Wire-up BWE feedback for audio receive streams.

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 "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/bitrate_controller/include/mock/mock_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/call_stats.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
@@ skipping 116 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_;
+  testing::NiceMock<MockBitrateObserver> 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));
@@ skipping 63 matching lines @@
       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