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webrtc_m130/modules/audio_device/android
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henrika f217903a67 Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC. 
Also ensures that audio parameters are accessed atomically.

Bug: b/113648245
Change-Id: Ic812bfe2b2c4cfb3b00d9d411bb4986dfeda1028
Reviewed-on: https://webrtc-review.googlesource.com/97331
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24550}
2018-09-04 11:22:53 +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
…
audio_device_template.h
Removes usage of AGC APIs in the ADM.
2017-12-13 16:32:21 +00:00
audio_device_unittest.cc
Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC.
2018-09-04 11:22:53 +00:00
audio_manager_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +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_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
…
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 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