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

#9153(undetermined:new): When specifying -acodec nellymoser ffmpeg does not recognize stream specifier
by FFmpeg 20 Mar '21

20 Mar '21

#9153: When specifying -acodec nellymoser ffmpeg does not recognize stream specifier --------------------------------------+---------------------------------- Reporter: haraldbre | Type: defect Status: new | Priority: normal Component: undetermined | Version: 4.2 Keywords: acodec | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | --------------------------------------+---------------------------------- Summary of the bug: Concatenating nellymoser encoded audio streams to .mp3 usually works when setting input codecs via "-acodex nellymoser". In some instances however it fails: Stream specifier ':3' in filtergraph description [0:0][0:2][0:3] concat=n=3:v=0:a=1 matches no streams. When setting input codecs individually via "-codec:a:0 nellymoser -codec:a:1 nellymoser .." there is no problem. How to reproduce: ffmpeg -acodec nellymoser -i http://www.hwjyw.com/textbook/dzjc/zh/cd12/main11/content/CzdB_001.swf -filter_complex "[0:0][0:2][0:3] concat=n=3:v=0:a=1" temp.mp3 ==> [swf @ 0x5620fa937240] SWF compressed file detected Input #0, swf, from 'http://www.hwjyw.com/textbook/dzjc/zh/cd12/main11/content/CzdB_001.swf': Duration: 00:00:02.74, bitrate: 3137 kb/s Stream #0:0: Audio: nellymoser, 44100 Hz, mono, flt Stream #0:1: Video: rawvideo (ARGB / 0x42475241), argb, 149x119, 25 fps, 25 tbr, 25 tbn Stream #0:2: Audio: nellymoser, 44100 Hz, mono, flt Stream specifier ':3' in filtergraph description [0:0][0:2][0:3] concat=n=3:v=0:a=1 matches no streams. ffmpeg -codec:a:0 nellymoser -codec:a:1 nellymoser -codec:a:2 nellymoser -i http://www.hwjyw.com/textbook/dzjc/zh/cd12/main11/content/CzdB_001.swf -filter_complex "[0:0][0:2][0:3] concat=n=3:v=0:a=1" temp.mp3 ==> [swf @ 0x556d10114240] SWF compressed file detected [swf @ 0x556d10114240] Could not find codec parameters for stream 0 (Audio: none, 44100 Hz, mono): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options [swf @ 0x556d10114240] Could not find codec parameters for stream 2 (Audio: none, 44100 Hz, mono): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options [swf @ 0x556d10114240] Could not find codec parameters for stream 3 (Audio: none, 44100 Hz, mono): unknown codec Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, swf, from 'http://www.hwjyw.com/textbook/dzjc/zh/cd12/main11/content/CzdB_001.swf': Duration: 00:00:04.31, bitrate: 1996 kb/s Stream #0:0: Audio: nellymoser, 44100 Hz, mono Stream #0:1: Video: rawvideo (ARGB / 0x42475241), argb, 149x119, 25 fps, 25 tbr, 25 tbn Stream #0:2: Audio: nellymoser, 44100 Hz, mono Stream #0:3: Audio: nellymoser, 44100 Hz, mono Stream mapping: Stream #0:0 (nellymoser) -> concat:in0:a0 Stream #0:2 (nellymoser) -> concat:in1:a0 Stream #0:3 (nellymoser) -> concat:in2:a0 concat -> Stream #0:0 (libmp3lame) ... -- Ticket URL: <https://trac.ffmpeg.org/ticket/9153> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#7919(ffmpeg:new): Live extracting subtitles from the stream
by FFmpeg 20 Mar '21

20 Mar '21

#7919: Live extracting subtitles from the stream -------------------------------------+------------------------------------- Reporter: | Type: defect MishkaPetran | Status: new | Priority: normal Component: ffmpeg | Version: 4.1 Keywords: ffmpeg, | Blocked By: subtitle, live, stream, m3u8, | webvtt, vtt | Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- '''Good day''' Experiencing problems when creating streaming playback. I get the input file in parts (from webtorrent), in the process of downloading it I will convert it using ffmpeg and give it to the user in the live mode. I can switch audio and video streams. There is a problem with the subtitles. '''I use the following options:''' {{{ ffmpeg -i '/home/mishkapetran/Загрузки/Rick.mkv' \ -map 0:v:0 -c:v libx264 -profile:v baseline -preset:v superfast -b:v 1000k -level 3.0 -strict -2 -s 480x360 -f hls -hls_time 10 -hls_list_size 0 -segment_list rick_360p -hls_segment_filename /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_360p_video_%d.ts /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_360p_video.m3u8 \ -map 0:v:0 -c:v libx264 -profile:v baseline -preset:v superfast -b:v 1000k -level 3.0 -strict -2 -s 858x480 -f hls -hls_time 10 -hls_list_size 0 -segment_list rick_480p -hls_segment_filename /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_480p_video_%d.ts /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_480p_video.m3u8 \ -map 0:v:0 -c:v libx264 -profile:v baseline -preset:v superfast -b:v 1000k -level 3.0 -strict -2 -s 1280x720 -f hls -hls_time 10 -hls_list_size 0 -segment_list rick_720p -hls_segment_filename /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_720p_video_%d.ts /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_720p_video.m3u8 \ -map 0:1 -c:a aac -b:a 128k -ar 44100 -ac 2 -f hls -hls_time 10 -hls_list_size 0 -segment_list rick_audio_rus -hls_segment_filename /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_rus_audio_%d.aac /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_rus_audio.m3u8 \ -map 0:2 -c:a aac -b:a 128k -ar 44100 -ac 2 -f hls -hls_time 10 -hls_list_size 0 -segment_list rick_audio_eng -hls_segment_filename /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_eng_audio_%d.aac /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_eng_audio.m3u8 }}} The average conversion speed on my local PC is about 2.5x (without subtitles). As soon as I add subtitles to the options, the speed drops to 0.5x, and this is not acceptable for streaming. '''My options for subtitles:''' {{{ -map 0:3 -f segment -segment_time 10 -segment_format webvtt /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_sub_eng-%d.vtt /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_sub_eng.m3u8 \ -map 0:4 -f segment -segment_time 10 -segment_format webvtt /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_sub_rus-%d.vtt /home/mishkapetran/m3u8_test/hls_test/video_stream_source/rick_sub_rus.m3u8 }}} '''I noticed:''' 1) If you convert to .webvtt, then .ts video elements are created along with the subtitle files. 2) If I use just .vtt, without fragments, then writing to the file occurs only after the file is completely processed. But for the .m3u8 playlist, the .vtt format is not suitable, so it will also stick to processing in parallel. Tell me how you can implement the extraction of subtitles in my case. '''My ffmpeg:''' {{{ ffmpeg version 4.1.3-0york1~18.04 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04) configuration: --prefix=/usr --extra-version='0york1~18.04' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable- libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable- libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable- libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable- libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-nonfree --enable-libfdk-aac --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared libavutil 56. 22.100 / 56. 22.100 libavcodec 58. 35.100 / 58. 35.100 libavformat 58. 20.100 / 58. 20.100 libavdevice 58. 5.100 / 58. 5.100 libavfilter 7. 40.101 / 7. 40.101 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 3.100 / 5. 3.100 libswresample 3. 3.100 / 3. 3.100 libpostproc 55. 3.100 / 55. 3.100 Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}... }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/7919> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9151(undetermined:new): Missing white space in the white list of tesseract configuration
by FFmpeg 19 Mar '21

