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• Build ffmpeg on a build machine

  18 juillet 2019, par RDI

  Build ffmpeg on build PC using libx264 and shared libraries (not static).
  I am building on a Red Hat 6.6 Server and final target machine is CentOS 6.6.
  I am trying, as said, to build ffmpeg with encoding enabled (with libx264) and shared libraries ; of course I do not want to install the libraries on the build PC, they should be only extracted and then delivered together with the final RPM.
  After the "./configure" I get all RPMs (related to ffmpeg) but when trying to installing ffmpeg-libs on the build pc it fails because the libx264.so.157 is not found, even if as test I installed it (configure/make/make install) and present at /usr/local/lib.

  Where am I wrong ?

  Thanks

  This is my SPEC file at the moment :

  ldconfig /usr/local/lib
  
  export LD_LIBRARY_PATH=/usr/local/lib:$LD_LIBRARY_PATH
  
  
  
  
  
  # configure 
  
  ./configure \
  
   --enable-gpl --disable-static --enable-shared --extra-cflags="-I/usr/local/include" --extra-ldflags="-L/usr/local/lib" --extra-libs=-ldl --disable-autodetect --disable-doc --disable-postproc --disable-ffplay --disable-everything --enable-encoder=aac --enable-encoder=png --enable-encoder=mjpeg --enable-encoder=libx264 --enable-decoder=aac --enable-decoder=h264 --enable-decoder=mpeg4 --enable-decoder=rawvideo --enable-decoder=png --enable-muxer=mp4 --enable-muxer=stream_segment --enable-muxer=image2 --enable-demuxer=aac --enable-demuxer=h264 --enable-demuxer=mov --enable-demuxer=rtp --enable-parser=aac --enable-parser=h264 --enable-parser=mpeg4video --enable-bsf=aac_adtstoasc --enable-protocol=file --enable-protocol=http --enable-protocol=tcp --enable-protocol=rtp --enable-protocol=udp --enable-indev=xcbgrab --disable-alsa --enable-libxcb --enable-libxcb-xfixes --enable-libxcb-shape --enable-zlib --prefix=%{_prefix} --bindir=%{_bindir} --datadir=%{_datadir}/%{name} --shlibdir=%{_libdir} --enable-alsa --enable-avfilter --enable-avresample --enable-libx264 --enable-filter=scale \

• Applying same filter_complex many times before output [duplicate]

  19 août 2019, par Fabián

  This question already has an answer here :

  • FFmpeg command to apply multiple filters and limit the final file size [closed]
    
    1 answer
    
  • Applying multiple filters at once with FFMPEG
    
    2 answers
    
  • FFMpeg - Combine multiple filter_complex and overlay functions
    
    1 answer
    
  • ffmpeg : multiple video filter order issue
    
    1 answer
    

  It’s not a duplicate. This is about using filter_complex, not -vf.

  In my video there’s an object that has shades of yellow (more orange-like) and a solid yellow as background.

  I need to output all frames into a png sequence, using a color key filter to replace the yellow from the background :

  ffmpeg -ss 4 -i original.mp4 -t 2 -filter_complex "[0:v]colorkey=0xfff31b:0.125:0[ckout]" -map "[ckout]" colorkey-%d.png

  This removes the specific color, but leaves some pints behind, and some items are yellow-themed, so blending value is a no-no for this scenario.

  I need to get rid of 4 specific yellow-colors from the frames : 0xfff31b, 0xfae56b, 0xfaec46 and 0xeee2a0, and I plan to run the same filter for specific colors before getting the final result.

  So first I tried this :

  ffmpeg -ss 4 -i original.mp4 -t 2 -filter_complex "[0:v]colorkey=0xfff31b:0.4:0[ckout1];[0:v]colorkey=0xfae56b:0.4:0[ckout2];[0:v]colorkey=0xfaec46:0.4:0[ckout3];[0:v]colorkey=0xeee2a0:0.4:0[ckout4]" -map "[ckout4]" colorkeyrefined-%d.png

  Then this :

  ffmpeg -ss 4 -i original.mp4 -t 2 -filter_complex "[0:v]colorkey=0xfff31b:0.4:0[ckout]" -filter_complex "[0:v]colorkey=0xfae56b:0.4:0[ckout]" -filter_complex "[0:v]colorkey=0xfaec46:0.4:0[ckout]" -filter_complex "[0:v]colorkey=0xeee2a0:0.4:0[ckout]" -map "[ckout]" colorkeyrefined-%d.png

  But both display the same error :

  Filter colorkey has an unconnected output.

  Is there a way to apply the colorkey feature 4 times (with the mentioned values) in one go ?

• MediaCodec AV Sync when decoding

  12 juin 2020, par ClassA

  All of the questions regarding syncing audio and video, when decoding using MediaCodec, suggests that we should use an "AV Sync" mechanism to sync the video and audio using their timestamps.

