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• avfilter/vf_delogo : round to the closest value

  9 décembre 2015, par Jean Delvare

  avfilter/vf_delogo : round to the closest value
  When the interpolated value is divided by the sum of weights, no
  
  rounding is done, which means the value is truncated. This results in
  
  a slight bias towards dark green in the interpolated area. Rounding
  
  properly removes the bias.
  I measured this change to reduce the interpolation error by 1 to 2 %
  
  on average on a number of sample input and logo area combinations.
  Signed-off-by : Jean Delvare <jdelvare@suse.de>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavfilter/vf_delogo.c
  • [DH] tests/ref/fate/filter-delogo

• Syncing Audio with Video iOS RTSP Player

  3 octobre 2015, par Dave Thomas

  I am combining two different classes from two different git projects to create an RTSP streamer for an iOS live streaming application.

  edit : Agree with the -1 this question is probably a shot in the dark. But, to answer the question if I am asked "why I am not using entirely the DFURTSPPlayer library ?" Because I would rather use the YUV display with opengl of the second project, hackcam, rather than decode the video frames into a UIImages like DFURTS does. Hackcam does not have audio

  Also please comment if you down vote, at least help me find an answer by telling me what I need to refine to be clear or point out if this question is inappropriate

  My current issue is that the audio playback has about a 1 second latency, and is out of sync with the video which is close to real time.

  I know that the audio is in sync because I’ve tested the RTSP streams in VLC. Something is wrong with my implementation. Mostly frankensteining these too projects together and the fact that I am not familiar with ffmpeg c library or AudioQueue for iOS.

  Any help would be greatly appreciated !

  I’ve taken the AudioStreamer class from this repository :
  https://github.com/durfu/DFURTSPPlayer

  https://github.com/durfu/DFURTSPPlayer/blob/master/DFURTSPPlayer/DFURTSPPlayer/FFMpegDecoder/AudioStreamer.m

  And I am trying to get it to work with this one :
  https://github.com/hackacam/ios_rtsp_player/blob/master/src/FfmpegWrapper.m

  I can post more code if needed, but my main loop in FfmpegWrapper now looks like this (_audioController is reference to AudioStreamer.m) :

  -(int) startDecodingWithCallbackBlock: (void (^) (AVFrameData *frame)) frameCallbackBlock
  
                        waitForConsumer: (BOOL) wait
  
                     completionCallback: (void (^)()) completion
  
  {
  
      OSMemoryBarrier();
  
      _stopDecode=false;
  
      dispatch_queue_t decodeQueue = dispatch_queue_create("decodeQueue", NULL);
  
      dispatch_async(decodeQueue, ^{
  
          int frameFinished;
  
          OSMemoryBarrier();
  
          while (self->_stopDecode==false){
  
              @autoreleasepool {
  
                  CFTimeInterval currentTime = CACurrentMediaTime();
  
                  if ((currentTime-_previousDecodedFrameTime) > MIN_FRAME_INTERVAL &&
  
                      av_read_frame(_formatCtx, &_packetFFmpeg)>=0) {
  
  
  
                      _previousDecodedFrameTime = currentTime;
  
                      // Is this a packet from the video stream?
  
                      if(_packetFFmpeg.stream_index==_videoStream) {
  
                          // Decode video frame
  
                          avcodec_decode_video2(_codecCtx, _frame, &frameFinished,
  
                                                &_packetFFmpeg);
  
  
  
                          // Did we get a video frame?
  
                          if(frameFinished) {
  
                              // create a frame object and call the block;
  
                              AVFrameData *frameData = [self createFrameData:_frame trimPadding:YES];
  
                              frameCallbackBlock(frameData);
  
                          }
  
  
  
                          // Free the packet that was allocated by av_read_frame
  
                          av_free_packet(&_packetFFmpeg);
  
  
  
                      } else if (_packetFFmpeg.stream_index==audioStream) {
  
  
  
                          // NSLog(@"audio stream");
  
                          [audioPacketQueueLock lock];
  
  
  
                          audioPacketQueueSize += _packetFFmpeg.size;
  
                          [audioPacketQueue addObject:[NSMutableData dataWithBytes:&_packetFFmpeg length:sizeof(_packetFFmpeg)]];
  
  
  
                          [audioPacketQueueLock unlock];
  
  
  
                          if (!primed) {
  
                              primed=YES;
  
                              [_audioController _startAudio];
  
                          }
  
  
  
                          if (emptyAudioBuffer) {
  
                              [_audioController enqueueBuffer:emptyAudioBuffer];
  
                          }
  
  
  
                          //av_free_packet(&_packetFFmpeg);
  
  
  
                      } else {
  
  
  
                          // Free the packet that was allocated by av_read_frame
  
                          av_free_packet(&_packetFFmpeg);
  
