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FFmpeg/libavformat/mux.c
Clément Bœsch 0072116ccb lavf: split packets before muxing. 
After demuxing, data and side are merged. Before decoding, they are
split. Encoder will perform with data and side split. This means that a
muxer can receive split data (after encoding) but also merged data (if
called directly after demuxing). This commit makes sure data and side
are split for the muxer.
2012-12-20 16:13:52 +01:00

816 lines
26 KiB
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/*
* muxing functions for use within FFmpeg
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* 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
*/
/* #define DEBUG */
#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 "libavutil/timestamp.h"
#include "metadata.h"
#include "id3v2.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.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 libavformat
*/
/* 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;
}
AVRational ff_choose_timebase(AVFormatContext *s, AVStream *st, int min_precission)
{
AVRational q;
int j;
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
q = (AVRational){1, st->codec->sample_rate};
} else {
q = st->codec->time_base;
}
for (j=2; j<14; j+= 1+(j>2))
while (q.den / q.num < min_precission && q.num % j == 0)
q.num /= j;
while (q.den / q.num < min_precission && q.den < (1<<24))
q.den <<= 1;
return q;
}
int avformat_alloc_output_context2(AVFormatContext **avctx, AVOutputFormat *oformat,
const char *format, const char *filename)
{
AVFormatContext *s = avformat_alloc_context();
int ret = 0;
*avctx = NULL;
if (!s)
goto nomem;
if (!oformat) {
if (format) {
oformat = av_guess_format(format, NULL, NULL);
if (!oformat) {
av_log(s, AV_LOG_ERROR, "Requested output format '%s' is not a suitable output format\n", format);
ret = AVERROR(EINVAL);
goto error;
}
} else {
oformat = av_guess_format(NULL, filename, NULL);
if (!oformat) {
ret = AVERROR(EINVAL);
av_log(s, AV_LOG_ERROR, "Unable to find a suitable output format for '%s'\n",
filename);
goto error;
}
}
}
s->oformat = oformat;
if (s->oformat->priv_data_size > 0) {
s->priv_data = av_mallocz(s->oformat->priv_data_size);
if (!s->priv_data)
goto nomem;
if (s->oformat->priv_class) {
*(const AVClass**)s->priv_data= s->oformat->priv_class;
av_opt_set_defaults(s->priv_data);
}
} else
s->priv_data = NULL;
if (filename)
av_strlcpy(s->filename, filename, sizeof(s->filename));
*avctx = s;
return 0;
nomem:
av_log(s, AV_LOG_ERROR, "Out of memory\n");
ret = AVERROR(ENOMEM);
error:
avformat_free_context(s);
return ret;
}
#if FF_API_ALLOC_OUTPUT_CONTEXT
AVFormatContext *avformat_alloc_output_context(const char *format,
AVOutputFormat *oformat, const char *filename)
{
AVFormatContext *avctx;
int ret = avformat_alloc_output_context2(&avctx, oformat, format, filename);
return ret < 0 ? NULL : avctx;
}
#endif
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;
if (s->priv_data && s->oformat->priv_class && *(const AVClass**)s->priv_data==s->oformat->priv_class &&
(ret = av_opt_set_dict(s->priv_data, &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)
&& FFABS(av_q2d(st->sample_aspect_ratio) - av_q2d(codec->sample_aspect_ratio)) > 0.004*av_q2d(st->sample_aspect_ratio)
) {
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) == 0
|| av_codec_get_tag(of->codec_tag, codec->codec_id) == MKTAG('r', 'a', 'w', ' '))
&& !validate_codec_tag(s, st)) {
// the current rawvideo encoding system ends up setting
// the wrong codec_tag for avi/mov, we override it here
codec->codec_tag = 0;
}
if (codec->codec_tag) {
if (!validate_codec_tag(s, st)) {
char tagbuf[32], cortag[32];
av_get_codec_tag_string(tagbuf, sizeof(tagbuf), codec->codec_tag);
av_get_codec_tag_string(cortag, sizeof(cortag), av_codec_get_tag(s->oformat->codec_tag, codec->codec_id));
av_log(s, AV_LOG_ERROR,
"Tag %s/0x%08x incompatible with output codec id '%d' (%s)\n",
tagbuf, codec->codec_tag, codec->codec_id, cortag);
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 (!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 && s->pb && s->pb->error < 0)
ret = s->pb->error;
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 > 0);
int num, den, frame_size, i;
av_dlog(s, "compute_pkt_fields2: pts:%s dts:%s cur_dts:%s b:%d size:%d st:%d\n",
av_ts2str(pkt->pts), av_ts2str(pkt->dts), av_ts2str(st->cur_dts), delay, pkt->size, pkt->stream_index);
/* 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) {
static int warned;
if (!warned) {
av_log(s, AV_LOG_WARNING, "Encoder did not produce proper pts, making some up.\n");
warned = 1;
}
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: %s >= %s\n",
st->index, av_ts2str(st->cur_dts), av_ts2str(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 (%s) < dts (%s) in stream %d\n",
av_ts2str(pkt->pts), av_ts2str(pkt->dts), st->index);
return AVERROR(EINVAL);
}
av_dlog(s, "av_write_frame: pts2:%s dts2:%s\n",
av_ts2str(pkt->pts), av_ts2str(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;
}
/**
* Move side data from payload to internal struct, call muxer, and restore
* original packet.
