This method will be called when PacedSender is using the new code path that directly owns the packets to be sent. It can be seen as combining a few features of the old code path: * It checks if this is the correct RTP module and then sends, without the need for PacketRouter to poll multiple methods for SSRC etc first. * It partly corresponds to TimeToSendPacket(), but RTX encapsulation now happens pre-pacer and FEC does not need to have a packet history, so most of that method is not used. * It implements most of PrepareAndSendPacket(), such as updating header extensions, reporting stats and of course forwards to Transport. It now also handles the history a bit differently, since media packets will only be stored for potential retransmission post-pacer. Bug: webrtc:10633 Change-Id: Ie97952eeef6e56e462e115d67f7c7929f36c1817 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142165 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28298}
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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