New beta versions of plugins and host application

There are now new beta versions of the plugins and host application available for download. These have a different communication method between the plugin and host and should work better in most situations, however, there are some limitations, like support of only 48 kHz sample rate at the moment.

The goal in the next few monts is to give users the possibility to choose the audio quality (compressed, uncompressed and transmission method (using webrtc or direct UDP connection) when playing. This allows the user to better choose the audio latency and quality depending on the situation, for example, when playing real-time or recording audio.

I have now updated the beta versions of plugins and host application. The plugin client has an option to choose from different types of audio codecs (Opus and raw audio) and different types of audio transports (webrtc and UDP). There are also unreliable and reliable transport types, the unreliable types are for the lowest latency while the reliable should not lose packets but require more latency, which means that the amount of buffers must be increased in the plugin.

I will do one more round of improvements and then I’ll update the actual plugin client and host applications.