October 2020 - ffmpeg-cvslog - lists.ffmpeg.org

avformat/mpegtsenc: check that not only first pts but also first dts is valid
by Limin Wang 30 Oct '20

30 Oct '20

ffmpeg | branch: master | Limin Wang <lance.lmwang(a)gmail.com> | Thu Oct 29 09:24:29 2020 +0800| [be4ff0f1b2eea3d4ab330eb2632bdb1c2fd30717] | committer: Limin Wang avformat/mpegtsenc: check that not only first pts but also first dts is valid Reviewed-by: Marton Balint <cus(a)passwd.hu> Signed-off-by: Limin Wang <lance.lmwang(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=be4ff0f1b2eea3d4a… --- libavformat/mpegtsenc.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/libavformat/mpegtsenc.c b/libavformat/mpegtsenc.c index 383181d2a2..45f8d5f373 100644 --- a/libavformat/mpegtsenc.c +++ b/libavformat/mpegtsenc.c @@ -232,7 +232,7 @@ typedef struct MpegTSWriteStream { int cc; int discontinuity; int payload_size; - int first_pts_checked; ///< first pts check needed + int first_timestamp_checked; ///< first pts/dts check needed int prev_payload_key; int64_t payload_pts; int64_t payload_dts; @@ -1699,11 +1699,11 @@ static int mpegts_write_packet_internal(AVFormatContext *s, AVPacket *pkt) dts += delay; } - if (!ts_st->first_pts_checked && pts == AV_NOPTS_VALUE) { - av_log(s, AV_LOG_ERROR, "first pts value must be set\n"); + if (!ts_st->first_timestamp_checked && (pts == AV_NOPTS_VALUE || dts == AV_NOPTS_VALUE)) { + av_log(s, AV_LOG_ERROR, "first pts and dts value must be set\n"); return AVERROR_INVALIDDATA; } - ts_st->first_pts_checked = 1; + ts_st->first_timestamp_checked = 1; if (st->codecpar->codec_id == AV_CODEC_ID_H264) { const uint8_t *p = buf, *buf_end = p + size;

1 0

avformat/mpegtsenc: first_pts_check -> first_pts_checked
by Limin Wang 30 Oct '20

30 Oct '20

ffmpeg | branch: master | Limin Wang <lance.lmwang(a)gmail.com> | Tue Oct 27 22:26:22 2020 +0800| [27e8a6c44bbe349827990783b7ebd9a44073c4c9] | committer: Limin Wang avformat/mpegtsenc: first_pts_check -> first_pts_checked change to first_pts_checked and reverse the logic. Reviewed-by: Marton Balint <cus(a)passwd.hu> Signed-off-by: Limin Wang <lance.lmwang(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=27e8a6c44bbe34982… --- libavformat/mpegtsenc.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/libavformat/mpegtsenc.c b/libavformat/mpegtsenc.c index 29dcaf6bec..383181d2a2 100644 --- a/libavformat/mpegtsenc.c +++ b/libavformat/mpegtsenc.c @@ -232,7 +232,7 @@ typedef struct MpegTSWriteStream { int cc; int discontinuity; int payload_size; - int first_pts_check; ///< first pts check needed + int first_pts_checked; ///< first pts check needed int prev_payload_key; int64_t payload_pts; int64_t payload_dts; @@ -1101,7 +1101,6 @@ static int mpegts_init(AVFormatContext *s) } ts_st->payload_pts = AV_NOPTS_VALUE; ts_st->payload_dts = AV_NOPTS_VALUE; - ts_st->first_pts_check = 1; ts_st->cc = 15; ts_st->discontinuity = ts->flags & MPEGTS_FLAG_DISCONT; if (st->codecpar->codec_id == AV_CODEC_ID_AAC && @@ -1700,11 +1699,11 @@ static int mpegts_write_packet_internal(AVFormatContext *s, AVPacket *pkt) dts += delay; } - if (ts_st->first_pts_check && pts == AV_NOPTS_VALUE) { + if (!ts_st->first_pts_checked && pts == AV_NOPTS_VALUE) { av_log(s, AV_LOG_ERROR, "first pts value must be set\n"); return AVERROR_INVALIDDATA; } - ts_st->first_pts_check = 0; + ts_st->first_pts_checked = 1; if (st->codecpar->codec_id == AV_CODEC_ID_H264) { const uint8_t *p = buf, *buf_end = p + size;

1 0

avformat/jacosubdec: Fix unintended fallthrough
by Andreas Rheinhardt 30 Oct '20

30 Oct '20

ffmpeg | branch: master | Andreas Rheinhardt <andreas.rheinhardt(a)gmail.com> | Thu Oct 29 13:36:22 2020 +0100| [11ea0ed20e11dd1b19af6e23eceabce7ec76f70d] | committer: Andreas Rheinhardt avformat/jacosubdec: Fix unintended fallthrough Regression since 715ff75e5dbbbefff7337351db596a9b7a5d4379. Fixes Coverity issues #1468654 and #1468656. Reviewed-by: Michael Niedermayer <michael(a)niedermayer.cc> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=11ea0ed20e11dd1b1… --- libavformat/jacosubdec.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/libavformat/jacosubdec.c b/libavformat/jacosubdec.c index e70ceeaafd..14221b166c 100644 --- a/libavformat/jacosubdec.c +++ b/libavformat/jacosubdec.c @@ -148,9 +148,15 @@ static int get_shift(int timeres, const char *buf) ret = 0; switch (n) { - case 4: ret = sign * (((int64_t)a*3600 + b*60 + c) * timeres + d); - case 3: ret = sign * (( (int64_t)a*60 + b) * timeres + c); - case 2: ret = sign * (( (int64_t)a) * timeres + b); + case 4: + ret = sign * (((int64_t)a*3600 + b*60 + c) * timeres + d); + break; + case 3: + ret = sign * (( (int64_t)a*60 + b) * timeres + c); + break; + case 2: + ret = sign * (( (int64_t)a) * timeres + b); + break; } if ((int)ret != ret) ret = 0;