  



  Here is what I do to achieve this :

  



  I have 2 threads, one for decoding video and one for audio. To sync the video and audio I'm using Extractor.getSampleTime() to determine if I should release the audio or video buffers, please see below :

  



  //This is called after configuring MediaCodec(both audio and video)
private void startPlaybackThreads(){
    //Audio playback thread
    mAudioWorkerThread = new Thread("AudioThread") {
        @Override
        public void run() {
            if (!Thread.interrupted()) {
                try {
                    //Check info below
                    if (shouldPushAudio()) {
                        workLoopAudio();
                    }
                } catch (Exception e) {
                    e.printStackTrace();
                }
            }
        }
    };
    mAudioWorkerThread.start();

    //Video playback thread
    mVideoWorkerThread = new Thread("VideoThread") {
        @Override
        public void run() {
            if (!Thread.interrupted()) {
                try {
                    //Check info below
                    if (shouldPushVideo()) {
                        workLoopVideo();
                    }
                } catch (Exception e) {
                    e.printStackTrace();
                }
            }
        }
    };
    mVideoWorkerThread.start();
}

//Check if more buffers should be sent to the audio decoder
private boolean shouldPushAudio(){
    int audioTime =(int) mAudioExtractor.getSampleTime();
    int videoTime = (int) mExtractor.getSampleTime();
    return audioTime <= videoTime;
}
//Check if more buffers should be sent to the video decoder
private boolean shouldPushVideo(){
    int audioTime =(int) mAudioExtractor.getSampleTime();
    int videoTime = (int) mExtractor.getSampleTime();
    return audioTime > videoTime;
}


  



  Inside workLoopAudio() and workLoopVideo() is all my MediaCodec logic (I decided not to post it because it's not relevant).

  



  So what I do is, I get the sample time of the video and the audio tracks, I then check which one is bigger(further ahead). If the video is "ahead" then I pass more buffers to my audio decoder and visa versa.

  



  This seems to be working fine - The video and audio are playing in sync.

  



  ---

  



  My question :

  



  I would like to know if my approach is correct(is this how we should be doing it, or is there another/better way) ? I could not find any working examples of this(written in java/kotlin), thus the question.

  



  ---

  



  EDIT 1 :

  



  I've found that the audio trails behind the video (very slightly) when I decode/play a video that was encoded using FFmpeg. If I use a video that was not encoded using FFmpeg then the video and audio syncs perfectly.

  



  The FFmpeg command is nothing out of the ordinary :

  



  -i inputPath -crf 18 -c:v libx264 -preset ultrafast OutputPath


  



  I will be providing additional information below :

  



  I initialize/create AudioTrack like this :

  



  //Audio
mAudioExtractor = new MediaExtractor();
mAudioExtractor.setDataSource(mSource);
int audioTrackIndex = selectAudioTrack(mAudioExtractor);
if (audioTrackIndex < 0){
    throw new IOException("Can't find Audio info!");
}
mAudioExtractor.selectTrack(audioTrackIndex);
mAudioFormat = mAudioExtractor.getTrackFormat(audioTrackIndex);
mAudioMime = mAudioFormat.getString(MediaFormat.KEY_MIME);

mAudioChannels = mAudioFormat.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
mAudioSampleRate = mAudioFormat.getInteger(MediaFormat.KEY_SAMPLE_RATE);

final int min_buf_size = AudioTrack.getMinBufferSize(mAudioSampleRate, (mAudioChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO), AudioFormat.ENCODING_PCM_16BIT);
final int max_input_size = mAudioFormat.getInteger(MediaFormat.KEY_MAX_INPUT_SIZE);
mAudioInputBufSize =  min_buf_size > 0 ? min_buf_size * 4 : max_input_size;
if (mAudioInputBufSize > max_input_size) mAudioInputBufSize = max_input_size;
final int frameSizeInBytes = mAudioChannels * 2;
mAudioInputBufSize = (mAudioInputBufSize / frameSizeInBytes) * frameSizeInBytes;

mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
    mAudioSampleRate,
    (mAudioChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO),
    AudioFormat.ENCODING_PCM_16BIT,
    AudioTrack.getMinBufferSize(mAudioSampleRate, mAudioChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT),
    AudioTrack.MODE_STREAM);

try {
    mAudioTrack.play();
} catch (final Exception e) {
    Log.e(TAG, "failed to start audio track playing", e);
    mAudioTrack.release();
    mAudioTrack = null;
}


  



  And I write to the AudioTrack like this :

  



  //Called from within workLoopAudio, when releasing audio buffers
if (bufferAudioIndex >= 0) {
    if (mAudioBufferInfo.size > 0) {
        internalWriteAudio(mAudioOutputBuffers[bufferAudioIndex], mAudioBufferInfo.size);
    }
    mAudioDecoder.releaseOutputBuffer(bufferAudioIndex, false);
}

private boolean internalWriteAudio(final ByteBuffer buffer, final int size) {
    if (mAudioOutTempBuf.length < size) {
        mAudioOutTempBuf = new byte[size];
    }
    buffer.position(0);
    buffer.get(mAudioOutTempBuf, 0, size);
    buffer.clear();
    if (mAudioTrack != null)
        mAudioTrack.write(mAudioOutTempBuf, 0, size);
    return true;
}


  



  "NEW" Question :

  



  The audio trails about 200ms behind the video if I use a video that was encoded using FFmpeg, is there a reason why this could be happening ?