Cleanup of iOS AudioDevice implementation

webrtc/modules/audio_device/audio_device_buffer.cc

 /*
  *  Copyright (c) 2012 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 112 matching lines @@
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //  SetRecordingSampleRate
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz)
 {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingSampleRate(fsHz=%u)", fsHz);
-
     CriticalSectionScoped lock(&_critSect);
     _recSampleRate = fsHz;
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //  SetPlayoutSampleRate
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz)
 {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutSampleRate(fsHz=%u)", fsHz);
-
     CriticalSectionScoped lock(&_critSect);
     _playSampleRate = fsHz;
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //  RecordingSampleRate
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::RecordingSampleRate() const
 {
     return _recSampleRate;
 }
 
 // ----------------------------------------------------------------------------
 //  PlayoutSampleRate
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::PlayoutSampleRate() const
 {
     return _playSampleRate;
 }
 
 // ----------------------------------------------------------------------------
 //  SetRecordingChannels
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels)
 {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingChannels(channels=%u)", channels);
-
     CriticalSectionScoped lock(&_critSect);
     _recChannels = channels;
     _recBytesPerSample = 2*channels;  // 16 bits per sample in mono, 32 bits in stereo
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //  SetPlayoutChannels
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::SetPlayoutChannels(uint8_t channels)
 {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutChannels(channels=%u)", channels);
-
     CriticalSectionScoped lock(&_critSect);
     _playChannels = channels;
     // 16 bits per sample in mono, 32 bits in stereo
     _playBytesPerSample = 2*channels;
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //  SetRecordingChannel
 //
@@ skipping 379 matching lines @@
     if (_playFile.Open())
     {
         // write to binary file in mono or stereo (interleaved)
         _playFile.Write(&_playBuffer[0], _playSize);
     }
 
     return static_cast<int32_t>(_playSamples);
 }
 
 }  // namespace webrtc