If a very large frame is sent (high res slide change) when the available send bitrate is very low, the it might take many seconds before any new frames are emitted as the accrued debt will take time to pay off. Add a bailout, so that if a frame hasn't been sent for 2 seconds, cancel the debt immediately, even if the target bitrate is then exceeded. BUG=webrtc:5750 Review URL: https://codereview.webrtc.org/1869003002 Cr-Commit-Position: refs/heads/master@{#12328}
Revert of CQ: Remove libfuzzer trybot from default trybot set. (patchset #1 id:1 of https://codereview.webrtc.org/1764093002/ )
Revert of Added webrtc/base/safe_conversions.h as a pseudonym (patchset #1 id:20001 of https://codereview.webrtc.org/1774933003/ )
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
Languages
C++
90.3%
Java
2.9%
C
2.2%
Objective-C++
2%
Python
1.3%
Other
1%