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webrtc/modules/audio_device/android
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Artem Titarenko 69540f4419 Use android Nullable instead of javax Nullable 
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.

Original comment from upstream change:

> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.

Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
2018-12-10 15:03:58 +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 Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
aaudio_recorder.cc Removes flaky thread checker in AudioDeviceBuffer. 2018-09-13 11:41:52 +00:00
aaudio_recorder.h Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +00:00
aaudio_wrapper.cc Add support of AAudio in native WebRTC on Android O and above 2018-03-16 10:20:27 +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 Eliminate use of EventWrapper from android audio device tests 2018-11-12 13:22:46 +00:00
audio_manager.cc Force alignment of JVM called functions. 2018-03-23 10:20:55 +00:00
audio_manager.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_manager_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_record_jni.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_record_jni.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_track_jni.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_track_jni.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
build_info.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
build_info.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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 absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +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 absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
opensles_recorder.h FineAudioBuffer now uses 16-bit audio samples to match the AudioDeviceBuffer. 2018-04-19 12:20:28 +00:00