*/
static inline int split_write_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret;
AVPacket spkt = *pkt;
av_packet_split_side_data(&spkt);
ret = s->oformat->write_packet(s, &spkt);
spkt.data = NULL;
av_destruct_packet(&spkt);
return ret;
}
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
{
int ret;
if (!pkt) {
if (s->oformat->flags & AVFMT_ALLOW_FLUSH) {
ret = s->oformat->write_packet(s, NULL);
if (ret >= 0 && s->pb && s->pb->error < 0)
ret = s->pb->error;
return ret;
}
return 1;
}
ret = compute_pkt_fields2(s, s->streams[pkt->stream_index], pkt);
if (ret < 0 && !(s->oformat->flags & AVFMT_NOTIMESTAMPS))
return ret;
ret = split_write_packet(s, pkt);
if (ret >= 0 && s->pb && s->pb->error < 0)
ret = s->pb->error;
if (ret >= 0)
s->streams[pkt->stream_index]->nb_frames++;
return ret;
}
#define CHUNK_START 0x1000
int ff_interleave_add_packet(AVFormatContext *s, AVPacket *pkt,
int (*compare)(AVFormatContext *, AVPacket *, AVPacket *))
{
AVPacketList **next_point, *this_pktl;
AVStream *st = s->streams[pkt->stream_index];
int chunked = s->max_chunk_size || s->max_chunk_duration;
this_pktl = av_mallocz(sizeof(AVPacketList));
if (!this_pktl)
return AVERROR(ENOMEM);
this_pktl->pkt = *pkt;
pkt->destruct = NULL; // do not free original but only the copy
av_dup_packet(&this_pktl->pkt); // duplicate the packet if it uses non-allocated memory
if (s->streams[pkt->stream_index]->last_in_packet_buffer) {
next_point = &(st->last_in_packet_buffer->next);
} else {
next_point = &s->packet_buffer;
}
if (*next_point) {
if (chunked) {
uint64_t max= av_rescale_q(s->max_chunk_duration, AV_TIME_BASE_Q, st->time_base);
if ( st->interleaver_chunk_size + pkt->size <= s->max_chunk_size-1U
&& st->interleaver_chunk_duration + pkt->duration <= max-1U) {
st->interleaver_chunk_size += pkt->size;
st->interleaver_chunk_duration += pkt->duration;
goto next_non_null;
} else {
st->interleaver_chunk_size =
st->interleaver_chunk_duration = 0;
this_pktl->pkt.flags |= CHUNK_START;
}
}
if (compare(s, &s->packet_buffer_end->pkt, pkt)) {
while ( *next_point
&& ((chunked && !((*next_point)->pkt.flags&CHUNK_START))
|| !compare(s, &(*next_point)->pkt, pkt)))
next_point = &(*next_point)->next;
if (*next_point)
goto next_non_null;
} else {
next_point = &(s->packet_buffer_end->next);
}
}
av_assert1(!*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;
return 0;
}
static int ff_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 (s->audio_preload && ((st->codec->codec_type == AVMEDIA_TYPE_AUDIO) != (st2->codec->codec_type == AVMEDIA_TYPE_AUDIO))) {
int64_t ts = av_rescale_q(pkt ->dts, st ->time_base, AV_TIME_BASE_Q) - s->audio_preload*(st ->codec->codec_type == AVMEDIA_TYPE_AUDIO);
int64_t ts2= av_rescale_q(next->dts, st2->time_base, AV_TIME_BASE_Q) - s->audio_preload*(st2->codec->codec_type == AVMEDIA_TYPE_AUDIO);
if (ts == ts2) {
ts= ( pkt ->dts* st->time_base.num*AV_TIME_BASE - s->audio_preload*(int64_t)(st ->codec->codec_type == AVMEDIA_TYPE_AUDIO)* st->time_base.den)*st2->time_base.den
