Add sender controlled playout delay limits

This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.

The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.

There are no tests at this time and most of testing is done with chromium
webrtc prototype.

On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.

BUG=webrtc:5895

Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}

webrtc/modules/video_coding/jitter_buffer.h

@@ -152,9 +152,8 @@ class VCMJitterBuffer {
   bool CompleteSequenceWithNextFrame();
 
   // Wait |max_wait_time_ms| for a complete frame to arrive.
-  // The function returns true once such a frame is found, its corresponding
-  // timestamp is returned. Otherwise, returns false.
-  bool NextCompleteTimestamp(uint32_t max_wait_time_ms, uint32_t* timestamp);
+  // If found, a pointer to the frame is returned. Returns nullptr otherwise.
+  VCMEncodedFrame* NextCompleteFrame(uint32_t max_wait_time_ms);
 
   // Locates a frame for decoding (even an incomplete) without delay.
   // The function returns true once such a frame is found, its corresponding