Today I’m sharing something I’ve been doing a lot lately and can make mixing a lot of fun.
Electro-Harmonix Memory Boy analog delay pedal
Use guitar pedals for mixing
Plugins are great but its just not the same as running sounds through real analog circuits. You can send sounds out of your audio interface, tweak the pedal settings and even ‘play’ the pedal to do realtime automation. It can be a lot of fun to work this way.
For the demonstration I’ve recorded an electric guitar directly into my DAW with Amplitube for amp and cabinet simulation. I’m going to then run the signal through an Electro-Harmonix Memory Boy analog delay pedal.
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Here is the guitar with the Amplitube 3 plugin added (stereo, amp+speaker+mic)
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Now I’m going to run the sound through the pedal. To do this in your home studio you need an audio interface
with a couple spare analog outputs, if your interface has 4 analog
outputs, that’s perfect.
Connect a guitar cable from output 3 of
the interface to the input of the pedal. Connect the ouput of the pedal
to one of the instrument inputs of your interface (usually in 1 or 2). In this case, I’m going from output 5 through the Memory Boy and into input 1 of my Profire 2626.
In the DAW you need to tell the signal where to go. Most DAWs will have a plugin for hardware inserts. In this next example I have the hardware insert before the amp, just like if I had the pedal before the amp. When you do this remember to keep the mix control of the pedal to about 50% or less, you still want to have the clean guitar get through.
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There’s another way to use pedals and that’s as a separate FX track. Add a new mono track to your project. You can either use the the hardware insert plugin again and have the other tracks send to this track, or set the track to monitor the analog input and have the other tracks send to the analog output.
Here I have the guitar track sending to my delay track, which has the hardware insert plugin set the same as before. The direct guitar signal goes into amp plugin, then to the master output. The signal from this track is also going to the delay track. It then goes into the insert plugin, out the interface, into the delay, (set to 100% wet, no clean sound) and then into the interface and to the same track. I hope this makes sense. The benefit of doing it this way is you can blend in as much of this signal as you want. You can also use plugins before and after to shape the sound independently of the original tracks. Also, you can send multiple tracks in at the same time.
Here is the guitar through the delay as a send. (delay after the amp in the chain)
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And finally, here is what the delay track sounds like soloed.
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Pedal are inexpensive, easy to use and a lot of them really sound great on more than just guitars. Have fun!
Introduction
T-RackS 3 is a suite of high-quality digital and analog-modelled VST/AU/RTAS Dynamics and EQ processors, for mixing and mastering. T-RackS can also be used outside of your DAW as a standalone mastering application. Version 3.5.1 is the latest at the time of this review.
T-RackS Standard comes with the 4 ‘classic’ processors and metering suite as individual plugins also available within the T-Racks Shell or standalone.
T-Racks Deluxe has all the same functionality but adds a few more processors for a total of 9 including two analog modelled devices, the Fairchild Limiter and Pultec Tube Equalizer.
Each of the processors are also available for $99 each. The two newest additions to the T-RackS family – Black 76 Limiting Amplifier (modelled after Urei 1176), and White 2A Leveling Amplifier (Modelled after Urei LA2A Tube compressor/limiter), are only available as add-on purchases.
The decision to offer the individual processors was based on user feedback and common use. When T-RackS 3 was first released it was considered a mastering plugin, but users started liking the effects for mixing as well. Splitting up the system outside of the T-RackS Shell has made things much more convenient.
Standard and Deluxe Processors Comparison
Standard
Deluxe
Classic T-RackS Compressor
Classic T-RackS Multi-band Limiter
Classic T-RackS Clipper
Classic T-RackS Equalizer
Vintage Tube Compressor / Limiter model 670
Vintage Tube Program Equalizer
Opto Compressor
Multi-algorithm Brickwall Llimiter
Linear Phase Equalizer
Classic T-RackS Compressor
Classic T-RackS Multi-band Limiter
Classic T-RackS Clipper
Classic T-RackS Equalizer
The T-RackS 3 Shell
The Shell contains all the processors – up to 12 can be loaded in any order, series and parallel. You can load and save presets for individual modules or globally. The shell also contains the full metering module with Peak, Perceived loudness, Phase, correlation and frequency spectrum metering. Each of these can be adjusted to suit the user. The Shell contains four buttons to AB(CD) 4 complete effect chains along with a “copy to” button. Lastly, at the top-right the compare button allows you to compare your processing to the volume adjustable source.
