May 2021 - FFmpeg-trac - lists.ffmpeg.org

#9239(documentation:new): guided filter
by FFmpeg 13 May '21

13 May '21

#9239: guided filter -------------------------------------+------------------------------------- Reporter: Michael | Type: defect Koch | Status: new | Priority: minor Component: | Version: git- documentation | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Wrong headline in documentation: The name of the filter is "guided" and not "guided filter". -- Ticket URL: <https://trac.ffmpeg.org/ticket/9239> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9102(undetermined:new): make HLS from multicast with ac3 audio track which have 0 channels
by FFmpeg 13 May '21

13 May '21

#9102: make HLS from multicast with ac3 audio track which have 0 channels -------------------------------------+------------------------------------- Reporter: krs_vital | Type: defect Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: ac3 | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- I want to make HLS with stream determined like this: [mpegts @ 0x2635780] Could not find codec parameters for stream 3 (Audio: ac3 (AC-3 / 0x332D4341), 0 channels, fltp): unspecified sample rate Consider increasing the value for the 'analyzeduration' (25000000) and 'probesize' (25000000) options Input #0, mpegts, from 'udp://239.192.39.3:1234': Duration: N/A, start: 42075.039333, bitrate: N/A Program 17104 Metadata: service_name : VIP Premiere service_provider: HTB+ Stream #0:1[0x130]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc Stream #0:2[0x194](rus): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, fltp, 192 kb/s Stream #0:3[0x1a8](rus): Audio: ac3 (AC-3 / 0x332D4341), 0 channels, fltp Stream #0:4[0x1f8](rus): Subtitle: dvb_subtitle ([6][0][0][0] / 0x0006) but ffmpeg stops and do nothing. sometimes in this stream on audio track [Stream #0:3] appears bitrate when TV provider make it, but sometimes this audio track is empty.This usually happens when content with 5.1 sound is turned on and then this track is not empty. How to make ffmpeg automatically detect the audio track on the fly and start working with it, since during decoding it can be empty or not? Ffmpeg starts like this: /usr/local/bin/ffmpeg -analyzeduration 25000000 -probesize 25000000 -dts_delta_threshold 3600 -err_detect ignore_err -correct_ts_overflow 0 -copytb 1 -overrun_nonfatal 1 -i udp://239.192.39.3:1234 -y -sn -dn -c:v copy -c:a aac -vsync 1 -hls_time 10 -map 0 -ignore_unknown -strftime 1 -strftime_mkdir 1 -max_interleave_delta 360000000000 -hls_flags omit_endlist -hls_segment_filename /store/records/Premium/%Y/%m/%d/%H/%s.ts /hls/tmp/Premium.m3u8 -- Ticket URL: <https://trac.ffmpeg.org/ticket/9102> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9232(undetermined:new): ERROR: librsvg-2.0 not found using pkg-config (Windows)
by FFmpeg 12 May '21

12 May '21

#9232: ERROR: librsvg-2.0 not found using pkg-config (Windows) -------------------------------------+------------------------------------- Reporter: qdras | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- I'm trying to compile ffmpeg with librsvg, but it shows error message: ERROR: librsvg-2.0 not found using pkg-config In the config.log file is this: require_pkg_config librsvg librsvg-2.0 librsvg-2.0/librsvg/rsvg.h rsvg_handle_render_cairo check_pkg_config librsvg librsvg-2.0 librsvg-2.0/librsvg/rsvg.h rsvg_handle_render_cairo test_pkg_config librsvg librsvg-2.0 librsvg-2.0/librsvg/rsvg.h rsvg_handle_render_cairo pkg-config --exists --print-errors librsvg-2.0 Package librsvg-2.0 was not found in the pkg-config search path. Perhaps you should add the directory containing `librsvg-2.0.pc' to the PKG_CONFIG_PATH environment variable No package 'librsvg-2.0' found ERROR: librsvg-2.0 not found using pkg-config Can tell me what to do? How to reproduce: {{{ I use command ./configure --prefix=ffmpeg/ --enable-cross-compile --disable-network --disable-debug --disable-yasm --enable-librsvg }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/9232> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9235(undetermined:new): commit 958bea5 broke fate-flv-demux
by FFmpeg 12 May '21

12 May '21

#9235: commit 958bea5 broke fate-flv-demux -------------------------------------+------------------------------------- Reporter: haihao | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- commit 958bea5 caused a regression when running `make fate-flv-demux V=2 KEEP=1` [flv @ 0x5612cb432e90] Packet corrupt (stream = 0, dts = 11678). [NULL @ 0x5612cb434c20] Invalid NAL unit size (2771 > 1186). [NULL @ 0x5612cb434c20] missing picture in access unit with size 1190 make: *** [tests/Makefile:255: fate-flv-demux] Error 1 `make fate-flv-demux V=2 KEEP=1` works well after resetting HEAD to 958bea5~1 The issue still exists after using the latest FFmpeg (commit 7fac6efa) -- Ticket URL: <https://trac.ffmpeg.org/ticket/9235> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9234(undetermined:new): Empty m3u8 after a Ctrl+C
by FFmpeg 12 May '21

12 May '21

#9234: Empty m3u8 after a Ctrl+C -------------------------------------+------------------------------------- Reporter: Gonzalo de | Type: defect Brito | Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: hls m3u8 | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- == Summary of the bug: == In ffmpeg version 4.4, if we kill ffmpeg during a m3u8 copy process, '''sometimes''' the output m3u8 file results completly empty. During the copy process a cat on the file exhibits the expected content, but after the Ctrl+C the file results completly empty. We couldn't reproduce this behaviour in ffmpeg v4.2.4, it seems to work perfectly fine in that version. == How to reproduce: == The command we're running is the following: {{{ ./ffmpeg_staging/ffmpeg -user_agent "Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/71.0.3554.0 Safari/537.36" -i <url> -c copy -f hls -hls_list_size 86400 -hls_flags delete_segments+append_list+program_date_time+temp_file -hls_segment_filename $segment_path $m3u8_path }}} == FFmpeg versions & builds: == This bug shows up in the following ffmpeg version, compiled from source: {{{ ffmpeg version n4.4-10-g75c3969292 Copyright (c) 2000-2021 the FFmpeg developers built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04) configuration: --arch=amd64 --enable-gpl --enable-gnutls --enable-libx264 --toolchain=hardened --disable-stripping libavutil 56. 70.100 / 56. 70.100 libavcodec 58.134.100 / 58.134.100 libavformat 58. 76.100 / 58. 76.100 libavdevice 58. 13.100 / 58. 13.100 libavfilter 7.110.100 / 7.110.100 libswscale 5. 9.100 / 5. 9.100 libswresample 3. 9.100 / 3. 9.100 libpostproc 55. 9.100 / 55. 9.100 }}} We '''CAN'T''' reproduce this bug (works perfectly fine) in the following ffmpeg version, also compiled from source: {{{ ffmpeg version 4.2.4 Copyright (c) 2000-2020 the FFmpeg developers built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04) configuration: --arch=amd64 --enable-gpl --enable-gnutls --enable-libx264 --toolchain=hardened --disable-stripping libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 / 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 / 55. 5.100 }}} Os: Ubuntu 20.04 If you need any further information, please ask me. Best regards, Gonzalo.- -- Ticket URL: <https://trac.ffmpeg.org/ticket/9234> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#8779(ffmpeg:new): Zscale transform to ST 2084 EOTF results in black video output
by FFmpeg 11 May '21

11 May '21

#8779: Zscale transform to ST 2084 EOTF results in black video output -------------------------------------+------------------------------------- Reporter: zambelli | Type: defect Status: new | Priority: important Component: ffmpeg | Version: git- Keywords: zscale, | master libzimg, z.lib, smpte2084, | Blocked By: st2084, hdr | Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- == Summary of the bug: == Sometime between releases 4.2.2 and 4.2.3 a bug was introduced into Zscale filter which causes the video output of the filter to be reduced to black when transforming video using SMPTE ST 2084 (PQ) EOTF. == How to reproduce: == Source video: https://hulu.box.com/shared/static/5o419x2dag93wx5f0msc69gbljfydri1.mp4 ffmpeg version git-2020-06-28-4cfcfb3 built with gcc 9.3.1 (GCC) 20200621 {{{ ffmpeg.exe -y -i zscale_test_source.mp4 -vf zscale=t=linear:w=640:h=360,format=gbrpf32le,zscale=t=smpte2084:p=bt709:m=bt709,format=yuv420p10le -vcodec ffv1 zscale_test_smpte2084.avi }}} Sample output: https://hulu.box.com/shared/static/6a8yjs5mel2scv63vvpmrn0438fqbiil.avi If t=smpte2084 is replaced with another EOTF, such as bt709, the filter produces the correct output: {{{ ffmpeg.exe -y -i zscale_test_source.mp4 -vf zscale=t=linear:w=640:h=360,format=gbrpf32le,zscale=t=bt709:p=bt709:m=bt709,format=yuv420p10le -vcodec ffv1 zscale_test_bt709.avi }}} Sample output: https://hulu.box.com/shared/static/uuy3gfhg6mwflrqmhvb2bvkl7009nlf4.avi The bug does not repro in FFmpeg release 4.2.2. To the best of my memory while working with nightly builds, this Zscale transform worked correctly until at least early May 2020. -- Ticket URL: <https://trac.ffmpeg.org/ticket/8779> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9228(avcodec:new): multiple output in single C/libavcodec xcode process offsets "start" in output files
by FFmpeg 11 May '21

