| commit | 7a12b5ad8eafec0b85e7e474e094d34141d14a56 | [log] [tgz] |
|---|---|---|
| author | kwiberg <kwiberg@webrtc.org> | Thu Apr 27 10:55:57 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Apr 27 10:55:57 2017 |
| tree | db40e4d59b0d10cee3e51ccb59e042891503b280 | |
| parent | 0f80a7a5a6d3a03efec40b1a91cf3eb1b82c642e [diff] |
Run some peer connection end-to-end tests with an empty audio decoder factory Specifically, the tests that only use data channels shouldn't need any audio codec support; by using an audio decoder factory that supports no codecs, we ensure that this is the case. For completeness, I tried doing the same to the two tests that actually use audio and video; as expected, they fail, with messages like this: [000:032] (webrtcsession.cc:334): Failed to set remote sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set local audio description recv parameters.. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2848563002 Cr-Commit-Position: refs/heads/master@{#17907}
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.