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SSL DUENDE MINI - FireWire DAW Expander - Now with conclusions!


Anderton

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I'd also like to hear how stable Duende PCIe is with Cubase 4 and Fireface 800

 

 

I've used it with Cubase while on the same FireWire bus as the MR816csx, without problems so far. I'll see if I can "break" it as I continue with the MR816csx/CC121 Pro Review.

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I wanted to post about a possible workaround with Wavelab and realtime bouncing. I won't lie: I didn't do it because I have never needed to do this trick on purpose (to fool Wavelab) mainly because I'm mixing pug-ins and hardware in wavelab master section. Still not clear? Well, in Wavelab 6 one of the greatest function is to be able to integrate your external gear (I'm doing it with my SPL Tube Vitalizer for premasters). And what do you know, it is actually the only way to put the render function in realtime mode. Now, when you're rendering, you're actually running all the master section plugins in real-time and Wavelab is printing it in a file. All this, just by inserting the External Gear plug-ins under /ASIO and setting up the routing in connections.


So, my solution is, with some advanced routing of your asio (to hook up your digital out on a digital in) you actually don't transform your sound staying in digital domain. It's like using a "null" gear, i.e. just hooking up an AES OUT on an AES IN.


sH

 

 

That's GREAT!! I liked Jim's suggestion of the VST plug-in, but your idea of doing a real-time bounce by going "outside the box" is a very universal solution. I had considered that but I didn't want to do the analog in/analog out conversions. I'm embarrassed to say that I didn't even think of using SPDIF or AES/EBU because two of the three interfaces I use the most don't have it. But of course, that's a way to do the transfer without loss.

 

I could probably figure out a way to do this internally with the Creamware SCOPE system and avoid going outside the box at all or using a plug-in.

 

Thanks for the tip!! Interaction is why I love Pro Reviews.

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All right - now that my NAMM videos are over, it's time to dig further into Duende. Let's move on to the Vocalstrip, shown in the first image.

 

I'll start off with a negative: If money was a big issue, this plug-in would be last on my list. Not because it's not good, but because it has some stiff competition: the bundled Channel Strip we just discussed. I used the Channel Strip a lot for vocals prior to the appearance of the Vocalstrip, and had become very comfortable with which settings to tweak to get good results with my voice.

 

However, there's a reason why SSL made the Vocalstrip plug-in, and my guess is it's about speed - it takes a lot less time & effort to get a good vocal sound with Vocalstrip than with the more general-purpose Channel Strip. We'll zoom out and look at the elements that make up Vocalstrip, then examine the elements in more detail.

 

The De-Esser does what de-essers do, but the De-Ploser is a less common processor unique to Vocalstrip. It's sort of like a "smart lowpass filter" for getting rid of plosive sounds - very handy if your singer didn't use a pop filter!

 

There's a three-band EQ, with the curves and ranges optimized for voice. Although you can get similar sounds with the Channel Strip (or other EQs), it's very easy to dial in a good EQ setting relatively fast.

 

The Compander is fairly typical, but its drive control has some interesting uses. The metering, on the other hand, is again unique to the Vocalstrip. This is definitely something we'll need to cover in detail.

 

Also, note that you can change the order of the processors -- see the second image. The highlighted area in the lower right is where you change the order; click on a module, then use the arrow keys to move it left or right in the chain. This takes care of the "compression before EQ or EQ before compression?" question once and for all -- try them both, and see which one you like.

 

More to come...

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I was looking for some vocals to process through the Vocalstrip, but gave up when I realized that with virtually all the music I've done, the vocals aren't "raw" but recorded with processing, or rendered to include processing. The kind of source material I'd want to use for applying Vocalstrip would be raw vocals compared to processing through Vocalstrip.

 

So, I think I'll work on a short song that can be used for demoing the Vocalstrip and some of the other plug-in effects - for example, the bus compressor on the overall mix, Drumstrip on the drums, X-Comp on guitar/bass, etc. I'll whip something up in the next day or two we can use in the review...stay tuned.

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Well, it took a while to get back here...sorry, my travel schedule has been pretty hectic lately. Anyway, at long last, now you hear Vocalstrip at work, turning a really unappealing vocal into something, well, more appealing :)

 

But first...in the latest version of Sonar, I've been able to do the "fast bounce" thing with Duende Mini, which I had not been able to do before. This contradicts my previous findings that I could only get Duende to work with real-time bounces. I do know that Cakewalk has been doing a lot of optimization of their audio engine, and maybe this is one result.

 

Anyway, this vocal was recorded live through a Shure SM58 dynamic mic, and I wanted to use Vocalstrip to give it a more polished, "studio" quality. The first attached image shows the EQ settings I used. Basically, I added a bit of a boost in the upper mids to give more intelligibility, and cut away some of the low end to reduce the somewhat muffled, boomy quality you'll hear in the audio examples. I didn't use the midrange control; it's a bit too "peaky" for my tastes with most vocal material.

 

The vocal level varied quite a bit, and the second attached image shows the compressor settings. I went for minimum attack (0.1ms) to even things out. The ratio is 3.6:1, with a threshold of -9dB. I went for a soft knee response, 222ms of release, and also set the expander threshold at -48dB - I wasn't singing down there anyway, but I figured it could help get rid of any background noise.

 

Next, the audio examples.

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The following examples not only show what Vocalstrip can do, but also give an idea of how to produce a vocal to improve its sound.

 

Example 1 is the original vocal and backing track. In all of these examples except the last one, I normalized the vocal in each case so you could hear how the processing was affecting perceived level.

 

Example 2 adds the Vocalstrip EQ and no other processing. You can hear the vocal become more intelligible and intimate, and lose some of its "tubbiness."

 

Example 3 adds the Vocalstrip Compander and no other processing. This definitely helps even out the level variations.

 

Example 4 is the Vocalstrip EQ and Compander. Now we're getting someplace! The voices cuts better through the track and the levels are smoother.

 

Example 5 doubles the Example 4 vocal (the one with EQ and compression), pans the two channels a little left and a little right, and adds a bit of reverb. I think you'll hear that overall, the vocal's tonal quality and dynamics are far better than the original sound in Example 1.

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I didn't use the midrange peak/notch function in processing the vocal with the previous examples, because it didn't seem like it was needed (remember, the most important control on any signal processor is the bypass switch!). However, the midrange section does indeed create some interesting effects - it harks back to what I was saying about Duende including extra unusual things that take what might appear to be a normal function into a different realm.

 

So, to give you an idea of how this feature affects the vocal. I took a portion of the previous equalized/compressed vocal, and automated a sweep of the midrange control over 200Hz to about 7kHz while applying various amounts of boost and cut. (The range actually goes up to about 10kHz, but anything over 7kHz made no significant difference to the vocal sound.)

 

The attached image shows the pertinent part of Vocalstrip - the middle control (with the notch frequency set to 3257Hz) and toward the upper right, the frequency response curve.

 

Regarding the audio examples:

 

Example 1 sweeps the midrange with a 6dB boost. You'll note this is a pretty subtle effect.

 

Example 2 sweeps the midrange with a 12dB boost.

 

Example 3 sweeps the midrange with a -18dB notch.

 

Example 4 sweeps the midrange with a -36dB notch.

 

All of these add a distinctive "character" to the vocal that you may or may not want to use, but the point is, it's there if you want it, and the -36dB notch is certainly more than you'll find on most general-purpose parametric EQs. It's unfortunate there isn't a bandwidth control, although you do have the option for a gentler boost in the third equalization stage.

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Jim - Thanks for checking in. I will be re-starting the Duende review next week, between the Frankfurt show, editing 40+ videos for same, and getting really sick for a month, I've kind of gotten behind...but there's much more to come.

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Well, that was quite a hiatus...but no traveling for a while, so it's back to Pro Reviews.

 

Let's take a look at the Bus Compressor, which is available in mono or stereo versions. When you ask Duende owners exactly what it does, they sort of get this dreamy look in their eyes and start waxing mystical..."It sort of, I don't know, glues the sound together." But really, we're just dealing with a compressor, right? What's the big deal?

 

We'll get to the audio examples in a bit to demonstrate what the big deal is, but first, let's look at the controls in the attached image[/blue]. It's pretty much what you'd expect: Threshold, Attack, Release, auto-release option, Ratio, and Make-Up Gain. There's also a bypass switch.

 

You'll note your options are restricted, presumably to be as faithful as possible to the hardware. For example, your compression ratio options are 2:1, 4:1, and 20:1 (the latter is, of course, wonderful for a drum room mic bus). Similarly, the Attack and Release times are stepped. At first, the limitations put me off - dude, this is software, you can do whatever you want! - but after playing with the bus compressor for a while, this actually turned out to be a strength. The settings are pretty much what I would use anyway, and I can't argue with either the sonic results, or the speed with which you can get the sound you want.

 

If you look at the waveforms behind the picture of the Bus Compressor front panel, you can get a visual idea of the compression. Note that both waveforms have been level-normalized to 0. The upper waveform doesn't have the bus compressor, while the lower one is a bounce of the upper waveform through the Bus Compressor. The settings on the Bus Compressor were the ones used to generate the lower waveform.

 

It's clear just by looking at the waveform that the lower one is "beefier." There's a higher average level for sure, but you'll also note that the dynamics haven't been destroyed or "flat-lined." Granted, part of that is because I didn't push the compression settings all that hard. But as you'll hear with the audio examples, it seems the main talent of the SSL Bus Compressor is to give a nice "lift" to the sound while retaining a solid sense of dynamics.

 

Now let's listen to it in action.

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I've created three audio examples.

 

The first audio example is a little demo piece I put together from Big Fish Audio's Funky House Grooves 2 loop library (I'm reviewing it for the August 09 issue of EQmagazine). There's no processing on this at all, and the peak levels have been normalized to 0.

 

The second audio example is the same file, but bounced through the Bus Compressor using the settings shown in the image in the previous post. No other processing was used.

 

The third audioexample alternates between segments. First there's the unprocessed sound, then a second of silence, then the same segment through the Bus Compressor. This makes it a little easier to do a head-to-head comparison.

 

As I think you'll hear, the Bus Compressor adds a gorgeous lift - not subtle, but it doesn't hit you over the head, either. The dynamics remain intact, and the sound is wonderful.

 

I would not recommend using the Bus Compressor as a substitute for a "maximizer" plug-in. The harder you push it, the easier it is to get the compressor out of its "comfort zone." But it's very transparent in its operation. When you hear the processed example, I doubt your first thought will be "it sounds compressed." Instead, when compared to the unprocessed example, you'll likely think "it sounds better."

 

There's not really much more to say about the Bus Compressor. It's sweet, effective, and improves your sound without degrading it. I'm not a fan of strapping a compressor across the stereo bus - I leave that for the mastering - but I'm definitely a fan of this plug-in. I've found that using it on a mixed track, and then following it with a dB or two of level maximizing (e.g., WAVES Lx series), gives a "loud," punchy sound that's can compete in the loudness wars, but without the sonic destruction you get using heavy maximization alone.

 

This plug-in gets a definite :thu: :thu: (that's two thumbs up).

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I like drums. Hey, who doesn't? And I'm always for ways for them to sound bigger, brighter, better, and ever-more bodacious. Having had good luck with Duende so far, it seemed like a good time to check out Drumstrip.

 

The attached image give a pretty good idea of what's going on. The first thing that struck me: This is a company that understands the virtues of parallel processing on drums - check out the Dry/Wet controls at the output, and for the Listen Mic Compressor.

 

Let's start with the HF (High Frequency) Enhancer and the LF (Low Frequency) enhancer, because it's easy to wrap your head around it: Pick a frequency, throw in some drive (which adds more of an edge to the sound), then dial in the amount of enhancement you want.

 

In use, these two processors can definitely add some low end and high end mojo and the easiest way to show this is with audio. So, the first audio example is the bypassed drum sound, and the second audio example is the same track but processed with the enhancers. By the way, I used a reasonable amount of enhancement - you can take it to a much more extreme level, if you want bass that may trigger small tremors in seismically-active areas.

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On to the Transient Shaper. This is arguably the most "drum specific" part of Drumstrip. It's an interesting effect, because I feel it really needs to be used subtly for best results; it's almost something you "feel" rather than hear.

