title: How to handle clock skew in audio streaming
body: <p>Problem : </p>
  
  <ol>
  <li>Its live audio-video streaming over WiFi + udp network.</li>
  <li>Stream : Mpeg2Ts </li>
  <li>Player Framework : gstreamer.</li>
  <li>Pipeline  Appsrc ----> tsdemuxer -----> audio-queue---->faad decoder--->alsasink
                        -         ----->video-queue->vpudecoder ---->videosink</li>
  <li>Audio device is configured for processing 48000 samples per second.</li>
  <li>Senders clock is faster than receiver clock  and i get this info by tracking the pcr value coming in the stream and receiver system clock. After 1 hour there is 8 second difference between sender and receiver clocks.</li>
  <li>So problem is sender is sending more samples in one seconds with respect to receiver clock due to this latency between sender and receiver is keep increasing with the time.</li>
  </ol>