Move responsibility for RTP header extensions on video receive.

webrtc/video/end_to_end_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 <algorithm>
@@ skipping 1776 matching lines @@
           send_config->rtp.ssrcs[0];
 
       if (stream_index == 0)
         first_media_ssrc_ = send_config->rtp.ssrcs[0];
     }
 
     void UpdateReceiveConfig(
         size_t stream_index,
         VideoReceiveStream::Config* receive_config) override {
       receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+#if 0
       receive_config->rtp.extensions.clear();
       receive_config->rtp.extensions.push_back(RtpExtension(
           RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+#endif
       receive_config->renderer = &fake_renderer_;
     }
 
     test::DirectTransport* CreateSendTransport(Call* sender_call) override {
       std::map<uint8_t, MediaType> payload_type_map =
           MultiStreamTest::payload_type_map_;
       RTC_DCHECK(payload_type_map.find(kSendRtxPayloadType) ==
                  payload_type_map.end());
       payload_type_map[kSendRtxPayloadType] = MediaType::VIDEO;
       observer_ =
@@ skipping 2468 matching lines @@
       << "Reduced-size RTCP require rtcp-rsize to be negotiated.";
   EXPECT_FALSE(default_receive_config.rtp.remb)
       << "REMB require rtcp-fb: goog-remb to be negotiated.";
   EXPECT_FALSE(
       default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
       << "RTCP XR settings require rtcp-xr to be negotiated.";
   EXPECT_EQ(0U, default_receive_config.rtp.rtx_ssrc)
       << "Enabling RTX requires ssrc-group: FID negotiation";
   EXPECT_TRUE(default_receive_config.rtp.rtx_payload_types.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
+#if 0
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
-
+#endif
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyUlpfecConfig(default_receive_config.rtp.ulpfec);
 }
 
 TEST_F(EndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) {
   test::NullTransport rtcp_send_transport;
   FlexfecReceiveStream::Config default_receive_config(&rtcp_send_transport);
   EXPECT_EQ(-1, default_receive_config.payload_type)
       << "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
   EXPECT_EQ(0U, default_receive_config.remote_ssrc)
@@ skipping 169 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc