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Doudou Kisabaka ae0d117d51 Implement the mixer-to-client per CSRC audio level extension (RFC 6465). 
This is loosely based on the similar implementation in gecko.

Bug: webrtc:9965
Change-Id: I5203a05e1c34ca6f97bd1b143790f95ff245e340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219791
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Cr-Commit-Position: refs/heads/master@{#34102}
2021-05-24 14:11:28 +00:00
api Implement the mixer-to-client per CSRC audio level extension (RFC 6465). 2021-05-24 14:11:28 +00:00
audio Make local to capturer clock offset a separate entry in PacketInfo. 2021-05-20 13:42:57 +00:00
build_overrides Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492) 2021-04-12 18:25:58 +00:00
call Update WebRTC code version (2021-05-24T04:02:02). 2021-05-24 04:59:35 +00:00
common_audio Avoid undefined behavior in a division operation. 2021-04-23 07:49:24 +00:00
common_video Update BitBuffer methods to style guide 2021-05-18 11:10:27 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Create a VideoFrameTrackingId RTP header extension. 2021-03-25 17:25:18 +00:00
examples Replace legacy getStats with standard getStats in the iOS example 2021-05-20 08:40:11 +00:00
g3doc Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
logging Revert "Deprecate microsecond timestamps in RTC event log." 2021-05-24 13:11:10 +00:00
media Revert "Fix race between enabled() and set_enabled() in VideoTrack." 2021-05-24 14:06:19 +00:00
modules Implement the mixer-to-client per CSRC audio level extension (RFC 6465). 2021-05-24 14:11:28 +00:00
net/dcsctp dcsctp: Implement Round Robin scheduler 2021-05-23 17:49:52 +00:00
p2p Remove cricket::DtlsTransportState. 2021-05-21 21:45:29 +00:00
pc Revert "Fix race between enabled() and set_enabled() in VideoTrack." 2021-05-24 14:06:19 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Update BitBuffer methods to style guide 2021-05-18 11:10:27 +00:00
rtc_tools Convert to NTP time using the real clock. 2021-05-21 19:29:32 +00:00
sdk Set nativeObserver to 0 to avoid double release. 2021-05-17 19:53:43 +00:00
stats Simplify reference counting implementation of PendingTaskSafetyFlag. 2021-04-21 07:04:01 +00:00
system_wrappers Make Clock::ConvertTimestampToNtpTime pure virtual 2021-05-21 09:55:14 +00:00
test Implement the mixer-to-client per CSRC audio level extension (RFC 6465). 2021-05-24 14:11:28 +00:00
tools_webrtc crc32c: Point the licensing script to the LICENSE file 2021-05-03 16:46:30 +00:00
video Make local to capturer clock offset a separate entry in PacketInfo. 2021-05-20 13:42:57 +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 Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +00:00
.vpython Update six library version 2021-04-26 16:39:07 +00:00
AUTHORS Revert "Ensure method which updates UI is called in main thread" 2021-04-30 09:26:03 +00:00
BUILD.gn Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +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 70eb2d0977..2826799ea1 (885736:885837) 2021-05-23 20:23:33 +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
g3doc.lua Improve webrtc documentation infra. Preview at: 2021-03-30 10:29:30 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Make the remote_bitrate_estimator build target private 2020-11-26 12:21:22 +00:00
OWNERS Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py sctp: Rename SctpTransport to UsrSctpTransport 2021-04-12 10:40:34 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +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 Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc.gni Turn on the RTC_ENABLE_WIN_WGC build flag. 2021-05-10 20:16:52 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +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