Where "TRANSPORT attributes" refers to: https://tools.ietf.org/html/draft-ietf-mmusic-sdp-mux-attributes-16 The BUNDLE draft now says that these attributes can (in fact, MUST) be omitted when m= sections are bundled (they only need to go in one of the bundled m= sections), so we should start accepting that SDP. This CL doesn't fix "a=rtcp-mux", unfortunately. That will be easier to fix once we've split apart an "RtpTransport" object from BaseChannel. BUG=webrtc:6351 Review-Url: https://codereview.webrtc.org/2647593003 Cr-Commit-Position: refs/heads/master@{#16782}
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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