Zandronum

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• zandronum-zdoom-sync
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Improved the synchronicity between the RMS volume bar and the actual volume of the client's voice during a microphone test.

Fixed the +voicerecord button not working properly in Linux.

Added CVars for adjusting the min/max rolloff distances of the proximity voice chat.

Fixed a typo where some stub member function definitions had semicolons at their ends.

Instead of not defining VOIPController at all when NO_SOUND is defined, let any necessary public member functions in the class be defined as stubs.

Give VoIP channels more priority over any other playing sounds.

When a client leaves the game, their VoIP channel is now deleted right away.

Made some changes to the microphone test bar:
- Removed parameters to the "MicTestBar" MENUDEF command that control the colors of the bar.
- A shadow of the "sensitivity" bar now appears wherever the "RMS" bar overlaps it.
- Added optimization checks to avoid drawing bars that aren't visible.

While proximity voice chat is enabled, a player is now "talking" only when their voice is audible to the listener.

VoIP audio packets that are transmitted in the same tick are now merged together using Opus's repacketizer function. This helps optimize the network traffic when sending commands to or from the server.

Significantly reduced the memory footprint of creating a sound for each VoIP channel by shortening its length from five seconds to one second.

Fixed the VOIP controller trying to start a transmission (with voice activity) while testing the microphone.

Made significant changes to how the VoIP controller and channels work:
- A VoIP channel no longer plays as a stream, but rather as a static sound in which new samples are occassionally copied into the sound's buffer. Playing too many streams at once was proving to be too unstable.
- Rather than letting a channel continuously play in the background even while muted, it stops when there's no samples left to play.
- The time when a VoIP channel stops playing is now sample-accurate and more efficient.
- Added a check so that the pitch of everyone's voices only changes if the new pitch differs from the old.

Added a template function for reading (a portion) of a sound's buffer and executing a callback function that manipulates the data.

Fixed a badly-formed initializer list.

Added a feature that lets the client test their own microphone while in the "voice chat options" menu.

Added a check to forbid the client from changing the volume of their own VoIP channel.

Added an option to change the input device in the "voice chat options" menu. It's now also possible to change the device without restarting the entire sound module.

Deleted an extra developer message that wasn't removed earlier.

Make sure to clamp the input audio samples in case they're amplified too much by snd_recordvolume.

Removed an option from the "voice chat options" menu that's no longer needed.

The VoIP controller is no longer usable during demo playback, so added a check to ensure that it can't be activated during playback.

Backed out changeset: 09ff3e4bd439
Removes: "Fixed the VoIP controller not working while playing a clientside demo."

Backed out changeset: b26ace30cb22
Removes: Added the CVar 'demo_savevoicechat' to (dis)allow VoIP audio packets to be saved into clientside demos. VoIP audio packets sent by the local player can now also be recorded into demos."

Added the CVar "snd_recordvolume" to control the volume of the recording device, up to 2.0, and moved the voice volume slider into "voice chat options".

Made it possible to amplify the voice chat volume, or that of a specific player's VoIP channel, up to 2.0.

Added "+voicerecord" to the "customize controls" menu.

Fixed CMake not finding the Opus header file automatically on Linux.

Make sure that the Opus library can be found on Unix systems.

Fixed a CMake warning.