Enable audio capture on Android

content/renderer/media/webrtc_audio_renderer.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_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "content/renderer/media/audio_device_factory.h"
@@ skipping 16 matching lines @@
 // Supported hardware sample rates for output sides.
 #if defined(OS_WIN) || defined(OS_MACOSX)
 // AudioHardwareConfig::GetOutputSampleRate() 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 Initialize() will fail if the user selects
 // any rate outside these ranges.
 const int kValidOutputRates[] = {96000, 48000, 44100, 32000, 16000};
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
 const int kValidOutputRates[] = {48000, 44100};
 #elif defined(OS_ANDROID)
-// On Android, the most popular sampling rate is 16000.
+// TODO(leozwang): We want to use native sampling rate on Android to achieve
+// low latency, currently 16000 is used to work around audio problem on some
+// Android devices.
 const int kValidOutputRates[] = {48000, 44100, 16000};
 #else
 const int kValidOutputRates[] = {44100};
 #endif
 
 // TODO(xians): Merge the following code to WebRtcAudioCapturer, or remove.
 enum AudioFramesPerBuffer {
   k160,
   k320,
   k440,  // WebRTC works internally with 440 audio frames at 44.1kHz.
@@ skipping 294 matching lines @@
   }
 
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(buffer_.get(),
                              audio_bus->frames(),
                              sizeof(buffer_[0]));
 }
 
 }  // namespace content