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Ivo Creusen 1518fd34d8 Add support for setting a custom NetEqFactory in PeerConnection level tests. 
This allows running Peerconnection level tests with a custom NetEqFactory.

Bug: webrtc:11005
Change-Id: If3063cf61a6274a137e4ab74f9ec2665425f21ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161307
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30028}
2019-12-06 12:34:02 +00:00
api Add support for setting a custom NetEqFactory in PeerConnection level tests. 2019-12-06 12:34:02 +00:00
audio Minor fixes to ChannelSend. 2019-12-02 09:30:51 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Exposing more features in the network emulation manager API. 2019-12-06 08:47:19 +00:00
common_audio Add ability to disable detailed error message in RTC_CHECKs 2019-11-28 17:51:00 +00:00
common_video Revert "VideoFrame: Store a reference to an encoded frame" 2019-11-21 14:55:21 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Fixing some typos. 2019-09-10 10:03:50 +00:00
examples Fix errorprone issues preventing Chromium Roll. 2019-11-27 12:52:48 +00:00
logging Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_. 2019-11-29 09:45:50 +00:00
media Enable SSRC 0 in MediaChannel methods 2019-12-04 23:49:04 +00:00
modules Remove unnecessary checks from AudioDeviceWindowsCore::CoreAudioIsSupported 2019-12-06 10:09:03 +00:00
p2p Replace hostCandidate with address and port in RTCPeerConnectionIceErrorEvent 2019-12-04 13:18:22 +00:00
pc Fix for defect found by clusterfuzz. 2019-12-05 14:12:33 +00:00
resources Changed the digital AGC1 gain to properly support multichannel 2019-11-23 08:42:59 +00:00
rtc_base Remove line number from rtc::Location 2019-12-03 19:43:12 +00:00
rtc_tools Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_. 2019-11-29 09:45:50 +00:00
sdk Deprecate RtpReceiver's SetParameters method 2019-12-03 19:50:42 +00:00
stats Add totalInterFrameDelay to RTCInboundRTPStreamStats 2019-11-25 10:50:37 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Expose methods to validate and merge FieldTrial strings. 2019-12-06 11:15:49 +00:00
test Add support for setting a custom NetEqFactory in PeerConnection level tests. 2019-12-06 12:34:02 +00:00
tools_webrtc Revert "Remove temporary workaround for generate_licenses." 2019-11-27 17:56:45 +00:00
video Use RtpPacket instead of legacy RtpHeaderParser in video/ tests 2019-12-06 10:54:39 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Update style guide for absl::make_unique. 2019-09-18 06:10:58 +00:00
AUTHORS Implement handling for MS PowerPoint Presentation Mode. 2019-12-03 19:17:24 +00:00
BUILD.gn Stop setting -Wextra (the toolchain already does that). 2019-12-05 15:14:32 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
common_types.h Format almost everything. 2019-07-08 13:45:15 +00:00
DEPS Roll chromium_revision 3f97848513..34a43a356e (720272:722057) 2019-12-05 16:47:34 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Add #COMPONENT to WebRTC. 2019-10-08 12:20:39 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Add possibility to skip check_includes presubmit check. 2019-10-22 19:35:31 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Update WebRTC's C++ style guide to reflect the switch to C++14. 2019-09-16 11:45:35 +00:00
WATCHLISTS Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc.gni Add ability to disable detailed error message in RTC_CHECKs 2019-11-28 17:51:00 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Revert "Whitespace change" 2019-11-11 14:58:20 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info