Enable audio capture 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"
@@ skipping 34 matching lines @@
     DCHECK_EQ(buffer_size * 100, sample_rate) <<
         "Sample rate not supported";
   }
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
   // Based on tests using the current ALSA implementation in Chrome, we have
   // found that the best combination is 20ms on the input side and 10ms on the
   // output side.
   // TODO(henrika): It might be possible to reduce the input buffer
   // size and reduce the delay even more.
   buffer_size = 2 * sample_rate / 100;
+#elif defined(OS_ANDROID)
+  // TODO(leozwang): Tune and adjust buffer size on Android.
+  buffer_size = 2 * sample_rate / 100;
 #endif
 
   return buffer_size;
 }
 
 // static
 scoped_refptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer() {
   scoped_refptr<WebRtcAudioCapturer> capturer = new  WebRtcAudioCapturer();
   return capturer;
 }
@@ skipping 352 matching lines @@
   // Inform the local renderer about the stopped device.
   // The renderer can then save resources by not asking for more data from
   // the stopped source. We are on the IO thread but the callback task will
   // be posted on the message loop of the main render thread thanks to
   // usage of BindToLoop() when the callback was initialized.
   if (!on_device_stopped_cb_.is_null())
     on_device_stopped_cb_.Run();
 }
 
 }  // namespace content