commit | 5894b6aad4b107418cfa34482a220ecc5f516c0c | [log] [tgz] |
---|---|---|
author | Rasmus Brandt <brandtr@webrtc.org> | Thu Jun 13 14:28:14 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Jun 13 15:45:00 2019 |
tree | ba321d2765375b3903ebb698485a1134bf50440b | |
parent | 5740afa0a453474ccb49470a42ec9b7f8ee4e128 [diff] |
Add kPayloadTypeGeneric to CallTest and use it in VideoQualityTest. - Add CallTest::kPayloadTypeGeneric - Allow for unrecognized codec names in VideoQualityTest. Generic packetization is implicitly enabled for these codecs. Tested: autoninja -C out/Debug && out/Debug/video_loopback Bug: webrtc:10738 Change-Id: I57001be997db2f0eed9197eb40801b5ad936d222 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141864 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28270}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.