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• Revision e1ff83f4b0 : vp9_full_search_sadx[38] : align sad arrays the sse4 code expects 16-byte aligne

  7 avril 2015, par James Zern

  Changed Paths :
   Modify /vp9/encoder/vp9_mcomp.c

  
  
  vp9_full_search_sadx[38] : align sad arrays

  the sse4 code expects 16-byte aligned arrays ; vp8 already had a similar
  change applied :
  b2aa401 Align SAD output array to be 16-byte aligned

  Change-Id : I5e902035e5a87e23309e151113f3c0d4a8372226

• Muxing Android MediaCodec encoded H264 packets into RTMP

  31 décembre 2015, par Vadym

  I am coming from a thread Encoding H.264 from camera with Android MediaCodec. My setup is very similar. However, I attempt to write mux the encoded frames and with javacv and broadcast them via rtmp.

  RtmpClient.java

  ...
  
  private volatile BlockingQueue mFrameQueue = new LinkedBlockingQueue(MAXIMUM_VIDEO_FRAME_BACKLOG);
  
  ...
  
  private void startStream() throws FrameRecorder.Exception, IOException {
  
      if (TextUtils.isEmpty(mDestination)) {
  
          throw new IllegalStateException("Cannot start RtmpClient without destination");
  
      }
  
  
  
      if (mCamera == null) {
  
          throw new IllegalStateException("Cannot start RtmpClient without camera.");
  
      }
  
  
  
      Camera.Parameters cameraParams = mCamera.getParameters();
  
  
  
      mRecorder = new FFmpegFrameRecorder(
  
              mDestination,
  
              mVideoQuality.resX,
  
              mVideoQuality.resY,
  
              (mAudioQuality.channelType.equals(AudioQuality.CHANNEL_TYPE_STEREO) ? 2 : 1));
  
  
  
      mRecorder.setFormat("flv");
  
  
  
      mRecorder.setFrameRate(mVideoQuality.frameRate);
  
      mRecorder.setVideoBitrate(mVideoQuality.bitRate);
  
      mRecorder.setVideoCodec(avcodec.AV_CODEC_ID_H264);
  
  
  
      mRecorder.setSampleRate(mAudioQuality.samplingRate);
  
      mRecorder.setAudioBitrate(mAudioQuality.bitRate);
  
      mRecorder.setAudioCodec(avcodec.AV_CODEC_ID_AAC);
  
  
  
      mVideoStream = new VideoStream(mRecorder, mVideoQuality, mFrameQueue, mCamera);
  
      mAudioStream = new AudioStream(mRecorder, mAudioQuality);
  
  
  
      mRecorder.start();
  
  
  
      // Setup a bufferred preview callback
  
      setupCameraCallback(mCamera, mRtmpClient, DEFAULT_PREVIEW_CALLBACK_BUFFERS,
  
              mVideoQuality.resX * mVideoQuality.resY * ImageFormat.getBitsPerPixel(
  
                      cameraParams.getPreviewFormat())/8);
  
  
  
      try {
  
          mVideoStream.start();
  
          mAudioStream.start();
  
      }
  
      catch(Exception e) {
  
          e.printStackTrace();
  
          stopStream();
  
      }
  
  }
  
  ...
  
  @Override
  
  public void onPreviewFrame(byte[] data, Camera camera) {
  
      boolean frameQueued = false;
  
  
  
      if (mRecorder == null || data == null) {
  
          return;
  
      }
  
  
  
      frameQueued = mFrameQueue.offer(data);
  
  
  
      // return the buffer to be reused - done in videostream
  
      //camera.addCallbackBuffer(data);
  
  }
  
  ...

  VideoStream.java

  ...
  
  @Override
  
  public void run() {
  
      try {
  
          mMediaCodec = MediaCodec.createEncoderByType("video/avc");
  
          MediaFormat mediaFormat = MediaFormat.createVideoFormat("video/avc", mVideoQuality.resX, mVideoQuality.resY);
  
          mediaFormat.setInteger(MediaFormat.KEY_BIT_RATE, mVideoQuality.bitRate);
  
          mediaFormat.setInteger(MediaFormat.KEY_FRAME_RATE, mVideoQuality.frameRate);
  
          mediaFormat.setInteger(MediaFormat.KEY_COLOR_FORMAT, MediaCodecInfo.CodecCapabilities.COLOR_FormatYUV420SemiPlanar);
  
          mediaFormat.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, 1);
  
          mMediaCodec.configure(mediaFormat, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
  
          mMediaCodec.start();
  
      }
  
      catch(IOException e) {
  
          e.printStackTrace();
  
      }
  
  
  
      long startTimestamp = System.currentTimeMillis();
  
      long frameTimestamp = 0;
  
      byte[] rawFrame = null;
  
  
  
      try {
  
          while (!Thread.interrupted()) {
  
              rawFrame = mFrameQueue.take();
  
  
  
              frameTimestamp = 1000 * (System.currentTimeMillis() - startTimestamp);
  
  
  
              encodeFrame(rawFrame, frameTimestamp);
  
  
  
              // return the buffer to be reused
  
              mCamera.addCallbackBuffer(rawFrame);
  
          }
  
      }
  
      catch (InterruptedException ignore) {
  
          // ignore interrup while waiting
  
      }
  
  
  
      // Clean up video stream allocations
  
      try {
  
          mMediaCodec.stop();
  
          mMediaCodec.release();
  
          mOutputStream.flush();
  
          mOutputStream.close();
  
      } catch (Exception e){
  
          e.printStackTrace();
  
      }
  
  }
  
  ...
  
  private void encodeFrame(byte[] input, long timestamp) {
  
      try {
  
          ByteBuffer[] inputBuffers = mMediaCodec.getInputBuffers();
  
          ByteBuffer[] outputBuffers = mMediaCodec.getOutputBuffers();
  
  
  
          int inputBufferIndex = mMediaCodec.dequeueInputBuffer(0);
  
  
  
          if (inputBufferIndex >= 0) {
  
              ByteBuffer inputBuffer = inputBuffers[inputBufferIndex];
  
              inputBuffer.clear();
  
              inputBuffer.put(input);
  
              mMediaCodec.queueInputBuffer(inputBufferIndex, 0, input.length, timestamp, 0);
  
          }
  
  
  
          MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
  
  
  
          int outputBufferIndex = mMediaCodec.dequeueOutputBuffer(bufferInfo, 0);
  
  
  
          if (outputBufferIndex >= 0) {
  
              while (outputBufferIndex >= 0) {
  
                  ByteBuffer outputBuffer = outputBuffers[outputBufferIndex];
  
  
  
                  // Should this be a direct byte buffer?
  
                  byte[] outData = new byte[bufferInfo.size - bufferInfo.offset];
  
                  outputBuffer.get(outData);
  
  
  
                  mFrameRecorder.record(outData, bufferInfo.offset, outData.length, timestamp);
  
  
  
                  mMediaCodec.releaseOutputBuffer(outputBufferIndex, false);
  
                  outputBufferIndex = mMediaCodec.dequeueOutputBuffer(bufferInfo, 0);
  
              }
  
          }
  
          else if (outputBufferIndex == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
  
              outputBuffers = mMediaCodec.getOutputBuffers();
  
          } else if (outputBufferIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
  
              // ignore for now
  
          }
  
      } catch (Throwable t) {
  
          t.printStackTrace();
  
      }
  
  
  
  }
  
  ...

  FFmpegFrameRecorder.java

  ...
  
  // Hackish codec copy frame recording function
  
  public boolean record(byte[] encodedData, int offset, int length, long frameCount) throws Exception {
  
      int ret;
  
  
  
      if (encodedData == null) {
  
          return false;
  
      }
  
  
  
      av_init_packet(video_pkt);
  
  
  
      // this is why i wondered whether I should get outputbuffer data into direct byte buffer
  
      video_outbuf.put(encodedData, 0, encodedData.length);
  
  
  
      video_pkt.data(video_outbuf);
  
      video_pkt.size(video_outbuf_size);
  
  
  
      video_pkt.pts(frameCount);
  
      video_pkt.dts(frameCount);
  
  
  
      video_pkt.stream_index(video_st.index());
  
  
  
      synchronized (oc) {
  
          /* write the compressed frame in the media file */
  
          if (interleaved && audio_st != null) {
  
              if ((ret = av_interleaved_write_frame(oc, video_pkt)) < 0) {
  
                  throw new Exception("av_interleaved_write_frame() error " + ret + " while writing interleaved video frame.");
  
              }
  
          } else {
  
              if ((ret = av_write_frame(oc, video_pkt)) < 0) {
  
                  throw new Exception("av_write_frame() error " + ret + " while writing video frame.");
  
              }
  
          }
  
      }
  
      return (video_pkt.flags() & AV_PKT_FLAG_KEY) == 1;
  
  }
  
  ...

  When I try to stream the video and run ffprobe on it, I get the following output :

  ffprobe version 2.5.3 Copyright (c) 2007-2015 the FFmpeg developers
  
    built on Jan 19 2015 12:56:57 with gcc 4.1.2 (GCC) 20080704 (Red Hat 4.1.2-55)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --disable-crystalhd --enable-libass --enable-libdc1394 --enable-libfaac --enable-nonfree --disable-indev=jack --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-openal --enable-libopencv --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --enable-libcaca --shlibdir=/usr/lib64 --enable-runtime-cpudetect
  
    libavutil      54. 15.100 / 54. 15.100
  
    libavcodec     56. 13.100 / 56. 13.100
  
    libavformat    56. 15.102 / 56. 15.102
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  2.103 /  5.  2.103
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Metadata:
  
    Server                NGINX RTMP (github.com/arut/nginx-rtmp-module)
  
    width                 320.00
  
    height                240.00
  
    displayWidth          320.00
  
    displayHeight         240.00
  
    duration              0.00
  
    framerate             0.00
  
    fps                   0.00
  
    videodatarate         261.00
  
    videocodecid          7.00
  
    audiodatarate         62.00
  
    audiocodecid          10.00
  
    profile
  
    level
  
  [live_flv @ 0x1edb0820] Could not find codec parameters for stream 0 (Video: none, none, 267 kb/s): unknown codec
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  Input #0, live_flv, from 'rtmp://<server>/input/<stream>':
  
    Metadata:
  
      Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
  
      displayWidth    : 320
  
      displayHeight   : 240
  
      fps             : 0
  
      profile         :
  
      level           :
  
    Duration: 00:00:00.00, start: 16.768000, bitrate: N/A
  
      Stream #0:0: Video: none, none, 267 kb/s, 1k tbr, 1k tbn, 1k tbc
  
      Stream #0:1: Audio: aac (LC), 16000 Hz, mono, fltp, 63 kb/s
  
  Unsupported codec with id 0 for input stream 0
  
  </stream></server>

  I am not, by any means, an expert in H264 or video encoding. I know that the encoded frames that come out from MediaCodec contain SPS NAL, PPS NAL, and frame NAL units. I’ve also written the MediaCodec output into a file and was able to play it back (I did have to specify the format and framerate as otherwise it would play too fast).

  My assumption is that things should work (see how little I know :)). Knowing that SPS and PPS are written out, decoder should know enough. Yet, ffprobe fails to recognize codec, fps, and other video information. Do I need to pass packet flag information to FFmpegFrameRecorder.java:record() function ? Or should I use direct buffer ? Any suggestion will be appreciated ! I should figure things out with a hint.

  PS : I know that some codecs use Planar and other SemiPlanar color formats. That distinction will come later if I get past this. Also, I didn’t go the Surface to MediaCodec way because I need to support API 17 and it requires more changes than this route, which I think helps me understand the more basic flow. Agan, I appreciate any suggestions. Please let me know if something needs to be clarified.

  Update #1

  So having done more testing, I see that my encoder outputs the following frames :

  000000016742800DDA0507E806D0A1350000000168CE06E2
  
  0000000165B840A6F1E7F0EA24000AE73BEB5F51CC7000233A84240...
  
  0000000141E2031364E387FD4F9BB3D67F51CC7000279B9F9CFE811...
  
  0000000141E40304423FFFFF0B7867F89FAFFFFFFFFFFCBE8EF25E6...
  
  0000000141E602899A3512EF8AEAD1379F0650CC3F905131504F839...
  
  ...

  The very first frame contains SPS and PPS. From what I was able to see, these are transmitted only once. The rest are NAL types 1 and 5. So, my assumption is that, for ffprobe to see stream info not only when the stream starts, I should capture SPS and PPS frames and re-transmit them myself periodically, after a certain number of frames, or perhaps before every I-frame. What do you think ?

  Update #2

  Unable to validate that I’m writing frames successfully. After having tried to read back the written packet, I cannot validate written bytes. As strange, on successful write of IPL image and streaming, I also cannot print out bytes of encoded packet after avcodec_encode_video2. Hit the official dead end.

• rtpdec : Don’t free the payload context in the .free function

  24 février 2015, par Martin Storsjö

  rtpdec : Don’t free the payload context in the .free function
  This makes it more consistent with depacketizers that don’t have any
  
  .free function at all, where the payload context is freed by the
  
  surrounding framework. Always free the context in the surrounding
  
  framework, having the individual depacketizers only free any data
  
  they’ve specifically allocated themselves.
  This is similar to how this works for demuxer/muxers/codecs - a
  
  component shouldn’t free the priv_data that the framework has
  
  allocated for it.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libavformat/rdt.c
  • [DH] libavformat/rtpdec.h
  • [DH] libavformat/rtpdec_ac3.c
  • [DH] libavformat/rtpdec_dv.c
  • [DH] libavformat/rtpdec_h261.c
  • [DH] libavformat/rtpdec_h263_rfc2190.c
  • [DH] libavformat/rtpdec_h264.c
  • [DH] libavformat/rtpdec_jpeg.c
  • [DH] libavformat/rtpdec_latm.c
  • [DH] libavformat/rtpdec_mpa_robust.c
  • [DH] libavformat/rtpdec_mpeg4.c
  • [DH] libavformat/rtpdec_mpegts.c
  • [DH] libavformat/rtpdec_qt.c
  • [DH] libavformat/rtpdec_svq3.c
  • [DH] libavformat/rtpdec_vp8.c
  • [DH] libavformat/rtpdec_xiph.c
  • [DH] libavformat/rtsp.c