cst_errmsg("audio_flush_alsa: Error: %s.\n", snd_strerror(result)); ++ } ++ return result; + } + + int audio_drain_alsa(cst_audiodev *ad) + { +- snd_pcm_t *pcm = ad->platform_data; +- +- return snd_pcm_plugin_playback_drain(pcm); ++ int result; ++ result = snd_pcm_drop((snd_pcm_t *) ad->platform_data); ++ if (result < 0) ++ { ++ cst_errmsg("audio_drain_alsa: Error: %s.\n", snd_strerror(result)); ++ } ++/* Prepare device for more data */ ++ result = snd_pcm_prepare((snd_pcm_t *) ad->platform_data); ++if (result < 0) ++ { ++ cst_errmsg("audio_drain_alsa: Error: %s.\n", snd_strerror(result)); ++ } ++ return result; + } +- +Index: doc/Makefile +=================================================================== +--- flite-1.2-release/doc/Makefile (.../flite-1.2-orig) (revision 10) ++++ flite-1.2-release/doc/Makefile (.../release-v1.2) (revision 10) +@@ -53,6 +53,7 @@ + @ if [ ! -d html ] ; \ + then mkdir -p html ; fi + (cd html; texi2html -number -split_chapter ../flite.texi) ++ mv html/flite/*.html html/ && rmdir html/flite + @ for i in html/*.html ; \ + do \ + sed 's/<BODY>/<BODY bgcolor="#ffffff">/' $$i >ttt.html; \ +Index: config/common_make_rules +=================================================================== +--- flite-1.2-release/config/common_make_rules (.../flite-1.2-orig) (revision 10) ++++ flite-1.2-release/config/common_make_rules (.../release-v1.2) (revision 10) +@@ -88,7 +88,7 @@ + @ rm -rf shared_os && mkdir shared_os + @ rm -f $@ $(LIBDIR)/$@.${PROJECT_VERSION} $(LIBDIR)/$@.${PROJECT_SHLIB_VERSION} + @ (cd shared_os && ar x ../$<) +- @ (cd shared_os && $(CC) -shared -Wl,-soname,$@.${PROJECT_SHLIB_VERSION} -o ../$@.${PROJECT_VERSION} *.os) ++ @ (cd shared_os && $(CC) -shared -Wl,-soname,$@.${PROJECT_SHLIB_VERSION} -o ../$@.${PROJECT_VERSION} *.os $(AUDIOLIBS)) + @ ln -s $(LIBDIR)/$@.${PROJECT_VERSION} $(LIBDIR)/$@.${PROJECT_SHLIB_VERSION} + @ ln -s $(LIBDIR)/$@.${PROJECT_SHLIB_VERSION} $(LIBDIR)/$@ + @ rm -rf shared_os |