May 2024 - FFmpeg-trac - lists.ffmpeg.org

#5413(ffmpeg:new): Image size, pixel format cannot be detected ( 0x0, -1 ) for short h264 video
by FFmpeg 02 Aug '25

02 Aug '25

#5413: Image size, pixel format cannot be detected ( 0x0,-1 ) for short h264 video -------------------------------------+------------------------------------- Reporter: sadensmol | Type: defect Status: new | Priority: normal Component: ffmpeg | Version: Keywords: image size | unspecified 0x0 | Blocked By: Blocking: | Reproduced by developer: 1 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Cannot convert video which plays fine with Windows Media Player. ffmpeg parameters ( splitted by comma) -probesize,100M,-analyzeduration,100M,-i,00443.MTS,-an,-y,-c:v,mjpeg,-q:v,8,frame%d.jpg ffmpeg output: [mpegts @ 0x7ffcd9808800] Could not find codec parameters for stream 0 (Video: h264 (HDMV / 0x564D4448), none): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from '/Users/saden/Documents/asimon/simplevideoed/test/folder8/00443.MTS': Duration: 00:00:00.51, start: 1.033367, bitrate: 9216 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (HDMV / 0x564D4448), none, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090), 1920x1080 [buffer @ 0x7ffcd96013c0] Unable to parse option value "0x0" as image size [buffer @ 0x7ffcd96013c0] Unable to parse option value "-1" as pixel format [buffer @ 0x7ffcd96013c0] Unable to parse option value "0x0" as image size [buffer @ 0x7ffcd96013c0] Error setting option video_size to value 0x0. [graph 0 input from stream 0:0 @ 0x7ffcd9601500] Error applying options to the filter. This problem is easily reproducible with the latest version of ffmpeg on both OSX and windows systems. At the same time exiftool returns correct values for size. -- Ticket URL: <https://trac.ffmpeg.org/ticket/5413> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#10990(undetermined:new): "Assertion best_input >= 0 failed" IFF upsampling AND complex filtergraph
by FFmpeg 02 Aug '25

02 Aug '25

#10990: "Assertion best_input >= 0 failed" IFF upsampling AND complex filtergraph -------------------------------------+------------------------------------- Reporter: Hunter | Type: defect Hogan | Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: If resampler is invoked at the start of the filtergraph to upsample, an error will be thrown at the very end despite the output file being successfully written. `Assertion best_input >= 0 failed at src/fftools/ffmpeg_filter.c:2075` How to reproduce: I reproduced it with a few different wav files, including a white noise file. All were stereo, however. Batch file and 22 logs attached. Only 4 logs have the error. The other logs are to compare changing only one variable but not getting an error. These quick tests all have filtergraphs loaded from a textfile, and I don't think I tested it without a text file. These tests all use `-/complex_filter` syntax, but I got the same error with the old syntax. I did not test downsampling at the start of the filtergraph. I first encountered this a couple of weeks ago after building version 7. Prior to that, however, I don't think I ever had a filtergraph that included upsampling and multiple input and/or output pads, so I am not claiming the error was recently introduced. If I recall correctly, originally, FFmpeg would crash, throw the error, and not write a usable file. IIRC, the output file was the size I expected (in bytes), but nothing could read it: ffprobe, ffmpeg, Goldwave all said the file was garbage. I interpreted it to mean that the "headers" were malformed. (By "headers", I probably mean metadata, but I don't know the precise term here.) I worked around it by scripting: 1. upsample my input file to a temp output file 2. filter as I want, output to a new temp 3. downsample IIRC, it was necessary for me to downsample in a different filtergraph. But, it now looks like downsampling doesn't throw an error. Today, I rebuilt, and noticed the differences. {{{ % FFmpeg -hide_banner -loglevel warning -stats -i test_input_44100_2ch.wav -/filter_complex test_filter.txt -map "[Z0Z_out]" -codec:a pcm_f32le "output\\test_input_44100_2ch.wav_filtergraph_192000.wav" }}} ffmpeg version: media autobuilder suite, built today on Windows {{{ ffmpeg version N-115035-g192f5b58b7-gb7debef3cc+4 Copyright (c) 2000-2024 the FFmpeg developers built with gcc 13.2.0 (Rev6, Built by MSYS2 project) configuration: --pkg-config=pkgconf --cc='ccache gcc' --cxx='ccache g++' --ld='ccache g++' --extra-cxxflags=-fpermissive --extra-cflags=-Wno-int- conversion --disable-autodetect --enable-amf --enable-bzlib --enable- d3d11va --enable-dxva2 --enable-iconv --enable-lzma --enable-zlib --enable-sdl2 --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libdav1d --enable-libaom --disable-debug --enable-libfdk-aac --enable-fontconfig --enable-libass --enable-libfreetype --enable-libmysofa --enable- libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable- libwebp --enable-libxml2 --enable-libzimg --enable-libshine --enable-gpl --enable-avisynth --enable-libxvid --enable-libopenmpt --enable-version3 --enable-librav1e --enable-libgsm --enable-libvmaf --enable-libsvtav1 --enable-chromaprint --enable-frei0r --enable-libaribb24 --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfribidi --enable-libgme --enable-libilbc --enable-libsvthevc --enable-libsvtvp9 --enable-libkvazaar --enable-libmodplug --enable-librtmp --enable- librubberband --enable-libtesseract --enable-libxavs --enable-libzmq --enable-libzvbi --enable-openal --enable-libcodec2 --enable-ladspa --enable-libglslang --enable-vulkan --enable-libdavs2 --enable-libxavs2 --enable-libuavs3d --enable-opengl --enable-libopenh264 --enable- libharfbuzz --enable-libfontconfig --enable-openssl --extra- cflags=-DLIBTWOLAME_STATIC --extra-cflags=-DCACA_STATIC --extra- cflags=-DMODPLUG_STATIC --extra-cflags=-DCHROMAPRINT_NODLL --extra- cflags=-DZMQ_STATIC --extra-libs=-lpsapi --extra-cflags=-DLIBXML_STATIC --extra-libs=-liconv --disable-w32threads --extra-cflags=-DKVZ_STATIC_LIB --enable-nonfree --extra-cflags=-DAL_LIBTYPE_STATIC --extra- cflags='-IC:/apps/media-autobuild_suite/local64/include' --extra- cflags='-IC:/apps/media-autobuild_suite/local64/include/AL' libavutil 59. 16.101 / 59. 16.101 libavcodec 61. 5.103 / 61. 5.103 libavformat 61. 3.103 / 61. 3.103 libavdevice 61. 2.100 / 61. 2.100 libavfilter 10. 2.101 / 10. 2.101 libswscale 8. 2.100 / 8. 2.100 libswresample 5. 2.100 / 5. 2.100 libpostproc 58. 2.100 / 58. 2.100 }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/10990> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#10988(undetermined:new): Acrossover crash
by FFmpeg 02 Aug '25

02 Aug '25

#10988: Acrossover crash -------------------------------------+------------------------------------- Reporter: JamesBond | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Setup: Windows 7 x64, FFMPEG g08781ebe1-2024-04-29 x64 Source: song.dsf https://fastupload.io/en/8hp1Dg1K3KNY3Zc/file Command: ffmpeg -i song.dsf -af "volume=6dB,aresample=96000:resampler=soxr:precision=28:osf=s32:dither_method=triangular_hp,acrossover=24000:order=8th[low][high];[low]anull;[high]anullsink" out.flac When the encoding is almost done, the red line appears. It says “Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:2075”. Then I see a GUI window that says “ffmpeg.exe has stopped working” with a button “Close the program”. An incomplete FLAC file remains on the disk, i.e. flac -t song.flac says “FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC” and “ FLAC__STREAM_DECODER_END_OF_STREAM”. If I remove that acrossover filter, then the encoding goes fine. -- Ticket URL: <https://trac.ffmpeg.org/ticket/10988> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9195(avfilter:new): vibrance (video filter) incorrect defaults for luma
by FFmpeg 31 Jul '25

31 Jul '25

#9195: vibrance (video filter) incorrect defaults for luma -------------------------------------+------------------------------------- Reporter: Jim | Type: defect DeLaHunt | Status: new | Priority: normal Component: avfilter | Version: git- | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: The "vibrance" video filter appears to have incorrect default values for the luma coefficient parameters "rlum" and "blum". They appear to be swapped. How to reproduce: {{{ % ffmpeg -h filter=vibrance ffmpeg version 4.4 Copyright (c) 2000-2021 the FFmpeg developers built with Apple LLVM version 10.0.0 (clang-1000.10.44.4) configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable- libopus --enable-librsvg --enable-libtheora --enable-libopenjpeg --enable- libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex --enable- libass --enable-libbluray --enable-lzma --enable-gnutls --enable- fontconfig --enable-libfreetype --enable-libfribidi --disable-libjack --disable-libopencore-amrnb --disable-libopencore-amrwb --disable- indev=jack --enable-opencl --disable-outdev=xv --enable-audiotoolbox --enable-videotoolbox --enable-sdl2 --disable-securetransport --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --enable-libdav1d --arch=x86_64 --enable-x86asm --enable-libx265 --enable-gpl --enable-postproc --enable-libx264 --enable- libxvid 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 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 9.100 / 5. 9.100 libswresample 3. 9.100 / 3. 9.100 libpostproc 55. 9.100 / 55. 9.100 Filter vibrance Boost or alter saturation. slice threading supported Inputs: #0: default (video) Outputs: #0: default (video) vibrance AVOptions: intensity <float> ..FV.....T. set the intensity value (from -2 to 2) (default 0) rbal <float> ..FV.....T. set the red balance value (from -10 to 10) (default 1) gbal <float> ..FV.....T. set the green balance value (from -10 to 10) (default 1) bbal <float> ..FV.....T. set the blue balance value (from -10 to 10) (default 1) rlum <float> ..FV.....T. set the red luma coefficient (from 0 to 1) (default 0.072186) glum <float> ..FV.....T. set the green luma coefficient (from 0 to 1) (default 0.715158) blum <float> ..FV.....T. set the blue luma coefficient (from 0 to 1) (default 0.212656) alternate <boolean> ..FV.....T. use alternate colors (default false) This filter has support for timeline through the 'enable' option. }}} The vibrance filter is documented at https://ffmpeg.org/ffmpeg- filters.html#vibrance . It looks like coefficients "rlum", "glum", and "blum" are supposed to match the luma computation as defined in BT.709-6 (https://en.wikipedia.org/wiki/Rec._709) or something similar. Wikipedia gives this formula as: > R’G’B’ coefficients 0.2126, 0.7152, and 0.0722 (together they add to 1) So, `luma = 0.2126 * r + 0.7152 * g + 0.0722 * b`, a value in [0.0, 1.0] for r, g, b in [0.0, 1.0] But the default values I see in the FFmpeg vibrance filter are (per output above): > rlum (default 0.072186), glum (default 0.715158), blum (default 0.212656) Thus it looks like the default values for rlum and blum are swapped. Vibrance would still compute a luma a value in a range of [0.0, 1.0], but it wouldn't be the correct value per BT.709. Since the vibrance adjustments are relative to luma, they might be slightly off. I believe the fix would be the simple, obvious exchange of default values in ffmpeg/libavfilter/vf_vibrance.c:372-374. But I haven't tried this. Present code: {{{ { "rlum", "set the red luma coefficient", OFFSET(lcoeffs[2]), AV_OPT_TYPE_FLOAT, {.dbl=0.072186}, 0, 1, VF }, { "glum", "set the green luma coefficient", OFFSET(lcoeffs[0]), AV_OPT_TYPE_FLOAT, {.dbl=0.715158}, 0, 1, VF }, { "blum", "set the blue luma coefficient", OFFSET(lcoeffs[1]), AV_OPT_TYPE_FLOAT, {.dbl=0.212656}, 0, 1, VF }, }}} Code with possible fix: {{{ { "rlum", "set the red luma coefficient", OFFSET(lcoeffs[2]), AV_OPT_TYPE_FLOAT, {.dbl=0.212656}, 0, 1, VF }, { "glum", "set the green luma coefficient", OFFSET(lcoeffs[0]), AV_OPT_TYPE_FLOAT, {.dbl=0.715158}, 0, 1, VF }, { "blum", "set the blue luma coefficient", OFFSET(lcoeffs[1]), AV_OPT_TYPE_FLOAT, {.dbl=0.072186}, 0, 1, VF }, }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/9195> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#10209(undetermined:new): EBUR128 and LOUDNORM filters compute LRA differently
by FFmpeg 25 Jul '25

25 Jul '25

#10209: EBUR128 and LOUDNORM filters compute LRA differently --------------------------------------+---------------------------------- Reporter: hsn | Type: defect Status: new | Priority: normal Component: undetermined | Version: 5.1.2 Keywords: LRA, ebur128 | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | --------------------------------------+---------------------------------- Summary of the bug: How to reproduce: {{{ ffmpeg -i song192k.mp4 -af ebur128,loudnorm=print_format=summary -f null - }}} Outputs: {{{ [Parsed_ebur128_0 @ 000001f413d21a40] Summary: Integrated loudness: I: -14.0 LUFS Threshold: -24.1 LUFS Loudness range: LRA: 3.6 LU Threshold: -33.9 LUFS LRA low: -15.6 LUFS LRA high: -12.0 LUFS [Parsed_loudnorm_1 @ 000001f413fc4e40] Input Integrated: -14.2 LUFS Input True Peak: +0.1 dBTP Input LRA: 5.9 LU Input Threshold: -24.3 LUFS }}} One LRA is ''5.9'' and second is ''3.6''. Song analysed by https://youlean.co/file-loudness-meter/ returns: {{{ Momentary Max = -9.81 LUFS Short Term Max = -11.50 LUFS Integrated = -14.00 LUFS }}} LRA definition is here: https://tech.ebu.ch/docs/tech/tech3342v2_0.pdf -- Ticket URL: <https://trac.ffmpeg.org/ticket/10209> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#8349(avcodec:new): Dolby AC-4 Support
by FFmpeg 21 Jul '25

21 Jul '25

#8349: Dolby AC-4 Support ----------------------------------+--------------------------------------- Reporter: Nomis101 | Type: enhancement Status: new | Priority: normal Component: avcodec | Version: git-master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+--------------------------------------- It would be nice to support the detection, muxing and demuxing of Dolby AC-4 audio streams (in MP4 and MKV). Information: https://dolbyac4.com/ Specs: https://www.dolby.com/us/en/technologies/ac-4/Next-Generation- Entertainment-Services.pdf Sample files: http://testassets.dashif.org/#testcase/details/586fb3879ae9045678eab64f MediaInfo 19.09 added support for AC-4, so it would be nice to have support in FFmpeg as well. https://mediaarea.net/MediaInfo/ChangeLog -- Ticket URL: <https://trac.ffmpeg.org/ticket/8349> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5970(undetermined:new): No valid multicast port obtained from RTSP
by FFmpeg 11 Jul '25

11 Jul '25

#5970: No valid multicast port obtained from RTSP -------------------------------------+------------------------------------- Reporter: ioeir2 | 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: Cannot get multicast RTSP stream from camera via ffmpeg/ffplay. ffmpeg/ffplay tries to connect to UDP port 0 instead of 40000. It sends in SETUP: Transport: RTP/AVP/UDP;multicast But should something like Transport: RTP/AVP;multicast;destination=224.1.2.4;port=40000-40001 How to reproduce: IP camera: Amcrest IPM-HX1, sw. version 2.420.AC00.15.R, build : 2016-09-08. Very popular at the moment IP camera. {{{ ./ffplay-3.2 -rtsp_transport udp_multicast 'rtsp://admin:password@10.0.1.20/cam/realmonitor?channel=1&subtype=0&proto=Dahua3&unicast=false' ffplay version 3.2 Copyright (c) 2003-2016 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: libavutil 55. 34.100 / 55. 34.100 libavcodec 57. 64.100 / 57. 64.100 libavformat 57. 56.100 / 57. 56.100 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 [udp @ 0x7fb20bc9d500] bind failed: Permission denied 0B f=0/0 [udp @ 0x7fb20bc9b1a0] bind failed: Permission denied [udp @ 0x7fb20bc9b220] bind failed: Permission denied Segmentation fault: 11 }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/5970> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9846(ffmpeg:new): DNxHR doesn't work with LB in Shutter Encoder
by FFmpeg 10 Jul '25

10 Jul '25

#9846: DNxHR doesn't work with LB in Shutter Encoder -------------------------------------+------------------------------------- Reporter: Soto Soto | Type: defect Status: new | Priority: normal Component: ffmpeg | Version: Keywords: ffmpeg | unspecified dnxhr | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- '''Summary of the bug:''' I've just been forwarded by the creator of Shutter Encoder who uses FFmpeg as the backend of their software. I wanted to create a small 480x270 file at low bitrate and the render stops after a few minutes. The last line the console says is "Assertion s->buf_ptr < s->buf_end failed at src/libavcodec/put_bits.h:150". How to reproduce: 1. Used a 1920x1080p, 60fps, h.264 10gb file. 2. Dropped it in Shutter Encoder, set to DNxHR > LB ("Low Bitrate" doesn't state the number of the bitrate itself). Change scale to 480x270p 3. Render 4. Error shown in the last line of the console (right click in progress bar and click "Console"). -- Ticket URL: <https://trac.ffmpeg.org/ticket/9846> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#10732(undetermined:new): avcodec_flush_buffers() not resetting E-AC-3 decoder
by FFmpeg 08 Jul '25

08 Jul '25

#10732: avcodec_flush_buffers() not resetting E-AC-3 decoder ----------------------------------------+---------------------------------- Reporter: Peter Krefting | Type: defect Status: new | Priority: minor Component: undetermined | Version: 5.0.3 Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------------+---------------------------------- Summary of the bug: I am reusing the same AVCodecContext to decode several parallel E-AC-3 audio streams. As per documentation I am calling [https://ffmpeg.org/doxygen/trunk/group__lavc__misc.html#gaf60b0e076f822abcb… avcodec_flush_buffers()] between each decoding run. This seems to work for most codecs, but when used with the E-AC-3 decoder the output is not identical between runs How to reproduce: {{{#!c #include <libavcodec/avcodec.h> #include <libavformat/avformat.h> #include <stdio.h> #include <assert.h> int main(int argc, char *argv[]) { const AVCodec *codec = avcodec_find_decoder(AV_CODEC_ID_EAC3); AVCodecContext *decoder = avcodec_alloc_context3(codec); avcodec_open2(decoder, codec, NULL); assert(decoder->sample_fmt == AV_SAMPLE_FMT_FLTP); for (int l = 0; l < 2; ++ l) { printf("[%d] Reading file\n", l); avcodec_flush_buffers(decoder); AVFormatContext *avf_ctx = NULL; avformat_open_input(&avf_ctx, "file:i28553p303t610156395819694.ac3", NULL, NULL); avformat_find_stream_info(avf_ctx, NULL); AVPacket avpkt = { 0 }; while (av_read_frame(avf_ctx, &avpkt) == 0) { avcodec_send_packet(decoder, &avpkt); AVFrame *frame = av_frame_alloc(); avcodec_receive_frame(decoder, frame); printf("Got %d samples in %d channels\n", frame->nb_samples, frame->channels); for (int i = 0; i < frame->nb_samples; ++ i) { printf("%4d:", i); for (int j = 0; j < frame->channels; ++ j) { printf(" [%d]%f", j, ((float *) frame->data[j])[i]); } printf("\n"); } av_frame_free(&frame); } avformat_close_input(&avf_ctx); } avcodec_free_context(&decoder); return 0; } }}} The test file is available here: https://e.pcloud.link/publink/show?code=XZS8y1ZlEryJIDYU0yIzDTUhw1sDfHDRtUV and contains a ten-second extract of an audio track from an MPEGTS container, captured off-air while looking at a problem where the encoder suddenly only outputs silence. I expect the output to be identical on the first and second run, but that is not what I am seeing: {{{ $ ./decode |grep -A5 'Reading file' [eac3 @ 0x55fc4b0a5ec0] Estimating duration from bitrate, this may be inaccurate [0] Reading file Got 1536 samples in 6 channels 0: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 1: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 2: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 3: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 [eac3 @ 0x55fc4b11a840] Estimating duration from bitrate, this may be inaccurate -- [1] Reading file Got 1536 samples in 6 channels 0: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 1: [0]-0.000000 [1]0.000000 [2]-0.000000 [3]-0.000000 [4]-0.000000 [5]-0.000000 2: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 3: [0]-0.000000 [1]-0.000000 [2]-0.000000 [3]-0.000000 [4]-0.000000 [5]-0.000000 }}} In real life we are also interleaving other E-AC-3 audio streams from the same MPEGTS between decodes, and see that the initial decoded samples differ even more than when just redecoding the same sample over and over. -- Ticket URL: <https://trac.ffmpeg.org/ticket/10732> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#10959(undetermined:new): ffmpeg 7.0: Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:1923
by FFmpeg 06 Jul '25

06 Jul '25

#10959: ffmpeg 7.0: Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:1923 -------------------------------------+------------------------------------- Reporter: Andreas | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- I try to split a video into 3 parts: {{{ % C:/Java/ffmpeg-n7.0-latest-win64-gpl-7.0/bin/ffmpeg.exe -report -nostdin -y -i video.mp4 -filter_complex "[v:0]select=n='3':e='gte(t,0.00)*lt(t,10.45)*1 + gte(t,10.45)*lt(t,58.50)*2 + gte(t,58.50)*lt(t,3661.00)*3'[vout001][vout002][vout003]" -map [vout001] v00-001.mp4 -map [vout002] v00-002.mp4 -map [vout003] v00-003.mp4 ... [out_0_0 @ 0000021b889bd600] 100 buffers queued in out_0_0, something may be wrong. ... [out_1_0 @ 0000021b889be200] 100 buffers queued in out_1_0, something may be wrong. [out_1_0 @ 0000021b889be200] 1000 buffers queued in out_1_0, something may be wrong. ... Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:1923 }}} It is independent of the input video, but you can find the video here: https://a-voss.de/ffmpeg/video.mp4 -- Ticket URL: <https://trac.ffmpeg.org/ticket/10959> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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