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avformat/audiointerleave: only keep the retime functionality of the audio interleaver

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And rename it to retimeinterleave, use the pcm_rechunk bitstream filter for
rechunking.

By seperating the two functions we hopefully get cleaner code.

Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint 2020-03-25 23:49:17 +01:00
parent 2035620b7c
commit c5324d92c5
7 changed files with 88 additions and 189 deletions
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2
configure vendored
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@ -2722,6 +2722,7 @@ fraps_decoder_select="bswapdsp huffman"
g2m_decoder_deps="zlib"
g2m_decoder_select="blockdsp idctdsp jpegtables"
g729_decoder_select="audiodsp"
gxf_encoder_select="pcm_rechunk_bsf"
h261_decoder_select="mpegvideo"
h261_encoder_select="mpegvideoenc"
h263_decoder_select="h263_parser h263dsp mpegvideo qpeldsp"
@ -2794,6 +2795,7 @@ mv30_decoder_select="aandcttables blockdsp"
mvha_decoder_deps="zlib"
mvha_decoder_select="llviddsp"
mwsc_decoder_deps="zlib"
mxf_encoder_select="pcm_rechunk_bsf"
mxpeg_decoder_select="mjpeg_decoder"
nellymoser_decoder_select="mdct sinewin"
nellymoser_encoder_select="audio_frame_queue mdct sinewin"

4
libavformat/Makefile
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@ -205,7 +205,7 @@ OBJS-$(CONFIG_GIF_DEMUXER) += gifdec.o
OBJS-$(CONFIG_GSM_DEMUXER) += gsmdec.o
OBJS-$(CONFIG_GSM_MUXER) += rawenc.o
OBJS-$(CONFIG_GXF_DEMUXER) += gxf.o
OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o audiointerleave.o
OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o retimeinterleave.o
OBJS-$(CONFIG_G722_DEMUXER) += g722.o rawdec.o
OBJS-$(CONFIG_G722_MUXER) += rawenc.o
OBJS-$(CONFIG_G723_1_DEMUXER) += g723_1.o
@ -347,7 +347,7 @@ OBJS-$(CONFIG_MUSX_DEMUXER) += musx.o
OBJS-$(CONFIG_MV_DEMUXER) += mvdec.o
OBJS-$(CONFIG_MVI_DEMUXER) += mvi.o
OBJS-$(CONFIG_MXF_DEMUXER) += mxfdec.o mxf.o
OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o audiointerleave.o avc.o
OBJS-$(CONFIG_MXF_MUXER) += mxfenc.o mxf.o retimeinterleave.o avc.o
OBJS-$(CONFIG_MXG_DEMUXER) += mxg.o
OBJS-$(CONFIG_NC_DEMUXER) += ncdec.o
OBJS-$(CONFIG_NISTSPHERE_DEMUXER) += nistspheredec.o pcm.o

148
libavformat/audiointerleave.c
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@ -1,148 +0,0 @@
/*
* 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 "libavutil/mathematics.h"
#include "avformat.h"
#include "audiointerleave.h"
#include "internal.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 (aic && st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
av_fifo_freep(&aic->fifo);
}
}
int ff_audio_interleave_init(AVFormatContext *s,
const int samples_per_frame,
AVRational time_base)
{
int i;
if (!time_base.num) {
av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
return AVERROR(EINVAL);
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
int max_samples = samples_per_frame ? samples_per_frame :
av_rescale_rnd(st->codecpar->sample_rate, time_base.num, time_base.den, AV_ROUND_UP);
aic->sample_size = (st->codecpar->channels *
av_get_bits_per_sample(st->codecpar->codec_id)) / 8;
if (!aic->sample_size) {
av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
return AVERROR(EINVAL);
}
aic->samples_per_frame = samples_per_frame;
aic->time_base = time_base;
if (!(aic->fifo = av_fifo_alloc_array(100, max_samples)))
return AVERROR(ENOMEM);
aic->fifo_size = 100 * max_samples;
}
}
return 0;
}
static int 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 ret;
int nb_samples = aic->samples_per_frame ? aic->samples_per_frame :
(av_rescale_q(aic->n + 1, av_make_q(st->codecpar->sample_rate, 1), av_inv_q(aic->time_base)) - aic->nb_samples);
int frame_size = nb_samples * aic->sample_size;
int size = FFMIN(av_fifo_size(aic->fifo), frame_size);
if (!size || (!flush && size == av_fifo_size(aic->fifo)))
return 0;
ret = av_new_packet(pkt, frame_size);
if (ret < 0)
return ret;
av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
if (size < pkt->size)
memset(pkt->data + size, 0, pkt->size - size);
pkt->dts = pkt->pts = aic->dts;
pkt->duration = av_rescale_q(nb_samples, st->time_base, aic->time_base);
pkt->stream_index = stream_index;
aic->dts += pkt->duration;
aic->nb_samples += nb_samples;
aic->n++;
return pkt->size;
}
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *))
{
int i, ret;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
AudioInterleaveContext *aic = st->priv_data;
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
if (new_size > aic->fifo_size) {
if (av_fifo_realloc2(aic->fifo, new_size) < 0)
return AVERROR(ENOMEM);
aic->fifo_size = new_size;
}
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;
if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
return ret;
}
pkt = NULL;
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
AVPacket new_pkt;
while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
return ret;
}
if (ret < 0)
return ret;
}
}
return get_packet(s, out, NULL, flush);
}

18
libavformat/gxfenc.c
View File

@ -27,8 +27,9 @@
#include "avformat.h"
#include "internal.h"
#include "gxf.h"
#include "audiointerleave.h"
#include "retimeinterleave.h"
#define GXF_SAMPLES_PER_FRAME 32768
#define GXF_AUDIO_PACKET_SIZE 65536
#define GXF_TIMECODE(c, d, h, m, s, f) \
@ -44,7 +45,7 @@ typedef struct GXFTimecode{
} GXFTimecode;
typedef struct GXFStreamContext {
AudioInterleaveContext aic;
RetimeInterleaveContext aic;
uint32_t track_type;
uint32_t sample_size;
uint32_t sample_rate;
@ -663,8 +664,6 @@ static int gxf_write_umf_packet(AVFormatContext *s)
return updatePacketSize(pb, pos);
}
static const int GXF_samples_per_frame = 32768;
static void gxf_init_timecode_track(GXFStreamContext *sc, GXFStreamContext *vsc)
{
if (!vsc)
@ -736,6 +735,9 @@ static int gxf_write_header(AVFormatContext *s)
av_log(s, AV_LOG_ERROR, "only mono tracks are allowed\n");
return -1;
}
ret = ff_stream_add_bitstream_filter(st, "pcm_rechunk", "n="AV_STRINGIFY(GXF_SAMPLES_PER_FRAME));
if (ret < 0)
return ret;
sc->track_type = 2;
sc->sample_rate = st->codecpar->sample_rate;
avpriv_set_pts_info(st, 64, 1, sc->sample_rate);
@ -813,14 +815,12 @@ static int gxf_write_header(AVFormatContext *s)
return -1;
}
}
ff_retime_interleave_init(&sc->aic, st->time_base);
/* FIXME first 10 audio tracks are 0 to 9 next 22 are A to V */
sc->media_info = media_info<<8 | ('0'+tracks[media_info]++);
sc->order = s->nb_streams - st->index;
}
if (ff_audio_interleave_init(s, GXF_samples_per_frame, (AVRational){ 1, 48000 }) < 0)
return -1;
if (tcr && vsc)
gxf_init_timecode(s, &gxf->tc, tcr->value, vsc->fields);
@ -877,8 +877,6 @@ static void gxf_deinit(AVFormatContext *s)
{
GXFContext *gxf = s->priv_data;
ff_audio_interleave_close(s);
av_freep(&gxf->flt_entries);
av_freep(&gxf->map_offsets);
}
@ -1016,7 +1014,7 @@ static int gxf_interleave_packet(AVFormatContext *s, AVPacket *out, AVPacket *pk
{
if (pkt && s->streams[pkt->stream_index]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
pkt->duration = 2; // enforce 2 fields
return ff_audio_rechunk_interleave(s, out, pkt, flush,
return ff_retime_interleave(s, out, pkt, flush,
ff_interleave_packet_per_dts, gxf_compare_field_nb);
}

