ffmpeg/fftools/ffmpeg_mux.c

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/*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdatomic.h>
#include <stdio.h>
#include <string.h>
#include "ffmpeg.h"
#include "ffmpeg_mux.h"
#include "ffmpeg_utils.h"
fftools/ffmpeg: rework -shortest implementation The -shortest option (which finishes the output file at the time the shortest stream ends) is currently implemented by faking the -t option when an output stream ends. This approach is fragile, since it depends on the frames/packets being processed in a specific order. E.g. there are currently some situations in which the output file length will depend unpredictably on unrelated factors like encoder delay. More importantly, the present work aiming at splitting various ffmpeg components into different threads will make this approach completely unworkable, since the frames/packets will arrive in effectively random order. This commit introduces a "sync queue", which is essentially a collection of FIFOs, one per stream. Frames/packets are submitted to these FIFOs and are then released for further processing (encoding or muxing) when it is ensured that the frame in question will not cause its stream to get ahead of the other streams (the logic is similar to libavformat's interleaving queue). These sync queues are then used for encoding and/or muxing when the -shortest option is specified. A new option – -shortest_buf_duration – controls the maximum number of queued packets, to avoid runaway memory usage. This commit changes the results of the following tests: - copy-shortest[12]: the last audio frame is now gone. This is correct, since it actually outlasts the last video frame. - shortest-sub: the video packets following the last subtitle packet are now gone. This is also correct.
2022-06-10 12:21:42 +00:00
#include "sync_queue.h"
#include "libavutil/avstring.h"
#include "libavutil/fifo.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "libavutil/mem.h"
#include "libavutil/time.h"
#include "libavutil/timestamp.h"
#include "libavcodec/packet.h"
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
typedef struct MuxThreadContext {
AVPacket *pkt;
AVPacket *fix_sub_duration_pkt;
} MuxThreadContext;
static Muxer *mux_from_of(OutputFile *of)
{
return (Muxer*)of;
}
static int64_t filesize(AVIOContext *pb)
{
int64_t ret = -1;
if (pb) {
ret = avio_size(pb);
if (ret <= 0) // FIXME improve avio_size() so it works with non seekable output too
ret = avio_tell(pb);
}
return ret;
}
static void mux_log_debug_ts(OutputStream *ost, const AVPacket *pkt)
{
static const char *desc[] = {
[LATENCY_PROBE_DEMUX] = "demux",
[LATENCY_PROBE_DEC_PRE] = "decode",
[LATENCY_PROBE_DEC_POST] = "decode",
[LATENCY_PROBE_FILTER_PRE] = "filter",
[LATENCY_PROBE_FILTER_POST] = "filter",
[LATENCY_PROBE_ENC_PRE] = "encode",
[LATENCY_PROBE_ENC_POST] = "encode",
[LATENCY_PROBE_NB] = "mux",
};
char latency[512];
*latency = 0;
if (pkt->opaque_ref) {
const FrameData *fd = (FrameData*)pkt->opaque_ref->data;
int64_t now = av_gettime_relative();
int64_t total = INT64_MIN;
int next;
for (unsigned i = 0; i < FF_ARRAY_ELEMS(fd->wallclock); i = next) {
int64_t val = fd->wallclock[i];
next = i + 1;
if (val == INT64_MIN)
continue;
if (total == INT64_MIN) {
total = now - val;
snprintf(latency, sizeof(latency), "total:%gms", total / 1e3);
}
// find the next valid entry
for (; next <= FF_ARRAY_ELEMS(fd->wallclock); next++) {
int64_t val_next = (next == FF_ARRAY_ELEMS(fd->wallclock)) ?
now : fd->wallclock[next];
int64_t diff;
if (val_next == INT64_MIN)
continue;
diff = val_next - val;
// print those stages that take at least 5% of total
if (100. * diff > 5. * total) {
av_strlcat(latency, ", ", sizeof(latency));
if (!strcmp(desc[i], desc[next]))
av_strlcat(latency, desc[i], sizeof(latency));
else
av_strlcatf(latency, sizeof(latency), "%s-%s:",
desc[i], desc[next]);
av_strlcatf(latency, sizeof(latency), " %gms/%d%%",
diff / 1e3, (int)(100. * diff / total));
}
break;
}
}
}
av_log(ost, AV_LOG_INFO, "muxer <- pts:%s pts_time:%s dts:%s dts_time:%s "
"duration:%s duration_time:%s size:%d latency(%s)\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, &ost->st->time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, &ost->st->time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, &ost->st->time_base),
pkt->size, *latency ? latency : "N/A");
}
static int mux_fixup_ts(Muxer *mux, MuxStream *ms, AVPacket *pkt)
{
OutputStream *ost = &ms->ost;
#if FFMPEG_OPT_VSYNC_DROP
if (ost->type == AVMEDIA_TYPE_VIDEO && ost->vsync_method == VSYNC_DROP)
pkt->pts = pkt->dts = AV_NOPTS_VALUE;
#endif
// rescale timestamps to the stream timebase
if (ost->type == AVMEDIA_TYPE_AUDIO && !ost->enc) {
// use av_rescale_delta() for streamcopying audio, to preserve
// accuracy with coarse input timebases
int duration = av_get_audio_frame_duration2(ost->st->codecpar, pkt->size);
if (!duration)
duration = ost->st->codecpar->frame_size;
pkt->dts = av_rescale_delta(pkt->time_base, pkt->dts,
(AVRational){1, ost->st->codecpar->sample_rate}, duration,
&ms->ts_rescale_delta_last, ost->st->time_base);
pkt->pts = pkt->dts;
pkt->duration = av_rescale_q(pkt->duration, pkt->time_base, ost->st->time_base);
} else
av_packet_rescale_ts(pkt, pkt->time_base, ost->st->time_base);
pkt->time_base = ost->st->time_base;
if (!(mux->fc->oformat->flags & AVFMT_NOTIMESTAMPS)) {
if (pkt->dts != AV_NOPTS_VALUE &&
pkt->pts != AV_NOPTS_VALUE &&
pkt->dts > pkt->pts) {
av_log(ost, AV_LOG_WARNING, "Invalid DTS: %"PRId64" PTS: %"PRId64", replacing by guess\n",
pkt->dts, pkt->pts);
pkt->pts =
pkt->dts = pkt->pts + pkt->dts + ms->last_mux_dts + 1
- FFMIN3(pkt->pts, pkt->dts, ms->last_mux_dts + 1)
- FFMAX3(pkt->pts, pkt->dts, ms->last_mux_dts + 1);
}
if ((ost->type == AVMEDIA_TYPE_AUDIO || ost->type == AVMEDIA_TYPE_VIDEO || ost->type == AVMEDIA_TYPE_SUBTITLE) &&
pkt->dts != AV_NOPTS_VALUE &&
ms->last_mux_dts != AV_NOPTS_VALUE) {
int64_t max = ms->last_mux_dts + !(mux->fc->oformat->flags & AVFMT_TS_NONSTRICT);
if (pkt->dts < max) {
int loglevel = max - pkt->dts > 2 || ost->type == AVMEDIA_TYPE_VIDEO ? AV_LOG_WARNING : AV_LOG_DEBUG;
if (exit_on_error)
loglevel = AV_LOG_ERROR;
av_log(ost, loglevel, "Non-monotonic DTS; "
"previous: %"PRId64", current: %"PRId64"; ",
ms->last_mux_dts, pkt->dts);
if (exit_on_error) {
return AVERROR(EINVAL);
}
av_log(ost, loglevel, "changing to %"PRId64". This may result "
"in incorrect timestamps in the output file.\n",
max);
if (pkt->pts >= pkt->dts)
pkt->pts = FFMAX(pkt->pts, max);
pkt->dts = max;
}
}
}
ms->last_mux_dts = pkt->dts;
if (debug_ts)
mux_log_debug_ts(ost, pkt);
return 0;
}
static int write_packet(Muxer *mux, OutputStream *ost, AVPacket *pkt)
{
MuxStream *ms = ms_from_ost(ost);
AVFormatContext *s = mux->fc;
int64_t fs;
uint64_t frame_num;
int ret;
fs = filesize(s->pb);
atomic_store(&mux->last_filesize, fs);
if (fs >= mux->limit_filesize) {
ret = AVERROR_EOF;
goto fail;
}
ret = mux_fixup_ts(mux, ms, pkt);
if (ret < 0)
goto fail;
ms->data_size_mux += pkt->size;
frame_num = atomic_fetch_add(&ost->packets_written, 1);
pkt->stream_index = ost->index;
if (ms->stats.io)
enc_stats_write(ost, &ms->stats, NULL, pkt, frame_num);
ret = av_interleaved_write_frame(s, pkt);
if (ret < 0) {
av_log(ost, AV_LOG_ERROR,
"Error submitting a packet to the muxer: %s\n",
av_err2str(ret));
goto fail;
}
return 0;
fail:
av_packet_unref(pkt);
return ret;
}
static int sync_queue_process(Muxer *mux, MuxStream *ms, AVPacket *pkt, int *stream_eof)
