Chiebot-Mirror
/
FFmpeg
mirror of https://github.com/FFmpeg/FFmpeg.git
8
0
Fork 0
Code Issues Projects Releases Wiki Activity

Merge commit 'd6e1d37100af568211f28ec0bcf7958a3a2a299e'

Browse Source 
* commit 'd6e1d37100af568211f28ec0bcf7958a3a2a299e':
  oss_audio: Split muxer and demuxer

Conflicts:
	libavdevice/Makefile
	libavdevice/oss_audio.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
pull/77/merge
Michael Niedermayer 10 years ago
parent 54cba3f53e d6e1d37100
commit 80acedae3e
5 changed files with 324 additions and 211 deletions
Whitespace
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. 4
      libavdevice/Makefile
  2. 219
      libavdevice/oss_audio.c
  3. 45
      libavdevice/oss_audio.h
  4. 149
      libavdevice/oss_audio_dec.c
  5. 118
      libavdevice/oss_audio_enc.c

4
libavdevice/Makefile
Unescape View File

@ -31,8 +31,8 @@ OBJS-$(CONFIG_JACK_INDEV) += jack_audio.o timefilter.o
OBJS-$(CONFIG_LAVFI_INDEV) += lavfi.o
OBJS-$(CONFIG_OPENAL_INDEV) += openal-dec.o
OBJS-$(CONFIG_OPENGL_OUTDEV) += opengl_enc.o
OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o
OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o
OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o oss_audio_dec.o
OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o oss_audio_enc.o
OBJS-$(CONFIG_PULSE_INDEV) += pulse_audio_dec.o \
pulse_audio_common.o
OBJS-$(CONFIG_PULSE_OUTDEV) += pulse_audio_enc.o \

219
libavdevice/oss_audio.c
Unescape View File

@ -20,47 +20,32 @@
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <stdint.h>
#include <string.h>
#include <errno.h>
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
#if HAVE_UNISTD_H
#include <unistd.h>
#endif
#include <fcntl.h>
#include <sys/ioctl.h>
#include "libavutil/internal.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavcodec/avcodec.h"
#include "avdevice.h"
#include "libavformat/internal.h"
#define AUDIO_BLOCK_SIZE 4096
#include "oss_audio.h"
typedef struct AudioData {
AVClass *class;
int fd;
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
enum AVCodecID codec_id;
unsigned int flip_left : 1;
uint8_t buffer[AUDIO_BLOCK_SIZE];
int buffer_ptr;
} AudioData;
static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
int ff_oss_audio_open(AVFormatContext *s1, int is_output,
const char *audio_device)
{
AudioData *s = s1->priv_data;
OSSAudioData *s = s1->priv_data;
int audio_fd;
int tmp, err;
char *flip = getenv("AUDIO_FLIP_LEFT");
@ -85,7 +70,7 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
}
}
s->frame_size = AUDIO_BLOCK_SIZE;
s->frame_size = OSS_AUDIO_BLOCK_SIZE;
#define CHECK_IOCTL_ERROR(event) \
if (err < 0) { \
@ -149,192 +134,8 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
#undef CHECK_IOCTL_ERROR
}
static int audio_close(AudioData *s)
int ff_oss_audio_close(OSSAudioData *s)
{
close(s->fd);
return 0;
}
/* sound output support */
static int audio_write_header(AVFormatContext *s1)
{
AudioData *s = s1->priv_data;
AVStream *st;
int ret;
st = s1->streams[0];
s->sample_rate = st->codec->sample_rate;
s->channels = st->codec->channels;
ret = audio_open(s1, 1, s1->filename);
if (ret < 0) {
return AVERROR(EIO);
} else {
return 0;
}
}
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = s1->priv_data;
int len, ret;
int size= pkt->size;
uint8_t *buf= pkt->data;
while (size > 0) {
len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
memcpy(s->buffer + s->buffer_ptr, buf, len);
s->buffer_ptr += len;
if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
for(;;) {
ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
if (ret > 0)
break;
if (ret < 0 && (errno != EAGAIN && errno != EINTR))
return AVERROR(EIO);
}
s->buffer_ptr = 0;
}
buf += len;
size -= len;
}
return 0;
}
static int audio_write_trailer(AVFormatContext *s1)
{
AudioData *s = s1->priv_data;
audio_close(s);
return 0;
}
/* grab support */
static int audio_read_header(AVFormatContext *s1)
{
AudioData *s = s1->priv_data;
AVStream *st;
int ret;
st = avformat_new_stream(s1, NULL);
if (!st) {
return AVERROR(ENOMEM);
}
ret = audio_open(s1, 0, s1->filename);
if (ret < 0) {
return AVERROR(EIO);
}
/* take real parameters */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = s->codec_id;
st->codec->sample_rate = s->sample_rate;
st->codec->channels = s->channels;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = s1->priv_data;
int ret, bdelay;
int64_t cur_time;
struct audio_buf_info abufi;
if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
return ret;
ret = read(s->fd, pkt->data, pkt->size);
if (ret <= 0){
av_free_packet(pkt);
pkt->size = 0;
if (ret<0) return AVERROR(errno);
else return AVERROR_EOF;
}
pkt->size = ret;
/* compute pts of the start of the packet */
cur_time = av_gettime();
bdelay = ret;
if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
bdelay += abufi.bytes;
}
/* subtract time represented by the number of bytes in the audio fifo */
cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
/* convert to wanted units */
pkt->pts = cur_time;
if (s->flip_left && s->channels == 2) {
int i;
short *p = (short *) pkt->data;
for (i = 0; i < ret; i += 4) {
*p = ~*p;
p += 2;
}
}
return 0;
}
static int audio_read_close(AVFormatContext *s1)
{
AudioData *s = s1->priv_data;
audio_close(s);
return 0;
}
#if CONFIG_OSS_INDEV
static const AVOption options[] = {
{ "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass oss_demuxer_class = {
.class_name = "OSS demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
};
AVInputFormat ff_oss_demuxer = {
.name = "oss",
.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
.priv_data_size = sizeof(AudioData),
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = audio_read_close,
.flags = AVFMT_NOFILE,
.priv_class = &oss_demuxer_class,
};
#endif
#if CONFIG_OSS_OUTDEV
static const AVClass oss_muxer_class = {
.class_name = "OSS muxer",
.item_name = av_default_item_name,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
};
AVOutputFormat ff_oss_muxer = {
.name = "oss",
.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
.priv_data_size = sizeof(AudioData),
/* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
other formats */
.audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
.video_codec = AV_CODEC_ID_NONE,
.write_header = audio_write_header,
.write_packet = audio_write_packet,
.write_trailer = audio_write_trailer,
.flags = AVFMT_NOFILE,
.priv_class = &oss_muxer_class,
};
#endif

45
libavdevice/oss_audio.h
Unescape View File

@ -0,0 +1,45 @@
/*
* 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
*/
#ifndef AVDEVICE_OSS_AUDIO_H
#define AVDEVICE_OSS_AUDIO_H
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#define OSS_AUDIO_BLOCK_SIZE 4096
typedef struct OSSAudioData {
AVClass *class;
int fd;
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
enum AVCodecID codec_id;
unsigned int flip_left : 1;
uint8_t buffer[OSS_AUDIO_BLOCK_SIZE];
int buffer_ptr;
} OSSAudioData;
int ff_oss_audio_open(AVFormatContext *s1, int is_output,
const char *audio_device);
int ff_oss_audio_close(OSSAudioData *s);
#endif /* AVDEVICE_OSS_AUDIO_H */

149
libavdevice/oss_audio_dec.c
Unescape View File

@ -0,0 +1,149 @@
/*
* Linux audio play interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* 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
*/
#include "config.h"
#include <stdint.h>
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
#if HAVE_UNISTD_H
#include <unistd.h>
#endif
#include <fcntl.h>
#include <sys/ioctl.h>
#include "libavutil/internal.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavcodec/avcodec.h"
#include "avdevice.h"
#include "libavformat/internal.h"
#include "oss_audio.h"
static int audio_read_header(AVFormatContext *s1)
{
OSSAudioData *s = s1->priv_data;
AVStream *st;
int ret;
st = avformat_new_stream(s1, NULL);
if (!st) {
return AVERROR(ENOMEM);
}
ret = ff_oss_audio_open(s1, 0, s1->filename);
if (ret < 0) {
return AVERROR(EIO);
}
/* take real parameters */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = s->codec_id;
st->codec->sample_rate = s->sample_rate;
st->codec->channels = s->channels;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
OSSAudioData *s = s1->priv_data;
int ret, bdelay;
int64_t cur_time;
struct audio_buf_info abufi;
if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
return ret;
ret = read(s->fd, pkt->data, pkt->size);
if (ret <= 0){
av_free_packet(pkt);
pkt->size = 0;
if (ret<0) return AVERROR(errno);
else return AVERROR_EOF;
}
pkt->size = ret;
/* compute pts of the start of the packet */
cur_time = av_gettime();
bdelay = ret;
if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
bdelay += abufi.bytes;
}
/* subtract time represented by the number of bytes in the audio fifo */
cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
/* convert to wanted units */
pkt->pts = cur_time;
if (s->flip_left && s->channels == 2) {
int i;
short *p = (short *) pkt->data;
for (i = 0; i < ret; i += 4) {
*p = ~*p;
p += 2;
}
}
return 0;
}
static int audio_read_close(AVFormatContext *s1)
{
OSSAudioData *s = s1->priv_data;
ff_oss_audio_close(s);
return 0;
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(OSSAudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass oss_demuxer_class = {
.class_name = "OSS demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
};
AVInputFormat ff_oss_demuxer = {
.name = "oss",
.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
.priv_data_size = sizeof(OSSAudioData),
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = audio_read_close,
.flags = AVFMT_NOFILE,
.priv_class = &oss_demuxer_class,
};

118
libavdevice/oss_audio_enc.c
Unescape View File

@ -0,0 +1,118 @@
/*
* Linux audio grab interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* 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
*/
#include "config.h"
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
#if HAVE_UNISTD_H
#include <unistd.h>
#endif
#include <fcntl.h>
#include <sys/ioctl.h>
#include "libavutil/internal.h"
#include "libavcodec/avcodec.h"
#include "avdevice.h"
#include "libavformat/internal.h"
#include "oss_audio.h"
static int audio_write_header(AVFormatContext *s1)
{
OSSAudioData *s = s1->priv_data;
AVStream *st;
int ret;
st = s1->streams[0];
s->sample_rate = st->codec->sample_rate;
s->channels = st->codec->channels;
ret = ff_oss_audio_open(s1, 1, s1->filename);
if (ret < 0) {
return AVERROR(EIO);
} else {
return 0;
}
}
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
OSSAudioData *s = s1->priv_data;
int len, ret;
int size= pkt->size;
uint8_t *buf= pkt->data;
while (size > 0) {
len = FFMIN(OSS_AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
memcpy(s->buffer + s->buffer_ptr, buf, len);
s->buffer_ptr += len;
if (s->buffer_ptr >= OSS_AUDIO_BLOCK_SIZE) {
for(;;) {
ret = write(s->fd, s->buffer, OSS_AUDIO_BLOCK_SIZE);
if (ret > 0)
break;
if (ret < 0 && (errno != EAGAIN && errno != EINTR))
return AVERROR(EIO);
}
s->buffer_ptr = 0;
}
buf += len;
size -= len;
}
return 0;
}
static int audio_write_trailer(AVFormatContext *s1)
{
OSSAudioData *s = s1->priv_data;
ff_oss_audio_close(s);
return 0;
}
static const AVClass oss_muxer_class = {
.class_name = "OSS muxer",
.item_name = av_default_item_name,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
};
AVOutputFormat ff_oss_muxer = {
.name = "oss",
.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
.priv_data_size = sizeof(OSSAudioData),
/* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
other formats */
.audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
.video_codec = AV_CODEC_ID_NONE,
.write_header = audio_write_header,
.write_packet = audio_write_packet,
.write_trailer = audio_write_trailer,
.flags = AVFMT_NOFILE,
.priv_class = &oss_muxer_class,
};
Write Preview
Loading…
Cancel
Save