Remove use of RtpHeaderExtension and clean up

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"
 
 #include <sstream>
 #include <string>
 
 namespace webrtc {
 std::string FecConfig::ToString() const {
   std::stringstream ss;
   ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
   ss << ", red_payload_type: " << red_payload_type;
   ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
   ss << '}';
   return ss.str();
 }
 
 std::string RtpExtension::ToString() const {
   std::stringstream ss;
-  ss << "{name: " << name;
+  ss << "{uri: " << uri;
   ss << ", id: " << id;
   ss << '}';
   return ss.str();
 }
 
-const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
-const char* RtpExtension::kAbsSendTime =
+const char* RtpExtension::kAudioLevelUri =
+    "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+const int RtpExtension::kAudioLevelDefaultId = 1;
+
+const char* RtpExtension::kTimestampOffsetUri =
+    "urn:ietf:params:rtp-hdrext:toffset";
+const int RtpExtension::kTimestampOffsetDefaultId = 2;
+
+const char* RtpExtension::kAbsSendTimeUri =
     "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
-const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
-const char* RtpExtension::kAudioLevel =
-    "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
-const char* RtpExtension::kTransportSequenceNumber =
+const int RtpExtension::kAbsSendTimeDefaultId = 3;
+
+const char* RtpExtension::kVideoRotationUri = "urn:3gpp:video-orientation";
+const int RtpExtension::kVideoRotationDefaultId = 4;
+
+const char* RtpExtension::kTransportSequenceNumberUri =
     "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
+const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
 
-bool RtpExtension::IsSupportedForAudio(const std::string& name) {
-  return name == webrtc::RtpExtension::kAbsSendTime ||
-         name == webrtc::RtpExtension::kAudioLevel ||
-         name == webrtc::RtpExtension::kTransportSequenceNumber;
+bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
+  return uri == webrtc::RtpExtension::kAbsSendTimeUri ||
+         uri == webrtc::RtpExtension::kAudioLevelUri ||
+         uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
 }
 
-bool RtpExtension::IsSupportedForVideo(const std::string& name) {
-  return name == webrtc::RtpExtension::kTOffset ||
-         name == webrtc::RtpExtension::kAbsSendTime ||
-         name == webrtc::RtpExtension::kVideoRotation ||
-         name == webrtc::RtpExtension::kTransportSequenceNumber;
+bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
+  return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
+         uri == webrtc::RtpExtension::kAbsSendTimeUri ||
+         uri == webrtc::RtpExtension::kVideoRotationUri ||
+         uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
 }
 
 VideoStream::VideoStream()
     : width(0),
       height(0),
       max_framerate(-1),
       min_bitrate_bps(-1),
       target_bitrate_bps(-1),
       max_bitrate_bps(-1),
       max_qp(-1) {}
@@ skipping 51 matching lines @@
   }
   ss << ", encoder_specific_settings: ";
   ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
 
   ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
   ss << '}';
   return ss.str();
 }
 
 }  // namespace webrtc