1 0

avcodec/nvenc: all presets are SDK10 presets now
by Timo Rothenpieler 29 Oct '20

29 Oct '20

ffmpeg | branch: master | Timo Rothenpieler <timo(a)rothenpieler.org> | Thu Oct 29 21:59:37 2020 +0100| [bb6edf618a2a2effbe857e1249fda05782a7dbbc] | committer: Timo Rothenpieler avcodec/nvenc: all presets are SDK10 presets now > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=bb6edf618a2a2effb… --- libavcodec/nvenc.c | 42 ++++++++++++++++-------------------------- 1 file changed, 16 insertions(+), 26 deletions(-) diff --git a/libavcodec/nvenc.c b/libavcodec/nvenc.c index 4e081dc393..e8b29fabee 100644 --- a/libavcodec/nvenc.c +++ b/libavcodec/nvenc.c @@ -43,12 +43,6 @@ rc == NV_ENC_PARAMS_RC_CBR_LOWDELAY_HQ || \ rc == NV_ENC_PARAMS_RC_CBR_HQ) -#ifdef NVENC_HAVE_NEW_PRESETS -#define IS_SDK10_PRESET(p) ((p) >= PRESET_P1 && (p) <= PRESET_P7) -#else -#define IS_SDK10_PRESET(p) 0 -#endif - const enum AVPixelFormat ff_nvenc_pix_fmts[] = { AV_PIX_FMT_YUV420P, AV_PIX_FMT_NV12, @@ -1268,30 +1262,26 @@ static av_cold int nvenc_setup_encoder(AVCodecContext *avctx) preset_config.version = NV_ENC_PRESET_CONFIG_VER; preset_config.presetCfg.version = NV_ENC_CONFIG_VER; - if (IS_SDK10_PRESET(ctx->preset)) { #ifdef NVENC_HAVE_NEW_PRESETS - ctx->init_encode_params.tuningInfo = ctx->tuning_info; + ctx->init_encode_params.tuningInfo = ctx->tuning_info; - if (ctx->flags & NVENC_LOWLATENCY) - ctx->init_encode_params.tuningInfo = NV_ENC_TUNING_INFO_LOW_LATENCY; + if (ctx->flags & NVENC_LOWLATENCY) + ctx->init_encode_params.tuningInfo = NV_ENC_TUNING_INFO_LOW_LATENCY; - nv_status = p_nvenc->nvEncGetEncodePresetConfigEx(ctx->nvencoder, - ctx->init_encode_params.encodeGUID, - ctx->init_encode_params.presetGUID, - ctx->init_encode_params.tuningInfo, - &preset_config); -#endif - } else { -#ifdef NVENC_HAVE_NEW_PRESETS - // Turn off tuning info parameter if older presets are on - ctx->init_encode_params.tuningInfo = 0; -#endif + nv_status = p_nvenc->nvEncGetEncodePresetConfigEx(ctx->nvencoder, + ctx->init_encode_params.encodeGUID, + ctx->init_encode_params.presetGUID, + ctx->init_encode_params.tuningInfo, + &preset_config); +#else + // Turn off tuning info parameter if older presets are on + ctx->init_encode_params.tuningInfo = 0; - nv_status = p_nvenc->nvEncGetEncodePresetConfig(ctx->nvencoder, - ctx->init_encode_params.encodeGUID, - ctx->init_encode_params.presetGUID, - &preset_config); - } + nv_status = p_nvenc->nvEncGetEncodePresetConfig(ctx->nvencoder, + ctx->init_encode_params.encodeGUID, + ctx->init_encode_params.presetGUID, + &preset_config); +#endif if (nv_status != NV_ENC_SUCCESS) return nvenc_print_error(avctx, nv_status, "Cannot get the preset configuration");

1 0

avcodec/utils: Check sample rate before use for AV_CODEC_ID_BINKAUDIO_DCT in get_audio_frame_duration()
by Michael Niedermayer 29 Oct '20

29 Oct '20

ffmpeg | branch: master | Michael Niedermayer <michael(a)niedermayer.cc> | Tue Oct 27 17:21:19 2020 +0100| [ec7e0d42884b40ce93b6b5e94de5f7849310f8a0] | committer: Michael Niedermayer avcodec/utils: Check sample rate before use for AV_CODEC_ID_BINKAUDIO_DCT in get_audio_frame_duration() Fixes: shift exponent 95 is too large for 32-bit type 'int' Fixes: 26590/clusterfuzz-testcase-minimized-ffmpeg_dem_SMACKER_fuzzer-5120609937522688 Reviewed-by: Peter Ross <pross(a)xvid.org> Signed-off-by: Michael Niedermayer <michael(a)niedermayer.cc> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ec7e0d42884b40ce9… --- libavcodec/utils.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/libavcodec/utils.c b/libavcodec/utils.c index fb40962c47..110496cc44 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -1633,8 +1633,11 @@ static int get_audio_frame_duration(enum AVCodecID id, int sr, int ch, int ba, if (ch > 0) { /* calc from sample rate and channels */ - if (id == AV_CODEC_ID_BINKAUDIO_DCT) + if (id == AV_CODEC_ID_BINKAUDIO_DCT) { + if (sr / 22050 > 22) + return 0; return (480 << (sr / 22050)) / ch; + } } if (id == AV_CODEC_ID_MP3)

1 0

ffmpeg: move field order decision making to encoder initialization
by Jan Ekström 29 Oct '20