19 Mar '21

#9151: Missing white space in the white list of tesseract configuration -------------------------------------+------------------------------------- Reporter: dominic108 | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: tesseract | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: I compiled ffmpeg on Ubuntu to have the tesseract module: {{{ ffmpeg version N-101412-gb7e7813 Copyright (c) 2000-2021 the FFmpeg developers built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04) configuration: --prefix=/home/working/app_download/ffmpeg_build --pkg- config-flags=--static --extra- cflags=-I/home/working/app_download/ffmpeg_build/include --extra- ldflags=-L/home/working/app_download/ffmpeg_build/lib --extra- libs='-lpthread -lm' --ld=g++ --bindir=/home/working/app_download/ffmpeg_build/bin --enable-gpl --enable-gnutls --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable- libsvtav1 --enable-libdav1d --enable-libvorbis --enable-libvpx --enable- libx264 --enable-libx265 --enable-nonfree --enable-libtesseract }}} I tested with the command line: {{{ % ffmpeg -i input -vf "ocr,metadata=mode=print:file=ocr.txt:direct=1" output }}} Here is an extract from ocr.txt after the above command: {{{ frame:53 pts:212752 pts_time:212.752 lavfi.ocr.text=Transcendingisthedeepsettlingoftheactivityofthemind whilethemindremainsawake. lavfi.ocr.confidence=0 0 95 }}} The white spaces were not recognized by Tesseract and I checked that Tesseract perfectly recognized the white spaces when directly applied on the frame image. So, I checked the code libavfilter/vf_ocr.c to see what is going on. The white space was not in the white list of characters. Even though it is a strange idea to consider a white space as a character in the context of ocr, I boldly added a white space to the list. The original code was : {{{ { "whitelist", "set character whitelist", OFFSET(whitelist), AV_OPT_TYPE_STRING, {.str="0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ.:;,-+_!?\"'[]{}()<>|/\\=*&%$#@!~"}, 0, 0, FLAGS }, }}} The modified code was {{{ { "whitelist", "set character whitelist", OFFSET(whitelist), AV_OPT_TYPE_STRING, {.str="0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ.:;,-+_!?\"'[]{}()<>|/\\=*&%$#@!~ "}, 0, 0, FLAGS }, }}} After recompilation I tried again and here was result: {{{ frame:53 pts:212752 pts_time:212.752 lavfi.ocr.text=Transcending is the deep settling of the activity of the mind while the mind remains awake. lavfi.ocr.confidence=96 96 97 96 96 96 96 96 96 96 96 96 96 96 96 96 95 }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/9151> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9156(avutil:new): Option flag usage
by FFmpeg 19 Mar '21

19 Mar '21

#9156: Option flag usage --------------------------------+--------------------------------------- Reporter: sceedy | Type: defect Status: new | Priority: minor Component: avutil | Version: unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | --------------------------------+--------------------------------------- I have either badly misunderstood or I have found a fault. I am using the public API. I have looked at the AVOption referenced by several examples of AVClass with particular interest in { x , x , x , AV_OPT_TYPE_CONST, x , x , x , ENC, x } (where x = don’t care) as this is how I believe options for bit encoded flags are stored. In all the cases I have looked at the flag settings are not mutually exclusive e.g. class "mpegts_muxer_class" flags "latm" and "system_b". I tried code like Options = NULL; av_dict_set(&Options,"mpegts_flags","system_b",0); av_dict_set(&Options,"mpegts_flags","latm", AV_DICT_DONT_OVERWRITE); ReturnCode = avformat_write_header(OutputContext,&Options); All that happened was the last flag setting in the list won. I then looked at opt.c and found flags are set in function "write_number" ... case AV_OPT_TYPE_BOOL: case AV_OPT_TYPE_FLAGS: case AV_OPT_TYPE_INT: *(int *)dst = llrint(num / den) * intnum; break; ... For flags to work the way I expect, i.e. you can "or" multiple values, I believe this should read ... case AV_OPT_TYPE_BOOL: case AV_OPT_TYPE_INT: *(int *)dst = llrint(num / den) * intnum; break; case AV_OPT_TYPE_FLAGS: *(int *)dst |= llrint(num / den) * intnum; break; ... As a work around I use av_opt_ptr to get the address of the *flags* variable and av_dict_get to get the values, "or" them together myself and then write them to the address using the pointer. So the question I wish answered is have I fundamentally misunderstood the way bit flags are set in options or have I found a problem? -- Ticket URL: <https://trac.ffmpeg.org/ticket/9156> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#7745(avformat:new): RIST - Reliable Internet Stream Transport Protocol
by FFmpeg 17 Mar '21

17 Mar '21

#7745: RIST - Reliable Internet Stream Transport Protocol -------------------------------------+------------------------------------- Reporter: mehmetb | Type: defect Status: new | Priority: wish Component: avformat | Version: git- Keywords: rist | master reliable internet stream | Blocked By: transport protocol | Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Quoting directly from the Video Services Forum Technical Recommendation TR-06-01: The Reliable Internet Stream Transport (RIST) project was initiated as an Activity Group under the auspices of the Video Services Forum in 2017. Many solutions exist in the market for reliable streaming over the Internet. These solutions all use the same types of techniques, but they are all proprietary and do not interoperate. This Technical Recommendation contains a protocol specification for reliable streaming over the Internet, so end users can mix and match solutions from different vendors. Document URL: [http://vsf.tv/download/technical_recommendations/VSF_TR-06-1_2018_10_17.pdf] I hope ffmpeg will support this protocol. -- Ticket URL: <https://trac.ffmpeg.org/ticket/7745> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9149(swscale:new): bugs in Altivec code
by FFmpeg 17 Mar '21

