Add bidirectional audio test and improve SCReAM test coverage 1. Refactor SendMediaInOneDirection to SendMedia to support bidirectional media and optional adaptive ptime. 2. Add CreateNetworkPathWithVaryingCapacity helper. 3. Add AvailableSendBitrateIsZeroOrBetween matcher. 4. Add AudioOnlyBothDirectionsLinkCapacity40To90KbpsRtt200ms test. 5. Increase accepted BWE upper bound in LowBweOnLinkWith5PercentUniformLoss to avoid test flakiness. 6. Update SendVideoOnlyReturnLinkWithBurstLoss to handle temporary zero BWE when feedback packets are lost and encoder pauses. Bug: webrtc:447037083 Change-Id: I4c9613f83212422c6f012ea4eadf5a4e7b170669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/506920 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#48776}
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.