Standalone showing signal chain.
Standalone
The standalone version adds several more features to the shell. Below the metering section is the playlist. You can drop in all your files to process, or reference. Changing tracks in the playlist will start a new empty chain with all settings saved for all previous tracks. There is a waveform display so you can trim in, and out points and fades. The waveform display is also where you can use snapshot automation for the processing chain.
The final notable feature in Standalone mode is ARC integration. ARC is a room compensation system sold separately
Basically, running T-RackS 3 outside of the DAW let’s you load any number of stereo audio files, process them and export to new files including sample rate, bit depth conversion with dithering.
One of the benefits of working in standalone mode is the semi-closed system without access to your hundreds of other VST plugins. You can focus on using the excellent tools available in T-RackS.
Mastering with T-RackS Singles
On the other hand, I already have a pretty good mastering workflow and template in REAPER. One of the things I’ve been doing lately is parallel MS compression which pumps up the RMS level, solidifies the centre channel and widens the overall image. I can do this with the parallel chain but I like having my reference tracks, source track, processed track and parallel effects layer out on faders in front of me. This is beyond the ability of T-RackS shell or standalone.
Another trick is to do band-specific upward compression. Kind of like splitting up a multi-band compressor to separate tracks for each range. By using the linear phase EQ (and latency compensation on in the DAW) you avoid the nasty phase shift you’d normally get trying to blend a band-passed signal in with the full-range source.
T-RackS 3 Vintage 670
5 of the 9 processors allow linked stereo, dual mono, and Mid-Side operation which opens up some creative possibilities without the hassle of encoding and decoding manually MS or splitting the tracks to dual mono manually.
All the single plugins could be successfully wrapped and used with Automap and my Nocturn Keyboard. There are probably too many parameters in the EQ’s to use automap but all the dynamics modules work great with some hands-on controls.
One feature I’d really like is a sidechain input for all the compressors, or at least an internal sidechain filter like the Classic Compressor.
T-RackS Mastering Strategy
The chain I usually start with is Linear Phase EQ, Vintage 670 (Fairchild), Vintage Tube EQ, Classic Clipper, Brickwall Limiter.
The Linear EQ is for corrections, 670 is for glue or widening if required, Tube EQ for wide tone shaping. The Clipper and Brickwall Limiter work together to get things loud. I find setting these up first, then jumping back to the corrective EQ to be a good strategy.
Authorization
I’m going to deviate from the review for a second to applaud IK Multimedia for dramatically improving their authorization process. The new Authorization Manager makes online activation very simple now. I’ve heard complaints and fumbled my way through the old method many times and it was confusing. Now, effortless. Thanks IK!
This is Awesome
The effects are easy to use, map well to controllers and sound great.
In the deluxe package you get both clean (opto and classic) compressors, and the colourful Fairchild model. You also get two clean EQ’s (classic and linear) and the vibey vintage Pultec model.
The Clipper module is excellent to push up your master levels but it’s also great in the mix where you don’t want to hear a change dynamics or tone but just want to set a ceiling for the track. Kick and snare is where I’m often using the Clipper, the hard knew can sharpen the attack and make it cut through the mix and soft knee is very transparent.
I don’t often need a linear phase EQ but when I do, I want something good. The T-RackS Linear EQ is great, so much better than the very expensive Waves ones I’ve used.
All the knobs smoothly respond to mousewheel.
T-RackS 3 Linear Phase Equalizer
Needs Improvement
There are a few areas where I see IK could stand to improve T-RackS, small things but I think they’re worth saying.
In the chain view of T-RackS Shell, an easy way to rearrange modules, such as drag & drop, would be nice.
Sidechain inputs (or at least HPF sidechain) for all the compressor modules
Button to open the user guide. There are no tool tips explaining what the various functions do and the manual is actually saved somewhere you won’t find it.
Install the documentation to the IK Multimedia/T-RackS folder in My Documents along with the presets instead of hidden in the Mac library documentation folder.
Adjustable graph scale for EQ modules. 40dB of gain is far more than you’d ever need in mastering. I’d like an option to limit the graph to +/- 6 and 12dB.
The Future of T-RackS
I’m interested to see where IK Multimedia takes T-RackS next. The Black 76 and White 2A compressor/limiters are not devices typically found in mastering but are essential rock mixing tools. I’d like to see more classic hardware like the Distressor, Massive Passive, and if we’re going with mixing tools, a Roland Space Echo and EMT Plate.
