@@ -10,7 +10,7 @@ OBJS = alldevices.o \ | |||
OBJS-$(HAVE_LIBC_MSVCRT) += file_open.o | |||
# input devices | |||
OBJS-$(CONFIG_ALSA_INDEV) += alsa_dec.o alsa.o | |||
OBJS-$(CONFIG_ALSA_INDEV) += alsa.o | |||
OBJS-$(CONFIG_AVFOUNDATION_INDEV) += avfoundation_dec.o | |||
OBJS-$(CONFIG_BKTR_INDEV) += bktr.o | |||
OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o | |||
@@ -1,5 +1,5 @@ | |||
/* | |||
* ALSA input and output | |||
* ALSA input | |||
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) | |||
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) | |||
* | |||
@@ -22,18 +22,39 @@ | |||
/** | |||
* @file | |||
* ALSA input and output: common code | |||
* ALSA input | |||
* @author Luca Abeni ( lucabe72 email it ) | |||
* @author Benoit Fouet ( benoit fouet free fr ) | |||
* @author Nicolas George ( nicolas george normalesup org ) | |||
*/ | |||
#include <alsa/asoundlib.h> | |||
#include "libavformat/avformat.h" | |||
#include "libavutil/avassert.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) | |||
{ | |||
@@ -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); | |||
} | |||
av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode, | |||
unsigned int *sample_rate, | |||
int channels, enum AVCodecID *codec_id) | |||
/** | |||
* 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 | |||
*/ | |||
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; | |||
const char *audio_device; | |||
@@ -315,7 +350,14 @@ fail1: | |||
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; | |||
@@ -324,7 +366,15 @@ av_cold int ff_alsa_close(AVFormatContext *s1) | |||
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; | |||
snd_pcm_t *handle = s->h; | |||
@@ -344,3 +394,125 @@ int ff_alsa_xrun_recover(AVFormatContext *s1, int 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, ×tamp); | |||
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, | |||
}; |
@@ -1,94 +0,0 @@ | |||
/* | |||
* 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 */ |
@@ -1,178 +0,0 @@ | |||
/* | |||
* 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, ×tamp); | |||
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, | |||
}; |