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• Revision e7bddba149 : new version of speed 1 This version of speed 1 only disables modes at higher re

  23 avril 2013, par Jim Bankoski

  Changed Paths : Modify /vp9/encoder/vp9_block.h Modify /vp9/encoder/vp9_rdopt.c new version of speed 1 This version of speed 1 only disables modes at higher resolution that had distortions >2x the best mode we found... The hope is that this could be a replacement for speed 0 ... Change-Id : (...)

• How can I concatenate two mp3 files with different bit rates

  31 décembre 2012, par Scott

  I have FFmpeg installed on my linux web server. When I execute the following code, I have intermittent results.

  I think I have figured out that the MP3s do not compile when they have different bitrates.

  exec ('cat '. $pair['source_file'] . ' ' . $pair['translated_word_file'] . '>' . $temp_mp3);

  I might have found some articles online that reference taking them apart and then bundling them back together at a consistent bitrates.

  I have confirmed that this won't really work with basic "cat" function and that "sox" can be used IF they have the same sample rate.

  The issue now becomes "What is the best way to get them to the same sample rate ?"

• 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