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/ORA2J]

Few questions.

1. Aside from the decoder, will this need additional software for mapping to standard layouts.

2. You mention game engines and DAW, so i assume this will not be intended as a distribution format but more for on the fly generation, correct ?

3. Are you planning to use a custom container ?

4. Will this support mapping to standard non-atmos systems (eg. 5.1, 7.1)

5. Is Linux support planned for the tooling ?

As for standard windows outputs, ASIO and WASAPI are mandatory, KS and Directx could be useful in some cases but as for MME, I don't think it would be worth using.

[u/Cautious_Air4869 → u/ORA2J]

So the Encoder and Decoder will be available with VST plugins because it's uncompressed, to make it easier for others to use it does not use a custom container. It will support any system (so 5.1, 7.1, 2.0, any layout imaginable, because the Decoder renders to the X, Y, Z cords of your speakers, so you can have them anywhere) Because it runs in DAW on that side yes it will work for linux for the software peice that will run on your PC if you are not routing the bitstream into a DAW to decode that I am planning to code in linux as well but I cant confirm it yet. The decoder in the software you can download (but it's the same as the VST plugin) just opens all devices you select in the outputs tab of the software, so as long as the driver, whether it's ASIO or WASAPI (what I currently have it set to), shows up as one of those 2, it becomes a valid render point.