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authorLuca Barbato <lu_zero@gentoo.org>2012-10-01 00:49:16 +0200
committerLuca Barbato <lu_zero@gentoo.org>2012-10-01 19:57:57 +0200
commit55f9037f38bc3beacb2f5a17408c1d24c077d7fd (patch)
tree8b50aa80ebc148470c35741fc95c004ca5f81ce2 /libavformat/mux.c
parentbfcd4b6a1691d20aebc6d2308424c2a88334a9f0 (diff)
avformat: split muxing functions from util.c
Diffstat (limited to 'libavformat/mux.c')
-rw-r--r--libavformat/mux.c564
1 files changed, 564 insertions, 0 deletions
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 <stdarg.h>
+#if CONFIG_NETWORK
+#include "network.h"
+#endif
+
+#undef NDEBUG
+#include <assert.h>
+
+/**
+ * @file
+ * muxing functions for use within Libav
+ */
+
+/* fraction handling */
+
+/**
+ * f = val + (num / den) + 0.5.
+ *
+ * 'num' is normalized so that it is such as 0 <= num < den.
+ *
+ * @param f fractional number
+ * @param val integer value
+ * @param num must be >= 0
+ * @param den must be >= 1
+ */
+static void frac_init(AVFrac *f, int64_t val, int64_t num, int64_t den)
+{
+ num += (den >> 1);
+ if (num >= den) {
+ val += num / den;
+ num = num % den;
+ }
+ f->val = val;
+ f->num = num;
+ f->den = den;
+}
+
+/**
+ * Fractional addition to f: f = f + (incr / f->den).
+ *
+ * @param f fractional number
+ * @param incr increment, can be positive or negative
+ */
+static void frac_add(AVFrac *f, int64_t incr)
+{
+ int64_t num, den;
+
+ num = f->num + incr;
+ den = f->den;
+ if (num < 0) {
+ f->val += num / den;
+ num = num % den;
+ if (num < 0) {
+ num += den;
+ f->val--;
+ }
+ } else if (num >= den) {
+ f->val += num / den;
+ num = num % den;
+ }
+ f->num = num;
+}
+
+static int validate_codec_tag(AVFormatContext *s, AVStream *st)
+{
+ const AVCodecTag *avctag;
+ int n;
+ enum AVCodecID id = AV_CODEC_ID_NONE;
+ unsigned int tag = 0;
+
+ /**
+ * Check that tag + id is in the table
+ * If neither is in the table -> OK
+ * If tag is in the table with another id -> FAIL
+ * If id is in the table with another tag -> FAIL unless strict < normal
+ */
+ for (n = 0; s->oformat->codec_tag[n]; n++) {
+ avctag = s->oformat->codec_tag[n];
+ while (avctag->id != AV_CODEC_ID_NONE) {
+ if (avpriv_toupper4(avctag->tag) == avpriv_toupper4(st->codec->codec_tag)) {
+ id = avctag->id;
+ if (id == st->codec->codec_id)
+ return 1;
+ }
+ if (avctag->id == st->codec->codec_id)
+ tag = avctag->tag;
+ avctag++;
+ }
+ }
+ if (id != AV_CODEC_ID_NONE)
+ return 0;
+ if (tag && (st->codec->strict_std_compliance >= FF_COMPLIANCE_NORMAL))
+ return 0;
+ return 1;
+}
+
+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; i<delay && st->pts_buffer[i] > st->pts_buffer[i + 1]; i++)
+ FFSWAP(int64_t, st->pts_buffer[i], st->pts_buffer[i + 1]);
+
+ pkt->dts = st->pts_buffer[0];
+ }
+
+ if (st->cur_dts && st->cur_dts != AV_NOPTS_VALUE &&
+ ((!(s->oformat->flags & AVFMT_TS_NONSTRICT) &&
+ st->cur_dts >= pkt->dts) || st->cur_dts > pkt->dts)) {
+ av_log(s, AV_LOG_ERROR,
+ "Application provided invalid, non monotonically increasing dts to muxer in stream %d: %" PRId64 " >= %" PRId64 "\n",
+ st->index, st->cur_dts, pkt->dts);
+ return AVERROR(EINVAL);
+ }
+ if (pkt->dts != AV_NOPTS_VALUE && pkt->pts != AV_NOPTS_VALUE && pkt->pts < pkt->dts) {
+ av_log(s, AV_LOG_ERROR, "pts < dts in stream %d\n", st->index);
+ return AVERROR(EINVAL);
+ }
+
+ av_dlog(s, "av_write_frame: pts2:%"PRId64" dts2:%"PRId64"\n",
+ pkt->pts, pkt->dts);
+ st->cur_dts = pkt->dts;
+ st->pts.val = pkt->dts;
+
+ /* update pts */
+ switch (st->codec->codec_type) {
+ case AVMEDIA_TYPE_AUDIO:
+ frame_size = ff_get_audio_frame_size(st->codec, pkt->size, 1);
+
+ /* HACK/FIXME, we skip the initial 0 size packets as they are most
+ * likely equal to the encoder delay, but it would be better if we
+ * had the real timestamps from the encoder */
+ if (frame_size >= 0 && (pkt->size || st->pts.num != st->pts.den >> 1 || st->pts.val)) {
+ frac_add(&st->pts, (int64_t)st->time_base.den * frame_size);
+ }
+ break;
+ case AVMEDIA_TYPE_VIDEO:
+ frac_add(&st->pts, (int64_t)st->time_base.den * st->codec->time_base.num);
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+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;
+}