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henrik.lundin 290d43aa14 Add a new UMA metric in APM to track incoming capture-side audio level 
This CL adds WebRTC.Audio.ApmCaptureInputLevelAverage and
WebRTC.Audio.ApmCaptureInputLevelPeak. The metrics are updated once
every 10 seconds.

BUG=webrtc:6622

Review-Url: https://codereview.webrtc.org/2534473004
Cr-Commit-Position: refs/heads/master@{#15300}
2016-11-29 16:09:17 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCStatsIntegrationTest added.
2016-11-29 09:57:08 +00:00
audio
Added webrtc/audio/utility directory and empty GN target.
2016-11-29 15:32:15 +00:00
base
Only use BoringSSL time callback in unit tests.
2016-11-29 01:38:39 +00:00
build
Landmine to clobber Windows builders
2016-11-29 10:14:47 +00:00
call
RTC_[D]CHECK_op: Remove "u" suffix on integer constants
2016-11-28 23:21:51 +00:00
common_audio
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
common_video
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
examples
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
modules
Add a new UMA metric in APM to track incoming capture-side audio level
2016-11-29 16:09:17 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
sdk
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
stats
Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t.
2016-11-28 09:55:05 +00:00
system_wrappers
RTC_[D]CHECK_op: Remove "u" suffix on integer constants
2016-11-28 23:21:51 +00:00
test
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
tools
Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly.
2016-11-29 10:01:26 +00:00
video
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
voice_engine
Refactor RMSLevel and give it new functionality
2016-11-29 12:26:31 +00:00
.gitignore
…
BUILD.gn
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
common_types.h
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +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
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
2016-11-22 09:43:06 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +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.