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• Latest FFmpeg fails to create a good MP4 out of JPG file (Windows 10)

  30 décembre 2023, par mengrie

  I am using FFmpeg to create an MP4 file out of several JPG files. The command to accomplish this

  


  ffmpeg.exe ^
-hide_banner -nostats -loglevel error -y ^
-loop 1 -t 6 -i 001.jpg ^
-loop 1 -t 6 -i 002.jpg ^
-loop 1 -t 6 -i 003.jpg ^
-loop 1 -t 6 -i 004.jpg ^
-loop 1 -t 6 -i 005.jpg ^
-loop 1 -t 6 -i 006.jpg ^
-filter_complex ^
"[0:v]scale=1920:1080:force_original_aspect_ratio=decrease,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,setsar=1,fade=t=out:st=5:d=1[v0]; ^
[1:v]scale=1920:1080:force_original_aspect_ratio=decrease,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,setsar=1,fade=t=in:st=0:d=1,fade=t=out:st=5:d=1[v1]; ^
[2:v]scale=1920:1080:force_original_aspect_ratio=decrease,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,setsar=1,fade=t=in:st=0:d=1,fade=t=out:st=5:d=1[v2]; ^
[3:v]scale=1920:1080:force_original_aspect_ratio=decrease,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,setsar=1,fade=t=in:st=0:d=1,fade=t=out:st=5:d=1[v3]; ^
[4:v]scale=1920:1080:force_original_aspect_ratio=decrease,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,setsar=1,fade=t=in:st=0:d=1,fade=t=out:st=5:d=1[v4]; ^
[5:v]scale=1920:1080:force_original_aspect_ratio=decrease,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,setsar=1,fade=t=in:st=0:d=1,fade=t=out:st=5:d=1[v5]; ^
[v0][v1][v2][v3][v4][v5]concat=n=6:v=1:a=0,format=yuv420p[v]" -map "[v]" -r 30 006.mp4


  


  Using "ffmpeg version 2022-12-25-git-eeb280f351-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers built with gcc 12.1.0 (Rev2, Built by MSYS2 project)" does the job as expected.

  


  However, yesterdag I downloaded (from gyan.dev - Windows - ffmpeg-git-full.7z) the latest version and upgrade the tool. "ffmpeg version 2023-12-28-git-c1340f3439-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers built with gcc 12.2.0 (Rev10, Built by MSYS2 project)"

  


  Using this version, the MP4 contains only a few JPGs and timing is also messed up, resulting in a useless MP4.

  


  Has something changed or are these parameters causing a new behaviour ?

  


  Thx

  


  In meanwhile I went back to older version

  


• Cutting and rejoining videos causes audio de-synchronization using ffmpeg [closed]

  10 décembre 2023, par Donotalo

  This is the original post. I'm not getting any answer there so I thought may be the programming community knows the answer.

  


  Main Post

  


  I'm trying to cut a video into pieces and rejoin them using ffmpeg on Windows 11 x64. Here's the details of ffmpeg :

  


  ffmpeg version 2023-11-22-git-0008e1c5d5-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers
  built with gcc 12.2.0 (Rev10, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --pkg-config=pkgconf --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-libuavs3d
--enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxavs2 --enable-libxvid --enable-libaom --enable-libjxl --enable-libopenjpeg --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-ffnvcodec --enable-nvdec --enable-nvenc
--enable-dxva2 --enable-d3d11va --enable-libvpl --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      58. 32.100 / 58. 32.100
  libavcodec     60. 34.100 / 60. 34.100
  libavformat    60. 17.100 / 60. 17.100
  libavdevice    60.  4.100 / 60.  4.100
  libavfilter     9. 13.100 /  9. 13.100
  libswscale      7.  6.100 /  7.  6.100
  libswresample   4. 13.100 /  4. 13.100
  libpostproc    57.  4.100 / 57.  4.100


  


  I've a video file in MKV format. I've cut it into 3 pieces using the following command :

