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• Android - Decoding via a pipe will not work : Could not find an ffmpeg binary for your system

  28 août 2017, par Daniele

  I’m trying to use TarsosDSP for real time pitch shifting on Android.

  This is my code :

  Uri song;
  
  // initialized in another method
  
  
  
  double rate = 1.0;
  
          RateTransposer rateTransposer;
  
          AudioDispatcher dispatcher;
  
          WaveformSimilarityBasedOverlapAdd wsola;
  
  
  
          dispatcher = AudioDispatcherFactory.fromPipe(getRealPathFromUri(song), 44100, 5000, 2500);
  
          rateTransposer = new RateTransposer(rate);
  
          wsola = new WaveformSimilarityBasedOverlapAdd(WaveformSimilarityBasedOverlapAdd.Parameters
  
                  .musicDefaults(rate, 44100));
  
  
  
          wsola.setDispatcher(dispatcher);
  
          dispatcher.addAudioProcessor(wsola);
  
          dispatcher.addAudioProcessor(rateTransposer);
  
          dispatcher.addAudioProcessor(new AndroidAudioPlayer(dispatcher.getFormat()));
  
          dispatcher.setZeroPadFirstBuffer(true);
  
          dispatcher.setZeroPadLastBuffer(true);

  I get an error here

  dispatcher = AudioDispatcherFactory.fromPipe(getRealPathFromUri(song), 44100, 5000, 2500);

  Decoding via a pipe will not work : Could not find an ffmpeg binary for
  your system

  Why does this happen and how should I fix it ?

  EDIT :

  As far as I was able to understand it’s because ffmpeg isn’t integrated within the app. I looked for a guide here on SO but I couldn’t find any updated one. Using NDK r15c and FFmpeg 3.3.3

• ffmpeg highest quality mp4 to mpg

  29 septembre 2017, par scopa

  I have a mp4 video that I need to convert to mpg (for windows PowerPoint2010)

  I have been trying to get best quality. But I keep getting error :

  [mpeg @ 0x2523620] buffer underflow st=0 bufi=420177 size=445860
  
  [mpeg @ 0x2523620] packet too large, ignoring buffer limits to mux it
  
  [mpeg @ 0x2523620] buffer underflow st=0 bufi=420177 size=445860
  
  [mpeg @ 0x2523620] buffer underflow st=0 bufi=422218 size=445860

  Could someone help me with the syntax for best quality ouput to mpg. Here is the output of the mp4 file :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4':
  
    Metadata:
  
      major_brand     : mp42
  
      minor_version   : 0
  
      compatible_brands: mp41isom
  
      creation_time   : 2016-06-10 11:15:06
  
    Duration: 00:04:20.86, start: 0.000000, bitrate: 18677 kb/s
  
      Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1920x1080, 18541 kb/s, 29.97 fps, 29.97 tbr, 29970 tbn, 59.94 tbc (default)
  
      Metadata:
  
        creation_time   : 2016-06-10 11:15:06
  
        handler_name    : VideoHandler
  
        encoder         : AVC Coding
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 132 kb/s (default)
  
      Metadata:
  
        creation_time   : 2016-06-10 11:15:06
  
        handler_name    : SoundHandler

  I have tried the following but keep getting that error :

  ffmpeg -i video.mp4 -c:v libx264 -c:a copy -qp 5 video.mpg
  
  ffmpeg -i video.mp4 -c:v libx264 -c:a copy -qscale:v 1 video.mpg
  
  ffmpeg -i video.mp4 -c:v libx264 -crf 0 -c:a copy -bf 2 -flags qprd -flags mv0 video.mpg
  
  ffmpeg -i video.mp4 -c:v libx264 -crf 0 -c:a copy video.mpg
  
  ffmpeg -i video.mp4 -c:v libx264 -preset slow -crf 5 -c:a copy video.mpg
  
  ffmpeg -i video.mp4 -c:v libx264 -preset slow -crf 5 -c:a copy -maxrate 11000k video.mpg
  
  ffmpeg -i video.mp4 -c:v libx264 -preset slow -crf 5 -c:a copy -maxrate 5000 -bufsize 11000 video.mpg

  Thanks,

• FFplay : WASAPI can't initialize audio client (FFmpeg 3.4 binaries)

  9 juillet 2022, par fve

  I am trying to playback a stream that is multicasted over RTP using ffplay on Windows 7 64 bits.
The computer that serves the audio over RTP runs :

  



  ffmpeg -re -f dshow -audio_buffer_size 15 -ac 1 -i audio="Mic in at front Mic-in (Realtek" -ar 8k -acodec pcm_alaw -vn -f rtp rtp://127.0.0.1:5000


  



  The client runs :

  



  ffplay rtp://127.0.0.1:5000


  



  While this was working correctly in release 3.3, release 3.4 fails with this error :

  



  SDL_OpenAudio (1 channels, 8000 Hz): WASAPI can't initialize audio client: CoInitialize has not been called.

No more combinations to try, audio open failed
Failed to open file 'rtp://127.0.0.1:5000' or configure filtergraph


  



  Anyone has an idea of whats going on ?

  



  Thanks

  



  UPDATE #1 :

  



  A workaround is to specify an alternate output driver.
Set SDL_AUDIODRIVER environment variable value to the driver you want to use.
Under windows 7, directsound and winmm both solved my issue.