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• Setting up audio queue for ffmpeg rtsp stream playing

  20 novembre 2011, par illew

  I'm working on an rtsp streaming(AAC format) client for iOS using ffmpeg. Right now I can only say my app is workable, but the streaming sound is very noisy and even a little distorted, far worse than when it's played by vlc or mplayer.

  The stream is read by av_read_frame(), decoded by avcodec_decode_audio3(). Then I just send the decoded raw audio to Audio Queue.

  When decoding a local aac file with my app, the sound seemed not so noisy at all. I know initial encoding would dramatically affect the result. However at least I should try to have it sounded like other streaming clients...

  Many parts in my implementation/modification actually came from try and error. I believe I'm doing something wrong in setting up Audio Queue, and the callback function for filling Audio Buffer.

  Any hints, suggestions or help are greatly appreciated.

  // —info of test materials dumped by av_dump_format() —

  Metadata:
  
      title           : /demo/test.3gp
  
    Duration: 00:00:30.11, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Audio: aac, 32000 Hz, stereo, s16
  
  aac  Advanced Audio Coding 

  // — the Audio Queue setup procedure —

  - (void) startPlayback
  
  {
  
      OSStatus err = 0;
  
      if(playState.playing) return;
  
  
  
      playState.started = false;
  
  
  
      if(!playState.queue) 
  
      {
  
  
  
          UInt32 bufferSize;
  
  
  
  
  
          playState.format.mSampleRate = _av->audio.sample_rate;
  
          playState.format.mFormatID = kAudioFormatLinearPCM;
  
          playState.format.mFormatFlags = kAudioFormatFlagsCanonical;
  
          playState.format.mChannelsPerFrame = _av->audio.channels_per_frame;
  
          playState.format.mBytesPerPacket = sizeof(AudioSampleType) *_av->audio.channels_per_frame;
  
          playState.format.mBytesPerFrame = sizeof(AudioSampleType) *_av->audio.channels_per_frame;
  
          playState.format.mBitsPerChannel = 8 * sizeof(AudioSampleType);
  
  
  
          playState.format.mFramesPerPacket = 1;        
  
          playState.format.mReserved = 0;
  
  
  
  
  
          pauseStart = 0;
  
          DeriveBufferSize(playState.format,playState.format.mBytesPerPacket,BUFFER_DURATION,&bufferSize,&numPacketsToRead);
  
          err= AudioQueueNewOutput(&playState.format, aqCallback, &playState, NULL, kCFRunLoopCommonModes, 0, &playState.queue);
  
  
  
          if(err != 0)
  
          {
  
              printf("AQHandler.m startPlayback: Error creating new AudioQueue: %d \n", (int)err);
  
          }
  
  
  
          for(int i = 0 ; i < NUM_BUFFERS ; i ++)
  
          {
  
              err = AudioQueueAllocateBufferWithPacketDescriptions(playState.queue, bufferSize, numPacketsToRead , &playState.buffers[i]);
  
  
  
              if(err != 0)
  
                  printf("AQHandler.m startPlayback: Error allocating buffer %d", i);
  
              fillAudioBuffer(&playState,playState.queue, playState.buffers[i]);
  
          }
  
  
  
      }
  
  
  
      startTime = mu_currentTimeInMicros();
  
  
  
      err=AudioQueueStart(playState.queue, NULL);
  
  
  
      if(err)
  
      {
  
  
  
          char sErr[4];
  
          printf("AQHandler.m startPlayback: Could not start queue %ld %s.", err, FormatError(sErr,err));
  
  
  
          playState.playing = NO;
  
      } 
  
      else
  
      {
  
          AudioSessionSetActive(true);
  
          playState.playing = YES;
  
      }           
  
  }

  // — callback for filling audio buffer —

  static int ct = 0;
  
  static void fillAudioBuffer(void *info,AudioQueueRef queue, AudioQueueBufferRef buffer)
  
  {
  
  
  
      int lengthCopied = INT32_MAX;
  
      int dts= 0;
  
      int isDone = 0;
  
  
  
      buffer->mAudioDataByteSize = 0;
  
      buffer->mPacketDescriptionCount = 0;
  
  
  
      OSStatus err = 0;
  
      AudioTimeStamp bufferStartTime;
  
  
  
      AudioQueueGetCurrentTime(queue, NULL, &bufferStartTime, NULL);
  
  
  
  
  
      PlayState *ps = (PlayState *)info;
  
  
  
      if (!ps->started)
  
          ps->started = true;
  
  
  
      while(buffer->mPacketDescriptionCount < numPacketsToRead && lengthCopied > 0)
  
      {
  
          lengthCopied = getNextAudio(_av,
  
                          buffer->mAudioDataBytesCapacity-buffer->mAudioDataByteSize,
  
                          (uint8_t*)buffer->mAudioData+buffeg->mAudioDataByteSize,
  
                          &dts,&isDone);
  