17
libavformat/mxfenc.c
View File

@ -52,7 +52,7 @@
#include "libavcodec/h264_ps.h"
#include "libavcodec/golomb.h"
#include "libavcodec/internal.h"
#include "audiointerleave.h"
#include "retimeinterleave.h"
#include "avformat.h"
#include "avio_internal.h"
#include "internal.h"
@ -79,7 +79,7 @@ typedef struct MXFIndexEntry {
} MXFIndexEntry;
typedef struct MXFStreamContext {
AudioInterleaveContext aic;
RetimeInterleaveContext aic;
UID track_essence_element_key;
int index; ///< index in mxf_essence_container_uls table
const UID *codec_ul;
@ -2538,6 +2538,7 @@ static int mxf_write_header(AVFormatContext *s)
if (mxf->signal_standard >= 0)
sc->signal_standard = mxf->signal_standard;
} else if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
char bsf_arg[32];
if (st->codecpar->sample_rate != 48000) {
av_log(s, AV_LOG_ERROR, "only 48khz is implemented\n");
return -1;
@ -2580,6 +2581,10 @@ static int mxf_write_header(AVFormatContext *s)
av_rescale_rnd(st->codecpar->sample_rate, mxf->time_base.num, mxf->time_base.den, AV_ROUND_UP) *
av_get_bits_per_sample(st->codecpar->codec_id) / 8;
}
snprintf(bsf_arg, sizeof(bsf_arg), "r=%d/%d", mxf->tc.rate.num, mxf->tc.rate.den);
ret = ff_stream_add_bitstream_filter(st, "pcm_rechunk", bsf_arg);
if (ret < 0)
return ret;
} else if (st->codecpar->codec_type == AVMEDIA_TYPE_DATA) {
AVDictionaryEntry *e = av_dict_get(st->metadata, "data_type", NULL, 0);
if (e && !strcmp(e->value, "vbi_vanc_smpte_436M")) {
@ -2593,6 +2598,7 @@ static int mxf_write_header(AVFormatContext *s)
return -1;
}
}
ff_retime_interleave_init(&sc->aic, av_inv_q(mxf->tc.rate));
if (sc->index == -1) {
sc->index = mxf_get_essence_container_ul_index(st->codecpar->codec_id);
@ -2646,9 +2652,6 @@ static int mxf_write_header(AVFormatContext *s)
return AVERROR(ENOMEM);
mxf->timecode_track->index = -1;
if (ff_audio_interleave_init(s, 0, av_inv_q(mxf->tc.rate)) < 0)
return -1;
return 0;
}
@ -3010,8 +3013,6 @@ static void mxf_deinit(AVFormatContext *s)
{
MXFContext *mxf = s->priv_data;
ff_audio_interleave_close(s);
av_freep(&mxf->index_entries);
av_freep(&mxf->body_partition_offset);
if (mxf->timecode_track) {
@ -3086,7 +3087,7 @@ static int mxf_compare_timestamps(AVFormatContext *s, const AVPacket *next,
static int mxf_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush)
{
return ff_audio_rechunk_interleave(s, out, pkt, flush,
return ff_retime_interleave(s, out, pkt, flush,
mxf_interleave_get_packet, mxf_compare_timestamps);
}

51
libavformat/retimeinterleave.c Normal file
View File

@ -0,0 +1,51 @@
/*
* Retime 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/mathematics.h"
#include "avformat.h"
#include "retimeinterleave.h"
#include "internal.h"
void ff_retime_interleave_init(RetimeInterleaveContext *aic, AVRational time_base)
{
aic->time_base = time_base;
}
int ff_retime_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *))
{
int ret;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
RetimeInterleaveContext *aic = st->priv_data;
pkt->duration = av_rescale_q(pkt->duration, st->time_base, aic->time_base);
// rewrite pts and dts to be decoded time line position
pkt->pts = pkt->dts = aic->dts;
aic->dts += pkt->duration;
if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
return ret;
}
return get_packet(s, out, NULL, flush);
}

33
libavformat/audiointerleave.h → libavformat/retimeinterleave.h
View File

@ -20,36 +20,31 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFORMAT_AUDIOINTERLEAVE_H
#define AVFORMAT_AUDIOINTERLEAVE_H
#ifndef AVFORMAT_RETIMEINTERLEAVE_H
#define AVFORMAT_RETIMEINTERLEAVE_H
#include "libavutil/fifo.h"
#include "avformat.h"
typedef struct AudioInterleaveContext {
AVFifoBuffer *fifo;
unsigned fifo_size; ///< size of currently allocated FIFO
int64_t n; ///< number of generated packets
int64_t nb_samples; ///< number of generated samples
typedef struct RetimeInterleaveContext {
uint64_t dts; ///< current dts
int sample_size; ///< size of one sample all channels included
int samples_per_frame; ///< samples per frame if fixed, 0 otherwise
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);
AVRational time_base; ///< time base of output packets
} RetimeInterleaveContext;
/**
* Rechunk audio PCM packets per AudioInterleaveContext->samples_per_frame
* and interleave them correctly.
* The first element of AVStream->priv_data must be AudioInterleaveContext
* Init the retime interleave context
*/
void ff_retime_interleave_init(RetimeInterleaveContext *aic, AVRational time_base);
/**
* Retime packets per RetimeInterleaveContext->time_base and interleave them
* correctly.
* The first element of AVStream->priv_data must be RetimeInterleaveContext
* when using this function.
*
* @param get_packet function will output a packet when streams are correctly interleaved.
* @param compare_ts function will compare AVPackets and decide interleaving order.
*/
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
int ff_retime_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
int (*compare_ts)(AVFormatContext *, const AVPacket *, const AVPacket *));