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• Xuggle - Concatenate two videos - Error - java.lang.RuntimeException : error -1094995529 decoding audio

  1er avril 2013, par user2232357

  I am using the Xuggle API to concatenate two MPEG videos (with Audio inbuilt in the MPEGs).
  I am referring to the https://code.google.com/p/xuggle/source/browse/trunk/java/xuggle-xuggler/src/com/xuggle/mediatool/demos/ConcatenateAudioAndVideo.java?r=929. (my both inputs and output are MPEGs).

  Getting the bellow error.

  14:06:50.139 [main] ERROR org.ffmpeg - [mp2 @ 0x7fd54693d000] incomplete frame
  
  java.lang.RuntimeException: error -1094995529 decoding audio
  
      at com.xuggle.mediatool.MediaReader.decodeAudio(MediaReader.java:549)
  
      at com.xuggle.mediatool.MediaReader.readPacket(MediaReader.java:469)
  
      at com.tav.factory.video.XuggleMediaCreator.concatenateAllVideos(XuggleMediaCreator.java:271)
  
      at com.tav.factory.video.XuggleMediaCreator.main(XuggleMediaCreator.java:446)

  Can anyone help mw with this ??? Thanks in Advance..

  Here is the complete code.

  public String concatenateAllVideos(ArrayList<tavtexttoavrequest> list){
  
          String finalPath="";
  
  
  
  
  
          String sourceUrl1 = "/Users/SSID/WS/SampleTTS/page2/AV_TAVImage2.mpeg";
  
          String sourceUrl2 = "/Users/SSID/WS/SampleTTS/page2/AV_TAVImage3.mpeg";
  
          String destinationUrl = "/Users/SSID/WS/SampleTTS/page2/z_AV_TAVImage_Final23.mpeg";
  
  
  
          out.printf("transcode %s + %s -> %s\n", sourceUrl1, sourceUrl2,
  
            destinationUrl);
  
  
  
          //////////////////////////////////////////////////////////////////////
  
          //                                                                  //
  
          // NOTE: be sure that the audio and video parameters match those of //
  
          // your input media                                                 //
  
          //                                                                  //
  
          //////////////////////////////////////////////////////////////////////
  
  
  
          // video parameters
  
  
  
          final int videoStreamIndex = 0;
  
          final int videoStreamId = 0;
  
          final int width = 400;
  
          final int height = 400;
  
  
  
          // audio parameters
  
  
  
          final int audioStreamIndex = 1;
  
          final int audioStreamId = 0;
  
          final int channelCount = 1;
  
          final int sampleRate = 16000 ; // Hz 16000 44100;
  
  
  
          // create the first media reader
  
  
  
          IMediaReader reader1 = ToolFactory.makeReader(sourceUrl1);
  
  
  
          // create the second media reader
  
  
  
          IMediaReader reader2 = ToolFactory.makeReader(sourceUrl2);
  
  
  
          // create the media concatenator
  
  
  
          MediaConcatenator concatenator = new MediaConcatenator(audioStreamIndex,
  
            videoStreamIndex);
  
  
  
          // concatenator listens to both readers
  
  
  
          reader1.addListener(concatenator);
  
          reader2.addListener(concatenator);
  
  
  
          // create the media writer which listens to the concatenator
  
  
  
          IMediaWriter writer = ToolFactory.makeWriter(destinationUrl);
  
          concatenator.addListener(writer);
  
  
  
          // add the video stream
  
  
  
          writer.addVideoStream(videoStreamIndex, videoStreamId, width, height);
  
  
  
          // add the audio stream
  
  
  
          writer.addAudioStream(audioStreamIndex, audioStreamId, channelCount,sampleRate);
  
  
  
          // read packets from the first source file until done
  
  
  
          try {
  
              while (reader1.readPacket() == null)
  
                ;
  
          } catch (Exception e) {
  
              // TODO Auto-generated catch block
  
              e.printStackTrace();
  
          }
  
  
  
          // read packets from the second source file until done
  
  
  
          try {
  
              while (reader2.readPacket() == null)
  
                ;
  
          } catch (Exception e) {
  
              // TODO Auto-generated catch block
  
              e.printStackTrace();
  
          }
  
  
  
          // close the writer
  
  
  
          writer.close();
  
  
  
  
  
          return finalPath;
  
      }
  
  
  
      static class MediaConcatenator extends MediaToolAdapter
  
        {
  
          // the current offset
  
  
  
          private long mOffset = 0;
  
  
  
          // the next video timestamp
  
  
  
          private long mNextVideo = 0;
  
  
  
          // the next audio timestamp
  
  
  
          private long mNextAudio = 0;
  
  
  
          // the index of the audio stream
  
  
  
          private final int mAudoStreamIndex;
  
  
  
          // the index of the video stream
  
  
  
          private final int mVideoStreamIndex;
  
  
  
          /**
  
           * Create a concatenator.
  
