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Commit d46cd249 authored by Diego Biurrun's avatar Diego Biurrun
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alsa: Coalesce source files after outdev removal

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libavdevice/Makefile
+ 1
− 1
View file @ d46cd249
...@@ -10,7 +10,7 @@ OBJS = alldevices.o \ ...@@ -10,7 +10,7 @@ OBJS = alldevices.o \
OBJS-$(HAVE_LIBC_MSVCRT) += file_open.o OBJS-$(HAVE_LIBC_MSVCRT) += file_open.o
# input devices # input devices
OBJS-$(CONFIG_ALSA_INDEV) += alsa_dec.o alsa.o OBJS-$(CONFIG_ALSA_INDEV) += alsa.o
OBJS-$(CONFIG_AVFOUNDATION_INDEV) += avfoundation_dec.o OBJS-$(CONFIG_AVFOUNDATION_INDEV) += avfoundation_dec.o
OBJS-$(CONFIG_BKTR_INDEV) += bktr.o OBJS-$(CONFIG_BKTR_INDEV) += bktr.o
OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o
......
libavdevice/alsa.c
+ 181
− 9
View file @ d46cd249
/* /*
* ALSA input and output * ALSA input
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
* *
...@@ -22,18 +22,39 @@ ...@@ -22,18 +22,39 @@
/** /**
* @file * @file
* ALSA input and output: common code * ALSA input
* @author Luca Abeni ( lucabe72 email it ) * @author Luca Abeni ( lucabe72 email it )
* @author Benoit Fouet ( benoit fouet free fr ) * @author Benoit Fouet ( benoit fouet free fr )
* @author Nicolas George ( nicolas george normalesup org ) * @author Nicolas George ( nicolas george normalesup org )
*/ */
#include <alsa/asoundlib.h> #include <alsa/asoundlib.h>
#include "libavformat/avformat.h"
#include "libavutil/avassert.h" #include "libavutil/avassert.h"
#include "libavutil/channel_layout.h" #include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "alsa.h" /* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
other formats */
#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
#define ALSA_BUFFER_SIZE_MAX 32768
typedef struct AlsaData {
AVClass *class;
snd_pcm_t *h;
int frame_size; ///< preferred size for reads and writes
int period_size; ///< bytes per sample * channels
int sample_rate; ///< sample rate set by user
int channels; ///< number of channels set by user
void (*reorder_func)(const void *, void *, int);
void *reorder_buf;
int reorder_buf_size; ///< in frames
} AlsaData;
static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id) static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
{ {
...@@ -183,9 +204,23 @@ static av_cold int find_reorder_func(AlsaData *s, int codec_id, uint64_t layout, ...@@ -183,9 +204,23 @@ static av_cold int find_reorder_func(AlsaData *s, int codec_id, uint64_t layout,
return s->reorder_func ? 0 : AVERROR(ENOSYS); return s->reorder_func ? 0 : AVERROR(ENOSYS);
} }
av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode, /**
unsigned int *sample_rate, * Open an ALSA PCM.
int channels, enum AVCodecID *codec_id) *
* @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 AVCodecID or AV_CODEC_ID_NONE;
* out: actually selected AVCodecID, changed only if
* AV_CODEC_ID_NONE was requested
*
* @return 0 if OK, AVERROR_xxx on error
*/
static av_cold int alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
unsigned int *sample_rate,
int channels, enum AVCodecID *codec_id)
{ {
AlsaData *s = ctx->priv_data; AlsaData *s = ctx->priv_data;
const char *audio_device; const char *audio_device;
...@@ -315,7 +350,14 @@ fail1: ...@@ -315,7 +350,14 @@ fail1:
return AVERROR(EIO); return AVERROR(EIO);
} }
av_cold int ff_alsa_close(AVFormatContext *s1) /**
* Close the ALSA PCM.
*
* @param s1 media file handle
*
* @return 0
*/
static av_cold int alsa_close(AVFormatContext *s1)
{ {
AlsaData *s = s1->priv_data; AlsaData *s = s1->priv_data;
...@@ -324,7 +366,15 @@ av_cold int ff_alsa_close(AVFormatContext *s1) ...@@ -324,7 +366,15 @@ av_cold int ff_alsa_close(AVFormatContext *s1)
return 0; return 0;
} }
int ff_alsa_xrun_recover(AVFormatContext *s1, int err) /**
* Try 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
*/
static int alsa_xrun_recover(AVFormatContext *s1, int err)
{ {
AlsaData *s = s1->priv_data; AlsaData *s = s1->priv_data;
snd_pcm_t *handle = s->h; snd_pcm_t *handle = s->h;
...@@ -344,3 +394,125 @@ int ff_alsa_xrun_recover(AVFormatContext *s1, int err) ...@@ -344,3 +394,125 @@ int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
} }
return err; return err;
} }
static av_cold int audio_read_header(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
AVStream *st;
int ret;
enum AVCodecID codec_id;
snd_pcm_sw_params_t *sw_params;
st = avformat_new_stream(s1, NULL);
if (!st) {
av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
codec_id = s1->audio_codec_id;
ret = alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = codec_id;
st->codecpar->sample_rate = s->sample_rate;
st->codecpar->channels = s->channels;
avpriv_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_packet_unref(pkt);
return AVERROR(EAGAIN);
}
if (alsa_xrun_recover(s1, res) < 0) {
av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
snd_strerror(res));
av_packet_unref(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->codecpar->sample_rate
- (int64_t)ts_delay * 1000000000LL + st->codecpar->sample_rate * 500LL)
/ (st->codecpar->sample_rate * 1000LL);
pkt->size = res * s->frame_size;
return 0;
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass alsa_demuxer_class = {
.class_name = "ALSA demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_alsa_demuxer = {
.name = "alsa",
.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
.priv_data_size = sizeof(AlsaData),
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = alsa_close,
.flags = AVFMT_NOFILE,
.priv_class = &alsa_demuxer_class,
};
libavdevice/alsa.h deleted 100644 → 0
+ 0
− 94
View file @ 6ce13070
/*
* 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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ALSA input and output: definitions and structures
* @author Luca Abeni ( lucabe72 email it )
* @author Benoit Fouet ( benoit fouet free fr )
*/
#ifndef AVDEVICE_ALSA_H
#define AVDEVICE_ALSA_H
#include <alsa/asoundlib.h>
#include "config.h"
#include "libavformat/avformat.h"
#include "libavutil/log.h"
/* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
other formats */
#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
#define ALSA_BUFFER_SIZE_MAX 32768
typedef struct AlsaData {
AVClass *class;
snd_pcm_t *h;
int frame_size; ///< preferred size for reads and writes
int period_size; ///< bytes per sample * channels
int sample_rate; ///< sample rate set by user
int channels; ///< number of channels set by user
void (*reorder_func)(const void *, void *, int);
void *reorder_buf;
int reorder_buf_size; ///< in frames
} AlsaData;
/**
* Open 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 AVCodecID or AV_CODEC_ID_NONE;
* out: actually selected AVCodecID, changed only if
* AV_CODEC_ID_NONE was requested
*
* @return 0 if OK, AVERROR_xxx on error
*/
int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
unsigned int *sample_rate,
int channels, enum AVCodecID *codec_id);
/**
* Close the ALSA PCM.
*
* @param s1 media file handle
*
* @return 0
*/
int ff_alsa_close(AVFormatContext *s1);
/**
* Try 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_H */
libavdevice/alsa_dec.c deleted 100644 → 0
+ 0
− 178
View file @ 6ce13070
/*
* 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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* 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 <alsa/asoundlib.h>
#include "libavutil/internal.h"
#include "libavutil/opt.h"
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "alsa.h"
static av_cold int audio_read_header(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
AVStream *st;
int ret;
enum AVCodecID codec_id;
snd_pcm_sw_params_t *sw_params;
st = avformat_new_stream(s1, NULL);
if (!st) {
av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
codec_id = s1->audio_codec_id;
ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = codec_id;
st->codecpar->sample_rate = s->sample_rate;
st->codecpar->channels = s->channels;
avpriv_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_packet_unref(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_packet_unref(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->codecpar->sample_rate
- (int64_t)ts_delay * 1000000000LL + st->codecpar->sample_rate * 500LL)
/ (st->codecpar->sample_rate * 1000LL);
pkt->size = res * s->frame_size;
return 0;
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass alsa_demuxer_class = {
.class_name = "ALSA demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_alsa_demuxer = {
.name = "alsa",
.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
.priv_data_size = sizeof(AlsaData),
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = ff_alsa_close,
.flags = AVFMT_NOFILE,
.priv_class = &alsa_demuxer_class,
};
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