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File flite-1.3-alsa_support.patch of Package flite
diff -uNr flite-1.3-release/configure.in flite-1.3-release-mod/configure.in --- flite-1.3-release/configure.in 2005-08-13 13:43:21.000000000 +0200 +++ flite-1.3-release-mod/configure.in 2006-11-13 21:16:27.000000000 +0200 @@ -206,10 +206,10 @@ AC_CHECK_HEADER(sys/audioio.h, [AUDIODRIVER="sun" AUDIODEFS=-DCST_AUDIO_SUNOS]) -dnl AC_CHECK_HEADER(sys/asoundlib.h, -dnl [AUDIODRIVER="alsa" -dnl AUDIODEFS=-DCST_AUDIO_ALSA -dnl AUDIOLIBS=-lasound]) +AC_CHECK_HEADER(alsa/asoundlib.h, + [AUDIODRIVER="alsa" + AUDIODEFS=-DCST_AUDIO_ALSA + AUDIOLIBS=-lasound]) AC_CHECK_HEADER(mmsystem.h, [AUDIODRIVER="wince" AUDIODEFS=-DCST_AUDIO_WINCE diff -uNr flite-1.3-release/src/audio/au_alsa.c flite-1.3-release-mod/src/audio/au_alsa.c --- flite-1.3-release/src/audio/au_alsa.c 1970-01-01 02:00:00.000000000 +0200 +++ flite-1.3-release-mod/src/audio/au_alsa.c 2006-11-13 21:16:54.000000000 +0200 @@ -0,0 +1,311 @@ +/*************************************************************************/ +/* */ +/* Language Technologies Institute */ +/* Carnegie Mellon University */ +/* Copyright (c) 2000 */ +/* All Rights Reserved. */ +/* */ +/* Permission is hereby granted, free of charge, to use and distribute */ +/* this software and its documentation without restriction, including */ +/* without limitation the rights to use, copy, modify, merge, publish, */ +/* distribute, sublicense, and/or sell copies of this work, and to */ +/* permit persons to whom this work is furnished to do so, subject to */ +/* the following conditions: */ +/* 1. The code must retain the above copyright notice, this list of */ +/* conditions and the following disclaimer. */ +/* 2. Any modifications must be clearly marked as such. */ +/* 3. Original authors' names are not deleted. */ +/* 4. The authors' names are not used to endorse or promote products */ +/* derived from this software without specific prior written */ +/* permission. */ +/* */ +/* CARNEGIE MELLON UNIVERSITY AND THE CONTRIBUTORS TO THIS WORK */ +/* DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING */ +/* ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT */ +/* SHALL CARNEGIE MELLON UNIVERSITY NOR THE CONTRIBUTORS BE LIABLE */ +/* FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES */ +/* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN */ +/* AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, */ +/* ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF */ +/* THIS SOFTWARE. */ +/* */ +/*********************************************************************** */ +/* Author: Lukas Loehrer ( */ +/* Date: January 2005 */ +/*************************************************************************/ +/* */ +/* Native access to alsa audio devices on Linux */ +/* Tested with libasound version 1.0.10 */ +/*************************************************************************/ + +#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 <alsa/asoundlib.h> + + +/*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) +{ + cst_audiodev *ad; + unsigned int real_rate; + int err; + + /* 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); + + /* 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; + } + + /* 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; + } + + /* 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; + } + + /* 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) + format = SND_PCM_FORMAT_S16_LE; + else + format = SND_PCM_FORMAT_S16_BE; + break; + case CST_AUDIO_LINEAR8: + format = SND_PCM_FORMAT_U8; + break; + 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; + } + + /* 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; + } + + /* 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); + + /* 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; + } + + /* 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) +{ + int result; + snd_pcm_t *pcm_handle; + + if (ad == NULL) + return 0; + + 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; +} + +/* 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) +{ + size_t frame_size; + ssize_t num_frames, res; + snd_pcm_t *pcm_handle; + char *buf = (char *) samples; + + /* 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) +{ + 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) +{ + 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; +}
