Using native sampling rate and optimal buffer size for audio on Android.

content/renderer/media/webrtc_audio_capturer.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "content/common/child_process.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/webrtc_audio_capturer_sink_owner.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/audio/audio_util.h"
 #include "media/audio/sample_rates.h"
 
 namespace content {
 
 // Supported hardware sample rates for input and output sides.
 #if defined(OS_WIN) || defined(OS_MACOSX)
 // media::GetAudioInputHardwareSampleRate() asks the audio layer
 // for its current sample rate (set by the user) on Windows and Mac OS X.
 // The listed rates below adds restrictions and WebRtcAudioDeviceImpl::Init()
 // will fail if the user selects any rate outside these ranges.
 static int kValidInputRates[] = {96000, 48000, 44100, 32000, 16000, 8000};
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
 static int kValidInputRates[] = {48000, 44100};
 #elif defined(OS_ANDROID)
-static int kValidInputRates[] = {48000, 44100, 16000};
+static int kValidInputRates[] = {48000, 44100};
 #else
 static int kValidInputRates[] = {44100};
 #endif
 
 static int GetBufferSizeForSampleRate(int sample_rate) {
   int buffer_size = 0;
 #if defined(OS_WIN) || defined(OS_MACOSX)
   // Use different buffer sizes depending on the current hardware sample rate.
   if (sample_rate == 44100) {
     // We do run at 44.1kHz at the actual audio layer, but ask for frames
@@ skipping 346 matching lines @@
 }
 
 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const {
   base::AutoLock auto_lock(lock_);
   // |buffer_| can be NULL when SetCapturerSource() or Initialize() has not
   // been called.
   return buffer_.get() ? buffer_->params() : media::AudioParameters();
 }
 
 }  // namespace content