Adds stereo support for Java-based input and output audio on Android

webrtc/modules/audio_device/android/audio_track_jni.h

 /*
  *  Copyright (c) 2013 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 121 matching lines @@
 
   // Number of bytes in the direct audio buffer owned by |j_audio_track_|.
   size_t direct_buffer_capacity_in_bytes_;
 
   // Number of audio frames per audio buffer. Each audio frame corresponds to
   // one sample of PCM mono data at 16 bits per sample. Hence, each audio
   // frame contains 2 bytes (given that the Java layer only supports mono).
   // Example: 480 for 48000 Hz or 441 for 44100 Hz.
   size_t frames_per_buffer_;
 
+  // Number of bytes per audio frame assuming that all sampels are 16-bit linear
+  // PCM samples.
+  // Example: 4 bytes for stereo and 2 bytes for mono.
+  size_t bytes_per_frame_;

[magjed_webrtc, 2016/11/16 13:42:09]

Do we need to store this as a member variable, or can we call
'audio_parameters_.channels() * sizeof(int16_t)' when we need it instead?

[henrika_webrtc, 2016/11/16 13:56:38]

Done.

+
   bool initialized_;
 
   bool playing_;
 
   // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
   // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
   // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
   // and therefore outlives this object.
   AudioDeviceBuffer* audio_device_buffer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_