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334 lines
11 KiB
334 lines
11 KiB
diff -uNr flite-1.3-release/configure.in flite-1.3-release-mod/configure.in
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--- flite-1.3-release/configure.in 2005-08-13 13:43:21.000000000 +0200
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+++ flite-1.3-release-mod/configure.in 2006-11-13 21:16:27.000000000 +0200
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@@ -206,10 +206,10 @@
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AC_CHECK_HEADER(sys/audioio.h,
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[AUDIODRIVER="sun"
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AUDIODEFS=-DCST_AUDIO_SUNOS])
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-dnl AC_CHECK_HEADER(sys/asoundlib.h,
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-dnl [AUDIODRIVER="alsa"
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-dnl AUDIODEFS=-DCST_AUDIO_ALSA
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-dnl AUDIOLIBS=-lasound])
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+AC_CHECK_HEADER(alsa/asoundlib.h,
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+ [AUDIODRIVER="alsa"
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+ AUDIODEFS=-DCST_AUDIO_ALSA
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+ AUDIOLIBS=-lasound])
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AC_CHECK_HEADER(mmsystem.h,
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[AUDIODRIVER="wince"
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AUDIODEFS=-DCST_AUDIO_WINCE
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diff -uNr flite-1.3-release/src/audio/au_alsa.c flite-1.3-release-mod/src/audio/au_alsa.c
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--- flite-1.3-release/src/audio/au_alsa.c 1970-01-01 02:00:00.000000000 +0200
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+++ flite-1.3-release-mod/src/audio/au_alsa.c 2006-11-13 21:16:54.000000000 +0200
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@@ -0,0 +1,311 @@
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+/*************************************************************************/
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+/* */
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+/* Language Technologies Institute */
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+/* Carnegie Mellon University */
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+/* Copyright (c) 2000 */
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+/* All Rights Reserved. */
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+/* */
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+/* Permission is hereby granted, free of charge, to use and distribute */
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+/* this software and its documentation without restriction, including */
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+/* without limitation the rights to use, copy, modify, merge, publish, */
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+/* distribute, sublicense, and/or sell copies of this work, and to */
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+/* permit persons to whom this work is furnished to do so, subject to */
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+/* the following conditions: */
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+/* 1. The code must retain the above copyright notice, this list of */
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+/* conditions and the following disclaimer. */
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+/* 2. Any modifications must be clearly marked as such. */
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+/* 3. Original authors' names are not deleted. */
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+/* 4. The authors' names are not used to endorse or promote products */
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+/* derived from this software without specific prior written */
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+/* permission. */
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+/* */
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+/* CARNEGIE MELLON UNIVERSITY AND THE CONTRIBUTORS TO THIS WORK */
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+/* DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING */
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+/* ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT */
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+/* SHALL CARNEGIE MELLON UNIVERSITY NOR THE CONTRIBUTORS BE LIABLE */
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+/* FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES */
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+/* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN */
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+/* AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, */
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+/* ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF */
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+/* THIS SOFTWARE. */
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+/* */
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+/*********************************************************************** */
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+/* Author: Lukas Loehrer ( */
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+/* Date: January 2005 */
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+/*************************************************************************/
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+/* */
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+/* Native access to alsa audio devices on Linux */
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+/* Tested with libasound version 1.0.10 */
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+/*************************************************************************/
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+
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+#include <stdlib.h>
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+#include <unistd.h>
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+#include <sys/types.h>
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+#include <assert.h>
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+#include <errno.h>
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+
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+#include "cst_string.h"
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+#include "cst_wave.h"
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+#include "cst_audio.h"
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+
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+#include <alsa/asoundlib.h>
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+
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+
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+/*static char *pcm_dev_name = "hw:0,0"; */
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+static char *pcm_dev_name ="default";
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+
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+static inline void print_pcm_state(snd_pcm_t *handle, char *msg)
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+{
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+ fprintf(stderr, "PCM state at %s = %s\n", msg,
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+ snd_pcm_state_name(snd_pcm_state(handle)));
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+}
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+
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+cst_audiodev *audio_open_alsa(int sps, int channels, cst_audiofmt fmt)
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+{
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+ cst_audiodev *ad;
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+ unsigned int real_rate;
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+ int err;
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+
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+ /* alsa specific stuff */
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+ snd_pcm_t *pcm_handle;
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+ snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
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+ snd_pcm_hw_params_t *hwparams;
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+ snd_pcm_format_t format;
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+ snd_pcm_access_t access = SND_PCM_ACCESS_RW_INTERLEAVED;
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+
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+ /* Allocate the snd_pcm_hw_params_t structure on the stack. */
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+ snd_pcm_hw_params_alloca(&hwparams);
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+
