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palmkvist a40672a120 Add UMA stats to bad call detection. 
Just simple "percentage of call that was bad" stats.

BUG=webrtc:6814

Review-Url: https://codereview.webrtc.org/2578213003
Cr-Commit-Position: refs/heads/master@{#16049}
2017-01-13 13:58:34 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Refactoring of RTCP options in BaseChannel.
2017-01-13 05:59:29 +00:00
audio
GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.
2017-01-05 14:03:24 +00:00
base
Update smoothed bitrate.
2017-01-12 18:17:38 +00:00
build
Remove lint ignores and add WrongCall ignore directly to the test.
2017-01-12 14:19:40 +00:00
call
Clean up ramp-up tests and make sure they all pass.
2017-01-09 12:19:24 +00:00
common_audio
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
2017-01-12 10:24:27 +00:00
common_video
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
examples
Remove lint ignores and add WrongCall ignore directly to the test.
2017-01-12 14:19:40 +00:00
logging
Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.
2016-12-20 13:03:58 +00:00
media
Remove WebRtcVideoSendStream2::VideoSink inheritance. Remove sending black frame on source removal.
2017-01-13 13:57:25 +00:00
modules
Update smoothed bitrate.
2017-01-12 18:17:38 +00:00
p2p
Remove BaseChannel's dependency on TransportController.
2017-01-13 03:37:48 +00:00
pc
Refactoring of RTCP options in BaseChannel.
2017-01-13 05:59:29 +00:00
sdk
Drop dependency to Chromium in WebRtcJniBootTest.
2017-01-13 13:56:16 +00:00
stats
Update smoothed bitrate.
2017-01-12 18:17:38 +00:00
system_wrappers
Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
2017-01-09 09:22:16 +00:00
test
Don't detect a new frame if a previous packet is used in a previous frame.
2017-01-11 10:01:56 +00:00
tools
Add method needed to extract frame capture and arrival timestamps from rtc event logs.
2016-12-20 16:51:52 +00:00
video
Add UMA stats to bad call detection.
2017-01-13 13:58:34 +00:00
voice_engine
Update smoothed bitrate.
2017-01-12 18:17:38 +00:00
.gitignore
…
BUILD.gn
Make P2PTransportChannel inherit from IceTransportInternal.
2017-01-12 23:58:31 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +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
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +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
…
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +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

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.