signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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C++ 91.4%
Java 2.8%
Objective-C++ 2%
C 1.4%
Python 1.2%
Other 1%
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Jason Long 577fc0c395 Moved Asynchronicity From Java to C++ for AndroidVoip Demo App 
Moved asynchronicity from Java to C++.

Bug: webrtc:11723
Change-Id: I985693dc7d4312b6072314088716167b9cdd9999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180774
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31958}
2020-08-18 17:54:56 +00:00
api Deprecate RtpTransceiver.setDirection() 2020-08-17 13:30:30 +00:00
audio Revert "Call OnReceivedOverhead after audio network adaptor is created." 2020-08-17 14:30:29 +00:00
build_overrides set perfetto flag to default value of false 2020-07-22 10:14:53 +00:00
call Revert "[Adaptation] Remove QualityScalerResource when disabled." 2020-08-17 13:46:30 +00:00
common_audio Reland "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-08-17 10:40:44 +00:00
common_video Move function PrintVideoFrame to the test file where it is used. 2020-08-06 15:20:19 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Updated IOS documentation with correct build_ios_libs.py path 2020-08-17 09:37:59 +00:00
examples Moved Asynchronicity From Java to C++ for AndroidVoip Demo App 2020-08-18 17:54:56 +00:00
logging Add missing tests for DTLS state logging in RTC event log. 2020-08-07 11:49:43 +00:00
media Remove RTPFragmentationHeader creation and propagation through webrtc 2020-08-17 16:37:33 +00:00
modules Reland "Add ability to state whether the APM output will be used" 2020-08-18 12:29:32 +00:00
p2p Delete left-over TODO comment 2020-08-07 11:36:51 +00:00
pc Make Objective-C interface use SetDirectionWithError 2020-08-17 10:01:49 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Removing NetworkMonitorFactory static global. 2020-08-13 20:50:23 +00:00
rtc_tools Use non-deprecated EncodedImageCallback::OnEncodedImage in rtc_tools 2020-08-12 09:17:15 +00:00
sdk Modify Android API to use SetDirectionWithError 2020-08-17 11:55:55 +00:00
stats Reland "Implement packets_(sent | received) for RTCTransportStats" 2020-07-10 11:50:59 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Reland "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-08-17 10:40:44 +00:00
test Remove RTPFragmentationHeader creation and propagation through webrtc 2020-08-17 16:37:33 +00:00
tools_webrtc Fix clang revision regexp in chromium autoroller to match new format 2020-08-07 09:22:03 +00:00
video Remove RTPFragmentationHeader creation and propagation through webrtc 2020-08-17 16:37:33 +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 Reenable libaom decoder by default 2020-03-18 18:04:41 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Use absl_deps in order to preapre to the Abseil component build release. 2020-06-08 12:59:40 +00:00
AUTHORS Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder 2020-08-05 09:41:49 +00:00
BUILD.gn Reland "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-08-17 10:40:44 +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 definition of SpatialLayer to api/video_codecs/spatial_layer.h 2020-08-17 09:45:19 +00:00
DEPS Roll chromium_revision 2ecce21d76..a0c9fe7f59 (798579:798709) 2020-08-17 18:52:47 +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
native-api.md Make transient suppression optionally excludable via defines 2020-04-02 11:44:07 +00:00
OWNERS Remove phoglund as root owner. 2020-03-30 12:15:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +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.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md Fix link in documentation. (take 2) 2020-04-16 11:08:43 +00:00
style-guide.md C++ style: We don't allow designated initializers 2020-06-03 09:11:09 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +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
webrtc.gni Reland "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-08-17 10:40:44 +00:00
whitespace.txt Trigger CI bots. 2020-07-15 17:50:55 +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 here 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