Remove unnecessary proxy+lock code around RtcpRttStats pointer

Change-Id: I9c7fdc695be1e424488fa46962d459c66cf4d1e7
Bug: webrtc:9068
Reviewed-on: https://webrtc-review.googlesource.com/64721
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22603}

8 files changed, 17 insertions, 59 deletions

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 00b745bdd5..352261e248 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -73,7 +73,7 @@ std::unique_ptr<voe::ChannelProxy> CreateChannelAndProxy(
   return std::unique_ptr<voe::ChannelProxy>(
       new voe::ChannelProxy(std::unique_ptr<voe::Channel>(new voe::Channel(
           module_process_thread, internal_audio_state->audio_device_module(),
-          config.jitter_buffer_max_packets,
+          nullptr /* RtcpRttStats */, config.jitter_buffer_max_packets,
           config.jitter_buffer_fast_accelerate, config.decoder_factory,
           config.codec_pair_id))));
 }
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 71a4319fd6..9cf73459ea 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -46,15 +46,15 @@ void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy,
 std::unique_ptr<voe::ChannelProxy> CreateChannelAndProxy(
     webrtc::AudioState* audio_state,
     rtc::TaskQueue* worker_queue,
-    ProcessThread* module_process_thread) {
+    ProcessThread* module_process_thread,
+    RtcpRttStats* rtcp_rtt_stats) {
   RTC_DCHECK(audio_state);
   internal::AudioState* internal_audio_state =
       static_cast<internal::AudioState*>(audio_state);
-  return std::unique_ptr<voe::ChannelProxy>(new voe::ChannelProxy(
-      std::unique_ptr<voe::Channel>(new voe::Channel(
-          worker_queue,
-          module_process_thread,
-          internal_audio_state->audio_device_module()))));
+  return std::unique_ptr<voe::ChannelProxy>(
+      new voe::ChannelProxy(std::unique_ptr<voe::Channel>(new voe::Channel(
+          worker_queue, module_process_thread,
+          internal_audio_state->audio_device_module(), rtcp_rtt_stats))));
 }
 }  // namespace
 
@@ -103,7 +103,8 @@ AudioSendStream::AudioSendStream(
                       overall_call_lifetime,
                       CreateChannelAndProxy(audio_state.get(),
                                             worker_queue,
-                                            module_process_thread)) {}
+                                            module_process_thread,
+                                            rtcp_rtt_stats)) {}
 
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
@@ -138,7 +139,6 @@ AudioSendStream::AudioSendStream(
   RTC_DCHECK(overall_call_lifetime_);
 
   channel_proxy_->SetRtcEventLog(event_log_);
-  channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
   channel_proxy_->SetRTCPStatus(true);
   RtpReceiver* rtpReceiver = nullptr;  // Unused, but required for call.
   channel_proxy_->GetRtpRtcp(&rtp_rtcp_module_, &rtpReceiver);
@@ -159,7 +159,6 @@ AudioSendStream::~AudioSendStream() {
   channel_proxy_->RegisterTransport(nullptr);
   channel_proxy_->ResetSenderCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
-  channel_proxy_->SetRtcpRttStats(nullptr);
   // Lifetime can only be updated after deregistering
   // |timed_send_transport_adapter_| in the underlying channel object to avoid
   // data races in |active_lifetime_|.
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 4d18c52030..8fb7e7e197 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -228,9 +228,6 @@ struct ConfigHelper {
     EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).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);
-    EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(testing::IsNull()))
-        .Times(1);  // Destructor resets the rtt stats.
   }
 
   void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
diff --git a/audio/channel.cc b/audio/channel.cc
index fd2c281530..8dbe12ed04 100644
--- a/audio/channel.cc
+++ b/audio/channel.cc
@@ -94,34 +94,6 @@ 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_)
-      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_ RTC_GUARDED_BY(crit_);
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcpRttStatsProxy);
-};
-
 class TransportFeedbackProxy : public TransportFeedbackObserver {
  public:
   TransportFeedbackProxy() : feedback_observer_(nullptr) {
@@ -539,9 +511,11 @@ int Channel::PreferredSampleRate() const {
 
 Channel::Channel(rtc::TaskQueue* encoder_queue,
                  ProcessThread* module_process_thread,
-                 AudioDeviceModule* audio_device_module)
+                 AudioDeviceModule* audio_device_module,
+                 RtcpRttStats* rtcp_rtt_stats)
     : Channel(module_process_thread,
               audio_device_module,
+              rtcp_rtt_stats,
               0,
               false,
               rtc::scoped_refptr<AudioDecoderFactory>(),
@@ -552,12 +526,12 @@ Channel::Channel(rtc::TaskQueue* encoder_queue,
 
 Channel::Channel(ProcessThread* module_process_thread,
                  AudioDeviceModule* audio_device_module,
+                 RtcpRttStats* rtcp_rtt_stats,
                  size_t jitter_buffer_max_packets,
                  bool jitter_buffer_fast_playout,
                  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
                  rtc::Optional<AudioCodecPairId> codec_pair_id)
     : event_log_proxy_(new RtcEventLogProxy()),
-      rtcp_rtt_stats_proxy_(new RtcpRttStatsProxy()),
       rtp_payload_registry_(new RTPPayloadRegistry()),
       rtp_receive_statistics_(
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
@@ -620,7 +594,7 @@ Channel::Channel(ProcessThread* module_process_thread,
     configuration.transport_feedback_callback = feedback_observer_proxy_.get();
   }
   configuration.event_log = &(*event_log_proxy_);
-  configuration.rtt_stats = &(*rtcp_rtt_stats_proxy_);
+  configuration.rtt_stats = rtcp_rtt_stats;
   configuration.retransmission_rate_limiter =
       retransmission_rate_limiter_.get();
 
@@ -1313,10 +1287,6 @@ 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::UpdateOverheadForEncoder() {
   size_t overhead_per_packet =
       transport_overhead_per_packet_ + rtp_overhead_per_packet_;
diff --git a/audio/channel.h b/audio/channel.h
index fa4539d83d..532ad156ed 100644
--- a/audio/channel.h
+++ b/audio/channel.h
@@ -90,7 +90,6 @@ struct ReportBlock {
 namespace voe {
 
 class RtcEventLogProxy;
-class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
 class TransportFeedbackProxy;
 class TransportSequenceNumberProxy;
@@ -151,10 +150,12 @@ class Channel
   // Used for send streams.
   Channel(rtc::TaskQueue* encoder_queue,
           ProcessThread* module_process_thread,
-          AudioDeviceModule* audio_device_module);
+          AudioDeviceModule* audio_device_module,
+          RtcpRttStats* rtcp_rtt_stats);
   // Used for receive streams.
   Channel(ProcessThread* module_process_thread,
           AudioDeviceModule* audio_device_module,
+          RtcpRttStats* rtcp_rtt_stats,
           size_t jitter_buffer_max_packets,
           bool jitter_buffer_fast_playout,
           rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
@@ -285,7 +286,6 @@ class Channel
   // Set a RtcEventLog logging object.
   void SetRtcEventLog(RtcEventLog* event_log);
 
-  void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
   void SetTransportOverhead(size_t transport_overhead_per_packet);
 
   // From OverheadObserver in the RTP/RTCP module
@@ -341,7 +341,6 @@ 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<RTPPayloadRegistry> rtp_payload_registry_;
   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
diff --git a/audio/channel_proxy.cc b/audio/channel_proxy.cc
index 1a546f6a37..f2575159d7 100644
--- a/audio/channel_proxy.cc
+++ b/audio/channel_proxy.cc
@@ -283,11 +283,6 @@ void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
   }
 }
 
-void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  channel_->SetRtcpRttStats(rtcp_rtt_stats);
-}
-
 bool ChannelProxy::GetRecCodec(CodecInst* codec_inst) const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return channel_->GetRecCodec(*codec_inst) == 0;
diff --git a/audio/channel_proxy.h b/audio/channel_proxy.h
index e11bd2d930..65f244d754 100644
--- a/audio/channel_proxy.h
+++ b/audio/channel_proxy.h
@@ -110,7 +110,6 @@ class ChannelProxy : public RtpPacketSinkInterface {
                           RtpReceiver** rtp_receiver) const;
   virtual uint32_t GetPlayoutTimestamp() const;
   virtual void SetMinimumPlayoutDelay(int delay_ms);
-  virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
   virtual bool GetRecCodec(CodecInst* codec_inst) const;
   virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
   virtual void OnRecoverableUplinkPacketLossRate(
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index 2f6754eea8..e12b0349c9 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -69,7 +69,6 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
   MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
   MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
-  MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats));
   MOCK_METHOD2(GetAudioFrameWithInfo,
       AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
                                          AudioFrame* audio_frame));