AudioDeviceBuffer now uses 16-bit buffers

webrtc/modules/audio_device/audio_device_buffer.h

Index: webrtc/modules/audio_device/audio_device_buffer.h
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h
index f5f8f1d2207d0be59cb85a761081a70f10850b75..621b9283f2b6cd242568dfa40a7a1a10da6477f0 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.h
+++ b/webrtc/modules/audio_device/audio_device_buffer.h
@@ -154,12 +154,13 @@ class AudioDeviceBuffer {
   bool recording_ ACCESS_ON(main_thread_checker_);
 
   // Buffer used for audio samples to be played out. Size can be changed
-  // dynamically.
-  rtc::Buffer play_buffer_ ACCESS_ON(playout_thread_checker_);
+  // dynamically. Each sample (or audio sample frame) has a size given by
+  // #channels * sizeof(uint16_t) bytes since WebRTC uses 16-bit PCM audio.
+  rtc::BufferT<uint16_t> play_buffer_ ACCESS_ON(playout_thread_checker_);
 
   // Byte buffer used for recorded audio samples. Size can be changed
   // dynamically.
-  rtc::Buffer rec_buffer_ ACCESS_ON(recording_thread_checker_);
+  rtc::BufferT<uint16_t> rec_buffer_ ACCESS_ON(recording_thread_checker_);
 
   // AGC parameters.
 #if !defined(WEBRTC_WIN)