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webrtc_m130/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
…
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_unittest.cc
Add RTC_ prefix to non-standard format specifier macro "PRIdNS"
2019-08-07 13:36:05 +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_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
…
ensure_initialized.h
…
opensles_common.cc
…
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