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• Ffmpeg freezes while decoding

  22 mars 2017, par J.Clarke

  FFmpeg freezes while decoding a small mp3 file. Thecrash happens conistantly after proscessing about 8 or 9 data buffers. I have no clue as to what is happening.

             await e.Channel.SendMessage("Playing");
  
              var server = e.Server;
  
              var voiceChannel = client.FindServers(server.Name).FirstOrDefault().VoiceChannels.FirstOrDefault();
  
              var _vClient = await client.GetService<audioservice>()
  
                  .Join(voiceChannel);
  
  
  
              var pathOrUrl = "a.mp3";
  
  
  
              var process = Process.Start(new ProcessStartInfo
  
              { // FFmpeg requires us to spawn a process and hook into its stdout, so we will create a Process
  
                  FileName = "ffmpeg",
  
                  Arguments = $"-i {pathOrUrl} " + // Here we provide a list of arguments to feed into FFmpeg. -i means the location of the file/URL it will read from
  
                              "-f s16le -ar 48000 -ac 1 pipe:1", // Next, we tell it to output 16-bit 48000Hz PCM, over 2 channels, to stdout.
  
                  UseShellExecute = false,
  
                  RedirectStandardOutput = true // Capture the stdout of the process
  
              });
  
              Thread.Sleep(2000); // Sleep for a few seconds to FFmpeg can start processing data.
  
  
  
              int blockSize = 3840; // The size of bytes to read per frame; 1920 for mono
  
              byte[] buffer = new byte[blockSize];
  
              int byteCount;
  
  
  
              while (true) // Loop forever, so data will always be read
  
              {
  
                  byteCount = process.StandardOutput.BaseStream // Access the underlying MemoryStream from the stdout of FFmpeg
  
                          .Read(buffer, 0, blockSize); // Read stdout into the buffer
  
  
  
                  if (byteCount == 0) { // FFmpeg did not output anything
  
                      Console.WriteLine("Finished");
  
                      break;
  
                      }// Break out of the while(true) loop, since there was nothing to read.
  
  
  
                  _vClient.Send(buffer, 0, byteCount); // Send our data to Discord
  
              }
  
              _vClient.Wait();
  
  </audioservice>

• Windows(WASAPI):Converting IEEE float PCM to PCM 16bit

  10 octobre 2017, par Michael IV

  I do a hook into some of WASAPI methods to read out audio buffer that contains system sound.

  The source is WAV with audio format = WAVE_FORMAT_IEEE_FLOAT and

  subformat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT

  I am trying to convert it to PCM 16bit on the fly using FFMPEG libswresample. The resulting soundtrack has alot of forground noise with the original track’s data playing in the background.

  Here is how I do it (I skip WASAPI init part) :

  First,the source WAVEFORMATEX struct has the following properties :

  Format tag:65534
  
  Sample rate:44100
  
  Num channels:2
  
  CB size:22
  
  Average bytes per sec:352800
  
  Block align:8
  
  Bits per sample:32

  Where 65534 stands for WAVE_FORMAT_IEEE_FLOAT.

   WAVEFORMATEX *w;//filled with the props above
  
   int samples = 0;  
  
   int bufferSize = 0;
  
   int buffer_frames = 0; 
  
   int bWritten = 0;
  
   char* bufferIn = NULL;
  
   char* bufferOut = NULL;
  
   BYTE* wavBuffer = NULL;
  
   void InitSWRContext()
  
   {
  
      swrctx = swr_alloc_set_opts(NULL,
  
      AV_CH_LAYOUT_STEREO, //stereo layout
  
      AV_SAMPLE_FMT_S16, // sample out format
  
      44100, // out sample rate
  
      2, //number of channels 
  
      AV_SAMPLE_FMT_FLT, //sample in format
  
      w->nSamplesPerSec ,//in sample rate
  
      0,
  
     NULL);
  
  
  
      swr_init(swrctx);
  
  
  
   //also tried like this:
  
    /*
  
     swrctx = swr_alloc();
  
     av_opt_set_int(swrctx, "in_channel_layout", CA2SWR_chlayout(w->nChannels), 0);
  
     av_opt_set_int(swrctx, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
  
     av_opt_set_int(swrctx, "in_sample_rate", 44100, 0);
  
     av_opt_set_int(swrctx, "out_sample_rate", 44100, 0);
  
     av_opt_set_sample_fmt(swrctx, "in_sample_fmt", AV_SAMPLE_FMT_FLT, 0);
  
     av_opt_set_sample_fmt(swrctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
  
  
  
     */
  
  
  
      samples = (int)av_rescale_rnd(CA_MAX_SAMPLES, 44100, w->nSamplesPerSec, 
  
      AV_ROUND_UP);
  
      bufferSize = av_samples_get_buffer_size(NULL,2, samples * 2,AV_SAMPLE_FMT_S16, 1/*no-alignment*/);
  
      bufferOut = ( char *)malloc(bufferSize ))
  
  
  
   }

  These two methods are invoked by the system and I hook them to access audio buffer :

    DEXPORT HRESULT __stdcall
  
  _GetBuffer(
  
  IAudioRenderClient *thiz,
  
  UINT32 NumFramesRequested,
  
  BYTE **ppData)
  
  {
  
      HRESULT hr;
  
      hr = orig_GetBuffer(thiz, NumFramesRequested, ppData);
  
      bufferIn = (char*)*ppData;
  
      buffer_frames = NumFramesRequested;
  
      return S_OK;
  
  }

  Then on buffer release I perform the conversion and readout.

   DEXPORT HRESULT __stdcall
  
  _ReleaseBuffer(IAudioRenderClient *thiz,
  
  UINT32 NumFramesWritten, DWORD dwFlags)
  
  {
  
      const unsigned char *srcplanes[2];
  
      unsigned char *dstplanes[2];
  
      int samples;
  
      srcplanes[0] = (unsigned char*)bufferIn;
  
      srcplanes[1] = NULL;
  
      dstplanes[0] = (unsigned char*)bufferOut; 
  
      dstplanes[1] = NULL;
  
      samples = (int)av_rescale_rnd(NumFramesWritten, 44100, 44100, AV_ROUND_UP);
  
      int samplesConverted = swr_convert(swrctx, dstplanes, samples , srcplanes, NumFramesWritten);
  
      int framesize = (/*PCM16*/16 / 8) * pwfx->nChannels * NumFramesWritten;
  
      if (!wavBuffer)
  
      {
  
        wavBuffer = (BYTE*)malloc(4096 * 1000); //write out a few seconds
  
      }
  
      //copy converted buffer into wavBuffer
  
      memcpy(wavBuffer + bWritten, bufferOut, framesize);
  
      bWritten += framesize;
  
      if (bWritten > 4096 * 1000)
  
      {
  
          pwfx->wFormatTag = WAVE_FORMAT_PCM;
  
          pwfx->wBitsPerSample = 16;
  
          pwfx->nBlockAlign = (pwfx->nChannels * pwfx->wBitsPerSample) / 8;
  
          pwfx->cbSize = 0;
  
          SaveWaveData(wavBuffer, bWritten, pwfx, L"test.wav");
  
  
  
          exit(0);
  
       }
  
  
  
    }

  SaveWaveData borrowed from here.

  Resulting sound file(WAV)

  Original file : a2002011001-e02.wav

  I tried to play and analyze the output in VLC,Audacity, and FMOD Studio.
  The strange thing is that VLC show (codec info) that it is PCM16, while FMOD studio interprets the data as PCM32. I also tried to store without conversion the original buffer which also produces a sound with noise,though not as significant as when converting to PCM16

• MPEG DASH (MPD) to MP4 in Node.js ?

  28 janvier 2024, par Hello World

  I am trying to convert streams to .mp4 files. I have successfully converted an HLS to MP4 using fluent-ffmpeg package on version 2.1.2. I did so with with the following code :

  


  var ffmpegCommand = ffmpeg();
            ffmpegCommand
            .input(HLS FILE HERE)
            .inputOptions('-protocol_whitelist file,http,https,tcp,tls,crypto,data')
            .on("error", (err, stdout, stderr) => {
                console.log(err)
                let error = new Error(err.message)
                error.code = err.code || 'FILE_CONVERSION_M3U8_TO_MP3/UNKNOWN_ERROR';
                error.status = err.status || 500;

                console.log('An error occurred: ' + err.message, err, stderr);
                reject(error)
            })
            .on("end", () => {
                console.log(temporaryFilePath2)
                resolve(temporaryFilePath2);
            })
            .on('progress', function(progress) {
                console.log('progress: ' + (progress.percent || 0).toFixed(2) + '%。');
                progressGathering.push((progress.percent || 0).toFixed(2))

                if(progressGathering.length == 10 && progressGathering[9] == 0.00){
                    let error = new Error("File is a live stream.")
                    error.code = 'FILE_CONVERSION_M3U8_TO_MP3/LIVESTREAM_SUPPORT'
                    error.status = 501

                    ffmpegCommand.kill()
                    reject(error)
                }
            })
            .inputOptions('-allowed_extensions ALL')
            .outputOptions("-c copy")
            .output(temporaryFilePath2)
            .run();

            setTimeout(() => {
                ffmpegCommand.kill();
                let error = new Error("The file selected was too large or the network was hanging. Conversion timeout.")
                error.code = 'FILE_CONVERSION_M3U8_TO_MP3/TIMEOUT'
                error.status = 599;
                reject(error)
            }, 300000) // 5 minutes


  


  However, when I attempt something similar with a mpd file url in the input, I get several errors that are all the same :

  


  ffmpeg exited with code 1: https://dash.akamaized.net/akamai/bbb_30fps/bbb_30fps.mpd: Invalid data found when processing input

  


  Is this something fluent-ffmpeg can't handle ? If so, I would like to know another package that can accomplish this.