fd58b61068
Instead of having to specify a bitrate and how many packets to use, the BitrateProber will now use the bitrate to calculate how many bytes it will use to probe that bitrate instead. For now, |kMinProbeDurationMs| is set to 15 ms which means that probing at 1900 kbps will result in 1900/8*0.015 = 3.5 kB used. Since we can expect packets to be smaller at the beginning of a stream (500 to 700 bytes) this will result in 7 to 5 packets sent for that bitrate, and should work very similar to how the current initial probing works. A minimum of 5 packets are always sent. BUG=webrtc:6822 Review-Url: https://codereview.webrtc.org/2609113003 Cr-Commit-Position: refs/heads/master@{#15899}
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