Interview: Ignite Amps

Ignite Amps is one of the most interesting artisan amp manufacturing realities in Italy, which not only builds Guitar and Bass Amps and stompboxes, but also releases the free Vst version of their products, with a very smart marketing strategy, to get the people to have an idea of how their products sound like before going to Rome to try them in person.
Here's our chat with Carlo Costabile and Federico Berti.

GuitarNerdingBlog: Can you give us some detail on how Ignite Amps was born?

IgniteAmps: It all started back in 2006. Right after finishing the university studies we kept in touch, eventually talking about guitar and bass amplifiers and home recording.
Being really passionate about engineering in general, one day we came out with the idea of building a clone of a famous tube pedal.
Our knowledge of electronics was quite limited at the time, especially regarding tubes and musical devices, which were absolutely not mentioned during our university courses, so the idea itself was
pretty ambitious already, but it turned completely insane when we decided to ditch the tube pedal project in order to build a dual channel tube preamplifier clone, which was obvoiusly even more risky due to the lethal tensions for which the circuit was designed...
Long story short, after a few months of work in an I-have-no-idea-what-I-m-doing fashion, we got the lead channel working. Considering the wiring and the board material used [we're not going through details about these, as it would be REALLY embarassing], getting the circuit to output sound could be seen as a miracle already, but the cool thing was that it sounded AWESOME and it wasn't noisy AT ALL especially if considering the insane amount of gain that thing had. We never got to finish its clean channel and that's where we got one of our most famous taglines: "clean channels are for pussies"!
After that experience, we continued with other projects, improving our knowledge, design skills and raising the bar of our builds every time.
That was incidentally the time when Stefano of 16th Cellar Studio put us in touch with Cristiano of Fleshgod Apocalypse.
He wanted a badass yet light amp for touring. That's when we came out with a 3 channel tube preamp kit we assembled with some mods [that was the ORIGINAL NRR-1!], mated to a class D 180W poweramp. The whole amp weighed like 9.5kg. Looking back we should have done a thousand things differently, but we were technically young and we still owe Cristiano for giving us his trust. That was something that really changed what was more of a hobby at the time into what it is now.

In December 2009 we decided to start working on some DSP algorithms to simulate tubes and circuits. As usual, we didn't realize the amount of work and knowledge required for such a task, but we eventually got a first tube stage prototype algorithm working after a few months, which was then extended to a complete preamp, the NRR-1 simulator. We didn't realize the potential of such algorithms until we did some A/B comparison with the hardware NRR-1 and after having an hard time discerning the two, we decided to use these modeling algorithms for circuit design and prototyping. This changed our workflow completely: being able to play through a real time simulation of the circuits before even switching on the soldering iron is a HUGE advantage, allowing us to drastically reduce the cost and time of hardware prototyping. We released the NRR-1 plug-in to the public for free in November 2010, receiving overwhelming feedback and bringing the Ignite Amps brand to the public, opening our official site and Facebook page.
I guess you know the rest...

GNB: What do you think about the amplification business nowadays?

IA: This is a complex question calling for an articulated answer. We'll try to make it short but it'll be a hard one!
Many customers are smart nowadays, this mostly applies to professionals, but there's also the average player who wants EVERYTHING from an amp: low cost, great quality/reliability and good tone. Realizing you just can't have these three together is the basic difference between being an amp expert and the average enthusiast.
Want low cost and good tone? You can get a 40 years old circuit with some mods, built by slaves in the far east, but don't come crying to me if it's noisy or when it breaks. And it will. Soon.
We're conscious we've almost been stating the obvious here, but when you get tons of emails from people asking the equivalent of a custom, 800hp sports car full of extras, built in Italy with top quality components and you give an absolutely honest quote FOR WHAT IS BEING ASKED and, if and when you get a reply, many a time you're told the price is too steep, you come to realize it ain't that clear to everybody out there.
Fortunately, the professional musicians and the people who have owned lots of amps know the real thing and honestly, that's the people we like to work with: conscious players who know what they want and know what it's worth.
On the other end of the spectrum, where the big money is, we see A LOT of marketing paired with small substance even in the pro field. There's big companies out there that, guess what, look for profit. Sure as hell we do like to make money out of our products too, but we also want our customers to be proud of their choice.
We see lots of cool videos on the net, huge endorsements, trendy signature series by many notorious brands, be them big or small. They all look uber cool on the internet. Then, many a time, if you get to play that particular piece of gear, you see it's the typical mass production amp/pedal, no more, no less. Build quality might be good, but components and design aren't always on par, or the tone isn't just THAT good as it seemed on the web. Great artists endorsing said gear are paid to say it's great stuff so I'd take those enthusiastic statements with a pinch of salt before buying. Mass production obviously has also its advantages: hopefully great customer service and good resale value retention, but a mass produced amp just won't be that beast of a head made in few numbers, customized around your needs, built with the best components and using the utmost care during design and assembly. And when one of your friends lets you in for a test drive with his custom amp, you'll definitely feel the difference! ;)

GNB: What is the design and build process of an Ignite Amps amplifier usually like?

AI: It always starts from the input of the musician. Once we know his taste and what he's really looking for, we turn this information into a prototype schematic.
Once we have the schematic drawn, we turn it into C++ code and build a circuit simulation in plug-in format, which we then send to the musician. In this way, he is able to play through it and provide us additional feedback which we then use to modify the circuit and tailor it to his exact needs. The loop goes on until the musician is 100% satisfied by the sound of the simulation. In this way, he can also try the sim with his own guitars and cabs.
We're pretty positive we are the only company in the world at the moment that can offer this type service, most of all with this level of real-time simulation precision.
Once the circuit design is finalized, the circuit layout is designed and a 3D model of the complete amp is created, containing almost every single component.
This prevents problems related to wrong measurements and erroneously calcutated clearances which may arise during the building phase. You can't really have that type of error in a one-off build.
The circuit is then assembled using turret boards built by us or industrial grade PCBs designed to our specs. The external look of the device is decided together with the musician, according to his taste.

GNB: Which components do you normally use for your products?

AI: Depending on the price range, we can use kits we assemble with some of our addons and mods, down to full custom amps built from scratch, and that's where the fun is, honestly.
In the latter case, we get almost all of the parts made on our spec: the chassis, the shell, the transformers. We got our transformer supplier which is an italian custom shop and we get all of our trannies wound on spec, this gives us the advantage of having the trannies with all of the features we need, with the right dimensions for the specific project.
Capacitors and resistors are usually finest quality altho we try to stay away from the hifi-like craze where people end up paying 50 bucks for a cap because some forum guru told'em to.
Lately, we've been also using laser engraving for the panels labeling. We think this is by far the most durable way to label the controls on an amp. It might be more expensive than silk screening but we've seen a good number of amps with the labels worn out after some years. This is another small aspect that underlines the difference between commercial stuff built for mere profit and gear that's built to last.
In the end, we try to get all of the custom parts made in Italy and then we use the usual web suppliers for the caps/resistors/pots.

GNB: What is the philosophy behind your software: analog modeling, black box approach or else?

IA: Analog modeling. Black box/Profiling has been used with pretty good results lately, but for our workflow it would turn completely useless. In order to create a "profile" one would need to have the hardware at his disposal, implicitly killing all the advantages of having a simulation algorithm before the actual build.
With all due respect to the developers who follow the black-box route, we consider analog modeling also vastly superior. Apart from the obvious advantages to create circuit prototypes at no cost, being analog modeling based on electric and math laws at single component level, the modeled circuit maintains its consistency for every parameters variation. With a black box approach you take a snapshot of the sound at defined settings, but once you start turning the knobs, the black-box algorithm may only guess what happens, eventually deviating from the real hardware behaviour by a significant amount. One may be thinking to circumvent this issue by simply taking multiple snapshots at various settings and then perform some sort of interpolation... well, good luck with that for devices with something like 10 controls, in which the position of each one of them influences the behavior of the others, like in all the amplifier tonestacks! 10 dimensional interpolation, right? Because fuck you, CPU usage and RAM allocation size, that's why!
When developing software for musicians/audio engineers, sound quality is indeed the most important aspect, but it's not the only one: usability, stability, responsiveness and resource usage are also key factors. Some developers like to hide behind their "100% accurate, no compromise tone" claims to justify the huge resource usage of their software, when it all comes down to their lack of coding/optimization skills.
We always pay a lot of attention to all of these aspects. In real-time audio processing, expecially in circuit simulation, the "no compromise" approach just doesn't exist yet. CPUs are not powerful enough, there is always a compromise to be made between accuracy and performance, if you want your software to be usable. The choice of the right balance is what discerns a professional developer from an amateur.

GNB: What have been your carreer highlights?

FedericoIA: I guess the Emissary plug-in release. The response has been really overwhelming, after 2 days we were litterally flooded by audio clips and videos by enthusiast users from all around the world, we scored the 3rd position over 37 entries during the KVR Developer Challange 2014 and the Emissary was the most clicked plug-in during the whole competition, reaching the 1st position in the absolute KVR rank 2 months later. Add to this that KVR is not a community of guitarists and bassists, but of all kind of musicians, most of which have no interest in amp-sims!
Plus, having the pleasure to work together with Voger Design for the Emissary GUI was such an inspiring experience! They really took it to the next level!
I also loved designing the circuit of this amp from scratch, as I've been able to put a lot of ideas I had in mind since quite some time into it and to see them gradually turn into a great product, thanks to my Ignite Amps mate!
We recently recorded some A/B comparison tests between the real amp and the plug-in... the only thing I can say is that they left me completely speachless! Keep an eye on our official Facebook page and you'll soon hear whan I mean.

CarloIA: as we're raising the bar build by build, I can only say that the Emissary is the showcase of what Ignite Amps can really do at this stage of its life.
We've built 1300W bass amps weighing less than 5kg in the past, we've had cleverly designed mods to well known circuits in the past, but this latest amp is the special one to me and I'm sure the next one will be too and so on.
The lead tone is very balanced, very mean and very precise at the same time, Fede really outdid himself with this. The tone shaping is quite complex here: the main circuit board alone is almost 50cm, can you imagine that for a 50 watter? That is a huge number of components! Then there's that cool, channel specific input loop thing so you can have a different set of pedals in front of each channel with a single switch press. There's the bias for each tube and the global bias control too. There's one really clever fx loop on the back which can go from full serial to full dry and everythng in between and then there's the switchable solo volume boost... that is a cool set of clever features over an already kick-ass circuit. The build itself took a lot of time as we're pretty obsessive-compulsive with the wiring haha!
The amp weighs just 12kg as it's all aluminum, including the shell. I think it also has a clean channel somewhere! :D
Yes, we had this feeling there's people who actually find a use to clean channels, so we spent some time designing what we thought a real clean needs to be. Turns out we pretty much nailed it with a super cristalline circuit that can also get pretty warm up to slightly overdriven if you push the gain up. The 14 people who tried it say it's got a great clean tone! :D There should also be a coupla clean clips out there on youtube I guess.
The whole project was unbelievably cool: first having a musician who knows what he wants as Ryan Huthnance ringing in to ask for an amp, trusting this small company from the other side of the world, then the Voger Design guys taking care of the sim GUI, to end up with the web spending flattering words about the tone of the sim. I swear I still have to read a negative review about it.
The whole project was a complete win on all fronts. It'll be a hard one to beat but rest assured we'll do our best to top it off.

GNB: What do you think the future of analog and digital amps will look like?

IA: Digital processors and plug-ins are taking over the guitarists/bassists needs for sure, but we think tube amps will hardly ever disappear. The convenience of digital processors (cost, weigth, dimensions, versatility, etc.) cannot be denied, but tube amps will keep being used for a long time, especially in recording studios for all the analog-enthusiasts.
We love digital technology, but we also love analog elecronics and handcrafted products, so we'll surely keep on focusing on both areas doing our best on each front.
Digital simulations have become better and better with the increase of processing power and this will continue in the near future, although with a lower rate than one may think, since microprocessors manufacturers are focusing on parallelism by increasing the number of cores, rather than the raw processing power on the single core. Unfortunately, audio processing doesn't take much advantage of parallelism due to feedback loops and systems stateful-ness in general, so developing increasingly complex algorithms to push this technology forward will keep being a highly challenging (but fun!) task.

GNB: Thank you guys and keep up with the good work!
You can find all the info about Ignite Amps and the free VST version of their products on their webpage.


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A different way of editing: BENDING AUDIO.

Hello and welcome to this week's article!
Today we're seeing in-depth how to edit a track, in a different and faster way than the "Slip Editing" method.
In Pro-Tools this method is called "Elastic Audio", in Presonus Studio One V2 is called "Audio Bend", but the way it works is more or less the same: we're talking about a tool capable of detecting the transients of a wave track ad to align them more or less automatically to the tempo track, avoiding us the need to do this long and boring task manually (click here for a dedicated article).

Today we're taking a look specifically in the method used for the Presonus Studio One V2.

- First off obviously we have to set the Tempo Track of the project, so the Daw knows where to move the transients.

- Then we Click on the Audio Bend icon on from the top menu, and from the sub menu that appears we click on "Analyze".

- After the software has analyzed the track and detected all the transients (for example all the snare hits), we can specify the threshold from where to start moving and the type of instruments (e.g. Drums). The software will generate a "Bend Marker" for each transient.

- From here we can choose between two things: Quantize, which will move all the transients on the midi grid, or Slice, that slices the events, turning the single track on a serie of separated events.
We can also eliminate the silence between the transients making the track a serie of slices in which only the audible part is shown.

- Once we have selected what to do with our track (the Quantize option is the standard choice) we just click on "Apply" and the software will move the transients according to the global click and quantization. You can also, with a specific function, choose the precision of the quantization (to make it feel a little less mechanical) or add some swing to the track.

Hope this was helpful!


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Review and tutorial: Fabfilter PRO L

Hello and welcome to this week's article! 

Today we're reviewing a different kind of limiter, the Fabfilter pro L.

When we have extensively talked about limiters, headroom and gain staging and mastering, the plugin always taken as our typical limiter was the Waves L2, an industry standard, but actually in the market there are many more limiters which have the same function, but radically different interfaces and mechanics.

The Fabfilter Pro L is one of those.

The first thing that captures the attention is the rich wave analyzer that takes most of the ui: to the right there are 2 metering tools: the brightest one is the input level, the darkest one is the output level once we have raised the gain, and the red part is the amount of limiting applied.

In the main window we can see the actual wave, with in red the parts that are being limited in real time, and it is possible to choose the range to show in the window, from the full dynamic range to only the peaks that gets reduced.
The gain fader is on the left: just scroll it upwards to add more gain, using all the available headroom, and stop when the moment is right: when only the peaks exceeding the average level are limited, and never more than 3 or 4 db of gain reduction, to avoid losing transients.
The output knob in the bottom right lets us set the ceiling, according to the main use we want to do with our song: -0.1db if we will hear it mainly on a cd, -1.0db if the main use will be internet streaming, and so on.

The typical use of this plugin is the same as the Waves L2 explained in the "How to use a Limiter" section of this article, which is linking the gain knob to the output one and raising them accordingly until the gain reduction is the right one for your material (sure there are many good presets too, but especially when limiting, presets are almost useless, since they cannot know exactly the loudness of your song, and they will often distort the material): luckily you can browse through the presets also using a function that will not affect gain and output level, just to see how the different settings sounds.

If you want to manually tweak the settings, as in the other Fabfilter products there is an "Advanced Tab" which lets us manually tweak attack, release, lookahead and other parameters as the "limiting algorithm": transparent, punchy (good for single tracks), dynamic (good for preserving transients), allround.

All in all, another very good and complete product from the Fabfilter team, which also integrates a complete monitoring system that will make our mastering chain thinner.


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How to use a De-esser (or Deesser), with Free Vst Plugins!

Hello and welcome to this week's article!
Today we are going to talk about de-essers!
A deesser is basically a compressor which is set to operate only on a very narrow range of frequences that corresponds to the sibilants of the human voice, which can be captured with high peaks of volume by the microphones.
Usually these sibilant "S" and "T" sounds are located in an area between the 9khz and the 15khz, and often the deessers have presets for male or female vocals, but the best ones sometimes features also a "learn" mode in which they automatically detects the right area to attenuate: the more the area is narrow, the more the deesser will be effective without processing unneeded areas of the mix.

There are also other, more creative ways to use a deesser: someone uses them on a cymbals track or on a distorted guitar track to tame the harsh frequences that sometimes can get a bit too prominent, giving a more smooth result, similar to the one you'd get from a Low-pass filter, but less invasive.

There are many Deessers on the market, and almost every DAW has one bundled, but if you want to try some free Vst, here are best ones the web has to offer:

Digitalfishphones Spitfish - Considered one of the best deesser ever made

Tonmann Deesser - One of the most transparent free deesser around

Sleepy-Time Lisp Deesser - A very good Deesser, available also in 64bit version

Antress Modern Deesser - An interesting rack-like deesser


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Review and Tutorial: Fabfilter PRO C

Hello everyone and welcome to this week's article!
Today we're going to talk about another Fabfilter product, the Pro C Compressor.
By now, after all the articles I've done, we all should know what a compressor is and how does it work: basically it's an automatic volume control that lowers the lowest peaks according to how we set it, and raises the quieter parts.

Plus it's an important tone shaping tool.

I have particularly liked this specific compressor because of its flexibility, which is similar to the Pro Q 2's: the interface is clean, easy, and at the same time there are tools to visually monitor what's going on that usually are not featured on other processors of this kind: I'm talking about the dual graph:

- the left one is pretty common on many compressors, and tells us when and how strong the compressor kicks in, and it is possible to choose between a soft and a hard knee.

- the right one is less common and it shows us 3 things: the original waveform, the amount of gain reduction we're applying, and the new waveform. This helps us a lot checking out if we're crushing the dynamics of our track too much, or on the contrary if the settings we're applying are uneffective.
The last part of the graph is a metering tool that tells us the level of our track and the gain reduction applied.

Beside the common controls that we can find on any other compressor (attack, release, ratio..), the Fabfilter Pro C features also some less common function:

- Automatic Release control, that changes adaptively according the part, and that is particularly useful if used on complex tracks

- Automatic Make up Gain to compensate the usual loss in volume due to the compression

- 3 Modes: Clean, Classic and Opto, which reacts differently, since they're modeled on different types of compressors.

- the A/B comparison function: we can set the compressor with two different settings and a/b compare them to choose the best one.

- the Parallel Compression function: we can experiment using some more extreme setting on our track, and then use the Dry Mix control to decide the amount of dry signal to blend with the processed one. This way we can add some body to the track leaving part of the transient to pass through uncompressed. This is particularly good with a snare drum, to keep the snap (the transient) intact, or with a whole drum buss.

- a Sidechain Function in which we can decide wether to use separated left and right gain control, or to use it Mid/side.

- a Midi Learn function to automate the settings: we need to push the Learn button, then move the knobs we want in real time while the song is playing, and the Pro C will record our movements and change the settings real time the same way we did, automatically.

There are even more functions to talk about (for example this plugin can be used as an effective Deesser too), and many presets to choose from, but it's already enough to say safely that this is the most complete Compressor ever made, and one of the most transparent sounding ones, too.


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How to record and mix clean guitar (with free Vst Plugins)

Hello and welcome to this week's article!

Today we're back with our mixing tutorials, and we'll see how to record and mix a clean guitar.

There are many ways to record a clean guitar, and many definitions of a "clean" tone, for example we could define "clean" the sound of an acoustic guitar (click here for a tutorial about how to mix an acoustic guitar), but today we will focus on how to how to mix a clean tone from an electric one.

First off let's say that we can track a clean guitar tone both Microphoning an amp (click here for a tutorial about how to mic an amp) or going straight to the audio interface, and we can do this directly from the guitar to the interface, or we can go from the line out of an amplifier, so that we can use the preamp sound, to the send of the audio interface, bypassing the interface preamp.