 

The first attached image highlights the Transient Shaper module. Gain essentially determines how much of the attack gets "caught" for the enhancement process; more gain means more of the attack gets shaped. Think of it similarly to how you would think of a noise gate threshold.

 

Speed controls how long it takes for the generated transient to decay, from a "glitch"-type spike to something more substantial. The Amount controls determines how much of the transient gets mixed in with the original signal (and can also be subtracted if you click on the Inv button). Check out the second attached image; the upper waveform (track 2) is the original drum loop, while the lower waveform (track 3) is the processed loop with no gain or change other than adding the Transient Shaper.

 

The Audition button is one of Drumstrip's coolest features - it's like being able to listen to a sidechain signal. You can hear the changes that happen to the attack in isolation, which makes it easier to adjust the controls.

 

And of course, you want to hear what it sounds like, right?

 

The first audio example has the Transient Shaper bypassed, while the second audio example has it active, using the settings shown in the first image.

 

Note that because the Transient Shaper can generate sharp spikes, I find it's a good idea to lower the drum level to create the headroom for these to be recorded properly. Then, adding a level maximizer can reclaim a few dB by taking off the very peaks of the attacks.

 

So...is it great? It's useful, but I find it works best if you don't hit people over the head with it, and if you follow it with some kind of maximizer (or the SSL Bus Compressor, if you're so inclined!) so you can still maintain a reasonable average level. It's definitely more of a team player that adds some enhancement rather than taking a sound into a whole different dimension.

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Now let's take a look at the Gate. Probably anyone who's in the market for a Duende is far enough along to know what a gate is, and SSL's implementation isn't all that different from the norm. The attached image shows the Gate controls.

 

The Attack control sets the time for the gate to ramp open after being triggered, while Release determines how long it takes to reach the full gate off setting after the trigger goes away. Hold controls a gate-on duration that starts when triggered and lasts for the Hold value. This would be useful if you want to make sure you always catch transients of a particular length.

 

However, it sure seems to me that the calibrations have the decimal point in the wrong place...Attack is said to go from "0.00ms" to "0.10ms," but I think they mean 0-10ms. Ditto Release: It reads 0.01ms to 1.00ms, but it sure sounds to me like 1-100ms. Hold seem to be calibrated correctly - 0ms to 4000ms.

 

Open sets the level at which the gate opens, while Close sets the level at which the gate closes. Range sets a maximum amount of attenuation when the gate is closed.

 

While this is all fairly straightforward, the metering is very helpful. The left meter shows the input; the small green line indicates the Open threshold, and the red line shows the Close threshold. As you can correlate this to what's happening with the signal level, the process of setting levels is pretty obvious. The right meter shows the amount of reduction being applied, with the white line indicating the maximum amount of attenuation.

 

Being able to attenuate by something less than full off is very useful for tightening up drum ambience, without making the drums sound artificially gated. Listen to the first audio example; the Gate is bypassed, and you can hear a ton of reverb on the drums. Now, check out the second audio example - this is with the Gate active, but attenuating by only 15dB or so. You'll hear how the reverb sounds way less prominent, but nonetheless, doesn't go away; in a way it resembles downward expansion. I also tried this gating setup with ringing toms, and the Gate was very effective for that, too.

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This is the last element of the Drumstrip, and it is a truly unusual "special effect." But we need to start with a little background...

 

First, you can download a free, somewhat less capable version of the Listen Mic Compressor from SSL's web site - all you need to do is register. It's available for Mac or Windows, so you can check out this effect for yourself. I'll be interested to see if anyone has any comments on it to add to this thread.

 

Second, the LMC effect has an interesting history. Here's the story, as quoted from SSL (you can find out more at the link given above for downloading the plug-in):

 

The SSL Listen Mic Compressor was the secret weapon in many producers' sonic arsenal of recording techniques. Originally designed to prevent overloading the return feed from a studio communications mic, its fixed attack and release curves were eminently suitable for use on ambient drums mics. Of course, we’d like to take all the credit for this great sound, but as usual, it was the creativity of SSL users that led to the idea.

