Adds support for OpenSL ES based audio capture on Android

webrtc/modules/audio_device/audio_device_impl.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.
  */
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 #include <assert.h>
 #include <string.h>
 
 #if defined(_WIN32)
 #include "audio_device_wave_win.h"
 #if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
 #include "audio_device_core_win.h"
 #endif
 #elif defined(WEBRTC_ANDROID)
 #include <stdlib.h>
 #include "webrtc/modules/audio_device/android/audio_device_template.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
 #include "webrtc/modules/audio_device/android/opensles_player.h"
+#include "webrtc/modules/audio_device/android/opensles_recorder.h"
 #elif defined(WEBRTC_LINUX)
 #if defined(LINUX_ALSA)
 #include "audio_device_alsa_linux.h"
 #endif
 #if defined(LINUX_PULSE)
 #include "audio_device_pulse_linux.h"
 #endif
 #elif defined(WEBRTC_IOS)
 #include "audio_device_ios.h"
 #elif defined(WEBRTC_MAC)
@@ skipping 185 matching lines @@
     }
   }
 #endif  // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
 #endif  // #if defined(_WIN32)
 
 #if defined(WEBRTC_ANDROID)
   // Create an Android audio manager.
   _audioManagerAndroid.reset(new AudioManager());
   // Select best possible combination of audio layers.
   if (audioLayer == kPlatformDefaultAudio) {
-    if (_audioManagerAndroid->IsLowLatencyPlayoutSupported()) {
-      // Always use OpenSL ES for output on devices that supports the
+    if (_audioManagerAndroid->IsLowLatencyPlayoutSupported() &&
+        _audioManagerAndroid->IsLowLatencyRecordSupported()) {
+      // Use OpenSL ES for both playout and recording.
+      audioLayer = kAndroidOpenSLESAudio;
+    } else if (_audioManagerAndroid->IsLowLatencyPlayoutSupported() &&
+               !_audioManagerAndroid->IsLowLatencyRecordSupported()) {
+      // Use OpenSL ES for output on devices that only supports the
       // low-latency output audio path.
       audioLayer = kAndroidJavaInputAndOpenSLESOutputAudio;
     } else {
-      // Use Java-based audio in both directions when low-latency output
-      // is not supported.
+      // Use Java-based audio in both directions when low-latency output is
+      // not supported.
       audioLayer = kAndroidJavaAudio;
     }
   }
   AudioManager* audio_manager = _audioManagerAndroid.get();
   if (audioLayer == kAndroidJavaAudio) {
     // Java audio for both input and output audio.
     ptrAudioDevice = new AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>(
         audioLayer, audio_manager);
+  } else if (audioLayer == kAndroidOpenSLESAudio) {
+    // OpenSL ES based audio for both input and output audio.
+    ptrAudioDevice = new AudioDeviceTemplate<OpenSLESRecorder, OpenSLESPlayer>(
+        audioLayer, audio_manager);
   } else if (audioLayer == kAndroidJavaInputAndOpenSLESOutputAudio) {
     // Java audio for input and OpenSL ES for output audio (i.e. mixed APIs).
     // This combination provides low-latency output audio and at the same
     // time support for HW AEC using the AudioRecord Java API.
     ptrAudioDevice = new AudioDeviceTemplate<AudioRecordJni, OpenSLESPlayer>(
         audioLayer, audio_manager);
   } else {
     // Invalid audio layer.
-    ptrAudioDevice = NULL;
+    ptrAudioDevice = nullptr;
   }
 // END #if defined(WEBRTC_ANDROID)
 
 // Create the *Linux* implementation of the Audio Device
 //
 #elif defined(WEBRTC_LINUX)
   if ((audioLayer == kLinuxPulseAudio) ||
       (audioLayer == kPlatformDefaultAudio)) {
 #if defined(LINUX_PULSE)
     LOG(INFO) << "attempting to use the Linux PulseAudio APIs...";
@@ skipping 1590 matching lines @@
 //  PlatformAudioLayer
 // ----------------------------------------------------------------------------
 
 AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer()
     const {
   LOG(INFO) << __FUNCTION__;
   return _platformAudioLayer;
 }
 
 }  // namespace webrtc