I’d also like to see IK’s take on console and tape simulations which yes, are the thing everyone is doing, but certainly essential mixing and mastering tools.
I have no insider info and I don’t want to start any rumours but I would not be surprised to see a couple new additions to T-Racks in 2012.
Mastering with T-RackS Tutorial
Learning T-RackS
Learning T-RackSI only skimmed through the manual but I did watch the great Groove3 tutorial hosted by Michael Costa on mastering with T-Racks, and it’s actually a great general mastering primer. It’s currently on sale for $10. Check that out here: Mastering With T-Racks
Also available is The Official Guide to T-RackS by Bobby Owsinski. If you’re looking to go beyond the manual and want an in-depth, plain english explanation of the effects and WHY you’d use them, check it out: Mixing and Mastering with IK Multimedia T-RackS – The Official Guide
TL;DR
T-Racks 3 Deluxe is an awesome bundle of audio effects for mixing and mastering. At the current promotional price, especially for upgrades, it’s a fantastic deal. The software is a few years old now but still holds up well vs the competition and is still being updated. There is currently a promotion and group buy for T-Racks 3 Deluxe, click here for more details.
In this article I’ll explain how I use Mid-Side processing on stereo sources for practical or creative effects.
Mid-Side? Two channels of audio can be combined in a way that gives us control over what is the same in each signal, the middle, and what is different, the sides. The middle is where the kick drum, snare, bass, vocals and a lot of other instruments are, the sides have any hard-panned instruments and spatial effects like reverb. It can be pretty interesting to listen to music like this, there can be a lot hidden in the side channel.
MS is also a stereo microphone technique using a cardioid microphone facing the source and a bidirectional mic turned 90 degrees away just picking up ambience. In this situation the two signals would need to be decoded into stereo. The side mic signal is duplicated, polarity inverted and the two side signals are then panned hard left and right. This is not a true stereo mic technique but can sound very nice. The balance of mid and side signals can be adjusted as needed by changing the level of the 3 tracks.
You can manually encode and decode stereo files to MS and use mono plugins to process mid or side individually. A lot more plugins have an MS mode now. Many of the modules in the T-Racks suite allow mid side processing, as does Ozone, a few compressors and equalizers and a distortion also come to mind.
You can do this for subtle or crazy effects, its a fun way to experiment with plugins and get some unique sounds.
Loud and wide
For a recent mastering job I used a Fairchild compressor plugin in MS mode (Lat/Vert) to compress the middle and increase the level of the sides. I did this in parallel so I could blend the effect in easily. I was also using this to get a lot of extra loudness. You can call this parallel MS Compression.
Compare the master without the parallel MS compression, then with, then the parallel compression soloed.
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Parallel Mid-Side Compression with Fairchild
No more messy verb
I had someone ask about clearing up the middle of a mix when using a lot of reverb. Using Mid-Side Compression on the reverb return can work well. Compress the middle more than the sides and increase the side volume if you want more width.
Here is an example of that on some drums. The drums are Steven Slate playing in KONTAKT. The whole kit is sent into Valhalla Room. With the Fairchild after the reverb I’m lowering the middle by 2dB and raising the sides by 2.
Here you can listen to this effect with lots of reverb on the drums.
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An now with MS compression on just the reverb bus.
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There is NO compression on the drums themselves, I’m only compressing the reverb return and widening it.
Wacky effects
Here is an example of what you can do with a stereo loop and any plugin. This is a little more complicated, and only works if there are hard panned sounds. The loop I started out with had a hihat that wasn’t panned very hard, I copied it to a new track, filtered out all the lows, boosted some highs and then panned it hard left. I recorded the combined original and panned track to a new file.
Here is what I’m starting out with
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Now that I had something on the sides I could mess around with Mid Side Processing.
The first thing you have to do is convert Left – Right to Mid and side. I use the free +matrix MS decoder from SoundHack.com. After that I used a delay plugin to add some filtered echoes just to the middle by disabling the right side input.
In the next insert I used a distortion on just the right side. This brought out a lot more of the reverb than was heard in the original loop. Lastly, second MS decoder was used to bring it back to stereo.
Soundhack +matrix MS encoder/decoder
Here is how the loop sounds now with delay in the middle and distortion on the sides.
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Pretty cool right!? I hope you have found these tricks useful.
