Implement FrameMarking header extension support

webrtc/config.cc

 /*
  *  Copyright (c) 2014 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/config.h"
@@ skipping 69 matching lines @@
     "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
 const int RtpExtension::kVideoContentTypeDefaultId = 7;
 
 const char RtpExtension::kVideoTimingUri[] =
     "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
 const int RtpExtension::kVideoTimingDefaultId = 8;
 
 const char RtpExtension::kEncryptHeaderExtensionsUri[] =
     "urn:ietf:params:rtp-hdrext:encrypt";
 
+// This extensions provides meta-information about the RTP streams outside the
+// encrypted media payload, an RTP switch can do codec-agnostic
+// selective forwarding without decrypting the payload
+const char* RtpExtension::kFrameMarkingUri =
+    "urn:ietf:params:rtp-hdrext:framemarking";
+const int RtpExtension::kFrameMarkingDefaultId = 9;
+
 const int RtpExtension::kMinId = 1;
 const int RtpExtension::kMaxId = 14;
 
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
   return uri == webrtc::RtpExtension::kAudioLevelUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
 }
 
 bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
   return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
          uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kVideoRotationUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
          uri == webrtc::RtpExtension::kPlayoutDelayUri ||
          uri == webrtc::RtpExtension::kVideoContentTypeUri ||
-         uri == webrtc::RtpExtension::kVideoTimingUri;
+         uri == webrtc::RtpExtension::kVideoTimingUri ||
+         uri == webrtc::RtpExtension::kFrameMarkingUri;
 }
 
 bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
   return uri == webrtc::RtpExtension::kAudioLevelUri ||
          uri == webrtc::RtpExtension::kTimestampOffsetUri ||
 #if !defined(ENABLE_EXTERNAL_AUTH)
          // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri"
          // here and filter out later if external auth is really used in
          // srtpfilter. External auth is used by Chromium and replaces the
          // extension header value of "kAbsSendTimeUri", so it must not be
@@ skipping 158 matching lines @@
 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings(
     const VideoCodecVP9& specifics)
     : specifics_(specifics) {}
 
 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
     VideoCodecVP9* vp9_settings) const {
   *vp9_settings = specifics_;
 }
 
 }  // namespace webrtc