The intended behavior of the pacer is that it should send audio (i.e high priority) packets immediatelly even if the pacer is paused and even if the available media budget has been used. The existing code will not send audio if the pacing budget has been used. It will normally send audio when the pacer is paused, but if the media budget was already used up when the pacer was paused, it would neither send audio nor update the budget. Updated the implementatation and the unit test. Added logging for when the pacer is paused, and when it receives a zero bitrate estimate. BUG=webrtc:6155 Review-Url: https://codereview.webrtc.org/2193183003 Cr-Commit-Position: refs/heads/master@{#13592}
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
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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