Graduated as sound technician in 2000, Raffaele Pezzella (aka Sonologyst) is also a musician working in the experimental ambient field. He runs the Unexplained Sounds group, a platform to investigate the current underground experimental music scene, and plays a weekly streaming radio program every Sunday (9.00 p.m. Italy time).
Excellent review of Sonologyst’s album “Silencers. The Conspiracy Theory Dossiers”.
Naples native Sonologyst (Raffaele Pezzella) has recorded for labels like PeopleSound, Eighth Tower, Petroglyph Music, Sirona Records and Sillage Intemporell – but this is his first on Cold Spring. Martin Bowes mastered Silencers: The Conspiracy Theory Dossiers which is available on limited edition CD (digipak/booklet) and in digital format via Bandcamp. Ten ambient sci-fi tracks snake and wander remotely as heard on the title track with evasive drone and pitchy contortions. Both Singularity and Monotape set the dramatic scene here, like a refined installation or film soundtrack of warped sonic waves, a geiger counter and lots of mystery. This has a similar vibe as themes experimented on the mid 90’s ambient project SETI (Savvas Ysatis and Taylor Deupree), though here it’s more documentary-type exploration and less fantastical. On Nocturnal Anomalies there’s a disturbance, an alien being of sorts, just whaling over a hybrid hiss. This is illustrated clearly…
An interesting article by L-Rox about mastering to cassette tape.
Beyond the Nostalgia
I grew up listening to vinyl and cassettes; I’m not that old, but growing up all we had was vinyl and cassettes to listen to. The first time I recorded anything, it was on cassette. I remember my dad letting me use his Hi-fi to tape some of our vinyl records and things from the radio when I was a child and when I grew up and wanted to record some music with friends, we all pitched in and bought a 4-track cassette portastudio. After bouncing tape tracks on that, and using one of the tracks to print timecode so it could trigger playback on my sampler, the mixdown went to another (stereo) cassette deck. That’s really when I became interested in getting the best sonic fidelity from cassette tape.
So why go back to using cassettes? It’s 2014 and today is the second year “Cassette Store Day” is observed around the world; why is this format all of a sudden making a (albeit small) comeback? I think people are becoming more interested in having a tangible product. Tapes are compact enough, are portable and require less maintenance than vinyl as a playback medium. Those who are getting into recording with tapes are realizing that there are many variables involved, and it’s possible to use the limitations of the format to get a unique sound.
Those new to cassettes are also finding out that it’s an inexpensive format to record with and reproduce. You can get tapes made for a lot less than Vinyl and CDs. For those who don’t currently own a cassette deck or walkman, you can find a cheap one in the used market. A good turntable in comparison, will usually set you back quite a bit more than a portable cassette player and good luck making it portable. I’ve seen some of the really cheap Sony cassette players on eBay sell for as little as $5 (although I seriously recommend those without one do a little bit of research and get something better than these low-quality players; you can score a very decent portable tape player for not much more, trust me).
The limitations of the format can give your project a “throwback” feel, and just like vinyl, you can’t replicate the sound of cassette tape effectively in the digital realm (but I wouldn’t be surprised if a plug-in developer comes up with some sort of emulation if cassettes keep getting popular). If you want your project to sound like it’s on vinyl or cassette, you have to put it on those formats; it’s similar to why some people still take pictures with film to this day. Technically, digital pictures are cleaner and sharper than film, but aesthetically, some people like the look and feel of photographic film. Like film, there are many variables that affect its quality; cassettes don’t all sound the same (different frequency response between types and the various formulas of tape that were produced and not all of them have an excessively “hissy” sound to them for example).
Mastering for Cassette (the right way)
If you’re going to create a master cassette to possibly be the source for cassette duplication, you should do it the right way. I wish I could tell you it was as easy as heading down to the nearest Goodwill, spending $30 on a used deck and recording your tape as hot as possible. You will get saturated recordings on cassette, sure, but it’s probably not going to sound great (yes, cassettes can sound good!)
My approach to mastering a cassette is to aim for a similar level of quality that was achieved in the peak days of the media. It was typical for Mastering Engineers then to audition a cassette after mastering to it, and make tonal and dynamic range adjustments as necessary to make the cassette recording sound as good as possible before it hit the bin loop duplicator for mass production.
Up until the early 1990’s, cassette bin loop duplicators were analog devices and they used a cassette master tape as the source, often this source master tape was “emphasized” a bit for the format. Digital bin loop duplicators started to become popular in the early to mid 90’s and these used a digital source, usually from DAT or hard drive using first generation ADCs/DACs. In the peak days of commercial cassette production, a degree of effort went into creating the source master cassette or digital source, since it was known that high-speed cassette duplication would degrade the quality of the tape copies to a degree and with the usage of noise reduction systems like Dolby B, emphasis was made between 4k – 10k to make up for the loss of frequencies in that range when encoding the source tape with the Dolby B NR system. Since it’s hard to predict how the NR profile will affect each recording, it was typical to make adjustments after a few test recordings.
These days, cassette duplication services will accept digital files (.wav, .aiff, .mp3, etc.) They should have an engineer on hand to make sure the cassettes that are being made sound as good as possible, but chances are they’ll just transfer your files “flat” using your source files. Ideally, they should make frequency adjustments to the source as needed if the tapes that are being made don’t sound optimal. If we’re talking about the sound of throwback Hip Hop tape releases, consider that the dynamic range of those older albums was bigger than the releases of today; people weren’t smashing levels as much as we do these days, so that’s going to have an impact on the way your tape will sound.
When mastering to cassette, I use the full resolution 24 bit masters to feed the recording deck an unbalanced line out from my mastering console, and drive the input to my cassette deck to allow the cassette format’s saturation characteristics to give the material that “crunch” that you might be familiar with, especially with older cassette releases from the 90’s, for example. The saturation that’s achieved on tape will help give it a sound of cassettes from back in the day and it will sound slightly different than your digital release.
On the processing side, it’s always useful to make test recordings and see what they sound like afterwards, and tweak your processing chain to get the best sound for this format. I usually like the way recordings sound with a little bit of compression focusing on clamping down percussive peaks slightly, and the UAD Fairchild is one that I like often. The UAD Neve 33609 sounds good as well, but it also depends on the material. Brightening the mids and highs is also something I’ll do, and for that my go-to is usually the UAD Pultec Pro. This is just a starting point for me, so if this doesn’t sound right I will try different bus compressors and EQs, then make a few recordings on tape and settle on whatever sounds best.
Overloading a cassette deck’s circuitry isn’t the same with all available cassette recorders out there. Higher end Hi-Fi and professional decks equipped with Dolby HX Pro are able to record hotter levels (about 6dB) on tape without added distortion, this also means we can saturate more tastefully. I have a restored Tascam 122 mkII recorder, which was a typical workhorse mixdown deck in many Mastering studios back in the days when record labels were interested in putting out the best possible sounding cassette releases. Many of the tapes I have to this day have a Dolby HX Pro logo on them, to suggest that the cassette master was mastered on a deck equipped with it and many were also encoded with Dolby B (although as tapes age, I find they sound better with Dolby B disengaged, even though they might have been encoded with it).
Dolby HX Pro was considered to be a major update to the compact cassette format when it started to be used in the early 80’s. Playback decks don’t have to have HX Pro built in to be able to play tapes that were recorded using this technology; it’s a process that happens during recording. Essentially, cassette decks equipped with HX Pro are able to produce louder cassette recordings with less noise than those that aren’t. Some high end consumer recorders like the Nakamichi Dragon, considered by many to be the best consumer cassette recorder ever made, didn’t use HX Pro because the quality of the recording head was so good that it could achieve similar recording levels with minimal noise and distortion. However, unlike professional-grade decks like those made by Tascam, bias selection isn’t automatic on the Dragon and it must be set manually for each tape type; cassette decks that automatically adjust bias for each type of tape do so by identifying a series of indentations on each cassette tape that is loaded. Scarcity of parts for servicing and cost of repairs (if you can find someone reliable that can do so) these days also make the Dragon not ideal for professional use.
