Register header extensions in RtpRtcpObserver to avoid log spam.

webrtc/video/video_send_stream_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>  // max
@@ skipping 145 matching lines @@
     void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for single RTP packet.";
     }
   } test;
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
-  static const uint8_t kTOffsetExtensionId = 13;
   static const int kEncodeDelayMs = 5;
   class TransmissionTimeOffsetObserver : public test::SendTest {
    public:
     TransmissionTimeOffsetObserver()
         : SendTest(kDefaultTimeoutMs),
           encoder_(Clock::GetRealTimeClock(), kEncodeDelayMs) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
-          kRtpExtensionTransmissionTimeOffset, kTOffsetExtensionId));
+          kRtpExtensionTransmissionTimeOffset, test::kTOffsetExtensionId));
     }
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       EXPECT_TRUE(header.extension.hasTransmissionTimeOffset);
       EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
       EXPECT_GT(header.extension.transmissionTimeOffset, 0);
       EXPECT_EQ(header.extension.absoluteSendTime, 0u);
       observation_complete_->Set();
 
       return SEND_PACKET;
     }
 
     void ModifyConfigs(VideoSendStream::Config* send_config,
                        std::vector<VideoReceiveStream::Config>* receive_configs,
                        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = &encoder_;
+      send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
+          RtpExtension(RtpExtension::kTOffset, test::kTOffsetExtensionId));
     }
 
     void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for a single RTP packet.";
     }
 
     test::DelayedEncoder encoder_;
   } test;
 
@@ skipping 21 matching lines @@
 
       observation_complete_->Set();
 
       return SEND_PACKET;
     }
 
     void ModifyConfigs(VideoSendStream::Config* send_config,
                        std::vector<VideoReceiveStream::Config>* receive_configs,
                        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = &encoder_;
+      send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
           RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
     }
 
     void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for a single RTP packet.";
     }
 
     test::FakeEncoder encoder_;
@@ skipping 1632 matching lines @@
       EXPECT_TRUE(vp9videoHeader->flexible_mode);
       observation_complete_->Set();
     }
 
   } test;
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 }  // namespace webrtc