| Index: webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
|
| diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
|
| index 1213f333d930778a8cba06d330b88afb9cf35b1a..67f079903d9a60ab72f2ba7f0aabff7c714038f6 100644
|
| --- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
|
| +++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
|
| @@ -80,6 +80,7 @@ public class WebRtcAudioManager {
|
| private boolean hardwareAEC;
|
| private boolean hardwareAGC;
|
| private boolean hardwareNS;
|
| + private boolean intelligibility;
|
| private boolean lowLatencyOutput;
|
| private int sampleRate;
|
| private int channels;
|
| @@ -98,7 +99,7 @@ public class WebRtcAudioManager {
|
| storeAudioParameters();
|
| nativeCacheAudioParameters(
|
| sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS,
|
| - lowLatencyOutput, outputBufferSize, inputBufferSize,
|
| + intelligibility, lowLatencyOutput, outputBufferSize, inputBufferSize,
|
| nativeAudioManager);
|
| }
|
|
|
| @@ -141,6 +142,7 @@ public class WebRtcAudioManager {
|
| hardwareAEC = isAcousticEchoCancelerSupported();
|
| hardwareAGC = isAutomaticGainControlSupported();
|
| hardwareNS = isNoiseSuppressorSupported();
|
| + intelligibility = isIntelligibilityEnhancerEnabled();
|
| lowLatencyOutput = isLowLatencyOutputSupported();
|
| outputBufferSize = lowLatencyOutput ?
|
| getLowLatencyOutputFramesPerBuffer() :
|
| @@ -237,6 +239,11 @@ public class WebRtcAudioManager {
|
| return WebRtcAudioEffects.canUseNoiseSuppressor();
|
| }
|
|
|
| + // Returns true if the Intelligibility Enhancer is enabled.
|
| + private static boolean isIntelligibilityEnhancerEnabled() {
|
| + return WebRtcAudioUtils.useWebRtcBasedIntelligibilityEnhancer();
|
| + }
|
| +
|
| // Returns the minimum output buffer size for Java based audio (AudioTrack).
|
| // This size can also be used for OpenSL ES implementations on devices that
|
| // lacks support of low-latency output.
|
| @@ -287,6 +294,6 @@ public class WebRtcAudioManager {
|
|
|
| private native void nativeCacheAudioParameters(
|
| int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC,
|
| - boolean hardwareNS, boolean lowLatencyOutput, int outputBufferSize,
|
| - int inputBufferSize, long nativeAudioManager);
|
| + boolean hardwareNS, boolean intelligibility, boolean lowLatencyOutput,
|
| + int outputBufferSize, int inputBufferSize, long nativeAudioManager);
|
| }
|
|
|