Improves stereo/mono audio support on Android

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 358 matching lines @@
   int16_t max_abs = 0;
   RTC_DCHECK_LT(play_stat_count_, 50);
   if (++play_stat_count_ >= 50) {
     // Returns the largest absolute value in a signed 16-bit vector.
     max_abs =
         WebRtcSpl_MaxAbsValueW16(play_buffer_.data(), play_buffer_.size());
     play_stat_count_ = 0;
   }
   // Update playout stats which is used as base for periodic logging of the
   // audio output state.
-  UpdatePlayStats(max_abs, num_samples_out);
-  return static_cast<int32_t>(num_samples_out);
+  UpdatePlayStats(max_abs, num_samples_out / play_channels_);
+  return static_cast<int32_t>(num_samples_out / play_channels_);
 }
 
 int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
   RTC_DCHECK_RUN_ON(&playout_thread_checker_);
   RTC_DCHECK_GT(play_buffer_.size(), 0);
 #ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
   const double phase_increment =
       k2Pi * 440.0 / static_cast<double>(play_sample_rate_);
   int16_t* destination_r = reinterpret_cast<int16_t*>(audio_buffer);
   for (size_t i = 0; i < play_buffer_.size(); ++i) {
@@ skipping 114 matching lines @@
   RTC_DCHECK_RUN_ON(&playout_thread_checker_);
   rtc::CritScope cs(&lock_);
   ++stats_.play_callbacks;
   stats_.play_samples += samples_per_channel;
   if (max_abs > stats_.max_play_level) {
     stats_.max_play_level = max_abs;
   }
 }
 
 }  // namespace webrtc