Convert channel counts to size_t.

webrtc/modules/audio_device/audio_device_buffer.h

 /*
  *  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 22 matching lines @@
     int32_t RegisterAudioCallback(AudioTransport* audioCallback);
 
     int32_t InitPlayout();
     int32_t InitRecording();
 
     virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
     virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
     int32_t RecordingSampleRate() const;
     int32_t PlayoutSampleRate() const;
 
-    virtual int32_t SetRecordingChannels(uint8_t channels);
-    virtual int32_t SetPlayoutChannels(uint8_t channels);
-    uint8_t RecordingChannels() const;
-    uint8_t PlayoutChannels() const;
+    virtual int32_t SetRecordingChannels(size_t channels);
+    virtual int32_t SetPlayoutChannels(size_t channels);
+    size_t RecordingChannels() const;
+    size_t PlayoutChannels() const;
     int32_t SetRecordingChannel(
         const AudioDeviceModule::ChannelType channel);
     int32_t RecordingChannel(
         AudioDeviceModule::ChannelType& channel) const;
 
     virtual int32_t SetRecordedBuffer(const void* audioBuffer,
                                       size_t nSamples);
     int32_t SetCurrentMicLevel(uint32_t level);
     virtual void SetVQEData(int playDelayMS,
                             int recDelayMS,
@@ skipping 16 matching lines @@
 private:
     int32_t                   _id;
     CriticalSectionWrapper&         _critSect;
     CriticalSectionWrapper&         _critSectCb;
 
     AudioTransport*                 _ptrCbAudioTransport;
 
     uint32_t                  _recSampleRate;
     uint32_t                  _playSampleRate;
 
-    uint8_t                   _recChannels;
-    uint8_t                   _playChannels;
+    size_t                   _recChannels;
+    size_t                   _playChannels;
 
     // selected recording channel (left/right/both)
     AudioDeviceModule::ChannelType _recChannel;
 
     // 2 or 4 depending on mono or stereo
     size_t                   _recBytesPerSample;
     size_t                   _playBytesPerSample;
 
     // 10ms in stereo @ 96kHz
     int8_t                          _recBuffer[kMaxBufferSizeBytes];
@@ skipping 19 matching lines @@
 
     int _playDelayMS;
     int _recDelayMS;
     int _clockDrift;
     int high_delay_counter_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H