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• Force GStreamer for IOS to use ffmpeg library outside of the framework

  24 avril 2014, par Michelle Cannon

  Gstreamer 1.0 for IOS is delivered in a static framework, the source to build the framework is around 1.2g , this framework is huge and tries to provide for any decoding server scenario you may have. Trouble is it tries to do to much and IMHO not enough thought was put into the IOS port.

  Here’s the problem we have an application that uses the GSTreamer avdec_h264 plugin for displaying an RTP over UDP stream . This works rather well. recently we were required to do some special recording functions so we introduced an api that had its own version of ffmpeg. Gstreamer has Libav compiled into the framework. When we place our api into the application with the gst_IOS_RESTICTED_PLUGINS disabled the code runs fine when we introduce the GStreamer.framework into the application code similar to that shown below fails with a protocol not found error.

  The problem is that the internal version of libav seems to disable all the protocols that ffmpeg supplies. because GSTreamer uses its own custom AVIO callback based on ffmpeg pipe protocol.

  According to Gstreamer support that has been somewhat helpful

  ) Add a new recipe with the libav version you want to use and disable the build of the internal libav in gst-libav-1.0 with :
  configure_options = ’—with-system-libav’

  You might need to comment out this part to prevent libav being packaged in the framewiork or make sure that your libav recipe creates these files in the correct place to include them in the framework :

  42 for f in [’libavcodec’, ’libavformat’, ’libavutil’, ’libswscale’] :
  43 for ext in [’.a’, ’.la’] :
  44 path = os.path.join(’lib’, f + ext)
  45 self.files_plugins_codecs_restricted_devel.append(path)

  2) Update libav submodule gst-libav to use the correct version you need.

  https://bugs.freedesktop.org/show_bug.cgi?id=77399

  The first method didn’t work , the recipe kept getting overwritten even after applying a patch for a bug fix that was made as result of this bug report.

  And I have no idea how to do the second method. Which is what I’d like some help with.

  Has anyone with GStreamer 1.0 for iOS

  1) Built the get-libav plugin against an external to to the framework set of ffmpeg static libs (.a)

  2) built the internal libav to allow for RTP , UDP and TCP protocols, or written a custom AVIO callback using the FFPipe protocol.

  3) just managed to somehow get the below code working with GStreamer.

  I don’t ask many questions, I’ve kind of implemented all kinds of encoders/decoders using ffmpeg , lib555 and a few hardware decoders. But this GStreamer issue is causing me more sleepless nights than I’ve had in a long time.

  AVFormatContext * avctx;
  
  avctx = avformat_alloc_context();
  
  
  
  av_register_all();
  
  avformat_network_init();
  
  avcodec_register_all();
  
  avdevice_register_all();
  
  
  
  
  
  
  
  // Set the RTSP Options
  
  AVDictionary *opts = 0;
  
  
  
  av_dict_set(&opts, "rtsp_transport", "udp", 0);
  
  
  
  
  
  int err = 0;
  
  err = avformat_open_input(&avctx, "rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov", NULL, &opts);
  
  av_dict_free(&opts);
  
  if (err) {
  
      NSLog(@"Error: Could not open stream: %d", err);
  
      char errbuf[400];
  
      av_strerror(err,errbuf,400);
  
      NSLog(@"%s failed with error %s","avformat_open_input",errbuf);
  
  
  
  
  
  
  
  }
  
  else {
  
      NSLog(@"Opened stream");
  
  }
  
  
  
  err = avformat_find_stream_info(avctx, NULL);
  
  if( err < 0)
  
  {
  
      char errbuf[400];
  
      av_strerror(err,errbuf,400);
  
      NSLog(@"%s failed with error %s","avformat_find_stream_info",errbuf);
  
      return ;
  
  }else {
  
       NSLog(@"found stream info");
  
  }

  }

• SRT : No Room to Store Incoming Packets when we stream SRT streams through VLC

  10 février, par Shankar

  Hope all you are doing well.

  


  Actually i was trying to stream a media file using VLC by SRT protocol. For which, srt-live-transmit is used as converter in between SRT Listener and VLC UDP streams. srt-live-transmit used to convert udp to srt streams(mpegts). But when i tried to do that, after few seconds, i got error in srt-live-transmit terminal that :
No room to store incoming packet :

  


  What should be the reason for this error ? And if anyone know anything about this problem, please share info. It would be helpful.
Thank you.

  


• RTMP to RTSP fluent-ffmpeg conversion

  13 avril 2020, par leon54

  I am trying to convert a real time video stream from RTMP protocol to RTSP protocol using node.js fluent-ffmpeg library.
I also incorporated node-media-server into my code, and called that ffmpegConversion method inside its "prePublish" hook.
Once I start my RTMP stream from OBS studio, conversion starts, and once I stop the stream it successfully ends.
But, the problem starts when I want to start the conversion again : It throws me a "Invalid argument error", even though the command was the same.
This is the command that gets executed first time I run it : "ffmpeg -i rtmp ://127.0.0.1/app/live -vcodec libx264 -filter:v scale=w=320:h=200 -f rtsp rtsp ://127.0.0.1:5554/mystream"
This is the command that gets runned every other time : "ffmpeg -i rtmp ://127.0.0.1/app/live -vcodec libx264 rtsp ://127.0.0.1:5554/mystream"
For some reason, it completely ignores the filters and format parameters.
This is how it looks like inside my code :

  



  function ffmpegConversion() {

    var one = ffmpeg('rtmp://127.0.0.1/app/live', { 
            timeout: 432000
        })
        .videoCodec('libx264')
        .on('start', function (commandLine) {
            console.log("FFMPEG Start has been triggered " + commandLine);
        })
        .on('stderr', function(stderrLine) {
            console.log('Stderr output: ' + stderrLine);
          })
        .on("progress", function (progress) {
            console.log('Timemark one: ' + progress.timemark + ' sec');
        })
        .on('codecData', function (data) {

            console.log('On codec data');
            // console.log(data);
        })
        .on('end', function (end) {
            console.log('file has ended converting succesfully ' + end);
        })
        .on('error', function (err) {
            console.log('an error happened: ' + err.message);
        })
        .save('rtsp://127.0.0.1:5554/mystream')
        .format("rtsp")
        .size('320x200');

}

//NODE MEDIA SERVER METHODS

nms.run();

nms.on('prePublish', (id, StreamPath, args) => {
    console.log('***NodeEvent on prePublish', `id=${id} StreamPath=${StreamPath} `);
    // let session = nms.getSession(id);
    // session.reject();
    ffmpegConversion(); 
});