Reland "Update rtt on audio only calls".

webrtc/voice_engine/channel.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 132 matching lines @@
     rtc::CritScope lock(&crit_);
     event_log_ = event_log;
   }
 
  private:
   rtc::CriticalSection crit_;
   RtcEventLog* event_log_ GUARDED_BY(crit_);
   RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy);
 };
 
+class RtcpRttStatsProxy final : public RtcpRttStats {
+ public:
+  RtcpRttStatsProxy() : rtcp_rtt_stats_(nullptr) {}
+
+  void OnRttUpdate(int64_t rtt) override {
+    rtc::CritScope lock(&crit_);
+    if (rtcp_rtt_stats_)
+      rtcp_rtt_stats_->OnRttUpdate(rtt);
+  }
+
+  int64_t LastProcessedRtt() const override {
+    rtc::CritScope lock(&crit_);
+    if (rtcp_rtt_stats_ == nullptr)

[the sun, 2016/11/28 13:20:58]

nit: !rtcp_rtt_stats_

[michaelt, 2016/11/28 13:47:35]

Done.

+      return 0;
+    return rtcp_rtt_stats_->LastProcessedRtt();
+  }
+
+  void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
+    rtc::CritScope lock(&crit_);
+    rtcp_rtt_stats_ = rtcp_rtt_stats;
+  }
+
+ private:
+  rtc::CriticalSection crit_;
+  RtcpRttStats* rtcp_rtt_stats_ GUARDED_BY(crit_);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtcpRttStatsProxy);
+};
+
 class TransportFeedbackProxy : public TransportFeedbackObserver {
  public:
   TransportFeedbackProxy() : feedback_observer_(nullptr) {
     pacer_thread_.DetachFromThread();
     network_thread_.DetachFromThread();
   }
 
   void SetTransportFeedbackObserver(
       TransportFeedbackObserver* feedback_observer) {
     RTC_DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 663 matching lines @@
                "Channel::RecordFileEnded() => output file recorder module is"
                " shutdown");
 }
 
 Channel::Channel(int32_t channelId,
                  uint32_t instanceId,
                  const VoEBase::ChannelConfig& config)
     : _instanceId(instanceId),
       _channelId(channelId),
       event_log_proxy_(new RtcEventLogProxy()),
+      rtcp_rtt_stats_proxy_(new RtcpRttStatsProxy()),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(new RTPPayloadRegistry()),
       rtp_receive_statistics_(
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
       rtp_receiver_(
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
                                            this,
                                            this,
                                            rtp_payload_registry_.get())),
       telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
@@ skipping 63 matching lines @@
   configuration.outgoing_transport = this;
   configuration.receive_statistics = rtp_receive_statistics_.get();
   configuration.bandwidth_callback = rtcp_observer_.get();
   if (pacing_enabled_) {
     configuration.paced_sender = rtp_packet_sender_proxy_.get();
     configuration.transport_sequence_number_allocator =
         seq_num_allocator_proxy_.get();
     configuration.transport_feedback_callback = feedback_observer_proxy_.get();
   }
   configuration.event_log = &(*event_log_proxy_);
+  configuration.rtt_stats = &(*rtcp_rtt_stats_proxy_);
   configuration.retransmission_rate_limiter =
       retransmission_rate_limiter_.get();
 
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
   _rtpRtcpModule->SetSendingMediaStatus(false);
 
   statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC()));
   rtp_receive_statistics_->RegisterRtcpStatisticsCallback(
       statistics_proxy_.get());
 }
@@ skipping 1889 matching lines @@
     // Channel ID, disassociate with it.
     ChannelOwner ref(NULL);
     associate_send_channel_ = ref;
   }
 }
 
 void Channel::SetRtcEventLog(RtcEventLog* event_log) {
   event_log_proxy_->SetEventLog(event_log);
 }
 
+void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
+  rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
+}
+
 void Channel::SetTransportOverhead(int transport_overhead_per_packet) {
   _rtpRtcpModule->SetTransportOverhead(transport_overhead_per_packet);
 }
 
 int Channel::RegisterExternalMediaProcessing(ProcessingTypes type,
                                              VoEMediaProcess& processObject) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterExternalMediaProcessing()");
 
   rtc::CritScope cs(&_callbackCritSect);
@@ skipping 389 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc