3 Commits

Author	SHA1	Message	Date
sophiechang	f935bcc2f7	Use strcmp instead of == operator for c.name and name to find appropriate classes for WebRtcAudio*.java ... . BUG= Review URL: https://codereview.webrtc.org/1229443002 Cr-Commit-Position: refs/heads/master@{#9543}	2015-07-07 08:10:21 +00:00
henrika	ee369e4277	Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes ... BUG=NONE TEST=./webrtc/build/android/test_runner.py gtest -s modules_unittests --gtest_filter=AudioDevice* R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/51079004 Cr-Commit-Position: refs/heads/master@{#9271}	2015-05-25 08:11:38 +00:00
henrika	b26198972c	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}	2015-05-18 14:49:04 +00:00