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• sharedarraybuffer error in edge and firefox but not in chrome when uploading with ffmpeg in nodejs

  29 juillet 2021, par Juliette

  I have the code below in React.js that uses ffmpeg to convert files :

  


  import { createFFmpeg, fetchFile } from '@ffmpeg/ffmpeg';

  const doTranscode = async () => {
    setMessage('Loading ffmpeg-core.js');
    await ffmpeg.load();
    setMessage('Uploading file');
    ffmpeg.FS('writeFile', filename, await fetchFile(file));
    await ffmpeg.run('-i', filename, 'test.wav');
    setMessage('Upload to manager complete. Sound file will be available and playable on the manger within 1-2 minutes.');
    const data = ffmpeg.FS('readFile', 'test.wav');
    setAudioSrc(URL.createObjectURL(new Blob([data.buffer], { type: 'audio/wav' })));

    var file_name = prompt('What would you like to call this file?');

    if (!file_name) {
      file_name = Date.now()
    }
    
    (async function(){
      let output = await getSoundID(customer_id, file_name);
      let sound_id = output.data;
      var bucket_file = new File([new Blob([data.buffer], { type: 'audio/wav' })], "sounds/" + customer_id + "/" + sound_id + ".wav");
      uploadFileToS3(bucket_file);
      updateSoundData(sound_id, customer_id);
    })();
  };

  useEffect(() => {
    if (file) {
      doTranscode()
    }
  }, [file])


  


  The code above works great in Chrome and the files are successfully converted. However, when I bring it to Firefox or Edge I get this error Unhandled Rejection (ReferenceError): SharedArrayBuffer is not defined.

  


  I looked up this issue and they said I need to modify my headers to include this :

  


  You need to set two response headers for your document:

Cross-Origin-Opener-Policy: same-origin Cross-Origin-Embedder-Policy: require-corp


  


  Not sure how I would this in my JS code ?

  


  Would love to hear what you guys think.

  


• Transfer audio with ffmpeg, unexpected silent at the beginning [duplicate]

  4 mai 2018, par LouisCJT

  This question already has an answer here :

  • ffmpeg wrong audio file after conversion in AAC
    
    1 answer
    
  • Why does libmp3lame add zeros to the start of the MP3 ?
    
    1 answer
    

  I was using ffmpeg to transfer the format of audio files, like from .wav to mp3. The transfer seemed fine and the audio sounds as expected. However when I check its waveform, I found that after the transfer,

  1. A small portion of silent (and with some very tiny noise) is at the beginning of the output audio. The length is about 1 ms 0.001 sec.

  2. As the silent part is appended, the end of the audio is altered - the length of the output audio is longer than the input file. (my input is exactly 10 sec, now the output is around 10.1 sec.

  Need the transfer precise because need to do further analysis frame by frame, and this situation could not go well. I used the following command to transfer a file.

  ffmpeg -I ..\wav_1K_32bit_24576kbps_384000Hz_stereo.wav -vn -ar 12000 -ac 2 -ab 320000 -f MP3 MP3_12000Hz_32kbps_stereo_VBROff.MP3

  ---

  Please refer to the screen crop of waveform view.

  Beginning of audio, input (U) & output (L)

  End of audio, input (U) & output (L)

  Thank you !

• Why am I getting FPE when using swresample 1.1 ?

  24 janvier 2013, par mystafer

  I am building a project to view a video feed from an IP camera in Android using FFmpeg 1.1.

  I'm attempting to use swresample in an Android project and getting a floating point exception when calling swr_convert. I stepped through the swresample code and found one line in libswresample/swresample.c function swri_realloc_audio where the variables a->bps and a->ch_count are zero causing the FPE.

  int swri_realloc_audio(AudioData *a, int count){
  
      int i, countb;
  
      AudioData old;
  
  
  
      LOGD("in swri_realloc_audio - bps[%d], ch_count[%d]", a->bps, a->ch_count);
  
      if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
  
      return AVERROR(EINVAL);

  01-21 17:29:09.612 : D/swresample.c(18789) : in swri_realloc_audio - bps[0], ch_count[0]

  I found bug ticket #1834 in the FFmpeg project that sounds like the exact same issue, but it was resolved by calling swr_init. However, my code does call this function and still crashes. Here is my JNI code :

  SwrContext* resampleCtx = swr_alloc_set_opts(NULL,
  
          AV_CH_LAYOUT_MONO, AV_SAMPLE_FMT_S16, pAudioCodecCtx->sample_rate,
  
          pAudioCodecCtx->channel_layout, pAudioCodecCtx->sample_fmt,
  
          pAudioCodecCtx->sample_rate, 0, 0);
  
  
  
  swr_init(resampleCtx);
  
  LOGD("Resample context initialized");
  
  
  
  int dataSize = swr_convert(resampleCtx,
  
          &pAudioOutBuffer, AVCODEC_MAX_AUDIO_FRAME_SIZE / 2,
  
          (const uint8_t**) &(pFrame->data[0]), pFrame->nb_samples);
  
  LOGD("Resample conversion complete");
  
  
  
  swr_free(&resampleCtx);
  
  LOGD("Obtained data size - dataSize[%d]", dataSize);

  I'm confused because I don't seem to have any control over the variable a in the swri_realloc_audio function. I stepped through the code and noticed that it is from the variable resampleCtx->postin. This variable is copied from resampleCtx->in in the swr_init function, but I don't see where in is ever set to anything.

  What am I doing wrong ? Is it in my code or is there a problem in swresample ?