Adding support for OpenSL ES output in native WebRTC

BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo

Summary:

- Removes dependency of the 'enable_android_opensl' compiler flag.
  Instead, OpenSL ES is always supported, and will enabled for devices that
  supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.

Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.

R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51759004

Cr-Commit-Position: refs/heads/master@{#9208}

webrtc/modules/audio_device/android/fine_audio_buffer.h

@@ -49,7 +49,8 @@ class FineAudioBuffer {
  private:
   // Device buffer that provides 10ms chunks of data.
   AudioDeviceBuffer* device_buffer_;
-  int desired_frame_size_bytes_;  // Number of bytes delivered per GetBufferData
+  // Number of bytes delivered per GetBufferData
+  int desired_frame_size_bytes_;
   int sample_rate_;
   int samples_per_10_ms_;
   // Convenience parameter to avoid converting from samples
@@ -57,8 +58,10 @@ class FineAudioBuffer {
 
   // Storage for samples that are not yet asked for.
   rtc::scoped_ptr<int8_t[]> cache_buffer_;
-  int cached_buffer_start_;  // Location of first unread sample.
-  int cached_bytes_;  // Number of bytes stored in cache.
+  // Location of first unread sample.
+  int cached_buffer_start_;
+  // Number of bytes stored in cache.
+  int cached_bytes_;
 };
 
 }  // namespace webrtc