Recherche avancée

Sur d’autres sites (13866)

• Error in converting audio file format from ogg to wav [on hold]

  9 juin 2014, par Sumit Bisht

  I am trying to convert an ogg format file that was created using webrtc (html5 usermedia content generated on firefox) and transferred and decoded on the server into a wav file through ffmpeg but am getting this error on cmmand line while trying to convert :

  $ ffmpeg -i 2014-6-5_16-17-54.ogg res1.wav
  
  ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on May  1 2014 13:12:12 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4)
  
    configuration: --enable-gpl --enable-version3 --enable-shared --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid
  
    libavutil      52. 38.100 / 52. 38.100
  
    libavcodec     55. 18.102 / 55. 18.102
  
    libavformat    55. 12.100 / 55. 12.100
  
    libavdevice    55.  3.100 / 55.  3.100
  
    libavfilter     3. 79.101 /  3. 79.101
  
    libswscale      2.  3.100 /  2.  3.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Guessed Channel Layout for  Input Stream #0.0 : mono
  
  Input #0, ogg, from '2014-6-5_16-17-54.ogg':
  
    Duration: 00:00:01.84, start: 0.000000, bitrate: 18 kb/s
  
      Stream #0:0: Audio: opus, 48000 Hz, mono
  
      Metadata:
  
        ENCODER         : Mozilla29.0.1
  
  [graph 0 input from stream 0:0 @ 0x18dca20] Invalid sample format (null)
  
  Error opening filters!

  Although, I am able to play the file on server and using the same command, am able to convert .ogg files generated somewhere else. What might be I missing ?

  Edit :
  Here’s the source code that is used to write to the file :

  1) During startup - use the methods of getUserMedia API.

  navigator.getUserMedia({
  
          audio: true,
  
          video: false
  
      }, function(stream) {
  
          audioStream = RecordRTC(stream, {
  
              bufferSize: 16384
  
          });
  
          audioStream.startRecording();

  2) During stopping of the recording - extracting the recorded information.

  function(audioDataURL) {
  
       var audioFile = {};
  
       audioFile = {
  
         contents: audioDataURL
  
       **strong text**};

  On server end, the following code is creating a file from this data :

  dataURL = dataURL.split(',').pop(); // dataURL is the audioDataURL as defined above
  
  fileBuffer = new Buffer(dataURL, 'base64');
  
  fs.writeFileSync(filePath, fileBuffer);

• Revision b98a1a43ce41afe3be303d168e308dacbb8d2d30 : Mais non, là on est revenu à la case départ. La confusion c’était entre ...

  15 septembre 2010, par Committo,Ergo:sum — Log

  Mais non, là on est revenu à la case départ. La confusion c’était entre true et continue de l’argument option de sql_getfetsel. Ce coup-ci on a bien à la fois le logout correct, le login automatique à l’install, et l’install forcée en cas de fichier corrompu. git-svn-id : (...)

• Podcast Producer 2 tip - running xgrid jobs as logged in user

  19 octobre 2009

  So I’ve been playing with an interesting "feature" in PCP2 - the "chapterize" command generates different results when it can talk to the window server versus when it can’t. In my case, it generates much better results in the case of the former.

  "But," you say, "my PCP2 xgrid jobs can’t talk to the window server !"

  Very true. However, you can change the user that PCP2 uses to submit Xgrid jobs, and Xgrid will run the job with that user’s permissions if everyone is single signon’d to the same kerberos domain.

  So, now we’ve got PCP2 jobs running as a real user. Next, log into the GUI as that user.

  Now, when PCP2 workflows run, they’ll be able to talk to the window server, and at least in the case of "chapterize," use what appears to be the "Good" code path. Faster, more accurate, more delightful.