fftools/ffmpeg: rework -shortest implementation The -shortest option (which finishes the output file at the time the shortest stream ends) is currently implemented by faking the -t option when an output stream ends. This approach is fragile, since it depends on the frames/packets being processed in a specific order. E.g. there are currently some situations in which the output file length will depend unpredictably on unrelated factors like encoder delay. More importantly, the present work aiming at splitting various ffmpeg components into different threads will make this approach completely unworkable, since the frames/packets will arrive in effectively random order. This commit introduces a "sync queue", which is essentially a collection of FIFOs, one per stream. Frames/packets are submitted to these FIFOs and are then released for further processing (encoding or muxing) when it is ensured that the frame in question will not cause its stream to get ahead of the other streams (the logic is similar to libavformat's interleaving queue). These sync queues are then used for encoding and/or muxing when the -shortest option is specified. A new option – -shortest_buf_duration – controls the maximum number of queued packets, to avoid runaway memory usage. This commit changes the results of the following tests: - copy-shortest[12]: the last audio frame is now gone. This is correct, since it actually outlasts the last video frame. - shortest-sub: the video packets following the last subtitle packet are now gone. This is also correct.
2022-06-10 12:21:42 +00:00
{
OutputFile *of = &mux->of;
if (ms->sq_idx_mux >= 0) {
int ret = sq_send(mux->sq_mux, ms->sq_idx_mux, SQPKT(pkt));
if (ret < 0) {
if (ret == AVERROR_EOF)
*stream_eof = 1;
return ret;
}
fftools/ffmpeg: rework -shortest implementation The -shortest option (which finishes the output file at the time the shortest stream ends) is currently implemented by faking the -t option when an output stream ends. This approach is fragile, since it depends on the frames/packets being processed in a specific order. E.g. there are currently some situations in which the output file length will depend unpredictably on unrelated factors like encoder delay. More importantly, the present work aiming at splitting various ffmpeg components into different threads will make this approach completely unworkable, since the frames/packets will arrive in effectively random order. This commit introduces a "sync queue", which is essentially a collection of FIFOs, one per stream. Frames/packets are submitted to these FIFOs and are then released for further processing (encoding or muxing) when it is ensured that the frame in question will not cause its stream to get ahead of the other streams (the logic is similar to libavformat's interleaving queue). These sync queues are then used for encoding and/or muxing when the -shortest option is specified. A new option – -shortest_buf_duration – controls the maximum number of queued packets, to avoid runaway memory usage. This commit changes the results of the following tests: - copy-shortest[12]: the last audio frame is now gone. This is correct, since it actually outlasts the last video frame. - shortest-sub: the video packets following the last subtitle packet are now gone. This is also correct.
2022-06-10 12:21:42 +00:00
while (1) {
ret = sq_receive(mux->sq_mux, -1, SQPKT(mux->sq_pkt));
if (ret < 0) {
/* n.b.: We forward EOF from the sync queue, terminating muxing.
* This assumes that if a muxing sync queue is present, then all
* the streams use it. That is true currently, but may change in
* the future, then this code needs to be revisited.
*/
return ret == AVERROR(EAGAIN) ? 0 : ret;
}
fftools/ffmpeg: rework -shortest implementation The -shortest option (which finishes the output file at the time the shortest stream ends) is currently implemented by faking the -t option when an output stream ends. This approach is fragile, since it depends on the frames/packets being processed in a specific order. E.g. there are currently some situations in which the output file length will depend unpredictably on unrelated factors like encoder delay. More importantly, the present work aiming at splitting various ffmpeg components into different threads will make this approach completely unworkable, since the frames/packets will arrive in effectively random order. This commit introduces a "sync queue", which is essentially a collection of FIFOs, one per stream. Frames/packets are submitted to these FIFOs and are then released for further processing (encoding or muxing) when it is ensured that the frame in question will not cause its stream to get ahead of the other streams (the logic is similar to libavformat's interleaving queue). These sync queues are then used for encoding and/or muxing when the -shortest option is specified. A new option – -shortest_buf_duration – controls the maximum number of queued packets, to avoid runaway memory usage. This commit changes the results of the following tests: - copy-shortest[12]: the last audio frame is now gone. This is correct, since it actually outlasts the last video frame. - shortest-sub: the video packets following the last subtitle packet are now gone. This is also correct.