29 Oct '20

ffmpeg | branch: master | Jan Ekström <jeebjp(a)gmail.com> | Sun Sep 20 21:00:52 2020 +0300| [fbb44bc51a647862eb05ae3f9d7d49a0be9bed57] | committer: Jan Ekström ffmpeg: move field order decision making to encoder initialization We now have the possibility of getting AVFrames here, and we should not touch the muxer's codecpar after writing the header. Results of FATE tests change as the MXF and Matroska muxers actually write down the field/frame coding type of a stream in their respective headers. Before this change, these values in codecpar would only be set after the muxer was initialized. Now, the information is also available for encoder and muxer initialization. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=fbb44bc51a647862e… --- fftools/ffmpeg.c | 27 +++++++++++----------- .../ref/fate/concat-demuxer-extended-lavf-mxf_d10 | 2 +- tests/ref/fate/concat-demuxer-simple1-lavf-mxf_d10 | 2 +- tests/ref/fate/rgb24-mkv | 4 ++-- tests/ref/lavf/mxf_d10 | 2 +- 5 files changed, 19 insertions(+), 18 deletions(-) diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c index 20d191bcf6..788cf9c665 100644 --- a/fftools/ffmpeg.c +++ b/fftools/ffmpeg.c @@ -1123,7 +1123,6 @@ static void do_video_out(OutputFile *of, int ret, format_video_sync; AVPacket pkt; AVCodecContext *enc = ost->enc_ctx; - AVCodecParameters *mux_par = ost->st->codecpar; AVRational frame_rate; int nb_frames, nb0_frames, i; double delta, delta0; @@ -1285,18 +1284,6 @@ static void do_video_out(OutputFile *of, if (!check_recording_time(ost)) return; - if (enc->flags & (AV_CODEC_FLAG_INTERLACED_DCT | AV_CODEC_FLAG_INTERLACED_ME) && - ost->top_field_first >= 0) - in_picture->top_field_first = !!ost->top_field_first; - - if (in_picture->interlaced_frame) { - if (enc->codec->id == AV_CODEC_ID_MJPEG) - mux_par->field_order = in_picture->top_field_first ? AV_FIELD_TT:AV_FIELD_BB; - else - mux_par->field_order = in_picture->top_field_first ? AV_FIELD_TB:AV_FIELD_BT; - } else - mux_par->field_order = AV_FIELD_PROGRESSIVE; - in_picture->quality = enc->global_quality; in_picture->pict_type = 0; @@ -3433,6 +3420,20 @@ static int init_output_stream_encode(OutputStream *ost, AVFrame *frame) enc_ctx->field_order = AV_FIELD_TT; } + if (frame) { + if (enc_ctx->flags & (AV_CODEC_FLAG_INTERLACED_DCT | AV_CODEC_FLAG_INTERLACED_ME) && + ost->top_field_first >= 0) + frame->top_field_first = !!ost->top_field_first; + + if (frame->interlaced_frame) { + if (enc_ctx->codec->id == AV_CODEC_ID_MJPEG) + enc_ctx->field_order = frame->top_field_first ? AV_FIELD_TT:AV_FIELD_BB; + else + enc_ctx->field_order = frame->top_field_first ? AV_FIELD_TB:AV_FIELD_BT; + } else + enc_ctx->field_order = AV_FIELD_PROGRESSIVE; + } + if (ost->forced_keyframes) { if (!strncmp(ost->forced_keyframes, "expr:", 5)) { ret = av_expr_parse(&ost->forced_keyframes_pexpr, ost->forced_keyframes+5, diff --git a/tests/ref/fate/concat-demuxer-extended-lavf-mxf_d10 b/tests/ref/fate/concat-demuxer-extended-lavf-mxf_d10 index e3e76f217a..f6efc00ca4 100644 --- a/tests/ref/fate/concat-demuxer-extended-lavf-mxf_d10 +++ b/tests/ref/fate/concat-demuxer-extended-lavf-mxf_d10 @@ -1 +1 @@ -d66177ea3922692bc91cd0f8aa907650 *tests/data/fate/concat-demuxer-extended-lavf-mxf_d10.ffprobe +84496cfe2d668db395280ea67e5c6fbe *tests/data/fate/concat-demuxer-extended-lavf-mxf_d10.ffprobe diff --git a/tests/ref/fate/concat-demuxer-simple1-lavf-mxf_d10 b/tests/ref/fate/concat-demuxer-simple1-lavf-mxf_d10 index 79ce1e2306..8f3f2e5265 100644 --- a/tests/ref/fate/concat-demuxer-simple1-lavf-mxf_d10 +++ b/tests/ref/fate/concat-demuxer-simple1-lavf-mxf_d10 @@ -78,5 +78,5 @@ video|0|34|1.360000|34|1.360000|1|0.040000|N/A|N/A|150000|1924096|K_|1 Strings Metadata audio|1|65280|1.360000|65280|1.360000|1920|0.040000|N/A|N/A|7680|2074624|K_|1 Strings Metadata -0|mpeg2video|0|video|1/25|[0][0][0][0]|0x0000|720|608|0|0|0|0|1:1|45:38|yuv422p|5|tv|unknown|unknown|unknown|topleft|tt|N/A|1|N/A|25/1|25/1|1/25|0|0.000000|N/A|N/A|30000000|N/A|N/A|N/A|N/A|35|0|0|0|0|0|0|0|0|0|0|0|0|0x060A2B340101010501010D001300000000000000000000000000000000000001 +0|mpeg2video|0|video|1/25|[0][0][0][0]|0x0000|720|608|0|0|0|0|1:1|45:38|yuv422p|5|tv|unknown|unknown|unknown|topleft|tb|N/A|1|N/A|25/1|25/1|1/25|0|0.000000|N/A|N/A|30000000|N/A|N/A|N/A|N/A|35|0|0|0|0|0|0|0|0|0|0|0|0|0x060A2B340101010501010D001300000000000000000000000000000000000001 1|pcm_s16le|unknown|audio|1/48000|[0][0][0][0]|0x0000|s16|48000|2|unknown|16|N/A|0/0|0/0|1/48000|0|0.000000|N/A|N/A|1536000|N/A|N/A|N/A|N/A|35|0|0|0|0|0|0|0|0|0|0|0|0|0x060A2B340101010501010D001300000000000000000000000000000000000001 diff --git a/tests/ref/fate/rgb24-mkv b/tests/ref/fate/rgb24-mkv index 34d028cbfd..3b14cd0ef0 100644 --- a/tests/ref/fate/rgb24-mkv +++ b/tests/ref/fate/rgb24-mkv @@ -1,5 +1,5 @@ -fdc02d700dbe99315a9f0d928a9b935e *tests/data/fate/rgb24-mkv.matroska -58213 tests/data/fate/rgb24-mkv.matroska +fde8903c4df0ba8235dafcfd8a2f368c *tests/data/fate/rgb24-mkv.matroska +58216 tests/data/fate/rgb24-mkv.matroska #tb 0: 1/10 #media_type 0: video #codec_id 0: rawvideo diff --git a/tests/ref/lavf/mxf_d10 b/tests/ref/lavf/mxf_d10 index 85e337d157..30701619e0 100644 --- a/tests/ref/lavf/mxf_d10 +++ b/tests/ref/lavf/mxf_d10 @@ -1,3 +1,3 @@ -36fc2a640368f6d33987d2b2d39df966 *tests/data/lavf/lavf.mxf_d10 +da0ebbebb50a530b14c0f06017f464b3 *tests/data/lavf/lavf.mxf_d10 5332013 tests/data/lavf/lavf.mxf_d10 tests/data/lavf/lavf.mxf_d10 CRC=0x6c74d488

