pbos@webrtc.org d5ce2e63df Remove EventWrapper::Reset(). ...

This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-02-13 14:58:38 +00:00

..

java/src/org/webrtc/voiceengine

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

audio_common.h

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

audio_device_template.h

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

audio_device_utility_android.cc

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

audio_device_utility_android.h

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

audio_manager_jni.cc

VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.

2013-10-03 18:23:13 +00:00

audio_manager_jni.h

VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.

2013-10-03 18:23:13 +00:00

audio_record_jni.cc

Fix problem where Android VoE can not record on multiple channels.

2015-02-12 12:33:28 +00:00

audio_record_jni.h

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

audio_track_jni.cc

Remove EventWrapper::Reset().

2015-02-13 14:58:38 +00:00

audio_track_jni.h

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

fine_audio_buffer_unittest.cc

Replace scoped_array<T> with scoped_ptr<T[]>.

2014-04-25 23:10:28 +00:00

fine_audio_buffer.cc

OpenSL (not default): Enables low latency audio on Android.

2013-09-10 18:24:07 +00:00

fine_audio_buffer.h

Replace scoped_array<T> with scoped_ptr<T[]>.

2014-04-25 23:10:28 +00:00

low_latency_event_posix.cc

Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close).

2013-12-03 19:10:20 +00:00

low_latency_event_unittest.cc

OpenSL (not default): Enables low latency audio on Android.

2013-09-10 18:24:07 +00:00

low_latency_event.h

OpenSL (not default): Enables low latency audio on Android.

2013-09-10 18:24:07 +00:00

opensles_common.cc

Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).

2013-12-11 21:42:44 +00:00

opensles_common.h

Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).

2013-12-11 21:42:44 +00:00

opensles_input.cc

WebRTC now compiles for enable_android_opensl=1.

2015-02-13 11:36:12 +00:00

opensles_input.h

WebRTC now compiles for enable_android_opensl=1.

2015-02-13 11:36:12 +00:00

opensles_output.cc

Volume buttons in AppRTCDemo should affect output audio volume (part II).

2014-11-06 15:48:05 +00:00

opensles_output.h

Replace scoped_array<T> with scoped_ptr<T[]>.

2014-04-25 23:10:28 +00:00

OWNERS

Remove former team members from OWNERS and WATCHLISTS

2014-08-26 06:12:08 +00:00

single_rw_fifo_unittest.cc

Replace scoped_array<T> with scoped_ptr<T[]>.

2014-04-25 23:10:28 +00:00

single_rw_fifo.cc

Make it possible to build webrtc for arm64.

2014-06-04 17:15:42 +00:00

single_rw_fifo.h

Replace scoped_array<T> with scoped_ptr<T[]>.

2014-04-25 23:10:28 +00:00