           * 
  
           * @param audioStreamIndex index of audio stream
  
           * @param videoStreamIndex index of video stream
  
           */
  
  
  
          public MediaConcatenator(int audioStreamIndex, int videoStreamIndex)
  
          {
  
            mAudoStreamIndex = audioStreamIndex;
  
            mVideoStreamIndex = videoStreamIndex;
  
          }
  
  
  
          public void onAudioSamples(IAudioSamplesEvent event)
  
          {
  
            IAudioSamples samples = event.getAudioSamples();
  
  
  
            // set the new time stamp to the original plus the offset established
  
            // for this media file
  
  
  
            long newTimeStamp = samples.getTimeStamp() + mOffset;
  
  
  
            // keep track of predicted time of the next audio samples, if the end
  
            // of the media file is encountered, then the offset will be adjusted
  
            // to this time.
  
  
  
            mNextAudio = samples.getNextPts();
  
  
  
            // set the new timestamp on audio samples
  
  
  
            samples.setTimeStamp(newTimeStamp);
  
  
  
            // create a new audio samples event with the one true audio stream
  
            // index
  
  
  
            super.onAudioSamples(new AudioSamplesEvent(this, samples,
  
              mAudoStreamIndex));
  
          }
  
  
  
          public void onVideoPicture(IVideoPictureEvent event)
  
          {
  
            IVideoPicture picture = event.getMediaData();
  
            long originalTimeStamp = picture.getTimeStamp();
  
  
  
            // set the new time stamp to the original plus the offset established
  
            // for this media file
  
  
  
            long newTimeStamp = originalTimeStamp + mOffset;
  
  
  
            // keep track of predicted time of the next video picture, if the end
  
            // of the media file is encountered, then the offset will be adjusted
  
            // to this this time.
  
            //
  
            // You&#39;ll note in the audio samples listener above we used
  
            // a method called getNextPts().  Video pictures don&#39;t have
  
            // a similar method because frame-rates can be variable, so
  
            // we don&#39;t now.  The minimum thing we do know though (since
  
            // all media containers require media to have monotonically
  
            // increasing time stamps), is that the next video timestamp
  
            // should be at least one tick ahead.  So, we fake it.
  
  
  
            mNextVideo = originalTimeStamp + 1;
  
  
  
            // set the new timestamp on video samples
  
  
  
            picture.setTimeStamp(newTimeStamp);
  
  
  
            // create a new video picture event with the one true video stream
  
            // index
  
  
  
            super.onVideoPicture(new VideoPictureEvent(this, picture,
  
              mVideoStreamIndex));
  
          }
  
  
  
          public void onClose(ICloseEvent event)
  
          {
  
            // update the offset by the larger of the next expected audio or video
  
            // frame time
  
  
  
            mOffset = Math.max(mNextVideo, mNextAudio);
  
  
  
            if (mNextAudio &lt; mNextVideo)
  
            {
  
              // In this case we know that there is more video in the
  
              // last file that we read than audio. Technically you
  
              // should pad the audio in the output file with enough
  
              // samples to fill that gap, as many media players (e.g.
  
              // Quicktime, Microsoft Media Player, MPlayer) actually
  
              // ignore audio time stamps and just play audio sequentially.
  
              // If you don&#39;t pad, in those players it may look like
  
              // audio and video is getting out of sync.
  
  
  
              // However kiddies, this is demo code, so that code
  
              // is left as an exercise for the readers. As a hint,
  
              // see the IAudioSamples.defaultPtsToSamples(...) methods.
  
            }
  
          }
  
  
  
          public void onAddStream(IAddStreamEvent event)
  
          {
  
            // overridden to ensure that add stream events are not passed down
  
            // the tool chain to the writer, which could cause problems
  
          }
  
  
  
          public void onOpen(IOpenEvent event)
  
          {
  
            // overridden to ensure that open events are not passed down the tool
  
            // chain to the writer, which could cause problems
  
          }
  
  
  
          public void onOpenCoder(IOpenCoderEvent event)
  
          {
  
            // overridden to ensure that open coder events are not passed down the
  
            // tool chain to the writer, which could cause problems
  
          }
  
  
  
          public void onCloseCoder(ICloseCoderEvent event)
  
          {
  
            // overridden to ensure that close coder events are not passed down the
  
            // tool chain to the writer, which could cause problems
  
          }
  
        }
  
  </tavtexttoavrequest>

• Using lcov With FFmpeg/Libav

  21 novembre 2011, par Multimedia Mike — Programming, code coverage, ffmpeg, lcov, libav

  Last year, I delved into code coverage tools and their usage with FFmpeg. I learned about using GNU gcov, which is powerful but pretty raw about the details it provides to you. I wrote a script to help interpret its output and later found another script called gcovr to do the same, only much better.

  I later found another tool called lcov which is absolutely amazing for understanding code coverage of your software. I’ve been meaning to use it to further FATE test coverage for the multimedia projects.

  
  
  Click for larger image
  

  Basic Instructions
  Install the lcov tool, of course. In Ubuntu, 'apt-get install lcov' will do the trick.

  Build the project with code coverage support, i.e.,

  ./configure —enable-gpl —samples=/path/to/fate/samples \
   —extra-cflags="-fprofile-arcs -ftest-coverage" \
   —extra-ldflags="-fprofile-arcs -ftest-coverage"
  make
  

  Clear the coverage data :

  lcov —directory . —zerocounters
  

  Run the software (in this case, the FATE test suite) :

  make fate
  

  Let lcov work its magic :

  lcov —directory . —capture —output-file coverage.info
  mkdir html-output
  genhtml -o html-output coverage.info
  

  At this point, you can aim your web browser at html-output/index.html to learn everything you could possibly want to know about code coverage of the test suite. You can sort various columns in order to see which modules have the least code coverage. You can drill into individual source files and see highlighted markup demonstrating which lines have been executed.

  As you can see from the screenshot above, FFmpeg / Libav are not anywhere close to full coverage. But lcov provides an exquisite roadmap.

• rtsp stream capturing

  12 avril 2016, par ДМИТРИЙ МАЛИКОВ

  I’m looking for some universal way to dump rtsp stream. I want to figure out, that some rtsp stream is working well and server is sending some watchable video.

  openRTSP

  At first, google recommends me openRTSP tool.

   openRTSP -4 ${stream_link} > ${output_file}

  But output video file dumped by that tool is not really correct. Video decoder (ffdec) returns many errors like "Failed to decode video packet" and "[h264] no frame !", which don’t suit me.

  ffmpeg

  Then I’ve tried to dump rtsp stream with ffmpeg tool.

  ffmpeg -loglevel debug -i "${stream_link}" -s 640x480 -vcodec copy -acodec copy -y ${output_file}

  But streaming process was interrupted often by error :

  Application provided invalid, non monotonically increasing dts to muxer in stream 0: 730672 >= 730672
  
  av_interleaved_write_frame(): Invalid argument

  I’m trying to use --fflags igndts but ffmpeg doesn’t ignore these errors. It doesn’t make any sense, because that error actually means that audio and video streams are sending asynchronously. The worst thing is that dumped file, resulted by that interrupted dump, is not correct too. Ffdec return some error :

  ERROR [mov,mp4,m4a,3gp,3g2,mj2] moov atom not found
  
  ERROR [ffdec] av_open_input_file: Operation not permitted

  After a nice cup of googling I’ve found, that it’s really old ffmpeg’s muxer bug.

  mplayer

  Than I’ve tried to use mplayer with LIVE_555 lib.

  mplayer -noframedrop -dumpfile ${output_file} -dumpstream ${stream_link}

  But I’ve got some errors too.

  Stream not seekable!
  
  Core dumped ;)

  Question

  I think I’m doing something wrong. It’s sounds really ridiculous, that there is no way to save rtsp stream in correct and playable video-file.

  Maybe there are some another tools which can help with that task ? Actually, I will be grateful for any advice for all kind of libs and languages. But that process should be automatic and have cli.

  Refinements

  Something about 50% experiments I’ve done on the localhost with vlc-streamer that emulates rtsp-broadcaster. Here is a manual which I try to follow.

  I have really fresh and latest ffmpeg with x264 support, that I’ve installed by that useful thread.