Adds support for OpenSL ES based audio capture 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 270 matching lines @@
   }
 
   // Update some stats but do it on the task queue to ensure that the members
   // are modified and read on the same thread.
   task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::UpdateRecStats, this,
                                  audio_buffer, num_samples));
   return 0;
 }
 
 int32_t AudioDeviceBuffer::DeliverRecordedData() {
-  RTC_DCHECK(audio_transport_cb_);
   rtc::CritScope lock(&_critSectCb);
 
   if (!audio_transport_cb_) {
     LOG(LS_WARNING) << "Invalid audio transport";
     return 0;
   }
 
   int32_t res(0);
   uint32_t newMicLevel(0);
   uint32_t totalDelayMS = play_delay_ms_ + rec_delay_ms_;
@@ skipping 189 matching lines @@
     int16_t max_abs = WebRtcSpl_MaxAbsValueW16(
         static_cast<int16_t*>(const_cast<void*>(audio_buffer)),
         num_samples * play_channels_);
     if (max_abs > max_play_level_) {
       max_play_level_ = max_abs;
     }
   }
 }
 
 }  // namespace webrtc