2022-06-10 12:21:42 +00:00
ret = write_packet(mux, of->streams[ret],
mux->sq_pkt);
if (ret < 0)
return ret;
fftools/ffmpeg: rework -shortest implementation The -shortest option (which finishes the output file at the time the shortest stream ends) is currently implemented by faking the -t option when an output stream ends. This approach is fragile, since it depends on the frames/packets being processed in a specific order. E.g. there are currently some situations in which the output file length will depend unpredictably on unrelated factors like encoder delay. More importantly, the present work aiming at splitting various ffmpeg components into different threads will make this approach completely unworkable, since the frames/packets will arrive in effectively random order. This commit introduces a "sync queue", which is essentially a collection of FIFOs, one per stream. Frames/packets are submitted to these FIFOs and are then released for further processing (encoding or muxing) when it is ensured that the frame in question will not cause its stream to get ahead of the other streams (the logic is similar to libavformat's interleaving queue). These sync queues are then used for encoding and/or muxing when the -shortest option is specified. A new option – -shortest_buf_duration – controls the maximum number of queued packets, to avoid runaway memory usage. This commit changes the results of the following tests: - copy-shortest[12]: the last audio frame is now gone. This is correct, since it actually outlasts the last video frame. - shortest-sub: the video packets following the last subtitle packet are now gone. This is also correct.
2022-06-10 12:21:42 +00:00
}
} else if (pkt)
return write_packet(mux, &ms->ost, pkt);
return 0;
fftools/ffmpeg: rework -shortest implementation The -shortest option (which finishes the output file at the time the shortest stream ends) is currently implemented by faking the -t option when an output stream ends. This approach is fragile, since it depends on the frames/packets being processed in a specific order. E.g. there are currently some situations in which the output file length will depend unpredictably on unrelated factors like encoder delay. More importantly, the present work aiming at splitting various ffmpeg components into different threads will make this approach completely unworkable, since the frames/packets will arrive in effectively random order. This commit introduces a "sync queue", which is essentially a collection of FIFOs, one per stream. Frames/packets are submitted to these FIFOs and are then released for further processing (encoding or muxing) when it is ensured that the frame in question will not cause its stream to get ahead of the other streams (the logic is similar to libavformat's interleaving queue). These sync queues are then used for encoding and/or muxing when the -shortest option is specified. A new option – -shortest_buf_duration – controls the maximum number of queued packets, to avoid runaway memory usage. This commit changes the results of the following tests: - copy-shortest[12]: the last audio frame is now gone. This is correct, since it actually outlasts the last video frame. - shortest-sub: the video packets following the last subtitle packet are now gone. This is also correct.
2022-06-10 12:21:42 +00:00
}
static int of_streamcopy(OutputFile *of, OutputStream *ost, AVPacket *pkt);
/* apply the output bitstream filters */
static int mux_packet_filter(Muxer *mux, MuxThreadContext *mt,
OutputStream *ost, AVPacket *pkt, int *stream_eof)
{
MuxStream *ms = ms_from_ost(ost);
const char *err_msg;
int ret = 0;
if (pkt && !ost->enc) {
ret = of_streamcopy(&mux->of, ost, pkt);
if (ret == AVERROR(EAGAIN))
return 0;
else if (ret == AVERROR_EOF) {
av_packet_unref(pkt);
pkt = NULL;
ret = 0;
*stream_eof = 1;
} else if (ret < 0)
goto fail;
}
// emit heartbeat for -fix_sub_duration;
// we are only interested in heartbeats on on random access points.
if (pkt && (pkt->flags & AV_PKT_FLAG_KEY)) {
mt->fix_sub_duration_pkt->opaque = (void*)(intptr_t)PKT_OPAQUE_FIX_SUB_DURATION;
mt->fix_sub_duration_pkt->pts = pkt->pts;
mt->fix_sub_duration_pkt->time_base = pkt->time_base;
ret = sch_mux_sub_heartbeat(mux->sch, mux->sch_idx, ms->sch_idx,
mt->fix_sub_duration_pkt);
if (ret < 0)
goto fail;
}
if (ms->bsf_ctx) {
int bsf_eof = 0;
if (pkt)
av_packet_rescale_ts(pkt, pkt->time_base, ms->bsf_ctx->time_base_in);
ret = av_bsf_send_packet(ms->bsf_ctx, pkt);
if (ret < 0) {
err_msg = "submitting a packet for bitstream filtering";
goto fail;
}
while (!bsf_eof) {
ret = av_bsf_receive_packet(ms->bsf_ctx, ms->bsf_pkt);
if (ret == AVERROR(EAGAIN))
return 0;
else if (ret == AVERROR_EOF)
bsf_eof = 1;
else if (ret < 0) {
av_log(ost, AV_LOG_ERROR,
"Error applying bitstream filters to a packet: %s",
av_err2str(ret));
if (exit_on_error)
return ret;
continue;
}
if (!bsf_eof)
ms->bsf_pkt->time_base = ms->bsf_ctx->time_base_out;
ret = sync_queue_process(mux, ms, bsf_eof ? NULL : ms->bsf_pkt, stream_eof);
if (ret < 0)
goto mux_fail;
}
*stream_eof = 1;
} else {
ret = sync_queue_process(mux, ms, pkt, stream_eof);
if (ret < 0)
goto mux_fail;
}
return *stream_eof ? AVERROR_EOF : 0;
mux_fail:
err_msg = "submitting a packet to the muxer";
fail:
if (ret != AVERROR_EOF)
av_log(ost, AV_LOG_ERROR, "Error %s: %s\n", err_msg, av_err2str(ret));
return ret;
}
2022-10-18 16:50:30 +00:00
static void thread_set_name(OutputFile *of)
{
char name[16];
snprintf(name, sizeof(name), "mux%d:%s", of->index, of->format->name);
ff_thread_setname(name);
}
static void mux_thread_uninit(MuxThreadContext *mt)
{
av_packet_free(&mt->pkt);
av_packet_free(&mt->fix_sub_duration_pkt);
memset(mt, 0, sizeof(*mt));
}
static int mux_thread_init(MuxThreadContext *mt)
{
memset(mt, 0, sizeof(*mt));
mt->pkt = av_packet_alloc();
if (!mt->pkt)
goto fail;
mt->fix_sub_duration_pkt = av_packet_alloc();
if (!mt->fix_sub_duration_pkt)
goto fail;
return 0;
fail:
mux_thread_uninit(mt);
return AVERROR(ENOMEM);
}
int muxer_thread(void *arg)
{
Muxer *mux = arg;
OutputFile *of = &mux->of;
MuxThreadContext mt;
int ret = 0;
ret = mux_thread_init(&mt);
if (ret < 0)
goto finish;
2022-10-18 16:50:30 +00:00
thread_set_name(of);
while (1) {
OutputStream *ost;
int stream_idx, stream_eof = 0;
ret = sch_mux_receive(mux->sch, of->index, mt.pkt);
stream_idx = mt.pkt->stream_index;
if (stream_idx < 0) {
av_log(mux, AV_LOG_VERBOSE, "All streams finished\n");
ret = 0;
break;
}
ost = of->streams[mux->sch_stream_idx[stream_idx]];
mt.pkt->stream_index = ost->index;
mt.pkt->flags &= ~AV_PKT_FLAG_TRUSTED;
ret = mux_packet_filter(mux, &mt, ost, ret < 0 ? NULL : mt.pkt, &stream_eof);
av_packet_unref(mt.pkt);
if (ret == AVERROR_EOF) {
if (stream_eof) {
sch_mux_receive_finish(mux->sch, of->index, stream_idx);
} else {
av_log(mux, AV_LOG_VERBOSE, "Muxer returned EOF\n");
ret = 0;
break;
}
} else if (ret < 0) {
av_log(mux, AV_LOG_ERROR, "Error muxing a packet\n");
break;
}
}
finish:
mux_thread_uninit(&mt);
return ret;
}
static int of_streamcopy(OutputFile *of, OutputStream *ost, AVPacket *pkt)
{
MuxStream *ms = ms_from_ost(ost);
FrameData *fd = pkt->opaque_ref ? (FrameData*)pkt->opaque_ref->data : NULL;
int64_t dts = fd ? fd->dts_est : AV_NOPTS_VALUE;
int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
int64_t ts_offset;
if (of->recording_time != INT64_MAX &&
dts >= of->recording_time + start_time)
return AVERROR_EOF;
if (!ms->streamcopy_started && !(pkt->flags & AV_PKT_FLAG_KEY) &&
!ms->copy_initial_nonkeyframes)
return AVERROR(EAGAIN);
if (!ms->streamcopy_started) {
if (!ms->copy_prior_start &&
(pkt->pts == AV_NOPTS_VALUE ?
