Patch by Nicolas George: name surname normalesup org Original thread: [FFmpeg-devel] [PATCH] ALSA for libavdevice Date: 12/09/2008 07:17 PM Originally committed as revision 16800 to svn://svn.ffmpeg.org/ffmpeg/trunktags/v0.5
@@ -838,6 +838,7 @@ ARCH_EXT_LIST=' | |||
HAVE_LIST=" | |||
$ARCH_EXT_LIST | |||
$THREADS_LIST | |||
alsa_asoundlib_h | |||
altivec_h | |||
arpa_inet_h | |||
bswap | |||
@@ -1069,6 +1070,10 @@ vdpau_deps="vdpau_vdpau_h vdpau_vdpau_x11_h" | |||
# demuxers / muxers | |||
ac3_demuxer_deps="ac3_parser" | |||
alsa_demuxer_deps="alsa_asoundlib_h snd_pcm_htimestamp" | |||
alsa_demuxer_extralibs="-lasound" | |||
alsa_muxer_deps="alsa_asoundlib_h" | |||
alsa_muxer_extralibs="-lasound" | |||
audio_beos_demuxer_deps="audio_beos" | |||
audio_beos_demuxer_extralibs="-lmedia -lbe" | |||
audio_beos_muxer_deps="audio_beos" | |||
@@ -2044,6 +2049,9 @@ check_header dev/ic/bt8xx.h | |||
check_header sys/soundcard.h | |||
check_header soundcard.h | |||
check_header alsa/asoundlib.h && | |||
check_lib2 alsa/asoundlib.h snd_pcm_htimestamp -lasound | |||
# deal with the X11 frame grabber | |||
enabled x11grab && | |||
check_header X11/Xlib.h && | |||
@@ -8,6 +8,8 @@ HEADERS = avdevice.h | |||
OBJS = alldevices.o | |||
# input/output devices | |||
OBJS-$(CONFIG_ALSA_DEMUXER) += alsa-audio-common.o alsa-audio-dec.o | |||
OBJS-$(CONFIG_ALSA_MUXER) += alsa-audio-common.o alsa-audio-enc.o | |||
OBJS-$(CONFIG_BKTR_DEMUXER) += bktr.o | |||
OBJS-$(CONFIG_DV1394_DEMUXER) += dv1394.o | |||
OBJS-$(CONFIG_OSS_DEMUXER) += oss_audio.o | |||
@@ -44,6 +44,7 @@ void avdevice_register_all(void) | |||
initialized = 1; | |||
/* devices */ | |||
REGISTER_MUXDEMUX (ALSA, alsa); | |||
REGISTER_MUXDEMUX (AUDIO_BEOS, audio_beos); | |||
REGISTER_DEMUXER (BKTR, bktr); | |||
REGISTER_DEMUXER (DV1394, dv1394); | |||
@@ -0,0 +1,186 @@ | |||
/* | |||
* ALSA input and output | |||
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) | |||
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) | |||
* | |||
* This file is part of FFmpeg. | |||
* | |||
* FFmpeg is free software; you can redistribute it and/or | |||
* modify it under the terms of the GNU Lesser General Public | |||
* License as published by the Free Software Foundation; either | |||
* version 2.1 of the License, or (at your option) any later version. | |||
* | |||
* FFmpeg is distributed in the hope that it will be useful, | |||
* but WITHOUT ANY WARRANTY; without even the implied warranty of | |||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |||
* Lesser General Public License for more details. | |||
* | |||
* You should have received a copy of the GNU Lesser General Public | |||
* License along with FFmpeg; if not, write to the Free Software | |||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |||
*/ | |||
/** | |||
* @file alsa-audio-common.c | |||
* ALSA input and output: common code | |||
* @author Luca Abeni ( lucabe72 email it ) | |||
* @author Benoit Fouet ( benoit fouet free fr ) | |||
* @author Nicolas George ( nicolas george normalesup org ) | |||
*/ | |||
#include "libavformat/avformat.h" | |||
#include <alsa/asoundlib.h> | |||
#include "alsa-audio.h" | |||
static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id) | |||
{ | |||
switch(codec_id) { | |||
case CODEC_ID_PCM_S16LE: return SND_PCM_FORMAT_S16_LE; | |||
case CODEC_ID_PCM_S16BE: return SND_PCM_FORMAT_S16_BE; | |||
case CODEC_ID_PCM_S8: return SND_PCM_FORMAT_S8; | |||
default: return SND_PCM_FORMAT_UNKNOWN; | |||
} | |||
} | |||
av_cold int ff_alsa_open(AVFormatContext *ctx, int mode, | |||
unsigned int *sample_rate, | |||
int channels, int *codec_id) | |||
{ | |||
AlsaData *s = ctx->priv_data; | |||
const char *audio_device; | |||
int res, flags = 0; | |||
snd_pcm_format_t format; | |||
snd_pcm_t *h; | |||
snd_pcm_hw_params_t *hw_params; | |||
snd_pcm_uframes_t buffer_size, period_size; | |||
if (ctx->filename[0] == 0) audio_device = "default"; | |||
else audio_device = ctx->filename; | |||
if (*codec_id == CODEC_ID_NONE) | |||
*codec_id = DEFAULT_CODEC_ID; | |||
format = codec_id_to_pcm_format(*codec_id); | |||
if (format == SND_PCM_FORMAT_UNKNOWN) { | |||
av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id); | |||
return AVERROR(ENOSYS); | |||
} | |||
s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels; | |||
if (ctx->flags & AVFMT_FLAG_NONBLOCK) { | |||
flags = O_NONBLOCK; | |||
} | |||
res = snd_pcm_open(&h, audio_device, mode, flags); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n", | |||
audio_device, snd_strerror(res)); | |||
return AVERROR_IO; | |||
} | |||
res = snd_pcm_hw_params_malloc(&hw_params); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot allocate hardware parameter structure (%s)\n", | |||
snd_strerror(res)); | |||
goto fail1; | |||
} | |||
res = snd_pcm_hw_params_any(h, hw_params); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot initialize hardware parameter structure (%s)\n", | |||
snd_strerror(res)); | |||
goto fail; | |||
} | |||
res = snd_pcm_hw_params_set_access(h, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot set access type (%s)\n", | |||
snd_strerror(res)); | |||
goto fail; | |||
} | |||
res = snd_pcm_hw_params_set_format(h, hw_params, format); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot set sample format 0x%04x %d (%s)\n", | |||
*codec_id, format, snd_strerror(res)); | |||
goto fail; | |||
} | |||
res = snd_pcm_hw_params_set_rate_near(h, hw_params, sample_rate, 0); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot set sample rate (%s)\n", | |||
snd_strerror(res)); | |||
goto fail; | |||
} | |||
res = snd_pcm_hw_params_set_channels(h, hw_params, channels); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n", | |||
channels, snd_strerror(res)); | |||
goto fail; | |||
} | |||
snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size); | |||
/* TODO: maybe use ctx->max_picture_buffer somehow */ | |||
res = snd_pcm_hw_params_set_buffer_size_near(h, hw_params, &buffer_size); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot set ALSA buffer size (%s)\n", | |||
snd_strerror(res)); | |||
goto fail; | |||
} | |||
snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL); | |||
res = snd_pcm_hw_params_set_period_size_near(h, hw_params, &period_size, NULL); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot set ALSA period size (%s)\n", | |||
snd_strerror(res)); | |||
goto fail; | |||
} | |||
s->period_size = period_size; | |||
res = snd_pcm_hw_params(h, hw_params); | |||
if (res < 0) { | |||
av_log(ctx, AV_LOG_ERROR, "cannot set parameters (%s)\n", | |||
snd_strerror(res)); | |||
goto fail; | |||
} | |||
snd_pcm_hw_params_free(hw_params); | |||
s->h = h; | |||
return 0; | |||
fail: | |||
snd_pcm_hw_params_free(hw_params); | |||
fail1: | |||
snd_pcm_close(h); | |||
return AVERROR_IO; | |||
} | |||
av_cold int ff_alsa_close(AVFormatContext *s1) | |||
{ | |||
AlsaData *s = s1->priv_data; | |||
snd_pcm_close(s->h); | |||
return 0; | |||
} | |||
int ff_alsa_xrun_recover(AVFormatContext *s1, int err) | |||
{ | |||
AlsaData *s = s1->priv_data; | |||
snd_pcm_t *handle = s->h; | |||
av_log(s1, AV_LOG_WARNING, "ALSA buffer xrun.\n"); | |||
if (err == -EPIPE) { | |||
err = snd_pcm_prepare(handle); | |||
if (err < 0) { | |||
av_log(s1, AV_LOG_ERROR, "cannot recover from underrun (snd_pcm_prepare failed: %s)\n", snd_strerror(err)); | |||
return AVERROR_IO; | |||
} | |||
} else if (err == -ESTRPIPE) { | |||
av_log(s1, AV_LOG_ERROR, "-ESTRPIPE... Unsupported!\n"); | |||
return -1; | |||
} | |||
return err; | |||
} |
@@ -0,0 +1,175 @@ | |||
/* | |||
* ALSA input and output | |||
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) | |||
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) | |||
* | |||
* This file is part of FFmpeg. | |||
* | |||
* FFmpeg is free software; you can redistribute it and/or | |||
* modify it under the terms of the GNU Lesser General Public | |||
* License as published by the Free Software Foundation; either | |||
* version 2.1 of the License, or (at your option) any later version. | |||
* | |||
* FFmpeg is distributed in the hope that it will be useful, | |||
* but WITHOUT ANY WARRANTY; without even the implied warranty of | |||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |||
* Lesser General Public License for more details. | |||
* | |||
* You should have received a copy of the GNU Lesser General Public | |||
* License along with FFmpeg; if not, write to the Free Software | |||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |||
*/ | |||
/** | |||
* @file alsa-audio-dec.c | |||
* ALSA input and output: input | |||
* @author Luca Abeni ( lucabe72 email it ) | |||
* @author Benoit Fouet ( benoit fouet free fr ) | |||
* @author Nicolas George ( nicolas george normalesup org ) | |||
* | |||
* This avdevice decoder allows to capture audio from an ALSA (Advanced | |||
* Linux Sound Architecture) device. | |||
* | |||
* The filename parameter is the name of an ALSA PCM device capable of | |||
* capture, for example "default" or "plughw:1"; see the ALSA documentation | |||
* for naming conventions. The empty string is equivalent to "default". | |||
* | |||
* The capture period is set to the lower value available for the device, | |||
* which gives a low latency suitable for real-time capture. | |||
* | |||
* The PTS are an Unix time in microsecond. | |||
* | |||
* Due to a bug in the ALSA library | |||
* (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this | |||
* decoder does not work with certain ALSA plugins, especially the dsnoop | |||
* plugin. | |||
*/ | |||
#include "libavformat/avformat.h" | |||
#include <alsa/asoundlib.h> | |||
#include "alsa-audio.h" | |||
av_cold static int audio_read_header(AVFormatContext *s1, | |||
AVFormatParameters *ap) | |||
{ | |||
AlsaData *s = s1->priv_data; | |||
AVStream *st; | |||
int ret; | |||
unsigned int sample_rate; | |||
int codec_id; | |||
snd_pcm_sw_params_t *sw_params; | |||
if (ap->sample_rate <= 0) { | |||
av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate); | |||
return AVERROR(EIO); | |||
} | |||
if (ap->channels <= 0) { | |||
av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels); | |||
return AVERROR(EIO); | |||
} | |||
st = av_new_stream(s1, 0); | |||
if (!st) { | |||
av_log(s1, AV_LOG_ERROR, "Cannot add stream\n"); | |||
return AVERROR(ENOMEM); | |||
} | |||
sample_rate = ap->sample_rate; | |||
codec_id = ap->audio_codec_id; | |||
ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels, | |||
&codec_id); | |||
if (ret < 0) { | |||
return AVERROR(EIO); | |||
} | |||
if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW) | |||
av_log(s1, AV_LOG_WARNING, | |||
"capture with some ALSA plugins, especially dsnoop, " | |||
"may hang.\n"); | |||
ret = snd_pcm_sw_params_malloc(&sw_params); | |||
if (ret < 0) { | |||
av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n", | |||
snd_strerror(ret)); | |||
goto fail; | |||
} | |||
snd_pcm_sw_params_current(s->h, sw_params); | |||
snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE); | |||
ret = snd_pcm_sw_params(s->h, sw_params); | |||
snd_pcm_sw_params_free(sw_params); | |||
if (ret < 0) { | |||
av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n", | |||
snd_strerror(ret)); | |||
goto fail; | |||
} | |||
/* take real parameters */ | |||
st->codec->codec_type = CODEC_TYPE_AUDIO; | |||
st->codec->codec_id = codec_id; | |||
st->codec->sample_rate = sample_rate; | |||
st->codec->channels = ap->channels; | |||
av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ | |||
return 0; | |||
fail: | |||
snd_pcm_close(s->h); | |||
return AVERROR(EIO); | |||
} | |||
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) | |||
{ | |||
AlsaData *s = s1->priv_data; | |||
AVStream *st = s1->streams[0]; | |||
int res; | |||
snd_htimestamp_t timestamp; | |||
snd_pcm_uframes_t ts_delay; | |||
if (av_new_packet(pkt, s->period_size) < 0) { | |||
return AVERROR(EIO); | |||
} | |||
while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) { | |||
if (res == -EAGAIN) { | |||
av_free_packet(pkt); | |||
return AVERROR(EAGAIN); | |||
} | |||
if (ff_alsa_xrun_recover(s1, res) < 0) { | |||
av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n", | |||
snd_strerror(res)); | |||
av_free_packet(pkt); | |||
return AVERROR(EIO); | |||
} | |||
} | |||
snd_pcm_htimestamp(s->h, &ts_delay, ×tamp); | |||
ts_delay += res; | |||
pkt->pts = timestamp.tv_sec * 1000000LL | |||
+ (timestamp.tv_nsec * st->codec->sample_rate | |||
- ts_delay * 1000000000LL + st->codec->sample_rate * 500LL) | |||
/ (st->codec->sample_rate * 1000LL); | |||
pkt->size = res * s->frame_size; | |||
return 0; | |||
} | |||
AVInputFormat alsa_demuxer = { | |||
"alsa", | |||
NULL_IF_CONFIG_SMALL("ALSA audio input"), | |||
sizeof(AlsaData), | |||
NULL, | |||
audio_read_header, | |||
audio_read_packet, | |||
ff_alsa_close, | |||
.flags = AVFMT_NOFILE, | |||
}; |
@@ -0,0 +1,108 @@ | |||
/* | |||
* ALSA input and output | |||
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) | |||
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) | |||
* | |||
* This file is part of FFmpeg. | |||
* | |||
* FFmpeg is free software; you can redistribute it and/or | |||
* modify it under the terms of the GNU Lesser General Public | |||
* License as published by the Free Software Foundation; either | |||
* version 2.1 of the License, or (at your option) any later version. | |||
* | |||
* FFmpeg is distributed in the hope that it will be useful, | |||
* but WITHOUT ANY WARRANTY; without even the implied warranty of | |||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |||
* Lesser General Public License for more details. | |||
* | |||
* You should have received a copy of the GNU Lesser General Public | |||
* License along with FFmpeg; if not, write to the Free Software | |||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |||
*/ | |||
/** | |||
* @file alsa-audio-enc.c | |||
* ALSA input and output: output | |||
* @author Luca Abeni ( lucabe72 email it ) | |||
* @author Benoit Fouet ( benoit fouet free fr ) | |||
* | |||
* This avdevice encoder allows to play audio to an ALSA (Advanced Linux | |||
* Sound Architecture) device. | |||
* | |||
* The filename parameter is the name of an ALSA PCM device capable of | |||
* capture, for example "default" or "plughw:1"; see the ALSA documentation | |||
* for naming conventions. The empty string is equivalent to "default". | |||
* | |||
* The playback period is set to the lower value available for the device, | |||
* which gives a low latency suitable for real-time playback. | |||
*/ | |||
#include "libavformat/avformat.h" | |||
#include <alsa/asoundlib.h> | |||
#include "alsa-audio.h" | |||
av_cold static int audio_write_header(AVFormatContext *s1) | |||
{ | |||
AlsaData *s = s1->priv_data; | |||
AVStream *st; | |||
unsigned int sample_rate; | |||
int codec_id; | |||
int res; | |||
st = s1->streams[0]; | |||
sample_rate = st->codec->sample_rate; | |||
codec_id = st->codec->codec_id; | |||
res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate, | |||
st->codec->channels, &codec_id); | |||
if (sample_rate != st->codec->sample_rate) { | |||
av_log(s1, AV_LOG_ERROR, | |||
"sample rate %d not available, nearest is %d\n", | |||
st->codec->sample_rate, sample_rate); | |||
goto fail; | |||
} | |||