Taming the Hissing Beast
Dolby NR (Noise Reduction) is an often misunderstood subject by many new to the format. Most consumer decks and portable players come with Dolby NR B. Many high-end consumer and professional decks often came with both B and C. For the sake of simplification, B reduces hiss during recording a bit less hiss than C, which extends the noise reduction frequency down to about 100 Hz. Both were part of an encoding (recording) and decoding (playback). If you record your tapes using B, the playback deck should also be set to B (and the same goes when using C). Commercial cassette tapes used the B profile, while C was aimed towards home recording gear. Fostex used the C system in many of its multitrack cassette and reel-to-reel recorders, so it was useful to have a stereo mixdown deck that was able to encode and decode both noise reduction systems.
Dolby B was developed in the late 60’s to help minimize tape noise. Dolby C was developed in 1980, and HX Pro came soon afterwards. By then, tape formulas had advanced quite a bit. As I mentioned earlier, not all cassettes sound the same and this is because there are different types, which are made with different materials that act as a magnetic element.
Before we go on to the different types of tapes, something that should be mentioned is bias. Bias is an inaudible, high frequency signal that is applied during recording. This signal is mixed in with the audio signal that is being recorded and moves it to the linear portion of the tape, so that the audio signal is recorded faithfully. The bias signal changes amplitude depending on the type of tape being used (lower bias for Type I, higher bias for Type II and even higher for Type IV tapes). Cassette decks either set bias automatically by reading indentations of the cassette shells themselves, or they allowed users to set the bias curve themselves, on these types of decks, there are controls usually labeled “normal” (for type I) “chrome” (for type II) and “high/metal” (for type IV).
Type I: This was the first type of cassette tape that was manufactured. The magnetic element in this type of cassette is gamma ferric oxide (commonly known as “ferric tape”). These kinds of tapes are usually labeled “normal bias” and tend to be noisier (more hiss) than Type II cassettes, but a lot of cassette tape enthusiasts prefer the sound of a well-made Type I tape for recording, like the Sony EF series, because it tends to warm up low frequencies in a way that Type II tapes don’t, and are able to record at slightly higher levels without saturation. High frequencies aren’t as bright as they can be on Type II tapes, which may be a desired effect depending on the type of music being recorded. When using one of the better Type I cassettes, it might be useful to use Dolby B (or C, which may produce slightly warmer recordings, but keep in mind what I said earlier about both NR profiles and their availability on consumer decks).
Type II: Developed not too long after Type I cassettes, this formula uses chromium dioxide and is commonly referred to as “chrome” tape. Type II tape is able to reproduce brighter high frequencies with less hiss, but it also reduces the response of low frequencies slightly. When using a high quality Type II tape, you may find that you’ll end up with better recordings when you don’t encode your recordings with a Dolby NR profile, and perhaps bump up the low end and the mids a little bit on your source recordings before hitting the tape.
Type III: This formula, known as “ferro-chrome” combined both “ferric” and “chrome” formulas on the same tape in hopes to get the best of both worlds: the better bass response of Type I and the better high frequency/reduced noise of Type II. The Type III had a short life span, from about the mid 1970’s to 1980. One of the main problems with it was bias; should you set your deck to normal (Type I) or chrome (Type II) bias? Those decks which set bias automatically would default to normal, and after a couple of years of consumers testing out this type of cassette (and manufacturers of cassette decks watching closely), they discovered some flaws, like the chrome layer of the tapes coming off with heavy use. They also discovered that when it came down to sound quality, the Type III didn’t offer an obvious improvement over the Type II cassette for those users that had decks that were able to adjust bias manually; many users felt that Type II, with its higher bias setting performed better. Manufacturers were reluctant to incorporate a middle ground bias setting for Type III in their tape decks because of the flaws being reported by consumers. They might have, if consumers would have bought into this particular type of cassette, but it was never popular and it struggled making worthy sales throughout its short life.
Type IV: Towards the end of the 1970’s, a completely different formulation of tape hit the market. This one used metal particles instead of oxides and consumers immediately saw a benefit from it. Known as “metal” tape, the Type IV was able to record even louder signals with less distortion in the upper frequencies than the Type II and the low frequencies also sounded better. This increase in quality did not come without some negative effects. Head wear was increased as the metal particles are more abrasive than oxides, and it was a bit more difficult to erase previously recorded material from it. The cost of these tapes was often more than double the cost of an average Type II cassette but it was worth it for a lot of users who heard the improvement in quality over the previous types of cassettes. It wasn’t long before manufacturers started including a metal bias selection in their decks, which happens to be an even higher bias signal than that which is used for the Type II cassette. It was definitely the best of all the types when it comes down to sonic fidelity.
After reading all of this, don’t you feel like giving your DAW a nice big hug? Isn’t it nice these days to just throw a good chunk of cash into a box with excellent Analog-to-Digital converters? Writing this article took me back to a time where you had to put in a lot of time and effort into getting decent recordings on cassette tape. I also remember lots of frustrating times with the format, like tapes stretching, dropouts and tapes being chewed up in the transport. I also remember what cassettes sound like when you play them loud through a nice system; the ones that were done right sounded excellent. With that, I can say that I see why this format is becoming increasingly appealing to artists, especially those that want a lo-fi feel from their recordings and are looking for that familiar vintage sound of the format.
Sometimes limitations inspire creativity, and the compact cassette tape format definitely had a lot of them.
Reverb is a classic mixing tool for adding width, but also that third dimension to your mix: depth.
By adding depth to your stereo image, you’re also expanding the stereo image as a whole. Reverb will give you more room for every sound to breathe and settle into the mix.
There are many different ways to use reverb and add space to your mix, but any reverb technique will add some degree of depth and spaciousness to your mix. And there are many types of reverb. Each is capable of adding a distinct vibe and depth to your mix.
Choosing the perfect type of reverb to give that extra space without drastically changing your audio’s character will take some practice. But when it comes to width, Hall reverb is a good place to start.
Don’t stop there though… all types of reverb can do wonders for adding three-dimensionality depending on your mix and production style. It can be useful to experiment with different reverbs for different tracks in the mix, or alternate dry tracks with reverb treated tracks. With small amount of effect, that can add unpredictable and variable spaciousness during the final mix.
Hot Tip: Using reverb with a short decay time will add a subtler reverb effect. It’s great for when you want to add width and depth without changing the overall character of a sound.
Some examples where reverbs are used with a creative and functional approach:
Microshifting is a clever technique for creating juicy stereo images that allow your channels to sound larger than life and extra wide.
Here’s how it’s done:
Take one stereo track, pan it center and keep it there. Next, duplicate that track twice (so you now have three versions) and patch a pitch shifting plugin inline on both copies.
Now, use the pitch shifter to pitch one copy down a few cents (5-10 cents is common) and pitch the other copy up the same amount of cents. Next, pan one copy hard left and the other hard right. That’s microshifting.
Listen to the three tracks back in stereo and revel in your clever trick and newly widened stereo image!
M.B.: What do you see as the differences between analog and digital creations of dark ambient music?
S.: There’s no difference from a creative point of view. Obviously there are a lot from the technical one.
M.B.: What are some of the key instruments/programs that you use to make analog dark ambient?
S.: Analog synthesizers, electrified string instruments, guitars, samples, editing software and plug ins, percussions, wind instruments parts (commissioned to other musicians), tapes, pedals, dronin.
M.B.: What are some of the key instruments/programs that you use to make digital dark ambient?
S.: Mainly plugins to work on noise parts and editing softwares.
M.B.: Do you see one or the other as being the “better” technique for creation of dark ambient music?
S.: Everyone has to develop the better process fitting with her/him attitude.
Drones? M.B.: What are some of the techniques you use to create drones?
S.: There are different ones, maybe infinite. It’s possible to make drones with stratifications of synth pads, by editing acoustic instruments like brasses, winds, string instruments and so on; playing heavily distorted bass and/or guitar; editing samples, using noise from modular synthesizers, editing field recording, recording the washing machine noise and on and on..
M.B.: Do you have a favorite program/instrument to use for creating drones?
S.: Not a specific one.
M.B.: As a beginner did you create drones the same way you do now?
S.: As a beginner I made a lot of mistakes before to find my way.
M.B.: Have you changed techniques/software/instruments for creating drones over the progress of your career?
S.: Yes I did it many times. And I continue to change to make the sound fabric different in any production I do.
M.B.: How important are drones to dark ambient music?
Probably drones are the dark ambient trade mark, as well the violin and piano are in the classical music for orchestra, or the electric guitar solos are in the rock music.
M.B.: How important are field recordings to dark ambient music?
S.: They are another fundamental component in dark ambient music. They are the ingredient to create visual atmospheres, vivid landscapes, even stories, and forge a solid concept when the musician has something interesting to tell through the music.
