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ilnik 6d5b4d6fe1 Piggybacking simulcast id and ALR experiment id into video content type extension. 
Use it to slice UMA video receive statis.

BUG=8032

Review-Url: https://codereview.webrtc.org/2986893002
Cr-Commit-Position: refs/heads/master@{#19598}
2017-08-30 10:32:14 +00:00
..
api
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
audio
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
call
Wiring discard rate of audio FEC/RED packets up to StatsReport.
2017-08-28 13:17:55 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
examples
Android: Replace webrtc_jni namespace with nested jni namespace
2017-08-28 23:24:06 +00:00
logging
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ )
2017-08-29 11:49:00 +00:00
media
Let VideoEncoderSoftwareFallbackWrapper own the wrapped encoder
2017-08-29 07:58:52 +00:00
modules
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Support a user-provided string for the TLS ALPN extension.
2017-08-29 20:11:16 +00:00
pc
Fix the Chromium crash when creating an answer without a remote description.
2017-08-29 20:23:53 +00:00
rtc_base
Support a user-provided string for the TLS ALPN extension.
2017-08-29 20:11:16 +00:00
rtc_tools
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ )
2017-08-29 11:49:00 +00:00
sdk
Support a user-provided string for the TLS ALPN extension.
2017-08-29 20:11:16 +00:00
stats
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
system_wrappers
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
test
Add a new frame generator that cycles through randomly generated slides.
2017-08-29 16:12:57 +00:00
video
Piggybacking simulcast id and ALR experiment id into video content type extension.
2017-08-30 10:32:14 +00:00
voice_engine
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Removing EXPAT_RELATIVE_PATH and rtc_relative_path.
2017-08-25 15:14:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )
2017-08-28 14:16:43 +00:00
common_types.h
Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )
2017-08-28 14:16:43 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Removing video_{receive,send}_stream.h stub headers.
2017-08-29 10:35:02 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
webrtc.gni
Removing EXPAT_RELATIVE_PATH and rtc_relative_path.
2017-08-25 15:14:04 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.