Currently, BitrateProber does not scale higher than 2 Mbps to 6 Mbps. The actual number is dependent on the size of the last packet. If a packet of around 250 bytes is used for probing, it fails above 2 Mbps. BitrateProber now provides a recommendation on probe size instead of a packet size. PacedSender utilizes this to decide on the number of packets per probe. This enables BitrateProber to scale up-to higher bitrates. Tests with chromoting show it stalls at about 10 Mbps (perhaps due to the limitation on the simulation pipeline to deliver packets). BUG=webrtc:6332 Review-Url: https://codereview.webrtc.org/2347023002 Cr-Commit-Position: refs/heads/master@{#14503}
Revert of Only expose gflags target in non-Chromium and non-fuzzer builds. (patchset #1 id:40001 of https://codereview.webrtc.org/2321963002/ )
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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