Move responsibility for RTP header extensions on video receive.

webrtc/logging/rtc_event_log/rtc_event_log_unittest.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.
  */
 
@@ skipping 153 matching lines @@
   config->rtp.remote_ssrc = prng->Rand<uint32_t>();
   config->rtp.local_ssrc = prng->Rand<uint32_t>();
   // Add extensions and settings for RTCP.
   config->rtp.rtcp_mode =
       prng->Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize;
   config->rtp.remb = prng->Rand<bool>();
   config->rtp.rtx_ssrc = prng->Rand<uint32_t>();
   // Add a map from a payload type to a new payload type for RTX.
   config->rtp.rtx_payload_types.insert(
       std::make_pair(prng->Rand(0, 127), prng->Rand(0, 127)));
+#if 0
   // Add header extensions.
   for (unsigned i = 0; i < kNumExtensions; i++) {
     if (extensions_bitvector & (1u << i)) {
       config->rtp.extensions.push_back(
           RtpExtension(kExtensionNames[i], prng->Rand<int>()));
     }
   }
+#endif
 }
 
 void GenerateVideoSendConfig(uint32_t extensions_bitvector,
                              VideoSendStream::Config* config,
                              Random* prng) {
   // Create a map from a payload type to an encoder name.
   config->encoder_settings.payload_type = prng->Rand(0, 127);
   config->encoder_settings.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
   // Add SSRCs for the stream.
   config->rtp.ssrcs.push_back(prng->Rand<uint32_t>());
@@ skipping 691 matching lines @@
   VideoSendConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
   AudioNetworkAdaptationReadWriteTest test;
   test.DoTest();
 }
 
 }  // namespace webrtc