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d440358cca
The feauture was added in https://webrtc-review.googlesource.com/c/src/+/291119 in order to ensure RTX packet is part of BWE even before the RTP stream is known. However, it cause an issue if media is signaled with an SSRC that has this RTX SSRC. Since BWE is now notified about received packets before demuxing to the correct receive stream, it is not necessary to demux RTX packets before the media SSRC is known. Note that WebRTC require at least one negotiated SSRC/MID before RTCP feedback can be sent. Ie, for BWE to work, at least one media SSRC must be known after this cl. It can either be unsignaled or signaled. BWE tested with BweRampupWithInitialProbeTest. Bug: webrtc:14795, webrtc:14817, b/320258158 Change-Id: Icf2c67bedc352720bf846b9ee38d509346af36f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340141 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41785} |
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| api | ||
| audio | ||
| build_overrides | ||
| call | ||
| common_audio | ||
| common_video | ||
| data | ||
| docs | ||
| examples | ||
| experiments | ||
| g3doc | ||
| infra | ||
| logging | ||
| media | ||
| modules | ||
| net/dcsctp | ||
| p2p | ||
| pc | ||
| resources | ||
| rtc_base | ||
| rtc_tools | ||
| sdk | ||
| stats | ||
| system_wrappers | ||
| test | ||
| tools_webrtc | ||
| video | ||
| .clang-format | ||
| .git-blame-ignore-revs | ||
| .gitignore | ||
| .gn | ||
| .mailmap | ||
| .style.yapf | ||
| .vpython3 | ||
| AUTHORS | ||
| BUILD.gn | ||
| CODE_OF_CONDUCT.md | ||
| codereview.settings | ||
| DEPS | ||
| DIR_METADATA | ||
| ENG_REVIEW_OWNERS | ||
| LICENSE | ||
| license_template.txt | ||
| native-api.md | ||
| OWNERS | ||
| OWNERS_INFRA | ||
| PATENTS | ||
| PRESUBMIT.py | ||
| presubmit_test.py | ||
| presubmit_test_mocks.py | ||
| pylintrc | ||
| pylintrc_old_style | ||
| README.chromium | ||
| README.md | ||
| WATCHLISTS | ||
| webrtc.gni | ||
| webrtc_lib_link_test.cc | ||
| whitespace.txt | ||
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation