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THis CL enables possibility to select full-duplex OpenSL ES audio in AppRTCDemo, i.e., it adds support for OpenSL ES for input as well. The user must explicitly select this new mode in the debug UI hence it is not the default selection. There is no separate UI for input and output; instead both are enabled/disabled by the same switch.

Browse Source 
BUG=webrtc:5925
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2366383002 .

Cr-Commit-Position: refs/heads/master@{#14390}
This commit is contained in:
henrika
2016-09-27 09:35:24 +02:00
parent 64ec8f826f
commit 0a52c7003d
1 changed files with 2 additions and 2 deletions
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4
webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
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@ -231,8 +231,8 @@ public class WebRtcAudioManager {
// as well. The NDK doc states that: "As of API level 21, lower latency
// audio input is supported on select devices. To take advantage of this
// feature, first confirm that lower latency output is available".
return false;
// return WebRtcAudioUtils.runningOnLollipopOrHigher() && isLowLatencyOutputSupported();
return WebRtcAudioUtils.runningOnLollipopOrHigher() &&
isLowLatencyOutputSupported();
}
// Returns true if the device has professional audio level of functionality