Split CongestionController into send- and receive-side classes.

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/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.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"
 #include "webrtc/voice_engine/transmit_mixer.h"
 
 namespace webrtc {
 namespace test {
 namespace {
@@ skipping 32 matching lines @@
 
 class MockTransmitMixer : public voe::TransmitMixer {
  public:
   MOCK_CONST_METHOD0(AudioLevelFullRange, int16_t());
 };
 
 struct ConfigHelper {
   explicit ConfigHelper(bool audio_bwe_enabled)
       : simulated_clock_(123456),
         stream_config_(nullptr),
-        congestion_controller_(&simulated_clock_,
-                               &bitrate_observer_,
-                               nullptr,
-                               &event_log_,
-                               &packet_router_),
+        send_side_cc_(&simulated_clock_,
+                      &bitrate_observer_,
+                      &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 29 matching lines @@
     // calls from the default ctor behavior.
     stream_config_.send_codec_spec.codec_inst = kIsacCodec;
     stream_config_.min_bitrate_bps = 10000;
     stream_config_.max_bitrate_bps = 65000;
   }
 
   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_;
-  }
+  SendSideCongestionController* send_side_cc() { return &send_side_cc_; }
   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(bool audio_bwe_enabled) {
     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);
 
     if (audio_bwe_enabled) {
       EXPECT_CALL(*channel_proxy_,
                   EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
           .Times(1);
-      EXPECT_CALL(*channel_proxy_,
-                  RegisterSenderCongestionControlObjects(
-                      congestion_controller_.pacer(), &congestion_controller_,
-                      packet_router(), Ne(nullptr)))
+      EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
+                                       send_side_cc_.pacer(), &send_side_cc_,
+                                       packet_router(), Ne(nullptr)))
           .Times(1);
     } else {
-      EXPECT_CALL(*channel_proxy_,
-                  RegisterSenderCongestionControlObjects(
-                      congestion_controller_.pacer(), &congestion_controller_,
-                      packet_router(), Eq(nullptr)))
+      EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
+                                       send_side_cc_.pacer(), &send_side_cc_,
+                                       packet_router(), Eq(nullptr)))
           .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);
     EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(testing::IsNull()))
@@ skipping 70 matching lines @@
   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_;
   MockAudioProcessing audio_processing_;
   MockTransmitMixer transmit_mixer_;
   AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
   PacketRouter packet_router_;
-  CongestionController congestion_controller_;
+  SendSideCongestionController send_side_cc_;
   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
 
@@ 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(false);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      helper.event_log(), helper.rtcp_rtt_stats());
 }
 
 TEST(AudioSendStreamTest, SendTelephoneEvent) {
   ConfigHelper helper(false);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      helper.event_log(), helper.rtcp_rtt_stats());
   helper.SetupMockForSendTelephoneEvent();
   EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
       kTelephoneEventPayloadFrequency, kTelephoneEventCode,
       kTelephoneEventDuration));
 }
 
 TEST(AudioSendStreamTest, SetMuted) {
   ConfigHelper helper(false);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      helper.event_log(), helper.rtcp_rtt_stats());
   EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
   send_stream.SetMuted(true);
 }
 
 TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
   ConfigHelper helper(true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      helper.event_log(), helper.rtcp_rtt_stats());
 }
 
 TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
   ConfigHelper helper(false);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      helper.event_log(), helper.rtcp_rtt_stats());
 }
 
 TEST(AudioSendStreamTest, GetStats) {
   ConfigHelper helper(false);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      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),
             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(false);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      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 32 matching lines @@
       .WillOnce(Return(true));
   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.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      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(false);
   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.channel_proxy(), SetVADStatus(true))
       .WillOnce(Return(true));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      helper.event_log(), helper.rtcp_rtt_stats());
 }
 
 TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
   ConfigHelper helper(false);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      helper.event_log(), helper.rtcp_rtt_stats());
   EXPECT_CALL(*helper.channel_proxy(),
               SetBitrate(helper.config().max_bitrate_bps, _));
   send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
                                6000);
 }
 
 TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
   ConfigHelper helper(false);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.packet_router(), helper.congestion_controller(),
-      helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
+      helper.packet_router(), helper.send_side_cc(), helper.bitrate_allocator(),
+      helper.event_log(), helper.rtcp_rtt_stats());
   EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
   send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
 }
 
 }  // namespace test
 }  // namespace webrtc