mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
1afddbe59e
This will allow us to avoid copying the packets in many cases. This breaks ABI.
606 lines
18 KiB
C
606 lines
18 KiB
C
/*
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* muxing functions for use within Libav
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* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/* #define DEBUG */
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#include "avformat.h"
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#include "avio_internal.h"
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#include "internal.h"
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#include "libavcodec/internal.h"
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#include "libavcodec/bytestream.h"
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#include "libavutil/opt.h"
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#include "libavutil/dict.h"
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#include "libavutil/pixdesc.h"
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#include "metadata.h"
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#include "id3v2.h"
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#include "libavutil/avassert.h"
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#include "libavutil/avstring.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/parseutils.h"
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#include "libavutil/time.h"
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#include "riff.h"
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#include "audiointerleave.h"
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#include "url.h"
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#include <stdarg.h>
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#if CONFIG_NETWORK
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#include "network.h"
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#endif
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#undef NDEBUG
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#include <assert.h>
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/**
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* @file
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* muxing functions for use within Libav
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*/
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/* fraction handling */
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/**
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* f = val + (num / den) + 0.5.
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*
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* 'num' is normalized so that it is such as 0 <= num < den.
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*
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* @param f fractional number
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* @param val integer value
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* @param num must be >= 0
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* @param den must be >= 1
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*/
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static void frac_init(AVFrac *f, int64_t val, int64_t num, int64_t den)
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{
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num += (den >> 1);
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if (num >= den) {
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val += num / den;
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num = num % den;
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}
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f->val = val;
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f->num = num;
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f->den = den;
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}
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/**
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* Fractional addition to f: f = f + (incr / f->den).
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*
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* @param f fractional number
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* @param incr increment, can be positive or negative
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*/
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static void frac_add(AVFrac *f, int64_t incr)
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{
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int64_t num, den;
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num = f->num + incr;
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den = f->den;
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if (num < 0) {
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f->val += num / den;
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num = num % den;
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if (num < 0) {
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num += den;
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f->val--;
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}
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} else if (num >= den) {
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f->val += num / den;
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num = num % den;
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}
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f->num = num;
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}
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static int validate_codec_tag(AVFormatContext *s, AVStream *st)
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{
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const AVCodecTag *avctag;
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int n;
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enum AVCodecID id = AV_CODEC_ID_NONE;
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unsigned int tag = 0;
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/**
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* Check that tag + id is in the table
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* If neither is in the table -> OK
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* If tag is in the table with another id -> FAIL
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* If id is in the table with another tag -> FAIL unless strict < normal
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*/
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for (n = 0; s->oformat->codec_tag[n]; n++) {
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avctag = s->oformat->codec_tag[n];
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while (avctag->id != AV_CODEC_ID_NONE) {
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if (avpriv_toupper4(avctag->tag) == avpriv_toupper4(st->codec->codec_tag)) {
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id = avctag->id;
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if (id == st->codec->codec_id)
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return 1;
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}
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if (avctag->id == st->codec->codec_id)
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tag = avctag->tag;
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avctag++;
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}
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}
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if (id != AV_CODEC_ID_NONE)
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return 0;
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if (tag && (st->codec->strict_std_compliance >= FF_COMPLIANCE_NORMAL))
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return 0;
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return 1;
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}
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static int init_muxer(AVFormatContext *s, AVDictionary **options)
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{
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int ret = 0, i;
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AVStream *st;
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AVDictionary *tmp = NULL;
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AVCodecContext *codec = NULL;
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AVOutputFormat *of = s->oformat;
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if (options)
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av_dict_copy(&tmp, *options, 0);
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if ((ret = av_opt_set_dict(s, &tmp)) < 0)
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goto fail;
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// some sanity checks
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if (s->nb_streams == 0 && !(of->flags & AVFMT_NOSTREAMS)) {
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av_log(s, AV_LOG_ERROR, "no streams\n");
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ret = AVERROR(EINVAL);
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goto fail;
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}
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for (i = 0; i < s->nb_streams; i++) {
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st = s->streams[i];
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codec = st->codec;
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switch (codec->codec_type) {
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case AVMEDIA_TYPE_AUDIO:
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if (codec->sample_rate <= 0) {
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av_log(s, AV_LOG_ERROR, "sample rate not set\n");
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ret = AVERROR(EINVAL);
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goto fail;
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}
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if (!codec->block_align)
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codec->block_align = codec->channels *
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av_get_bits_per_sample(codec->codec_id) >> 3;
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break;
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case AVMEDIA_TYPE_VIDEO:
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if (codec->time_base.num <= 0 ||
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codec->time_base.den <= 0) { //FIXME audio too?