-( next->dts*st2->time_base.num*AV_TIME_BASE - s->audio_preload*(int64_t)(st2->codec->codec_type == AVMEDIA_TYPE_AUDIO)*st2->time_base.den)* st->time_base.den;
ts2=0;
}
comp= (ts>ts2) - (ts<ts2);
}
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, noninterleaved_count = 0;
int64_t delta_dts_max = 0;
int i, ret;
if (pkt) {
ret = ff_interleave_add_packet(s, pkt, ff_interleave_compare_dts);
if (ret < 0)
return ret;
}
for (i = 0; i < s->nb_streams; i++) {
if (s->streams[i]->last_in_packet_buffer) {
++stream_count;
} else if (s->streams[i]->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
++noninterleaved_count;
}
}
if (s->nb_streams == stream_count) {
flush = 1;
} else if (!flush) {
for (i=0; i < s->nb_streams; i++) {
if (s->streams[i]->last_in_packet_buffer) {
int64_t delta_dts =
av_rescale_q(s->streams[i]->last_in_packet_buffer->pkt.dts,
s->streams[i]->time_base,
AV_TIME_BASE_Q) -
av_rescale_q(s->packet_buffer->pkt.dts,
s->streams[s->packet_buffer->pkt.stream_index]->time_base,
AV_TIME_BASE_Q);
delta_dts_max= FFMAX(delta_dts_max, delta_dts);
}
}
if (s->nb_streams == stream_count+noninterleaved_count &&
delta_dts_max > 20*AV_TIME_BASE) {
av_log(s, AV_LOG_DEBUG, "flushing with %d noninterleaved\n", noninterleaved_count);
flush = 1;
}
}
if (stream_count && flush) {
AVStream *st;
pktl = s->packet_buffer;
*out = pktl->pkt;
st = s->streams[out->stream_index];
s->packet_buffer = pktl->next;
if (!s->packet_buffer)
s->packet_buffer_end = NULL;
if (st->last_in_packet_buffer == pktl)
st->last_in_packet_buffer = NULL;
av_freep(&pktl);
if (s->avoid_negative_ts > 0) {
if (out->dts != AV_NOPTS_VALUE) {
if (!st->mux_ts_offset && out->dts < 0) {
for (i = 0; i < s->nb_streams; i++) {
s->streams[i]->mux_ts_offset =
av_rescale_q_rnd(-out->dts,
st->time_base,
s->streams[i]->time_base,
AV_ROUND_UP);
}
}
out->dts += st->mux_ts_offset;
}
if (out->pts != AV_NOPTS_VALUE)
out->pts += st->mux_ts_offset;
}
return 1;
} else {
av_init_packet(out);
return 0;
}
}
#if FF_API_INTERLEAVE_PACKET
int av_interleave_packet_per_dts(AVFormatContext *s, AVPacket *out,
AVPacket *pkt, int flush)
{
return ff_interleave_packet_per_dts(s, out, pkt, flush);
}
#endif
/**
* 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;
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:%s pts:%s\n",
pkt->size, av_ts2str(pkt->dts), av_ts2str(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 = split_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;
if(s->pb && s->pb->error)
return s->pb->error;
}
}
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 = split_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->pb && s->pb->error)
goto fail;
}
if (s->oformat->write_trailer)
ret = s->oformat->write_trailer(s);
fail:
if (s->pb)
avio_flush(s->pb);
if (ret == 0)
ret = s->pb ? s->pb->error : 0;
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 av_get_output_timestamp(struct AVFormatContext *s, int stream,
int64_t *dts, int64_t *wall)
{
if (!s->oformat || !s->oformat->get_output_timestamp)
return AVERROR(ENOSYS);
s->oformat->get_output_timestamp(s, stream, dts, wall);
return 0;
}
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