Allow applications to limit the ICE check rate through RTCConfiguration

webrtc/pc/peerconnection.cc

 /*
  *  Copyright 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 487 matching lines @@
     bool enable_rtp_data_channel;
     bool enable_quic;
     rtc::Optional<int> screencast_min_bitrate;
     rtc::Optional<bool> combined_audio_video_bwe;
     rtc::Optional<bool> enable_dtls_srtp;
     int ice_candidate_pool_size;
     bool prune_turn_ports;
     bool presume_writable_when_fully_relayed;
     bool enable_ice_renomination;
     bool redetermine_role_on_ice_restart;
+    rtc::Optional<int> ice_check_min_interval;

[Taylor Brandstetter, 2017/02/02 07:03:55]

Out of curiosity, did my scheme here work, or did you remember to update
operator== by yourself?

[skvlad, 2017/02/02 08:51:25]

Peter did this change, I'll let him answer :)

   };
   static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
                 "Did you add something to RTCConfiguration and forget to "
                 "update operator==?");
   return type == o.type && servers == o.servers &&
          bundle_policy == o.bundle_policy &&
          rtcp_mux_policy == o.rtcp_mux_policy &&
          tcp_candidate_policy == o.tcp_candidate_policy &&
          candidate_network_policy == o.candidate_network_policy &&
          audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
@@ skipping 11 matching lines @@
          enable_rtp_data_channel == o.enable_rtp_data_channel &&
          enable_quic == o.enable_quic &&
          screencast_min_bitrate == o.screencast_min_bitrate &&
          combined_audio_video_bwe == o.combined_audio_video_bwe &&
          enable_dtls_srtp == o.enable_dtls_srtp &&
          ice_candidate_pool_size == o.ice_candidate_pool_size &&
          prune_turn_ports == o.prune_turn_ports &&
          presume_writable_when_fully_relayed ==
              o.presume_writable_when_fully_relayed &&
          enable_ice_renomination == o.enable_ice_renomination &&
-         redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart;
+         redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
+         ice_check_min_interval == o.ice_check_min_interval;
 }
 
 bool PeerConnectionInterface::RTCConfiguration::operator!=(
     const PeerConnectionInterface::RTCConfiguration& o) const {
   return !(*this == o);
 }
 
 // Generate a RTCP CNAME when a PeerConnection is created.
 std::string GenerateRtcpCname() {
   std::string cname;
@@ skipping 2011 matching lines @@
 
 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
                                         int64_t max_size_bytes) {
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   event_log_->StopLogging();
 }
 }  // namespace webrtc