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Add ALSA support in libavdevice.

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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/trunk
pull/126/head
Nicolas George 16 years ago committed by Benoit Fouet
parent 1db2c5c9ef
commit 35fd81224a
7 changed files with 564 additions and 0 deletions
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  1. 8
      configure
  2. 2
      libavdevice/Makefile
  3. 1
      libavdevice/alldevices.c
  4. 186
      libavdevice/alsa-audio-common.c
  5. 175
      libavdevice/alsa-audio-dec.c
  6. 108
      libavdevice/alsa-audio-enc.c
  7. 84
      libavdevice/alsa-audio.h

8
configure vendored
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@ -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 &&

2
libavdevice/Makefile
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@ -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

1
libavdevice/alldevices.c
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@ -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);

186
libavdevice/alsa-audio-common.c
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@ -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;
}

175
libavdevice/alsa-audio-dec.c
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@ -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, &timestamp);
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,
};

108
libavdevice/alsa-audio-enc.c
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@ -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,
};

84
libavdevice/alsa-audio.h
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@ -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 */
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