<html> <head> <base href="https://bugs.freedesktop.org/"> </head> <body> <div> <a class="bz_bug_link bz_status_NEW " title="NEW - Implement opus audio compression" href="https://bugs.freedesktop.org/show_bug.cgi?id=56993#c10">Comment # 10</a> on <a class="bz_bug_link bz_status_NEW " title="NEW - Implement opus audio compression" href="https://bugs.freedesktop.org/show_bug.cgi?id=56993">bug 56993</a> from <a class="email" href="mailto:tanuk@iki.fi" title="Tanu Kaskinen <tanuk@iki.fi>"> Tanu Kaskinen</a> <pre>(In reply to Arun Raghavan from <a href="show_bug.cgi?id=56993#c6">comment #6</a>) > I'm quite against the idea of having codec support in PulseAudio itself. > > In my opinion, the right way to do this is to first move our RTP support to > use GStreamer under the hood, and then potentially use that to do encoding > if needed. The RTP modules are not useful when talking about a tunnel setup or a direct client-server connection over TCP. Can you clarify, are you against any compressed audio implementation in the native protocol, and if yes, why exactly? There's a new version of the opus patch, and I thought I'd start reviewing it: <a href="https://patchwork.freedesktop.org/patch/169038/">https://patchwork.freedesktop.org/patch/169038/</a></pre> </div> <hr> You are receiving this mail because: <ul> <li>You are the QA Contact for the bug.</li> <li>You are the assignee for the bug.</li> </ul> </body> </html>