Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]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.
  */
 
@@ skipping 40 matching lines @@
     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);
           return channel_proxy_;
         }));
-    EXPECT_CALL(voice_engine_,
-        SetSendAbsoluteSenderTimeStatus(kChannelId, true, kAbsSendTimeId))
-            .WillOnce(Return(0));
-    EXPECT_CALL(voice_engine_,
-        SetSendAudioLevelIndicationStatus(kChannelId, true, kAudioLevelId))
-            .WillOnce(Return(0));
     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));
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
 
   void SetupMockForGetStats() {
     using testing::DoAll;
-    using testing::SetArgPointee;
     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.
     block.fraction_lost = 0;
     report_blocks.push_back(block);  // Duplicate SSRC, bad fraction_lost.
 
-    EXPECT_CALL(voice_engine_, GetRTCPStatistics(kChannelId, _))
-        .WillRepeatedly(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
+    EXPECT_TRUE(channel_proxy_);
+    EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
+        .WillRepeatedly(Return(kCallStats));
+    EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
+        .WillRepeatedly(Return(report_blocks));
+
     EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
         .WillRepeatedly(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
-    EXPECT_CALL(voice_engine_, GetRemoteRTCPReportBlocks(kChannelId, _))
-        .WillRepeatedly(DoAll(SetArgPointee<1>(report_blocks), Return(0)));
     EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
     EXPECT_CALL(voice_engine_, GetEcMetricsStatus(_))
         .WillRepeatedly(DoAll(SetArgReferee<0>(true), Return(0)));
     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),
@@ skipping 67 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