1 0

ffmpeg: pass decoded or filtered AVFrame to output stream initialization
by Jan Ekström 29 Oct '20

29 Oct '20

ffmpeg | branch: master | Jan Ekström <jeebjp(a)gmail.com> | Fri Sep 11 00:36:23 2020 +0300| [7369595c556516b90bf3acdf85043dc11add7d89] | committer: Jan Ekström ffmpeg: pass decoded or filtered AVFrame to output stream initialization Additionally, reap the first rewards by being able to set the color related encoding values based on the passed AVFrame. The only tests that seem to have changed their results with this change seem to be the MXF tests. There, the muxer writes the limited/full range flag to the output container if the encoder is not set to "unspecified". > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7369595c556516b90… --- fftools/ffmpeg.c | 33 ++++++++++++++++++++++----------- tests/ref/lavf/mxf_d10 | 2 +- tests/ref/lavf/mxf_dv25 | 2 +- tests/ref/lavf/mxf_dvcpro50 | 2 +- tests/ref/lavf/mxf_opatom | 2 +- 5 files changed, 26 insertions(+), 15 deletions(-) diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c index 52d1b09c78..20d191bcf6 100644 --- a/fftools/ffmpeg.c +++ b/fftools/ffmpeg.c @@ -947,9 +947,11 @@ early_exit: return float_pts; } -static int init_output_stream(OutputStream *ost, char *error, int error_len); +static int init_output_stream(OutputStream *ost, AVFrame *frame, + char *error, int error_len); -static int init_output_stream_wrapper(OutputStream *ost, unsigned int fatal) +static int init_output_stream_wrapper(OutputStream *ost, AVFrame *frame, + unsigned int fatal) { int ret = AVERROR_BUG; char error[1024] = {0}; @@ -957,7 +959,7 @@ static int init_output_stream_wrapper(OutputStream *ost, unsigned int fatal) if (ost->initialized) return 0; - ret = init_output_stream(ost, error, sizeof(error)); + ret = init_output_stream(ost, frame, error, sizeof(error)); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Error initializing output stream %d:%d -- %s\n", ost->file_index, ost->index, error); @@ -1131,7 +1133,7 @@ static void do_video_out(OutputFile *of, InputStream *ist = NULL; AVFilterContext *filter = ost->filter->filter; - init_output_stream_wrapper(ost, 1); + init_output_stream_wrapper(ost, next_picture, 1); sync_ipts = adjust_frame_pts_to_encoder_tb(of, ost, next_picture); if (ost->source_index >= 0) @@ -1513,7 +1515,7 @@ static int reap_filters(int flush) * the encoder earlier than receiving the first AVFrame. */ if (av_buffersink_get_type(filter) == AVMEDIA_TYPE_AUDIO) - init_output_stream_wrapper(ost, 1); + init_output_stream_wrapper(ost, NULL, 1); if (!ost->filtered_frame && !(ost->filtered_frame = av_frame_alloc())) { return AVERROR(ENOMEM); @@ -1936,7 +1938,7 @@ static void flush_encoders(void) finish_output_stream(ost); } - init_output_stream_wrapper(ost, 1); + init_output_stream_wrapper(ost, NULL, 1); } if (enc->codec_type != AVMEDIA_TYPE_VIDEO && enc->codec_type != AVMEDIA_TYPE_AUDIO) @@ -3309,7 +3311,7 @@ static void init_encoder_time_base(OutputStream *ost, AVRational default_time_ba enc_ctx->time_base = default_time_base; } -static int init_output_stream_encode(OutputStream *ost) +static int init_output_stream_encode(OutputStream *ost, AVFrame *frame) { InputStream *ist = get_input_stream(ost); AVCodecContext *enc_ctx = ost->enc_ctx; @@ -3406,6 +3408,14 @@ static int init_output_stream_encode(OutputStream *ost) enc_ctx->bits_per_raw_sample = FFMIN(dec_ctx->bits_per_raw_sample, av_pix_fmt_desc_get(enc_ctx->pix_fmt)->comp[0].depth); + if (frame) { + enc_ctx->color_range = frame->color_range; + enc_ctx->color_primaries = frame->color_primaries; + enc_ctx->color_trc = frame->color_trc; + enc_ctx->colorspace = frame->colorspace; + enc_ctx->chroma_sample_location = frame->chroma_location; + } + enc_ctx->framerate = ost->frame_rate; ost->st->avg_frame_rate = ost->frame_rate; @@ -3463,7 +3473,8 @@ static int init_output_stream_encode(OutputStream *ost) return 0; } -static int init_output_stream(OutputStream *ost, char *error, int error_len) +static int init_output_stream(OutputStream *ost, AVFrame *frame, + char *error, int error_len) { int ret = 0; @@ -3472,7 +3483,7 @@ static int init_output_stream(OutputStream *ost, char *error, int error_len) AVCodecContext *dec = NULL; InputStream *ist; - ret = init_output_stream_encode(ost); + ret = init_output_stream_encode(ost, frame); if (ret < 0) return ret; @@ -3724,7 +3735,7 @@ static int transcode_init(void) output_streams[i]->enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO)) continue; - ret = init_output_stream_wrapper(output_streams[i], 0); + ret = init_output_stream_wrapper(output_streams[i], NULL, 0); if (ret < 0) goto dump_format; } @@ -4657,7 +4668,7 @@ static int transcode_step(void) * early encoder initialization. */ if (av_buffersink_get_type(ost->filter->filter) == AVMEDIA_TYPE_AUDIO) - init_output_stream_wrapper(ost, 1); + init_output_stream_wrapper(ost, NULL, 1); if ((ret = transcode_from_filter(ost->filter->graph, &ist)) < 0) return ret; diff --git a/tests/ref/lavf/mxf_d10 b/tests/ref/lavf/mxf_d10 index aea469bb58..85e337d157 100644 --- a/tests/ref/lavf/mxf_d10 +++ b/tests/ref/lavf/mxf_d10 @@ -1,3 +1,3 @@ -e597f73ef9c9819710d2f815813eb91f *tests/data/lavf/lavf.mxf_d10 +36fc2a640368f6d33987d2b2d39df966 *tests/data/lavf/lavf.mxf_d10 5332013 tests/data/lavf/lavf.mxf_d10 tests/data/lavf/lavf.mxf_d10 CRC=0x6c74d488 diff --git a/tests/ref/lavf/mxf_dv25 b/tests/ref/lavf/mxf_dv25 index db6b76c6f8..d4559df862 100644 --- a/tests/ref/lavf/mxf_dv25 +++ b/tests/ref/lavf/mxf_dv25 @@ -1,3 +1,3 @@ -0fc964fa22bc8b3a389b81b9a2efccb3 *tests/data/lavf/lavf.mxf_dv25 +57623b3b968c0bb0d8a0bbaeef6fe657 *tests/data/lavf/lavf.mxf_dv25 3834413 tests/data/lavf/lavf.mxf_dv25 tests/data/lavf/lavf.mxf_dv25 CRC=0xbdaf7f52 diff --git a/tests/ref/lavf/mxf_dvcpro50 b/tests/ref/lavf/mxf_dvcpro50 index 09999914bf..8bcf67d17f 100644 --- a/tests/ref/lavf/mxf_dvcpro50 +++ b/tests/ref/lavf/mxf_dvcpro50 @@ -1,3 +1,3 @@ -aa81ea83af44a69e73849e327cc4bd12 *tests/data/lavf/lavf.mxf_dvcpro50 +6e82b4cc962199e2593e30851ff7ccfb *tests/data/lavf/lavf.mxf_dvcpro50 7431213 tests/data/lavf/lavf.mxf_dvcpro50 tests/data/lavf/lavf.mxf_dvcpro50 CRC=0xe3bbe4b4 diff --git a/tests/ref/lavf/mxf_opatom b/tests/ref/lavf/mxf_opatom index 05794a4e5e..1aa843a22a 100644 --- a/tests/ref/lavf/mxf_opatom +++ b/tests/ref/lavf/mxf_opatom @@ -1,3 +1,3 @@ -06a1816aa91c733e1ef7e45d82e4f1d3 *tests/data/lavf/lavf.mxf_opatom +d5f56215c2b16dee204f03bfa653dd1b *tests/data/lavf/lavf.mxf_opatom 4717625 tests/data/lavf/lavf.mxf_opatom tests/data/lavf/lavf.mxf_opatom CRC=0xf55aa22a

1 0

ffmpeg: move A/V non-streamcopy initialization to a later point
by Jan Ekström 29 Oct '20

29 Oct '20

ffmpeg | branch: master | Jan Ekström <jeebjp(a)gmail.com> | Fri Sep 11 00:25:21 2020 +0300| [67be1ce0c6de330b1c10d1d121819d8a74a7b1f5] | committer: Jan Ekström ffmpeg: move A/V non-streamcopy initialization to a later point - For video, this means a single initialization point in do_video_out. - For audio we unfortunately need to do it in two places just before the buffer sink is utilized (if av_buffersink_get_samples would still work according to its specification after a call to avfilter_graph_request_oldest was made, we could at least remove the one in transcode_step). Other adjustments to make things work: - As the AVFrame PTS adjustment to encoder time base needs the encoder to be initialized, so it is now moved to do_{video,audio}_out, right after the encoder has been initialized. Due to this, the additional parameter in do_video_out is removed as it is no longer necessary. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=67be1ce0c6de330b1… --- fftools/ffmpeg.c | 112 ++++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 77 insertions(+), 35 deletions(-) diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c index 498e5f08a6..52d1b09c78 100644 --- a/fftools/ffmpeg.c +++ b/fftools/ffmpeg.c @@ -947,6 +947,28 @@ early_exit: return float_pts; } +static int init_output_stream(OutputStream *ost, char *error, int error_len); + +static int init_output_stream_wrapper(OutputStream *ost, unsigned int fatal) +{ + int ret = AVERROR_BUG; + char error[1024] = {0}; + + if (ost->initialized) + return 0; + + ret = init_output_stream(ost, error, sizeof(error)); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error initializing output stream %d:%d -- %s\n", + ost->file_index, ost->index, error); + + if (fatal) + exit_program(1); + } + + return ret; +} + static void do_audio_out(OutputFile *of, OutputStream *ost, AVFrame *frame) { @@ -958,6 +980,8 @@ static void do_audio_out(OutputFile *of, OutputStream *ost, pkt.data = NULL; pkt.size = 0; + adjust_frame_pts_to_encoder_tb(of, ost, frame); + if (!check_recording_time(ost)) return; @@ -1092,8 +1116,7 @@ static void do_subtitle_out(OutputFile *of, static void do_video_out(OutputFile *of, OutputStream *ost, - AVFrame *next_picture, - double sync_ipts) + AVFrame *next_picture) { int ret, format_video_sync; AVPacket pkt; @@ -1103,10 +1126,14 @@ static void do_video_out(OutputFile *of, int nb_frames, nb0_frames, i; double delta, delta0; double duration = 0; + double sync_ipts = AV_NOPTS_VALUE; int frame_size = 0; InputStream *ist = NULL; AVFilterContext *filter = ost->filter->filter; + init_output_stream_wrapper(ost, 1); + sync_ipts = adjust_frame_pts_to_encoder_tb(of, ost, next_picture); + if (ost->source_index >= 0) ist = input_streams[ost->source_index]; @@ -1440,28 +1467,6 @@ static void do_video_stats(OutputStream *ost, int frame_size) } } -static int init_output_stream(OutputStream *ost, char *error, int error_len); - -static int init_output_stream_wrapper(OutputStream *ost, unsigned int fatal) -{ - int ret = AVERROR_BUG; - char error[1024] = {0}; - - if (ost->initialized) - return 0; - - ret = init_output_stream(ost, error, sizeof(error)); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Error initializing output stream %d:%d -- %s\n", - ost->file_index, ost->index, error); - - if (fatal) - exit_program(1); - } - - return ret; -} - static void finish_output_stream(OutputStream *ost) { OutputFile *of = output_files[ost->file_index]; @@ -1498,7 +1503,17 @@ static int reap_filters(int flush) continue; filter = ost->filter->filter; - init_output_stream_wrapper(ost, 1); + /* + * Unlike video, with audio the audio frame size matters. + * Currently we are fully reliant on the lavfi filter chain to + * do the buffering deed for us, and thus the frame size parameter + * needs to be set accordingly. Where does one get the required + * frame size? From the initialized AVCodecContext of an audio + * encoder. Thus, if we have gotten to an audio stream, initialize + * the encoder earlier than receiving the first AVFrame. + */ + if (av_buffersink_get_type(filter) == AVMEDIA_TYPE_AUDIO) + init_output_stream_wrapper(ost, 1); if (!ost->filtered_frame && !(ost->filtered_frame = av_frame_alloc())) { return AVERROR(ENOMEM); @@ -1506,7 +1521,6 @@ static int reap_filters(int flush) filtered_frame = ost->filtered_frame; while (1) { - double float_pts = AV_NOPTS_VALUE; // this is identical to filtered_frame.pts but with higher precision ret = av_buffersink_get_frame_flags(filter, filtered_frame, AV_BUFFERSINK_FLAG_NO_REQUEST); if (ret < 0) { @@ -1515,7 +1529,7 @@ static int reap_filters(int flush) "Error in av_buffersink_get_frame_flags(): %s\n", av_err2str(ret)); } else if (flush && ret == AVERROR_EOF) { if (av_buffersink_get_type(filter) == AVMEDIA_TYPE_VIDEO) - do_video_out(of, ost, NULL, AV_NOPTS_VALUE); + do_video_out(of, ost, NULL); } break; } @@ -1524,15 +1538,12 @@ static int reap_filters(int flush) continue; } - float_pts = adjust_frame_pts_to_encoder_tb(of, ost, - filtered_frame); - switch (av_buffersink_get_type(filter)) { case AVMEDIA_TYPE_VIDEO: if (!ost->frame_aspect_ratio.num) enc->sample_aspect_ratio = filtered_frame->sample_aspect_ratio; - do_video_out(of, ost, filtered_frame, float_pts); + do_video_out(of, ost, filtered_frame); break; case AVMEDIA_TYPE_AUDIO: if (!(enc->codec->capabilities & AV_CODEC_CAP_PARAM_CHANGE) && @@ -3698,10 +3709,19 @@ static int transcode_init(void) goto dump_format; } - /* open each encoder */ + /* + * initialize stream copy and subtitle/data streams. + * Encoded AVFrame based streams will get initialized as follows: + * - when the first AVFrame is received in do_video_out + * - just before the first AVFrame is received in either transcode_step + * or reap_filters due to us requiring the filter chain buffer sink + * to be configured with the correct audio frame size, which is only + * known after the encoder is initialized. + */ for (i = 0; i < nb_output_streams; i++) { - // skip streams fed from filtergraphs until we have a frame for them - if (output_streams[i]->filter) + if (!output_streams[i]->stream_copy && + (output_streams[i]->enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO || + output_streams[i]->enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO)) continue; ret = init_output_stream_wrapper(output_streams[i], 0); @@ -4615,7 +4635,29 @@ static int transcode_step(void) } if (ost->filter && ost->filter->graph->graph) { - init_output_stream_wrapper(ost, 1); + /* + * Similar case to the early audio initialization in reap_filters. + * Audio is special in ffmpeg.c currently as we depend on lavfi's + * audio frame buffering/creation to get the output audio frame size + * in samples correct. The audio frame size for the filter chain is + * configured during the output stream initialization. + * + * Apparently avfilter_graph_request_oldest (called in + * transcode_from_filter just down the line) peeks. Peeking already + * puts one frame "ready to be given out", which means that any + * update in filter buffer sink configuration afterwards will not + * help us. And yes, even if it would be utilized, + * av_buffersink_get_samples is affected, as it internally utilizes + * the same early exit for peeked frames. + * + * In other words, if avfilter_graph_request_oldest would not make + * further filter chain configuration or usage of + * av_buffersink_get_samples useless (by just causing the return + * of the peeked AVFrame as-is), we could get rid of this additional + * early encoder initialization. + */ + if (av_buffersink_get_type(ost->filter->filter) == AVMEDIA_TYPE_AUDIO) + init_output_stream_wrapper(ost, 1); if ((ret = transcode_from_filter(ost->filter->graph, &ist)) < 0) return ret;

