Wire up RTP keep-alive in ortc api.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 193 matching lines @@
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
   void OnTransportOverheadChanged(MediaType media,
                                   int transport_overhead_per_packet) override;
 
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
 
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
+  bool SetRtpKeepAliveConfig(const RtpKeepAliveConfig& config) override;
 
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t bitrate_bps,
                         uint8_t fraction_loss,
                         int64_t rtt_ms,
                         int64_t probing_interval_ms) override;
 
   // Implements BitrateAllocator::LimitObserver.
   void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
                                  uint32_t max_padding_bitrate_bps) override;
@@ skipping 910 matching lines @@
 
   transport_send_->send_side_cc()->SignalNetworkState(aggregate_state);
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
                                         clock_->TimeInMilliseconds());
   transport_send_->send_side_cc()->OnSentPacket(sent_packet);
 }
 
+bool Call::SetRtpKeepAliveConfig(const RtpKeepAliveConfig& config) {
+  RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
+
+  ReadLockScoped lock(*send_crit_);
+  if (config != config_.keepalive_config &&
+      (!video_send_streams_.empty() || !audio_send_ssrcs_.empty())) {
+    LOG(LS_WARNING) << "RTP keep-alive settings cannot be altered after "
+                       "creating send streams.";
+    return false;
+  }
+  config_.keepalive_config = config;
+  return true;
+}
+
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt_ms,
                             int64_t probing_interval_ms) {
   // TODO(perkj): Consider making sure CongestionController operates on
   // |worker_queue_|.
   if (!worker_queue_.IsCurrent()) {
     worker_queue_.PostTask(
         [this, target_bitrate_bps, fraction_loss, rtt_ms, probing_interval_ms] {
           OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt_ms,
@@ skipping 254 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc