Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should…

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 57 matching lines @@
     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_, GetRemoteSSRC(kChannelId, _))
-        .WillOnce(DoAll(SetArgReferee<1>(0), Return(0)));
     EXPECT_CALL(voice_engine_, GetRTCPStatistics(kChannelId, _))
         .WillOnce(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
     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)));
@@ skipping 125 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