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• x264 Downsides of a high CRF (22) intermediary codec between conversions instead of lossless

  18 décembre 2019, par bobtheencoder

  I have a huge collection of video files that are in the range of CRF 16-20 taking up TB’s of space. The only need I have for these originals is that I have to encode them from time to time but the CRF of these final encodes is very low (CRF 26-28).

  I understand that a lossy to lossy converstion ALWAYS results in some quality loss but my question is what if the intermediate file is almost visually lossless compared to the final output.

  So to sum up, what quality difference should I expect from the following routes ?

  CRF 18 (original) -----> CRF 28 (final)
  
  CRF 18 (original) -----> CRF 22 (long-term storage) -----> Lossy  CRF 28 (final)

• Issue in FFmpegAndroid library when i compress video it convert video time into 1 or 2 second

  25 mai 2017, par Fateh Singh Saini

  Used this dependencies :
  compile ’com.writingminds:FFmpegAndroid:0.3.2’

  I used blow code for video compress
  public static final String VIDEOCODEC = "-vcodec" ;
  public static final String AUDIOCODEC = "-acodec" ;

  public static final String VIDEOBITSTREAMFILTER = "-vbsf";
  
  public static final String AUDIOBITSTREAMFILTER = "-absf";
  
  
  
  public static final String VERBOSITY = "-v";
  
  public static final String FILE_INPUT = "-i";
  
  public static final String SIZE = "-s";
  
  public static final String FRAMERATE = "-r";
  
  public static final String FORMAT = "-f";
  
  public static final String BITRATE_VIDEO = "-b:v";
  
  
  
  public static final String BITRATE_AUDIO = "-b:a";
  
  public static final String CHANNELS_AUDIO = "-ac";
  
  public static final String FREQ_AUDIO = "-ar";

  String[] complexCommand = "-y", FILE_INPUT, yourRealPath, SIZE, "480x360", FRAMERATE, "25", VIDEOCODEC, "mpeg4", BITRATE_VIDEO, "150k", BITRATE_AUDIO, "48000", CHANNELS_AUDIO, "2", FREQ_AUDIO, "22050", filePath ;

  /**
  
   * Executing ffmpeg binary
  
   */
  
  private static String execFFmpegBinary(final String[] command) {
  
  
  
  
  
      try {
  
          ffmpeg.execute(command, new ExecuteBinaryResponseHandler() {
  
              @Override
  
              public void onFailure(String s) {
  
                  Log.d(TAG, "FAILED with output : " + s);
  
              }
  
  
  
              @Override
  
              public void onSuccess(String s) {
  
                  Log.d(TAG, "SUCCESS with output : " + s);
  
              }
  
  
  
              @Override
  
              public void onProgress(String s) {
  
                  Log.d(TAG, "Started command : ffmpeg " + command);
  
                  Log.d(TAG, "progress : " + s);
  
              }
  
  
  
              @Override
  
              public void onStart() {
  
                  Log.d(TAG, "Started command : ffmpeg " + command);
  
              }
  
  
  
              @Override
  
              public void onFinish() {
  
                  Log.d(TAG, "Finished command : ffmpeg " + command);
  
  
  
              }
  
          });
  
      } catch (FFmpegCommandAlreadyRunningException e) {
  
          // do nothing for now
  
      }
  
      return filePath;
  
  }

• Normalizing audio in ffmpeg - how ?

  11 novembre 2020, par Betty Crokker

  I'm creating one of those "Brady Bunch" videos for a choir using a C# application I'm writing that uses ffmpeg for all the heavy lifting, and for the most part it's working great but I'm having trouble getting the audio levels just right.

  


  What I'm doing right now, is first "normalizing" the audio from the individual singers like this :

  


  

  • Extract audio into a WAV file using ffmpeg

  


  • Load the WAV file into my application using NAudio

  


  • Find the maximum 16-bit value

  


  • When I create the merged video, specify a volume for this stream that boosts the maximum value to 32767

  


  


  So, for example, if I have 3 streams : stream A's maximum audio is 32767 already, stream B's maximum audio is 32000, and stream C's maximum audio is 16000, then when I merge these videos I will specify

  


  [0:a]volume=1.0,aresample=async=1:first_pts=0[aud0]
[1:a]volume=1.02,aresample=async=1:first_pts=0[aud1]
[2:a]volume=2.05,aresample=async=1:first_pts=0[aud2]
[aud0][aud1][aud2]amix=inputs=3[a]


  


  (I have an additional "volume tweak" that lets me adjust the volume level of individual singers as necessary, but we can ignore that for this question)

  


  I am reading the ffmpeg wiki on Audio Volume Manipulation, and I will implement that next, but I don't know what to do with the output it generates. It looks like I'm going to get mean and max volume levels in dB and while I understand decibels in a "yeah, I learned about those in college 30 years ago" kind of way, I don't know how to use those values to normalize the audio of my input videos.

  


  The problem is, in the ffmpeg output video, the audio level is quite low. If I do the same process of extracting the audio and looking at the WAV file in the merged video that ffmpeg generated, the maximum value is only 4904.

  


  How do I implement an algorithm that automatically sets the output volume to a "reasonable" level ? I realize I can simply add a manual volume filter and have the human set the level, but that's going to be a lot of back & forth of generating the merged video, listening to it, adjusting the level, merging again, etc. I want a way where my application figures out an appropriate output volume (possibly with human adjustment allowed).

  


  EDIT

  


  Asking ffmpeg to determine the mean and max volume of each clip does provide mean and max volume in dB, and I can then use those values to scale each input clip :

  


  [0:a]volume=3.40dB,aresample=async=1:first_pts=0[aud0]
[1:a]volume=3.90dB,aresample=async=1:first_pts=0[aud1]
[2:a]volume=4.40dB,aresample=async=1:first_pts=0[aud2]
[3:a]volume=-0.00dB,aresample=async=1:first_pts=0[aud3]


  


  But my final video is still strangely quiet. For now, I've added a manually-entered volume factor that gets applied at the very end :

  


  [aud0][aud1][aud2]amix=inputs=3[a]
[a]volume=volume=3.00[b]


  


  So my question is, in effect, how do I determine algorithmically what this final volume factor needs to be ?

  


  MORE EDIT

  


  There's something deeper going on here, I just set the volume filter to 100 and the output is only slightly louder. Here are my filters, and the relevant portions of the command line :

  


  color=size=1920x1080:c=0x0000FF [base];
[0:v] scale=576x324 [clip0];
[0:a]volume=1.48,aresample=async=1:first_pts=0[aud0];
[1:v] crop=808:1022:202:276,scale=384x486 [clip1];
[1:a]volume=1.57,aresample=async=1:first_pts=0[aud1];
[2:v] crop=1160:1010:428:70,scale=558x486 [clip2];
[2:a]volume=1.66,aresample=async=1:first_pts=0[aud2];
[3:v] crop=1326:1080:180:0,scale=576x469 [clip3];
[3:a]volume=1.70,aresample=async=1:first_pts=0[aud3];
[4:a]volume=0.20,aresample=async=1:first_pts=0[aud4];
[5:a]volume=0.73,aresample=async=1:first_pts=0[aud5];
[6:v] crop=1326:1080:276:0,scale=576x469 [clip4];
[6:a]volume=1.51,aresample=async=1:first_pts=0[aud6];
[base][clip0] overlay=shortest=1:x=32:y=158 [tmp0];
[tmp0][clip1] overlay=shortest=1:x=768:y=27 [tmp1];
[tmp1][clip2] overlay=shortest=1:x=1321:y=27 [tmp2];
[tmp2][clip3] overlay=shortest=1:x=32:y=625 [tmp3];
[tmp3][clip4] overlay=shortest=1:x=672:y=625 [tmp4];
[aud0][aud1][aud2][aud3][aud4][aud5][aud6]amix=inputs=7[a];
[a]adelay=delays=200:all=1[b];
[b]volume=volume=100.00[c];
[c]asplit[a1][a2];

ffmpeg -y ....
   -map "[tmp4]" -map "[a1]" -c:v libx264 "D:\voutput.mp4" 
   -map "[a2]" "D:\aoutput.mp3""


  


  When I do this, the audio I want is louder (loud enough to clip and get distorted), but definitely not 100x louder.