  
  
          ct+= lengthCopied;
  
  
  
          if(lengthCopied < 0 || isDone) 
  
          {
  
              printf("nothing to read....\n\n");
  
              PlayState *ps = (PlayState *)info;
  
              ps->finished = true;
  
              ps->started = false;
  
              break;
  
          }
  
  
  
          if(aqStartDts < 0) aqStartDts = dts;
  
  
  
          if(buffer->mPacketDescriptionCount ==0)
  
          {
  
              bufferStartTime.mFlags = kAudioTimeStampSampleTimeValid;
  
              bufferStartTime.mSampleTime = (Float64)(dts-aqStartDts);//* _av->audio.frame_size;
  
  
  
              if (bufferStartTime.mSampleTime <0 ) 
  
                  bufferStartTime.mSampleTime = 0;
  
  
  
              printf("AQHandler.m fillAudioBuffer: DTS for %x: %lf time base: %lf StartDTS: %d\n", 
  
                      (unsigned int)buffer, 
  
                      bufferStartTime.mSampleTime, 
  
                      _av->audio.time_base, 
  
                      aqStartDts);
  
  
  
          }
  
  
  
          buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mStartOffset = buffer->mAudioDataByteSize;
  
          buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mDataByteSize = lengthCopied;
  
  
  
  
  
  
  
          buffer->mPacketDescriptions[buffer->mPacketDescriptionCount].mVariableFramesInPacket = 0;
  
  
  
          buffer->mPacketDescriptionCount++;
  
  
  
          buffer->mAudioDataByteSize += lengthCopied;
  
      }
  
  
  
      int audioBufferCount, audioBufferTotal,  videoBufferCount, videoBufferTotal;
  
      bufferCheck(_av,&videoBufferCount, &videoBufferTotal, &audioBufferCount, &audioBufferTotal);
  
  
  
      if(buffer->mAudioDataByteSize)
  
      {
  
  
  
          err = AudioQueueEnqueueBufferWithParameters(queue, buffer, 0, NULL, 0, 0, 0, NULL, &bufferStartTime, NULL);
  
  
  
          if(err)
  
          {
  
              char sErr[10];
  
              printf("AQHandler.m fillAudioBuffer: Could not enqueue buffer 0x%x: %d %s.", buffer, err, FormatError(sErr, err));
  
  
  
          }
  
  
  
      }
  
  
  
  }
  
  
  
  
  
  
  
  
  
  int getNextAudio(video_data_t* vInst, int maxlength, uint8_t* buf, int* pts, int* isDone) 
  
  {
  
  
  
      struct video_context_t  *ctx = vInst->context;
  
      int    datalength            = 0;
  
  
  
      while(ctx->audio_ring.lock || (ctx->audio_ring.count <= 0 && ((ctx->play_state & STATE_DIE) != STATE_DIE)))
  
      {
  
  
  
          if (ctx->play_state & STATE_EOF) return -1;        
  
          usleep(100);
  
      }
  
  
  
      *pts = 0;
  
      ctx->audio_ring.lock = kLocked;
  
  
  
      if(ctx->audio_ring.count>0 && maxlength > ctx->audio_buffer[ctx->audio_ring.read].size)
  
      {    
  
          memcpy(buf, ctx->audio_buffer[ctx->audio_ring.read].data,ctx->audio_buffer[ctx->audio_ring.read].size);
  
  
  
          *pts = ctx->audio_buffer[ctx->audio_ring.read].pts;
  
  
  
          datalength = ctx->audio_buffer[ctx->audio_ring.read].size;
  
  
  
          ctx->audio_ring.read++;        
  
          ctx->audio_ring.read %= ABUF_SIZE;        
  
          ctx->audio_ring.count--;
  
  
  
      }
  
      ctx->audio_ring.lock = kUnlocked;
  
  
  
      if((ctx->play_state & STATE_EOF) == STATE_EOF && ctx->audio_ring.count == 0) *isDone = 1;
  
  
  
      return datalength;
  
  }

• ffmpeg filename output format

  10 janvier 2017, par Yuval Adam

  When using ffmpeg to output a series of frames as images, the only format I can find documentation for is frame_%d.jpg. The %d identifier is replaced by the sequential frame count.

  Are there other parameters I can use in the output file format ? Specifically, I need the ability to add the timestamp of the specific frame.

• How can i know the duration of any h264 file ?

  26 septembre 2011, par Mr.32

  I have one file in which only h264 frames are there in form of NAL unit.
  So now is there any method so i can count the duration of that file ?

  I dont know how many frames are there in file.
  I have only file size.

  PS : All i want to do is in C language and on Linux platform.