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Harald Alvestrand c0d44d9d63 Split audio and video channels into Send and Receive APIs.
The implementation here has a number of changes that force the callers
that called the "channel" functions into specific interfaces rather than
just letting C++ take care of it; this should go away once there stops
being a common implementation class for those interfaces.

Bug: webrtc:13931
Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38888}
2022-12-14 11:00:17 +00:00
api Make requested_resolution_alignment of webrtc::EncoderInfo as uint32_t. 2022-12-12 06:02:59 +00:00
audio Remove pending packets from the pacer when an RTP module is removed. 2022-12-13 11:32:58 +00:00
build_overrides Use default values provided by PartitionAlloc instead of hard-coded ones 2022-12-07 09:11:35 +00:00
call Update WebRTC code version (2022-12-14T04:03:07). 2022-12-14 05:51:33 +00:00
common_audio Make header files self contained. 2022-10-08 08:38:36 +00:00
common_video Make header files self contained. 2022-10-08 08:38:36 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update webrtc build command instruction to autoninja. 2022-11-25 09:19:37 +00:00
examples pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Explicitly forbid dynamic_cast 2022-11-04 12:37:57 +00:00
infra Let iOS bots run unittests with XCTest. 2022-12-09 10:44:23 +00:00
logging Add missing dependencies. 2022-10-10 15:51:33 +00:00
media Split audio and video channels into Send and Receive APIs. 2022-12-14 11:00:17 +00:00
modules Don't send abs capture time when capture time unset 2022-12-13 12:37:45 +00:00
net/dcsctp Add missing dependencies. 2022-10-10 10:18:37 +00:00
p2p Improve logging of DTLS transport handshake 2022-12-14 07:41:36 +00:00
pc Split audio and video channels into Send and Receive APIs. 2022-12-14 11:00:17 +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 Make requested_resolution_alignment of webrtc::EncoderInfo as uint32_t. 2022-12-12 06:02:59 +00:00
rtc_tools Delete RGB to I420 and YUV to IVF converters 2022-12-02 08:21:21 +00:00
sdk AndroidNetworkMonitor - loosen assumptions even more 2022-12-05 10:01:01 +00:00
stats Add scalability mode to RTCOutboundRtpStreamStats stats 2022-12-08 11:46:06 +00:00
system_wrappers Rename test helper for registering field trial keys 2022-12-02 13:21:28 +00:00
test Remove webrtc::webrtc_pc_e2e::GetCurrentTestName(). 2022-12-11 09:36:52 +00:00
tools_webrtc Only instrument changed lines on coverage bot 2022-12-12 10:13:49 +00:00
video Call OnRenderedFrame after frame is passed to renderer 2022-12-13 18:18:44 +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 Fix bugs in discarding stun address 2022-11-30 13:51:41 +00:00
BUILD.gn Add fuchsia_perf_test to the default build group. 2022-12-05 10:09:00 +00:00
CODE_OF_CONDUCT.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +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 78850b7cdf..6255348f06 (1082669:1082965) 2022-12-14 08:33:04 +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
g3doc.lua Improve webrtc documentation infra. Preview at: 2021-03-30 10:29:30 +00:00
LICENSE
license_template.txt
native-api.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +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

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