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• AAC : Fix M/S stereo encoding

  3 mars 2015, par Claudio Freire

  AAC : Fix M/S stereo encoding
  This patch fixes a pointer arithmetic bug in adjust_frame_information that resulted in heavily corrupted audio when using M/S encoding. Also, a backup copy of untransformed coefficients has to be kept around or attempts at re-processing the frame (which happens when hevavily overspending bits during transients) will result in re-encoding of the coefficients and subsequent corruption of the resulting stream.
  A/B testing shows the bug as corrected, but still cannot prove that M/S coding is a win at least in numbers. Limited listening tests do show improvement on M/S encoded samples in lower bitrates, but they’re hidden among the other artifacts that remain to be corrected in the encoder.
  Some of the regressions flagged in the report do show poor stereo image (but not buggy), so M/S encoding is clearly not good enough yet to be defaulted to auto.
  In numbers, Patched against Unpatched, stereo_mode auto :
    Files : 114
  
    Bitrates : 6
  
    Tests : 683
    Serious Regressions : 0 (0%)
  
    Regressions : 0 (0%)
  
    Improvements : 227 (33%)
  
    Big improvements : 92 (13%)
  
    Worst regression - mybloodrusts.wv - 256k
  
      - StdDev : 28.61       pSNR : -0.43     maxdiff : 1372.00
  
    Best improvement - 60.wv - 384k
  
      - StdDev : -369.57     pSNR : 45.02     maxdiff : -13322.00
  
    Average          - StdDev : -80.56       pSNR : 2.49      maxdiff : -8858.00
  Patched against Unpatched stereo_mode ms_off shows no difference.
  Patched stereo_mode auto vs Unpatched stereo_mode ms_off shows a small average improvement, just not too significant :
    Serious Regressions : 0 (0%)
  
    Regressions : 10 (1%)
  
    Improvements : 45 (6%)
  
    Big improvements : 2 (0%)
  
    Worst regression - Illinois.wv - 256k
  
      - StdDev : 33.20       pSNR : -2.03     maxdiff : 477.00
  
    Best improvement - song_of_circomstances.flac - 384k
  
      - StdDev : -3.97       pSNR : 7.61      maxdiff : -826.00
  
    Average          - StdDev : -10.25       pSNR : 0.20      maxdiff : -281.00
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavcodec/aac.h
  • [DH] libavcodec/aaccoder.c
  • [DH] libavcodec/aacenc.c

• Facing issue with mp4Parser

  20 novembre 2014, par Rohit

  I am trying to trim video with the help of mp4Parser. In output am getting trimmed video but just before video finishes it give error "Sorry cant Play this video". Video am making is H264.

      public static void main(String args) throws IOException {
  
  
  
      Movie movie = new MovieCreator()
  
              .build(new RandomAccessFileIsoBufferWrapperImpl(
  
                      new File(
  
                              "/sdcard/Videos11/"+args+".mp4")));
  
  
  
      List<track> tracks = movie.getTracks();
  
      Log.e("Tracks","Following are the Tracks:- "+tracks);
  
      movie.setTracks(new LinkedList<track>());
  
      // remove all tracks we will create new tracks from the old
  
  
  
      double startTime = 0.000;
  
      double endTime = 6.000;
  
  
  
      boolean timeCorrected = false;
  
  
  
      // Here we try to find a track that has sync samples. Since we can only
  
      // start decoding
  
      // at such a sample we SHOULD make sure that the start of the new
  
      // fragment is exactly
  
      // such a frame
  
      for (Track track : tracks) {
  
          if (track.getSyncSamples() != null
  
                  &amp;&amp; track.getSyncSamples().length > 0) {
  
              if (timeCorrected) {
  
                  // This exception here could be a false positive in case we
  
                  // have multiple tracks
  
                  // with sync samples at exactly the same positions. E.g. a
  
                  // single movie containing
  
                  // multiple qualities of the same video (Microsoft Smooth
  
                  // Streaming file)
  
  
  
                  throw new RuntimeException(
  
                          "The startTime has already been corrected by another track with SyncSample. Not Supported.");
  
              }
  
              startTime = correctTimeToNextSyncSample(track, startTime);
  
              endTime = correctTimeToNextSyncSample(track, endTime);
  
              timeCorrected = true;
  
          }
  
      }
  
  
  
      for (Track track : tracks) {
  
          long currentSample = 0;
  
          double currentTime = 0;
  
          long startSample = -1;
  
          long endSample = -1;
  
  
  
          for (int i = 0; i &lt; track.getDecodingTimeEntries().size(); i++) {
  
              TimeToSampleBox.Entry entry = track.getDecodingTimeEntries().get(i);
  
              for (int j = 0; j &lt; entry.getCount(); j++) {
  
                  // entry.getDelta() is the amount of time the current sample
  
                  // covers.
  
  
  
                  if (currentTime &lt;= startTime) {
  
                      // current sample is still before the new starttime
  
                      startSample = currentSample;
  
                  }
  
                  if (currentTime &lt;= endTime) {
  
                      // current sample is after the new start time and still
  
                      // before the new endtime
  
                      endSample = currentSample;
  
                  } else {
  
                      // current sample is after the end of the cropped video
  
                      break;
  
                  }
  
                  currentTime += (double) entry.getDelta()
  
                          / (double) track.getTrackMetaData().getTimescale();
  
                  currentSample++;
  
              }
  
          }
  
          movie.addTrack(new CroppedTrack(track, startSample, endSample));
  
      }
  
  
  
      IsoFile out = new DefaultMp4Builder().build(movie);
  
  
  
      String filePath = "sdcard/test"+i+".mp4";
  
      i++;
  
      File f = new File(filePath);
  
      FileOutputStream fos = new FileOutputStream(f);
  
      BufferedOutputStream bos = new BufferedOutputStream(fos, 65535);
  
      out.getBox(new IsoOutputStream(fos));
  
      bos.close();
  
      fos.flush();
  
      fos.close();
  
  
  
  }
  
  
  
  private static double correctTimeToNextSyncSample(Track track,
  
          double cutHere) {
  
      double[] timeOfSyncSamples = new double[track.getSyncSamples().length];
  
      long currentSample = 0;
  
      double currentTime = 0;
  
      for (int i = 0; i &lt; track.getDecodingTimeEntries().size(); i++) {
  
          TimeToSampleBox.Entry entry = track.getDecodingTimeEntries().get(i);
  
          for (int j = 0; j &lt; entry.getCount(); j++) {
  
              if (Arrays.binarySearch(track.getSyncSamples(),
  
                      currentSample + 1) >= 0) {
  
                  // samples always start with 1 but we start with zero
  
                  // therefore +1
  
                  timeOfSyncSamples[Arrays.binarySearch(
  
                          track.getSyncSamples(), currentSample + 1)] = currentTime;
  
              }
  
              currentTime += (double) entry.getDelta()
  
                      / (double) track.getTrackMetaData().getTimescale();
  
              currentSample++;
  
          }
  
      }
  
      for (double timeOfSyncSample : timeOfSyncSamples) {
  
          if (timeOfSyncSample > cutHere) {
  
              return timeOfSyncSample;
  
          }
  
      }
  
      return timeOfSyncSamples[timeOfSyncSamples.length - 1];
  
  }
  
  </track></track>

  Making video of 30 seconds and want to trim first 6 seconds. Working but in last(between 5 to 6 sec) it shows Cant play this video. Any help will be appreciated.

• Revision 9f79259e54 : adapt the adjustment limit for rate correction factor in RTC mode Rate correcti

  13 novembre 2014, par Yaowu Xu

  Changed Paths :
   Modify /vp9/encoder/vp9_ratectrl.c

  
  
  adapt the adjustment limit for rate correction factor in RTC mode

  Rate correction factor is used to correct the estimated rate for any
  given quantizer, and feeds into rate control for quantizer selection.
  We make use of the actual bits used to calculate this rate correction
  factor with an adjustment limit to prevent over-adjustment.

  This commit adapts the adjustment limit to the difference between the
  estimated bits and the actual bits, allows the adjustment limit to vary
  between 0.125 (when estimate is close to actual) and 0.625 (when there
  is >10X factor off between estimated and actual bits). By doing this,
  the commit appears to have largely corrected two observed issues :
  1. Adjustment is too slow when the actual bits used is way off from
  estimate due to the small adjustment limit.
  2. Extreme oscillating quantizer choices due to the feedback loop.

  Change-Id : I4ee148d2c9d26d173b6c48011313ddb07ce2d7d6