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  • 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.

    


    enter image description here

    


    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 ?

      • 


    


    


  • ffmpeg Audiosegment error in get audio chunks in socketIo server in python

    26 janvier 2024, par a_crszkvc30Last_NameCol

    I want to send each audio chunk every minute.
this is the test code and i want to save audiofile and audio chunk file.
then, i will combine two audio files stop button was worked correctly but with set time function is not worked in python server.
there is python server code with socketio

    


    def handle_voice(sid,data): # blob 으로 들어온 데이터 
    # BytesIO를 사용하여 메모리 상에서 오디오 데이터를 로드
    audio_segment = AudioSegment.from_file(BytesIO(data), format="webm")
    directory = "dddd"
    # 오디오 파일로 저장
    #directory = str(names_sid.get(sid))
    if not os.path.exists(directory):
        os.makedirs(directory)
 
    # 오디오 파일로 저장
    file_path = os.path.join(directory, f'{sid}.wav')
    audio_segment.export(file_path, format='wav') 
    print('오디오 파일 저장 완료')`
 


    


    and there is client

    


    &#xA;&#xA;&#xA;&#xA;    &#xA;    &#xA;    <code class="echappe-js">&lt;script src=&quot;https://cdnjs.cloudflare.com/ajax/libs/socket.io/4.5.2/socket.io.js&quot;&gt;&lt;/script&gt;&#xA;&#xA;&#xA;&#xA;    &#xA;    

    &#xA;
    


    &#xA;
    

    

