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Jeremy Leconte 47701c8c9b Chromium LocalRobolectricTestRunner has been removed. 
This is a follow up on https://chromium-review.googlesource.com/c/chromium/src/+/44503.

Change-Id: I28a0789a0af43cfac27081c9b5bcf695e9798910
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303020
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39933}
2023-04-24 11:26:11 +00:00
api [DVQA] Add a GetSenderPeerName method. 2023-04-17 13:05:33 +00:00
audio Extract TestADM into a separate target 2023-04-20 10:45:37 +00:00
build_overrides Always check out google_benchmark, part 3. 2023-03-14 12:14:51 +00:00
call Update WebRTC code version (2023-04-24T04:05:22). 2023-04-24 06:02:33 +00:00
common_audio Check FMA3 support before use it in SincResampler 2023-01-31 17:28:55 +00:00
common_video Changed sps parser and sps parser unit test case for h264, and it is working 2023-03-14 12:15:54 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs doc: rename index.md to README.md 2023-03-13 13:16:22 +00:00
examples Chromium LocalRobolectricTestRunner has been removed. 2023-04-24 11:26:11 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Add docs about adding a new test binary. 2023-03-07 11:12:33 +00:00
infra iOS64 Perf bot build15-a7 is replaced by mac-254-e504. 2023-04-24 07:44:12 +00:00
logging Use DD encoder/decoder in RTC event log encoder/parser. 2023-04-24 10:35:22 +00:00
media Use GlobalSimulatedTimeController in more webrtc video engine unittests 2023-04-21 14:41:27 +00:00
modules Add slightly more constness to SourceFrame and the embedded AudioFrame 2023-04-24 10:05:35 +00:00
net/dcsctp dcsctp: Add handover state for zero checksum 2023-04-24 10:06:40 +00:00
p2p Default enable WebRTC-IPv6NetworkResolutionFixes 2023-04-17 20:28:53 +00:00
pc Call PrepareShutdown in the dtor just in case Close() hasn't been called 2023-04-24 11:06:42 +00:00
resources Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00
rtc_base Format /rtc_base 2023-04-21 06:17:42 +00:00
rtc_tools Delete ReportBlock::cumulative_lost_signed accessor 2023-04-20 10:39:37 +00:00
sdk Enable RTC mode in Google HW AV1 encoder 2023-04-21 16:38:06 +00:00
stats Remove deprecated RTCStatsReport(int64) and timestamp_us 2023-03-22 08:00:53 +00:00
system_wrappers Add option to log a warning for unregistered field trials 2023-02-28 15:43:18 +00:00
test Extract TestADM into a separate target 2023-04-20 10:45:37 +00:00
tools_webrtc Noop change to trigger bots 2023-02-13 10:30:38 +00:00
video Add AV1 perf tests. 2023-04-24 10:45:15 +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 Increase android32_ndk_api_level to 21. 2023-03-13 12:37:57 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Fix mb.py presubmit issues. 2021-12-08 08:53:00 +00:00
.vpython Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
.vpython3 Add python grpc to .vpython3 for ios test runner 2022-09-16 12:26:48 +00:00
AUTHORS Video: add new metric for VP9/AV1 hw encoding with softwareBRC 2023-04-20 12:54:06 +00:00
BUILD.gn Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +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
DEPS Roll chromium_revision 2598ff6c3a..1cc6fa230c (1133653:1133778) 2023-04-21 14:45:42 +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 phoglund from ENG_REVIEW_OWNERS 2021-10-08 08:29:42 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Add infra owners file 2022-12-02 09:21:47 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Update portaudio to the latest 2022-05-13 09:01:34 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
pylintrc tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md doc: add g3doc sitemap to toplevel readme 2021-07-23 07:55:17 +00:00
WATCHLISTS Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc.gni Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
webrtc_lib_link_test.cc Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
whitespace.txt Trigger builds 2023-04-12 07:09:41 +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