Implement RTP keepalive in native stack.

webrtc/video/video_send_stream.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.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 39 matching lines @@
     RtcpIntraFrameObserver* intra_frame_callback,
     RtcpBandwidthObserver* bandwidth_callback,
     RtpTransportControllerSendInterface* transport,
     RtcpRttStats* rtt_stats,
     FlexfecSender* flexfec_sender,
     SendStatisticsProxy* stats_proxy,
     SendDelayStats* send_delay_stats,
     RtcEventLog* event_log,
     RateLimiter* retransmission_rate_limiter,
     OverheadObserver* overhead_observer,
-    size_t num_modules) {
+    size_t num_modules,
+    VideoSendStream::Config::Rtp::KeepAlive keepalive_config) {
   RTC_DCHECK_GT(num_modules, 0);
   RtpRtcp::Configuration configuration;
   ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
   configuration.audio = false;
   configuration.receiver_only = false;
   configuration.flexfec_sender = flexfec_sender;
   configuration.receive_statistics = null_receive_statistics;
   configuration.outgoing_transport = outgoing_transport;
   configuration.intra_frame_callback = intra_frame_callback;
   configuration.bandwidth_callback = bandwidth_callback;
   configuration.transport_feedback_callback =
       transport->transport_feedback_observer();
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer = stats_proxy;
   configuration.paced_sender = transport->packet_sender();
   configuration.transport_sequence_number_allocator =
       transport->packet_router();
   configuration.send_bitrate_observer = stats_proxy;
   configuration.send_frame_count_observer = stats_proxy;
   configuration.send_side_delay_observer = stats_proxy;
   configuration.send_packet_observer = send_delay_stats;
   configuration.event_log = event_log;
   configuration.retransmission_rate_limiter = retransmission_rate_limiter;
   configuration.overhead_observer = overhead_observer;
+  configuration.keepalive_config = keepalive_config;
   std::vector<RtpRtcp*> modules;
   for (size_t i = 0; i < num_modules; ++i) {
     RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
     rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
     modules.push_back(rtp_rtcp);
   }
   return modules;
 }
@@ skipping 693 matching lines @@
           &encoder_feedback_,
           bandwidth_observer_.get(),
           transport,
           call_stats_->rtcp_rtt_stats(),
           flexfec_sender_.get(),
           stats_proxy_,
           send_delay_stats,
           event_log,
           transport->send_side_cc()->GetRetransmissionRateLimiter(),
           this,
-          config_->rtp.ssrcs.size())),
+          config_->rtp.ssrcs.size(),
+          config_->rtp.keep_alive)),
       payload_router_(rtp_rtcp_modules_,
                       config_->encoder_settings.payload_type),
       weak_ptr_factory_(this),
       overhead_bytes_per_packet_(0),
       transport_overhead_bytes_per_packet_(0) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
   weak_ptr_ = weak_ptr_factory_.GetWeakPtr();
   module_process_thread_checker_.DetachFromThread();
 
@@ skipping 532 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc