| commit | 2c1b4dac57407134986c0251d823d4f9d8b782cf | [log] [tgz] |
|---|---|---|
| author | Diep Bui <diepbp@webrtc.org> | Tue Aug 23 13:42:00 2022 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Sep 29 10:24:13 2022 |
| tree | 56e2955b9b64a334c2e9c64bceae89c490ed9b01 | |
| parent | 6c2dae21e9f80cc38d10ff282117de0dfcb46243 [diff] |
Apply stricter bandwidth cap for high loss. When loss rate is above a certain threshold, set instant_limit = 500 - 1000 * average_loss_rate, which returns 200kbps at 30% loss rate, or 100kbps at 40% loss rate. When the loss rate is above 50%, use the min_bitrate from send_side_bandwidth_estimation. The high_loss_rate_threshold is set to 1.0, so the change is not activated by default. Tested the change with hamrit, when average loss rate is above 50%, bandwidth backed to 10kbps, and it took ~10s to ramp up to 1.5Mbps. https://screenshot.googleplex.com/7dvPoWa2b5SgMSL Bug: webrtc:12707 Change-Id: I5eea04ef709a183bdf696246094dbd4a204e48f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272061 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Diep Bui <diepbp@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38243}
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.
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.