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+ /* Open pcm device */
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+ err = snd_pcm_open(&pcm_handle, pcm_dev_name, stream, 0);
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+ if (err < 0)
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+ {
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+ cst_errmsg("audio_open_alsa: failed to open audio device %s. %s\n",
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+ pcm_dev_name, snd_strerror(err));
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+ return NULL;
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+ }
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+
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+ /* Init hwparams with full configuration space */
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+ err = snd_pcm_hw_params_any(pcm_handle, hwparams);
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+ if (err < 0)
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+ {
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+ snd_pcm_close(pcm_handle);
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+ cst_errmsg("audio_open_alsa: failed to get hardware parameters from audio device. %s\n", snd_strerror(err));
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+ return NULL;
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+ }
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+
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+ /* Set access mode */
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+ err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, access);
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+ if (err < 0)
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+ {
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+ snd_pcm_close(pcm_handle);
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+ cst_errmsg("audio_open_alsa: failed to set access mode. %s.\n", snd_strerror(err));
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+ return NULL;
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+ }
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+
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+ /* Determine matching alsa sample format */
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+ /* This could be implemented in a more */
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+ /* flexible way (byte order conversion). */
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+ switch (fmt)
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+ {
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+ case CST_AUDIO_LINEAR16:
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+ if (CST_LITTLE_ENDIAN)
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+ format = SND_PCM_FORMAT_S16_LE;
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+ else
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+ format = SND_PCM_FORMAT_S16_BE;
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+ break;
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+ case CST_AUDIO_LINEAR8:
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+ format = SND_PCM_FORMAT_U8;
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+ break;
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+ case CST_AUDIO_MULAW:
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+ format = SND_PCM_FORMAT_MU_LAW;
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+ break;
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+ default:
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+ snd_pcm_close(pcm_handle);
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+ cst_errmsg("audio_open_alsa: failed to find suitable format.\n");
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+ return NULL;
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+ break;
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+ }
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+
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+ /* Set samble format */
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+ err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, format);
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+ if (err <0)
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+ {
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+ snd_pcm_close(pcm_handle);
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+ cst_errmsg("audio_open_alsa: failed to set format. %s.\n", snd_strerror(err));
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+ return NULL;
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+ }
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+
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+ /* Set sample rate near the disired rate */
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+ real_rate = sps;
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+ err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &real_rate, 0);
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+ if (err < 0)
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+ {
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+ snd_pcm_close(pcm_handle);
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+ cst_errmsg("audio_open_alsa: failed to set sample rate near %d. %s.\n", sps, snd_strerror(err));
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+ return NULL;
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+ }
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+ /*FIXME: This is probably too strict */
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+ assert(sps == real_rate);
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+
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+ /* Set number of channels */
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+ assert(channels >0);
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+ err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, channels);
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+ if (err < 0)
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+ {
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+ snd_pcm_close(pcm_handle);
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+ cst_errmsg("audio_open_alsa: failed to set number of channels to %d. %s.\n", channels, snd_strerror(err));
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+ return NULL;
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+ }
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+
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+ /* Commit hardware parameters */
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+ err = snd_pcm_hw_params(pcm_handle, hwparams);
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+ if (err < 0)
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+ {
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+ snd_pcm_close(pcm_handle);
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+ cst_errmsg("audio_open_alsa: failed to set hw parameters. %s.\n", snd_strerror(err));
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+ return NULL;
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+ }
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+
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+ /* Make sure the device is ready to accept data */
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+ assert(snd_pcm_state(pcm_handle) == SND_PCM_STATE_PREPARED);
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+
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+ /* Write hardware parameters to flite audio device data structure */
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+ ad = cst_alloc(cst_audiodev, 1);
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+ assert(ad != NULL);
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+ ad->real_sps = ad->sps = sps;
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+ ad->real_channels = ad->channels = channels;
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+ ad->real_fmt = ad->fmt = fmt;
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+ ad->platform_data = (void *) pcm_handle;
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+
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+ return ad;
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+}
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+
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+int audio_close_alsa(cst_audiodev *ad)
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+{
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+ int result;
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+ snd_pcm_t *pcm_handle;
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+
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+ if (ad == NULL)
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+ return 0;
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+
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+ pcm_handle = (snd_pcm_t *) ad->platform_data;
