These structs will be used for ORTC objects (and their WebRTC equivalents). This CL also introduces some minor changes to the existing implemented structs: - max_bitrate_bps uses rtc::Optional instead of "-1 means unset" - "mime_type" turned into "name"/"kind" (which can be used to form the MIME type string, if needed). - clock_rate and channels changed to rtc::Optional, since they will need to be for RtpSender.send(). - Renamed "channels" to "num_channels" (the ORTC name, which I prefer). BUG=webrtc:7013, webrtc:7112 Review-Url: https://codereview.webrtc.org/2651883010 Cr-Commit-Position: refs/heads/master@{#16437}
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
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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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