| commit | 66753c34ad91d82919cefab13c08c28492c0a1a1 | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Thu Apr 27 08:47:13 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Apr 27 08:47:13 2017 |
| tree | c31468f030929409f006834af92594b13b2bb924 | |
| parent | 15f0c6815d089640230e625a4a9faa53b7dd7f62 [diff] |
Normalize codec names to those used by AcmCodecDatabase. With the upcoming changes to WebRtcVoiceEngine due to injectable audio encoders, codec names will no longer be normalized against the ACM's codec database. This not only breaks a couple of downstream tests (which I've written fixes for) but also runs the risk of breaking other external usage, like SDP mangling etc. Eventually, we should change to using the spelling from the relevant RFCs, although not as part of this other, large change. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2845603002 Cr-Commit-Position: refs/heads/master@{#17903}
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.