    &#xA;    &lt;script&gt;&amp;#xA;        var socket = io(&amp;#x27;http://127.0.0.1:5000&amp;#x27;);&amp;#xA;        const record = document.getElementById(&quot;record&quot;)&amp;#xA;        const stop = document.getElementById(&quot;stop&quot;)&amp;#xA;        const soundClips = document.getElementById(&quot;sound-clips&quot;)&amp;#xA;        const chkHearMic = document.getElementById(&quot;chk-hear-mic&quot;)&amp;#xA;&amp;#xA;        const audioCtx = new(window.AudioContext || window.webkitAudioContext)() // 오디오 컨텍스트 정의&amp;#xA;&amp;#xA;        const analyser = audioCtx.createAnalyser()&amp;#xA;        //        const distortion = audioCtx.createWaveShaper()&amp;#xA;        //        const gainNode = audioCtx.createGain()&amp;#xA;        //        const biquadFilter = audioCtx.createBiquadFilter()&amp;#xA;&amp;#xA;        function makeSound(stream) {&amp;#xA;            const source = audioCtx.createMediaStreamSource(stream)&amp;#xA;            socket.connect()&amp;#xA;            source.connect(analyser)&amp;#xA;            //            analyser.connect(distortion)&amp;#xA;            //            distortion.connect(biquadFilter)&amp;#xA;            //            biquadFilter.connect(gainNode)&amp;#xA;            //            gainNode.connect(audioCtx.destination) // connecting the different audio graph nodes together&amp;#xA;            analyser.connect(audioCtx.destination)&amp;#xA;&amp;#xA;        }&amp;#xA;&amp;#xA;        if (navigator.mediaDevices) {&amp;#xA;            console.log(&amp;#x27;getUserMedia supported.&amp;#x27;)&amp;#xA;&amp;#xA;            const constraints = {&amp;#xA;                audio: true&amp;#xA;            }&amp;#xA;            let chunks = []&amp;#xA;&amp;#xA;            navigator.mediaDevices.getUserMedia(constraints)&amp;#xA;                .then(stream =&gt; {&amp;#xA;&amp;#xA;                    const mediaRecorder = new MediaRecorder(stream)&amp;#xA;                    &amp;#xA;                    chkHearMic.onchange = e =&gt; {&amp;#xA;                        if(e.target.checked == true) {&amp;#xA;                            audioCtx.resume()&amp;#xA;                            makeSound(stream)&amp;#xA;                        } else {&amp;#xA;                            audioCtx.suspend()&amp;#xA;                        }&amp;#xA;                    }&amp;#xA;                    &amp;#xA;                    record.onclick = () =&gt; {&amp;#xA;                        mediaRecorder.start(1000)&amp;#xA;                        console.log(mediaRecorder.state)&amp;#xA;                        console.log(&quot;recorder started&quot;)&amp;#xA;                        record.style.background = &quot;red&quot;&amp;#xA;                        record.style.color = &quot;black&quot;&amp;#xA;                    }&amp;#xA;&amp;#xA;                    stop.onclick = () =&gt; {&amp;#xA;                        mediaRecorder.stop()&amp;#xA;                        console.log(mediaRecorder.state)&amp;#xA;                        console.log(&quot;recorder stopped&quot;)&amp;#xA;                        record.style.background = &quot;&quot;&amp;#xA;                        record.style.color = &quot;&quot;&amp;#xA;                    }&amp;#xA;&amp;#xA;                    mediaRecorder.onstop = e =&gt; {&amp;#xA;                        console.log(&quot;data available after MediaRecorder.stop() called.&quot;)&amp;#xA;                        const bb = new Blob(chunks, { &amp;#x27;type&amp;#x27; : &amp;#x27;audio/wav&amp;#x27; })&amp;#xA;                        socket.emit(&amp;#x27;voice&amp;#x27;,bb)&amp;#xA;                        const clipName = prompt(&quot;오디오 파일 제목을 입력하세요.&quot;, new Date())&amp;#xA;&amp;#xA;                        const clipContainer = document.createElement(&amp;#x27;article&amp;#x27;)&amp;#xA;                        const clipLabel = document.createElement(&amp;#x27;p&amp;#x27;)&amp;#xA;                        const audio = document.createElement(&amp;#x27;audio&amp;#x27;)&amp;#xA;                        const deleteButton = document.createElement(&amp;#x27;button&amp;#x27;)&amp;#xA;&amp;#xA;                        clipContainer.classList.add(&amp;#x27;clip&amp;#x27;)&amp;#xA;                        audio.setAttribute(&amp;#x27;controls&amp;#x27;, &amp;#x27;&amp;#x27;)&amp;#xA;                        deleteButton.innerHTML = &quot;삭제&quot;&amp;#xA;                        clipLabel.innerHTML = clipName&amp;#xA;&amp;#xA;                        clipContainer.appendChild(audio)&amp;#xA;                        clipContainer.appendChild(clipLabel)&amp;#xA;                        clipContainer.appendChild(deleteButton)&amp;#xA;                        soundClips.appendChild(clipContainer)&amp;#xA;&amp;#xA;                        audio.controls = true&amp;#xA;                        const blob = new Blob(chunks, {&amp;#xA;                            &amp;#x27;type&amp;#x27;: &amp;#x27;audio/ogg codecs=opus&amp;#x27;&amp;#xA;                        })&amp;#xA;&amp;#xA;                        chunks = []&amp;#xA;                        const audioURL = URL.createObjectURL(blob)&amp;#xA;                        audio.src = audioURL&amp;#xA;                        console.log(&quot;recorder stopped&quot;)&amp;#xA;&amp;#xA;                        deleteButton.onclick = e =&gt; {&amp;#xA;                            evtTgt = e.target&amp;#xA;                            evtTgt  .parentNode.parentNode.removeChild(evtTgt.parentNode)&amp;#xA;                        }&amp;#xA;                    }&amp;#xA;&amp;#xA;                  mediaRecorder.ondataavailable = function(e) {&amp;#xA;                    chunks.push(e.data)&amp;#xA;                    if (chunks.length &gt;= 5)&amp;#xA;                    {&amp;#xA;                        const bloddb = new Blob(chunks, { &amp;#x27;type&amp;#x27; : &amp;#x27;audio/wav&amp;#x27; })&amp;#xA;                        socket.emit(&amp;#x27;voice&amp;#x27;, bloddb)&amp;#xA;                         &amp;#xA;                        chunks = []&amp;#xA;                    }&amp;#xA;                    mediaRecorder.sendData = function(buffer) {&amp;#xA;                        const bloddb = new Blob(buffer, { &amp;#x27;type&amp;#x27; : &amp;#x27;audio/wav&amp;#x27; })&amp;#xA;                        socket.emit(&amp;#x27;voice&amp;#x27;, bloddb)&amp;#xA;}&amp;#xA;};&amp;#xA;                })&amp;#xA;                .catch(err =&gt; {&amp;#xA;                    console.log(&amp;#x27;The following error occurred: &amp;#x27; &amp;#x2B; err)&amp;#xA;                })&amp;#xA;        }&amp;#xA;    &lt;/script&gt;&#xA;&#xA;

    &#xA;

    ask exception was never retrieved&#xA;future: <task finished="finished" coro="<InstrumentedAsyncServer._handle_event_internal()" defined="defined" at="at"> exception=CouldntDecodeError(&#x27;Decoding failed. ffmpeg returned error code: 3199971767\n\nOutput from ffmpeg/avlib:\n\nffmpeg version 6.1.1-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers\r\n  built with gcc 12.2.0 (Rev10, Built by MSYS2 project)\r\n  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\r\n  libavutil      58. 29.100 / 58. 29.100\r\n  libavcodec     60. 31.102 / 60. 31.102\r\n  libavformat    60. 16.100 / 60. 16.100\r\n  libavdevice    60.  3.100 / 60.  3.100\r\n  libavfilter     9. 12.100 /  9. 12.100\r\n  libswscale      7.  5.100 /  7.  5.100\r\n  libswresample   4. 12.100 /  4. 12.100\r\n  libpostproc    57.  3.100 / 57.  3.100\r\n[cache @ 000001d9828efe40] Inner protocol failed to seekback end : -40\r\n[matroska,webm @ 000001d9828efa00] EBML header parsing failed\r\n[cache @ 000001d9828efe40] Statistics, cache hits:0 cache misses:3\r\n[in#0 @ 000001d9828da3c0] Error opening input: Invalid data found when processing input\r\nError opening input file cache:pipe:0.\r\nError opening input files: Invalid data found when processing input\r\n&#x27;)>&#xA;Traceback (most recent call last):&#xA;  File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_admin.py", line 276, in _handle_event_internal&#xA;    ret = await self.sio.__handle_event_internal(server, sid, eio_sid,&#xA;          ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_server.py", line 597, in _handle_event_internal&#xA;    r = await server._trigger_event(data[0], namespace, sid, *data[1:])&#xA;        ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_server.py", line 635, in _trigger_event&#xA;    ret = handler(*args)&#xA;          ^^^^^^^^^^^^^^&#xA;  File "f:\fastapi-socketio-wb38\Python-Javascript-Websocket-Video-Streaming--main\poom2.py", line 153, in handle_voice&#xA;    audio_segment = AudioSegment.from_file(BytesIO(data), format="webm")&#xA;                    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^&#xA;  File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\pydub\audio_segment.py", line 773, in from_file&#xA;    raise CouldntDecodeError(&#xA;pydub.exceptions.CouldntDecodeError: Decoding failed. ffmpeg returned error code: 3199971767&#xA;&#xA;Output from ffmpeg/avlib:&#xA;&#xA;ffmpeg version 6.1.1-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers&#xA;  built with gcc 12.2.0 (Rev10, Built by MSYS2 project)&#xA;  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&#xA;  libavutil      58. 29.100 / 58. 29.100&#xA;  libavcodec     60. 31.102 / 60. 31.102&#xA;  libavformat    60. 16.100 / 60. 16.100&#xA;  libavdevice    60.  3.100 / 60.  3.100&#xA;  libavfilter     9. 12.100 /  9. 12.100&#xA;  libswscale      7.  5.100 /  7.  5.100&#xA;  libswresample   4. 12.100 /  4. 12.100&#xA;  libpostproc    57.  3.100 / 57.  3.100&#xA;[cache @ 000001d9828efe40] Inner protocol failed to seekback end : -40&#xA;[matroska,webm @ 000001d9828efa00] EBML header parsing failed&#xA;[cache @ 000001d9828efe40] Statistics, cache hits:0 cache misses:3&#xA;[in#0 @ 000001d9828da3c0] Error opening input: Invalid data found when processing input&#xA;Error opening input file cache:pipe:0.&#xA;Error opening input files: Invalid data found when processing input&#xA;</task>

    &#xA;

    im using version of ffmpeg-6.1.1-full_build.&#xA;i dont know this error exist the stop button sent event correctly. but chunk data was not work correctly in python server.&#xA;my english was so bad. sry

    &#xA;

  • ffmpeg giving Error while decoding stream #0:1 : Invalid data found when processing input

