Budget Tactile, Mono Simhub Profile

Some of the tactile DIY rigs on here are just awesome but a little out of my budget. I only have a single BK LFE (EDIT: see post 27), it's bolted to my seat powered by an Inuke 1000 in bridged mode. I'm mainly playing Pcars2 and Dirt Rally 2.0.

I've been doing some experimentation with Simhub and its effects, for now I'm just using a selection of the default effects with modified settings. I see it has a function for custom effects but it's a bit beyond me atm. Is anyone in a similar position (using only mono effects) and has created something they wouldn't mind sharing? Or can give me some tips on how to get started making custom effects?

I've attached my (very basic) profile if anyone wants to try it. All effects are mono, under 60Hz, with some as low as 8Hz. Obviously remove the ".txt" from the end of the filename to make it work. Comments/feedback welcome.

simhub1.png


In terms of DSP, I have +6dB of gain (Q=6.0) at 60Hz but it should work well enough without this. No other crossovers or anything else are used.

Tip: No need to buy an amp with built-in DSP or a separate DSP box (e.g. miniDSP). Free software called EqualizerAPO allows use of DSP filters in software on your PC. EqAPO is easier to use than the clunky DSPs on behringer and similar amplifiers, and much more capable.

I'm new-ish to sim racing but I've been using EqAPO (and buttkickers) for home cinema multi-channel audio and tactile bass for several years so I'm quite familiar with that aspect, it's simhub where I'm a novice and need some advice.
 

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I've put my responses to mr latte in this spoiler so it clogs up my thread no further. If latte or anyone else wants to debate these points please do the same, much appreciated.
Im not convinced recommending EQ APO is suitable for many users and you certainly have a lot more to do to cover points illustrated here below. No disrespect is intended, no name calling, no personal insults. Shared perspectives and views from two people with passion for audio, expressing different opinions, different ideas and experiences.

OK, so you're not convinced. Doesn't change the facts that it is suitable for many users.

The above clearly shows you're informing others to try alternative options via free software options. That the software is going to work as well if not better, you say and save people money.

I'm informing others that it is an option that could potentially save money. I'm not telling anyone it's the only option, or even the best option for them. Just a free option, that's all. Why are you so upset about it? You don't like people saving money?

People reading this, while no names are mentioned. Who does it look like, this is reflected at?
As who on these forums has covered amps with DSP or separate DSP boxes in several posts? Here you also make it clear that Equalizer APO is the software you use. You state Its not only as capable but potentially better to use than the "clunky" software for the Behringer amps or other amps. The software is also much more capable.

It is now clear you are recommending EQ APO as the software you use to others.

My comments about the clunkiness of the behringer DSP's UI is just my opinion and I made that clear at the time. I recommend EQ APO because it works and it costs nothing. If a user is looking for/evaluating DSP options, tried EQ APO on my recommendation and it didn't work for them, nothing is lost. If you don't like it, don't use it. It works for me, so why shouldn't I recommend it based on my experience?

Confirmation that you are not just showcasing what "you" do, as the topic has never been presented as just "your sole journey" into all this.

Not sure what you mean. I showed what I did, and at the same time showed that EQ APO is a viable option for others.

We have further confirmation of your recommendation to others and highlighting that people with standard amps could use EQ APO. You state in several posts how EQ APO is better, easier, more capable and also free. Yet, no, no, no your not criticising what others have recommend/showed even though you directly used them as examples and your not trying to steer people towards anything particular. Any sign of self-contradiction here at all?

I'm not seeing a contradiction. Apart from power-limiting features (which must be part of the amplifier by definition) EQ APO is more capable in terms of functions than the DSP in the behringer amps and the MiniDSP units, that is a simple fact. That doesn't mean EQ APO is the best option for everybody, there are many other factors to consider. When I point out the fact the EQ APO is more capable, that does NOT mean I'm criticizing people who use the behringer DSP or people who recommend it.


So you state all my 4 asked scenarios/queries, do indeed work.
How can you prove it to me. So now we have a personal interest that you are wanting to prove to me that EQ APO can do it all and not only that but much more

No personal interest, just trying to be helpful. And what has trying to be helpful brought me? Nothing but attacks from you.

So the truth is, you have not used Simhub with 8 transducers and been able to apply individual EQ and crossover controls to each channel to confirm it works as you said it can. So we have another self-contradiction here.

Yes, and I made that clear from the start, I didn't misrepresent anything, no contradiction. I've used it control 6 audio channels with individual EQ and crossover controls, plus a 7th channel on a separate sound card with separate EQ and crossovers. Simhub output is just audio as far as the software is concerned, so what logical reason can you think of why it wouldn't work? As far as EQ APO is concerned, it doesn't matter if it's a media player or a game or simhub that is outputting the audio, so why would it matter? By analogy, this is like you saying "yes your bucket holds water, but how can we know it holds milk until you actually try it?" Doesn't make much sense. If you still don't believe EQ APO has this capability, don't use it.

It's becoming apparent what's questionable is your own experience so far with Simhub. How incorrect you have been on certain things and making claims or speculating some things like they are indeed fact.

I also made this clear from the start, yes I am new to simhub. I made a few incorrect assumptions about it, when I realised my mistake I admitted it straight away and made a correction. What more do you want. Sorry I'm not infallible like you are lol.

What help in settings or configuration have you given to them? After all your the guy promoting the use of EQ APO

All the posts in this thread where I have shared my configurations and settings. Has someone asked me a question or asked for help and been ignored by me?


You said you understand not only harmonics but were well educated in physics. You said Simhub alone gives you full control of what Simub puts out. You expressed "Your Reason" was due to using only a single unit and you highlighted how you were using only low frequencies which can be seen in your settings at the very beginning.

I do understand harmonics. I made an incorrect assumption about simhub and I retracted my statement when I realised my mistake. What's you problem here? People aren't allowed to get things wrong?

Fact nobody stated you were questioning anyone else's specific use of a crossover

False, You stated it, right here...

@Ormy has questioned me in other discussions, why we would apply a crossover filter..


It would not matter what unit was being used or what the amplifier was. Being a single unit is irrelevant but you did not know this, you appeared to assume this was a factor. It was nothing to do with splitting certain frequencies to more than one unit.

Yes, I made an incorrect assumption regarding simhub. I deeply apologise if this upset you. Can we get over it now? You've brought it up three times already in one post.

My Own Conclusions To All This:...

For someone who types so much, pretends to know all the answers and attacks others for presenting alternative solutions, you've barely answered any of my actual questions, given me next to no help at all. All you've done is say how I am getting everything wrong because I didn't do it the way you did it. You can't seem to accept there are alternative methods to solving a problem than the method you used. In addition, you misrepresented my statements to make me look bad on other threads, and made many false statements. For example...

Well, oh dear, so that's another thing it can't do then...

Another point to highlight is that early adopters to EQ APO are not going to be able to do what you can do and use code to configure its settings/controls?

It can do that, I gave the code to do it. I didn't write that code, I found it with 2 minutes of searching on the EQ APO sourceforge page, something anyone could do.

It was never raised in that perspective but you did previously state you would need another 7 amps if trying to test an 8 channel configuration. Your words, not mine.

You really are trying your hardest to nitpick at every little thing aren't you? I would need another 7 more channels of amplification, whether I buy 7 low power 2ch amps and bridge each one, or buy 4x bigger 2ch amps, or 2x 4ch amps, what difference does it really make? Oh I forgot sorry, I have to do it the way you did it or else.
 
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@Ormy, if you have JRiver Media Center, just install WDM Driver and use that as an output for SImHub.
Jriver besides being one of the best media player on the market, also comes with an excellent DSP Studio that features things like PEQ with very intuitive and easy to use interface

Ah, thanks so much. I am aware of Jriver but I personally prefer MPC-HC for my media watching. I know many people on the Audio forums I frequent use Jriver as an all in one alternative to MPC or VLC combined with EQ APO.

I didn't know about this WDM feature though, that's cool. So in theory one could use Jriver/WDM to implement DSP for simhub, although I can think of a certain someone who probably wouldn't approve haha.

EDIT: Ahaha this is fantastic, reasonably good spectral analysis without the need for loopback (physical connection from an output to an input), been looking for a way to do this for years. Some limitations though to Jriver DSP studio analyzer though, for a start the vertical axis isn't labeled, I've been able to determine through trial and error that each vertical division is 3dB. I'll be doing lots more testing using this method and the loopback method to definitively answer the question of what exactly simhub is outputting compared to it's configuration. Is it putting out significant harmonics (unlike a true tone generator) that need to be curtailed with low-pass/high-pass filters, or not? Answers coming soon.

Shame Jriver is not free though. At the moment I'm using the trial version and should be able to get my testing done before the trial period ends. Is there any software with this virtual-audio-output(WDM) and analyzer/spectrum analysis functionality that is free?

EDIT 2: So when I said "reasonably good spectral analysis" above I may have been a bit overzealous, turns out it's not really that good at all due to low FFT sample size (see details below). Useful for ballpark-level analysis only. Still really appreciate the suggestion though @Andrew_WOT, please don't take this as a criticism :p (Unless there's a way to increase the FFT sample size/accuracy of the analysis somewhere in the options I haven't found?)
 
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Preface: I'm taking this thread in a slightly different direction now. I want to test more rigorously what simhub is actually outputting to our tactile transducers. If simhub is set to make a pulse of xHz for y milliseconds (in response to a gear change or whatever), does it make a nice clean tone at xHz like a tone generator would? Or does it also produce significant harmonic content that requires the use of a crossover or low pass filter to block out? We're about to find out.

This post will contain many images, they will all be thumbnails to improve readability.

I have reason to believe my BK LFE my be faulty or at least worn out and tired, therefore I am removing it from the equation completely. I have a particularly capable DIY subwoofer setup (ask if you want more details), so I can just point simhubs output to my subwoofers instead of the BK and measure the response with my UMIK-1 calibrated measurement microphone (link).

Measurement methods: I will be using three separate and independent methods to measure simhubs output which I will refer to throughout this post. I will also be doing preliminary and control testing on all three methods using REW (free audio analysis/measurement/correction software used by audio professionals, link). I will use its tone generator function which is known to be accurate, good quality and free from unwanted harmonics to perform control tests on the three measurement methods.

1.) Creating a virtual audio device using Jriver WDM (thanks to @Andrew_WOT for showing me this). Simhub can output directly to this audio device, Jriver includes basic spectral analysis. This has the benefit of being entirely within software so many factors that could affect the final output are removed including characteristics of the soundcard and amplifier. Therefore this should provide the 'cleanest' measurement with the least distortions. (Jriver has frequency labels on the horizontal axis, but none on the vertical amplitude axis. I have determined the vertical divisions are every 3dB)

2.) The loop-back method. I will connect the output of the soundcard directly back to it's input with a short cable, then analyse that input with a spectral analysis software. In this case I am using Visual Analyser (link) which I have used before for spectral analysis and software-oscilloscope functions. It's basic and a bit slow but it's accurate. This method excludes the factors of the amp and the transducer (modeled here by a subwoofer) but includes the factor of the soundcard output.

3.)Sending the output from the soundcard to my subwoofers (via their amplifier of course), I then measure their output with the UMIK-1 calibrated USB microphone, microphone input is analysed and displayed as frequency spectrum by REW. This is probably the most revealing method as it includes everything in the signal chain including amplifiers. (I'm using the same model amplifier for subwoofers and tactile transducer). Therefore it likely to be the least 'clean' of the measurement methods as it has the most possibilities for distortion or other reproduction error.

Preliminary and control measurements:
So lets start of with method no.3, sending output to a subwoofer which performs exactly the same function as a tactile transducer except it puts it's vibration energy into the air instead of into an object like a seat.

Since the microphone picks up every sound in the room, not just what I want it measure, I need to take a baseline measurement with no sound playing at all, no audio output, nothing. Here it is.
subon.png
Ideally I should measure nothing at all, absolute silence. In reality there are always ambient noises, we are accustomed to ignoring them, but the microphone picks them up all the same, I've circled the main noises in yellow. The first peak at 25Hz and 45dB, I have no clue what that is, maybe airflow from the airbricks, maybe my breath, maybe someone using a powertool a few houses away, who knows. The second peak at 50Hz and 35dB is unmistakably mains hum (UK mains electricity is 50Hz) coming through the subwoofers, I can't hear it at all but the mic can. The peak at 100Hz is the 1st harmonic of the mains hum. The noise at 600-1000Hz at 20dB is my computer case fans.

Just to make sure those 50Hz and 100Hz peaks are indeed mains hum, I'll power off the amps driving the subwoofers and take another baseline measurement.
sub off.png
Yep, as expected the 50Hz and 100Hz peaks are significantly diminished. Still some tiny amount of hum at those frequencies from other electrical equipment, screen, PSU in the PC etc.

Right so lets make some tones and see how they measure. Here is a 400Hz sine-wave tone played from REW and measured by method no.1.
40b.png
A nice sharp peak at 400Hz with no harmonics or distortion, just as expected, so far so good.

The same 400Hz tone measured by method no.2
40c.png
Uh oh, significant odd-order harmonics present, not good. But looking at the top graph, it's obvious the signal is just clipped. Obviously I am clipping the soundcard input. I will lower the soundcard output from 100% to 50% and try again.
40d.png
There we go, much better. A nice, sharp peak at 400Hz, as expected.

400Hz tone again now measured by method no.3.
sub 400.png
Nice sharp peak at 400Hz again, awesome. All three measurement methods working well so far.

Just to double check, lets check all three methods again, this time at 40Hz.

Method no.1 and no.2 shown in the same screenshot
40e.png
Hmm, how strange. Method 2 (visual analyser, green plot) is showing a nice clean peak. Jriver analyser (yellow plot) is showing a very broad peak. Why the difference? Which is correct? Let's look the at result from the microphone (method no.3) to find out.
sub 400.png
A nice sharp peak that agrees with method no.2. So why is method no.1 showing a different result? The answer is it is using a smaller FFT sample size (and I'm not sure it can be increased? Maybe I missed that setting?)

A little background: all three methods use what's called a Fast Fourier Transform in order to calculate the data needed to plot a spectral analysis. The accuracy of data decreases as the frequency of the measured signal decreases. To increase accuracy of the calculations more samples can be taken but this means more calculations which means it's slower.

Lets look at this in more detail. Here is the 40Hz tone measured by method no.2 with a sample size of 65536 (the sample size used for all measurements with method no.2 unless otherwise stated)
40hz2.png
Nice clean 40Hz peak. Now look at the same tone but with the sample size reduced to 4096.
40hz.png
We can see the peak has broadened because the data at this low frequency is less accurate due to insufficient FFT sample size. It looks just like the result from method no.1 from before.
1.png
We can now conclude that method no.1 is unsuitable for measuring tones at low frequencies due to low FFT sample size. Perhaps there is an option hidden in Jriver to increase the sample size? I looked and couldn't find it.

So I have validated that methods 2 and 3 give results consistent with each other. Moving on now to the actual tests of Simhub I will only be using testing methods 2 and 3.

Actual Simhub Tests:
I'm going to start simple and just look at one frequency and the 'add white noise' function for today. Later I will look at other frequencies and combined effects that are overlapping each other.

I set simhub to make a 42Hz tone. Result from method no.2
sh42lp2.png
Lots of harmonic noise, but look at the top trace, the signal is clipped again. I'll reduce the gain on the effect in simhub and try again.
sh42lp3.png
Ah, much better. There is a little bit of noise but it is centred around the fundamental. No significant harmonics.

And the same 42Hz effect measured by method no.3
sh45nown2.png
Aha! A nice sharp 42Hz peak. Some low level noise but no harmonics.

Let's turn on simhub's 'add white noise' feature and see what happens. I'm adding white noise at 25hz, so ideally we should be getting output between 17 and 67Hz and not much outside that range. Let's find out.
Result from method no2 and no3 side by side.
sh45=25.png
Pretty much as expected. Wide bandwidth noise from 17 to 67Hz and not much output outside of that range. Certainly no harmonics or other serious issues.

Measurements from simultaneous simhub effects overlapping to follow at a later date.
 
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  • Deleted member 197115

Yes, it works
1597429279560.png


Just to save you some time, dowlnoad SPAN from the link above, install it. In JRiver DSP studio click Manage Plugins / VST ....
and choose c:\Program Files\Common Files\VST2\Voxengo\SPAN.dll

My understanding that Block Size in screenshot above controls your FFT.
 
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Just to save you some time, dowlnoad SPAN from the link above, install it. In JRiver DSP studio click Manage Plugins / VST ....
and choose c:\Program Files\Common Files\VST2\Voxengo\SPAN.dll

My understanding that Block Size in screenshot above controls your FFT.

Awesome, much appreciated, I didn't realise Jriver was also a VST host. Will give this a whirl this weekend when I do some more testing.
 
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I've mentioned before that I suspect my BK LFE unit to be faulty or at least worn and not working quite right. This is because I don't seem to be getting much output above 60Hz unless I apply large PEQ boosts, and also I'm getting plenty of output below 20Hz without applying boosts there. Both of these findings are contrary to what other users of the BK LFE have reported. But why? Why is my BK LFE different. BK say they are "nearly indestructible and maintenance free". What could it be? It's not the amplifier as I've tested it previously with subwoofers and it's working fine.

Today I did some testing on the BK unit itself, impedance sweeps and so on, I quickly discovered the answer, it's simple but rather embarrassing haha. Turns out I don't have a BK LFE at all, I have a BK Concert. The previous owner had removed the labels and listed it on ebay as an LFE. The colour of the labels is the only visually distinguishing feature, without the label they look identical which is why I didn't notice till now when I investigated the impedance.

The Concert is designed for more output in the lower frequencies compared to the LFE, which explains why my unit is behaving differently to the BK LFEs that other users have reported on. This also explains why low-passing at 60Hz seemed to have no effect on the sensation.

I'm not going to hassle the ebay seller over it, for one thing the concert is slightly more expensive than the LFE, and the used price I paid was so good I can't in good conscience complain to the seller.

Further testing of simhub effects following on from post 23 to follow soon.
 
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