Enable cpplint and fix cpplint errors in webrtc/*audio

webrtc/audio/audio_transport_proxy.cc

 /*
  *  Copyright (c) 2016 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 34 matching lines @@
 int32_t AudioTransportProxy::RecordedDataIsAvailable(
     const void* audioSamples,
     const size_t nSamples,
     const size_t nBytesPerSample,
     const size_t nChannels,
     const uint32_t samplesPerSec,
     const uint32_t totalDelayMS,
     const int32_t clockDrift,
     const uint32_t currentMicLevel,
     const bool keyPressed,
-    uint32_t& newMicLevel) {
+    uint32_t& newMicLevel) {  // NOLINT: to avoid changing APIs
   // Pass call through to original audio transport instance.
   return voe_audio_transport_->RecordedDataIsAvailable(
       audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
       totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
 }
 
 int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples,
                                               const size_t nBytesPerSample,
                                               const size_t nChannels,
                                               const uint32_t samplesPerSec,
@@ skipping 58 matching lines @@
   mixer_->Mix(number_of_channels, &mixed_frame_);
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
   *ntp_time_ms = mixed_frame_.ntp_time_ms_;
 
   const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_,
                                        static_cast<int16_t*>(audio_data));
   RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
 }
 
 }  // namespace webrtc