..
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
Remove rtc_base/scoped_ref_ptr.h.
2019-01-25 20:29:58 +00:00
audio_manager_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_manager.cc
Force alignment of JVM called functions.
2018-03-23 10:20:55 +00:00
audio_manager.h
Rename AttachCurrentThreadIfNeeded to avoid clash with function.
2019-01-11 19:09:23 +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