 

Long-time SSL user Hugh Padgham was one of the first to capture this new drum sound on tape,while working with Steve Lilywhite on Peter Gabriel’s ‘Intruder’, he told Mix magazine: "On a normal console, you have a button to press to talk to the musicians in the headphones, but you did not have a button to press for us to listen to the musicians. To do that, you'd plug a microphone into a spare channel on the desk and listen to your musicians through that. But the SSL had a reverse talkback button and there was a microphone hanging up in the studio already, a dedicated input into the reverse mic input on the console. And on this microphone, they had the most unbelievably heavy compressor, so you could hear somebody who was over in the corner.

 

"One day, Phil (Collins) was playing the drums,” Hugh recalls, “and I had the reverse talkback on because he was speaking, and then he started playing the drums. The most unbelievable sound came out because of the heavy compressor. I said, 'My God, this is the most amazing sound! Steve, listen to this.' But the way the reverse talkback was setup, you couldn't record it. So I had the desk modified that night. I got one of the maintenance guys to take the desk apart and get a split output of this compressor and feed it into a patch point on the jack field so I could then patch it into a channel on the board. From there, we were able to route that to the tape recorder."

 

All right...now let's get into the LMC itself.

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Let's look at the difference between the free version and the one included in the Drumstrip plug-in.

 

The first attached image shows the freebie plug-in. Controls are Compression (goes from "less" to "more"), Input level, and Output level - that's it.

 

The second attached image shows the version in the Drumstrip. There's no Input level control because of course, the Drumstrip itself has one at the beginning of the plug-in. The Compressor control provides the same function as on the LMC-1, and Makeup is equivalent to the LMC-1 Output control.

 

The Mix control, which the LMC-1 doesn't have, mixes the LMC in parallel with the dry drum sound. The control ranges from dry only to effect only, or anywhere in between. (If you want to be able to do what the Mix control does with the LMC-1, you'd need to use the LMC-1 as a send effect, and mix in the desired amount of processed sound by changing the aux send or aux return level.)

 

The LMC-1 includes EQ that provides a more "room-like" sound, but cannot be defeated for full-range operation. The version in the Drumstrip includes an EQ in/out button so you can remove the EQ if desired, and go for a more full-range compression sound.

 

However, there's a caution here. The compressor has extremely tight time constants because it was designed to pick up musicians talking from far away in a room; no thought was given at the time to using this "reverse talkback" compressor with regular instruments. As a result, you can't lengthen the release time to minimize distortion with low-frequency sounds. If you apply the LMC to drums with the EQ out and there's a massive kick going on, it will have a distorted character that you may or may not like. I realize that the whole point of the LMC is to re-create a specific effect, but I would have much preferred if when the EQ is bypassed, you would have the option to adjust the compressor release time. As a result, I tended to leave the EQ in more often than not.

 

[Hey SSL - any chance of adding a release control in an update? You could make room for this control in the plug-in by abbreviating "Listen Mic Compressor" to LMC, moving the EQ in/out button up, and putting a Release control underneath the button.]

 

Okay...time for some audio examples! As a post can have only five attachments and we've already used up two for the images, we'll put the audio examples in the next post.

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Here we go...

 

The first audio example is the drum loop without any processing, for comparative purposes.

 

The second audio example has the LMC sound only (no dry mixed in), with about 40% compression.

 

The third audio example mixes the dry and processed sounds together, with about 40% wet, 60% dry.

 

The fourth audio example takes out the EQ and monitors the LMC sound only. I've kicked the Compression up to 50% so you can really hear the low-frequency distortion that occurs due to the short time constants.

 

The fifth audio example has the EQ switched out. The output is a mix of the dry and processed sounds, with about 40% wet, 60% dry; the Compression amount is around 43%.

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First of all, thanks Jim for checking out the Gate calibration issue. I knew my ears were telling me something different from my eyes!

 

We've pretty covered everything about the Drumstrip except for the routing option, which is similar to other SSL modules. The attached image shows the "Process Order" section of the module, where you can arbitrarily move any module to the left or right in the chain. You just click on the module, it becomes highlighted in yellow, and you move it using the and buttons.

