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 cf2f03a2f18ac86e21dfc2bbb145dfafa22f62d4..10fe8ca33fcc06f5d57315bae6ea681337b38283 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 |
@@ -71,7 +71,6 @@ |
private int channels; |
private int outputBufferSize; |
private int inputBufferSize; |
- private int outputStreamType; |
WebRtcAudioManager(Context context, long nativeAudioManager) { |
Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo()); |
@@ -85,7 +84,7 @@ |
storeAudioParameters(); |
nativeCacheAudioParameters( |
sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS, |
- lowLatencyOutput, outputBufferSize, inputBufferSize, outputStreamType, |
+ lowLatencyOutput, outputBufferSize, inputBufferSize, |
nativeAudioManager); |
} |
@@ -133,8 +132,6 @@ |
getMinOutputFrameSize(sampleRate, channels); |
// TODO(henrika): add support for low-latency input. |
inputBufferSize = getMinInputFrameSize(sampleRate, channels); |
- outputStreamType = WebRtcAudioUtils.getOutputStreamTypeFromAudioMode( |
- audioManager.getMode()); |
} |
// Gets the current earpiece state. |
@@ -270,5 +267,5 @@ |
private native void nativeCacheAudioParameters( |
int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC, |
boolean hardwareNS, boolean lowLatencyOutput, int outputBufferSize, |
- int inputBufferSize, int outputStreamType, long nativeAudioManager); |
+ int inputBufferSize, long nativeAudioManager); |
} |