Several times in the past few years of doing the Home Recording Show podcast, we’ve had listeners write in asking about phase and often wondering why we can’t just move the tracks around after recording. Usually we answer saying that it can be done for guitars and a few other sources but never on drums and its not the same as actually moving the mic.
In the past month I’ve actually done this technique a few times on some tracks that were a little carelessly recorded and some others that just needed a little help. In all cases it has helped. Even if you’re super careful about mic positioning, this can be a very useful technique to know.
Click to view
Before I get into techniques and examples, I’ll give you the best reason I can think of for why shifting a recorded track is not the same as moving a mic.
Its not the same in any case where there is bleed or off axis sound. If you move the mic, you’re changing the off axis sound as well as the direct sound. If you time adjust, you’re just changing the relationship of that sound to another, the direct sound and bleed move together. If you time adjust by any large amount you could end up causing more problems because while the direct sound sources are in phase, the off axis sounds are not. You may also run into a situation where that causes an echoing effect when combined with other mics.
With that said, if there is already a problem with the tracks, its worth a try.
If you’re confused about what this is all about, let me play some examples.
Guitar Example
I was given some guitar tracks as part of a mix. There is an SM57 and an AT4033 on the amp. I don’t know the exact positions other than that the 4033 was a little further away. The mics sound ok on their own but are completely useless when combined.
Shure SM57
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Audio-Technica 4033
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Both mics combined.
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Inverting polarity on one mic.
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Techniques
As you can hear, there’s an obvious comb filtering issue and it doesn’t go away with the polarity switch alone.
At this point we have a few options
get rid of one of the mics. The downside being that either mics is kinda boring
time adjust by nudging the second file earlier
time adjust by delaying the first track
Options 2 or 3 will have the same result. One method is not really better or worse.
I don’t like option 1. It makes me feel lazy.
With either option 2 or 3 I need to get a rough estimate of how much offset there is. This means finding a transient and zooming in close.
When it’s the same source with two mics the waveforms should be fairly similar. Find a transient on the first track and drag a selection to that peak on the second track.
Set your timebase in the DAW to samples and you should see how much of a delay you need to compensate for.
You can use either method to time adjust.
In Reaper there is brilliant feature that makes nudging the audio in this type of situation very easy. Reaper has an option to show a mono waveform of the combined active tracks within a folder. You can actually see the two waveforms stacked. Simply drag one of the tracks and line up the waveforms. As far as I know this only works in Reaper. [see image up at the top]
In Pro Tools you would set your nudge value to be the same number of samples we calculated earlier. Nudge either the close mic later, or far mic earlier.
If the mics were fairly close together it should be under 300 samples. In this case it was just 56 samples.
If you want to use a plugin for this, find one that works in samples and enter the value.
Let’s here how these mics combine after adjustment.
Time adjusted combined mics
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SO much better, and we have a sound that is more tonally interesting than either mic alone.
Ensemble example
Here is another situation. This is a 3 track live recording in Cuba I was given to mix and master. There are two Neumann M150s omni tube condensers in front of the musicians and a cardioid mic within the group to pick up vocals and percussion.
Original mix
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I found that shifting the tracks slightly improved the center image and made the recording sound a little more focused, it is a fairly subtle change. In this case I used the left side mic as the target, as it was latest and adjusted the right side by 194 samples and center mic by 513.
Time adjusted
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Drum kit example
If you’re still interested in this at all, you probably want to know how it works on a drum kit. Until today I haven’t tried.
I have a drum kit with 2 overheads, kick, snare and 3 tom mics. I’m going to use the snare as the standard and move the other tracks around to match.
Here are the drums with just levels, panning and polarity. No EQ or any other processing .
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On the drums, at least on this recording, the change was very subtle. The longest delay was 190 samples between the snare and overheads. Only 4ms. That’s a tiny amount. The result is a little less wide having removed the distance from the overheads to the snare.
After timing adjustment
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Sometimes it makes a huge difference, sometimes its subtle. I think its worth a few minutes of experimenting even when you are very careful about your mic placement.
This guest post comes from Barry Gardner, mastering engineer at Safe And Sound online mastering.
You may also like his previous contribution to AGZ, The 24 Bit Advantage.
Analogue warmth, what it is, and how to inject it into your recordings and mixes.