return res; | |||
fail: | |||
snd_pcm_close(s->h); | |||
return AVERROR(EIO); | |||
} | |||
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) | |||
{ | |||
AlsaData *s = s1->priv_data; | |||
int res; | |||
int size = pkt->size; | |||
uint8_t *buf = pkt->data; | |||
while((res = snd_pcm_writei(s->h, buf, size / s->frame_size)) < 0) { | |||
if (res == -EAGAIN) { | |||
return AVERROR(EAGAIN); | |||
} | |||
if (ff_alsa_xrun_recover(s1, res) < 0) { | |||
av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n", | |||
snd_strerror(res)); | |||
return AVERROR(EIO); | |||
} | |||
} | |||
return 0; | |||
} | |||
AVOutputFormat alsa_muxer = { | |||
"alsa", | |||
NULL_IF_CONFIG_SMALL("ALSA audio output"), | |||
"", | |||
"", | |||
sizeof(AlsaData), | |||
DEFAULT_CODEC_ID, | |||
CODEC_ID_NONE, | |||
audio_write_header, | |||
audio_write_packet, | |||
ff_alsa_close, | |||
.flags = AVFMT_NOFILE, | |||
}; |
@@ -0,0 +1,84 @@ | |||
/* | |||
* ALSA input and output | |||
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) | |||
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) | |||
* | |||
* This file is part of FFmpeg. | |||
* | |||
* FFmpeg is free software; you can redistribute it and/or | |||
* modify it under the terms of the GNU Lesser General Public | |||
* License as published by the Free Software Foundation; either | |||
* version 2.1 of the License, or (at your option) any later version. | |||
* | |||
* FFmpeg is distributed in the hope that it will be useful, | |||
* but WITHOUT ANY WARRANTY; without even the implied warranty of | |||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |||
* Lesser General Public License for more details. | |||
* | |||
* You should have received a copy of the GNU Lesser General Public | |||
* License along with FFmpeg; if not, write to the Free Software | |||
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |||
*/ | |||
/** | |||
* @file alsa-audio.h | |||
* ALSA input and output: definitions and structures | |||
* @author Luca Abeni ( lucabe72 email it ) | |||
* @author Benoit Fouet ( benoit fouet free fr ) | |||
*/ | |||
#ifndef AVDEVICE_ALSA_AUDIO_H | |||
#define AVDEVICE_ALSA_AUDIO_H | |||
/* XXX: we make the assumption that the soundcard accepts this format */ | |||
/* XXX: find better solution with "preinit" method, needed also in | |||
other formats */ | |||
#ifdef WORDS_BIGENDIAN | |||
#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16BE | |||
#else | |||
#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16LE | |||
#endif | |||
typedef struct { | |||
snd_pcm_t *h; | |||
int frame_size; ///< preferred size for reads and writes | |||
int period_size; ///< bytes per sample * channels | |||
} AlsaData; | |||
/** | |||
* Opens an ALSA PCM. | |||
* | |||
* @param s media file handle | |||
* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK | |||
* @param sample_rate in: requested sample rate; | |||
* out: actually selected sample rate | |||
* @param channels number of channels | |||
* @param codec_id in: requested CodecID or CODEC_ID_NONE; | |||
* out: actually selected CodecID, changed only if | |||
* CODEC_ID_NONE was requested | |||
* | |||
* @return 0 if OK, AVERROR_xxx on error | |||
*/ | |||
int ff_alsa_open(AVFormatContext *s, int mode, unsigned int *sample_rate, | |||
int channels, int *codec_id); | |||
/** | |||
* Closes the ALSA PCM. | |||
* | |||
* @param s1 media file handle | |||
* | |||
* @return 0 | |||
*/ | |||
int ff_alsa_close(AVFormatContext *s1); | |||
/** | |||
* Tries to recover from ALSA buffer underrun. | |||
* | |||
* @param s1 media file handle | |||
* @param err error code reported by the previous ALSA call | |||
* | |||
* @return 0 if OK, AVERROR_xxx on error | |||
*/ | |||
int ff_alsa_xrun_recover(AVFormatContext *s1, int err); | |||
#endif /* AVDEVICE_ALSA_AUDIO_H */ |