DAW - ASiO and Scope/Pulsar outputs

[Music Manic]

Hi guys

I was wondering about the audio signal and what happens when it reaches the analogue output of the sound card.

Ok so DAW’s are 32 bit floating point but what happens with the different Scope ASIO modules we choose from? Obviously the scope float keeps the integrity but what happens when it is converted to analogue or it’s output to the digital outs at 16 bit?
Does scope do the conversion and how does it go about it? I must say the quality is good for analogue. Is there a special process when it’s converted to 16 bit?

Would appreciate it being clarified and if anyone uses ASIO other than the floating point ones.

Thanks

[valis]

Coming out of the DAW typically converts to fixed point (and thus clipping can occur) when using all Scope ASIO drivers aside from the 32 float ones.

To be frank, I'm not sure if the float modules now handle 64bit FLT coming into Scope. Many DAWs are now up to 64bit float internally, and there's little concern about outputting 32 float from a 64bit DAW. 32 float should be bit-transparent when using DAWs that are still 32bit float internally.

Bob Katz's book Mastering Audio is a good read, and covers in brief how floating point and fixed point audio processing occurs. But essentially when you are in a fixed point arena, keeping the input and output signal into a reasonable bounds to avoid audio discontinuities (clipping, quantization distortion from moving to low amplitude to high amplitude values and vice versa). Most of this is handled for us as the end user, and we just attend to levels in our working domain. Floating point math simplifies this for the devs (although there is still a type of quantization distortion at the extremes it's much less of a concern). Converting between the two introduces similar concerns, but in Scope all of this is handled for you again. DSP chips will handle internal calculations at many depths, and it's largely a fixed point arena between them. The AD sharc chips have some great algorithms and also have good dither for when scaling & conversions occurs, this is one of the attributes we covered in past discussions that might make Scope sound different to solutions built on Motorola & etc.

For the most part, in our uses this all comes down to gain-staging. Massive mult-mic arrays and/or extreme amounts of processing will reveal these issues but for most users mixing and doing synth work in Scope the gain staging and signal flow is similar to working on outboard gear.

There were some proponents of the floating point modules over the years, and you can dig on those threads for a lot of the information. Or I'll defer to others to correct my mistakes and point to this information.

[Spindrift]

Unlike sample rate conversion, reducing bit depth is doesn't require any algorithm. You can simply truncate the extra bits...that will give you some truncation errors that can be helped by dithering, but that you should do as one final pass, not on each channel.

As I understand, when going from float to fixed, a 32 bit float is 24 bit of actual value, and the rest is mantissa which scales the value. So you have 24 bit resolution signal which you can multiply by 8 bits. So to get the fixed 24 bit value from a 32 bit float, you can simply discard the mantissa.

So conversion between different bit depths, as well as between float and fixed, is pretty much seamless, the only thing to keep in mind then going into a fixed bit depth is that levels matter, unlike float where signal is one thing, and level of signal another.
And if going to 16 bit you should apply dithering once for the whole track during mastering.