Warmth in the context of audio production, is a hot topic. With the popularity of almost all digital signal paths, it has become much cheaper to record, mix and master your music. However many musicians, producers and engineers feel that there is sometimes an elusive sound quality missing from modern digital production methods. In modern times, three common techniques in music recording and production have changed since domestic and DIY audio production has proliferated. These are namely, the use of multi-track tape machines, large format analogue consoles and large recording studio spaces. These changes are the more obvious ones and have definitely changed the quality of audio.
Defining warmth in recordings and mixes.
Analogue warmth is subjective and difficult to describe in words and everyone’s interpretation is slightly different. However there are a few statements which appear to be commonly accepted as characterizing warmth within a mix or recording.
A gelling of instruments, yet without compromising their own defined space in both tone and stereo image.
A lack of harshness and rounded quality.
A punchy and smooth sounding lower mid range.
Not excessively bright and the brightness that is present is smooth.
A punchy low frequency range.
To me personally, warmth can be a number of things. I can recall analogue tape recordings from the 1970′s which I would define as being warm. I can also produce something I define as warm with simple attenuation of high frequencies. In some instances a very rounded sound with a strong lower mid presence can sound warm to the ear. With this in mind I would like to suggest some pointers on how to create warmer sounding mixes and recordings.
What equipment and techniques can we use to enhance warmth?
All music recording starts with setting up microphones, experiment with different mic positioning or mic choices in order to get less bright recordings, although take care not to box yourself into a corner, double mic instruments with secondary ribbon or dynamic mics in addition to your usual choices.
A common source of warmth can be certain audio transformers which can reduce harshness in the upper registers and provide additional body in the lower mid range. There is a wide selection of vintage and retro styled mic preamps that utilize audio transformers at the input stage. Audio transformers are usually used at the inputs and outputs of equipment and can be found in many outboard equipment types such as equalizers and compressors as well as microphones. They are often overlooked in the quest for warmth.
One of the most powerful tools which is overlooked for generation of warmth is equalization, you have the power to sculpt and adjust sounds as is required. Do not be afraid to experiment with rolling off high frequencies to reduce harshness, presence and brittleness in a mix. You can also employ EQ on effects returns to soften them and make them gel better with the source.
Compression has the ability to smooth transients in recordings and fast attack times with 1dB or so of gain reduction can work wonders in smoothing out abrasive, harsh and aggressive transients in a mix. Analogue tape applied a natural form of compression when overloaded gently. It is a technique that can be used to good effect. Very gentle group or master bus compression can also provide a sense of “wholeness”.
In addition to these essential tools, in software form there is an emulation of virtually every piece of classic analogue studio equipment ever built. Often these software emulations rely on some kind of valve/tube saturation. In my experience valves do not add warmth as such but they can give a perception of thickening a sound as harmonics are added. Some emulations are better than others and I suggest keeping an open mind and downloading some demo’s and spending some quality listening time with them. Try and discern which ones seem to add that special something in terms of tone.
By experimenting with these techniques and equipment choices you should be able to start adding some warmth to your mixes. As always when experimenting in audio production take some time to rest your ears over night and double check that you have not laid the processing on too thick.
This guest post comes from Barry Gardner, mastering engineer at Safe And Sound online mastering.
You may also like his previous contribution to AGZ, The 24 Bit Advantage.
This guest post comes from Steve Hillier, a songwriter, DJ and record producer, who has worked with everyone from Keane to Gary Numan. Steve is also a journalist and music technology expert, writing for Future Music & BBC Worldwide. Steve teaches Music Business and Logic Music Production Online at Point Blank Music School
Master the use of reverb and your lifeless, two-dimensional mix will become a three dimensional panorama, says Steve Hillier.
Things that people do wrong with their music:
1. Write a composition starting with the drums. This is madness. Can you imagine Lennon and McCartney waiting for Ringo to set up his drum kit before writing their next Beatles smash? Obviously not.
2. Compress everything. At least twice. Anyone doing this in their mixes should stop now. Modern DAWs have an internal dynamic range that’s comparable to a pin dropping versus the sound of the big bang. Try using it, rather than squashing your music to the flatness of a pancake being sucked into a black hole . Compressors are like guns…only the sane should ever pick one up.
