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• Transcoding audio using xuggler

  23 juin 2014, par amd

  I am trying to convert an audio file with the header

  Opening audio decoder: [pcm] Uncompressed PCM audio decoder
  
  AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400)
  
  Selected audio codec: [pcm] afm: pcm (Uncompressed PCM)

  I want to transcode this file to mp3 format. I have following code snippet but its not working well. I have written it using XUGGLER code snippet for transcoding audio and video.

  Audio decoder is

      audioDecoder = IStreamCoder.make(IStreamCoder.Direction.DECODING, ICodec.findDecodingCodec(ICodec.ID.CODEC_ID_PCM_S16LE));
  
      audioDecoder.setSampleRate(44100);
  
      audioDecoder.setBitRate(176400);
  
      audioDecoder.setChannels(2);
  
      audioDecoder.setTimeBase(IRational.make(1,1000));
  
      if (audioDecoder.open(IMetaData.make(), IMetaData.make()) < 0)
  
          return false;
  
      return true; 

  Audio encoder is

      outContainer = IContainer.make();
  
      outContainerFormat = IContainerFormat.make();
  
      outContainerFormat.setOutputFormat("mp3", urlOut, null);
  
      int retVal = outContainer.open(urlOut, IContainer.Type.WRITE, outContainerFormat);
  
      if (retVal < 0) {
  
          System.out.println("Could not open output container");
  
          return false;
  
      }
  
      outAudioCoder = IStreamCoder.make(IStreamCoder.Direction.ENCODING, ICodec.findEncodingCodec(ICodec.ID.CODEC_ID_MP3));
  
      outAudioStream = outContainer.addNewStream(outAudioCoder);
  
      outAudioCoder.setSampleRate(new Integer(44100));
  
      outAudioCoder.setChannels(2);
  
      retVal = outAudioCoder.open(IMetaData.make(), IMetaData.make());
  
      if (retVal < 0) {
  
          System.out.println("Could not open audio coder");
  
          return false;
  
      }
  
      retVal = outContainer.writeHeader();
  
      if (retVal < 0) {
  
          System.out.println("Could not write output FLV header: ");
  
          return false;
  
      }
  
      return true;

  And here is encode method where i send packets of 32 byte to transcode

  public void encode(byte[] audioFrame){
  
      //duration of 1 video frame
  
      long lastVideoPts = 0;
  
  
  
      IPacket packet_out = IPacket.make();
  
      int lastPos = 0;
  
      int lastPos_out = 0;
  
  
  
      IAudioSamples audioSamples = IAudioSamples.make(48000, audioDecoder.getChannels());
  
      IAudioSamples audioSamples_resampled = IAudioSamples.make(48000, audioDecoder.getChannels());
  
  
  
      //we always have 32 bytes/sample
  
      int pos = 0;
  
      int audioFrameLength = audioFrame.length;
  
      int audioFrameCnt = 1;
  
      iBuffer = IBuffer.make(null, audioFrame, 0, audioFrameLength);
  
      IPacket packet = IPacket.make(iBuffer);
  
      //packet.setKeyPacket(true);
  
      packet.setTimeBase(IRational.make(1,1000));
  
      packet.setDuration(20);
  
      packet.setDts(audioFrameCnt*20);
  
      packet.setPts(audioFrameCnt*20);
  
      packet.setStreamIndex(1);
  
      packet.setPosition(lastPos);
  
      lastPos+=audioFrameLength;
  
      int pksz = packet.getSize();
  
      packet.setComplete(true, pksz);
  
      /*
  
      * A packet can actually contain multiple samples
  
      */
  
      int offset = 0;
  
      int retVal;
  
      while(offset < packet.getSize())
  
      {
  
          int bytesDecoded = audioDecoder.decodeAudio(audioSamples, packet, offset);
  
          if (bytesDecoded < 0)
  
              throw new RuntimeException("got error decoding audio ");
  
          offset += bytesDecoded;
  
          if (audioSamples.isComplete())
  
          {
  
              int samplesConsumed = 0;
  
              while (samplesConsumed < audioSamples.getNumSamples()) {
  
                  retVal = outAudioCoder.encodeAudio(packet_out, audioSamples, samplesConsumed);
  
                  if (retVal <= 0)
  
                      throw new RuntimeException("Could not encode audio");
  
                  samplesConsumed += retVal;
  
                  if (packet_out.isComplete()) {
  
                      packet_out.setPosition(lastPos_out);
  
                      packet_out.setStreamIndex(1);
  
                      lastPos_out+=packet_out.getSize();
  
                      retVal = outContainer.writePacket(packet_out);
  
                      if(retVal < 0){
  
                          throw new RuntimeException("Could not write data packet");
  
                      }
  
                  }
  
              }
  
          }
  
  
  
      }
  
  
  
  }

  I get an output file but it doesnt get played. I have very little experience of audio encoding and sampling. Thanks in advance.

• When using ffmpeg to create mp4 video file from batch of images the whole process is very slow how can i make it faster ?

  29 juin 2015, par Brubaker Haim

  The whole process is slow and also in the end the video file when playing it the frames moving very slow.

  ffmpeg -framerate 1/5 -i screenshot%06d.jpg -c:v libx264 -r 30 -p
  
  ix_fmt yuv420p out2.mp4

  Is that mean 1 frames each 5 seconds ?
  So if i will make 5/1 it will be 5 frames in a second ?
  What should be the best result ?

  And the second problem is that for testing i have 70 images but in the original i have over 1000 images is there any way to make all this process faster ?

• Android how to increase ffmpeg mp4 perfromance ?

  4 janvier 2013, par testCoder

  I have detected that function avcodec_decode_audio3 works slow with mp4 format, here my code cycle for decoding audio :

  while (av_read_frame(av_format_context, &packet) >= 0 && is_play == 1) {
  
          if (av_codec_context->codec_type == AVMEDIA_TYPE_AUDIO
  
                  && is_play == 1) {
  
              int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
              int size = packet.size;
  
              int n;
  
              int dataLength = size;
  
              int decoded = 0;
  
              while (size > 0) {
  
  
  
                  //start measure time
  
                  gettimeofday(&tvBegin, NULL);
  
  
  
                  int len = avcodec_decode_audio3(av_codec_context,
  
                          (int16_t *) pAudioBuffer, &out_size, &packet);
  
  
  
                  //stop measure time
  
                  gettimeofday(&tvEnd, NULL);
  
                  timeval_subtract(&tvDiff, &tvEnd, &tvBegin);
  
  
  
                  LOGI("%d", tvDiff.tv_usec / 1000);
  
                  LOGI("len='%d'", len);
  
                  LOGI("out_size='%d'", out_size);
  
  
  
                  if (len < 0) {
  
                      break;
  
                      return 1;
  
                  }
  
                  if (out_size > 0) {
  
  
  
  
  
  
  
                      jbyte *bytes = (*env)->GetByteArrayElements(env, array,
  
                              NULL);
  
                      memcpy(bytes, (int16_t *) pAudioBuffer, out_size);
  
                      (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
                      (*env)->CallVoidMethod(env, obj, play, array, out_size,
  
                              is_play);
  
  
  
                  }
  
                  size -= len;
  
              }
  
  
  
          }
  
          if (packet.data)
  
              av_free_packet(&packet);
  
  
  
      }

  But with other formats like flac and mp3 it works fine. avcodec_decode_audio3 take about 1-2 milisecounds for decoding mp3 frame with out_size = 4608 but with the same frame size in mp4 decoding take about 6-7 millisecounds. I got my build script from here.

  Does it normal behavior ? Is any way to increase performance of decoding mp4 ?