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FFmpeg/libavformat/mux.c
Luca Barbato d9ae1031f5 lavf: improve handling of sparse streams when muxing
Currently ff_interleave_packet_per_dts() waits until it gets a frame for
each stream before outputting packets in interleaved order.

Sparse streams (i.e. streams with much fewer packets than the other
streams, like subtitles or audio with DTX) tend to add up latency and in
specific cases end up allocating a large amount of memory.

Emit the top packet from the packet_buffer if it has a time delta
larger than a specified threshold.

Original report of the issue and initial proposed solution by
mus.svz@gmail.com.

Bug-id: 31
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2014-02-04 22:01:42 +01:00

717 lines
22 KiB
C

/*
* muxing functions for use within Libav
* Copyright (c) 2000, 2001, 2002 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 "avformat.h"
#include "avio_internal.h"
#include "internal.h"
#include "libavcodec/internal.h"
#include "libavcodec/bytestream.h"
#include "libavutil/opt.h"
#include "libavutil/dict.h"
#include "libavutil/pixdesc.h"
#include "metadata.h"
#include "id3v2.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/internal.h"
#include "libavutil/mathematics.h"
#include "libavutil/parseutils.h"
#include "libavutil/time.h"
#include "riff.h"
#include "audiointerleave.h"
#include "url.h"
#include <stdarg.h>
#if CONFIG_NETWORK
#include "network.h"
#endif
#undef NDEBUG
#include <assert.h>
/**
* @file
* muxing functions for use within Libav
*/
/* fraction handling */
/**
* f = val + (num / den) + 0.5.
*
* 'num' is normalized so that it is such as 0 <= num < den.
*
* @param f fractional number
* @param val integer value
* @param num must be >= 0
* @param den must be >= 1
*/
static void frac_init(AVFrac *f, int64_t val, int64_t num, int64_t den)
{
num += (den >> 1);
if (num >= den) {
val += num / den;
num = num % den;
}
f->val = val;
f->num = num;
f->den = den;
}
/**
* Fractional addition to f: f = f + (incr / f->den).
*
* @param f fractional number
* @param incr increment, can be positive or negative
*/
static void frac_add(AVFrac *f, int64_t incr)
{
int64_t num, den;
num = f->num + incr;
den = f->den;
if (num < 0) {
f->val += num / den;
num = num % den;
if (num < 0) {
num += den;
f->val--;
}
} else if (num >= den) {
f->val += num / den;
num = num % den;
}
f->num = num;
}
static int validate_codec_tag(AVFormatContext *s, AVStream *st)
{
const AVCodecTag *avctag;
int n;
enum AVCodecID id = AV_CODEC_ID_NONE;
unsigned int tag = 0;
/**
* Check that tag + id is in the table
* If neither is in the table -> OK
* If tag is in the table with another id -> FAIL
* If id is in the table with another tag -> FAIL unless strict < normal
*/
for (n = 0; s->oformat->codec_tag[n]; n++) {
avctag = s->oformat->codec_tag[n];
while (avctag->id != AV_CODEC_ID_NONE) {
if (avpriv_toupper4(avctag->tag) == avpriv_toupper4(st->codec->codec_tag)) {
id = avctag->id;
if (id == st->codec->codec_id)
return 1;
}
if (avctag->id == st->codec->codec_id)
tag = avctag->tag;
avctag++;
}
}
if (id != AV_CODEC_ID_NONE)
return 0;
if (tag && (st->codec->strict_std_compliance >= FF_COMPLIANCE_NORMAL))
return 0;
return 1;
}
static int init_muxer(AVFormatContext *s, AVDictionary **options)
{
int ret = 0, i;
AVStream *st;
AVDictionary *tmp = NULL;
AVCodecContext *codec = NULL;
AVOutputFormat *of = s->oformat;
if (options)
av_dict_copy(&tmp, *options, 0);
if ((ret = av_opt_set_dict(s, &tmp)) < 0)
goto fail;
// some sanity checks
if (s->nb_streams == 0 && !(of->flags & AVFMT_NOSTREAMS)) {
av_log(s, AV_LOG_ERROR, "no streams\n");
ret = AVERROR(EINVAL);
goto fail;
}
for (i = 0; i < s->nb_streams; i++) {
st = s->streams[i];
codec = st->codec;
switch (codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
if (codec->sample_rate <= 0) {
av_log(s, AV_LOG_ERROR, "sample rate not set\n");
ret = AVERROR(EINVAL);
goto fail;
}
if (!codec->block_align)
codec->block_align = codec->channels *
av_get_bits_per_sample(codec->codec_id) >> 3;
break;
case AVMEDIA_TYPE_VIDEO:
if (codec->time_base.num <= 0 ||
codec->time_base.den <= 0) { //FIXME audio too?
av_log(s, AV_LOG_ERROR, "time base not set\n");
ret = AVERROR(EINVAL);
goto fail;
}
if ((codec->width <= 0 || codec->height <= 0) &&
!(of->flags & AVFMT_NODIMENSIONS)) {
av_log(s, AV_LOG_ERROR, "dimensions not set\n");
ret = AVERROR(EINVAL);
goto fail;
}
if (av_cmp_q(st->sample_aspect_ratio,
codec->sample_aspect_ratio)) {
if (st->sample_aspect_ratio.num != 0 &&
st->sample_aspect_ratio.den != 0 &&
codec->sample_aspect_ratio.den != 0 &&
codec->sample_aspect_ratio.den != 0) {
av_log(s, AV_LOG_ERROR, "Aspect ratio mismatch between muxer "
"(%d/%d) and encoder layer (%d/%d)\n",
st->sample_aspect_ratio.num, st->sample_aspect_ratio.den,
codec->sample_aspect_ratio.num,
codec->sample_aspect_ratio.den);
ret = AVERROR(EINVAL);
goto fail;
}
}
break;
}
if (of->codec_tag) {
if (codec->codec_tag &&
codec->codec_id == AV_CODEC_ID_RAWVIDEO &&
!av_codec_get_tag(of->codec_tag, codec->codec_id) &&
!validate_codec_tag(s, st)) {
// the current rawvideo encoding system ends up setting
// the wrong codec_tag for avi, we override it here
codec->codec_tag = 0;
}
if (codec->codec_tag) {
if (!validate_codec_tag(s, st)) {
char tagbuf[32];
av_get_codec_tag_string(tagbuf, sizeof(tagbuf), codec->codec_tag);
av_log(s, AV_LOG_ERROR,
"Tag %s/0x%08x incompatible with output codec id '%d'\n",
tagbuf, codec->codec_tag, codec->codec_id);
ret = AVERROR_INVALIDDATA;
goto fail;
}
} else
codec->codec_tag = av_codec_get_tag(of->codec_tag, codec->codec_id);
}
if (of->flags & AVFMT_GLOBALHEADER &&
!(codec->flags & CODEC_FLAG_GLOBAL_HEADER))
av_log(s, AV_LOG_WARNING,
"Codec for stream %d does not use global headers "
"but container format requires global headers\n", i);
if (codec->codec_type != AVMEDIA_TYPE_ATTACHMENT)
s->internal->nb_interleaved_streams++;
}
if (!s->priv_data && of->priv_data_size > 0) {
s->priv_data = av_mallocz(of->priv_data_size);
if (!s->priv_data) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (of->priv_class) {
*(const AVClass **)s->priv_data = of->priv_class;
av_opt_set_defaults(s->priv_data);
if ((ret = av_opt_set_dict(s->priv_data, &tmp)) < 0)
goto fail;
}
}
/* set muxer identification string */
if (s->nb_streams && !(s->streams[0]->codec->flags & CODEC_FLAG_BITEXACT)) {
av_dict_set(&s->metadata, "encoder", LIBAVFORMAT_IDENT, 0);
}
if (options) {
av_dict_free(options);
*options = tmp;
}
return 0;
fail:
av_dict_free(&tmp);
return ret;
}
static int init_pts(AVFormatContext *s)
{
int i;
AVStream *st;
/* init PTS generation */
for (i = 0; i < s->nb_streams; i++) {
int64_t den = AV_NOPTS_VALUE;
st = s->streams[i];
switch (st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
den = (int64_t)st->time_base.num * st->codec->sample_rate;
break;
case AVMEDIA_TYPE_VIDEO:
den = (int64_t)st->time_base.num * st->codec->time_base.den;
break;
default:
break;
}
if (den != AV_NOPTS_VALUE) {
if (den <= 0)
return AVERROR_INVALIDDATA;
frac_init(&st->pts, 0, 0, den);
}
}
return 0;
}
int avformat_write_header(AVFormatContext *s, AVDictionary **options)
{
int ret = 0;
if (ret = init_muxer(s, options))
return ret;
if (s->oformat->write_header) {
ret = s->oformat->write_header(s);
if (ret < 0)
return ret;
}
if ((ret = init_pts(s)) < 0)
return ret;
return 0;
}
//FIXME merge with compute_pkt_fields
static int compute_pkt_fields2(AVFormatContext *s, AVStream *st, AVPacket *pkt)
{
int delay = FFMAX(st->codec->has_b_frames, !!st->codec->max_b_frames);
int num, den, frame_size, i;
av_dlog(s, "compute_pkt_fields2: pts:%" PRId64 " dts:%" PRId64 " cur_dts:%" PRId64 " b:%d size:%d st:%d\n",
pkt->pts, pkt->dts, st->cur_dts, delay, pkt->size, pkt->stream_index);
/* if(pkt->pts == AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE)
* return AVERROR(EINVAL);*/
/* duration field */
if (pkt->duration == 0) {
ff_compute_frame_duration(&num, &den, st, NULL, pkt);
if (den && num) {
pkt->duration = av_rescale(1, num * (int64_t)st->time_base.den * st->codec->ticks_per_frame, den * (int64_t)st->time_base.num);
}
}
if (pkt->pts == AV_NOPTS_VALUE && pkt->dts != AV_NOPTS_VALUE && delay == 0)
pkt->pts = pkt->dts;
//XXX/FIXME this is a temporary hack until all encoders output pts
if ((pkt->pts == 0 || pkt->pts == AV_NOPTS_VALUE) && pkt->dts == AV_NOPTS_VALUE && !delay) {
pkt->dts =
// pkt->pts= st->cur_dts;
pkt->pts = st->pts.val;
}
//calculate dts from pts
if (pkt->pts != AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE && delay <= MAX_REORDER_DELAY) {
st->pts_buffer[0] = pkt->pts;
for (i = 1; i < delay + 1 && st->pts_buffer[i] == AV_NOPTS_VALUE; i++)
st->pts_buffer[i] = pkt->pts + (i - delay - 1) * pkt->duration;
for (i = 0; i<delay && st->pts_buffer[i] > st->pts_buffer[i + 1]; i++)
FFSWAP(int64_t, st->pts_buffer[i], st->pts_buffer[i + 1]);
pkt->dts = st->pts_buffer[0];
}
if (st->cur_dts && st->cur_dts != AV_NOPTS_VALUE &&
((!(s->oformat->flags & AVFMT_TS_NONSTRICT) &&
st->cur_dts >= pkt->dts) || st->cur_dts > pkt->dts)) {
av_log(s, AV_LOG_ERROR,
"Application provided invalid, non monotonically increasing dts to muxer in stream %d: %" PRId64 " >= %" PRId64 "\n",
st->index, st->cur_dts, pkt->dts);
return AVERROR(EINVAL);
}
if (pkt->dts != AV_NOPTS_VALUE && pkt->pts != AV_NOPTS_VALUE && pkt->pts < pkt->dts) {
av_log(s, AV_LOG_ERROR, "pts < dts in stream %d\n", st->index);
return AVERROR(EINVAL);
}
av_dlog(s, "av_write_frame: pts2:%"PRId64" dts2:%"PRId64"\n",
pkt->pts, pkt->dts);
st->cur_dts = pkt->dts;
st->pts.val = pkt->dts;
/* update pts */
switch (st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
frame_size = ff_get_audio_frame_size(st->codec, pkt->size, 1);
/* HACK/FIXME, we skip the initial 0 size packets as they are most
* likely equal to the encoder delay, but it would be better if we
* had the real timestamps from the encoder */
if (frame_size >= 0 && (pkt->size || st->pts.num != st->pts.den >> 1 || st->pts.val)) {
frac_add(&st->pts, (int64_t)st->time_base.den * frame_size);
}
break;
case AVMEDIA_TYPE_VIDEO:
frac_add(&st->pts, (int64_t)st->time_base.den * st->codec->time_base.num);
break;
default:
break;
}
return 0;
}
/*
* FIXME: this function should NEVER get undefined pts/dts beside when the
* AVFMT_NOTIMESTAMPS is set.
* Those additional safety checks should be dropped once the correct checks
* are set in the callers.
*/
static int write_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret;
if (!(s->oformat->flags & (AVFMT_TS_NEGATIVE | AVFMT_NOTIMESTAMPS))) {
AVRational time_base = s->streams[pkt->stream_index]->time_base;
int64_t offset = 0;
if (!s->offset && pkt->dts != AV_NOPTS_VALUE && pkt->dts < 0) {
s->offset = -pkt->dts;
s->offset_timebase = time_base;
}
if (s->offset)
offset = av_rescale_q(s->offset, s->offset_timebase, time_base);
if (pkt->dts != AV_NOPTS_VALUE)
pkt->dts += offset;
if (pkt->pts != AV_NOPTS_VALUE)
pkt->pts += offset;
}
ret = s->oformat->write_packet(s, pkt);
if (s->pb && ret >= 0 && s->flags & AVFMT_FLAG_FLUSH_PACKETS)
avio_flush(s->pb);
return ret;
}
static int check_packet(AVFormatContext *s, AVPacket *pkt)
{
if (!pkt)
return 0;
if (pkt->stream_index < 0 || pkt->stream_index >= s->nb_streams) {
av_log(s, AV_LOG_ERROR, "Invalid packet stream index: %d\n",
pkt->stream_index);
return AVERROR(EINVAL);
}
if (s->streams[pkt->stream_index]->codec->codec_type == AVMEDIA_TYPE_ATTACHMENT) {
av_log(s, AV_LOG_ERROR, "Received a packet for an attachment stream.\n");
return AVERROR(EINVAL);
}
return 0;
}
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
{
int ret;
ret = check_packet(s, pkt);
if (ret < 0)
return ret;
if (!pkt) {
if (s->oformat->flags & AVFMT_ALLOW_FLUSH)
return s->oformat->write_packet(s, pkt);
return 1;
}
ret = compute_pkt_fields2(s, s->streams[pkt->stream_index], pkt);
if (ret < 0 && !(s->oformat->flags & AVFMT_NOTIMESTAMPS))
return ret;
ret = write_packet(s, pkt);
if (ret >= 0)
s->streams[pkt->stream_index]->nb_frames++;
return ret;
}
void ff_interleave_add_packet(AVFormatContext *s, AVPacket *pkt,
int (*compare)(AVFormatContext *, AVPacket *, AVPacket *))
{
AVPacketList **next_point, *this_pktl;
this_pktl = av_mallocz(sizeof(AVPacketList));
this_pktl->pkt = *pkt;
#if FF_API_DESTRUCT_PACKET
FF_DISABLE_DEPRECATION_WARNINGS
pkt->destruct = NULL; // do not free original but only the copy
FF_ENABLE_DEPRECATION_WARNINGS
#endif
pkt->buf = NULL;
av_dup_packet(&this_pktl->pkt); // duplicate the packet if it uses non-alloced memory
if (s->streams[pkt->stream_index]->last_in_packet_buffer) {
next_point = &(s->streams[pkt->stream_index]->last_in_packet_buffer->next);
} else
next_point = &s->packet_buffer;
if (*next_point) {
if (compare(s, &s->packet_buffer_end->pkt, pkt)) {
while (!compare(s, &(*next_point)->pkt, pkt))
next_point = &(*next_point)->next;
goto next_non_null;
} else {
next_point = &(s->packet_buffer_end->next);
}
}
assert(!*next_point);
s->packet_buffer_end = this_pktl;
next_non_null:
this_pktl->next = *next_point;
s->streams[pkt->stream_index]->last_in_packet_buffer =
*next_point = this_pktl;
}
static int interleave_compare_dts(AVFormatContext *s, AVPacket *next,
AVPacket *pkt)
{
AVStream *st = s->streams[pkt->stream_index];
AVStream *st2 = s->streams[next->stream_index];
int comp = av_compare_ts(next->dts, st2->time_base, pkt->dts,
st->time_base);
if (comp == 0)
return pkt->stream_index < next->stream_index;
return comp > 0;
}
int ff_interleave_packet_per_dts(AVFormatContext *s, AVPacket *out,
AVPacket *pkt, int flush)
{
AVPacketList *pktl;
int stream_count = 0;
int i;
if (pkt) {
ff_interleave_add_packet(s, pkt, interleave_compare_dts);
}
if (s->max_interleave_delta > 0 && s->packet_buffer && !flush) {
AVPacket *top_pkt = &s->packet_buffer->pkt;
int64_t delta_dts = INT64_MIN;
int64_t top_dts = av_rescale_q(top_pkt->dts,
s->streams[top_pkt->stream_index]->time_base,
AV_TIME_BASE_Q);
for (i = 0; i < s->nb_streams; i++) {
int64_t last_dts;
const AVPacketList *last = s->streams[i]->last_in_packet_buffer;
if (!last)
continue;
last_dts = av_rescale_q(last->pkt.dts,
s->streams[i]->time_base,
AV_TIME_BASE_Q);
delta_dts = FFMAX(delta_dts, last_dts - top_dts);
stream_count++;
}
if (delta_dts > s->max_interleave_delta) {
av_log(s, AV_LOG_DEBUG,
"Delay between the first packet and last packet in the "
"muxing queue is %"PRId64" > %"PRId64": forcing output\n",
delta_dts, s->max_interleave_delta);
flush = 1;
}
} else {
for (i = 0; i < s->nb_streams; i++)
stream_count += !!s->streams[i]->last_in_packet_buffer;
}
if (stream_count && (s->internal->nb_interleaved_streams == stream_count || flush)) {
pktl = s->packet_buffer;
*out = pktl->pkt;
s->packet_buffer = pktl->next;
if (!s->packet_buffer)
s->packet_buffer_end = NULL;
if (s->streams[out->stream_index]->last_in_packet_buffer == pktl)
s->streams[out->stream_index]->last_in_packet_buffer = NULL;
av_freep(&pktl);
return 1;
} else {
av_init_packet(out);
return 0;
}
}
/**
* Interleave an AVPacket correctly so it can be muxed.
