| commit | e1405ad0d164230c77c4ac10efd92db3ef2f9030 | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Mon Jan 23 08:55:48 2017 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Jan 23 16:55:48 2017 +0000 |
| tree | 45592eaa47159537b69beeb4ed1caf95779b8428 | |
| parent | cb893ee6346f27e7e660578c802bbe455fcf9102 [diff] |
Removed double-special-casing of ISAC in libjingle and WebRtcVoE. webrtcvoiceengine.cc ensured that if the bitrate set for ISAC was 0, it was changed to -1 so that the codec could manage the bitrate itself. webrtcsdp.cc ensured that if the bitrate set for ISAC was 0, it was explicitly set to default values to avoid the codec's built in bitrate management. Eventually, there'll be no codec specific code like this in these layers. This is one step towards that goal. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2642923003 Cr-Commit-Position: refs/heads/master@{#16220}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.