    11 octobre 2023, par user1432181

    I am trying to merge two video files into one using ffmpeg on Windows. The process has been proven to work over and over (with over 100 files merged together at some points) - but I have come across an input file that is causing the process to fail with the errors :

    &#xA;

    _[aac @ 00000142532f74c0] channel element 1.0 is not allocated&#xA;Error while decoding stream #0:1: Invalid data found when processing input&#xA;[aac @ 00000142532f74c0] channel element 1.0 is not allocated&#xA;Error while decoding stream #0:1: Invalid data found when processing input&#xA;[aac @ 00000142532f74c0] channel element 1.0 is not allocated&#xA;.&#xA;.&#xA;.&#xA;

    &#xA;

    There seems to be 3 command line steps to get here, using a concats-inputs.dat file containing :

    &#xA;

    file E:/..../snippet A.mp4&#xA;file E:/..../snippet B.mp4&#xA;

    &#xA;

    (Copies of these files can be found at https://filebin.net/77wbowvh7vbklkey/snippet_A.mp4 and https://filebin.net/77wbowvh7vbklkey/snippet_B.mp4)

    &#xA;

    Step 1 :

    &#xA;

    > ffmpeg-6.0-full_build/bin/ffmpeg -y -progress ".Default.mp4.progressinfo.dat" -vsync 0 -f concat -safe 0 -i "E:/...../concat-inputs.dat" -c:v copy -c:a copy -crf 0 -b:v 10M "E:/...../video.Default.mp4"&#xA;

    &#xA;

    with the output....

    &#xA;

    built with gcc 12.2.0 (Rev10, Built by MSYS2 project)&#xA;&#xA;  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-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-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --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-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&#xA;&#xA;  libavutil      58.  2.100 / 58.  2.100&#xA;&#xA;  libavcodec     60.  3.100 / 60.  3.100&#xA;&#xA;  libavformat    60.  3.100 / 60.  3.100&#xA;&#xA;  libavdevice    60.  1.100 / 60.  1.100&#xA;&#xA;  libavfilter     9.  3.100 /  9.  3.100&#xA;&#xA;  libswscale      7.  1.100 /  7.  1.100&#xA;&#xA;  libswresample   4. 10.100 /  4. 10.100&#xA;&#xA;  libpostproc    57.  1.100 / 57.  1.100&#xA;&#xA;-vsync is deprecated. Use -fps_mode&#xA;&#xA;Passing a number to -vsync is deprecated, use a string argument as described in the manual.&#xA;&#xA;[mov,mp4,m4a,3gp,3g2,mj2 @ 000001bf88ffe240] Auto-inserting h264_mp4toannexb bitstream filter&#xA;&#xA;Input #0, concat, from &#x27;E:/...../concat-inputs.dat&#x27;:&#xA;&#xA;  Duration: N/A, start: -0.010667, bitrate: 20382 kb/s&#xA;&#xA;  Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], 20043 kb/s, 50 fps, 50 tbr, 12800 tbn&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : VideoHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;      encoder         : Lavc60.3.100 libx264&#xA;&#xA;  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 339 kb/s&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : SoundHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;Output #0, mp4, to &#x27;E:/...../video.Default.mp4&#x27;:&#xA;&#xA;  Metadata:&#xA;&#xA;    encoder         : Lavf60.3.100&#xA;&#xA;  Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 10000 kb/s, 50 fps, 50 tbr, 12800 tbn&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : VideoHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;      encoder         : Lavc60.3.100 libx264&#xA;&#xA;  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 339 kb/s&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : SoundHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;Stream mapping:&#xA;&#xA;  Stream #0:0 -> #0:0 (copy)&#xA;&#xA;  Stream #0:1 -> #0:1 (copy)&#xA;&#xA;Press [q] to stop, [?] for help&#xA;&#xA;frame=    0 fps=0.0 q=-1.0 size=       0kB time=00:00:00.00 bitrate=N/A speed=N/A    &#xA;_[mov,mp4,m4a,3gp,3g2,mj2 @ 000001bf890653c0] Auto-inserting h264_mp4toannexb bitstream filter&#xA;&#xA;[mp4 @ 000001bf89000580] Non-monotonous DTS in output stream 0:1; previous: 180224, current: 180192; changing to 180225. This may result in incorrect timestamps in the output file.&#xA;&#xA;frame=  210 fps=0.0 q=-1.0 Lsize=   11537kB time=00:00:04.21 bitrate=22433.7kbits/s speed=41.9x&#xA;&#xA;video:11417kB audio:114kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.053312%&#xA;

    &#xA;

    Step 2

    &#xA;

    > ffmpeg-6.0-full_build/bin/ffmpeg -y -progress ".Default.mp4.progressinfo.dat" -vsync 0 -f concat -safe 0 -i "E:/...../concat-inputs.dat" -c:v copy -c:a copy -crf 0 -b:v 10M "E:/...../audio.Default.wav"&#xA;

    &#xA;

    which outputs...

    &#xA;

    ffmpeg version 6.0-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers&#xA;&#xA;  built with gcc 12.2.0 (Rev10, Built by MSYS2 project)&#xA;&#xA;  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-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-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --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-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&#xA;&#xA;  libavutil      58.  2.100 / 58.  2.100&#xA;&#xA;  libavcodec     60.  3.100 / 60.  3.100&#xA;&#xA;  libavformat    60.  3.100 / 60.  3.100&#xA;&#xA;  libavdevice    60.  1.100 / 60.  1.100&#xA;&#xA;  libavfilter     9.  3.100 /  9.  3.100&#xA;&#xA;  libswscale      7.  1.100 /  7.  1.100&#xA;&#xA;  libswresample   4. 10.100 /  4. 10.100&#xA;&#xA;  libpostproc    57.  1.100 / 57.  1.100&#xA;&#xA;-vsync is deprecated. Use -fps_mode&#xA;&#xA;Passing a number to -vsync is deprecated, use a string argument as described in the manual.&#xA;&#xA;[mov,mp4,m4a,3gp,3g2,mj2 @ 00000246d314e240] Auto-inserting h264_mp4toannexb bitstream filter&#xA;&#xA;Input #0, concat, from &#x27;E:/...../concat-inputs.dat&#x27;:&#xA;&#xA;  Duration: N/A, start: -0.010667, bitrate: 20382 kb/s&#xA;&#xA;  Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], 20043 kb/s, 50 fps, 50 tbr, 12800 tbn&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : VideoHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;      encoder         : Lavc60.3.100 libx264&#xA;&#xA;  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 339 kb/s&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : SoundHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;[out#0/wav @ 00000246d31bd240] Codec AVOption b (set bitrate (in bits/s)) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.&#xA;&#xA;Output #0, wav, to &#x27;E:/...../audio.Default.wav&#x27;:&#xA;&#xA;  Metadata:&#xA;&#xA;    ISFT            : Lavf60.3.100&#xA;&#xA;  Stream #0:0(und): Audio: aac (LC) ([255][0][0][0] / 0x00FF), 96000 Hz, 5.1, fltp, 339 kb/s&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : SoundHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;Stream mapping:&#xA;&#xA;  Stream #0:1 -> #0:0 (copy)&#xA;&#xA;Press [q] to stop, [?] for help&#xA;&#xA;size=       0kB time=00:00:00.00 bitrate=N/A speed=N/A    &#xA;_[mov,mp4,m4a,3gp,3g2,mj2 @ 00000246d3b009c0] Auto-inserting h264_mp4toannexb bitstream filter&#xA;&#xA;[wav @ 00000246d3150580] Non-monotonous DTS in output stream 0:0; previous: 180224, current: 180192; changing to 180224. This may result in incorrect timestamps in the output file.&#xA;&#xA;size=     114kB time=00:00:04.21 bitrate= 222.4kbits/s speed= 128x&#xA;&#xA;video:0kB audio:114kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.102561%&#xA;

    &#xA;

    Step 3

    &#xA;

    > ffmpeg-6.0-full_build/bin/ffmpeg -y -progress ".Default.mp4.progressinfo.dat" -i "E:/...../video.Default.mp4" -i "E:/...../audio.Default.wav" -crf 0 -c:v copy -c:a aac "E:/...../Default.mp4"&#xA;

    &#xA;

    ... which then gives the errors....

