Misc. small cleanups

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 25 matching lines @@
 const int kAudioLevelId = 2;
 const int kAbsSendTimeId = 3;
 const int kTransportSequenceNumberId = 4;
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
 const int kEchoReturnLoss = -65;
 const int kEchoReturnLossEnhancement = 101;
 const unsigned int kSpeechInputLevel = 96;
 const CallStatistics kCallStats = {
     1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
-const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, -451, -671};
+const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 const int kTelephoneEventPayloadType = 123;
 const uint8_t kTelephoneEventCode = 45;
 const uint32_t kTelephoneEventDuration = 6789;
 
 struct ConfigHelper {
   ConfigHelper()
       : stream_config_(nullptr),
         call_stats_(Clock::GetRealTimeClock()),
         process_thread_(ProcessThread::Create("AudioTestThread")),
@@ skipping 186 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