I don’t know why DAWs equate humanizing with randomness. Sure, humans don’t have the same metronomic precision as quantized MIDI data, but avoiding that metronomic precision is only a small part of what makes playing “humanized.” Really good musicians have great control over timing, and use that talent to move timing around the beat, either consciously or subconsciously, which indeed adds a more human quality. (On the other hand, if the goal is to emulate musicians who've had too much to drink, then randomization does a superb job!)
You can alter note timings manually (e.g., draw a marquee around the notes you want to move, then move them) or use a “slide” editing function; note that snap needs to be turned off, and these changes should be subtle. For example:
Transient shapers have tended to fly under the radar a bit, but as they become more common, there's increasing interest in using them to enhance mixes. A transient shaper is a specialized dynamics processor that affects a signal’s attack, but unlike a compressor or limiter, doesn’t necessarily change the overall signal level. It can either emphasize or soften the initial transient; there’s typically a single control where the center position does nothing, clockwise sharpens the attack by amplifying it, and counter-clockwise softens the attack by ramping up to the full level over time. Some transient shapers also include a sustain control that brings up the average level after the initial decay, which turns the transient shaper into something more like a compressor that separates the initial attack and sustain elements
There are two main cautions when using transient shapers.
Transient shaping has many uses for mixing.
Transient shapers can help emphasize or de-emphasize sounds in the mix without the traditional level and dynamics controls. If you haven't found out how helpful transient shaping can be, give it a shot.
There are two kinds of people in this world: Those who like to divide people into two kinds, and those who don’t. Okay, I’m just kidding, but there are some people who gravitate toward sampled sounds and some who gravitate toward analog-type waveforms. But why not enjoy the best of both worlds?
Why Sine and Triangle Waves Deserve RespectSine and triangle waves may seem like the most boring waveforms in the world, but they actually have many uses. For a fuller acoustic guitar or piano sound with a more authoritative low end, layer a sine wave along with the lower notes. To attenuate the sine wave at higher notes, modulate the wave’s amplitude negatively according to keyboard note position (i.e., the higher you play on the keyboard, the lower the level). Also keep the overall level low—just enough to provide a subtle psycho-acoustic boost.
But that’s not all; sine and triangle waves can add more depth to almost any sample because digitally-generated waveforms can have more presence than digitally-recorded sounds. For example, harp samples may lack a bit of “you are there” presence due to mic limitations, room acoustics, etc. Layer a triangle wave with the harp (adjust the triangle’s amplitude envelope so that it mimics the harp’s natural envelope); the triangle wave provides presence, while the sample provides detail and realism. Initially set the triangle wave level to 0, and then bring it up slowly to taste. Keep it subtle—we’re talking background reinforcement, not something obvious.
Here’s another triangle trick: To add some male voices to an ethereal female choir, layer a triangle wave tuned an octave lower. This gives a powerful bottom end that sounds like guys singing along. To maintain the ethereal quality in the upper registers, consider modulating the triangle wave amplitude according to keyboard position so that the triangle wave isn’t apparent on higher notes.
Better Strings Through Layering
String synthesizers of the 70s, based on sawtooth or pulse waves, created rich, syrupy string sounds that weren’t super-lifelike, but nonetheless sounded pretty cool. Sampled strings may sound more realistic, but often lack the smoothness of analog simulations. For the best of both worlds, dial up a sawtooth or pulse wave, and adjust its envelope for as realistic a string sound as possible. Now layer it behind a string section sample, and the synthesized waveform will “fill in the cracks” in the digital waveform.
Pitched Percussive Transients
To close out, let’s do the reverse—layer a sample with typical synthesizer waveforms. Percussion instruments, when played across a keyboard, acquire a sense of pitch. Adding a short amplitude decay, and layering these with interesting waveforms, can yield hybrid sounds that are synth-like but have complex and interesting transients. Cowbell is one candidate for this application. Claves, triangle dropped down an octave, struck metal, and just about any other pitchable percussion can also give good results.
171027 More Expressive Electronic Drums
So let’s make your life easier—a much better way to introduce that pause is to drop the tempo waaaaaaaay down for just a fraction of a second in a beat (different note values can work too), which will create a pause before the music returns to the beat again.
It’s of course best if the section over the pause sustains—something like a pad, held note, and the like. Delay with feedback works well, and with MIDI, if you choose a math-friendly tempo change (e.g., drop it to half-speed), you can compensate mentally and add MIDI notes where they make musical sense.
181102 More Touch Sensitivity for Compression and Distortion
180907 Hardware not Compatible with Windows 10? Maybe It Is After All...
171110 Tempo Change "Time Traps" for Extra Drama
180413 Instant Open-Back Amp to Closed-Back Amp Conversion in the Studio
To mic a cab, you just point a mic in its general direction, right? Yes...and to get to the moon, just jump real high. Actually, there’s much more to miking than meets the ear, as you’ll find out from these tips.
1 Choose the right mic. For dynamic mics, the inexpensive Shure SM57 is the classic guitar cabinet mic—many engineers choose it even when cost is no object. The Electrovoice RE20 and Sennheiser MD421 are more upscale. Condenser mics, though often too sensitive for close miking loud amps, make good “secondary” mics—placing one further back from the amp adds definition to the primary dynamic mic. AKG’s C414B-ULS is a great, but pricey, choice; their C214 gives similar performance for much less. Neumann’s U87 is beyond most budgets, but the more affordable Audio-Technica AT 4051 has a similar character (it’s great for vocals, too). And don’t forget ribbon mics: they have a “warm” personality, and a polar pattern that picks up sounds from the front and back—but not the sides. In multi-cab guitar setups, ribbon mics let you do cool tricks by choosing which sounds to accept and which to reject based on mic placement. Among newer mics, Royer’s R-121 and R-101 are popular for miking cabs.
