Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.

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"
+#include "webrtc/voice_engine/transmit_mixer.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
 using testing::Eq;
 using testing::Ne;
 using testing::Return;
 
 const int kChannelId = 1;
 const uint32_t kSsrc = 1234;
 const char* kCName = "foo_name";
 const int kAudioLevelId = 2;
 const int kTransportSequenceNumberId = 4;
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
 const int kEchoReturnLoss = -65;
 const int kEchoReturnLossEnhancement = 101;
 const float kResidualEchoLikelihood = -1.0f;
-const unsigned int kSpeechInputLevel = 96;
+const int32_t kSpeechInputLevel = 96;
 const CallStatistics kCallStats = {
     1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 const int kTelephoneEventPayloadType = 123;
 const int kTelephoneEventPayloadFrequency = 65432;
 const int kTelephoneEventCode = 45;
 const int kTelephoneEventDuration = 6789;
 const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000};
 
 class MockLimitObserver : public BitrateAllocator::LimitObserver {
  public:
   MOCK_METHOD2(OnAllocationLimitsChanged,
                void(uint32_t min_send_bitrate_bps,
                     uint32_t max_padding_bitrate_bps));
 };
 
+class MockTransmitMixer : public voe::TransmitMixer {
+ public:
+  MockTransmitMixer() = default;
+  virtual ~MockTransmitMixer() = default;

[kwiberg-webrtc, 2017/02/28 13:59:18]

Do you need these two lines?

[the sun, 2017/03/02 00:36:25]

Nope.

I'm tempted to make it a struct to remove one more. :)

+
+  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_,
                                &remote_bitrate_observer_,
                                &event_log_,
                                &packet_router_),
         bitrate_allocator_(&limit_observer_),
@@ skipping 130 matching lines @@
     block.fraction_lost = 0;
     report_blocks.push_back(block);  // Duplicate SSRC, bad fraction_lost.
 
     EXPECT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
         .WillRepeatedly(Return(kCallStats));
     EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
         .WillRepeatedly(Return(report_blocks));
     EXPECT_CALL(*channel_proxy_, GetSendCodec(_))
         .WillRepeatedly(DoAll(SetArgPointee<0>(kIsacCodec), Return(true)));
-    EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
-        .WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
+    EXPECT_CALL(voice_engine_, transmit_mixer())
+        .WillRepeatedly(Return(&transmit_mixer_));
     EXPECT_CALL(voice_engine_, audio_processing())
         .WillRepeatedly(Return(&audio_processing_));
 
+    EXPECT_CALL(transmit_mixer_, AudioLevelFullRange())
+        .WillRepeatedly(Return(kSpeechInputLevel));
+
     // We have to set the instantaneous value, the average, min and max. We only
     // care about the instantaneous value, so we set all to the same value.
     audio_processing_stats_.echo_return_loss.Set(
         kEchoReturnLoss, kEchoReturnLoss, kEchoReturnLoss, kEchoReturnLoss);
     audio_processing_stats_.echo_return_loss_enhancement.Set(
         kEchoReturnLossEnhancement, kEchoReturnLossEnhancement,
         kEchoReturnLossEnhancement, kEchoReturnLossEnhancement);
     audio_processing_stats_.delay_median = kEchoDelayMedian;
     audio_processing_stats_.delay_standard_deviation = kEchoDelayStdDev;
 
     EXPECT_CALL(audio_processing_, GetStatistics())
         .WillRepeatedly(Return(audio_processing_stats_));
   }
 
  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_;
+  MockTransmitMixer transmit_mixer_;
   AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
   PacketRouter packet_router_;
   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_;
@@ skipping 205 matching lines @@
   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());
   EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
   send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
 }
 
 }  // namespace test
 }  // namespace webrtc