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Revolution of Open-source and film making towards open film making
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Sur d’autres sites (10725)
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ffmpeg giving Error while decoding stream #0:1 : Invalid data found when processing input
11 octobre 2023, par user1432181I 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 :


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


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


file E:/..../snippet A.mp4
file E:/..../snippet B.mp4


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


Step 1 :


> 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"


with the output....


built with gcc 12.2.0 (Rev10, 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-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

 libavutil 58. 2.100 / 58. 2.100

 libavcodec 60. 3.100 / 60. 3.100

 libavformat 60. 3.100 / 60. 3.100

 libavdevice 60. 1.100 / 60. 1.100

 libavfilter 9. 3.100 / 9. 3.100

 libswscale 7. 1.100 / 7. 1.100

 libswresample 4. 10.100 / 4. 10.100

 libpostproc 57. 1.100 / 57. 1.100

-vsync is deprecated. Use -fps_mode

Passing a number to -vsync is deprecated, use a string argument as described in the manual.

[mov,mp4,m4a,3gp,3g2,mj2 @ 000001bf88ffe240] Auto-inserting h264_mp4toannexb bitstream filter

Input #0, concat, from 'E:/...../concat-inputs.dat':

 Duration: N/A, start: -0.010667, bitrate: 20382 kb/s

 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

 Metadata:

 handler_name : VideoHandler

 vendor_id : [0][0][0][0]

 encoder : Lavc60.3.100 libx264

 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 339 kb/s

 Metadata:

 handler_name : SoundHandler

 vendor_id : [0][0][0][0]

Output #0, mp4, to 'E:/...../video.Default.mp4':

 Metadata:

 encoder : Lavf60.3.100

 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

 Metadata:

 handler_name : VideoHandler

 vendor_id : [0][0][0][0]

 encoder : Lavc60.3.100 libx264

 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 339 kb/s

 Metadata:

 handler_name : SoundHandler

 vendor_id : [0][0][0][0]

Stream mapping:

 Stream #0:0 -> #0:0 (copy)

 Stream #0:1 -> #0:1 (copy)

Press [q] to stop, [?] for help

frame= 0 fps=0.0 q=-1.0 size= 0kB time=00:00:00.00 bitrate=N/A speed=N/A 
_[mov,mp4,m4a,3gp,3g2,mj2 @ 000001bf890653c0] Auto-inserting h264_mp4toannexb bitstream filter

[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.

frame= 210 fps=0.0 q=-1.0 Lsize= 11537kB time=00:00:04.21 bitrate=22433.7kbits/s speed=41.9x

video:11417kB audio:114kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.053312%


Step 2


> 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"


which outputs...


ffmpeg version 6.0-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 --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

 libavutil 58. 2.100 / 58. 2.100

 libavcodec 60. 3.100 / 60. 3.100

 libavformat 60. 3.100 / 60. 3.100

 libavdevice 60. 1.100 / 60. 1.100

 libavfilter 9. 3.100 / 9. 3.100

 libswscale 7. 1.100 / 7. 1.100

 libswresample 4. 10.100 / 4. 10.100

 libpostproc 57. 1.100 / 57. 1.100

-vsync is deprecated. Use -fps_mode

Passing a number to -vsync is deprecated, use a string argument as described in the manual.

[mov,mp4,m4a,3gp,3g2,mj2 @ 00000246d314e240] Auto-inserting h264_mp4toannexb bitstream filter

Input #0, concat, from 'E:/...../concat-inputs.dat':

 Duration: N/A, start: -0.010667, bitrate: 20382 kb/s

 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

 Metadata:

 handler_name : VideoHandler

 vendor_id : [0][0][0][0]

 encoder : Lavc60.3.100 libx264

 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 339 kb/s

 Metadata:

 handler_name : SoundHandler

 vendor_id : [0][0][0][0]

[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.

Output #0, wav, to 'E:/...../audio.Default.wav':

 Metadata:

 ISFT : Lavf60.3.100

 Stream #0:0(und): Audio: aac (LC) ([255][0][0][0] / 0x00FF), 96000 Hz, 5.1, fltp, 339 kb/s

 Metadata:

 handler_name : SoundHandler

 vendor_id : [0][0][0][0]

Stream mapping:

 Stream #0:1 -> #0:0 (copy)

Press [q] to stop, [?] for help

size= 0kB time=00:00:00.00 bitrate=N/A speed=N/A 
_[mov,mp4,m4a,3gp,3g2,mj2 @ 00000246d3b009c0] Auto-inserting h264_mp4toannexb bitstream filter

[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.

size= 114kB time=00:00:04.21 bitrate= 222.4kbits/s speed= 128x

video:0kB audio:114kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.102561%


Step 3


> 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"


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


ffmpeg version 6.0-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 --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

 libavutil 58. 2.100 / 58. 2.100

 libavcodec 60. 3.100 / 60. 3.100

 libavformat 60. 3.100 / 60. 3.100

 libavdevice 60. 1.100 / 60. 1.100

 libavfilter 9. 3.100 / 9. 3.100

 libswscale 7. 1.100 / 7. 1.100

 libswresample 4. 10.100 / 4. 10.100

 libpostproc 57. 1.100 / 57. 1.100

Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'E:/...../video.Default.mp4':

 Metadata:

 major_brand : isom

 minor_version : 512

 compatible_brands: isomiso2avc1mp41

 encoder : Lavf60.3.100

 Duration: 00:00:04.23, start: 0.000000, bitrate: 22359 kb/s

 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)

 Metadata:

 handler_name : VideoHandler

 vendor_id : [0][0][0][0]

 encoder : Lavc60.3.100 libx264

 Stream #0:1[0x2](und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 221 kb/s (default)

 Metadata:

 handler_name : SoundHandler

 vendor_id : [0][0][0][0]

[aac @ 000001425315e580] Multiple frames in a packet.

Input #1, wav, from 'E:/...../audio.Default.wav':

 Metadata:

 encoder : Lavf60.3.100

 Duration: 00:00:04.22, bitrate: 221 kb/s

 Stream #1:0: Audio: aac (LC) ([255][0][0][0] / 0x00FF), 96000 Hz, 5.1, fltp, 339 kb/s

Stream mapping:

 Stream #0:0 -> #0:0 (copy)

 Stream #0:1 -> #0:1 (aac (native) -> aac (native))

Press [q] to stop, [?] for help

Output #0, mp4, to 'E:/...../Default.mp4':

 Metadata:

 major_brand : isom

 minor_version : 512

 compatible_brands: isomiso2avc1mp41

 encoder : Lavf60.3.100

 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)

 Metadata:

 handler_name : VideoHandler

 vendor_id : [0][0][0][0]

 encoder : Lavc60.3.100 libx264

 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 96000 Hz, 5.1, fltp, 341 kb/s (default)

 Metadata:

 handler_name : SoundHandler

 vendor_id : [0][0][0][0]

 encoder : Lavc60.3.100 aac

frame= 0 fps=0.0 q=-1.0 size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
_[aac @ 00000142532f74c0] channel element 1.0 is not allocated

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

[aac @ 00000142532f74c0] channel element 1.0 is not allocated
.
.
.


If I was to do this to merge
snippet Bwithsnippet Bthen it would work - it's something aboutsnippet Athat is causing the problem.

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

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.


-
ffmpeg Audiosegment error in get audio chunks in socketIo server in python
26 janvier 2024, par a_crszkvc30Last_NameColI 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






 
 
 <code class="echappe-js"><script src="https://cdnjs.cloudflare.com/ajax/libs/socket.io/4.5.2/socket.io.js"></script>



 






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



ask exception was never retrieved
future: <task finished="finished" coro="<InstrumentedAsyncServer._handle_event_internal()" defined="defined" at="at"> exception=CouldntDecodeError('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')>
Traceback (most recent call last):
 File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_admin.py", line 276, in _handle_event_internal
 ret = await self.sio.__handle_event_internal(server, sid, eio_sid,
 ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
 File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_server.py", line 597, in _handle_event_internal
 r = await server._trigger_event(data[0], namespace, sid, *data[1:])
 ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
 File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\socketio\async_server.py", line 635, in _trigger_event
 ret = handler(*args)
 ^^^^^^^^^^^^^^
 File "f:\fastapi-socketio-wb38\Python-Javascript-Websocket-Video-Streaming--main\poom2.py", line 153, in handle_voice
 audio_segment = AudioSegment.from_file(BytesIO(data), format="webm")
 ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
 File "f:\fastapi-socketio-wb38\.vent\Lib\site-packages\pydub\audio_segment.py", line 773, in from_file
 raise CouldntDecodeError(
pydub.exceptions.CouldntDecodeError: Decoding failed. ffmpeg returned error code: 3199971767

Output from ffmpeg/avlib:

ffmpeg version 6.1.1-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. 29.100 / 58. 29.100
 libavcodec 60. 31.102 / 60. 31.102
 libavformat 60. 16.100 / 60. 16.100
 libavdevice 60. 3.100 / 60. 3.100
 libavfilter 9. 12.100 / 9. 12.100
 libswscale 7. 5.100 / 7. 5.100
 libswresample 4. 12.100 / 4. 12.100
 libpostproc 57. 3.100 / 57. 3.100
[cache @ 000001d9828efe40] Inner protocol failed to seekback end : -40
[matroska,webm @ 000001d9828efa00] EBML header parsing failed
[cache @ 000001d9828efe40] Statistics, cache hits:0 cache misses:3
[in#0 @ 000001d9828da3c0] Error opening input: Invalid data found when processing input
Error opening input file cache:pipe:0.
Error opening input files: Invalid data found when processing input
</task>

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


-
Transcode webcam blob to RTMP using ffmpeg.wasm
29 novembre 2023, par hassan moradnezhadI'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
transcodeit's goal is going to send data to rtmp server using useffmpeg.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.wasmtry to execute the last command.

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

it just calls aGET Request, I will send further details about this request.

as u can see, i try to use lots of arg sample withffmpeg?.exec, but non of them works.

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



it send a
GET requesttows://localhost:1935/
and nothing happened after that.

for backend, i use node-media-server and here is my output logs when
ffmpeg.wasmtrying 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 ?








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