Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_utility.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 230 matching lines @@
 
   // May not be present in packet.
   header->extension.hasAudioLevel = false;
   header->extension.voiceActivity = false;
   header->extension.audioLevel = 0;
 
   // May not be present in packet.
   header->extension.hasVideoRotation = false;
   header->extension.videoRotation = 0;
 
+  // May not be present in packet.
+  header->extension.playout_delay.min_ms = -1;
+  header->extension.playout_delay.max_ms = -1;
+
   if (X) {
     /* RTP header extension, RFC 3550.
      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |      defined by profile       |           length              |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |                        header extension                       |
     |                             ....                              |
     */
@@ skipping 139 matching lines @@
           //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
           //  |  ID   | L=1   |transport wide sequence number |
           //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
           uint16_t sequence_number = ptr[0] << 8;
           sequence_number += ptr[1];
           header->extension.transportSequenceNumber = sequence_number;
           header->extension.hasTransportSequenceNumber = true;
           break;
         }
+        case kRtpExtensionPlayoutDelay: {
+          if (len != 2) {
+            LOG(LS_WARNING) << "Incorrect playout delay len: " << len;
+            return;
+          }
+          //   0                   1                   2                   3
+          //   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+          //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+          //  |  ID   | len=2 |   MIN delay           |   MAX delay           |
+          //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+          int min_playout_delay = (ptr[0] << 4) | ((ptr[1] >> 4) & 0xf);
+          int max_playout_delay = ((ptr[1] & 0xf) << 8) | ptr[2];
+          header->extension.playout_delay.min_ms =
+              min_playout_delay * kPlayoutDelayGranularityMs;
+          header->extension.playout_delay.max_ms =
+              max_playout_delay * kPlayoutDelayGranularityMs;
+          break;
+        }
         default: {
           LOG(LS_WARNING) << "Extension type not implemented: " << type;
           return;
         }
       }
     }
     ptr += (len + 1);
     uint8_t num_bytes = ParsePaddingBytes(ptrRTPDataExtensionEnd, ptr);
     ptr += num_bytes;
   }
 }
 
 uint8_t RtpHeaderParser::ParsePaddingBytes(
     const uint8_t* ptrRTPDataExtensionEnd,
     const uint8_t* ptr) const {
   uint8_t num_zero_bytes = 0;
   while (ptrRTPDataExtensionEnd - ptr > 0) {
     if (*ptr != 0) {
       return num_zero_bytes;
     }
     ptr++;
     num_zero_bytes++;
   }
   return num_zero_bytes;
 }
 }  // namespace RtpUtility
 }  // namespace webrtc