signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Oleh Prypin eec86cdd96 Fix platform-specific header dependencies to be more precise 
Make the GN conditionals match what happens in sources, or the other way around. Include headers only when they're used.

Bug: None
Change-Id: Ib8e3346e3c24eaa7e61ac4776dcd66efe2cc5c65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28500}
2019-07-08 11:21:30 +00:00
api Remove deprecation notice for VideoFrame::ntp_time_ms 2019-07-05 18:37:41 +00:00
audio Introduce PC level audio quality test. 2019-07-05 11:41:38 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call [getStats] Implement "media-source" audio levels, fixing Chrome bug. 2019-07-04 08:13:45 +00:00
common_audio Initialize signal processing function pointers statically 2019-06-28 14:20:03 +00:00
common_video Extend structures to store updated version of the dependency descriptor 2019-06-27 12:10:38 +00:00
crypto Adding new top-level directory crypto/ 2019-03-08 00:35:05 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Fix platform-specific header dependencies to be more precise 2019-07-08 11:21:30 +00:00
logging Delete RtcEventLogFactory factory as now unused 2019-06-27 10:03:22 +00:00
media Remove trial WebRTC-SimulcastMaxLayers and make its behavior default 2019-07-05 14:55:46 +00:00
modules Fix platform-specific header dependencies to be more precise 2019-07-08 11:21:30 +00:00
p2p Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly 2019-06-28 19:12:14 +00:00
pc Reduce kMaxSimulcastStreams to 3 2019-07-04 15:54:36 +00:00
resources Cleanup of resources from removed remote bitrate estimate test framework. 2019-06-18 10:22:01 +00:00
rtc_base Make rtc_base/base_java public. 2019-07-03 14:44:46 +00:00
rtc_tools Switch unpack_aecdump to ABSL_FLAG. 2019-07-04 14:19:40 +00:00
sdk Android: Add error callback for GL_OUT_OF_MEMORY in EglRenderer 2019-07-05 13:04:17 +00:00
stats [getStats] Implement "media-source" audio levels, fixing Chrome bug. 2019-07-04 08:13:45 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Sanity-check field trial string at initialization. 2019-06-11 14:11:06 +00:00
test Fix platform-specific header dependencies to be more precise 2019-07-08 11:21:30 +00:00
tools_webrtc Remove android_tools deps 2019-07-02 06:48:30 +00:00
video Disable VP9 2nd profile test for ios arm64 2019-07-03 14:48:16 +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 format commit. 2018-06-20 09:26:44 +00:00
.gitignore Add Visual Studio Code project folder to gitignore file. 2019-01-21 18:42:33 +00:00
.gn Remove last mention of ortc from the codebase. 2019-05-25 07:28:05 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
AUTHORS Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly 2019-06-28 19:12:14 +00:00
BUILD.gn Move network emulation framework under test/network 2019-07-05 11:08:42 +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 Move kRtpCsrcSize from common_types.h to rtp_headers.h 2019-05-10 09:43:39 +00:00
DEPS Roll chromium_revision 48f88caf2e..619b1dc518 (674992:675093) 2019-07-05 22:46:33 +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 juberti@ to webrtc root owners 2019-05-17 18:11:58 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
PRESUBMIT.py Convert file objects to strings, before passing to PresubmitNotifyResult 2019-06-26 11:31:21 +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 Remove rule that discourages passing optional by const reference 2019-02-05 11:58:05 +00:00
WATCHLISTS Remove myself from OWNERS in a few places. 2019-06-10 07:57:46 +00:00
webrtc.gni Specify min_sdk_version for unittest apks also in GN configs 2019-06-26 07:18:15 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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