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Philipp Hancke f0f435e983 Remove deprecated RTCStatsReport(int64) and timestamp_us 
BUG=webrtc:14813

Change-Id: I80c2ba8f57354ef63cf2cc7b767d1f64dd0dd766
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298444
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39633}
2023-03-22 08:00:53 +00:00
api Remove deprecated RTCStatsReport(int64) and timestamp_us 2023-03-22 08:00:53 +00:00
audio stats: use uint64_t for RTCSentRtpStreamStats.packetsSent 2023-03-16 06:46:19 +00:00
build_overrides Always check out google_benchmark, part 3. 2023-03-14 12:14:51 +00:00
call Update WebRTC code version (2023-03-22T04:02:52). 2023-03-22 05:36:24 +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 Roll chromium_revision 598cedadf7..c3980d4b97 (1118297:1119913) 2023-03-21 19:37:45 +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 Add webrtc_nonparallel_tests to Fuchsia bots. 2023-03-15 08:44:59 +00:00
logging RtcEventLog: Separate LogToMemory from TaskQueue to current thread 2023-03-21 23:07:40 +00:00
media Update SctpTransportInternal to use RTCError. 2023-03-21 13:57:47 +00:00
modules Dump codec input 2023-03-21 16:54:19 +00:00
net/dcsctp dcsctp: Make use of log_prefix consistent 2023-03-15 22:15:05 +00:00
p2p Replace use of test-only connections() with P2PTransportChannel member. 2023-02-27 16:49:05 +00:00
pc Add thread checker to SctpSidAllocator 2023-03-22 00:28:02 +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 Per default endable reading incoming packet timestamp from socket 2023-03-21 14:41:37 +00:00
rtc_tools Handling NetEqSetMinimumDelay events in neteq_rtpplay. 2023-02-09 09:39:29 +00:00
sdk Add H265 to VideoCodecMimeType 2023-03-17 15:28:11 +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 [PCLF] Introduce test video source and make it more controllable 2023-03-21 14:15:24 +00:00
tools_webrtc Noop change to trigger bots 2023-02-13 10:30:38 +00:00
video Keep SVC max bitrate if number of spatial layers are reduced. 2023-03-21 12:00:17 +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 Changed OutputToDebug to create a CFString at compile-time, rather than runtime 2023-02-19 22:41:59 +00:00
BUILD.gn Always check out google_benchmark, part 5. 2023-03-15 07:52:04 +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 9fa5a7fb6a..f35f0851b0 (1120027:1120331) 2023-03-22 03:03:20 +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
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
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 Always check out google_benchmark, part 3. 2023-03-14 12:14:51 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots 2022-11-17 21:29:53 +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