March 2025 - ffmpeg-user - lists.ffmpeg.org

Would it be possible to load xattr-related media-files like a container?
by Peter B. 02 Aug '25

02 Aug '25

Hi everyone :) I'm working a lot with extended attributes on filesystems, and I'm amazed! I was wondering if it would be possible for ffmpeg to load "related media-streams" as related objects: Having a 0-Byte "videofile.aha": * video = /path/to/my_movie.h264 * audio = ./my_movie-DE_AT.flac * audio.1 = my_movie-EN_GB.mp3 * video.default = video * audio.default = audio.1 * dc.title = "My Movie" * dc.creator = "..." * ... Like an input demuxer which resolves the related objects and treats this like loading a container? I imagine this to be quite fun (and useful actually). One could then drag-and-drop to remux tracks, and right-click-edit metadata, I imagine. Feedback greatly appreciated :) Have a great day! Peter

1 1

ffmpeg and dv
by Dave Rice 30 Mar '25

30 Mar '25

Hi all, I love ffmpeg and I work with DV. Two years ago this commit, https://git.ffmpeg.org/gitweb/ffmpeg.git/commit/c7a852b638b88f129790af31fc4…, caused a regression so that the DV frame rate would be misinterpreted, for instance as 60000/1 fps rather than 30000/1001 fps. I reported this as a ticket at https://trac.ffmpeg.org/ticket/11339. For the meanwhile I’ve been staying at ffmpeg 5 to use a version without the commit, but although DV is a bit of an obsolete codec I’d love to see some consideration for the use of DV in ffmpeg’s future. Would any of the following options work: - Is there any workaround to force a frame rate of an input? For instance, could I use a non-ffmpeg tool to identify the frame rate and force ffmpeg to use a user-supplied one (such as 25 or 30000/1001) rather than the one is detects (such as 60000/1 fps). - Anyone willing to fix the reported issue? I could offer some sponsorship of the work so that ffmpeg could once again work with DV? Kind Regards, Dave Rice

3 2

has setpts filter been changed?
by William C Bonner 21 Mar '25

21 Mar '25

I use setpts to speed up some video that I'm concatenating from my car's dashcam. The process is consistent and hasn't changed recently. Here's an example command line: C:\Windows\ffmpeg.exe -hide_banner -i D:\ROAV\MOVIE\2024_0803_173613_229.MP4 -i D:\ROAV\MOVIE\2024_0803_173713_230.MP4 -i D:\ROAV\MOVIE\2024_0803_173813_231.MP4 -i D:\ROAV\MOVIE\2024_0803_173913_232.MP4 -i D:\ROAV\MOVIE\2024_0803_174013_233.MP4 -i D:\ROAV\MOVIE\2024_0803_174113_234.MP4 -i D:\ROAV\MOVIE\2024_0803_174213_235.MP4 -i D:\ROAV\MOVIE\2024_0803_174313_236.MP4 -i D:\ROAV\MOVIE\2024_0803_174413_237.MP4 -i D:\ROAV\MOVIE\2024_0803_174513_238.MP4 -i D:\ROAV\MOVIE\2024_0803_174613_239.MP4 -i D:\ROAV\MOVIE\2024_0803_174713_240.MP4 -i D:\ROAV\MOVIE\2024_0803_174813_241.MP4 -i D:\ROAV\MOVIE\2024_0803_174913_242.MP4 -i D:\ROAV\MOVIE\2024_0803_175013_243.MP4 -i D:\ROAV\MOVIE\2024_0803_175113_244.MP4 -i D:\ROAV\MOVIE\2024_0803_175213_245.MP4 -i D:\ROAV\MOVIE\2024_0803_175313_246.MP4 -i D:\ROAV\MOVIE\2024_0803_175413_247.MP4 -i D:\ROAV\MOVIE\2024_0803_175513_248.MP4 -i D:\ROAV\MOVIE\2024_0803_175613_249.MP4 -i D:\ROAV\MOVIE\2024_0803_175713_250.MP4 -i D:\ROAV\MOVIE\2024_0803_175813_251.MP4 -i D:\ROAV\MOVIE\2024_0803_175913_252.MP4 -i D:\ROAV\MOVIE\2024_0803_180013_253.MP4 -i D:\ROAV\MOVIE\2024_0803_180113_254.MP4 -i D:\ROAV\MOVIE\2024_0803_180213_255.MP4 -filter_complex [0:v][1:v][2:v][3:v][4:v][5:v][6:v][7:v][8:v][9:v][10:v][11:v][12:v][13:v][14:v][15:v][16:v][17:v][18:v][19:v][20:v][21:v][22:v][23:v][24:v][25:v][26:v]concat=n=27:v=1[v];[v]setpts=(1/60)*PTS,drawtext=fontfile=C\\:/Windows/Fonts/consola.ttf:fontcolor=white:fontsize=80:y=main_h-text_h-50:x=50:text=WimsWorld[o] -map [o] -map_metadata -1 -c:v libx265 -crf 23 -preset veryfast -movflags +faststart -bf 2 -g 15 -pix_fmt yuv420p -n D:\ROAV\20240803-BMW.mp4 The important part is setpts=(1/60)*PTS which should convert 1 minute of the source video to 1 second in the destination I noticed that watching recent videos strangely seemed much smoother, then realized that they were also significantly larger. https://trac.ffmpeg.org/wiki/How%20to%20speed%20up%20/%20slow%20down%20a%20… still says that using this method will drop frames, but when I ran ffprobe on the current file I got: C:\Users\Wim>ffprobe d:\ROAV\20240803-BMW.mp4 ffprobe version 2024-07-18-git-fa5a605542-full_build-www.gyan.dev Copyright (c) 2007-2024 the FFmpeg developers built with gcc 13.2.0 (Rev5, Built by MSYS2 project) configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libaribcaption --enable-libdav1d --enable-libdavs2 --enable-libopenjpeg --enable-libquirc --enable-libuavs3d --enable-libxevd --enable-libzvbi --enable-libqrencode --enable-librav1e --enable-libsvtav1 --enable-libvvenc --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxeve --enable-libxvid --enable-libaom --enable-libjxl --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libcodec2 --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint libavutil 59. 28.100 / 59. 28.100 libavcodec 61. 10.100 / 61. 10.100 libavformat 61. 5.101 / 61. 5.101 libavdevice 61. 2.100 / 61. 2.100 libavfilter 10. 2.102 / 10. 2.102 libswscale 8. 2.100 / 8. 2.100 libswresample 5. 2.100 / 5. 2.100 libpostproc 58. 2.100 / 58. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'd:\ROAV\20240803-BMW.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 encoder : Lavf61.5.101 Duration: 00:00:27.07, start: 0.000000, bitrate: 889896 kb/s Stream #0:0[0x1](und): Video: hevc (Main) (hev1 / 0x31766568), yuv420p(tv, progressive), 2560x1440, 889591 kb/s, 1800.04 fps, 1000k tbr, 1000k tbn (default) Metadata: handler_name : VideoHandler vendor_id : [0][0][0][0] encoder : Lavc61.10.100 libx265 Running it on an older file produced: C:\Users\Wim>ffprobe d:\ROAV\20240523-BMW.mp4 ffprobe version 2024-07-18-git-fa5a605542-full_build-www.gyan.dev Copyright (c) 2007-2024 the FFmpeg developers built with gcc 13.2.0 (Rev5, Built by MSYS2 project) configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libaribcaption --enable-libdav1d --enable-libdavs2 --enable-libopenjpeg --enable-libquirc --enable-libuavs3d --enable-libxevd --enable-libzvbi --enable-libqrencode --enable-librav1e --enable-libsvtav1 --enable-libvvenc --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxeve --enable-libxvid --enable-libaom --enable-libjxl --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libcodec2 --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint libavutil 59. 28.100 / 59. 28.100 libavcodec 61. 10.100 / 61. 10.100 libavformat 61. 5.101 / 61. 5.101 libavdevice 61. 2.100 / 61. 2.100 libavfilter 10. 2.102 / 10. 2.102 libswscale 8. 2.100 / 8. 2.100 libswresample 5. 2.100 / 5. 2.100 libpostproc 58. 2.100 / 58. 2.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'd:\ROAV\20240523-BMW.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 encoder : Lavf61.3.103 Duration: 00:09:40.88, start: 0.000000, bitrate: 26675 kb/s Stream #0:0[0x1](und): Video: hevc (Main) (hev1 / 0x31766568), yuv420p(tv, progressive), 2560x1440, 26672 kb/s, 25 fps, 25 tbr, 12800 tbn (default) Metadata: handler_name : VideoHandler vendor_id : [0][0][0][0] encoder : Lavc61.5.104 libx265 The important difference is that the older file reports 2560x1440, 26672 kb/s, 25 fps, 25 tbr, 12800 tbn and the newer file reports 2560x1440, 889591 kb/s, 1800.04 fps, 1000k tbr, 1000k tbn. This seems that the new code speed up the reported fps without dropping any frames.

3 4

Continue audio even when RTSP video stutters?
by Steven Kan 20 Mar '25

20 Mar '25

Hi! For the last 7 years I've been streaming RTSP from an IP camera to one of my 4 YouTube channels: https://www.youtube.com/@IAmTheWaterbug/live I am adding a series of royalty-free MP3 to the stream because the camera lacks audio, but YouTube Stream Now requires an audio channel, and this was better (IMNSHO) than adding an empty audio track. I also like the music :-). This has worked very well for the last 7 years. About 6 months ago the camera started malfunctioning, and the RTSP stream now stutters every few seconds. It's definitely a problem with the stream from this particular camera because my other 3 streams from different IP cameras, on the same network, and from the same streaming computer, are working fine at minimal CPU load. If I paste the URL for the problem camera into VLC, the video stutters just like the YT stream. ffmpeg reports: vt decoder cb: output image buffer is null repeatedly (see command string, below). As you can see on the stream, the video stuttering is annoying, but not crippling, but the audio interruptions are intolerable. I am still looking to replace the camera, but while I'm looking, is there a way to modify my processing chain so that, when the RTSP video stutters, the audio does _not_ pause? The audio is being sourced directly from MP3s on the local HD of the computer running ffmpeg, so there's no issue with that data being accessible in a timely manner. When the video stream stutters I'd like it to continue doing what it does now, e.g. just display the last available frame until a new one comes along. If I output to DeleteMe.flv, then the FLV video is a mess, while the audio continues to play normally. If I open that FLV in VLC the video will just stall completely. If I open it in IINA I can skip around in the timeline, and it will play a few frames, but it won't play through: https://www.kan.org/download/DeleteMe.flv I don't care about latency, within reason, because it's already at least 10 seconds behind real-time, based on the timestamp. Thanks! ./ffmpeg1 -thread_queue_size 2048 -hwaccel videotoolbox -i "rtsp://anonymous:password@192.168.1.11:554" -f concat -safe 0 -i playlist.txt -vf drawtext="fontfile=/Users/steven/Library/Fonts/Nouveau_IBM.ttf: text='%{localtime\:%Y-%m-%d\ %H\\\\\:%M\\\\\:%S}': fontcolor=white: fontsize=66: x=(w-text_w)*0.01: y=(h-text_h)*0.08" -vcodec h264_videotoolbox -b:v 5000k -acodec copy -t 00:1:00 -f flv DeleteMe.flv ffmpeg version N-109676-g40512dbd96-tessus https://evermeet.cx/ffmpeg/ Copyright (c) 2000-2023 the FFmpeg developers built with Apple clang version 11.0.0 (clang-1100.0.33.17) configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvmaf --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3 --pkg-config-flags=--static --disable-ffplay libavutil 57. 44.100 / 57. 44.100 libavcodec 59. 57.100 / 59. 57.100 libavformat 59. 36.100 / 59. 36.100 libavdevice 59. 8.101 / 59. 8.101 libavfilter 8. 54.100 / 8. 54.100 libswscale 6. 8.112 / 6. 8.112 libswresample 4. 9.100 / 4. 9.100 libpostproc 56. 7.100 / 56. 7.100 Input #0, rtsp, from 'rtsp://anonymous:password@192.168.1.11:554': Metadata: title : Session streamed by "preview" comment : Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: h264 (High), yuv420p(progressive), 3072x1728, 20 fps, 20 tbr, 90k tbn Stream #0:1: Audio: aac (LC), 16000 Hz, mono, fltp [mp3 @ 0x7fc0f9804c80] Estimating duration from bitrate, this may be inaccurate Input #1, concat, from 'playlist.txt': Duration: N/A, start: 0.000000, bitrate: 320 kb/s Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 320 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (h264_videotoolbox)) Stream #1:0 -> #0:1 (copy) Press [q] to stop, [?] for help Output #0, flv, to 'DeleteMe.flv': Metadata: title : Session streamed by "preview" comment : encoder : Lavf59.36.100 Stream #0:0: Video: h264 ([7][0][0][0] / 0x0007), nv12(progressive), 3072x1728, q=2-31, 5000 kb/s, 20 fps, 1k tbn Metadata: encoder : Lavc59.57.100 h264_videotoolbox Stream #0:1: Audio: mp3 ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp, 320 kb/s [h264_videotoolbox @ 0x7fc0f8f0e5c0] Color range not set for nv12. Using MPEG range. [h264 @ 0x7fc0f8f10ac0] vt decoder cb: output image buffer is null: -12909kbits/s speed=0.44x [h264 @ 0x7fc0f8f1b680] hardware accelerator failed to decode picture [h264 @ 0x7fc0f8f10ac0] vt decoder cb: output image buffer is null: -12909kbits/s speed=0.414x [h264 @ 0x7fc0f8f26040] hardware accelerator failed to decode picture Error while decoding stream #0:0: Unknown error occurred.29 bitrate=5329.5kbits/s speed=0.41x Last message repeated 1 times [h264 @ 0x7fc0f8f10ac0] vt decoder cb: output image buffer is null: -12909kbits/s speed=0.742x [h264 @ 0x7fc0e90043c0] hardware accelerator failed to decode picture Error while decoding stream #0:0: Unknown error occurred.84 bitrate=4727.5kbits/s speed=0.729x frame= 315 fps=7.9 q=-0.0 size= 19712kB time=00:00:39.96 bitrate=4040.4kbits/s speed=0.996x

2 1

multiple readrate
by Jonathan Baecker 19 Mar '25

19 Mar '25

Hi, does anyone know if it makes sense to use the -readrate option for each input file or is it enough to use it only once. For example, if I have a video and a webvtt file: ffmpeg -readrate 1 -i video.mp4 -readrate 1 -i subtitle.vtt ... For a single file I don't see any difference, but what if I sometimes use -ss or -t option and write many files into a single m3u8/ts stream? Have a good day! Jonathan

2 2

Decoding slice based frames
by d.gordenin＠ngslab.ru 18 Mar '25

18 Mar '25

Hi My decoder receives slice based frame, and it cannot decode them (the result is a greased picture). How should I decode them and which are differences from decoding a usual frame? My code of dedoding is canonical: if (size >= sizeof(kh264Prefix) && memcmp(data, kh264Prefix, sizeof(kh264Prefix)) != 0) { if (size > prefix_buffer_.size() - sizeof(kh264Prefix)) { logger_->Error("Too large data packet for decoding"); return false; } memcpy(prefix_buffer_.data() + sizeof(kh264Prefix), data, size); data = prefix_buffer_.data(); size += sizeof(kh264Prefix); } AVPacket avpkt; av_init_packet(&avpkt); avpkt.pts = 0; avpkt.dts = 0; avpkt.data = data; avpkt.size = size; int gotFrame = false; while (avpkt.size > 0) { // We must use positive result to move through the data buffer auto len = avcodec_decode_video2(av_context_, av_frame_, &gotFrame, &avpkt); if (len < 0) { // In case of an error/warning FFmpeg will call // decoder::LogCallback(...) to inform about the issue break; } avpkt.data += len; avpkt.size -= len; } av_free_packet(&avpkt); I tried the AV_CODEC_CAP_SLICE_THREADS flag, not sure is it for decoding or not.

1 1

ffmpeg doesn't reconnect when capturing live connections
by arthurpeabody 16 Mar '25

16 Mar '25

It begins with: ffmpeg -i https://listen.wruu.org/stream -t 3720 -reconnect true -reconnect_streamed true -reconnect_on_network_error true -timelimit 3720 -loglevel warning wruu_JoeFrank-20250130-2100.mp3 ffmpeg version 7.1 Copyright (c) 2000-2024 the FFmpeg developers built with gcc 14.2.0 (GCC) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --docdir=/usr/doc/ffmpeg-7.1/html --mandir=/usr/man --disable-debug --enable-shared --disable-static --enable-gpl --enable-version3 --arch=x86_64 --disable-encoder=aac --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-gnutls --enable-libass --enable-libbluray --enable-libcaca --enable-libcdio --enable-frei0r --enable-openal --enable-libopus --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libmp3lame --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libpulse --enable-libsmbclient --enable-libxml2 --enable-librsvg --enable-vulkan --enable-libdrm --enable-libaom --enable-libdav1d --enable-libshaderc --enable-gcrypt --enable-libplacebo --enable-lcms2 libavutil 59. 39.100 / 59. 39.100 libavcodec 61. 19.100 / 61. 19.100 libavformat 61. 7.100 / 61. 7.100 libavdevice 61. 3.100 / 61. 3.100 libavfilter 10. 4.100 / 10. 4.100 libswscale 8. 3.100 / 8. 3.100 libswresample 5. 3.100 / 5. 3.100 libpostproc 58. 3.100 / 58. 3.100 Splitting the commandline. Reading option '-i' ... matched as input url with argument 'https://listen.wruu.org/stream'. Reading option '-t' ... matched as option 't' (stop transcoding after specified duration) with argument '3720'. Reading option '-reconnect' ... matched as AVOption 'reconnect' with argument 'true'. Reading option '-reconnect_streamed' ... matched as AVOption 'reconnect_streamed' with argument 'true'. Reading option '-reconnect_on_network_error' ... matched as AVOption 'reconnect_on_network_error' with argument 'true'. Reading option '-timelimit' ... matched as option 'timelimit' (set max runtime in seconds in CPU user time) with argument '3720'. Reading option '-v' ... matched as option 'v' (set logging level) with argument '48'. Reading option 'wruu_JoeFrank-20250130-2100.mp3' ... matched as output url. Finished splitting the commandline. Parsing a group of options: global . Applying option timelimit (set max runtime in seconds in CPU user time) with argument 3720. Applying option v (set logging level) with argument 48. Successfully parsed a group of options. Parsing a group of options: input url https://listen.wruu.org/stream. Successfully parsed a group of options. Opening an input file: https://listen.wruu.org/stream. [AVFormatContext @ 0x36542540] Opening 'https://listen.wruu.org/stream' for reading [https @ 0x365388c0] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy' size= 9984KiB time=00:10:48.15 bitrate= 126.2kbits/s speed=0.0019x [tcp @ 0x36542e00] Original list of addresses: [tcp @ 0x36542e00] Address 165.227.18.155 port 443 [tcp @ 0x36542e00] Interleaved list of addresses: [tcp @ 0x36542e00] Address 165.227.18.155 port 443 [tcp @ 0x36542e00] Starting connection attempt to 165.227.18.155 port 443 [tcp @ 0x36542e00] Successfully connected to 165.227.18.155 port 443 [https @ 0x365388c0] request: GET /stream HTTP/1.1 User-Agent: Lavf/61.7.100 Accept: */* Range: bytes=0- Connection: close Host: listen.wruu.org Icy-MetaData: 1 [mp3 @ 0x36542540] Format mp3 probed with size=4096 and score=51 [mp3 @ 0x36542540] Skipping 0 bytes of junk at 0. [mp3 @ 0x36542540] Before avformat_find_stream_info() pos: 0 bytes read:7569 seeks:0 nb_streams:1 [https @ 0x365388c0] Metadata update for StreamTitle: - WRUU-WRUU [mp3 @ 0x36542540] All info found [mp3 @ 0x36542540] After avformat_find_stream_info() pos: 31744 bytes read:32000 seeks:0 frames:50 Input #0, mp3, from 'https://listen.wruu.org/stream': Metadata: icy-br : 192 icy-channels : 2 icy-name : WRUU 2022 icy-pub : 1 icy-samplerate : 44100 StreamTitle : - WRUU-WRUU Duration: N/A, start: 0.000000, bitrate: 192 kb/s Stream #0:0, 50, 1/14112000: Audio: mp3 (mp3float), 44100 Hz, stereo, fltp, 192 kb/s Successfully opened the file. Parsing a group of options: output url wruu_JoeFrank-20250130-2100.mp3. Applying option t (stop transcoding after specified duration) with argument 3720. Successfully parsed a group of options. Opening an output file: wruu_JoeFrank-20250130-2100.mp3. [out#0/mp3 @ 0x36888b80] No explicit maps, mapping streams automatically... [aost#0:0/libmp3lame @ 0x36889c80] Created audio stream from input stream 0:0 [file @ 0x3689d1c0] Setting default whitelist 'file,crypto,data' Successfully opened the file. Stream mapping: Stream #0:0 -> #0:0 (mp3 (mp3float) -> mp3 (libmp3lame)) [aost#0:0/libmp3lame @ 0x36889c80] Starting thread... [af#0:0 @ 0x36546c00] Starting thread... [aist#0:0/mp3 @ 0x3655f340] [dec:mp3float @ 0x368786c0] Starting thread... [in#0/mp3 @ 0x36542280] Starting thread... Press [q] to stop, [?] for help detected 4 logical cores [graph_-1_in_0:0 @ 0x7fdb580022c0] Setting 'time_base' to value '1/44100' [graph_-1_in_0:0 @ 0x7fdb580022c0] Setting 'sample_rate' to value '44100' [graph_-1_in_0:0 @ 0x7fdb580022c0] Setting 'sample_fmt' to value 'fltp' [graph_-1_in_0:0 @ 0x7fdb580022c0] Setting 'channel_layout' to value 'stereo' [graph_-1_in_0:0 @ 0x7fdb580022c0] tb:1/44100 samplefmt:fltp samplerate:44100 chlayout:stereo [format_out_#0:0 @ 0x7fdb58002c80] Setting 'sample_fmts' to value 's32p|fltp|s16p' [format_out_#0:0 @ 0x7fdb58002c80] Setting 'sample_rates' to value '44100|48000|32000|22050|24000|16000|11025|12000|8000' [format_out_#0:0 @ 0x7fdb58002c80] Setting 'channel_layouts' to value 'mono|stereo' [AVFilterGraph @ 0x7fdb58001000] query_formats: 5 queried, 9 merged, 3 already done, 0 delayed Output #0, mp3, to 'wruu_JoeFrank-20250130-2100.mp3': Metadata: icy-br : 192 icy-channels : 2 icy-name : WRUU 2022 icy-pub : 1 icy-samplerate : 44100 StreamTitle : - WRUU-WRUU TSSE : Lavf61.7.100 Stream #0:0, 0, 1/44100: Audio: mp3, 44100 Hz, stereo, fltp, delay 1105 Metadata: encoder : Lavc61.19.100 libmp3lame [out#0/mp3 @ 0x36888b80] Starting thread... Then there are thousands of entries like this: out#0/mp3 @ 0x36888b80] sq: send 0 ts 0.0261224 [out#0/mp3 @ 0x36888b80] sq: receive 0 ts 0.0261224 queue head -1 ts N/A [out#0/mp3 @ 0x36888b80] sq: send 0 ts 0.0522449 [out#0/mp3 @ 0x36888b80] sq: receive 0 ts 0.0522449 queue head -1 ts N/A [out#0/mp3 @ 0x36888b80] sq: send 0 ts 0.0783673 Then thousands that look like this: size= 54272KiB time=00:58:02.98 bitrate= 127.6kbits/s speed=0.679x No error is reported. ffmpeg is still running after its specified time.

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Issue with MJPEG Video Conversion in FFmpeg
by Enrico Schreier 12 Mar '25

12 Mar '25

Dear FFmpeg Development Team, I hope this message finds you well. We are encountering an issue when processing a video file using FFmpeg, and we are hoping you can provide some assistance. The video in question is in MJPEG format and features a transition from 8-bit greyscale to an YUV420 color space at some point during playback. Here is the issue we are experiencing: Conversion with FFmpeg: When we convert the video using FFmpeg, the filter graph appears to adjust correctly to the different color spaces. The video plays correctly when processed in forward motion, but it ends up being completely in greyscale. Reversing the Video: When we attempt to reverse the video using the following command: ffmpeg -ss 380 -to 480 -i input.avi -filter_complex "[0]reverse" out4.mp4 the video file becomes corrupted. Specifically, the greyscale portions of the video are skipped, and the non-greyscale sections are displayed in color. This results in a fragmented output where parts of the video are missing entirely (the greyscale segments), and other parts are displayed in color. We are unable to share the video file due to privacy and data protection reasons, but we have attached the relevant FFmpeg output as a .txt file for your reference. We hope that this will help you better understand the issue. Our request: Could you please assist us in identifying the cause of this behavior? It seems that FFmpeg may not be properly handling the transition between the different color spaces when reversing the video. We would appreciate any insights or potential solutions you can provide to help us resolve this issue. Thank you very much for your time and assistance. We look forward to your guidance and support. Best regards, Enrico Schreier (c) ERDMANN-Softwaregesellschaft mbH Gesch?ftsf?hrer: Ulrich Erdmann Dr.-Kahlbaum-Allee 16 02826 G?RLITZ DEUTSCHLAND Telefon: +49 3581 4761-0 Telefax: +49 3581 4761-99 E-Mail: info(a)erdmannsoftware.com Rechtsform: Gesellschaft mit beschr?nkter Haftung Handelsregister: Amtsgericht Dresden HRB 9857

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Issue with Delayed Packet Writing to MP4 Container Using Hardware Encoder (h264_v4l2m2m) in FFmpeg API
by Lisa 05 Mar '25

05 Mar '25

Hello. I am trying to store the encoded packets to a mp4 container file using av_interlenced_write_packet() ffmpeg api. With Hardware encoder[h264_v4l2m2m] there is an issue when I add 1 second delay to write encoded Packets to the mp4 container. While writing the packets there are no errors but the encoded video is not playing.When verified with *mediainfo* CLI, got some partial metadata about the video. NOTE: Without delay there is no issue with recorded video. - *Mediainfo output of video without delay* [image: mediainfo_of_no_delay.png] - *Mediainfo output of video with delay* [image: mediainfo_of_with_delay.png] The flow of my code is as described in the text and illustrated in the flow diagram. ... ->send packet to encode ->Receive packet ->then clone the packet with av_packet_clone(sourcePacket) ->modified the pts & dts and ->writing frames to disk using av_interleaved_write_frame <https://www.ffmpeg.org/doxygen/4.0/group__lavf__encoding.html#ga37352ed2c63…> (AVFormatContext <https://www.ffmpeg.org/doxygen/4.0/structAVFormatContext.html> *s <https://www.ffmpeg.org/doxygen/4.0/llviddspenc_8c.html#a517a4433fe5466cc6af…> , AVPacket <https://www.ffmpeg.org/doxygen/4.0/structAVPacket.html> *pkt <https://www.ffmpeg.org/doxygen/4.0/demuxing__decoding_8c.html#a3d4c6562f0b2…> ) ... [image: ffmpeg.png] 1. Why is the issue occurring with a delay of about 1 second? 2. How can this issue be resolved?

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Extracting slides from a slideshow recording
by Avid Seeker 04 Mar '25

04 Mar '25

I have a video recording of someone explaining a slideshow. Is there an ffmpeg command to extract individual unique frames from the video so that I get the slideshow in slide_%d.png files? The solutions listed in - https://stackoverflow.com/questions/11904531/ffmpeg-how-to-extract-frames-a… - https://video.stackexchange.com/questions/20958/ffmpeg-dropping-duplicate-f… Do not work when the slide show have fade animation or other kinds of animation. As a simple algorithm, it should extract frames only that stayed unchanged for say 2 seconds. As that would mean those frames were part of the slideshow. Avid

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