If we're going straight to the audio interface, the idea is to record on a stereo track if our clean guitar track will not be double tracked and it will be set dead center;
if we are recording two different takes instead, one to be played on the left side and one on the right one, we should use two mono tracks.
Plus, if we're going straight from the guitar to the interface, we should use a Vst Amp simulator, for example the free Ignite NRR-1, which has a beautiful clean channel, and then we can choose wether to use a cabinet simulator or not: unlike what happens with distorted guitars, for clean guitars a cabinet simulator is not mandatory, the same way is not mandatory for D.I. Bass.

Once we have tracked a good tone (the better it is at the source, the less we will have to process it later), we can add some stereo widening (if we have recorded a single track in stereo), to make it sound a bit more large in the soundscape, then is time to use some equalization: the idea is to use a high pass filter, not too invasive, to get rid of the unnecessary lows (for example we can roll off from 60-to 100hz), then we can add some snap around 2500hz and sometimes, if needed, some "air" in the 7,5-12khz area.
Remember, though, that if a correction can be done with the real or virtual amp controls, is better to fix it that way instead of using the vst equalizer: the sound will mantain its realism much more.
Next step: Compression.
The settings of the compressor vary according to the music genre we're mixing: if we're mixing a song with just few notes resonating we won't need a lot of compression, a 2:1 to 4:1 ratio should do the trick, but if we're mixing fast strumming music, such as funky, the levels must be kept even all the time, so we can go up to a 12:1 ratio to keep the performance stable.
Last link of the chain: Reverb.
As for all effects, we should use them with taste, otherwise we would screw up all the work done until now.
The reverb should have a short tail, and we should carefully set the "wet/dry" ratio, in order to give weight and space to the sound without making it drown on it.

If needed, clean guitar accepts processing with modulation effects very well, so we can use a bit of Chorus to add a vibe similar to Metallica or Pink Floyd classic clean sound, a Delay to obtain a U2 sounding guitar, some Tremolo to add a bit of western mood, and so on.

Signal Chain: SOURCE -> (STEREO WIDENER) -> EQUALIZATION -> COMPRESSION -> REVERB -> (OTHER EFFECTS)


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Review and Tutorial: Fabfilter PRO Q 2

Hello and welcome to this week's article!
Today we're going to talk about my to-go Eq Vst plugin, the Fabfilter Pro Q 2, which is basically nowadays the only Equalizer I use, both in the Mixing and in the Mastering phase.

Why do I think this Eq is better than the others?
Apart from the fact that it sounds good and it doesn't colour my source sounds too much (but this is a characteristic that many good quality Vsts have, for example some of the Waves ones), I love this Eq and I use it on every project because of its unique tools, that makes my job so much easier:

- Built in Spectrum Analizer: it lets you see the original curve and the Post Eq one, even at the same time, to see if there are peaks or resonances to tame.
I don't know of any other Equalizer that gives that much flexibility, so far I had to use at least a Spectrum analizer on each chain, and it has been a pleasure to take 'em out.

- The fact that if you're working on a stereo track (i.e. Mastering), you can have each band to work on the whole track or just on the left or right channel.

- It has an Eq Matching function, much like the Voxengo Curve Eq: you can make it analyze another track via the Sidechain Input, and it will apply the same curve to yours.

- Solo mode: this function allows you to isolate just one "slice" of the spectrum and hear it.
I find this function, which is quite rare, to be the most useful feature in the whole plugin, and it works by clicking and holding down the headophone icon that appears whenever you highlight a single band. You can move left and right to move the "slice" of the curve you're listening to, and use the mouse wheel to make the audible part more wide or more narrow.
This is the best and most precise way to point out a single area of the sound to intervene.

All in all, this plugin makes my tracks sound better and my signal chain shorter, which is very important since the more processors you need to stack on a track to make it sound good, the more problems you will have, and ultimately the more "unnatural" the final result will sound (plus the project will be heavier on the cpu).


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