11 May '21

#9228: multiple output in single C/libavcodec xcode process offsets "start" in output files -------------------------------------+------------------------------------- Reporter: Ray | Type: defect Status: new | Priority: normal Component: avcodec | Version: 4.2.4 Keywords: AAC | Blocked By: transcoding | Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: I have a C/libavcodec program that transcodes multiple input files to resulting mp3 or m4a using the inbuilt ffmeg libmp3lame or aac encoders respectively. No libavcodec constructs are reused (new AVCodecContext, AVFormatContext, AVAudioFifo used for every transcode) When generating .m4a output, examining the output using ffprobe, I observe the "start" is incrementally offset. the first m4a file is at 00:00:00 but the subsequent m4a file has a "start" that is offset by a time very close to the length of the first file. This does not happen when transcoding to mp3. I have been able to reproduce this using the examples/transcode_aac.c and modified it so that it creates multiple files. For example, when transcoding the same 5sec wav file to m4a within the same process generates a set of files but the start time is offset further and further as each new file is generated - note the "start" time {{{ $ for i in /tmp/a/foo*m4a; do ffprobe -hide_banner -i $i; done Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/tmp/a/foo00.m4a': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf58.29.100 Duration: 00:00:05.02, start: 0.000000, bitrate: 79 kb/s Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 76 kb/s (default) Metadata: handler_name : SoundHandler Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/tmp/a/foo01.m4a': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf58.29.100 Duration: 00:00:05.02, start: 4.976009, bitrate: 79 kb/s Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 76 kb/s (default) Metadata: handler_name : SoundHandler Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/tmp/a/foo02.m4a': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf58.29.100 Duration: 00:00:05.02, start: 9.976009, bitrate: 79 kb/s Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 76 kb/s (default) Metadata: handler_name : SoundHandler }}} I would expect ALL files to be identical. I tested this with built in aac and also libfdk_aac and both exhibit this behaviour How to reproduce: {{{ # have a single process that will transocde mulitple files to aac within a single process. # see attached patched transcode_aac.c $ ./transcode_aac input.wav output m4a 3 # generates 3x m4a encoded by inbuilt "aac" encoder called output{00,01,02}.m4a # examine the output files, observing the start time via ffprobe }}} Attached are the wav input, m4a outputs and the sameple src code that demonstrates this - code is examples/doc/transcode_aac.c with a loop around old main() function with a new output file name on next iteration. -- Ticket URL: <https://trac.ffmpeg.org/ticket/9228> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9236(avformat:new): Appending new segments into old hls segment list is very slow for large m3u8 file
by FFmpeg 11 May '21

11 May '21

#9236: Appending new segments into old hls segment list is very slow for large m3u8 file ----------------------------------+--------------------------------------- Reporter: litzh | Type: enhancement Status: new | Priority: normal Component: avformat | Version: git-master Keywords: HLS | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+--------------------------------------- Summary of the bug: I am trying to record rtmp live stream into HLS with `append_list` flag. When the recording is interrupted and the m3u8 file is very large (see attached file), it would take several minutes to resume recording. With `append_list` enabled, `find_segment_by_filename` finds duplicate segment filenames by linearly traversing the singly linked list. `av_strcasecmp` is a simple loop and cannot be accelerated by SMID. The time complexity is O(n^3^), where n is the total bytes of the segment filenames. How to reproduce: {{{ % ffmpeg -rw_timeout 10000000 -y -flv_ignore_prevtag 1 -i "rtmp://example.com/live/12544134-c2397sl2lrnb4obtt020" -vcodec copy -acodec copy -hls_time 4 -hls_flags append_list -hls_list_size 0 -hls_segment_filename 12544134-c2397sl2lrnb4obtt020-0_540p-%04d.ts -f hls 12544134-c2397sl2lrnb4obtt020-0_540p.m3u8 ffmpeg version N-102493-g7ce0f246f4 Copyright (c) 2000-2021 the FFmpeg developers built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04) configuration: --pkg-config-flags=--static --extra-ldflags='-ldl -lm -lpthread -lrt -lstdc++ -static' --enable-gpl --enable-nonfree --enable- version3 --extra-libs=-ldl --disable-programs --enable-ffmpeg --enable- ffprobe --enable-static libavutil 57. 0.100 / 57. 0.100 libavcodec 59. 1.100 / 59. 1.100 libavformat 59. 0.101 / 59. 0.101 libavdevice 59. 0.100 / 59. 0.100 libavfilter 8. 0.101 / 8. 0.101 libswscale 6. 0.100 / 6. 0.100 libswresample 4. 0.100 / 4. 0.100 libpostproc 56. 0.100 / 56. 0.100 Input #0, flv, from 'rtmp://example.com/live/12544134-c2397sl2lrnb4obtt020': Metadata: |RtmpSampleAccess: true encoder : DyEncoder58.40.100 Duration: 00:00:00.00, start: 661322.354000, bitrate: N/A Stream #0:0: Data: none Stream #0:1: Subtitle: text Stream #0:2: Audio: aac (LC), 44100 Hz, stereo, fltp, 160 kb/s Stream #0:3: Video: h264 (Main), yuv420p(progressive), 1920x1080, 8001 kb/s, 62.50 fps, 62 tbr, 1k tbn }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/9236> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9233(avformat:new): MPEG-2 audio stream probed as AC-3 due to PMT descriptor
by FFmpeg 10 May '21

10 May '21

#9233: MPEG-2 audio stream probed as AC-3 due to PMT descriptor -------------------------------------+------------------------------------- Reporter: Chris | Type: defect Olekas | Status: new | Priority: normal Component: avformat | Version: Keywords: mpegts AAC | unspecified ac3 | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: I have a stream that within the PMT has an AC-3 descriptor but the audio type is MPEG-2 audio (IEC 13818-3) How to reproduce: Use the attached ac3-mpeg2-issue.ts {{{ Input #0, mpegts, from 'ac3-mpeg2-issue.ts': Duration: 00:00:03.27, start: 22824.824789, bitrate: 10034 kb/s Program 1 Metadata: service_name : HRD_NUESTRT_SD_33431_0 service_provider: Stream #0:0[0x3e9]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, top first), 720x480 [SAR 8:9 DAR 4:3], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Side data: cpb: bitrate max/min/avg: 3750000/0/0 buffer size: 1835008 vbv_delay: N/A Stream #0:1[0x3ea](spa): Audio: ac3 (AC-3 / 0x332D4341), 0 channels, fltp }}} Tested on: f8d910e90f599f338438833dfc92e2f1915ce414 -- Ticket URL: <https://trac.ffmpeg.org/ticket/9233> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#8769(avcodec:new): FFmpge crash when use VideoToolBox to encode frames
by FFmpeg 09 May '21

09 May '21

#8769: FFmpge crash when use VideoToolBox to encode frames ---------------------------------+-------------------------------------- Reporter: yxibng | Type: defect Status: new | Priority: critical Component: avcodec | Version: git-master Keywords: crash | Blocked By: Blocking: | Reproduced by developer: 1 Analyzed by developer: 1 | ---------------------------------+-------------------------------------- Summary of the bug: How to reproduce: {{{ static void vtenc_output_callback( void *ctx, void *sourceFrameCtx, OSStatus status, VTEncodeInfoFlags flags, CMSampleBufferRef sample_buffer) { AVCodecContext *avctx = ctx; VTEncContext *vtctx = avctx->priv_data; ExtraSEI *sei = sourceFrameCtx; if (vtctx->async_error) { if(sample_buffer) CFRelease(sample_buffer); //This line cause crash, should remove return; } if (status) { av_log(avctx, AV_LOG_ERROR, "Error encoding frame: %d\n", (int)status); set_async_error(vtctx, AVERROR_EXTERNAL); return; } if (!sample_buffer) { return; } if (!avctx->extradata && (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER)) { int set_status = set_extradata(avctx, sample_buffer); if (set_status) { set_async_error(vtctx, set_status); return; } } vtenc_q_push(vtctx, sample_buffer, sei); } }}} Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker. -- Ticket URL: <https://trac.ffmpeg.org/ticket/8769> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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