Move responsibility for RTP header extensions on video receive.

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 110 matching lines @@
       video_send_config_.encoder_settings.payload_name = "FAKE";
       video_send_config_.encoder_settings.payload_type =
           kFakeVideoSendPayloadType;
       test::FillEncoderConfiguration(1, &video_encoder_config_);
 
       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;
+#if 0
       receive_config_.rtp.extensions.push_back(
           RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
       receive_config_.rtp.extensions.push_back(
           RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
+#endif
     });
   }
 
   virtual void TearDown() {
     task_queue_.SendTask([this]() {
       std::for_each(streams_.begin(), streams_.end(),
                     std::mem_fun(&Stream::StopSending));
 
       while (!streams_.empty()) {
         delete streams_.back();
@@ skipping 170 matching lines @@
     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
     receiver_log_.PushExpectedLogLine(
         "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
     streams_.push_back(new Stream(this));
     streams_[0]->StopSending();
     streams_[1]->StopSending();
   });
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc