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Sergey Silkin 8566e779e3 Add samples sum calculation 
Bug: b/261160916, webrtc:14852
Change-Id: I88e464fce4673dd9b9683219b8d2837206579ba5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293942
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39386}
2023-02-24 11:48:39 +00:00
api Add samples sum calculation 2023-02-24 11:48:39 +00:00
audio Create unit test for the population of capture_start_ntp_time 2023-02-06 14:00:39 +00:00
build_overrides Use default values provided by PartitionAlloc instead of hard-coded ones 2022-12-07 09:11:35 +00:00
call Reland "Make SimulcastIndex() and SpatialIndex() distinct (remove fallback)." 2023-02-21 18:30:35 +00:00
common_audio Check FMA3 support before use it in SincResampler 2023-01-31 17:28:55 +00:00
common_video Add 444 10 bits support for H264 and VP9 2023-01-17 12:32:26 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update documentation links in docs/native-code/development/index.md 2023-02-20 16:35:23 +00:00
examples Remove all usage of //rtc_base target 2023-01-16 14:36:06 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Fix doc path 2023-01-31 10:14:47 +00:00
infra Remove use of refs/heads/master mirror in WebRTC infra. 2023-02-21 09:24:41 +00:00
logging RtcEventLogImpl: Add test cases 2023-02-03 09:55:33 +00:00
media Separate last_stats_log_ms_ for send and receive stats. 2023-02-23 16:13:27 +00:00
modules Switch WGC to ScreenCaptureFrameQueue 2023-02-24 09:39:00 +00:00
net/dcsctp Remove all usage of //rtc_base target 2023-01-16 14:36:06 +00:00
p2p Add an ICE switch reason for a switch requested by an application. 2023-02-06 16:19:49 +00:00
pc Add plumbing for video NACK to be coupled between channels. 2023-02-22 14:54:38 +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 Add samples sum calculation 2023-02-24 11:48:39 +00:00
rtc_tools Handling NetEqSetMinimumDelay events in neteq_rtpplay. 2023-02-09 09:39:29 +00:00
sdk Revert "Remove ISAC media constant and payload type mapping" 2023-02-23 15:00:38 +00:00
stats stats: Deprecate RTCStatsReport(int64 timestamp_us) 2023-02-22 12:32:02 +00:00
system_wrappers Check FMA3 support before use it in SincResampler 2023-01-31 17:28:55 +00:00
test Allow setting the network labels in NetworkQualityMetricsReporter 2023-02-21 20:01:54 +00:00
tools_webrtc Noop change to trigger bots 2023-02-13 10:30:38 +00:00
video Introduce capture_time_identifier in webrtc::EncodedImage 2023-02-22 17:08:53 +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 Set Fuchsia Api level + update SDK version 2022-09-14 08:49:56 +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 Delete unused Audio Bwe integration test. 2023-01-26 09:31:44 +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 bf9471517a..34f1d79720 (1109037:1109485) 2023-02-24 10:55:17 +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 Verify field trials looked up through field_trial::FindFullName 2022-10-20 10:46:01 +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