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64e739aeae5629cbbebf2a19e1d3e6b452bb6d0b
platform-external-webrtc/webrtc
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ilnik 64e739aeae Add content type information to Encoded Images and add corresponding RTP extension header. 
Use it to separate UMA e2e delay metric between screenshare from video.
Content type extension is set based on encoder settings and processed and decoders.

Also,
Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2772033002
Cr-Commit-Position: refs/heads/master@{#17640}
2017-04-11 08:46:04 +00:00
..
api
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
audio
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
base
Make safe_cmp::* constexpr
2017-04-11 05:44:07 +00:00
call
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
common_audio
Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
2017-04-11 06:21:43 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Removing unnecessary parameters from initializeAndroidGlobals.
2017-04-10 22:08:02 +00:00
logging
Resolve dependency between rtc_event_log_api and remote_bitrate_estimator
2017-04-11 07:49:44 +00:00
media
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
modules
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
ortc
Disable the ORTC integration tests on TSan.
2017-03-28 20:45:18 +00:00
p2p
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
pc
Make RtpTransport actually implement RtpTransportInterface
2017-04-10 23:17:57 +00:00
sdk
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
2017-04-11 06:21:43 +00:00
test
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
tools
Revert of Add first part of the network_tester functionality. (patchset #13 id:260001 of https://codereview.webrtc.org/2779233002/ )
2017-04-11 07:16:51 +00:00
video
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
voice_engine
Removing unnecessary parameters from initializeAndroidGlobals.
2017-04-10 22:08:02 +00:00
.gitignore
…
BUILD.gn
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
2017-04-07 07:59:12 +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
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
common_types.h
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
config.cc
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
config.h
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +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
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_decoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_encoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_frame.h
Add content type information to Encoded Images and add corresponding RTP extension header.
2017-04-11 08:46:04 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.