webrtc_m130

History

Zijie He 05ec178808 Add CroppedDesktopFrameTest

CroppedDesktopFrame was broken by recent changes unexpectedly, it should be
covered by test cases to avoid this kind of accident.

Bug: chromium:761703
Change-Id: Ib95e4fe19a8bb9799db24fefd3ad154ddd15e53e
Reviewed-on: https://chromium-review.googlesource.com/651448
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19725}

2017-09-07 14:24:54 +00:00

api

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

audio

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

call

Add reporting of googContentType via GetStats on send side

2017-09-06 19:32:35 +00:00

common_audio

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

common_video

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

examples

Make PeerConnectionClient non-singleton.

2017-08-31 15:03:46 +00:00

logging

Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread ( https://codereview.webrtc.org/3007473002/ )

2017-09-06 12:18:15 +00:00

media

Add reporting of googContentType via GetStats on send side

2017-09-06 19:32:35 +00:00

modules

Add CroppedDesktopFrameTest

2017-09-07 14:24:54 +00:00

ortc

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

p2p

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

Thread-checkers for PeerConnectionFactory::worker_thread_

2017-09-06 19:33:43 +00:00

rtc_base

Update thread annotiation macros in rtc_base to use RTC_ prefix

2017-09-06 12:46:29 +00:00

rtc_tools

Add visualization of late feedback packets, and avoid plotting incorrect delays computed based on those.

2017-09-06 09:06:32 +00:00

sdk

Improve the style of the objc wrapper for PeerConnectionInterface::SetBitrate.

2017-09-05 21:43:03 +00:00

stats

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

system_wrappers

Revert of Rename thread annotation macros to have RTC prefix for syncrhonization primitives. (patchset #1 id:1 of https://codereview.webrtc.org/3004393002/ )

2017-09-06 08:38:35 +00:00

test

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

video

Add reporting of googContentType via GetStats on send side

2017-09-06 19:32:35 +00:00

voice_engine

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

.gitignore

…

BUILD.gn

Enable UBSan float-cast-overflow warnings and fix existing ones

2017-09-05 16:49:30 +00:00

codereview.settings

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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

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

config.h

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

DEPS

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

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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

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PRESUBMIT.py

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README.chromium

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typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

webrtc.gni

Revert of Add //build/android/pylib/results/ to Android isolate packaging (patchset #2 id:40001 of https://codereview.webrtc.org/3011023002/ )

2017-09-06 19:11:27 +00:00

whitespace.txt

…

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.