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Philipp Hancke ae566cd831 audio/red: provide default fmtp line 
otherwise the generated codec won't match the preassigned codec
and red will use 96 as payload type, increasing the payload type
congestion in the upper range.

BUG=webrtc:11640

Change-Id: I466ed6d4e025ef116f3099e85855e10493408ab1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233560
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35130}
2021-10-01 13:02:30 +00:00
api ice: fix comment about relay preference 2021-09-30 07:45:19 +00:00
audio Add Direction indicator to TransformableFrames 2021-09-27 17:40:36 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Update WebRTC code version (2021-10-01T04:03:55). 2021-10-01 05:34:50 +00:00
common_audio Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
common_video Fix integer overflow in h264 pps parser 2021-09-20 11:28:36 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update Mac prerequisites 2021-08-23 19:52:17 +00:00
examples Do not use UI thread as signaling thread. 2021-09-23 14:42:39 +00:00
g3doc Update missed link to chromium C++11 styleguide 2021-09-20 21:05:39 +00:00
logging Allow encoding string fields in new event log format. 2021-10-01 12:26:10 +00:00
media audio/red: provide default fmtp line 2021-10-01 13:02:30 +00:00
modules video: Implement bandwidth based scaler 2021-09-29 10:39:27 +00:00
net/dcsctp dcsctp: Add sequence checker to socket 2021-09-30 11:25:18 +00:00
p2p Implement missing candidate pair packets/bytes sent/received stats. 2021-09-28 23:27:05 +00:00
pc Eliminate a temporary std::string in ParsedFailed helper. 2021-09-30 09:31:28 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base video: Implement bandwidth based scaler 2021-09-29 10:39:27 +00:00
rtc_tools Delete BasicPacketSocketFactory constructor with thread argument 2021-09-22 12:15:06 +00:00
sdk [ios] Fix two rtc_unittests that fail when using lld as linker 2021-09-22 19:45:44 +00:00
stats Implement missing candidate pair packets/bytes sent/received stats. 2021-09-28 23:27:05 +00:00
system_wrappers Use GTEST_SKIP() instead of early return. 2021-08-12 15:24:13 +00:00
test AGC2: Remove unused parameters 2021-09-27 11:14:35 +00:00
tools_webrtc Revert "Fix iOS sim bot to for the new Chromium roll" 2021-09-22 20:31:04 +00:00
video video: Implement bandwidth based scaler 2021-09-29 10:39:27 +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 iOS deployment target from 10 to 12. 2021-07-02 17:02:27 +00:00
.vpython Update links to point at main branch 2021-07-22 16:41:26 +00:00
.vpython3 Add .vpython3 file to webrtc 2021-09-15 16:56:30 +00:00
AUTHORS video: Implement bandwidth based scaler 2021-09-29 10:39:27 +00:00
BUILD.gn Force -Wno-shadow to avoid variable shadowing warnings. 2021-09-28 17:53:24 +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 6c17347f8a..e2c8b6114c (925841:925965) 2021-09-28 22:45:25 +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
license_template.txt
native-api.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Add .vpython3 file to webrtc 2021-09-15 16:56:30 +00:00
PATENTS
PRESUBMIT.py fix some typos 2021-08-12 18:37:10 +00:00
presubmit_test.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
presubmit_test_mocks.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +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 doc: add g3doc sitemap to toplevel readme 2021-07-23 07:55:17 +00:00
WATCHLISTS Update WATCHLISTS 2021-08-23 13:37:55 +00:00
webrtc.gni rename use_x11 to ozone_platform_x11 2021-09-20 15:13:55 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Test LKGR finder. 2021-09-24 20:09:34 +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