Stereo Tool

Would this be useful?

A secondary input with limited functionality and lower quality processing, which has a very low latency. For example only noise filter + pre limiter + limiter + a single Loudness step.

Basically I could create 2 processing paths, and AFTER doing the current Loudness processing add this new signal at the desired volume. Then after that perform a second "Final Limiter" step.

This would increase the CPU load and the latency of the rest of the sound a bit.

This is possible because MOST of the latency in the processing is needed to avoid distortion caused by very low frequencies. For voices, that's not really an issue.


This would give:
- NORMAL music with the same latency that you have now (slightly more)
- A secondary input for sounds that don't require high quality processing such as voices etc.

When starting a new song the response would still be very slow, but the latency for voices would be below 100 ms, while the quality of the rest of the sound would not degrade.

Another MIC input and different processing ... Great idea !

I assume that latency calculation (100ms) when is ASIO used !? Is it possible to be less? I do quick test with "breakaway asio" and i get about 20ms latency on output.
If we speak about Voice (mic) processing, i think loudness is not so needed.
How will it be when i have loudness included on mic and master goes again to ST which again have another loudness ?

I'll do tomorow quick test with mic processing only and ST, but i think needed processes is noise reduction, multiband, sb compressor, and Final limiter.

Hi Bojcha,

Without ANY Loudness-like function, the microphone input would be way too soft if music is playing.

I've done a quick test, and a simplified (lower latency) Final Limiter combined with a singleband compressor (!) would suffice.

So:
Microphone signal -> pre limiter -> 5-bands multiband compressor -> add to original signal with peak level around 3* that of the original signal -> Singleband compressor which kicks in if the signal gets above ~ 150% of the original peak level -> simplified Final Limiter (corresponds to a loudness of 2) -> Hard Limit.

That's based on normal processing with a loudness of 3 - which has the extra advantage that voices now get processed at 2/3rds the Loudness of the rest of the sound.

This has almost no effect on the normal audio, and (see the Low Latency processing) can be done with very little latency.

I hope I can get it below 100 ms, but I'll have to test that. ASIO latency itself in Stereo Tool is currently near 20 ms. But any processing that is done (except for the singleband compressor) adds a bit.


I already have the 2nd sound card working here, now all I need to do is create a secondary processing path. So I hope to get something working quickly.

By the way, I suppose the Breakaway latency is with all the processing disabled?

Based on the current low latency processing, I would get an extra processing delay of 102 ms (20 + 102 = 122 ms total latency).

8 ms for the noise gate
8 ms for dualband processing
6 ms for removing remaining loud peaks in the dualband signal
16 ms for multiband processing
64 ms for the final limiter

BUT, even low latency processing needs to process low bass sounds properly, and for voices that's not necessary. So, several of these can be reduced a bit (I think I can divide the multiband by 2, and the final limiter too. That would give a total latency that's 40 ms lower, 82 ms total).

NOTE: This is the latency UP TO the moment where the MPX signal is generated. That also takes some time, I've never measured how much though. (Just checked, seems to be another 50 ms. Can be lowered though).

I get it, but, i was not planing to use this in that way.
I already have mic on mixer without any processing .. only master signal from mixer is processed with ST on transmitter site.
So i was thinking just to add another PC with good asio soundcard only for microphone.
Anyway with current setup i don't know how can i use this MIC input with current ST on transmitter site. ? No, actually i was trying with "Breakaway Live". with "ultra low latency mode" enabled (but lower quality) overall latency is ~20ms.

Really, 20!? Wow, then I have some work to do here also...

I've had some ideas for a while already to create a 0-latency pre limiter and a 0-latency multiband compressor (without or with much lower quality clippers though), that would also use far less CPU processing power. I guess it's time to start working on that!

The Loudness and Final Limiter filters will never have 0-latency, but if you can live with some bass distortion I can make it very low.

Reported latency for BL is 17ms.

I just tested again with E-MU 0202 and BL is really fast! But i can't get ST to work with this soundcard in asio for some curently unknown reason!
Only what i can hear for now is that ST have much better sound quality, more natural. (with lowlatency mode).

btw mic in my hend is Sennheiser MD421 II and hp is AKG 141 Studio.

Hans, do you have some ideas why i can't use ASIO with this E-MU 0202 ? It working but sound is very strange, like no enough asio buffer or something. In all other programs asio works!

That's good to know. I can greatly reduce the latency by sacrificing more quality. Maybe I should make that configurable too.
No idea... Could you record a few seconds of it and mail it to me? So I can hear/analyze what you hear?

Just found that in low latency mode, HQ ON, if i rise first and second slider in EQ (bass) .. i hear artifacts. In HQ ON and lolatency off that is not case.

E-MU 0202.
I solve it!
For some reason it work when i set ASIO to work in 32bit , at 16bit it's not good.
Another problem with E-MU, is drivers, i don't have ASIO panel, but i can "call" asio panel from Breakaway Live. (ASIO Config) It will be nice to have button in ST for that.