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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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philipel ca18809ee5 Move RtpFrameObject and EncodedFrame out of video_coding namespace. 
Bug: webrtc:12579
Change-Id: Ib7ecd624eb5c54abb77fe08440a014aa1e963865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33542}
2021-03-23 14:22:47 +00:00
api Move RtpFrameObject and EncodedFrame out of video_coding namespace. 2021-03-23 14:22:47 +00:00
audio Fix timestamps for the remote outbound audio stream stats 2021-03-23 10:53:59 +00:00
build_overrides Allow webrtc mac cross compile 2021-03-10 18:42:58 +00:00
call Update WebRTC code version (2021-03-23T04:03:37). 2021-03-23 06:02:50 +00:00
common_audio Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
common_video Provide a default implementation of NV12BufferInterface::CropAndScale. 2021-03-22 11:09:36 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Allow empty video layer allocation extension 2021-03-23 12:36:59 +00:00
examples Expose addIceCandidate with completion handler. 2021-03-11 16:15:44 +00:00
g3doc Introduce WebRTC documentation structure and how-to 2021-03-12 12:07:52 +00:00
logging Replace RTC_DEPRECATED with ABSL_DEPRECATED 2021-02-22 12:53:23 +00:00
media Change default adaptive ptime min bitrate to 16kbps. 2021-03-23 11:31:49 +00:00
modules Move RtpFrameObject and EncodedFrame out of video_coding namespace. 2021-03-23 14:22:47 +00:00
p2p Delete AsyncInvoker usage in TurnServer 2021-03-23 08:01:00 +00:00
pc stats: add address as alias for ip 2021-03-23 06:29:10 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Check if ifa_addr field is null. 2021-03-22 09:41:27 +00:00
rtc_tools Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium" 2021-02-25 10:48:55 +00:00
sdk Make GL errors thrown by checkNoGLES2Error inherit GLException. 2021-03-23 11:48:19 +00:00
stats stats: add address as alias for ip 2021-03-23 06:29:10 +00:00
style-guide Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
system_wrappers Don't use SystemTimeNanos() for current wallclock time on WINUWP 2021-02-23 14:53:22 +00:00
test Move RtpFrameObject and EncodedFrame out of video_coding namespace. 2021-03-23 14:22:47 +00:00
tools_webrtc Update apply-iwyu tool to report compile errors 2021-03-22 13:01:56 +00:00
video Move RtpFrameObject and EncodedFrame out of video_coding namespace. 2021-03-23 14:22:47 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h 2020-09-07 08:37:14 +00:00
.vpython Reland "Add protobuf-py2_py3 3.13.0 to .vpython." 2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md Polish the "Using Abseil in WebRTC" docs 2020-10-16 13:42:00 +00:00
AUTHORS Roll src/third_party/libjpeg_turbo/ fa0de0767..7b4981b65 (2 commits) 2021-02-24 08:40:13 +00:00
BUILD.gn Move SequenceChecker header to API: step 1, move header only 2021-02-08 11:49:58 +00:00
CODE_OF_CONDUCT.md
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
DEPS Roll chromium_revision b7f0a0c111..c0436807ae (865116:865247) 2021-03-22 18:57:43 +00:00
DIR_METADATA Move metadata in OWNERS files to DIR_METADATA files. 2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Make the remote_bitrate_estimator build target private 2020-11-26 12:21:22 +00:00
OWNERS Move metadata in OWNERS files to DIR_METADATA files. 2021-02-08 19:09:33 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
PRESUBMIT.py Add deprecation section to webrtc style guide 2021-02-22 13:34:40 +00:00
pylintrc Undo enforcing of PEP-8 pylint changes for method and function names. 2020-11-10 18:26:25 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md doc: move bug reporting instructions to the repository 2020-10-21 14:47:49 +00:00
style-guide.md Add deprecation section to webrtc style guide 2021-02-22 13:34:40 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +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 build: improve rtc_include_tests documentation 2021-03-05 13:54:20 +00:00
whitespace.txt Reland "Triggering CI." 2021-03-22 11:57:23 +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