Move responsibility for RTP header extensions on video receive.

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 109 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
+    // TODO(nisse): Configure receiver_call_ instead.
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
+#endif
   }
 
   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 153 matching lines @@
       RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   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