Move ownership of PacketRouter from CongestionController to Call.

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.
  */
 
@@ skipping 49 matching lines @@
                     uint32_t max_padding_bitrate_bps));
 };
 
 struct ConfigHelper {
   ConfigHelper()
       : simulated_clock_(123456),
         stream_config_(nullptr),
         congestion_controller_(&simulated_clock_,
                                &bitrate_observer_,
                                &remote_bitrate_observer_,
-                               &event_log_),
+                               &event_log_,
+                               &packet_router_),
         bitrate_allocator_(&limit_observer_),
         worker_queue_("ConfigHelper_worker_queue") {
     using testing::Invoke;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_, audio_device_module());
     EXPECT_CALL(voice_engine_, audio_processing());
@@ skipping 22 matching lines @@
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
     // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
     // calls from the default ctor behavior.
     stream_config_.send_codec_spec.codec_inst = kIsacCodec;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
+  PacketRouter* packet_router() { return &packet_router_; }
   CongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
   MockVoiceEngine* voice_engine() { return &voice_engine_; }
 
   void SetupDefaultChannelProxy() {
     using testing::StrEq;
     channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
     EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
     EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
     EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
     EXPECT_CALL(*channel_proxy_,
                 SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
         .Times(1);
     EXPECT_CALL(*channel_proxy_,
                 EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
         .Times(1);
     EXPECT_CALL(*channel_proxy_,
                 RegisterSenderCongestionControlObjects(
                     congestion_controller_.pacer(),
                     congestion_controller_.GetTransportFeedbackObserver(),
-                    congestion_controller_.packet_router()))
+                    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
   }
 
   void SetupMockForSetupSendCodec() {
@@ skipping 62 matching lines @@
  private:
   SimulatedClock simulated_clock_;
   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_;
+  PacketRouter packet_router_;
   CongestionController congestion_controller_;
   MockRtcEventLog event_log_;
   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
 
@@ skipping 26 matching lines @@
       "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: "
       "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.packet_router(), helper.congestion_controller(),
+      helper.bitrate_allocator(), helper.event_log());
 }
 
 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.packet_router(), helper.congestion_controller(),
+      helper.bitrate_allocator(), helper.event_log());
   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.packet_router(), helper.congestion_controller(),
+      helper.bitrate_allocator(), helper.event_log());
   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.packet_router(), helper.congestion_controller(),
+      helper.bitrate_allocator(), helper.event_log());
   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),
             stats.ext_seqnum);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
                                  (kIsacCodec.plfreq / 1000)),
             stats.jitter_ms);
   EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
   EXPECT_EQ(static_cast<int32_t>(kSpeechInputLevel), stats.audio_level);
   EXPECT_EQ(-1, stats.aec_quality_min);
   EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms);
   EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms);
   EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss);
   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.packet_router(), helper.congestion_controller(),
+      helper.bitrate_allocator(), helper.event_log());
   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 32 matching lines @@
       .WillOnce(Return(0));
   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.packet_router(), helper.congestion_controller(),
+      helper.bitrate_allocator(), helper.event_log());
 }
 
 // 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.packet_router(), helper.congestion_controller(),
+      helper.bitrate_allocator(), helper.event_log());
 }
 
 }  // namespace test
 }  // namespace webrtc