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extract audio interleaving code from mxf muxer, will be used by gxf and dv

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Originally committed as revision 17038 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2009-02-08 04:31:44 +00:00
parent baf2ffd329
commit f1544e79f2
4 changed files with 178 additions and 105 deletions
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2
libavformat/Makefile
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@ -116,7 +116,7 @@ OBJS-$(CONFIG_MSNWC_TCP_DEMUXER) += msnwc_tcp.o
OBJS-$(CONFIG_MTV_DEMUXER) += mtv.o OBJS-$(CONFIG_MTV_DEMUXER) += mtv.o
OBJS-$(CONFIG_MVI_DEMUXER) += mvi.o OBJS-$(CONFIG_MVI_DEMUXER) += mvi.o
OBJS-$(CONFIG_MXF_DEMUXER) += mxfdec.o mxf.o OBJS-$(CONFIG_MXF_DEMUXER) += mxfdec.o mxf.o
OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o audiointerleave.o
OBJS-$(CONFIG_NSV_DEMUXER) += nsvdec.o OBJS-$(CONFIG_NSV_DEMUXER) += nsvdec.o
OBJS-$(CONFIG_NULL_MUXER) += raw.o OBJS-$(CONFIG_NULL_MUXER) += raw.o
OBJS-$(CONFIG_NUT_DEMUXER) += nutdec.o nut.o riff.o OBJS-$(CONFIG_NUT_DEMUXER) += nutdec.o nut.o riff.o

125
libavformat/audiointerleave.c Normal file
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@ -0,0 +1,125 @@
/*
* Audio Interleaving functions
*
* Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
*
* 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 "libavutil/fifo.h"
#include "avformat.h"
#include "audiointerleave.h"
void ff_audio_interleave_close(AVFormatContext *s)
{
int i;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO)
av_fifo_free(&aic->fifo);
}
}
int ff_audio_interleave_init(AVFormatContext *s,
const int *samples_per_frame,
AVRational time_base)
{
int i;
if (!samples_per_frame)
return -1;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
aic->sample_size = (st->codec->channels *
av_get_bits_per_sample(st->codec->codec_id)) / 8;
if (!aic->sample_size) {
av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
return -1;
}
aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame;
aic->time_base = time_base;
av_fifo_init(&aic->fifo, 100 * *aic->samples);
}
}
return 0;
}
int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
int stream_index, int flush)
{
AVStream *st = s->streams[stream_index];
AudioInterleaveContext *aic = st->priv_data;
int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
return 0;
av_new_packet(pkt, size);
av_fifo_read(&aic->fifo, pkt->data, size);
pkt->dts = pkt->pts = aic->dts;
pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
pkt->stream_index = stream_index;
aic->dts += pkt->duration;
aic->samples++;
if (!*aic->samples)
aic->samples = aic->samples_per_frame;
return size;
}
int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
{
int i;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
} else {
// rewrite pts and dts to be decoded time line position
pkt->dts = aic->dts;
aic->dts += pkt->duration;
ff_interleave_add_packet(s, pkt, compare_ts);
}
pkt = NULL;
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
AVPacket new_pkt;
while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
ff_interleave_add_packet(s, &new_pkt, compare_ts);
}
}
return get_packet(s, out, pkt, flush);
}

49
libavformat/audiointerleave.h Normal file
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@ -0,0 +1,49 @@
/*
* Audio Interleaving prototypes and declarations
*
* Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
*
* 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 AVFORMAT_AUDIOINTERLEAVE_H
#define AVFORMAT_AUDIOINTERLEAVE_H
#include "libavutil/fifo.h"
#include "avformat.h"
typedef struct {
AVFifoBuffer fifo;
unsigned fifo_size; ///< current fifo size allocated
uint64_t dts; ///< current dts
int sample_size; ///< size of one sample all channels included
const int *samples_per_frame; ///< must be 0 terminated
const int *samples; ///< current samples per frame, pointer to samples_per_frame
AVRational time_base; ///< time base of output audio packets
} AudioInterleaveContext;
int ff_audio_interleave_init(AVFormatContext *s, const int *samples_per_frame, AVRational time_base);
void ff_audio_interleave_close(AVFormatContext *s);
int ff_interleave_compare_dts(AVFormatContext *s, AVPacket *next, AVPacket *pkt);
int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
int stream_index, int flush);
int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *));
#endif // AVFORMAT_AUDIOINTERLEAVE_H

107
libavformat/mxfenc.c
View File

@ -36,6 +36,7 @@
#include <time.h> #include <time.h>
#include "libavutil/fifo.h" #include "libavutil/fifo.h"
#include "audiointerleave.h"
#include "mxf.h" #include "mxf.h"
static const int NTSC_samples_per_frame[] = { 1602, 1601, 1602, 1601, 1602, 0 }; static const int NTSC_samples_per_frame[] = { 1602, 1601, 1602, 1601, 1602, 0 };
@ -44,16 +45,6 @@ static const int PAL_samples_per_frame[] = { 1920, 0 };
#define MXF_INDEX_CLUSTER_SIZE 4096 #define MXF_INDEX_CLUSTER_SIZE 4096
#define KAG_SIZE 512 #define KAG_SIZE 512
typedef struct {
AVFifoBuffer fifo;
unsigned fifo_size; ///< current fifo size allocated
uint64_t dts; ///< current dts
int sample_size; ///< size of one sample all channels included
const int *samples_per_frame; ///< must be 0 terminated
const int *samples; ///< current samples per frame, pointer to samples_per_frame
AVRational time_base; ///< time base of output audio packets
} AudioInterleaveContext;
typedef struct { typedef struct {
int local_tag; int local_tag;
UID uid; UID uid;
@ -1110,49 +1101,6 @@ static int mxf_parse_mpeg2_frame(AVFormatContext *s, AVStream *st, AVPacket *pkt
return !!sc->codec_ul; return !!sc->codec_ul;
} }
static int ff_audio_interleave_init(AVFormatContext *s,
const int *samples_per_frame,
AVRational time_base)
{
int i;
if (!samples_per_frame)
return -1;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
aic->sample_size = (st->codec->channels *
av_get_bits_per_sample(st->codec->codec_id)) / 8;
if (!aic->sample_size) {
av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
return -1;
}
aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame;
aic->time_base = time_base;
av_fifo_init(&aic->fifo, 100 * *aic->samples);
}
}
return 0;
}
static void ff_audio_interleave_close(AVFormatContext *s)
{
int i;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO)
av_fifo_free(&aic->fifo);
}
}
static uint64_t mxf_parse_timestamp(time_t timestamp) static uint64_t mxf_parse_timestamp(time_t timestamp)
{ {
struct tm *time = localtime(&timestamp); struct tm *time = localtime(&timestamp);
@ -1428,31 +1376,6 @@ static int mxf_write_footer(AVFormatContext *s)
return 0; return 0;
} }
static int mxf_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
int stream_index, int flush)
{
AVStream *st = s->streams[stream_index];
AudioInterleaveContext *aic = st->priv_data;
int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
return 0;
av_new_packet(pkt, size);
av_fifo_read(&aic->fifo, pkt->data, size);
pkt->dts = pkt->pts = aic->dts;
pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
pkt->stream_index = stream_index;
aic->dts += pkt->duration;
aic->samples++;
if (!*aic->samples)
aic->samples = aic->samples_per_frame;
return size;
}
static int mxf_interleave_get_packet(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush) static int mxf_interleave_get_packet(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{ {
AVPacketList *pktl; AVPacketList *pktl;
@ -1517,32 +1440,8 @@ static int mxf_compare_timestamps(AVFormatContext *s, AVPacket *next, AVPacket *
static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush) static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{ {
int i; return ff_audio_interleave(s, out, pkt, flush,
mxf_interleave_get_packet, mxf_compare_timestamps);
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
} else {
// rewrite pts and dts to be decoded time line position
pkt->pts = pkt->dts = aic->dts;
aic->dts += pkt->duration;
ff_interleave_add_packet(s, pkt, mxf_compare_timestamps);
}
pkt = NULL;
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
AVPacket new_pkt;
while (mxf_interleave_new_audio_packet(s, &new_pkt, i, flush))
ff_interleave_add_packet(s, &new_pkt, mxf_compare_timestamps);
}
}
return mxf_interleave_get_packet(s, out, pkt, flush);
} }
AVOutputFormat mxf_muxer = { AVOutputFormat mxf_muxer = {