3. Use reverb badly, or not at all…
Unlike compression, everyone likes reverb. How can I say this with such confidence? Because nearly everything you’ve ever heard has been covered with reverb. Everything. Reverberation is what you hear when the sound from an event, such as a gun shot, bounces off a reflective surface, such as a wall, and then into our ears. It’s a fundamental attribute of how we experience sound, and our brains have evolved to use the information contained in reverb to help us survive in our everyday lives. If we’re hearing lots of sounds with long reverb tails on them, that suggests we’re in a large room, such as a church. Lots of short ‘early reflections’, we’re probably in a small room. Everything we hear has some reverberation on it before it ends up in our ears (we’ll ignore scientists who work in anechoic chambers for today).
Too many novice programmers don’t know how to use reverb, so they shy away from it, leaving their mixes dryer than Stewart Lee. Or they go the other way and use completely the wrong reverb sound, and get wetter than a Michael McIntyre show. Maybe programmers are confusing acoustic size with acoustic impact? Imagine this text on your page is your tune:
This is your mix,
This is your mix with the correct use of reverb on it,
Here’s your mix with a little too much reverb on it,
And here it is with way too much!
The effective use of reverb will make a component of a mix sound bigger, fuller and more comfortable for your audience. Without it, the sound will be tiny and illogical; think about it, in real life when will you ever hear a big dry sound? The answer is never. Ever! On the other hand, too much reverb and the mix will be wet and flabby, too big for anyone to comprehend.
How to use reverb:
So what do we do then? First, you need a decent reverb unit or plugin, don’t use just any old reverb plugin. I have a theory that the reason that reverb went out of fashion was related to the fact everyone used way to much of it in the eighties. And many of them were using horrible cheap digital units*. There’s no excuse for that today. Invest some money and buy one each of both of these:
A convolution reverb unit works by generating reverb tails based on impulse responses, recordings of reverberations from a real-world environment. They sound amazing; the best are extremely realistic and open up a world of possibilities. But you’ll need a traditional digital reverb too, probably a plugin based on classic hardware form the past. Since the late 1970s and up until about five years ago pretty much all reverb on records was simulated in some way, often by a microprocessor delaying audio, feeding it back into itself, doing some clever filtering and sticking it out the other end. It sounds great, if a little synthetic. But who cares? This is the sound of records, and they still sound great now.
Here’s how I use reverb in my own work. Your mileage may vary but most mix engineers I know use this approach or a variation on it:
1. Set up three reverb plugins as send effects on a bus, not as insert effects. The first will be short (less than 0.5 sec) and come from a convolution reverb using a room impulse response. The second will be a traditional digital reverb sound, such as a plate reverb, set to around 1.5 seconds decay. The last will be a ‘third option’, normally reserved for vocals and normally another plate or hall sound.
2. I then balance my sounds without reverb. Please note that I only use the bare minimum of compression at this point too!
3. When I’m happy with my mix, I then start placing my sounds in an imaginary three dimensional space. The shorter reverb sound places the drums and other high energy or rhythmical sound sources at the front of my stage, the larger reverbs put those sounds slightly further back and into a supporting role. The more reverb, the bigger the sound but also how far away it is.
Thinking of your mix as a three dimensional illusion is crucial for a comfortable and exciting result. Without reverb, your mix will sound like it’s stuck inside the speakers. Reverb brings the sounds alive and gives them the opportunity to leap out of headphones!
Why do so many programmers get this bit wrong?
What this all comes down to, time and time again, is the disconnect that bad programmers have between their brains and their ears and their music. They get into the habit of searching for answers to why their work isn’t working with the same cognitive tools that they use to explain why their internet router isn’t connecting to their laptop. This is not how music works. Our ears and our hearts should guide 99% of our musical work, the remaining 1% comes from experience and knowing how to use our equipment. So, from here on, start listening carefully to what’s going on around you. Listen to the difference between the sound of talking voices in a car and in the street. That’s reverb. Listen to the difference between the sound of tune in a club or in your iPhone headphones. That’s reverb. Listen to the sound of you brushing your teeth in a tiled bathroom. That’s reverb. And then, listen very carefully to the difference between your lifeless, static, two-dimensional mixes and three dimensional panoramas of the artists you most admire.
*Actually, the MIDIverb does have some great uses and you can probably pick one up for nothing at a jumble sale now if you look hard enough. Just don’t use it as your primary reverb tool.
For more inspiration on using reverb have a look at these videos:
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This guest post comes from Steve Hillier, a songwriter, DJ and record producer, who has worked with everyone from Keane to Gary Numan. Steve is also a journalist and music technology expert, writing for Future Music & BBC Worldwide. Steve teaches Music Business and Logic Music Production Online at Point Blank Music School