 

So...let's draw some conclusions. First off, I find this a very useful module. That usefulness is mitigated somewhat by your being able to get a more basic version of the LMC for free, and that the Channel Strip module already includes a Gate (albeit without the same degree of control over threshold) as well as EQ. However, the Transient Shaper has no equivalent in the Duende processing arsenal, and (in typical SSL fashion, I might add!) the HF and LF Enhancers have their own special mojo that the Channel Strip EQ doesn't duplicate.

 

Being a big fan of parallel processing with drums, I also like the Dry/Wet controls on the LMC and the overall output. Being able to change the Process Order is also very handy.

 

My "wish list" is limited to two things: Fixing the Gate calibrations, but it seems SSL is already on that, and having time constant controls on the LMC which would do things: 1) allow it to act more like a standard compressor, a feature that Drumstrip lacks; and 2) minimize distortion on low-frequency material. And while I'm in wish list mode, I'd like to see two buttons on the output: Limiter on/off that traps peaks, and Saturation on/off that would add more of a clipping effect with peaks. Of course you could have a plug-in after Drumstrip that did these things, but it would be convenient to have these features on Drumstrip itself.

 

Overall, though, I have to say that you can indeed make drums sound better with Drumstrip...and that's what it's all about, right?

 

Next, let's take a look at the X-EQ, and follow that up with the amazing X-Comp.

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Does the world really need another equalizer plug-in? Let's find out.

 

X-EQ is a 10-band, non linear-phase EQ. The ten bands are highpass, low shelf, six bands of parametric EQ, high shelf, and lowpass. Of course we'll get into details shortly, but let's take an overview first.

 

The X-EQ's main claim to fame, aside from eschewing linear-phase response, is that there are five different curves for the highpass and lowpass filters and nine different curves for the midrange. Referring to the first attached image, selecting a band causes a drop-down menu to appear toward the upper right (circled in red for clarity); this is where you make your selection.

 

The overall philosophy reminds me a bit of the Liquid Mix hardware, which allows for multiple equalizer responses. In the case of the Duende Mini, some emulate specific analog responses, while others have responses that could be done only through digital technology. I'm not surprised that SSL has decided X-EQ should have a "character" - that seems to be a common theme here amongst all the plug-ins.

 

As appropriate for different functions, Gain, Frequency, and Q controls appear. There's also input metering, output metering, on-off switches for each band, and standard load/save/compare functions.

 

You can also enable a post-EQ spectrum analysis display (second attached image) which makes it easy to see "rogue resonances" and the like right after loading it, rather than listening carefully for them to get "in the ballpark." You'll also see one of the nine possible parametric responses being selected.

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Let's listen to how the choice of the midrange "bell" affects the sound. The sonic character depends on two main factors: Whether the EQ is boosting or cutting, and the amount of boost or cut. I tried to think of the most obvious way to show this, and decided that the constant hiss of white noise would highlight any differences the best. As you can imagine, the same kind of "character" will be impressed upon instruments.

 

Duende's documentation is very helpful in this respect, giving full descriptions and charts of the various responses, and indicating the goal of the emulation.

 

The first audio example plays three sections back-to-back, with a slight silence in between: White noise boosted around 2kHz with the Proportional 1 bell, then the same thing with the Proportional 2 bell, and finally, the same thing with the Proportional 3 bell. You won't hear much difference between 1 and 2, but you will definitely hear a difference with 3.

 

The second audio example again plays three sections back-to-back, but this time white noise is cut around 2kHz with the Proportional 1 bell, then with the Proportional 2 bell, and finally, with the Proportional 3 bell. This time you'll almost certainly hear the difference between 1 and 2, but you will definitely hear a difference with 3.

 

The third audio example plays four sections back-to-back using the classic curve (standard and asymmetrical). The first is standard cut, the second is asymmetrical cut; the difference is obvious. The third is standard boost, and the fourth is asymmetrical boost. You'll hardly hear any difference at all.

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For the five different highpass and lowpass filter responses, I thought showing the curves themselves would be useful because the main difference is the steepness of the response dropoff; see the five attached screen shots. These show the highpass response; the lowpass response is simply the reverse.

 

Also, note that there are some sonic differences, so the next post compares the Critical, Butterworth, and Chebychev responses.

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