Convert channel counts to size_t.

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 151 matching lines @@
 
 int32_t AudioDeviceBuffer::PlayoutSampleRate() const
 {
     return _playSampleRate;
 }
 
 // ----------------------------------------------------------------------------
 //  SetRecordingChannels
 // ----------------------------------------------------------------------------
 
-int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels)
+int32_t AudioDeviceBuffer::SetRecordingChannels(size_t 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)
+int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels)
 {
     CriticalSectionScoped lock(&_critSect);
     _playChannels = channels;
     // 16 bits per sample in mono, 32 bits in stereo
     _playBytesPerSample = 2*channels;
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //  SetRecordingChannel
@@ skipping 37 matching lines @@
 int32_t AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& channel) const
 {
     channel = _recChannel;
     return 0;
 }
 
 // ----------------------------------------------------------------------------
 //  RecordingChannels
 // ----------------------------------------------------------------------------
 
-uint8_t AudioDeviceBuffer::RecordingChannels() const
+size_t AudioDeviceBuffer::RecordingChannels() const
 {
     return _recChannels;
 }
 
 // ----------------------------------------------------------------------------
 //  PlayoutChannels
 // ----------------------------------------------------------------------------
 
-uint8_t AudioDeviceBuffer::PlayoutChannels() const
+size_t AudioDeviceBuffer::PlayoutChannels() const
 {
     return _playChannels;
 }
 
 // ----------------------------------------------------------------------------
 //  SetCurrentMicLevel
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level)
 {
@@ skipping 218 matching lines @@
 }
 
 // ----------------------------------------------------------------------------
 //  RequestPlayoutData
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples)
 {
     uint32_t playSampleRate = 0;
     size_t playBytesPerSample = 0;
-    uint8_t playChannels = 0;
+    size_t playChannels = 0;
     {
         CriticalSectionScoped lock(&_critSect);
 
         // Store copies under lock and use copies hereafter to avoid race with
         // setter methods.
         playSampleRate = _playSampleRate;
         playBytesPerSample = _playBytesPerSample;
         playChannels = _playChannels;
 
         // Ensure that user has initialized all essential members
@@ skipping 74 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