1 0

ffmpeg: move AVFrame time base adjustment into a function
by Jan Ekström 29 Oct '20

29 Oct '20

ffmpeg | branch: master | Jan Ekström <jeebjp(a)gmail.com> | Sun Sep 13 11:41:41 2020 +0300| [3360c9a5679cc9bd83ab5860757ebab8f64ea8ba] | committer: Jan Ekström ffmpeg: move AVFrame time base adjustment into a function This will have to be called later for video down the line. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3360c9a5679cc9bd8… --- fftools/ffmpeg.c | 72 ++++++++++++++++++++++++++++++++++++-------------------- 1 file changed, 47 insertions(+), 25 deletions(-) diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c index 3af189e7f2..498e5f08a6 100644 --- a/fftools/ffmpeg.c +++ b/fftools/ffmpeg.c @@ -903,6 +903,50 @@ static int check_recording_time(OutputStream *ost) return 1; } +static double adjust_frame_pts_to_encoder_tb(OutputFile *of, OutputStream *ost, + AVFrame *frame) +{ + double float_pts = AV_NOPTS_VALUE; // this is identical to frame.pts but with higher precision + AVCodecContext *enc = ost->enc_ctx; + if (!frame || frame->pts == AV_NOPTS_VALUE || + !enc || !ost->filter || !ost->filter->graph->graph) + goto early_exit; + + { + AVFilterContext *filter = ost->filter->filter; + + int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time; + AVRational filter_tb = av_buffersink_get_time_base(filter); + AVRational tb = enc->time_base; + int extra_bits = av_clip(29 - av_log2(tb.den), 0, 16); + + tb.den <<= extra_bits; + float_pts = + av_rescale_q(frame->pts, filter_tb, tb) - + av_rescale_q(start_time, AV_TIME_BASE_Q, tb); + float_pts /= 1 << extra_bits; + // avoid exact midoints to reduce the chance of rounding differences, this can be removed in case the fps code is changed to work with integers + float_pts += FFSIGN(float_pts) * 1.0 / (1<<17); + + frame->pts = + av_rescale_q(frame->pts, filter_tb, enc->time_base) - + av_rescale_q(start_time, AV_TIME_BASE_Q, enc->time_base); + } + +early_exit: + + if (debug_ts) { + av_log(NULL, AV_LOG_INFO, "filter -> pts:%s pts_time:%s exact:%f time_base:%d/%d\n", + frame ? av_ts2str(frame->pts) : "NULL", + frame ? av_ts2timestr(frame->pts, &enc->time_base) : "NULL", + float_pts, + enc ? enc->time_base.num : -1, + enc ? enc->time_base.den : -1); + } + + return float_pts; +} + static void do_audio_out(OutputFile *of, OutputStream *ost, AVFrame *frame) { @@ -1479,37 +1523,15 @@ static int reap_filters(int flush) av_frame_unref(filtered_frame); continue; } - if (filtered_frame->pts != AV_NOPTS_VALUE) { - int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time; - AVRational filter_tb = av_buffersink_get_time_base(filter); - AVRational tb = enc->time_base; - int extra_bits = av_clip(29 - av_log2(tb.den), 0, 16); - - tb.den <<= extra_bits; - float_pts = - av_rescale_q(filtered_frame->pts, filter_tb, tb) - - av_rescale_q(start_time, AV_TIME_BASE_Q, tb); - float_pts /= 1 << extra_bits; - // avoid exact midoints to reduce the chance of rounding differences, this can be removed in case the fps code is changed to work with integers - float_pts += FFSIGN(float_pts) * 1.0 / (1<<17); - - filtered_frame->pts = - av_rescale_q(filtered_frame->pts, filter_tb, enc->time_base) - - av_rescale_q(start_time, AV_TIME_BASE_Q, enc->time_base); - } + + float_pts = adjust_frame_pts_to_encoder_tb(of, ost, + filtered_frame); switch (av_buffersink_get_type(filter)) { case AVMEDIA_TYPE_VIDEO: if (!ost->frame_aspect_ratio.num) enc->sample_aspect_ratio = filtered_frame->sample_aspect_ratio; - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "filter -> pts:%s pts_time:%s exact:%f time_base:%d/%d\n", - av_ts2str(filtered_frame->pts), av_ts2timestr(filtered_frame->pts, &enc->time_base), - float_pts, - enc->time_base.num, enc->time_base.den); - } - do_video_out(of, ost, filtered_frame, float_pts); break; case AVMEDIA_TYPE_AUDIO:

1 0

ffmpeg: add a data size threshold for muxing queue size
by Jan Ekström 29 Oct '20

29 Oct '20

ffmpeg | branch: master | Jan Ekström <jeebjp(a)gmail.com> | Thu Oct 15 21:21:05 2020 +0300| [453b2f3c154f6b83221940ad411599ded91f7413] | committer: Jan Ekström ffmpeg: add a data size threshold for muxing queue size This way the old max queue size limit based behavior for streams where each individual packet is large is kept, while for smaller streams more packets can be buffered (current default is at 50 megabytes per stream). For some explanation, by default ffmpeg copies packets from before the appointed seek point/start time and puts them into the local muxing queue. Before, it getting utilized was much less likely since as soon as the filter chain was initialized, the encoder (and thus output stream) was also initialized. Now, since we will be pushing the encoder initialization to when the first AVFrame is decoded and filtered - which only happens after the exact seek point is hit as packets are ignored until then - this queue will be seeing much more usage. In more layman's terms, this attempts to fix cases such as where: - seek point ends up being 5 seconds before requested time. - audio is set to copy, and thus immediately begins filling the muxing queue. - video is being encoded, and thus all received packets are skipped until the requested time is hit. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=453b2f3c154f6b832… --- doc/ffmpeg.texi | 5 +++++ fftools/ffmpeg.c | 11 +++++++++-- fftools/ffmpeg.h | 11 +++++++++++ fftools/ffmpeg_opt.c | 8 ++++++++ 4 files changed, 33 insertions(+), 2 deletions(-) diff --git a/doc/ffmpeg.texi b/doc/ffmpeg.texi index 96b3257e79..95d6463685 100644 --- a/doc/ffmpeg.texi +++ b/doc/ffmpeg.texi @@ -1746,6 +1746,11 @@ this buffer, in packets, for the matching output stream. The default value of this option should be high enough for most uses, so only touch this option if you are sure that you need it. +@item -muxing_queue_data_threshold @var{bytes} (@emph{output,per-stream}) +This is a minimum threshold until which the muxing queue size is not taken into +account. Defaults to 50 megabytes per stream, and is based on the overall size +of packets passed to the muxer. + @item -auto_conversion_filters (@emph{global}) Enable automatically inserting format conversion filters in all filter graphs, including those defined by @option{-vf}, @option{-af}, diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c index cb7644de6a..3af189e7f2 100644 --- a/fftools/ffmpeg.c +++ b/fftools/ffmpeg.c @@ -725,8 +725,13 @@ static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int u AVPacket tmp_pkt = {0}; /* the muxer is not initialized yet, buffer the packet */ if (!av_fifo_space(ost->muxing_queue)) { - int new_size = FFMIN(2 * av_fifo_size(ost->muxing_queue), - ost->max_muxing_queue_size); + unsigned int are_we_over_size = + (ost->muxing_queue_data_size + pkt->size) > ost->muxing_queue_data_threshold; + int new_size = are_we_over_size ? + FFMIN(2 * av_fifo_size(ost->muxing_queue), + ost->max_muxing_queue_size) : + 2 * av_fifo_size(ost->muxing_queue); + if (new_size <= av_fifo_size(ost->muxing_queue)) { av_log(NULL, AV_LOG_ERROR, "Too many packets buffered for output stream %d:%d.\n", @@ -741,6 +746,7 @@ static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int u if (ret < 0) exit_program(1); av_packet_move_ref(&tmp_pkt, pkt); + ost->muxing_queue_data_size += tmp_pkt.size; av_fifo_generic_write(ost->muxing_queue, &tmp_pkt, sizeof(tmp_pkt), NULL); return; } @@ -2991,6 +2997,7 @@ static int check_init_output_file(OutputFile *of, int file_index) while (av_fifo_size(ost->muxing_queue)) { AVPacket pkt; av_fifo_generic_read(ost->muxing_queue, &pkt, sizeof(pkt), NULL); + ost->muxing_queue_data_size -= pkt.size; write_packet(of, &pkt, ost, 1); } } diff --git a/fftools/ffmpeg.h b/fftools/ffmpeg.h index 8665218dcf..3b54dab7fc 100644 --- a/fftools/ffmpeg.h +++ b/fftools/ffmpeg.h @@ -215,6 +215,8 @@ typedef struct OptionsContext { int nb_passlogfiles; SpecifierOpt *max_muxing_queue_size; int nb_max_muxing_queue_size; + SpecifierOpt *muxing_queue_data_threshold; + int nb_muxing_queue_data_threshold; SpecifierOpt *guess_layout_max; int nb_guess_layout_max; SpecifierOpt *apad; @@ -547,6 +549,15 @@ typedef struct OutputStream { /* the packets are buffered here until the muxer is ready to be initialized */ AVFifoBuffer *muxing_queue; + /* + * The size of the AVPackets' buffers in queue. + * Updated when a packet is either pushed or pulled from the queue. + */ + size_t muxing_queue_data_size; + + /* Threshold after which max_muxing_queue_size will be in effect */ + size_t muxing_queue_data_threshold; + /* packet picture type */ int pict_type; diff --git a/fftools/ffmpeg_opt.c b/fftools/ffmpeg_opt.c index d58462b6ae..7ee034c9c9 100644 --- a/fftools/ffmpeg_opt.c +++ b/fftools/ffmpeg_opt.c @@ -87,6 +87,7 @@ static const char *opt_name_canvas_sizes[] = {"canvas_size", NULL}; static const char *opt_name_pass[] = {"pass", NULL}; static const char *opt_name_passlogfiles[] = {"passlogfile", NULL}; static const char *opt_name_max_muxing_queue_size[] = {"max_muxing_queue_size", NULL}; +static const char *opt_name_muxing_queue_data_threshold[] = {"muxing_queue_data_threshold", NULL}; static const char *opt_name_guess_layout_max[] = {"guess_layout_max", NULL}; static const char *opt_name_apad[] = {"apad", NULL}; static const char *opt_name_discard[] = {"discard", NULL}; @@ -1564,6 +1565,11 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e MATCH_PER_STREAM_OPT(max_muxing_queue_size, i, ost->max_muxing_queue_size, oc, st); ost->max_muxing_queue_size *= sizeof(AVPacket); + ost->muxing_queue_data_size = 0; + + ost->muxing_queue_data_threshold = 50*1024*1024; + MATCH_PER_STREAM_OPT(muxing_queue_data_threshold, i, ost->muxing_queue_data_threshold, oc, st); + if (oc->oformat->flags & AVFMT_GLOBALHEADER) ost->enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; @@ -3761,6 +3767,8 @@ const OptionDef options[] = { { "max_muxing_queue_size", HAS_ARG | OPT_INT | OPT_SPEC | OPT_EXPERT | OPT_OUTPUT, { .off = OFFSET(max_muxing_queue_size) }, "maximum number of packets that can be buffered while waiting for all streams to initialize", "packets" }, + { "muxing_queue_data_threshold", HAS_ARG | OPT_INT | OPT_SPEC | OPT_EXPERT | OPT_OUTPUT, { .off = OFFSET(muxing_queue_data_threshold) }, + "set the threshold after which max_muxing_queue_size is taken into account", "bytes" }, /* data codec support */ { "dcodec", HAS_ARG | OPT_DATA | OPT_PERFILE | OPT_EXPERT | OPT_INPUT | OPT_OUTPUT, { .func_arg = opt_data_codec },

1 0