M.B.: What electronics do you use to capture field recordings?
S.: I’m not a professional of field recordings, so I use simply an IPhone when I’m around to catch everything could be interesting.
M.B.: Do you leave the field recordings raw or do you add effects treatment to them?
S.: I usually treat field recordings with additional reverbs. But the most important thing is to find the right level for the field recording layer in the mix. Mixing is by all means a crucial part in the process.
M.B.: Do you use field recordings in the creation of drone or do you only use them as a secondary layer of sound?
S.: It’s a possible choice to use f.r. for drones, why not?
M.B.: Do you use human vocals in dark ambient?
S.: Yes human vocals.
M.B.: How important are human vocals to dark ambient?
S.: It depends of the concept behind the work, but I find human vocals important in my music, especially the spoken words.
M.B.: Do you create your own passages to recite?
S.: Just sometimes.
M.B.: Do you use your own voice, hire a voice actor, or use samples from films/television/speeches?
S.: Yes, samples from old documentaries, movies, speeches are my favorite. But I also asked singers to send to me parts for specific uses.
M.B.: Is it necessary to ask permission of the original copyright holder before using samples of vocals in your music?
S.: No, for they are usually very short samples or free samples.
I prefer to escape all questions about DAW, computer and so on, simply because there are not peculiarities for dark ambient music. The logic of hardware and software is the same for all kind of music. Just I can add that I’m a graduated sound technician, so I learned technique of recording, mixing and mastering through regular courses. But as in all studies, the experience is the most important factor. Do it, do it and do it again. And after some years everyone will find the right set up and process. And for people like me, who don’t have big amounts of money to invest in expensive hardware and software, the experience will help to do more, using less. And this is a big advantage for creativity; when you have poor instruments and have to use your brain to find out something good. Take a few small stones, beat each other and record the sound by using some freeware delay and reverb. Probably you will be very positively surprised of the result.
M.B.: Where do you go to find samples?
S.: Everywhere: everyday life, music, movies, documentaries, vinyl, VHS, NASA web site, specialized platforms for samples sharing…
M.B.: What samples would be off-limits in a legal sense?
S.: There are a lot of free samples around, or simply usable by asking the owner permission. But for more specific knowledge of the argument I suggest to read the related laws of the source origin country.
M.B.: How do you extract samples from movies, games, speechs?
S.: Through Youtube when it’s possible by a software, but for more original sources by connecting the source (turntable, VHS player, microphones,…) to the audio interface.
M.B.: How important are samples to dark ambient music?
S.: Important, but not necessary.
M.B.: When you use instruments in your music do you play a real instrument yourself?
S.: Yes I do it.M.B.: If you want to have violin, for exampe, (or any other instrument) in a song, but don’t own one and can’t play one, is there another option? (some sort of program that will create violin sounds for you?)
S.: I prefer to directly ask other musicians to realize the part, so to have a more natural and warm sound effect.
MasteringM.B.: How important is mastering in dark ambient?
S.: It’s fundamental.
M.B.: Can a musician master their own album with limited training?
S.: It’s not an easy job without a little of training.
M.B.:What programs do you use to master an album?
S.: I prefer to not tell that. It risks to be a commercial advertising for software companies
M.B.:If paying another person to master an album, what credentials should they have? (ie. do they need to make dark ambient themselves to understand how to master dark ambient?)
S.: It would be better if the mastering service comes from a person with a good sensibility for that kind of music. If a musician who plays himself that music, that’s even better.
M.B.: What are the differences between mastering an album that is digital, CD, cassette or vinyl? Should each have a separate mastering?
S.: There’s a certain difference about mastering a vinyl compared with cd or cassette mastering. It’s related to the output levels that differ, in the vinyl case, depending of the track position (closer to the edge or the center). So as matter of fact, they are two completely different mastering. But for this I suggest to read this article from Sonologyst blog: https://wordpress.com/post/sonologyst.com/168
M.B.: What advice would you give to a person just coming into dark ambient as a potential artist?
S.: Just to work with passion and not to be hurry, releasing huge amount of music, just to show the audience what is going on. That is a mistake many people do, while the process to improve the own style should be something private. M.B.: What are the best aspects of creating dark ambient? S.: It gives to you the possibility to be in deep connection with you profound states of mind. M.B.: What are the worst/hardest aspects of creating dark ambient?
S.: There aren’t worst aspect to me. M.B.: What are somethings an amatuer should avoid doing at all costs?
S.: I replied to this question in the previous point. M.B.: How frequently should an artist aim for releasing albums (several times a year?, once a year?, once a month?)
S.: Every artist has to find the own way for that. It’s impossible to give a general advice. In my case I found the good and natural rhythm working on one release a year. And I don’t exclude to increase the interval between two works. That lets me a major deepness, awareness and consciousness of what I’m going to do. Basically I start a work when I have really something to communicate, and after I’m aware of that, I need time to explore how to communicate it.
M.B.: Should a musician know the history of the genre before creating their own music?
S.: Not necessarily, but it would be a crime to ignore all that beautiful music created in latest decades.
Reverb is necessary in order to create the impression of distance and separation between elements, but it also contributes a lot to the ‘glamour factor’ you’ll need for a modern commercial production. Quite simply, making the wrong reverb choices is a strong indicator of a non-professional mix. It is potentially so destructive that many of us are either too conservative when we use it – resulting in no real benefit – or else it’s applied too liberally and smears over all your previous delicate mixing manoeuvres.
What follows then are 10 essential tips to help you steer clear of the pitfalls and build a more effective reverb workflow:
1. Long and Short Reverbs
A good general piece of advice would be to use short reverbs in busy mixes, longer reverbs in music with more space. What can be deceiving though is to judge the validity of reverbs by name i.e Halls and Chambers as long, plates and rooms as short – if you think you need short reverbs you could find exactly what you want from a short Hall and you could find just the tail your after in a mix using a long plate. Rather than thinking purely in terms of long and short, think in terms of the quality of the tails, longer tails can disguise the presence of reverb where short ones can draw attention to it.
2. Pre Delay
Pre-delay is the single most powerful feature in most reverbs, setting a pre-delay allows for a certain amount of dry signal to get through before it is washed in reverb, this means greater intelligibility. It is particularly useful for keeping the attack of words from a lead vocal upfront and clear. In this sense it is much like how you use a compressor, setting a slower attack time or in the case of reverb – pre delay, lets vocal information through which is re-assuring to listen to. Anything from 20ms to about 80ms will be the area you need to work in – beyond this you will create a distinctive slap back effect that could be cool in the right circumstances but less suitable for most.
3. 3D Reverb
For a far more dimensional reverb and richer spaces you should try using multiple reverbs together. For example, using three you can create a much more convincing ambience.
To get you started – first find a small room reverb, this to give a little air around the source, second use a plate reverb and blend it as you might pour sauce into pasta, adding flavour. Third use a hall with a long tail to add ceiling, be careful not to overdo it. How do you know when you’re overdoing it? Read on…
4. How much?
It is a taste thing of course, if the reverb is a big feature of your production ala Phil Spector, you’ll use a lot more than if you were just using it to blend, glue and create a believable ambiance. Remember that a little reverb goes a long way. When it comes to applying reverb, solo the instrument or voice, then bring in the reverb until you can hear it, then back it off a smidge until you sort of feel you want a bit more. That’ll probably be about right.
5. High Pass Filtering
Hopefully you are aware of the benefits of high and low pass filters, and you’ll have been applying these to instruments in your mix already to keep low end rumble and other toxic frequencies at bay. The same wisdom works on reverb, high passing reverb with an EQ – i.e. rolling off its low end will keep the space open. Leaving the low end in, could mean you lose definition as you add more reverb to more channels.
6. Brightening with Reverb
Reverb can be especially useful for brightening vocals where you might feel that EQing the vocal directly is working against you. This same approach can of course be used on any instrument, the trick is to identify the presence EQ range and then boost that in the reverb. For example, vocals usually have a strong presence around 3K, so, rather than EQ the vocal audio, instead, insert an EQ after your reverb and push that in the 3K area to get a more airy, transparent lift.
7. Impact with Reverb
Snare drums can regularly benefit from gated reverb, whereby a half second or so pre-delay followed then by a reverb that is gated, i.e. cut short abruptly – creates an un-natural, but useful artefact that might be described as smashed glass. The effect if used proportionally and blended well will give dimension to the snare without adding body. Used heavily it is a very distinctive effect brought to popular consciousness by Phil Collins and David Bowie, however, despite how dated it can make a drum sound, it is still used a lot for sound re-enforcement even in today’s most cutting edge productions.
8. Grouping Reverb
A useful tip for gauging the effectiveness of your reverb, especially if you are using multiple reverbs is to group them together so you can then solo or mute them with a single mouse click. Being able to A/B processing in this way can be very informative and help you reign in your levels or feel confident about adding more. Having control of all your reverbs on a single fader will allow you to fine tune how you want them, plus you can of course apply EQ as mentioned in tips 5 & 6.
9. Springs and other ‘Dirty’ Verbs
Certain instruments take better to reverb than others, some don’t play so nicely. Often we like electric guitars to feel upfront, but regular reverb tends to soften their impact. Spring reverb and certain other lo-fi processors work especially well, the harshness of them can add body and presence to the audio, don’t discount cheap sounding processors and springs for use on vocals either, for a vintage lo-fi vocal reverb springs are extremely fashionable right now.
10. Mono Reverb
We often think of reverb being stereo but there is huge benefit to setting up mono reverbs. Mono reverbs are great for spot lighting or where you want to draw attention to an instrument without swamping the mix. If you wanted to spot light a keyboard solo that was panned off to the right, setting up a mono dedicated processor and then the pan to match the pan setting of the keyboard, will really help retain the dynamics in your production.
Reverb fashions come and go and presently we seem to be in a phase of rediscovering reverb. Reverb today is generally very musical and subtle: We want it, but we don’t want it to be over bearing. Modern vocal tracks and spot FX will tolerate a good deal, but we like drums to feel natural, textural and upfront. The main thing to keep in mind is that successful reverb should enhance the mood you are aiming for; it is not just about adding size, but embellishment of your central theme.
Basic considerations on compression and parallel compression.
The first thing to establish is what a compressor of any form actually does, and the answer is that it reduces the dynamic range of the input signal. Whether it’s configured to make the loud bits quieter, or the quiet bits louder, fundamentally it exists to reduce the overall dynamic range from something large and unmanageable to something smaller and more appropriate for the intended application.
The vast majority of compressors apply ‘downward compression’ which means, in essence, that loud stuff is made quieter. More specifically, signals below the threshold level are left alone, while those above are ‘squashed’ by an amount determined by the ratio setting.
The more conventional way of illustrating compression is with a ‘transfer plot’, which is a graph with the input level on the horizontal axis and the output level on the vertical axis. The graph in Figure 2 was obtained by measuring the amplitude response of a ‘hard-knee’ compressor plug-in in a DAW, using an Audio Precision test system. The dotted straight red line at 45 degrees shows the response with the compressor bypassed — clearly illustrating that what goes in comes out, unchanged in level!
The different coloured solid lines were obtained with the compressor switched in and the threshold set to -20dBFS. The light-blue line is the result of a 2:1 ratio, and it clearly shows that when the input is 10dB above the threshold (ie. a ‘Generator Level’ of -10dBFS on the horizontal axis), the output or ‘Measured Level’ on the vertical axis is at -15dBFS, which is 5dB above the threshold. In other words a rise of 10dB at the input results in a rise of only 5dB at the output — which is half as much, and hence a compression ratio of 2:1.
The other traces show progressively ‘stiffer’ ratios, of 3:1, 5:1, 10:1, and 40:1. Anything above 20:1 is generally referred to as ‘limiting’ because the output level barely rises regardless of how much the input level exceeds the threshold.
Incidentally, a ‘hard-knee’ compressor, like the one used for these measurements, switches abruptly from doing nothing to squashing the audio, and that is revealed by the very distinct change of angle on the transfer curves. A ‘soft-knee’ compressor moves more gently from inaction to action, so its transfer plots would curve smoothly away from the 45 degree linear slope instead of diverting abruptly.
Now let’s imagine a musical signal where the quietest element measures -35dBFS and the loudest is -5dBFS, so that we have a starting dynamic range of 30dB. If we were to pass that signal through a 2:1 compressor with a threshold at -20dBFS, the output signal will range between -35dBFS (this level is below the threshold and thus unchanged) and -12.5dBFS. The latter figure arises because the source peak level (at -5dBFS) is 15dB above the threshold, and thus will be reduced by half to 7.5dB above the -20dBFS threshold, which is -12.5dBFS).
