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f06f35a161b753696103615d8a40532c4fb4266a
platform-external-webrtc/webrtc
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henrika f06f35a161 Adds logging of native audio levels and UMA stats to track issues. 
This changes added a simple measurement of levels "close to the audio hardware"
both for playout and for recording. These levels are logged once each 10 seconds.

It also adds WebRTC.Audio.RecordedOnlyZeros UMA stat and it is updated at
destuction. It will report true iff all reported recording leves are zero.

BUG=NONE
R=peah@webrtc.org

Review URL: https://codereview.webrtc.org/2328433003 .

Cr-Commit-Position: refs/heads/master@{#14160}
2016-09-09 12:23:24 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Add sakal as an OWNER to some Android files.
2016-09-09 07:22:33 +00:00
audio
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
base
rtc::Optional: Tell sanitizers that unset values aren't OK to access
2016-09-09 10:35:32 +00:00
build
MB: Disable more parts of the GYP build.
2016-09-09 08:42:09 +00:00
call
Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats).
2016-09-09 07:13:39 +00:00
common_audio
VadCore: Allow signed multiplication overflow that we don't know how to fix
2016-09-09 09:04:45 +00:00
common_video
Add magjed@ as owner of webrtc/common_video
2016-09-08 21:15:34 +00:00
examples
Renamed RTCStatsReport to RTCLegacyStatsReport in objc files.
2016-09-09 08:36:32 +00:00
libjingle
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
media
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
modules
Adds logging of native audio levels and UMA stats to track issues.
2016-09-09 12:23:24 +00:00
p2p
Fixed flaky StunRequestTests which depended on the wall clock
2016-09-07 00:18:58 +00:00
pc
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
sdk
Renamed RTCStatsReport to RTCLegacyStatsReport in objc files.
2016-09-09 08:36:32 +00:00
stats
GN Templates: Move common_inherited_config to the template.
2016-09-05 13:10:23 +00:00
system_wrappers
Make UMA stats creation available in the Java interface.
2016-09-09 07:11:54 +00:00
test
Change owner of webrtc/test/channel_transport to solenberg@.
2016-09-08 15:41:59 +00:00
tools
Add a chart for packet loss on incoming streams.
2016-09-09 11:38:00 +00:00
video
Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats).
2016-09-09 07:13:39 +00:00
voice_engine
FilePlayer: Remove backwards compatibility stuff that we no longer need
2016-09-08 11:32:40 +00:00
.gitignore
…
BUILD.gn
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
…
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
DEPS
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
webrtc_examples.gyp
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
webrtc.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.