Description:

Our first goal is to implement an adaptive playout mechanism for the receiver end of the telephony system. This will involve testing different versions of playout algorithms. Secondly, we want to collect data on different forward error correction schemes in order to fine-tune our data exchange system. Ideally, we would like an adaptive FEC scheme, which requires feedback from the receiver to the sender.

Pseudocode:

EstablishConnection(port, input, output, destination)
{
  attempt_UDP_connection(destination, port);
  while (unsuccessful) 
  {
     wait_for_connection(port);
     if !connection_from(destination)
	reject;
  }
  // This is the thread version (see below). 
  spawn(encode_and_send_process);   // Yu and Vincent handle this
  spawn(receive_and_playout_process);
}

receive_and_playout_process()
{
  while(event)
  {
    if (event == incoming_packet)
      decode_packet;
      determine_location_in_buffer(sample); // uses adaptive playout algorithm
      store_sample_in_buffer;	
      decode_FEC_information;
      for each FEC sample
        if (buffer_has_higher_quality_version)
          discard_sample;
        else
          store_sample_in_buffer;
    if (event == time_to_play_sample)
      if (sample_in_buffer)
        play_sample;
      else
        take_action_determined_by_adaptive_playout_algorithm;
  }
}

SetParameter(parameter, value)
{
  update_paramater(value);
}

Interface provided to other teams:

• EstablishConnection(port, input, output, destination)
  • port : port number
  • input : input source for packets sent on this connection
  • output : packets sent on this connection go to this output
  • destination : remote host
  • If A calls this function with port p and destination B, then B must also call this function with port p and destination A. The function is blocking and will either spawn a new thread to handle the data exchange, or require that an update function be called periodically.
• SetParameter(parameter, value)
  • parameter : packet timelength, base packet encoding quality, spike detection parameters

Interface needed from other teams:

Schedule:

• 9/28-10/4 : Resolve behavior of EstablishConnection by consultation with other components in our group.
• 10/5-10/25 : (Three weeks) Write basic code for forward error correction and adaptive playout algorithms. Confer with the other group in component 1 to ensure that the packet format we're using can support adaptive FEC.
• 10/26-11/22 : (Four weeks) Network measurements : round-trip packet times, packet losses, etc. Determine how to measure quality of playback, so we can analyze the results of different approaches. Verify the assumptions in the adaptive playout algorithm that characterize internet traffic (are losses bursty? are spikes the biggest concern for data loss?), or modify the algorithm to handle the characterization we find.
• 11/23-12/4: (Two weeks) Either implement an adaptive FEC scheme, or allow selection of an appropriate scheme based on the user's description of the current network state.

Needed from the course staff:

• Reflectors set up to allow measurements over long-distance paths.

References:

1. Meeting with S. Keshav on goals for this component.
2. Ramachandran Ramjee, Jim Kurose, Don Towsley, Henning Schulzrinne, "Adaptive Playout Mechanisms for Packetized Audio Applications in Wide-Area Networks", Proceedings of IEEE INFOCOMM '94, April 1994.
3. Sue B. Moon, Jim Kurose, and Don Towsley, "Packet Audio Playout Delay Adjustment: Performance Bounds and Algorithms", to appear on ACM/Springer Multimedia Systems.
4. Vicky Hardman, Martina Angela Sasse, Mark Handley, Anna Watson, "Reliable Audio for Use over the Internet", Proceedings of INET '95.
5. Matthew Podolsky, Cynthia Romer, Steven MaCanna, "Simulation of FEC-Based Error Control for Packet Audio on the Internet", Proceedings of IEEE INFOCOMM '98.
6. Jean-C. Bolot, Andres Vega-Garcia, "The Case for FEC-Based Error Control for Packet Audio in the Internet", ACM Multimedia Systems, 1997.
7. Jean-C. Bolot, Andres Vega-Garcia, "Control Mechanisms for Packet Audio in the Internet", Proceedings of IEEE INFOCOMM '96.