Therefore the dynamic range has been reduced, in this case from 30dB to 22.5dB, and at the same time the peak level has been reduced in level by 7.5dB. To help illustrate those points, I’ve added coloured bars to the previous plot to illustrate how the dynamic range and peak levels have been reduced.
Useful though this form of compression is, often we want to reduce the dynamic range without reducing the peak level. In other words, we want to raise the level of quieter signal components rather than turn down the loud ones. The usual way to achieve this is to introduce ‘make-up gain’ at the output of the compressor. A good way of understanding the concept is to return to the ladder diagram.
I prefer to call this ‘uplift compression’ to avoid confusion with true ‘upwards compression’ (which I’ll come back to in a moment). As the lowest ladder diagram in Figure 1 shows, the input signal is first downwardly compressed in the usual way, but then the output is raised in level by a fixed amount of ‘make-up gain’. The overall result is that the dynamic range has been reduced, again, but this time the peak level is restored to the same as the input while lower-level signals have been raised — so the quiet bits have been made louder.
However, this diagram reveals clearly that it is still the louder elements that have been ‘squashed’ — the quieter signals have simply been raised in level. That’s the key difference between ‘uplift compression’ and ‘upwards compression’. By the latter term, I mean processing that squashes the quieter elements while leaving the loud bits alone, as the final diagram in Figure 1 illustrates.
Normal downward compression — whether it’s used on its own or with make-up gain — inherently changes the character of loud signals to some extent by squashing them. Rule number one for any downward compressor is to squash anything loud! However, the action of turning the level down (and back up again afterwards) isn’t instantaneous; it takes place over a timescale that is governed by the compressor’s attack and release time constants. The inevitable result is that the sound and shape of complex but delicate and loud transient signals can be altered quite drastically. This is a significant part of the reason why different compressor designs can sound so different from each other, and why one compressor may be preferred in a given situation over another.
Using compression effectively is fairly easy once you get your head around the principles of what it does to your signals, and it’s the simplest way to give your sounds some of that elusive pro punch.Parallel compression is one technique that can help here. It sounds complicated but it’s not – you simply duplicate your drum track (or any other type of track), and then heavily compress the duplicate, leaving the original uncompressed. When you play them back together, you get the powerful ‘breathing’ dynamic sound of the compressed version, whilst still retaining the detail, brightness and clarity of the uncompressed version. The best of both worlds…
Obviously a vinyl record is a different thing from a CD or a WAV file, but does it require a separate, dedicated master, or are the two formats basically made from the same mastered file?
The answer is YES! Making a mix post production for vinyl means to make an alternate version after the digital one, with little to no digital peak limiting, and a little more headroom in the analog domain. Sending a loud and aggressive CD master to a lathe will only cause the cutting engineer to have to turn things down significantly, and in many cases they’ll be forced to cut an even quieter record than they would have with a more dynamic premaster. Your mastering for vinyl doesn’t need to be as loud as your CD master because the volume of your vinyl will be determined by the length of the sides, which means to keep louder more aggressive material near the outer edge of the record (early in the side sequence), and the more subdued and less aggressive tracks near the inner grooves (where noise and distortion become more of a consideration).
A digital master for CD has to have a 16-bit word length, and it can be as loud and as limited as the client’s taste or insecurity dictates; with the vinyl master there is a physical limit to what can be fed to the cutting head of the lathe, and so heavily clipped masters are not welcome and can only be accommodated, if at all, by serious level reduction. For vinyl, the optimum source is 24-bit. Things can get more tricky if the primary focus is the digital master, and especially when that is required to be fairly loud. You can’t simply take an unlimited file and add 4 or 6 dB of limiting without sonic consequences, and so for loud CD masters, we normally add another step of gain-staging and include some light limiting during the initial processing run, the result being a louder master to begin with for the second stage of adding gain. In this case the difference between CD/digital mastering and the vinyl one will be more relevant.
So before to go on with an alternate version for vinyl mastering, think about what’s the priority in your music distribution strategy and then talk about that with the sound technician who will care about your project.