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av_log(s, AV_LOG_ERROR, "time base not set\n");
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ret = AVERROR(EINVAL);
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goto fail;
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}
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if ((codec->width <= 0 || codec->height <= 0) &&
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!(of->flags & AVFMT_NODIMENSIONS)) {
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av_log(s, AV_LOG_ERROR, "dimensions not set\n");
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ret = AVERROR(EINVAL);
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goto fail;
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}
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if (av_cmp_q(st->sample_aspect_ratio,
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codec->sample_aspect_ratio)) {
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av_log(s, AV_LOG_ERROR, "Aspect ratio mismatch between muxer "
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"(%d/%d) and encoder layer (%d/%d)\n",
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st->sample_aspect_ratio.num, st->sample_aspect_ratio.den,
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codec->sample_aspect_ratio.num,
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codec->sample_aspect_ratio.den);
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ret = AVERROR(EINVAL);
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goto fail;
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}
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break;
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}
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if (of->codec_tag) {
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if (codec->codec_tag &&
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codec->codec_id == AV_CODEC_ID_RAWVIDEO &&
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!av_codec_get_tag(of->codec_tag, codec->codec_id) &&
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!validate_codec_tag(s, st)) {
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// the current rawvideo encoding system ends up setting
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// the wrong codec_tag for avi, we override it here
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codec->codec_tag = 0;
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}
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if (codec->codec_tag) {
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if (!validate_codec_tag(s, st)) {
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char tagbuf[32];
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av_get_codec_tag_string(tagbuf, sizeof(tagbuf), codec->codec_tag);
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av_log(s, AV_LOG_ERROR,
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"Tag %s/0x%08x incompatible with output codec id '%d'\n",
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tagbuf, codec->codec_tag, codec->codec_id);
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ret = AVERROR_INVALIDDATA;
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goto fail;
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}
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} else
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codec->codec_tag = av_codec_get_tag(of->codec_tag, codec->codec_id);
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}
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if (of->flags & AVFMT_GLOBALHEADER &&
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!(codec->flags & CODEC_FLAG_GLOBAL_HEADER))
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av_log(s, AV_LOG_WARNING,
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"Codec for stream %d does not use global headers "
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"but container format requires global headers\n", i);
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}
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if (!s->priv_data && of->priv_data_size > 0) {
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s->priv_data = av_mallocz(of->priv_data_size);
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if (!s->priv_data) {
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ret = AVERROR(ENOMEM);
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goto fail;
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}
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if (of->priv_class) {
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*(const AVClass **)s->priv_data = of->priv_class;
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av_opt_set_defaults(s->priv_data);
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if ((ret = av_opt_set_dict(s->priv_data, &tmp)) < 0)
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goto fail;
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}
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}
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/* set muxer identification string */
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if (s->nb_streams && !(s->streams[0]->codec->flags & CODEC_FLAG_BITEXACT)) {
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av_dict_set(&s->metadata, "encoder", LIBAVFORMAT_IDENT, 0);
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}
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if (options) {
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av_dict_free(options);
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*options = tmp;
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}
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return 0;
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fail:
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av_dict_free(&tmp);
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return ret;
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}
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static int init_pts(AVFormatContext *s)
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{
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int i;
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AVStream *st;
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/* init PTS generation */
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for (i = 0; i < s->nb_streams; i++) {
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int64_t den = AV_NOPTS_VALUE;
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st = s->streams[i];
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switch (st->codec->codec_type) {
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case AVMEDIA_TYPE_AUDIO:
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den = (int64_t)st->time_base.num * st->codec->sample_rate;
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break;
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case AVMEDIA_TYPE_VIDEO:
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den = (int64_t)st->time_base.num * st->codec->time_base.den;
