Remove use of RtpHeaderExtension and clean up

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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 <functional>
@@ skipping 116 matching lines @@
     video_send_config_.encoder_settings.payload_type =
         kFakeVideoSendPayloadType;
     video_encoder_config_.streams = test::CreateVideoStreams(1);
 
     receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
     // receive_config_.decoders will be set by every stream separately.
     receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
     receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     receive_config_.rtp.remb = true;
     receive_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
+        RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
     receive_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
+        RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
   }
 
   virtual void TearDown() {
     std::for_each(streams_.begin(), streams_.end(),
                   std::mem_fun(&Stream::StopSending));
 
     send_transport_->StopSending();
     receive_transport_->StopSending();
 
     while (!streams_.empty()) {
@@ skipping 32 matching lines @@
       frame_generator_capturer_->Start();
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
         receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
         // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
         // the AudioReceiveStream. Every receive stream has to correspond to
         // an underlying channel id.
         receive_config.voe_channel_id = 0;
         receive_config.rtp.extensions.push_back(
-            RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
+            RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
         audio_receive_stream_ =
             test_->receiver_call_->CreateAudioReceiveStream(receive_config);
       } else {
         VideoReceiveStream::Decoder decoder;
         decoder.decoder = &fake_decoder_;
         decoder.payload_type =
             test_->video_send_config_.encoder_settings.payload_type;
         decoder.payload_name =
             test_->video_send_config_.encoder_settings.payload_name;
         test_->receive_config_.decoders.clear();
@@ skipping 55 matching lines @@
   std::vector<Stream*> streams_;
 };
 
 static const char* kAbsSendTimeLog =
     "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
 static const char* kSingleStreamLog =
     "RemoteBitrateEstimatorSingleStream: Instantiating.";
 
 TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
   video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
+      RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
   video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
+      RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
 TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
   video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
+      RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 
   video_send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
+      RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
 // This test is flaky. See webrtc:5790.
 TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
   video_send_config_.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
+      RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 
   video_send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
+      RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 
   video_send_config_.rtp.extensions[0] =
-      RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
+      RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc