Prepare the AudioSendStream to be hooked up to send-side BWE.

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 "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video_engine/call_stats.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
 using testing::Return;
 
 const int kChannelId = 1;
 const uint32_t kSsrc = 1234;
 const char* kCName = "foo_name";
 const int kAudioLevelId = 2;
 const int kAbsSendTimeId = 3;
+const int kTransportSequenceNumberId = 4;
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
 const int kEchoReturnLoss = -65;
 const int kEchoReturnLossEnhancement = 101;
 const unsigned int kSpeechInputLevel = 96;
 const CallStatistics kCallStats = {
     1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
 const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, -451, -671};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 
 struct ConfigHelper {
-  ConfigHelper() : stream_config_(nullptr) {
+  ConfigHelper()
+      : stream_config_(nullptr),
+        process_thread_(ProcessThread::Create("AudioTestThread")),
+        congestion_controller_(process_thread_.get(),
+                               &call_stats_,
+                               &bitrate_observer_) {
     using testing::Invoke;
     using testing::StrEq;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
 
     EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
         .WillOnce(Invoke([this](int channel_id) {
           EXPECT_FALSE(channel_proxy_);
           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_,
               SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);
+          EXPECT_CALL(*channel_proxy_, SetSendTransportSequenceNumber(
+                                           kTransportSequenceNumberId))
+              .Times(1);
+          EXPECT_CALL(*channel_proxy_,
+                      SetCongestionControlObjects(
+                          congestion_controller_.pacer(),
+                          congestion_controller_.GetTransportFeedbackObserver(),
+                          congestion_controller_.packet_router()))
+              .Times(1);
+          EXPECT_CALL(*channel_proxy_,
+                      SetCongestionControlObjects(nullptr, nullptr, nullptr))
+              .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+    stream_config_.rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
+  CongestionController* congestion_controller() {
+    return &congestion_controller_;
+  }
 
   void SetupMockForGetStats() {
     using testing::DoAll;
     using testing::SetArgReferee;
 
     std::vector<ReportBlock> report_blocks;
     webrtc::ReportBlock block = kReportBlock;
     report_blocks.push_back(block);  // Has wrong SSRC.
     block.source_SSRC = kSsrc;
     report_blocks.push_back(block);  // Correct block.
@@ skipping 15 matching lines @@
     EXPECT_CALL(voice_engine_, GetEchoMetrics(_, _, _, _))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoReturnLoss),
                         SetArgReferee<1>(kEchoReturnLossEnhancement),
                         Return(0)));
     EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
                         SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
   }
 
  private:
+  class NullBitrateObserver : public BitrateObserver {
+   public:
+    virtual void OnNetworkChanged(uint32_t bitrate_bps,
+                                  uint8_t fraction_loss,
+                                  int64_t rtt_ms) {}
+  };
+
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
+  CallStats call_stats_;
+  NullBitrateObserver bitrate_observer_;
+  rtc::scoped_ptr<ProcessThread> process_thread_;
+  CongestionController congestion_controller_;
 };
 }  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = kSsrc;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
   config.rtp.c_name = kCName;
   config.voe_channel_id = kChannelId;
   config.cng_payload_type = 42;
   config.red_payload_type = 17;
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extensions: [{name: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
       "c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42, "
       "red_payload_type: 17}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
   ConfigHelper helper;
-  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        helper.congestion_controller());
 }
 
 TEST(AudioSendStreamTest, GetStats) {
   ConfigHelper helper;
-  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        helper.congestion_controller());
   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(kCodecInst.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 /
                                  (kCodecInst.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_FALSE(stats.typing_noise_detected);
 }
 
 TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
   ConfigHelper helper;
-  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        helper.congestion_controller());
   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);
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
 }  // namespace test
 }  // namespace webrtc