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oss_audio: Split muxer and demuxer 

Signed-off-by: Diego Biurrun <diego@biurrun.de>
tags/n2.4
Nidhi Makhijani Diego Biurrun 11 years ago
parent
e0a2e60c0a
commit
d6e1d37100
5 changed files with 312 additions and 202 deletions
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  1. +2
    -2
      libavdevice/Makefile
  2. +11
    -200
      libavdevice/oss_audio.c
  3. +45
    -0
      libavdevice/oss_audio.h
  4. +146
    -0
      libavdevice/oss_audio_dec.c
  5. +108
    -0
      libavdevice/oss_audio_enc.c

+ 2
- 2
libavdevice/Makefile View File

@@ -15,8 +15,8 @@ OBJS-$(CONFIG_BKTR_INDEV) += bktr.o
OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o
OBJS-$(CONFIG_FBDEV_INDEV) += fbdev.o
OBJS-$(CONFIG_JACK_INDEV) += jack_audio.o timefilter.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-$(CONFIG_PULSE_INDEV) += pulse.o
OBJS-$(CONFIG_SNDIO_INDEV) += sndio_common.o sndio_dec.o
OBJS-$(CONFIG_SNDIO_OUTDEV) += sndio_common.o sndio_enc.o


+ 11
- 200
libavdevice/oss_audio.c View File

@@ -20,45 +20,31 @@
*/

#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

#include <unistd.h>
#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 "libavformat/avformat.h"
#include "libavformat/internal.h"

#define AUDIO_BLOCK_SIZE 4096
#include "libavformat/avformat.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;
#include "oss_audio.h"

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");
@@ -80,7 +66,7 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
if (!is_output)
fcntl(audio_fd, F_SETFL, O_NONBLOCK);

s->frame_size = AUDIO_BLOCK_SIZE;
s->frame_size = OSS_AUDIO_BLOCK_SIZE;

/* select format : favour native format */
err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
@@ -143,183 +129,8 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi
return AVERROR(EIO);
}

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,
};

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
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,
};
#endif

+ 45
- 0
libavdevice/oss_audio.h View File

@@ -0,0 +1,45 @@
/*
* 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
*/

#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 */

+ 146
- 0
libavdevice/oss_audio_dec.c View File

@@ -0,0 +1,146 @@
/*
* Linux audio play interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* 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
*/

#include "config.h"

#include <stdint.h>

#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif

#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>

#include "libavutil/internal.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"

#include "libavcodec/avcodec.h"

#include "libavformat/avformat.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,
};

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,
};

+ 108
- 0
libavdevice/oss_audio_enc.c View File

@@ -0,0 +1,108 @@
/*
* Linux audio grab interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* 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
*/

#include "config.h"

#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif

#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>

#include "libavutil/internal.h"

#include "libavcodec/avcodec.h"

#include "libavformat/avformat.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;
}

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,
};

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