Move responsibility for RTP header extensions on video receive.

call/rampup_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.
  */
 
@@ skipping 178 matching lines @@
     if (rtx_) {
       send_config->rtp.ulpfec.red_rtx_payload_type =
           test::CallTest::kRtxRedPayloadType;
     }
   }
 
   size_t i = 0;
   for (VideoReceiveStream::Config& recv_config : *receive_configs) {
     recv_config.rtp.remb = remb;
     recv_config.rtp.transport_cc = transport_cc;
+#if 0
+    // TODO(nisse): Configured for Call instead
     recv_config.rtp.extensions = send_config->rtp.extensions;
-
+#endif
     recv_config.rtp.remote_ssrc = video_ssrcs_[i];
     recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms;
 
     if (red_) {
       recv_config.rtp.red_payload_type =
           send_config->rtp.ulpfec.red_payload_type;
       recv_config.rtp.ulpfec_payload_type =
           send_config->rtp.ulpfec.ulpfec_payload_type;
       if (rtx_) {
         recv_config.rtp.rtx_associated_payload_types
@@ skipping 54 matching lines @@
     std::vector<FlexfecReceiveStream::Config>* receive_configs) {
   if (num_flexfec_streams_ == 0)
     return;
   RTC_DCHECK_EQ(1, num_flexfec_streams_);
   (*receive_configs)[0].payload_type = test::CallTest::kFlexfecPayloadType;
   (*receive_configs)[0].remote_ssrc = test::CallTest::kFlexfecSendSsrc;
   (*receive_configs)[0].protected_media_ssrcs = {video_ssrcs_[0]};
   (*receive_configs)[0].local_ssrc = video_ssrcs_[0];
   if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
     (*receive_configs)[0].transport_cc = false;
+#if 0
     (*receive_configs)[0].rtp_header_extensions.push_back(
         RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
+#endif
   } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) {
     (*receive_configs)[0].transport_cc = true;
+#if 0
     (*receive_configs)[0].rtp_header_extensions.push_back(RtpExtension(
         extension_type_.c_str(), kTransportSequenceNumberExtensionId));
+#endif
   }
 }
 
 void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) {
   sender_call_ = sender_call;
 }
 
 void RampUpTester::BitrateStatsPollingThread(void* obj) {
   static_cast<RampUpTester*>(obj)->PollStats();
 }
@@ skipping 353 matching lines @@
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, AudioTransportSequenceNumber) {
   RampUpTester test(0, 1, 0, 300000, 10000,
                     RtpExtension::kTransportSequenceNumberUri, false, false,
                     false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc