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https://git.ffmpeg.org/ffmpeg.git
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0072116ccb
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.
816 lines
26 KiB
C
816 lines
26 KiB
C
/*
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* muxing functions for use within FFmpeg
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* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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 "libavutil/timestamp.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 libavformat
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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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AVRational ff_choose_timebase(AVFormatContext *s, AVStream *st, int min_precission)
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{
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AVRational q;
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int j;
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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q = (AVRational){1, st->codec->sample_rate};
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} else {
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q = st->codec->time_base;
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}
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for (j=2; j<14; j+= 1+(j>2))
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while (q.den / q.num < min_precission && q.num % j == 0)
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q.num /= j;
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while (q.den / q.num < min_precission && q.den < (1<<24))
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q.den <<= 1;
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return q;
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}
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int avformat_alloc_output_context2(AVFormatContext **avctx, AVOutputFormat *oformat,
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const char *format, const char *filename)
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{
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AVFormatContext *s = avformat_alloc_context();
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int ret = 0;
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*avctx = NULL;
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if (!s)
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goto nomem;
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if (!oformat) {
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if (format) {
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oformat = av_guess_format(format, NULL, NULL);
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if (!oformat) {
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av_log(s, AV_LOG_ERROR, "Requested output format '%s' is not a suitable output format\n", format);
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ret = AVERROR(EINVAL);
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goto error;
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}
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} else {
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oformat = av_guess_format(NULL, filename, NULL);
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if (!oformat) {
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ret = AVERROR(EINVAL);
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av_log(s, AV_LOG_ERROR, "Unable to find a suitable output format for '%s'\n",
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filename);
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goto error;
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}
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}
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}
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s->oformat = oformat;
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if (s->oformat->priv_data_size > 0) {
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s->priv_data = av_mallocz(s->oformat->priv_data_size);
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if (!s->priv_data)
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goto nomem;
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if (s->oformat->priv_class) {
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*(const AVClass**)s->priv_data= s->oformat->priv_class;
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av_opt_set_defaults(s->priv_data);
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}
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} else
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s->priv_data = NULL;
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if (filename)
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av_strlcpy(s->filename, filename, sizeof(s->filename));
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*avctx = s;
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return 0;
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nomem:
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av_log(s, AV_LOG_ERROR, "Out of memory\n");
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ret = AVERROR(ENOMEM);
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error:
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avformat_free_context(s);
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return ret;
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}
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#if FF_API_ALLOC_OUTPUT_CONTEXT
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AVFormatContext *avformat_alloc_output_context(const char *format,
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AVOutputFormat *oformat, const char *filename)
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{
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AVFormatContext *avctx;
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int ret = avformat_alloc_output_context2(&avctx, oformat, format, filename);
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return ret < 0 ? NULL : avctx;
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}
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#endif
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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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if (s->priv_data && s->oformat->priv_class && *(const AVClass**)s->priv_data==s->oformat->priv_class &&
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(ret = av_opt_set_dict(s->priv_data, &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, codec->sample_aspect_ratio)
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&& FFABS(av_q2d(st->sample_aspect_ratio) - av_q2d(codec->sample_aspect_ratio)) > 0.004*av_q2d(st->sample_aspect_ratio)
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) {
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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) == 0
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|| av_codec_get_tag(of->codec_tag, codec->codec_id) == MKTAG('r', 'a', 'w', ' '))
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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/mov, 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], cortag[32];
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av_get_codec_tag_string(tagbuf, sizeof(tagbuf), codec->codec_tag);
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av_get_codec_tag_string(cortag, sizeof(cortag), av_codec_get_tag(s->oformat->codec_tag, codec->codec_id));
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av_log(s, AV_LOG_ERROR,
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"Tag %s/0x%08x incompatible with output codec id '%d' (%s)\n",
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tagbuf, codec->codec_tag, codec->codec_id, cortag);
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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 && s->pb && s->pb->error < 0)
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ret = s->pb->error;
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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 > 0);
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int num, den, frame_size, i;
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av_dlog(s, "compute_pkt_fields2: pts:%s dts:%s cur_dts:%s b:%d size:%d st:%d\n",
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av_ts2str(pkt->pts), av_ts2str(pkt->dts), av_ts2str(st->cur_dts), delay, pkt->size, pkt->stream_index);
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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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static int warned;
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if (!warned) {
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av_log(s, AV_LOG_WARNING, "Encoder did not produce proper pts, making some up.\n");
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warned = 1;
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}
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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: %s >= %s\n",
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st->index, av_ts2str(st->cur_dts), av_ts2str(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 (%s) < dts (%s) in stream %d\n",
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av_ts2str(pkt->pts), av_ts2str(pkt->dts), st->index);
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return AVERROR(EINVAL);
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}
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av_dlog(s, "av_write_frame: pts2:%s dts2:%s\n",
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av_ts2str(pkt->pts), av_ts2str(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
|
|
* 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;
|
|
}
|