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• AAC encoder : various fixes in M/S coding

  8 janvier 2016, par Claudio Freire

  AAC encoder : various fixes in M/S coding
  1. Fix sf_idx and band_type addressing to address only the first
  
  subwindow in the group (others could hold garbage values)
  2. Don’t step on ms_mask when is_mask is set. I/S selection
  
  already sets the ms_mask properly and shouldn’t be overridden.
  3. Use mid/sid cb/sf when computing coding error, as should be
  
  since those are the cb/sfs that will eventually be set.
  4. Fix distortion computation on multi-subwindow groups (was
  
  subtracting the bits terms multiple times)
  5. Clear ms_mask when one side uses PNS and the other doesn’t.
  
  When using PNS, ms_mask signals correlated noise, which can be
  
  detected just like regular M/S detection, so we don’t skip
  
  noise bands, but when only one side uses PNS setting the flag
  
  can confuse some encoders, so avoid that.
  

  • [DH] libavcodec/aaccoder.c
  • [DH] libavcodec/mips/aaccoder_mips.c
  • [DH] tests/fate/aac.mak

• Fast transformation of BGR BufferedImage to YUV using FFMpeg

  3 août 2016, par Hossein

  I wanted to transform a TYPE_3BYTE_BGR BufferedImage in Java to yuv using the sws_scale function of FFMpeg through JNI. I first extract the data of my image from the BufferedImage as

      byte[] imgData = ((DataBufferByte) myImage.getRaster().getDataBuffer()).getData();
  
  
  
      byte[] output = processImage(toSend,0);     

  Then I pass it to the processImage function which is a native function. The C++ side looks like this :

  JNIEXPORT jbyteArray JNICALL Java_jni_JniExample_processData
  
    (JNIEnv *env, jobject obj, jbyteArray data, jint index)
  
  {
  
  
  
      jboolean isCopy;
  
      uint8_t *test  = (uint8_t *)env->GetPrimitiveArrayCritical(data, &isCopy);
  
      uint8_t *inData[1]; // RGB24 have one plane
  
      inData[0] = test;
  
  
  
  
  
  SwsContext * ctx = sws_getContext(width,height,AV_PIX_FMT_BGR24, (int)width, (int)width,
  
          AV_PIX_FMT_YUV420P, 0, 0, 0, 0);
  
  
  
      int lumaPlaneSize = width *height;
  
      uint8_t *yuv[3];
  
  
  
      yuv[0] = new uint8_t[lumaPlaneSize];
  
      yuv[1] = new uint8_t[lumaPlaneSize/4];
  
      yuv[2] = new uint8_t[lumaPlaneSize/4];
  
  
  
      int inLinesize[1] = { 3*nvEncoder->width }; // RGB stride
  
      int outLinesize[3] = { 3*width ,3*width ,3*width }; // YUV stride
  
  
  
      sws_scale(ctx, inData, inLinesize, 0, height , yuv, outLinesize);

  However, after running the code, I get the warning : [swscaler @ 0x7fb598659480] Warning: data is not aligned! This can lead to a speedloss, everything crashes., and everything crashes on the last line. Am I doing things properly in terms of passing the correct arguments to sws_scale ? (specially the strides).

  Update :
  There was a separate bug here : SwsContext * ctx = sws_getContext(width,height,AV_PIX_FMT_BGR24, (int)width, (int)width,0,NULL,NULL,NULL) which should be changed to : SwsContext * ctx = sws_getContext(width,height,AV_PIX_FMT_BGR24, (int)height, (int)width,0,NULL,NULL,NULL)

• convert lower bitrate mp3 to higher bit rate [closed]

  6 mars 2016, par janeman

  i had 2000 songs in 128 kbps but in my server space was not there so i have compressed those mp3 to 16kbps and found ok quality on my pc but when i was listening those songs on my iphone i found that sound quality is not good so now i wan t to convert those 16kbps songs to 48kbps which i think will give good sound quality so that users cant differentiate that songs are compressed and it will load fast as well .here is the ffmpeg code i have used to convert 16kbps file to 48kbps . i can see that the size of converted file has increased approx 3 fold but not sure whether it will give the better quality mp3 when i increase the bit rate through conversion .please suggest any way to get my file back with good mp3 quality

    <?php  
  
   if ( exec('ffmpeg  -i /pathtomp3folder/test.mp3  -acodec libmp3lame  -ab 48k -ac 1 -ar 44100  /pathtomp3folder/oputput.mp3'))
  
  
  
     {
  
    echo 'done!!';
  
    }
  
  
  
    ?>

  i can do below code as well wthout specifying the frequency which is best way to do

     ffmpeg  -i /pathtomp3folder/test.mp3  -acodec libmp3lame  -ab 48k -ac 1   /pathtomp3folder/oputput.mp3
  
  
  
    {
  
    echo 'done!!';
  
     }

  i dont see much difference in 16kbps and converted file quality however size got increased 3 times in converted file

  what is the minimum kbps we should prefer so that users cant detect the difference and i get high streaming rate as well without buffering .i am targetting mobile device so please suggest about mobile device

  size of 16 kbps file was 745kb and size of 48kbps file is 1840kb

  where i am doing mistake any idea or if anyone can share his experinene it will be a great help .i have stuggled to get my collection of 2000 songs back as i dont have backup and in my pc the quality of mp3 seems to be ok but when i play in iphone it clearly shows that its of low quality

  in my server ffmpeg is installed and working properly so it is easily converting the bit rate .one more thing i want to know when the bit rate is getting changed from 16kbps to 48 kbps what is the changes in quality ???/ as i can clearly see that size is increasing but the quality remains same