17 Mar '21

#9149: bugs in Altivec code -------------------------------------+------------------------------------- Reporter: | Type: defect ilyakurdyukov | Status: new | Priority: normal Component: swscale | Version: | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Is the Altivec code still in use? I am making a SIMD optimization patch for the Elbrus architecture (e2k). It's easier to port from existing code with intrinsic functions, so I'm rewriting the Altivec code to Elbrus. When doing this, I noticed three places in swscale that are definitely wrong in Altivec/VSX (compared to the C reference code). I can't check that for sure because I don't have Altivec hardware. I check swscale with: {{{ $ libswscale/tests/swscale -cpuflags 0 2> /dev/null > ../swscale_ref.log $ libswscale/tests/swscale 2> /dev/null > ../swscale.log $ diff -u swscale_ref.log swscale.log > swscale.diff }}} It's pretty slow. After completing the tests, "swscale.diff" shows the problems. Some yuv2rgb conversions are more accurate, it's okay. But check the lines with big SSD values, this is what is not working correctly. == libswscale/ppc/swscale_vsx.c:2110 Here's should be added "dstFormat != AV_PIX_FMT_P010LE && dstFormat != AV_PIX_FMT_P010BE", because this code doesn't handle these formats. {{{ if (!(c->flags & (SWS_BITEXACT | SWS_FULL_CHR_H_INT)) && !c->needAlpha) { switch (c->dstBpc) { case 8: c->yuv2plane1 = yuv2plane1_8_vsx; break; }}} == libswscale/ppc/swscale_vsx.c:1060 In yuv2rgb_full_1_vsx_template() there is a code: {{{ const vec_s16 mul8 = vec_splat_s16(8); ... vu32_l = vec_mule(vu, mul8); vu32_r = vec_mulo(vu, mul8); vv32_l = vec_mule(vv, mul8); vv32_r = vec_mulo(vv, mul8); }}} But should be: {{{ const vec_s16 mul2 = vec_splat_s16(2); ... tmp32 = vec_mule(vu, mul2); tmp32_2 = vec_mulo(vu, mul2); vu32_l = vec_mergeh(tmp32, tmp32_2); vu32_r = vec_mergel(tmp32, tmp32_2); tmp32 = vec_mule(vv, mul2); tmp32_2 = vec_mulo(vv, mul2); vv32_l = vec_mergeh(tmp32, tmp32_2); vv32_r = vec_mergel(tmp32, tmp32_2); }}} Actually, this code may be written a lot better, by multiplying coef multipliers before the for loop. And using vec_unpack instead of vec_mul/vec_merge. == libswscale/ppc/swscale_vsx.c:1513 In yuv2422_2_vsx_template() there is a wrong macro SETUP, that multiplies by (4096 - alpha) and (4096 - alpha), should be (4096 - alpha) and (alpha). You can see yuv2rgb_full_2_vsx_template() for the correct code. -- Ticket URL: <https://trac.ffmpeg.org/ticket/9149> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9148(undetermined:new): Downloading an https file with gnutls is too slow
by FFmpeg 17 Mar '21

17 Mar '21

#9148: Downloading an https file with gnutls is too slow -------------------------------------+------------------------------------- Reporter: tanersener | Type: defect Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: gnutls, | Blocked By: https | Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: Downloading a file using https protocol is too slow. Download times out frequently. This happens when ffmpeg is built with gnutls on macOS. This performance decrease is caused by https://git.ffmpeg.org/gitweb/ffmpeg.git/commit/70d8077b795766e2486e6ec8110…. If you checkout master using a previous commit e.g. a1553b0cfbff92223caaed4103b92373b2339173 the same file is downloaded in 35 seconds. See attachments provided. How to reproduce: {{{ % ffmpeg -y -report -i https://snapcellvideos.s3.amazonaws.com/manual/24/manualupload_160985869629… -c:v mpeg4 /tmp/latest.mp4 ffmpeg version N-101485-g16e8ea0dde Copyright (c) 2000-2021 the FFmpeg developers built with Apple clang version 12.0.0 (clang-1200.0.32.29) configuration: --enable-gnutls libavutil 56. 68.100 / 56. 68.100 libavcodec 58.130.100 / 58.130.100 libavformat 58. 72.100 / 58. 72.100 libavdevice 58. 12.100 / 58. 12.100 libavfilter 7.109.100 / 7.109.100 libswscale 5. 8.100 / 5. 8.100 libswresample 3. 8.100 / 3. 8.100 }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/9148> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9147(undetermined:new): ffmpeg gets into infinite loop with some audio chain
by FFmpeg 16 Mar '21

16 Mar '21

#9147: ffmpeg gets into infinite loop with some audio chain -------------------------------------+------------------------------------- Reporter: bakaiadam | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: If you have a short audio and try to get only the non silent parts then amplify the non silent part, it just hangs. If the duration is 1.5 seconds for example, it works just fine. How to reproduce: {{{ % ffmpeg -f lavfi -i "sine=frequency=1000:duration=0.5" -y test.wav ; ffmpeg -i test.wav -af 'silenceremove=stop_periods=-1:stop_duration=0.5:stop_threshold=-70dB,loudnorm,dynaudnorm=g=3:f=200' -y b.flac ffmpeg version N-56320-ge937457b7b-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the FFmpeg developers built with gcc 8 (Debian 8.3.0-6) configuration: --enable-gpl --enable-version3 --enable-static --disable- debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg }}} Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker. -- Ticket URL: <https://trac.ffmpeg.org/ticket/9147> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9154(undetermined:new): SVQ1 fourcc is wrong for AVI container
by FFmpeg 16 Mar '21

16 Mar '21

#9154: SVQ1 fourcc is wrong for AVI container -------------------------------------+------------------------------------- Reporter: cecodere | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: SVQ1, | Blocked By: Sorenson | Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: When using the svq1 codec and using AVI as output container, the fourcc output may be invalid, as it is in lower case while in Qt it is in upper case, so it cannot be recognized by ffdshow. VLC does accept both svq1 and SVQ1 as AVI fourcc. How to reproduce: {{{ % ffmpeg -y -benchmark -i foreman_cif.avi -codec:v svq1 foreman_cif_svq1_ffmpeg.avi 2>foreman_cif_svq1_ffmpeg.avi.txt ffmpeg version : ffmpeg-4.4-2460-2c6f532-win32-static-xpmod-sse.7z built on 2021-01-02 }}} Workaround: {{{ % ffmpeg -y -benchmark -i foreman_cif.avi -codec:v svq1 -vtag SVQ1 foreman_cif_svq1_ffmpeg2.avi }}} See attached mediainfo exports. -- Ticket URL: <https://trac.ffmpeg.org/ticket/9154> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#8911(avfilter:new): showspectrumpic innacurate zoom of resulting image with stop and size options
by FFmpeg 13 Mar '21

13 Mar '21

#8911: showspectrumpic innacurate zoom of resulting image with stop and size options -------------------------------------+------------------------------------- Reporter: semiaddict | Type: defect Status: new | Priority: normal Component: avfilter | Version: git- Keywords: | master showspectrumpic stop zoom | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- '''Summary of the bug:''' When using the [https://ffmpeg.org/ffmpeg-filters.html#showspectrumpic showspectrumpic] filter with a value for the ''stop'' option along with a ''size'', the resulting image seems to always stop at a much lower frequency than the one specified. The legend on the top right corner (Zoom: ...) is, however, correct. This doesn't seem to be the case if the ''size'' option is not specified, and the issue seems to occur whether or not the ''legend'' option is active. ''' How to reproduce:''' {{{ ffmpeg -i http://commondatastorage.googleapis.com/gtv-videos- bucket/sample/BigBuckBunny.mp4 -filter_complex showspectrumpic=stop=5000:size=1024x768:legend=1 output.png }}} ffmpeg version 4.3 built on commit a3a6b56 This was tested on multiple files including http://commondatastorage.googleapis.com/gtv-videos- bucket/sample/BigBuckBunny.mp4 -- Ticket URL: <https://trac.ffmpeg.org/ticket/8911> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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