                      }
  
  
  
  
  
                  } else{
  
                      usleep(1000);
  
                  }
  
              }
  
          }
  
          completion();
  
      });
  
      return 0;
  
  }

  Enqueue Buffer in AudioStreamer :

  - (OSStatus)enqueueBuffer:(AudioQueueBufferRef)buffer
  
  {
  
      OSStatus status = noErr;
  
  
  
      if (buffer) {
  
          AudioTimeStamp bufferStartTime;
  
          buffer->mAudioDataByteSize = 0;
  
          buffer->mPacketDescriptionCount = 0;
  
  
  
          if (_streamer.audioPacketQueue.count <= 0) {
  
              _streamer.emptyAudioBuffer = buffer;
  
              return status;
  
          }
  
  
  
          _streamer.emptyAudioBuffer = nil;
  
  
  
          while (_streamer.audioPacketQueue.count && buffer->mPacketDescriptionCount < buffer->mPacketDescriptionCapacity) {
  
              AVPacket *packet = [_streamer readPacket];
  
  
  
              if (buffer->mAudioDataBytesCapacity - buffer->mAudioDataByteSize >= packet->size) {
  
                  if (buffer->mPacketDescriptionCount == 0) {
  
                      bufferStartTime.mSampleTime = packet->dts * _audioCodecContext->frame_size;
  
                      bufferStartTime.mFlags = kAudioTimeStampSampleTimeValid;
  
                  }
  
  
  
                  memcpy((uint8_t *)buffer->mAudioData + buffer->mAudioDataByteSize, packet->data, packet->size);
  
                  buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mStartOffset = buffer->mAudioDataByteSize;
  
                  buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mDataByteSize = packet->size;
  
                  buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mVariableFramesInPacket = _audioCodecContext->frame_size;
  
  
  
                  buffer->mAudioDataByteSize += packet->size;
  
                  buffer->mPacketDescriptionCount++;
  
  
  
  
  
                  _streamer.audioPacketQueueSize -= packet->size;
  
  
  
                  av_free_packet(packet);
  
              }
  
              else {
  
  
  
                  //av_free_packet(packet);
  
                  break;
  
              }
  
          }
  
  
  
          [decodeLock_ lock];
  
          if (buffer->mPacketDescriptionCount > 0) {
  
              status = AudioQueueEnqueueBuffer(audioQueue_, buffer, 0, NULL);
  
              if (status != noErr) {
  
                  NSLog(@"Could not enqueue buffer.");
  
              }
  
          } else {
  
              AudioQueueStop(audioQueue_, NO);
  
              finished_ = YES;
  
          }
  
  
  
          [decodeLock_ unlock];
  
      }
  
  
  
      return status;
  
  }

  Read packet in ffmpegwrapper :

  - (AVPacket*)readPacket
  
  {
  
      if (_currentPacket.size > 0 || _inBuffer) return &_currentPacket;
  
  
  
      NSMutableData *packetData = [audioPacketQueue objectAtIndex:0];
  
      _packet = [packetData mutableBytes];
  
  
  
      if (_packet) {
  
          if (_packet->dts != AV_NOPTS_VALUE) {
  
              _packet->dts += av_rescale_q(0, AV_TIME_BASE_Q, _audioStream->time_base);
  
          }
  
  
  
          if (_packet->pts != AV_NOPTS_VALUE) {
  
              _packet->pts += av_rescale_q(0, AV_TIME_BASE_Q, _audioStream->time_base);
  
          }
  
  
  
          [audioPacketQueueLock lock];
  
          audioPacketQueueSize -= _packet->size;
  
          if ([audioPacketQueue count] > 0) {
  
              [audioPacketQueue removeObjectAtIndex:0];
  
          }
  
          [audioPacketQueueLock unlock];
  
  
  
          _currentPacket = *(_packet);
  
      }
  
  
  
      return &_currentPacket;
  
  }

• ffmpeg streaming camera with directshow

  17 novembre 2015, par atu0830

  I am trying use ffmpeg to streaming one camera. The command is

  ffmpeg.exe -y  -f dshow -i video="AmCam" -c:v copy  -framerate 7.5 -map 0:0 -f ssegment -segment_time 4 -segment_format mpegts  -segment_list "web\stream.m3u8"  -segment_list_size 720  -segment_list_flags live  -segment_wrap 10 -segment_list_type m3u8  "web\segments\s%%d.ts"

  And I create a html in web folder

                                        
  
                                  
  
                                       
  
                                    
  
      <video controls="controls" width="720" height="405" autoplay="autoplay">   
  
          <source src="stream.m3u8" type="application/x-mpegURL"></source>           
  
      </video>                                    
  
                                       
  
   

  All ts file generated but looking Safari on iPad looding but it always show dark player and loading