Implement timing frames.

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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
-#include <string.h>
-
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 
 namespace webrtc {
 
 RtpFeedback* NullObjectRtpFeedback() {
   static NullRtpFeedback null_rtp_feedback;
   return &null_rtp_feedback;
@@ skipping 221 matching lines @@
   header->extension.videoRotation = kVideoRotation_0;
 
   // May not be present in packet.
   header->extension.playout_delay.min_ms = -1;
   header->extension.playout_delay.max_ms = -1;
 
   // May not be present in packet.
   header->extension.hasVideoContentType = false;
   header->extension.videoContentType = VideoContentType::UNSPECIFIED;
 
+  header->extension.hasVideoTiming = false;
+  header->extension.videoTiming = {0u, 0u, 0u, 0u, 0u, 0u, false};
+
   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 191 matching lines @@
           //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 
           if (ptr[0] <
               static_cast<uint8_t>(VideoContentType::TOTAL_CONTENT_TYPES)) {
             header->extension.hasVideoContentType = true;
             header->extension.videoContentType =
                 static_cast<VideoContentType>(ptr[0]);
           }
           break;
         }
+        case kRtpExtensionVideoTiming: {
+          if (len != VideoTimingExtension::kValueSizeBytes - 1) {
+            LOG(LS_WARNING) << "Incorrect video timing len: " << len;
+            return;
+          }
+          header->extension.hasVideoTiming = true;
+          VideoTimingExtension::Parse(rtc::MakeArrayView(ptr, len + 1),
+                                      &header->extension.videoTiming);
+          break;
+        }
         case kRtpExtensionRtpStreamId: {
           header->extension.stream_id.Set(rtc::MakeArrayView(ptr, len + 1));
           break;
         }
         case kRtpExtensionRepairedRtpStreamId: {
           header->extension.repaired_stream_id.Set(
               rtc::MakeArrayView(ptr, len + 1));
           break;
         }
         case kRtpExtensionNone:
         case kRtpExtensionNumberOfExtensions: {
           RTC_NOTREACHED() << "Invalid extension type: " << type;
           return;
         }
       }
     }
     ptr += (len + 1);
   }
 }
 
 }  // namespace RtpUtility
 }  // namespace webrtc