Move the voice pipelines into core

Capture processing, the jitter buffer and decoding were desktop classes,
so any other platform would have had to copy them. They now live in core
and reach the platform only through two interfaces:

- AudioIo lists devices and opens capture/playback lines; the desktop's
  AudioDevices implements it over PipeWire and Java Sound.
- OpusCodec creates encoders and decoders; the desktop binds libopus
  through the FFM API as before.

DesktopVoiceInput becomes CaptureVoiceInput unchanged in behaviour.
DesktopVoiceOutput splits into VoiceStream, one speaker's jitter buffer
and decoder, paced by whoever pulls it, and StreamingVoiceOutput around
it. Speakers reach the device either on a line each (the desktop, so
each shows up in the PipeWire mixer) or mixed onto one shared line, as
mobile audio APIs want.

The settings dialog's device lists and microphone test now go through
the AudioBackend instead of desktop classes.

Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>

ts3-client/swing/src/main/java/com/ts3client/ui/MainFrame.java

@@ -834,7 +834,7 @@ public final class MainFrame extends JFrame implements ServerTabPane.Listener {
     private void showSettings() {
         // Device and voice-gating changes apply to the capturing connection and the
         // playback of the visible one; the rest pick the new settings up on connect.
-        SettingsDialog dlg = new SettingsDialog(this, settings,
+        SettingsDialog dlg = new SettingsDialog(this, settings, audio,
                 micTab == null ? null : micTab.connection().getMicrophone(),
                 selected == null ? null : selected.connection().getPlayback(),
                 sounds, hotkeys,