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break;
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default:
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break;
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}
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if (den != AV_NOPTS_VALUE) {
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if (den <= 0)
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return AVERROR_INVALIDDATA;
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frac_init(&st->pts, 0, 0, den);
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}
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}
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return 0;
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}
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int avformat_write_header(AVFormatContext *s, AVDictionary **options)
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{
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int ret = 0;
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if (ret = init_muxer(s, options))
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return ret;
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if (s->oformat->write_header) {
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ret = s->oformat->write_header(s);
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if (ret < 0)
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return ret;
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}
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if ((ret = init_pts(s)) < 0)
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return ret;
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return 0;
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}
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//FIXME merge with compute_pkt_fields
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static int compute_pkt_fields2(AVFormatContext *s, AVStream *st, AVPacket *pkt)
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{
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int delay = FFMAX(st->codec->has_b_frames, !!st->codec->max_b_frames);
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int num, den, frame_size, i;
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av_dlog(s, "compute_pkt_fields2: pts:%" PRId64 " dts:%" PRId64 " cur_dts:%" PRId64 " b:%d size:%d st:%d\n",
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pkt->pts, pkt->dts, st->cur_dts, delay, pkt->size, pkt->stream_index);
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/* if(pkt->pts == AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE)
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* return AVERROR(EINVAL);*/
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/* duration field */
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if (pkt->duration == 0) {
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ff_compute_frame_duration(&num, &den, st, NULL, pkt);
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if (den && num) {
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pkt->duration = av_rescale(1, num * (int64_t)st->time_base.den * st->codec->ticks_per_frame, den * (int64_t)st->time_base.num);
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}
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}
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if (pkt->pts == AV_NOPTS_VALUE && pkt->dts != AV_NOPTS_VALUE && delay == 0)
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pkt->pts = pkt->dts;
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//XXX/FIXME this is a temporary hack until all encoders output pts
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if ((pkt->pts == 0 || pkt->pts == AV_NOPTS_VALUE) && pkt->dts == AV_NOPTS_VALUE && !delay) {
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pkt->dts =
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// pkt->pts= st->cur_dts;
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pkt->pts = st->pts.val;
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}
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//calculate dts from pts
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if (pkt->pts != AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE && delay <= MAX_REORDER_DELAY) {
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st->pts_buffer[0] = pkt->pts;
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for (i = 1; i < delay + 1 && st->pts_buffer[i] == AV_NOPTS_VALUE; i++)
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st->pts_buffer[i] = pkt->pts + (i - delay - 1) * pkt->duration;
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for (i = 0; i<delay && st->pts_buffer[i] > st->pts_buffer[i + 1]; i++)
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FFSWAP(int64_t, st->pts_buffer[i], st->pts_buffer[i + 1]);
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pkt->dts = st->pts_buffer[0];
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}
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if (st->cur_dts && st->cur_dts != AV_NOPTS_VALUE &&
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((!(s->oformat->flags & AVFMT_TS_NONSTRICT) &&
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st->cur_dts >= pkt->dts) || st->cur_dts > pkt->dts)) {
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av_log(s, AV_LOG_ERROR,
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"Application provided invalid, non monotonically increasing dts to muxer in stream %d: %" PRId64 " >= %" PRId64 "\n",
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st->index, st->cur_dts, pkt->dts);
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return AVERROR(EINVAL);
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}
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if (pkt->dts != AV_NOPTS_VALUE && pkt->pts != AV_NOPTS_VALUE && pkt->pts < pkt->dts) {
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av_log(s, AV_LOG_ERROR, "pts < dts in stream %d\n", st->index);
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return AVERROR(EINVAL);
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}
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av_dlog(s, "av_write_frame: pts2:%"PRId64" dts2:%"PRId64"\n",
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pkt->pts, pkt->dts);
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st->cur_dts = pkt->dts;
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st->pts.val = pkt->dts;
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/* update pts */
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switch (st->codec->codec_type) {
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case AVMEDIA_TYPE_AUDIO:
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frame_size = ff_get_audio_frame_size(st->codec, pkt->size, 1);
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/* HACK/FIXME, we skip the initial 0 size packets as they are most
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* likely equal to the encoder delay, but it would be better if we
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* had the real timestamps from the encoder */
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if (frame_size >= 0 && (pkt->size || st->pts.num != st->pts.den >> 1 || st->pts.val)) {
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frac_add(&st->pts, (int64_t)st->time_base.den * frame_size);
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}
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break;
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case AVMEDIA_TYPE_VIDEO:
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frac_add(&st->pts, (int64_t)st->time_base.den * st->codec->time_base.num);
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break;
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default:
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break;
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}
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return 0;
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}
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int av_write_frame(AVFormatContext *s, AVPacket *pkt)
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{
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int ret;
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if (!pkt) {
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if (s->oformat->flags & AVFMT_ALLOW_FLUSH)
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return s->oformat->write_packet(s, pkt);
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return 1;
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}
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ret = compute_pkt_fields2(s, s->streams[pkt->stream_index], pkt);
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if (ret < 0 && !(s->oformat->flags & AVFMT_NOTIMESTAMPS))
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return ret;
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ret = s->oformat->write_packet(s, pkt);
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if (ret >= 0)
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s->streams[pkt->stream_index]->nb_frames++;
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return ret;
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}
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void ff_interleave_add_packet(AVFormatContext *s, AVPacket *pkt,
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int (*compare)(AVFormatContext *, AVPacket *, AVPacket *))
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{
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AVPacketList **next_point, *this_pktl;
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this_pktl = av_mallocz(sizeof(AVPacketList));
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this_pktl->pkt = *pkt;
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#if FF_API_DESTRUCT_PACKET
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pkt->destruct = NULL; // do not free original but only the copy
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#endif
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pkt->buf = NULL;
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av_dup_packet(&this_pktl->pkt); // duplicate the packet if it uses non-alloced memory
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if (s->streams[pkt->stream_index]->last_in_packet_buffer) {
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next_point = &(s->streams[pkt->stream_index]->last_in_packet_buffer->next);
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} else
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next_point = &s->packet_buffer;
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if (*next_point) {
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if (compare(s, &s->packet_buffer_end->pkt, pkt)) {
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while (!compare(s, &(*next_point)->pkt, pkt))
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next_point = &(*next_point)->next;
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goto next_non_null;
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} else {
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next_point = &(s->packet_buffer_end->next);
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}
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}
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assert(!*next_point);
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s->packet_buffer_end = this_pktl;
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next_non_null:
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this_pktl->next = *next_point;
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s->streams[pkt->stream_index]->last_in_packet_buffer =
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*next_point = this_pktl;
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}
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static int ff_interleave_compare_dts(AVFormatContext *s, AVPacket *next, AVPacket *pkt)
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{
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AVStream *st = s->streams[pkt->stream_index];
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AVStream *st2 = s->streams[next->stream_index];
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int comp = av_compare_ts(next->dts, st2->time_base, pkt->dts,
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st->time_base);
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if (comp == 0)
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return pkt->stream_index < next->stream_index;
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return comp > 0;
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}
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int ff_interleave_packet_per_dts(AVFormatContext *s, AVPacket *out,
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AVPacket *pkt, int flush)
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{
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AVPacketList *pktl;
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int stream_count = 0;
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int i;
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if (pkt) {
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ff_interleave_add_packet(s, pkt, ff_interleave_compare_dts);
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}
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for (i = 0; i < s->nb_streams; i++)
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stream_count += !!s->streams[i]->last_in_packet_buffer;
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if (stream_count && (s->nb_streams == stream_count || flush)) {
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pktl = s->packet_buffer;
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*out = pktl->pkt;
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|
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;
|
|
}
|
|
}
|
|
|
|
#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:%" 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 = s->oformat->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 = s->oformat->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;
|
|
}
|