* @param out the interleaved packet will be output here
* @param in the input packet
* @param flush 1 if no further packets are available as input and all
* remaining packets should be output
* @return 1 if a packet was output, 0 if no packet could be output,
* < 0 if an error occurred
*/
static int interleave_packet(AVFormatContext *s, AVPacket *out, AVPacket *in, int flush)
{
if (s->oformat->interleave_packet) {
int ret = s->oformat->interleave_packet(s, out, in, flush);
if (in)
av_free_packet(in);
return ret;
} else
return ff_interleave_packet_per_dts(s, out, in, flush);
}
int av_interleaved_write_frame(AVFormatContext *s, AVPacket *pkt)
{
int ret, flush = 0;
ret = check_packet(s, pkt);
if (ret < 0)
return ret;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
//FIXME/XXX/HACK drop zero sized packets
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && pkt->size == 0)
return 0;
av_dlog(s, "av_interleaved_write_frame size:%d dts:%" PRId64 " pts:%" PRId64 "\n",
pkt->size, pkt->dts, pkt->pts);
if ((ret = compute_pkt_fields2(s, st, pkt)) < 0 && !(s->oformat->flags & AVFMT_NOTIMESTAMPS))
return ret;
if (pkt->dts == AV_NOPTS_VALUE && !(s->oformat->flags & AVFMT_NOTIMESTAMPS))
return AVERROR(EINVAL);
} else {
av_dlog(s, "av_interleaved_write_frame FLUSH\n");
flush = 1;
}
for (;; ) {
AVPacket opkt;
int ret = interleave_packet(s, &opkt, pkt, flush);
if (ret <= 0) //FIXME cleanup needed for ret<0 ?
return ret;
ret = write_packet(s, &opkt);
if (ret >= 0)
s->streams[opkt.stream_index]->nb_frames++;
av_free_packet(&opkt);
pkt = NULL;
if (ret < 0)
return ret;
}
}
int av_write_trailer(AVFormatContext *s)
{
int ret, i;
for (;; ) {
AVPacket pkt;
ret = interleave_packet(s, &pkt, NULL, 1);
if (ret < 0) //FIXME cleanup needed for ret<0 ?
goto fail;
if (!ret)
break;
ret = write_packet(s, &pkt);
if (ret >= 0)
s->streams[pkt.stream_index]->nb_frames++;
av_free_packet(&pkt);
if (ret < 0)
goto fail;
}
if (s->oformat->write_trailer)
ret = s->oformat->write_trailer(s);
if (!(s->oformat->flags & AVFMT_NOFILE))
avio_flush(s->pb);
fail:
for (i = 0; i < s->nb_streams; i++) {
av_freep(&s->streams[i]->priv_data);
av_freep(&s->streams[i]->index_entries);
}
if (s->oformat->priv_class)
av_opt_free(s->priv_data);
av_freep(&s->priv_data);
return ret;
}
int ff_write_chained(AVFormatContext *dst, int dst_stream, AVPacket *pkt,
AVFormatContext *src)
{
AVPacket local_pkt;
local_pkt = *pkt;
local_pkt.stream_index = dst_stream;
if (pkt->pts != AV_NOPTS_VALUE)
local_pkt.pts = av_rescale_q(pkt->pts,
src->streams[pkt->stream_index]->time_base,
dst->streams[dst_stream]->time_base);
if (pkt->dts != AV_NOPTS_VALUE)
local_pkt.dts = av_rescale_q(pkt->dts,
src->streams[pkt->stream_index]->time_base,
dst->streams[dst_stream]->time_base);
return av_write_frame(dst, &local_pkt);
}