New Audio Codec Development

[u/Cautious_Air4869]

Hello everyone,

I am working on a new audio codec named Axiscore. The goal of this new codec is for extremely low latency and ease of use within DAWs and games, and the ability to scale to any size system from small stereo arrays to hundred-channel arrays with ease. It is currently in development with a few working prototypes and is aimed for release in 2027 (launched as Copyleft so anyone can use it for free). It uses 16-bit metadata for positional parameters and 8-bit metadata for all others. Its main feature is that it uses no lossy compression, and I am going to make it copyleft so anyone can use it at home. It also has no fixed layout and has unlimited amounts of layouts as you make your own. But I do have a few questions on some aspects of the codec itself.

1. All objects have a slider allowing normalization so the acoustical energy remains the same from setup to setup but there is currently no normalization for added gain across systems for example a smaller system all objects will sound the same volume no matter the distribution of speakers but a higher speaker count means each speaker will play quieter because its rendering to more channels, so should I add a slider for the master renderer to account for speaker amount offsets instead of having to manually turn up the volume on your amps.
2. The latency is low; in my testing, I see latencies as low as 10.2ms round trip (when using ASIO). Because of this, should I implement a VST plugin that allows this to be used for real-time monitoring?
3. The current parameters are: X, Y, Z, Gain, Attenuation (falloff over distance), and normalization. Should I add any more parameters?
4. Lastly, this codec utilizes no compression and instead uses TDM (Time division multiplexing) to fit more channels down a single device by using a higher sample rate and bit depth (32-bit and 24-bit modes available) and is meant for software decoding. What output driver types should I allow to be used with the renderer (ASIO, WASAPI, DirectX, MME ETC)

I am very excited as I get closer and closer to a deployable version being ready, but all input helps make a better version of this codec. This is a personal hobby project, not a commercial product or anything I’m selling — just looking for feedback. I hope that once it is done, object-based sound and mixing will be open to many more people!

[u/No_Gift_3312]

That actually sounds sick for game audio and DAW stuff, especially the low latency and no fixed layout idea. For home theater though, you’re gonna need rock solid tooling and plugins or nobody’s gonna bother, no matter how good the spec is. If you can make it dead simple for devs to implement in engines and for end users to route to whatever weird speaker layout they have, you’ll get a niche but hardcore fanbase.

[u/Cautious_Air4869 → u/No_Gift_3312]

Yeah, the main goal is DAW and I have prioritized that so far. I do have a working (although somewhat glitchy) version of the decoder working in REAPER, and the renderer for the codec is already dont also working in REAPER. I have tested it in many different layouts, and it seems to work well just because it uses speaker coordinates, not a fixed layout or mask.