Add RtcpRttStats to AudioStream

BUG=webrtc:6508

Review-Url: https://codereview.webrtc.org/2402333002
Cr-Commit-Position: refs/heads/master@{#14595}

webrtc/audio/audio_send_stream.cc

@@ -64,7 +64,8 @@ AudioSendStream::AudioSendStream(
     rtc::TaskQueue* worker_queue,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator,
-    RtcEventLog* event_log)
+    RtcEventLog* event_log,
+    RtcpRttStats* rtcp_rtt_stats)
     : worker_queue_(worker_queue),
       config_(config),
       audio_state_(audio_state),
@@ -77,6 +78,7 @@ AudioSendStream::AudioSendStream(
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log);
+  channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
   channel_proxy_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(),
       congestion_controller->GetTransportFeedbackObserver(),

webrtc/audio/audio_send_stream.h

@@ -23,6 +23,7 @@ namespace webrtc {
 class CongestionController;
 class VoiceEngine;
 class RtcEventLog;
+class RtcpRttStats;
 
 namespace voe {
 class ChannelProxy;
@@ -37,7 +38,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
                   rtc::TaskQueue* worker_queue,
                   CongestionController* congestion_controller,
                   BitrateAllocator* bitrate_allocator,
-                  RtcEventLog* event_log);
+                  RtcEventLog* event_log,
+                  RtcpRttStats* rtcp_rtt_stats);
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.

webrtc/audio/audio_send_stream_unittest.cc

@@ -20,6 +20,7 @@
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
@@ -109,6 +110,8 @@ struct ConfigHelper {
               .Times(1);
           EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
               .Times(1);  // Destructor resets the event log
+          EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(&rtcp_rtt_stats_))
+              .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
@@ -132,6 +135,7 @@ struct ConfigHelper {
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
+  RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; }
 
   void SetupMockForSendTelephoneEvent() {
     EXPECT_TRUE(channel_proxy_);
@@ -186,6 +190,7 @@ struct ConfigHelper {
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   CongestionController congestion_controller_;
   MockRtcEventLog event_log_;
+  MockRtcpRttStats rtcp_rtt_stats_;
   testing::NiceMock<MockLimitObserver> limit_observer_;
   BitrateAllocator bitrate_allocator_;
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
@@ -215,7 +220,7 @@ TEST(AudioSendStreamTest, ConstructDestruct) {
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
+      helper.event_log(), helper.rtcp_rtt_stats());
 }
 
 TEST(AudioSendStreamTest, SendTelephoneEvent) {
@@ -223,7 +228,7 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) {
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
+      helper.event_log(), helper.rtcp_rtt_stats());
   helper.SetupMockForSendTelephoneEvent();
   EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
       kTelephoneEventCode, kTelephoneEventDuration));
@@ -234,7 +239,7 @@ TEST(AudioSendStreamTest, SetMuted) {
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
+      helper.event_log(), helper.rtcp_rtt_stats());
   EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
   send_stream.SetMuted(true);
 }
@@ -244,7 +249,7 @@ TEST(AudioSendStreamTest, GetStats) {
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
+      helper.event_log(), helper.rtcp_rtt_stats());
   helper.SetupMockForGetStats();
   AudioSendStream::Stats stats = send_stream.GetStats();
   EXPECT_EQ(kSsrc, stats.local_ssrc);
@@ -274,7 +279,7 @@ TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
+      helper.event_log(), helper.rtcp_rtt_stats());
   helper.SetupMockForGetStats();
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 

webrtc/call/call.cc

@@ -373,7 +373,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
   event_log_->LogAudioSendStreamConfig(config);
   AudioSendStream* send_stream = new AudioSendStream(
       config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
-      bitrate_allocator_.get(), event_log_);
+      bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());
   {
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==

webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
+#define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockRtcpRttStats : public RtcpRttStats {
+ public:
+  MOCK_METHOD1(OnRttUpdate, void(int64_t rtt));
+
+  MOCK_CONST_METHOD0(LastProcessedRtt, int64_t());
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_

webrtc/test/mock_voe_channel_proxy.h

@@ -59,6 +59,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
                      const rtc::scoped_refptr<AudioDecoderFactory>&());
   MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
   MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
+  MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats));
   MOCK_METHOD1(SetBitrate, void(int bitrate_bps));
 };
 }  // namespace test

webrtc/voice_engine/channel.cc

@@ -157,6 +157,34 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog {
   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)
+      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) {
@@ -815,6 +843,7 @@ Channel::Channel(int32_t channelId,
     : _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(RTPPayloadStrategy::CreateStrategy(true))),
@@ -900,6 +929,7 @@ Channel::Channel(int32_t channelId,
     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();
 
@@ -2811,6 +2841,10 @@ 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);
+}
+
 int Channel::RegisterExternalMediaProcessing(ProcessingTypes type,
                                              VoEMediaProcess& processObject) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),

webrtc/voice_engine/channel.h

@@ -67,6 +67,7 @@ namespace voe {
 
 class OutputMixer;
 class RtcEventLogProxy;
+class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
 class Statistics;
 class StatisticsProxy;
@@ -417,6 +418,8 @@ class Channel
   // Set a RtcEventLog logging object.
   void SetRtcEventLog(RtcEventLog* event_log);
 
+  void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
+
  protected:
   void OnIncomingFractionLoss(int fraction_lost);
 
@@ -452,6 +455,7 @@ class Channel
   ChannelState channel_state_;
 
   std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_;
+  std::unique_ptr<voe::RtcpRttStatsProxy> rtcp_rtt_stats_proxy_;
 
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;

webrtc/voice_engine/channel_proxy.cc

@@ -214,6 +214,11 @@ void ChannelProxy::SetRtcEventLog(RtcEventLog* event_log) {
   channel()->SetRtcEventLog(event_log);
 }
 
+void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->SetRtcpRttStats(rtcp_rtt_stats);
+}
+
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();

webrtc/voice_engine/channel_proxy.h

@@ -25,6 +25,7 @@ namespace webrtc {
 class AudioSinkInterface;
 class PacketRouter;
 class RtcEventLog;
+class RtcpRttStats;
 class RtpPacketSender;
 class Transport;
 class TransportFeedbackObserver;
@@ -91,6 +92,8 @@ class ChannelProxy {
 
   virtual void SetRtcEventLog(RtcEventLog* event_log);
 
+  virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
+
  private:
   Channel* channel() const;