Move responsibility for RTP header extensions on video receive.

webrtc/video/rtp_stream_receiver.cc

 /*
  *  Copyright (c) 2012 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 110 matching lines @@
          "reserved for internal usage.";
   RTC_DCHECK(config_.rtp.remote_ssrc != 0);
   // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
   RTC_DCHECK(config_.rtp.local_ssrc != 0);
   RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
 
   rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
   rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
   rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
 
+#if 0
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     EnableReceiveRtpHeaderExtension(config_.rtp.extensions[i].uri,
                                     config_.rtp.extensions[i].id);
   }
-
+#endif
   static const int kMaxPacketAgeToNack = 450;
   const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0)
                                            ? kMaxPacketAgeToNack
                                            : kDefaultMaxReorderingThreshold;
   rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
 
   if (config_.rtp.rtx_ssrc) {
     rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc);
 
     for (const auto& kv : config_.rtp.rtx_payload_types) {
@@ skipping 470 matching lines @@
         "WebRTC.Video.ReceivedFecPacketsInPercent",
         static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
   }
   if (counter.num_fec_packets > 0) {
     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
                              static_cast<int>(counter.num_recovered_packets *
                                               100 / counter.num_fec_packets));
   }
 }
 
+#if 0
 void RtpStreamReceiver::EnableReceiveRtpHeaderExtension(
     const std::string& extension, int id) {
   // One-byte-extension local identifiers are in the range 1-14 inclusive.
   RTC_DCHECK_GE(id, 1);
   RTC_DCHECK_LE(id, 14);
   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
       StringToRtpExtensionType(extension), id));
 }
+#endif
 
 void RtpStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) {
   auto codec_params_it = pt_codec_params_.find(payload_type);
   if (codec_params_it == pt_codec_params_.end())
     return;
 
   LOG(LS_INFO) << "Found out of band supplied codec parameters for"
                << " payload type: " << static_cast<int>(payload_type);
 
   H264SpropParameterSets sprop_decoder;
   auto sprop_base64_it =
       codec_params_it->second.find(cricket::kH264FmtpSpropParameterSets);
 
   if (sprop_base64_it == codec_params_it->second.end())
     return;
 
   if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
     return;
 
   tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                              sprop_decoder.pps_nalu());
 }
 
 }  // namespace webrtc