AudioDeviceBuffer now uses 16-bit buffers

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 42 matching lines @@
   int32_t PlayoutSampleRate() const;
 
   int32_t SetRecordingChannels(size_t channels);
   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* audio_buffer,
-                                    size_t num_samples);
+                                    size_t samples_per_channel);
   int32_t SetCurrentMicLevel(uint32_t level);
   virtual void SetVQEData(int play_delay_ms, int rec_delay_ms, int clock_drift);
   virtual int32_t DeliverRecordedData();
   uint32_t NewMicLevel() const;
 
-  virtual int32_t RequestPlayoutData(size_t num_samples);
+  virtual int32_t RequestPlayoutData(size_t samples_per_channel);
   virtual int32_t GetPlayoutData(void* audio_buffer);
 
   // TODO(henrika): these methods should not be used and does not contain any
   // valid implementation. Investigate the possibility to either remove them
   // or add a proper implementation if needed.
   int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopInputFileRecording();
   int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
   int32_t StopOutputFileRecording();
 
   int32_t SetTypingStatus(bool typing_status);
 
  private:
   // Starts/stops periodic logging of audio stats.
   void StartPeriodicLogging();
   void StopPeriodicLogging();
 
   // Called periodically on the internal thread created by the TaskQueue.
   // Updates some stats but dooes it on the task queue to ensure that access of
   // members is serialized hence avoiding usage of locks.
   // state = LOG_START => members are initialized and the timer starts.
   // state = LOG_STOP => no logs are printed and the timer stops.
   // state = LOG_ACTIVE => logs are printed and the timer is kept alive.
   void LogStats(LogState state);
 
   // Updates counters in each play/record callback but does it on the task
   // queue to ensure that they can be read by LogStats() without any locks since
   // each task is serialized by the task queue.
-  void UpdateRecStats(int16_t max_abs, size_t num_samples);
-  void UpdatePlayStats(int16_t max_abs, size_t num_samples);
+  void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
+  void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
 
   // Clears all members tracking stats for recording and playout.
   // These methods both run on the task queue.
   void ResetRecStats();
   void ResetPlayStats();
 
   // This object lives on the main (creating) thread and most methods are
   // called on that same thread. When audio has started some methods will be
   // called on either a native audio thread for playout or a native thread for
   // recording. Some members are not annotated since they are "protected by
@@ skipping 37 matching lines @@
   size_t rec_channels_;
   size_t play_channels_;
 
   // Keeps track of if playout/recording are active or not. A combination
   // of these states are used to determine when to start and stop the timer.
   // Only used on the creating thread and not used to control any media flow.
   bool playing_ ACCESS_ON(main_thread_checker_);
   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. The 16-bit samples are interleaved, hence the size is
+  // proportional to the number of channels.
+  rtc::BufferT<int16_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<int16_t> rec_buffer_ ACCESS_ON(recording_thread_checker_);
 
   // AGC parameters.
 #if !defined(WEBRTC_WIN)
   uint32_t current_mic_level_ ACCESS_ON(recording_thread_checker_);
 #else
   // Windows uses a dedicated thread for volume APIs.
   uint32_t current_mic_level_;
 #endif
   uint32_t new_mic_level_ ACCESS_ON(recording_thread_checker_);
 
@@ skipping 58 matching lines @@
   uint64_t rec_start_time_ ACCESS_ON(main_thread_checker_);
 
   // Set to true at construction and modified to false as soon as one audio-
   // level estimate larger than zero is detected.
   bool only_silence_recorded_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_