mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-19 08:37:54 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/modules/audio_device/android
History
Danil Chapovalov 1c41be6e05 Propagate TaskQueueFactory to AudioDeviceBuffer 
keep using GlobalTaskQueueFactory in android/ios bindings.
Switch to DefaultTaskQueueFactory in tests.

Bug: webrtc:10284
Change-Id: I034c70542be5eeb830be86527830d51204fb2855
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130223
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27380}
2019-04-01 08:00:49 +00:00
..
java/src/org/webrtc/voiceengine Use android Nullable instead of javax Nullable 2018-12-10 15:03:58 +00:00
aaudio_player.cc Removes flaky thread checker in AudioDeviceBuffer. 2018-09-13 11:41:52 +00:00
aaudio_player.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
aaudio_recorder.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
aaudio_recorder.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
aaudio_wrapper.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
aaudio_wrapper.h Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
audio_common.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_device_template.h Removes usage of AGC APIs in the ADM. 2017-12-13 16:32:21 +00:00
audio_device_unittest.cc Propagate TaskQueueFactory to AudioDeviceBuffer 2019-04-01 08:00:49 +00:00
audio_manager.cc Prefix AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO with WEBRTC_. 2019-03-28 08:44:27 +00:00
audio_manager.h Rename AttachCurrentThreadIfNeeded to avoid clash with function. 2019-01-11 19:09:23 +00:00
audio_manager_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_record_jni.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_record_jni.h Rename AttachCurrentThreadIfNeeded to avoid clash with function. 2019-01-11 19:09:23 +00:00
audio_track_jni.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_track_jni.h Rename AttachCurrentThreadIfNeeded to avoid clash with function. 2019-01-11 19:09:23 +00:00
build_info.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
build_info.h Rename AttachCurrentThreadIfNeeded to avoid clash with function. 2019-01-11 19:09:23 +00:00
ensure_initialized.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ensure_initialized.h Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
opensles_common.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
opensles_common.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opensles_player.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
opensles_player.h FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer. 2018-04-19 12:20:28 +00:00
opensles_recorder.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
opensles_recorder.h FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer. 2018-04-19 12:20:28 +00:00