Reland "Update rtt on audio only calls".

webrtc/audio/audio_send_stream_unittest.cc

 /*
  *  Copyright (c) 2015 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 <string>
 #include <vector>
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #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"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
 using testing::Return;
@@ skipping 101 matching lines @@
                     congestion_controller_.pacer(),
                     congestion_controller_.GetTransportFeedbackObserver(),
                     congestion_controller_.packet_router()))
         .Times(1);
     EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()).Times(1);
     EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1);
     EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()).Times(1);
     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);
   }
 
   void SetupMockForSetupSendCodec() {
     EXPECT_CALL(voice_engine_, SetVADStatus(kChannelId, false, _, _))
         .WillOnce(Return(0));
     EXPECT_CALL(voice_engine_, SetFECStatus(kChannelId, false))
         .WillOnce(Return(0));
     EXPECT_CALL(*channel_proxy_, DisableAudioNetworkAdaptor());
     // Let |GetSendCodec| return -1 for the first time to indicate that no send
     // codec has been set.
     EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
         .WillOnce(Return(-1));
     EXPECT_CALL(voice_engine_, SetSendCodec(kChannelId, _)).WillOnce(Return(0));
   }
+  RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; }
 
   void SetupMockForSendTelephoneEvent() {
     EXPECT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_,
         SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType,
                                          kTelephoneEventPayloadFrequency))
             .WillOnce(Return(true));
     EXPECT_CALL(*channel_proxy_,
         SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
             .WillOnce(Return(true));
@@ skipping 43 matching lines @@
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
   testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   MockAudioProcessing audio_processing_;
   AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
   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
   // and deleted before any other members.
   rtc::TaskQueue worker_queue_;
 };
 }  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
   AudioSendStream::Config config(nullptr);
@@ skipping 25 matching lines @@
       "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: "
       "320, channels: 1, rate: 32000}}}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   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) {
   ConfigHelper helper;
   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,
       kTelephoneEventPayloadFrequency, kTelephoneEventCode,
       kTelephoneEventDuration));
 }
 
 TEST(AudioSendStreamTest, SetMuted) {
   ConfigHelper helper;
   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);
 }
 
 TEST(AudioSendStreamTest, GetStats) {
   ConfigHelper helper;
   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);
   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
   EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost),
             stats.packets_lost);
   EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
   EXPECT_EQ(std::string(kIsacCodec.plname), stats.codec_name);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
@@ skipping 10 matching lines @@
   EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement);
   EXPECT_EQ(kResidualEchoLikelihood, stats.residual_echo_likelihood);
   EXPECT_FALSE(stats.typing_noise_detected);
 }
 
 TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
   ConfigHelper helper;
   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);
 
   internal::AudioState* internal_audio_state =
       static_cast<internal::AudioState*>(helper.audio_state().get());
   VoiceEngineObserver* voe_observer =
       static_cast<VoiceEngineObserver*>(internal_audio_state);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
   EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
@@ skipping 33 matching lines @@
   EXPECT_CALL(
       *helper.channel_proxy(),
       SetReceiverFrameLengthRange(stream_config.send_codec_spec.min_ptime_ms,
                                   stream_config.send_codec_spec.max_ptime_ms));
   EXPECT_CALL(
       *helper.channel_proxy(),
       EnableAudioNetworkAdaptor(*stream_config.audio_network_adaptor_config));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
+      helper.event_log(), helper.rtcp_rtt_stats());
 }
 
 // VAD is applied when codec is mono and the CNG frequency matches the codec
 // sample rate.
 TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
   ConfigHelper helper;
   auto stream_config = helper.config();
   const CodecInst kG722Codec = {9, "g722", 8000, 160, 1, 16000};
   stream_config.send_codec_spec.codec_inst = kG722Codec;
   stream_config.send_codec_spec.cng_plfreq = 8000;
   stream_config.send_codec_spec.cng_payload_type = 105;
   EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
       .WillOnce(Return(0));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
+      helper.event_log(), helper.rtcp_rtt_stats());
 }
 
 }  // namespace test
 }  // namespace webrtc