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• Revision b04d766800 : Use small transform size in non-rd real-time mode In non-rd real-time mode, cho

  30 mai 2014, par Yunqing Wang

  Changed Paths :
   Modify /vp9/encoder/vp9_encodeframe.c

  
   Modify /vp9/encoder/vp9_speed_features.c

  
   Modify /vp9/encoder/vp9_speed_features.h

  
  
  Use small transform size in non-rd real-time mode

  In non-rd real-time mode, choosing smaller transform size in
  encoding gives better video quality and good speed gain than
  choosing larger transform size. This patch set tx size search
  method to ALLOW_8X8, which is better than using 4x4 or other
  larger sizes.

  Borg tests on rtc set at speed 6 showed significant gain on quality.
  PSNR gain : 11.034% and SSIM gain : 15.466%.

  The speed gain is 5% - 12% for <720p clips, and 2% - 7% for
  720p clips.

  Change-Id : If4dc74ed2df359346b059f47fb73b4a0193ec548

• samplerate conversion function fails to produce an audible sound but only small pieces of audio

  2 juillet 2014, par user3749290

  playmp3() using libmpg123

  if (isPaused==0 && mpg123_read(mh, buffer, buffer_size, &done) == MPG123_OK)
  
  {
  
      char * resBuffer=&buffer[0]; //22100=0,5s
  
      buffer = resample(resBuffer,22050,22050); // I think the result is 1/2 of audio speed
  
      if((ao_play(dev, (char*)buffer, done)==0)){
  
          return 1;
  
  }

  resample() Using avcodec from ffmpeg

  #define LENGTH_MS 500       // how many milliseconds of speech to store 0,5s:x=1:44100 x=22050 sample to store
  
  #define RATE 44100      // the sampling rate (input)
  
  #define FORMAT PA_SAMPLE_S16NE  // sample size: 8 or 16 bits
  
  #define CHANNELS 2      // 1 = mono 2 = stereo
  
  
  
  struct AVResampleContext* audio_cntx = 0;
  
  //(LENGTH_MS*RATE*16*CHANNELS)/8000
  
  
  
      void resample(char in_buffer[],int out_rate,int nsamples,char out_buffer[])
  
      {
  
          //char out_buffer[ sizeof( in_buffer ) * 4];
  
          audio_cntx = av_resample_init( out_rate, //out rate
  
              RATE, //in rate
  
              16, //filter length
  
              10, //phase count
  
              0, //linear FIR filter
  
              1.0 ); //cutoff frequency
  
          assert( audio_cntx && "Failed to create resampling context!");
  
          int samples_consumed;
  
          //*out_buffer = malloc(sizeof(in_buffer));
  
          int samples_output = av_resample( audio_cntx, //resample context
  
              (short*)out_buffer, //buffout
  
              (short*)in_buffer,  //buffin
  
              &samples_consumed,  //&consumed
  
              nsamples,       //nb_samples
  
              sizeof(out_buffer)/2,//lenout sizeof(out_buffer)/2
  
              0);//is_last
  
          assert( samples_output > 0 && "Error calling av_resample()!" );
  
          av_resample_close( audio_cntx );    
  
      }

  When I run this code, the application part, the problem is that I hear the sound jerky, why ?
  The size of the array I think is right, I calculated considering that in the second half should be 22050 samples from store.

• FFmpeg : How to convert horizontal video 16:9 to vertical video 9:16, with blurred background on top and bottom sides

  13 janvier 2016, par gomezo

  I know there is a post to do the contrary :
  FFmpeg : How to convert vertical video with black sides, to video 16:9, with blurred background sides

  but i can’t figure out how to do this with ffmpeg

  here is the input :

  

  and here is the expected result (the background video should be zoomed with the same aspect ratio, cropped to 9/16 and blurred) :

  

  Thanks for your help !