Move responsibility for RTP header extensions on video receive.

webrtc/call/bitrate_estimator_tests.cc

Index: webrtc/call/bitrate_estimator_tests.cc
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index ef7aba83e3f12023a1b0bba737b61923e6dc439c..3c921c41d42de15cadd86a30f9243186c9292f93 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -127,10 +127,13 @@ class BitrateEstimatorTest : public test::CallTest {
     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() {