Move responsibility for RTP header extensions on video receive.

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

 /*
  *  Copyright (c) 2016 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 343 matching lines @@
     auto pred = [&config](uint32_t ssrc) {
       return ssrc == config->rtp.rtx_ssrc;
     };
     if (!std::all_of(rtx_ssrcs.cbegin(), rtx_ssrcs.cend(), pred)) {
       LOG(LS_WARNING) << "RtcEventLog protobuf contained different SSRCs for "
                          "different received RTX payload types. Will only use "
                          "rtx_ssrc = "
                       << config->rtp.rtx_ssrc << ".";
     }
   }
+#if 0
   // Get header extensions.
   GetHeaderExtensions(&config->rtp.extensions,
                       receiver_config.header_extensions());
+#endif
   // Get decoders.
   config->decoders.clear();
   for (int i = 0; i < receiver_config.decoders_size(); i++) {
     RTC_CHECK(receiver_config.decoders(i).has_name());
     RTC_CHECK(receiver_config.decoders(i).has_payload_type());
     VideoReceiveStream::Decoder decoder;
     decoder.payload_name = receiver_config.decoders(i).name();
     decoder.payload_type = receiver_config.decoders(i).payload_type();
     config->decoders.push_back(decoder);
   }
@@ skipping 206 matching lines @@
         rtc::Optional<ProbeFailureReason>(kInvalidSendReceiveRatio);
   } else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
     res.failure_reason = rtc::Optional<ProbeFailureReason>(kTimeout);
   } else {
     RTC_NOTREACHED();
   }
 
   return res;
 }
 }  // namespace webrtc