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webrtc/modules/audio_device/android
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henrika cfbd26df1e Relands Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC 
First version was reverted in https://webrtc-review.googlesource.com/c/src/+/97941.
The issue is now fixed.

TBR=ivoc

Bug: b/113648245
Change-Id: If631fdea95aa963952f15e48e9d2d678797dc225
Reviewed-on: https://webrtc-review.googlesource.com/97942
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24573}
2018-09-05 10:24:35 +00:00
..
java/src/org/webrtc/voiceengine Temporary suppress bytebuffer warnings. 2018-04-20 11:45:28 +00:00
aaudio_player.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +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 Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +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 Relands Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC 2018-09-05 10:24:35 +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