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webrtc/modules/audio_device/android
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Markus Handell 5f61282687 Migrate modules/audio_device to webrtc::Mutex. 
Bug: webrtc:11567
Change-Id: I6d1a7145aaaae2e4cd0c8658fa31a673f857dbd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178814
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31664}
2020-07-08 09:32:12 +00:00
..
java/src/org/webrtc/voiceengine Add UMA histogram for actual Android buffer size 2020-05-29 11:14:55 +00:00
aaudio_player.cc Concatenate string literals at compile time. 2020-01-14 14:47:48 +00:00
aaudio_player.h Format almost everything. 2019-07-08 13:45:15 +00:00
aaudio_recorder.cc Concatenate string literals at compile time. 2020-01-14 14:47:48 +00:00
aaudio_recorder.h Format almost everything. 2019-07-08 13:45:15 +00:00
aaudio_wrapper.cc Deprecating ThreadChecker specific interface. 2019-04-08 16:58:07 +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 Deprecating ThreadChecker specific interface. 2019-04-08 16:58:07 +00:00
audio_device_unittest.cc Migrate modules/audio_device to webrtc::Mutex. 2020-07-08 09:32:12 +00:00
audio_manager.cc Concatenate string literals at compile time. 2020-01-14 14:47:48 +00:00
audio_manager.h Deprecating ThreadChecker specific interface. 2019-04-08 16:58:07 +00:00
audio_manager_unittest.cc Add RTC_ prefix to non-standard format specifier macro "PRIdNS" 2019-08-07 13:36:05 +00:00
audio_record_jni.cc Deprecating ThreadChecker specific interface. 2019-04-08 16:58:07 +00:00
audio_record_jni.h Format almost everything. 2019-07-08 13:45:15 +00:00
audio_track_jni.cc Add UMA histogram for actual Android buffer size 2020-05-29 11:14:55 +00:00
audio_track_jni.h Add UMA histogram for actual Android buffer size 2020-05-29 11:14:55 +00:00
build_info.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
build_info.h Format almost everything. 2019-07-08 13:45:15 +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
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 Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +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 Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
opensles_recorder.h FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer. 2018-04-19 12:20:28 +00:00