dts < ms->ts_copy_start :
pkt->pts < av_rescale_q(ms->ts_copy_start, AV_TIME_BASE_Q, pkt->time_base)))
return AVERROR(EAGAIN);
if (of->start_time != AV_NOPTS_VALUE && dts < of->start_time)
return AVERROR(EAGAIN);
}
ts_offset = av_rescale_q(start_time, AV_TIME_BASE_Q, pkt->time_base);
if (pkt->pts != AV_NOPTS_VALUE)
pkt->pts -= ts_offset;
if (pkt->dts == AV_NOPTS_VALUE) {
pkt->dts = av_rescale_q(dts, AV_TIME_BASE_Q, pkt->time_base);
} else if (ost->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
pkt->pts = pkt->dts - ts_offset;
}
pkt->dts -= ts_offset;
ms->streamcopy_started = 1;
return 0;
}
int print_sdp(const char *filename);
int print_sdp(const char *filename)
{
char sdp[16384];
int j = 0, ret;
AVIOContext *sdp_pb;
AVFormatContext **avc;
avc = av_malloc_array(nb_output_files, sizeof(*avc));
if (!avc)
return AVERROR(ENOMEM);
for (int i = 0; i < nb_output_files; i++) {
if (!strcmp(output_files[i]->format->name, "rtp")) {
avc[j] = mux_from_of(output_files[i])->fc;
j++;
}
}
if (!j) {
av_log(NULL, AV_LOG_ERROR, "No output streams in the SDP.\n");
ret = AVERROR(EINVAL);
goto fail;
}
ret = av_sdp_create(avc, j, sdp, sizeof(sdp));
if (ret < 0)
goto fail;
if (!filename) {
printf("SDP:\n%s\n", sdp);
fflush(stdout);
} else {
ret = avio_open2(&sdp_pb, filename, AVIO_FLAG_WRITE, &int_cb, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open sdp file '%s'\n", filename);
goto fail;
}
avio_print(sdp_pb, sdp);
avio_closep(&sdp_pb);
}
fail:
av_freep(&avc);
return ret;
}
int mux_check_init(void *arg)
{
Muxer *mux = arg;
OutputFile *of = &mux->of;
AVFormatContext *fc = mux->fc;
int ret;
ret = avformat_write_header(fc, &mux->opts);
if (ret < 0) {
av_log(mux, AV_LOG_ERROR, "Could not write header (incorrect codec "
"parameters ?): %s\n", av_err2str(ret));
return ret;
}
//assert_avoptions(of->opts);
mux->header_written = 1;
av_dump_format(fc, of->index, fc->url, 1);
atomic_fetch_add(&nb_output_dumped, 1);
return 0;
}
static int bsf_init(MuxStream *ms)
{
OutputStream *ost = &ms->ost;
AVBSFContext *ctx = ms->bsf_ctx;
int ret;
if (!ctx)
return avcodec_parameters_copy(ost->st->codecpar, ost->par_in);
ret = avcodec_parameters_copy(ctx->par_in, ost->par_in);
if (ret < 0)
return ret;
ctx->time_base_in = ost->st->time_base;
ret = av_bsf_init(ctx);
if (ret < 0) {
av_log(ms, AV_LOG_ERROR, "Error initializing bitstream filter: %s\n",
ctx->filter->name);
return ret;
}
ret = avcodec_parameters_copy(ost->st->codecpar, ctx->par_out);
if (ret < 0)
return ret;
ost->st->time_base = ctx->time_base_out;
ms->bsf_pkt = av_packet_alloc();
if (!ms->bsf_pkt)
return AVERROR(ENOMEM);
return 0;
}
int of_stream_init(OutputFile *of, OutputStream *ost)
{
Muxer *mux = mux_from_of(of);
MuxStream *ms = ms_from_ost(ost);
int ret;
/* initialize bitstream filters for the output stream
* needs to be done here, because the codec id for streamcopy is not
* known until now */
ret = bsf_init(ms);
if (ret < 0)
return ret;
if (ms->stream_duration) {
ost->st->duration = av_rescale_q(ms->stream_duration, ms->stream_duration_tb,
ost->st->time_base);
}
if (ms->sch_idx >= 0)
return sch_mux_stream_ready(mux->sch, of->index, ms->sch_idx);
return 0;
}
static int check_written(OutputFile *of)
{
int64_t total_packets_written = 0;
int pass1_used = 1;
int ret = 0;
for (int i = 0; i < of->nb_streams; i++) {
OutputStream *ost = of->streams[i];
uint64_t packets_written = atomic_load(&ost->packets_written);
total_packets_written += packets_written;
if (ost->enc_ctx &&
(ost->enc_ctx->flags & (AV_CODEC_FLAG_PASS1 | AV_CODEC_FLAG_PASS2))
!= AV_CODEC_FLAG_PASS1)
pass1_used = 0;
if (!packets_written &&
