Register header extensions in RtpRtcpObserver to avoid log spam.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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 <stdio.h>
@@ skipping 14 matching lines @@
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/test/layer_filtering_transport.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/video/video_quality_test.h"
 
 namespace webrtc {
 
-static const int kTransportSeqExtensionId =
-    VideoQualityTest::kAbsSendTimeExtensionId + 1;
 static const int kSendStatsPollingIntervalMs = 1000;
 static const int kPayloadTypeVP8 = 123;
 static const int kPayloadTypeVP9 = 124;
 
 class VideoAnalyzer : public PacketReceiver,
                       public Transport,
                       public VideoRenderer,
                       public VideoCaptureInput,
                       public EncodedFrameObserver,
                       public EncodingTimeObserver {
@@ skipping 574 matching lines @@
   send_config_.encoder_settings.encoder = encoder_.get();
   send_config_.encoder_settings.payload_name = params.common.codec;
   send_config_.encoder_settings.payload_type = payload_type;
 
   send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
   send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
   send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
 
   send_config_.rtp.extensions.clear();
   if (params.common.send_side_bwe) {
+    send_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTransportSequenceNumber,
+                     test::kTransportSequenceNumberExtensionId));
+  } else {
     send_config_.rtp.extensions.push_back(RtpExtension(
-        RtpExtension::kTransportSequenceNumber, kTransportSeqExtensionId));
-  } else {
-    send_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
+        RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId));
   }
 
   // Automatically fill out streams[0] with params.
   VideoStream* stream = &encoder_config_.streams[0];
   stream->width = params.common.width;
   stream->height = params.common.height;
   stream->min_bitrate_bps = params.common.min_bitrate_bps;
   stream->target_bitrate_bps = params.common.target_bitrate_bps;
   stream->max_bitrate_bps = params.common.max_bitrate_bps;
   stream->max_framerate = static_cast<int>(params.common.fps);
@@ skipping 211 matching lines @@
   send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc