commit | 3e61f881cd2ba9040a07371e0ba6dda902aa60ae | [log] [tgz] |
---|---|---|
author | Per Kjellander <perkj@webrtc.org> | Thu Jan 19 10:08:35 2023 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Jan 19 11:41:42 2023 |
tree | 3ef92937b74dc454905a019211b7364680726b57 | |
parent | fc5d627cef71f906e921476c2e6b1e01d07732fe [diff] |
Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio Original change's description: > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > Therefore DirectTransport is provided with the extension mapping. > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > Bug: webrtc:7135, webrtc:14795 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39137} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 Owners-Override: Björn Terelius <terelius@webrtc.org> Auto-Submit: Per Kjellander <perkj@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#39146}
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.