  


  ffmpeg.exe -ss 0:0:0 -i input.mkv -t 0:54:15 -c:v hevc_amf -b:v 3M -c:s mov_text seg-01.mp4
ffmpeg.exe -ss 0:54:29 -i input.mkv -t 0:35:35 -c:v hevc_amf -b:v 3M -c:s mov_text seg-02.mp4
ffmpeg.exe -ss 1:30:12 -i input.mkv -t 0:4:10 -c:v hevc_amf -b:v 3M -c:s mov_text seg-03.mp4


  


  The audio is copied in all 3 pieces like original. Now I'm joining them using the following command :

  


  ffmpeg.exe -y -f concat -safe 0 -i .\join.txt -c:v hevc_amf -c:a copy -c:s copy -fflags +genpts out.mp4


  


  where, join.txt is :

  


  file seg-01.mp4
file seg-02.mp4
file seg-03.mp4


  


  ffmpeg throws the following warning :

  


  [mp4 @ 000002a5cf9cc040] Non-monotonic DTS in output stream 0:1; previous: 156240896, current: 156240084; changing to 156240897.
This may result in incorrect timestamps in the output file.
[mp4 @ 000002a5cf9cc040] Non-monotonic DTS in output stream 0:1; previous: 258720980, current: 258720462; changing to 258720981.
This may result in incorrect timestamps in the output file.


  


  I've observed that audio is de-synchronized after the places where I cut.

  


  How to keep audio, video and subtitle synchronized after cutting and rejoining the video using ffmpeg ?

  


• Transcode webcam blob to RTMP using ffmpeg.wasm

  29 novembre 2023, par hassan moradnezhad

  I'm trying transcode webcam blob data to a rtmp server from browser by using ffmpeg.wasm .
  
first, i create a MediaStream.

  


          const stream = await navigator.mediaDevices.getUserMedia({
            video: true,
        });


  


  then, i create a MediaRecorder.

  


          const recorder = new MediaRecorder(stream, {mimeType: "video/webm; codecs:vp9"});
        recorder.ondataavailable = handleDataAvailable;
        recorder.start(0)


  


  when data is available, i call a function called handleDataAvailable.
  
here is the function.

  


      const handleDataAvailable = (event: BlobEvent) => {
        console.log("data-available");
        if (event.data.size > 0) {
            recordedChunksRef.current.push(event.data);
            transcode(event.data)
        }
    };


  


  in above code, i use another function which called transcode it's goal is going to send data to rtmp server using use ffmpeg.wasm.
  
here it is.

  


  const transcode = async (inputVideo: Blob | undefined) => {
        const ffmpeg = ffmpegRef.current;
        const fetchFileOutput = await fetchFile(inputVideo)
        ffmpeg?.writeFile('input.webm', fetchFileOutput)

        const data = await ffmpeg?.readFile('input.webm');
        if (videoRef.current) {
            videoRef.current.src =
                URL.createObjectURL(new Blob([(data as any)?.buffer], {type: 'video/webm'}));
        }

        // execute by node-media-server config 1
        await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c', 'copy', '-f', 'flv', "rtmp://localhost:1935/live/ttt"])

        // execute by node-media-server config 2
        // await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c:v', 'libx264', '-preset', 'veryfast', '-tune', 'zerolatency', '-c:a', 'aac', '-ar', '44100', '-f', 'flv', 'rtmp://localhost:1935/live/ttt']);

        // execute by stack-over-flow config 1
        // await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c:v', 'h264', '-c:a', 'aac', '-f', 'flv', "rtmp://localhost:1935/live/ttt"]);

        // execute by stack-over-flow config 2
        // await ffmpeg?.exec(['-i', 'input.webm', '-c:v', 'libx264', '-flags:v', '+global_header', '-c:a', 'aac', '-ac', '2', '-f', 'flv', "rtmp://localhost:1935/live/ttt"]);

        // execute by stack-over-flow config 3
        // await ffmpeg?.exec(['-i', 'input.webm', '-acodec', 'aac', '-ac', '2', '-strict', 'experimental', '-ab', '160k', '-vcodec', 'libx264', '-preset', 'slow', '-profile:v', 'baseline', '-level', '30', '-maxrate', '10000000', '-bufsize', '10000000', '-b', '1000k', '-f', 'flv', 'rtmp://localhost:1935/live/ttt']);

    }


  


  after running app and start streaming, console logs are as below.

  


  ffmpeg >>>  ffmpeg version 5.1.3 Copyright (c) 2000-2022 the FFmpeg developers index.tsx:81:20
ffmpeg >>>    built with emcc (Emscripten gcc/clang-like replacement + linker emulating GNU ld) 3.1.40 (5c27e79dd0a9c4e27ef2326841698cdd4f6b5784) index.tsx:81:20
ffmpeg >>>    configuration: --target-os=none --arch=x86_32 --enable-cross-compile --disable-asm --disable-stripping --disable-programs --disable-doc --disable-debug --disable-runtime-cpudetect --disable-autodetect --nm=emnm --ar=emar --ranlib=emranlib --cc=emcc --cxx=em++ --objcc=emcc --dep-cc=emcc --extra-cflags='-I/opt/include -O3 -msimd128' --extra-cxxflags='-I/opt/include -O3 -msimd128' --disable-pthreads --disable-w32threads --disable-os2threads --enable-gpl --enable-libx264 --enable-libx265 --enable-libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libopus --enable-zlib --enable-libwebp --enable-libfreetype --enable-libfribidi --enable-libass --enable-libzimg index.tsx:81:20
ffmpeg >>>    libavutil      57. 28.100 / 57. 28.100 index.tsx:81:20
ffmpeg >>>    libavcodec     59. 37.100 / 59. 37.100 index.tsx:81:20
ffmpeg >>>    libavformat    59. 27.100 / 59. 27.100 index.tsx:81:20
ffmpeg >>>    libavdevice    59.  7.100 / 59.  7.100 index.tsx:81:20
ffmpeg >>>    libavfilter     8. 44.100 /  8. 44.100 index.tsx:81:20
ffmpeg >>>    libswscale      6.  7.100 /  6.  7.100 index.tsx:81:20
ffmpeg >>>    libswresample   4.  7.100 /  4.  7.100 index.tsx:81:20
ffmpeg >>>    libpostproc    56.  6.100 / 56.  6.100 index.tsx:81:20
ffmpeg >>>  Input #0, matroska,webm, from 'input.webm': index.tsx:81:20
ffmpeg >>>    Metadata: index.tsx:81:20
ffmpeg >>>      encoder         : QTmuxingAppLibWebM-0.0.1 index.tsx:81:20
ffmpeg >>>    Duration: N/A, start: 0.000000, bitrate: N/A index.tsx:81:20
ffmpeg >>>    Stream #0:0(eng): Video: vp8, yuv420p(progressive), 640x480, SAR 1:1 DAR 4:3, 15.50 tbr, 1k tbn (default)


  


  the problem is when ffmpeg.wasm try to execute the last command.
  
await ffmpeg?.exec(['-re', '-i', 'input.webm', '-c', 'copy', '-f', 'flv', "rtmp://localhost:1935/live/ttt"]).
  
it just calls a GET Request, I will send further details about this request.
  
as u can see, i try to use lots of arg sample with ffmpeg?.exec, but non of them works.

  


  the network tab in browser, after ffmpeg.wasm execute the command is as below.

  


  

  


  it send a GET request to ws://localhost:1935/
and nothing happened after that.

  


  for backend, i use node-media-server and here is my output logs when ffmpeg.wasm trying to execute the args

  


  11/28/2023 19:33:18 55301 [INFO] [rtmp disconnect] id=JL569YOF
[NodeEvent on doneConnect] id=JL569YOF args=undefined


  


  at last here are my ques

  


  


  

  • how can i achive this option ?

  


  • is it possible to share webcam to rtmp server ?