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+ result = snd_pcm_close(pcm_handle);
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+ if (result < 0)
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+ {
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+ cst_errmsg("audio_close_alsa: Error: %s.\n", snd_strerror(result));
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+ }
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+ cst_free(ad);
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+ return result;
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+}
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+
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+/* Returns zero if recovery was successful. */
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+static int recover_from_error(snd_pcm_t *pcm_handle, ssize_t res)
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+{
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+ if (res == -EPIPE) /* xrun */
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+ {
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+ res = snd_pcm_prepare(pcm_handle);
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+ if (res < 0)
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+ {
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+ /* Failed to recover from xrun */
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+ cst_errmsg("recover_from_write_error: failed to recover from xrun. %s\n.", snd_strerror(res));
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+ return res;
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+ }
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+ }
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+ else if (res == -ESTRPIPE) /* Suspend */
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+ {
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+ while ((res = snd_pcm_resume(pcm_handle)) == -EAGAIN)
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+ {
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+ snd_pcm_wait(pcm_handle, 1000);
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+ }
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+ if (res < 0)
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+ {
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+ res = snd_pcm_prepare(pcm_handle);
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+ if (res <0)
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+ {
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+ /* Resume failed */
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+ cst_errmsg("audio_recover_from_write_error: failed to resume after suspend. %s\n.", snd_strerror(res));
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+ return res;
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+ }
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+ }
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+ }
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+ else if (res < 0)
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+ {
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+ /* Unknown failure */
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+ cst_errmsg("audio_recover_from_write_error: %s.\n", snd_strerror(res));
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+ return res;
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+ }
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+ return 0;
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+}
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+
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+int audio_write_alsa(cst_audiodev *ad, void *samples, int num_bytes)
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+{
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+ size_t frame_size;
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+ ssize_t num_frames, res;
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+ snd_pcm_t *pcm_handle;
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+ char *buf = (char *) samples;
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+
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+ /* Determine frame size in bytes */
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+ frame_size = audio_bps(ad->real_fmt) * ad->real_channels;
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+ /* Require that only complete frames are handed in */
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+ assert((num_bytes % frame_size) == 0);
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+ num_frames = num_bytes / frame_size;
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+ pcm_handle = (snd_pcm_t *) ad->platform_data;
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+
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+ while (num_frames > 0)
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+ {
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+ res = snd_pcm_writei(pcm_handle, buf, num_frames);
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+ if (res != num_frames)
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+ {
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+ if (res == -EAGAIN || (res > 0 && res < num_frames))
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+ {
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+ snd_pcm_wait(pcm_handle, 100);
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+ }
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+ else if (recover_from_error(pcm_handle, res) < 0)
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+ {
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+ return -1;
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+ }
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+ }
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+
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+ if (res >0)
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+ {
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+ num_frames -= res;
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+ buf += res * frame_size;
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+ }
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+ }
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+ return num_bytes;
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+}
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+
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+int audio_flush_alsa(cst_audiodev *ad)
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+{
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+ int result;
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+ result = snd_pcm_drain((snd_pcm_t *) ad->platform_data);
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+ if (result < 0)
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+ {
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+ cst_errmsg("audio_flush_alsa: Error: %s.\n", snd_strerror(result));
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+ }
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+ /* Prepare device for more data */
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+ result = snd_pcm_prepare((snd_pcm_t *) ad->platform_data);
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+if (result < 0)
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+ {
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+ cst_errmsg("audio_flush_alsa: Error: %s.\n", snd_strerror(result));
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+ }
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+ return result;
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+}
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+
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+int audio_drain_alsa(cst_audiodev *ad)
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+{
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+ int result;
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+ result = snd_pcm_drop((snd_pcm_t *) ad->platform_data);
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+ if (result < 0)
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+ {
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+ cst_errmsg("audio_drain_alsa: Error: %s.\n", snd_strerror(result));
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+ }
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+/* Prepare device for more data */
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+ result = snd_pcm_prepare((snd_pcm_t *) ad->platform_data);
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+if (result < 0)
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+ {
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+ cst_errmsg("audio_drain_alsa: Error: %s.\n", snd_strerror(result));
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+ }
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+ return result;
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+}
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