Ivo Creusen f1393e23a2 Add UMA histogram for actual Android buffer size ... Previously a histogram was added to track the requested buffer size, this CL adds a histogram for the actually used buffer size. Bug: b/157429867 Change-Id: I04016760982a4c43b8ba8f0e095fe1171b705258 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176227 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31385}		2020-05-29 11:14:55 +00:00
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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	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_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	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
ensure_initialized.h
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