    &#xA;

    ffmpeg version 6.0-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers&#xA;&#xA;  built with gcc 12.2.0 (Rev10, Built by MSYS2 project)&#xA;&#xA;  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-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-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --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-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&#xA;&#xA;  libavutil      58.  2.100 / 58.  2.100&#xA;&#xA;  libavcodec     60.  3.100 / 60.  3.100&#xA;&#xA;  libavformat    60.  3.100 / 60.  3.100&#xA;&#xA;  libavdevice    60.  1.100 / 60.  1.100&#xA;&#xA;  libavfilter     9.  3.100 /  9.  3.100&#xA;&#xA;  libswscale      7.  1.100 /  7.  1.100&#xA;&#xA;  libswresample   4. 10.100 /  4. 10.100&#xA;&#xA;  libpostproc    57.  1.100 / 57.  1.100&#xA;&#xA;Input #0, mov,mp4,m4a,3gp,3g2,mj2, from &#x27;E:/...../video.Default.mp4&#x27;:&#xA;&#xA;  Metadata:&#xA;&#xA;    major_brand     : isom&#xA;&#xA;    minor_version   : 512&#xA;&#xA;    compatible_brands: isomiso2avc1mp41&#xA;&#xA;    encoder         : Lavf60.3.100&#xA;&#xA;  Duration: 00:00:04.23, start: 0.000000, bitrate: 22359 kb/s&#xA;&#xA;  Stream #0:0[0x1](und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], 22178 kb/s, 49.80 fps, 50 tbr, 12800 tbn (default)&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : VideoHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;      encoder         : Lavc60.3.100 libx264&#xA;&#xA;  Stream #0:1[0x2](und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 221 kb/s (default)&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : SoundHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;[aac @ 000001425315e580] Multiple frames in a packet.&#xA;&#xA;Input #1, wav, from &#x27;E:/...../audio.Default.wav&#x27;:&#xA;&#xA;  Metadata:&#xA;&#xA;    encoder         : Lavf60.3.100&#xA;&#xA;  Duration: 00:00:04.22, bitrate: 221 kb/s&#xA;&#xA;  Stream #1:0: Audio: aac (LC) ([255][0][0][0] / 0x00FF), 96000 Hz, 5.1, fltp, 339 kb/s&#xA;&#xA;Stream mapping:&#xA;&#xA;  Stream #0:0 -> #0:0 (copy)&#xA;&#xA;  Stream #0:1 -> #0:1 (aac (native) -> aac (native))&#xA;&#xA;Press [q] to stop, [?] for help&#xA;&#xA;Output #0, mp4, to &#x27;E:/...../Default.mp4&#x27;:&#xA;&#xA;  Metadata:&#xA;&#xA;    major_brand     : isom&#xA;&#xA;    minor_version   : 512&#xA;&#xA;    compatible_brands: isomiso2avc1mp41&#xA;&#xA;    encoder         : Lavf60.3.100&#xA;&#xA;  Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 22178 kb/s, 49.80 fps, 50 tbr, 12800 tbn (default)&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : VideoHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;      encoder         : Lavc60.3.100 libx264&#xA;&#xA;  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 341 kb/s (default)&#xA;&#xA;    Metadata:&#xA;&#xA;      handler_name    : SoundHandler&#xA;&#xA;      vendor_id       : [0][0][0][0]&#xA;&#xA;      encoder         : Lavc60.3.100 aac&#xA;&#xA;frame=    0 fps=0.0 q=-1.0 size=       0kB time=-577014:32:22.77 bitrate=  -0.0kbits/s speed=N/A    &#xA;_[aac @ 00000142532f74c0] channel element 1.0 is not allocated&#xA;&#xA;Error while decoding stream #0:1: Invalid data found when processing input&#xA;&#xA;[aac @ 00000142532f74c0] channel element 1.0 is not allocated&#xA;.&#xA;.&#xA;.&#xA;

    &#xA;

    If I was to do this to merge snippet B with snippet B then it would work - it's something about snippet A that is causing the problem.

    &#xA;

    Is there any way to get around this... what is it about snippet A that is causing a problem... and is there a way to "normalize" it so that it can be merged as part of the "set".

    &#xA;

    Note, I just upgraded to ffmpeg6 after a previous version was giving the same problems - so I will also work on the deprecated messages when I can.

    &#xA;

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