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Stefan Holmer 99f8e0841a Add a chart for packet loss on incoming streams. 
Also add stream names to labels.

R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2317113003 .

Cr-Commit-Position: refs/heads/master@{#14159}
2016-09-09 11:38:00 +00:00
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androidjunit
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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
Avoids crash at device switch on iOS by ensuring that audio parameters can be updated on the fly driven by e.g. switching audio device.
2016-09-09 11:15:49 +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
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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
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codereview.settings
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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
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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
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LICENSE_THIRD_PARTY
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OWNERS
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PATENTS
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PRESUBMIT.py
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README.chromium
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rtc_unittests_apk.isolate
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rtc_unittests.isolate
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supplement.gypi
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transport.h
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typedefs.h
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video_decoder.h
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video_encoder.h
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video_engine_tests_apk.isolate
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video_engine_tests.isolate
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video_frame.h
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video_receive_stream.h
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video_send_stream.h
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webrtc_examples.gyp
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webrtc_nonparallel_tests_apk.isolate
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webrtc_nonparallel_tests.isolate
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webrtc_perf_tests_apk.isolate
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webrtc_perf_tests.isolate
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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
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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.