From 55f9037f38bc3beacb2f5a17408c1d24c077d7fd Mon Sep 17 00:00:00 2001 From: Luca Barbato Date: Mon, 1 Oct 2012 00:49:16 +0200 Subject: avformat: split muxing functions from util.c --- libavformat/mux.c | 564 ++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 564 insertions(+) create mode 100644 libavformat/mux.c (limited to 'libavformat/mux.c') diff --git a/libavformat/mux.c b/libavformat/mux.c new file mode 100644 index 0000000000..ee5352c868 --- /dev/null +++ b/libavformat/mux.c @@ -0,0 +1,564 @@ +/* + * 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 +#if CONFIG_NETWORK +#include "network.h" +#endif + +#undef NDEBUG +#include + +/** + * @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; +} + +int avformat_write_header(AVFormatContext *s, AVDictionary **options) +{ + int ret = 0, i; + AVStream *st; + AVDictionary *tmp = NULL; + + 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 && !(s->oformat->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]; + + switch (st->codec->codec_type) { + case AVMEDIA_TYPE_AUDIO: + if (st->codec->sample_rate <= 0) { + av_log(s, AV_LOG_ERROR, "sample rate not set\n"); + ret = AVERROR(EINVAL); + goto fail; + } + if (!st->codec->block_align) + st->codec->block_align = st->codec->channels * + av_get_bits_per_sample(st->codec->codec_id) >> 3; + break; + case AVMEDIA_TYPE_VIDEO: + if (st->codec->time_base.num <= 0 || st->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 ((st->codec->width <= 0 || st->codec->height <= 0) && !(s->oformat->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, st->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, + st->codec->sample_aspect_ratio.num, + st->codec->sample_aspect_ratio.den); + ret = AVERROR(EINVAL); + goto fail; + } + break; + } + + if (s->oformat->codec_tag) { + if (st->codec->codec_tag && st->codec->codec_id == AV_CODEC_ID_RAWVIDEO && av_codec_get_tag(s->oformat->codec_tag, st->codec->codec_id) == 0 && !validate_codec_tag(s, st)) { + //the current rawvideo encoding system ends up setting the wrong codec_tag for avi, we override it here + st->codec->codec_tag = 0; + } + if (st->codec->codec_tag) { + if (!validate_codec_tag(s, st)) { + char tagbuf[32]; + av_get_codec_tag_string(tagbuf, sizeof(tagbuf), st->codec->codec_tag); + av_log(s, AV_LOG_ERROR, + "Tag %s/0x%08x incompatible with output codec id '%d'\n", + tagbuf, st->codec->codec_tag, st->codec->codec_id); + ret = AVERROR_INVALIDDATA; + goto fail; + } + } else + st->codec->codec_tag = av_codec_get_tag(s->oformat->codec_tag, st->codec->codec_id); + } + + if (s->oformat->flags & AVFMT_GLOBALHEADER && + !(st->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 && s->oformat->priv_data_size > 0) { + s->priv_data = av_mallocz(s->oformat->priv_data_size); + if (!s->priv_data) { + ret = AVERROR(ENOMEM); + goto fail; + } + if (s->oformat->priv_class) { + *(const AVClass **)s->priv_data = s->oformat->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 (s->oformat->write_header) { + ret = s->oformat->write_header(s); + if (ret < 0) + goto fail; + } + + /* 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) { + ret = AVERROR_INVALIDDATA; + goto fail; + } + frac_init(&st->pts, 0, 0, den); + } + } + + if (options) { + av_dict_free(options); + *options = tmp; + } + return 0; +fail: + av_dict_free(&tmp); + return ret; +} + +//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; ipts_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; +} + +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 = s->oformat->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; + pkt->destruct = NULL; // do not free original but only the copy + 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 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 (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, ff_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; + } +} + +#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; +} -- cgit v1.2.3