Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.

webrtc/audio/audio_receive_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 51 matching lines @@
         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_, SetLocalSSRC(kLocalSsrc)).Times(1);
+          EXPECT_CALL(*channel_proxy_,
+              SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
+                  .Times(1);
+          EXPECT_CALL(*channel_proxy_,
+              SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
+                  .Times(1);
           return channel_proxy_;
         }));
-    EXPECT_CALL(voice_engine_,
-        SetReceiveAbsoluteSenderTimeStatus(kChannelId, true, kAbsSendTimeId))
-            .WillOnce(Return(0));
-    EXPECT_CALL(voice_engine_,
-        SetReceiveAudioLevelIndicationStatus(kChannelId, true, kAudioLevelId))
-            .WillOnce(Return(0));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
   }
 
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
 
   void SetupMockForGetStats() {
     using testing::DoAll;
-    using testing::SetArgPointee;
     using testing::SetArgReferee;
 
-    EXPECT_CALL(voice_engine_, GetRTCPStatistics(kChannelId, _))
-        .WillOnce(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
+    EXPECT_TRUE(channel_proxy_);
+    EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
+        .WillOnce(Return(kCallStats));
+    EXPECT_CALL(*channel_proxy_, GetDelayEstimate())
+        .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
+    EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange())
+        .WillOnce(Return(kSpeechOutputLevel));
+    EXPECT_CALL(*channel_proxy_, GetNetworkStatistics())
+        .WillOnce(Return(kNetworkStats));
+    EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
+        .WillOnce(Return(kAudioDecodeStats));
+
     EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
         .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
-    EXPECT_CALL(voice_engine_, GetDelayEstimate(kChannelId, _, _))
-        .WillOnce(DoAll(SetArgPointee<1>(kJitterBufferDelay),
-                        SetArgPointee<2>(kPlayoutBufferDelay), Return(0)));
-    EXPECT_CALL(voice_engine_,
-        GetSpeechOutputLevelFullRange(kChannelId, _)).WillOnce(
-            DoAll(SetArgReferee<1>(kSpeechOutputLevel), Return(0)));
-    EXPECT_CALL(voice_engine_, GetNetworkStatistics(kChannelId, _))
-        .WillOnce(DoAll(SetArgReferee<1>(kNetworkStats), Return(0)));
-    EXPECT_CALL(voice_engine_, GetDecodingCallStatistics(kChannelId, _))
-        .WillOnce(DoAll(SetArgPointee<1>(kAudioDecodeStats), Return(0)));
   }
 
  private:
   MockRemoteBitrateEstimator remote_bitrate_estimator_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioReceiveStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
 };
 
@@ skipping 110 matching lines @@
   EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
   EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
   EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc