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oss: Coalesce source files after outdev removal

tags/n4.0
Diego Biurrun 8 years ago
parent
8d3db95f20
commit
6ce13070bd
4 changed files with 117 additions and 198 deletions
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  1. +1
    -1
      libavdevice/Makefile
  2. +116
    -6
      libavdevice/oss.c
  3. +0
    -45
      libavdevice/oss.h
  4. +0
    -146
      libavdevice/oss_dec.c

+ 1
- 1
libavdevice/Makefile View File

@@ -16,7 +16,7 @@ OBJS-$(CONFIG_BKTR_INDEV) += bktr.o
OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o
OBJS-$(CONFIG_FBDEV_INDEV) += fbdev.o
OBJS-$(CONFIG_JACK_INDEV) += jack.o timefilter.o
OBJS-$(CONFIG_OSS_INDEV) += oss_dec.o oss.o
OBJS-$(CONFIG_OSS_INDEV) += oss.o
OBJS-$(CONFIG_PULSE_INDEV) += pulse.o
OBJS-$(CONFIG_SNDIO_INDEV) += sndio.o
OBJS-$(CONFIG_V4L2_INDEV) += v4l2.o


+ 116
- 6
libavdevice/oss.c View File

@@ -28,15 +28,30 @@
#include <sys/soundcard.h>

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

#include "libavcodec/avcodec.h"

#include "libavformat/avformat.h"

#include "oss.h"

int ff_oss_audio_open(AVFormatContext *s1, int is_output,
const char *audio_device)
#include "libavformat/internal.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;

static int oss_audio_open(AVFormatContext *s1, int is_output,
const char *audio_device)
{
OSSAudioData *s = s1->priv_data;
int audio_fd;
@@ -126,8 +141,103 @@ int ff_oss_audio_open(AVFormatContext *s1, int is_output,
#undef CHECK_IOCTL_ERROR
}

int ff_oss_audio_close(OSSAudioData *s)
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 = oss_audio_open(s1, 0, s1->filename);
if (ret < 0) {
return AVERROR(EIO);
}

/* take real parameters */
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = s->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;
}

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_packet_unref(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;

close(s->fd);
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,
};

+ 0
- 45
libavdevice/oss.h View File

@@ -1,45 +0,0 @@
/*
* 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_H
#define AVDEVICE_OSS_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_H */

+ 0
- 146
libavdevice/oss_dec.c View File

@@ -1,146 +0,0 @@
/*
* 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.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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = s->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;
}

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_packet_unref(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,
};

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