[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
365381fdf1
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867} |
||
|---|---|---|
| api | ||
| audio | ||
| build_overrides | ||
| call | ||
| common_audio | ||
| common_video | ||
| data | ||
| examples | ||
| infra | ||
| logging | ||
| media | ||
| modules | ||
| ortc | ||
| p2p | ||
| pc | ||
| resources | ||
| rtc_base | ||
| rtc_tools | ||
| sdk | ||
| stats | ||
| style-guide | ||
| system_wrappers | ||
| test | ||
| tools_webrtc | ||
| video | ||
| .clang-format | ||
| .git-blame-ignore-revs | ||
| .gitignore | ||
| .gn | ||
| .vpython | ||
| AUTHORS | ||
| BUILD.gn | ||
| CODE_OF_CONDUCT.md | ||
| codereview.settings | ||
| common_types.cc | ||
| common_types.h | ||
| DEPS | ||
| LICENSE | ||
| license_template.txt | ||
| LICENSE_THIRD_PARTY | ||
| native-api.md | ||
| OWNERS | ||
| PATENTS | ||
| presubmit_test_mocks.py | ||
| presubmit_test.py | ||
| PRESUBMIT.py | ||
| pylintrc | ||
| README.chromium | ||
| README.md | ||
| style-guide.md | ||
| typedefs.h | ||
| WATCHLISTS | ||
| webrtc.gni | ||
| 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 http://www.webrtc.org/native-code/development 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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct