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FFmpeg/libavformat/mux.c

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/*
* 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
*/
/* #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 "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 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)) {
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 (!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)
{
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;
}
return s->oformat->write_packet(s, pkt);
}
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
{
int 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
pkt->destruct = NULL; // do not free original but only the copy
#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);
}
for (i = 0; i < s->nb_streams; i++)
stream_count += !!s->streams[i]->last_in_packet_buffer;
if (stream_count && (s->nb_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;
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;
}
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