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• Publish RTMP stream to Red5 Server form iOS camera

  7 septembre 2015, par Mohammad Asif

  Please look at following code, I have transformed CMSampleBufferRef into AV_CODEC_ID_H264 but I don’t know how to transmit it to Red5 server.

  Thanks,

  - (void)  captureOutput:(AVCaptureOutput *)captureOutput
  
    didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
  
       fromConnection:(AVCaptureConnection *)connection {
  
  
  
  //NSLog(@"This is working ....");
  
  
  
    // [connection setVideoOrientation: [self deviceOrientation] ];
  
  
  
  if( !CMSampleBufferDataIsReady(sampleBuffer) )
  
  {
  
      NSLog( @"sample buffer is not ready. Skipping sample" );
  
      return;
  
  } else {
  
  
  
      if (captureOutput == videoOutput) {
  
  
  
  
  
          CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
  
          CVPixelBufferLockBaseAddress(pixelBuffer, 0);
  
  
  
          // access the data
  
          float width = CVPixelBufferGetWidth(pixelBuffer);
  
          float height = CVPixelBufferGetHeight(pixelBuffer);
  
  
  
          //float bytesPerRow = CVPixelBufferGetBytesPerRowOfPlane(pixelBuffer, 0);
  
          unsigned char *rawPixelBase = (unsigned char *)CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 0);
  
  
  
  
  
          // Convert the raw pixel base to h.264 format
  
  
  
          if (codec == nil) {
  
  
  
              codec = 0;
  
              context = 0;
  
              frame = 0;
  
  
  
              fmt = avformat_alloc_context();
  
              //avformat_write_header(fmt, NULL);
  
  
  
              codec = avcodec_find_encoder(AV_CODEC_ID_H264);
  
  
  
              if (codec == 0) {
  
                  NSLog(@"Codec not found!!");
  
                  return;
  
              }
  
  
  
              context = avcodec_alloc_context3(codec);
  
  
  
              if (!context) {
  
                  NSLog(@"Context no bueno.");
  
                  return;
  
              }
  
  
  
              // Bit rate
  
              context->bit_rate = 400000; // HARD CODE
  
              context->bit_rate_tolerance = 10;
  
              // Resolution
  
  
  
              // Frames Per Second
  
              context->time_base = (AVRational) {1,25};
  
              context->gop_size = 1;
  
              //context->max_b_frames = 1;
  
              context->width = width;
  
              context->height = height;
  
              context->pix_fmt = PIX_FMT_YUV420P;
  
  
  
              // Open the codec
  
              if (avcodec_open2(context, codec, 0) < 0) {
  
                  NSLog(@"Unable to open codec");
  
                  return;
  
              }
  
  
  
              // Create the frame
  
              frame = av_frame_alloc();
  
              if (!frame) {
  
                  NSLog(@"Unable to alloc frame");
  
                  return;
  
              }
  
          }
  
  
  
          context->width = width;
  
          context->height = height;
  
  
  
          frame->format = context->pix_fmt;
  
          frame->width = context->width;
  
          frame->height = context->height;
  
  
  
          //int nbytes = avpicture_get_size(context->pix_fmt, context->width, context->height);
  
          //uint8_t* outbuffer = (uint8_t*)av_malloc(nbytes);
  
  //            AVFrame *pFrameDecoded = avcodec_alloc_frame();
  
  //            int num_bytes2 = avpicture_get_size(context->pix_fmt, frame->width, frame->height);
  
  //            uint8_t* frame2_buffer2 = (uint8_t *)av_malloc(num_bytes2 * sizeof(uint8_t));
  
  //            avpicture_fill((AVPicture*)pFrameDecoded, frame2_buffer2, PIX_FMT_YUVJ422P, 320, 240);
  
  
  
          frame->pts = (1.0 / 30) * 60 * count;
  
          avpicture_fill((AVPicture *) frame, rawPixelBase, context->pix_fmt, frame->width, frame->height);
  
  
  
          int got_output = 0;
  
          av_init_packet(&packet);
  
          //avcodec_encode_video2(context, &packet, frame, &got_output);
  
  
  
          do {
  
              avcodec_encode_video2(context, &packet, frame, &got_output);
  
              //avcodec_decode_video2(context, &packet, NULL, &got_output);
  
              //*... handle received packet*/
  
  
  
              if (isFirstPacket) {
  
                  [rtmp sendCreateStreamPacket];
  
                  isFirstPacket = false;
  
                  //av_dump_format(fmt, 0, [kRtmpEP UTF8String], 1);
  
                  avformat_alloc_output_context2(&ofmt_ctx, NULL, "flv", [kRtmpEP UTF8String]); //RTMP
  
              }
  
  
  
              packet.stream_index = ofmt_ctx->nb_streams;
  
              av_interleaved_write_frame(ofmt_ctx, &packet);
  
              count ++;
  
  
  
              //[rtmp write:[NSData dataWithBytes:packet.data length:packet.size]];
  
  
  
  
  
          } while(got_output);
  
  
  
          // Unlock the pixel data
  
          CVPixelBufferUnlockBaseAddress(pixelBuffer, 0);
  
  
  
          //[rtmp write:[NSData dataWithBytes:packet.data length:packet.size]];
  
  
  
      } else {
  
  
  
      }
  
  }   }   

• Don't throw an error when attached picture isn't recognized.

  10 avril 2023, par Dale Curtis

  Don't throw an error when attached picture isn't recognized.
  The MIME type field is required per the FLAC standard, but it's
  
  not an error just because ffmpeg doesn't recognize it.
  Signed-off-by : Dale Curtis <dalecurtis@chromium.org>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/flac_picture.c

• torchaudio.io.StreamReader doesn't throw error when seeking to time stamp more than the duration of audio file

  27 mars 2023, par lokesh

  I am trying to get the audio chunk of audio file between specific start time and end time

  


  Consider a audio of duration 10 seconds. Now i need to get chunk from 4 sec to 7 sec

  


  torchaudio.info doesn't give correct num_frames for io.BytesIO flac audio file. So there is no way to find the total number of frames in the given audio to check for out of bounds start offset
Ref : https://github.com/pytorch/audio/issues/2524

  


  What I did to get the chunk of audio with start and end offsets.

  


  def read_audio(audio_file, start, end):
    audio_file.seek(0)
    reader = StreamReader(audio_file)
    sample_rate = int(reader.get_src_stream_info(reader.default_audio_stream).sample_rate)

    reader.seek(start * sample_rate)
    reader.add_basic_audio_stream(frames_per_chunk=(end - start) * sample_rate)

    return list(reader.stream())[0].pop()


  


  This is working as intended for start time less than the duration of audio file. But when we give the start time more than the duration of audio file, It doesn't throw error or return empty tensor.
Is there any way to know the given offsets are out of bounds.