webrtc_m130

History

terelius 661d94996b Only use BBRs pacer if the unit test is running BBR.

Otherwise use WebRTCs default pacer.

BUG=None

Review-Url: https://codereview.webrtc.org/3009363002
Cr-Commit-Position: refs/heads/master@{#19797}

2017-09-12 12:47:34 +00:00

api

Reland of Prepare for injectable SW decoders (patchset #1 id:1 of https://codereview.webrtc.org/3010953002/ )

2017-09-11 18:50:51 +00:00

audio

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

call

Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )

2017-09-11 09:32:16 +00:00

common_audio

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +00:00

common_video

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

examples

Make PeerConnectionClient non-singleton.

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

logging

Move StreamConfig into its own file

2017-09-11 14:25:26 +00:00

media

Simplify passing video coded factories in media engine

2017-09-12 11:42:15 +00:00

modules

Only use BBRs pacer if the unit test is running BBR.

2017-09-12 12:47:34 +00:00

ortc

Move optional.h to webrtc/api/

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

p2p

Add support for changing the TLS elliptic curve set.

2017-09-08 20:02:03 +00:00

Move SDP m= line matching from BaseChannel to WebRtcSession

2017-09-11 20:46:16 +00:00

rtc_base

Decoupling rtc_base_approved from Obj-C code

2017-09-11 10:43:34 +00:00

rtc_tools

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

sdk

Decoupling audio_device from Obj-C code

2017-09-12 11:45:24 +00:00

stats

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

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

system_wrappers

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

test

Add |RTCUIApplicationStatusObserver sharedInstance| call in ios test AppDelegate.

2017-09-11 13:43:28 +00:00

video

Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )

2017-09-11 09:32:16 +00:00

voice_engine

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

.gitignore

…

BUILD.gn

Add VideoProcessorIntegrationTest for MediaCodec implementations.

2017-09-07 14:50:06 +00:00

codereview.settings

…

common_types.cc

Add explicit copy constructors to RTPHeader and RTPHeaderExtension

2017-09-11 15:48:26 +00:00

common_types.h

Add explicit copy constructors to RTPHeader and RTPHeaderExtension

2017-09-11 15:48:26 +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

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

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PRESUBMIT.py

…

README.chromium

…

typedefs.h

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +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

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.