(abort_on_flags & ABORT_ON_FLAG_EMPTY_OUTPUT_STREAM)) {
av_log(ost, AV_LOG_FATAL, "Empty output stream\n");
ret = err_merge(ret, AVERROR(EINVAL));
}
}
if (!total_packets_written) {
int level = AV_LOG_WARNING;
if (abort_on_flags & ABORT_ON_FLAG_EMPTY_OUTPUT) {
ret = err_merge(ret, AVERROR(EINVAL));
level = AV_LOG_FATAL;
}
av_log(of, level, "Output file is empty, nothing was encoded%s\n",
pass1_used ? "" : "(check -ss / -t / -frames parameters if used)");
}
return ret;
}
static void mux_final_stats(Muxer *mux)
{
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OutputFile *of = &mux->of;
uint64_t total_packets = 0, total_size = 0;
uint64_t video_size = 0, audio_size = 0, subtitle_size = 0,
extra_size = 0, other_size = 0;
uint8_t overhead[16] = "unknown";
int64_t file_size = of_filesize(of);
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av_log(of, AV_LOG_VERBOSE, "Output file #%d (%s):\n",
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of->index, of->url);
for (int j = 0; j < of->nb_streams; j++) {
OutputStream *ost = of->streams[j];
MuxStream *ms = ms_from_ost(ost);
const AVCodecParameters *par = ost->st->codecpar;
const enum AVMediaType type = par->codec_type;
const uint64_t s = ms->data_size_mux;
switch (type) {
case AVMEDIA_TYPE_VIDEO: video_size += s; break;
case AVMEDIA_TYPE_AUDIO: audio_size += s; break;
case AVMEDIA_TYPE_SUBTITLE: subtitle_size += s; break;
default: other_size += s; break;
}
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extra_size += par->extradata_size;
total_size += s;
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total_packets += atomic_load(&ost->packets_written);
av_log(of, AV_LOG_VERBOSE, " Output stream #%d:%d (%s): ",
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of->index, j, av_get_media_type_string(type));
if (ost->enc) {
av_log(of, AV_LOG_VERBOSE, "%"PRIu64" frames encoded",
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ost->frames_encoded);
if (type == AVMEDIA_TYPE_AUDIO)
av_log(of, AV_LOG_VERBOSE, " (%"PRIu64" samples)", ost->samples_encoded);
av_log(of, AV_LOG_VERBOSE, "; ");
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}
av_log(of, AV_LOG_VERBOSE, "%"PRIu64" packets muxed (%"PRIu64" bytes); ",
atomic_load(&ost->packets_written), s);
av_log(of, AV_LOG_VERBOSE, "\n");
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}
av_log(of, AV_LOG_VERBOSE, " Total: %"PRIu64" packets (%"PRIu64" bytes) muxed\n",
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total_packets, total_size);
if (total_size && file_size > 0 && file_size >= total_size) {
snprintf(overhead, sizeof(overhead), "%f%%",
100.0 * (file_size - total_size) / total_size);
}
av_log(of, AV_LOG_INFO,
"video:%1.0fKiB audio:%1.0fKiB subtitle:%1.0fKiB other streams:%1.0fKiB "
"global headers:%1.0fKiB muxing overhead: %s\n",
video_size / 1024.0,
audio_size / 1024.0,
subtitle_size / 1024.0,
other_size / 1024.0,
extra_size / 1024.0,
overhead);
}
int of_write_trailer(OutputFile *of)
{
Muxer *mux = mux_from_of(of);
AVFormatContext *fc = mux->fc;
int ret, mux_result = 0;
if (!mux->header_written) {
av_log(mux, AV_LOG_ERROR,
"Nothing was written into output file, because "
"at least one of its streams received no packets.\n");
return AVERROR(EINVAL);
}
ret = av_write_trailer(fc);
if (ret < 0) {
av_log(mux, AV_LOG_ERROR, "Error writing trailer: %s\n", av_err2str(ret));
mux_result = err_merge(mux_result, ret);
}
mux->last_filesize = filesize(fc->pb);
if (!(of->format->flags & AVFMT_NOFILE)) {
ret = avio_closep(&fc->pb);
if (ret < 0) {
av_log(mux, AV_LOG_ERROR, "Error closing file: %s\n", av_err2str(ret));
mux_result = err_merge(mux_result, ret);
}
}
mux_final_stats(mux);
// check whether anything was actually written
ret = check_written(of);
mux_result = err_merge(mux_result, ret);
return mux_result;
}
static void enc_stats_uninit(EncStats *es)
{
for (int i = 0; i < es->nb_components; i++)
av_freep(&es->components[i].str);
av_freep(&es->components);
if (es->lock_initialized)
pthread_mutex_destroy(&es->lock);
es->lock_initialized = 0;
}
static void ost_free(OutputStream **post)
{
OutputStream *ost = *post;
MuxStream *ms;
if (!ost)
return;
ms = ms_from_ost(ost);
enc_free(&ost->enc);
if (ost->logfile) {
if (fclose(ost->logfile))
av_log(ms, AV_LOG_ERROR,
"Error closing logfile, loss of information possible: %s\n",
av_err2str(AVERROR(errno)));
ost->logfile = NULL;
}
avcodec_parameters_free(&ost->par_in);
av_bsf_free(&ms->bsf_ctx);
av_packet_free(&ms->bsf_pkt);
av_packet_free(&ms->pkt);
av_dict_free(&ost->encoder_opts);
av_freep(&ost->kf.pts);
av_expr_free(ost->kf.pexpr);
av_freep(&ost->logfile_prefix);
av_freep(&ost->apad);
av_freep(&ost->attachment_filename);
av_dict_free(&ost->sws_dict);
av_dict_free(&ost->swr_opts);
if (ost->enc_ctx)
av_freep(&ost->enc_ctx->stats_in);
avcodec_free_context(&ost->enc_ctx);
enc_stats_uninit(&ost->enc_stats_pre);
enc_stats_uninit(&ost->enc_stats_post);
enc_stats_uninit(&ms->stats);
av_freep(post);
}
static void fc_close(AVFormatContext **pfc)
{
AVFormatContext *fc = *pfc;
if (!fc)
return;
if (!(fc->oformat->flags & AVFMT_NOFILE))
avio_closep(&fc->pb);
avformat_free_context(fc);
*pfc = NULL;
}
void of_free(OutputFile **pof)
{
OutputFile *of = *pof;
Muxer *mux;
if (!of)
return;
mux = mux_from_of(of);
sq_free(&mux->sq_mux);
for (int i = 0; i < of->nb_streams; i++)
ost_free(&of->streams[i]);
av_freep(&of->streams);
av_freep(&mux->sch_stream_idx);
av_dict_free(&mux->opts);
fftools/ffmpeg: rework -shortest implementation The -shortest option (which finishes the output file at the time the shortest stream ends) is currently implemented by faking the -t option when an output stream ends. This approach is fragile, since it depends on the frames/packets being processed in a specific order. E.g. there are currently some situations in which the output file length will depend unpredictably on unrelated factors like encoder delay. More importantly, the present work aiming at splitting various ffmpeg components into different threads will make this approach completely unworkable, since the frames/packets will arrive in effectively random order. This commit introduces a "sync queue", which is essentially a collection of FIFOs, one per stream. Frames/packets are submitted to these FIFOs and are then released for further processing (encoding or muxing) when it is ensured that the frame in question will not cause its stream to get ahead of the other streams (the logic is similar to libavformat's interleaving queue). These sync queues are then used for encoding and/or muxing when the -shortest option is specified. A new option – -shortest_buf_duration – controls the maximum number of queued packets, to avoid runaway memory usage. This commit changes the results of the following tests: - copy-shortest[12]: the last audio frame is now gone. This is correct, since it actually outlasts the last video frame. - shortest-sub: the video packets following the last subtitle packet are now gone. This is also correct.
2022-06-10 12:21:42 +00:00
av_packet_free(&mux->sq_pkt);
fc_close(&mux->fc);
av_freep(pof);
}
int64_t of_filesize(OutputFile *of)
{
Muxer *mux = mux_from_of(of);
return atomic_load(&mux->last_filesize);
}