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webrtc/modules/audio_device/android
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Mirko Bonadei 9f9e20a3dc Fix errorprone issues preventing Chromium Roll. 
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889

Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
2019-11-27 12:52:48 +00:00
..
java/src/org/webrtc/voiceengine Fix errorprone issues preventing Chromium Roll. 2019-11-27 12:52:48 +00:00
aaudio_player.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
aaudio_player.h Format almost everything. 2019-07-08 13:45:15 +00:00
aaudio_recorder.cc Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +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 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" 2019-08-07 13:36:05 +00:00
audio_manager.cc Deprecating ThreadChecker specific interface. 2019-04-08 16:58:07 +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 Allows configuration of playout audio buffer 2019-08-29 12:57:14 +00:00
audio_track_jni.h Format almost everything. 2019-07-08 13:45:15 +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 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 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