181130 Dealing with VST Scan Problems
180810 Now There's a Standard for MIDI Over TRS Connectors!
171208 Beyond Level Changes with Mixing
180824 How to Fix Windows Groove Music Issues
Yes, there is a way to get sustain with amp sims! Use a Y-cable (or other splitter) to send one guitar split into your audio interface’s input, and the other into a small practice amp set to a high-gain setting (using a compressor between the split and amp can also help promote sustain). When you want to sustain a note, touch the guitar headstock to the amp’s speaker cabinet. The cab’s vibrations transfer into the neck, which vibrates the strings and causes them to sustain. Because the strings are sustaining, when you record the dry guitar sound into the audio interface, the sustain gets recorded into your DAW.
For resistor values, try 4.7k, 10k, 22k, 47k, and 100k. Good luck finding a rotary switch locally...but you can get them from parts-express.com, or if you’re in a nostalgic mood, radioshack.com still exists and has a 2-pole, 6-position switch (part #2750034).
Even if you’re a fan of hi-fi sound, you might be pleasantly surprised at how much a little loading can smooooth out your sound when feeding distortion. Try it sometime - for the cost of butchering a cable and a 10 cent resistor, duller sound can be yours!
Sometimes doing just a little bit of a pause before a section of a song comes crashing in can add a major element of drama...the listener expects the section to start on the beat, but even a tiny pause can add significant tension before the release.
Most programs let you insert measures to “open up” the project via ripple editing, but you can also insert something a lot shorter to add the needed dramatic pause. However, there’s a major downside because this will throw off your timeline timings.
180112 How to Build Your Ideal Amp Sim Cabinet
180727 One Tone Control, Two Tone Options
Maybe you’ve listened to the bright, glassy sound of out-of-phase pickup sounds and wish you could get that sound, too. Yeah, I know...rewiring guitars can be a hassle, and besides, you don’t want to void your warranty or nuke the resale value. But if your guitar doesn’t have an out-of-phase switch, or you’re a keyboard player and want to get this sound out of a sampled guitar, you can come close with a studio-type EQ that offers a high and low shelving response along with a parametric stage.
1. Select both pickups on the guitar.
2. For the EQ, dial in a notch filter around 1,200Hz with a fairly broad Q (0.6 or so) and severe cut—around -15 to -18dB.
3. Use a high shelf to boost about 8dB starting at 2kHz, and a low shelf to cut by -18dB starting at 140Hz. If you have a choice for the rolloff slope, 12 dB/octave seems like a good choice .
4. Tweak as needed for your particular guitar and pickups.
5. Boost the level—like a real out-of-phase switch, this thins out the sound.
The screen shot shows these settings translated to Studio One’s ProEQ and Cakewalk’s Sonitus EQ, but the principle is the same for other EQs in other DAWs. And you don’t even need a soldering iron!
180803 Attention, Komplete Kontrol Users on Windows
When mixing, the usual way to make an instrument stand out is to raise its level. But there are other ways to make an instrument leap out at you, or settle demurely into the background, that don’t involve level in the usual sense. These options give you additional control over a mix that can be very helpful.
CHANGING START TIMES CHANGES PERCEIVED LOUDNESS
The ear is most interested in the first few hundred milliseconds of a sound, then moves on to the next sound. This may have roots that go way back into our history, when it was important to know if a new sound was leaves rustling in the wind—or a saber-tooth tiger looking for fast food.
What happens during those first few hundred milliseconds affects the perception of how “loud” that signal is. Given two sounds that play at almost the same time, the one that started first will appear to be more prominent. For example, suppose kick drum and bass hit at the same time. Move the bass a tiny bit ahead of the kick for a more melodic feel, and behind the kick to emphasize the rhythm.
When your computer isn’t working properly, it’s pretty annoying. Fortunately, many times keeping a cool head will get you out of trouble and back to work again in minutes instead of hours—particularly if you panic and go down the wrong path, which could even make matters worse.
One of my less favorite Windows “features” is when it won’t let you write to a hard drive because it says the drive is write-protected, or gives some other error message like not letting you drag a file from the desktop to the disk. This happens mostly with USB drives, and I suspect is has to do with not ejecting it properly (or the computer thinking you didn’t eject it properly).
I can’t tell you how many hours I rummaged through the drive’s “Security” tab trying to figure out what permissions weren’t being granted before finding out that there’s an easy fix. Note that to do this, you need to have administrator privileges. (Note: Although this tip assumes Windows 10, it works with previous versions.)
A lot of people ask me whether dithering really makes a difference, because it happens at such a low level it's really hard to do comparisons of dithered and non-dithered sounds. Whether you'll perceive a difference is hard to say - a lot of it depends on the program material and the listening environment - but there's an easy way to hear what effects dithering has on low-level audio.
180504 Windows DAW Playback via Bluetooth
180223 5 "Wrong" FX Orders
There are a lot of modifications you can do to a stock guitar to greatly alter the sound, such as using tapped pickups, rewiring the tone control, changing pickup phase, etc. The following schematic shows a simple way to obtain two different tone options from a single tone control.
Hook up two different capacitors (the values shown are suggestions; adjust to taste). Turning the control clockwise connects to one capacitor, counterclockwise to the other capacitor, and centering the control effectively takes both capacitors out of the circuit. The tone control can range from 250k to 1 Meg. The higher the value, the more you can be assured that the center position does indeed take both capacitors out of the signal path.
171215 How to Coddle Your Computer
If you use Cakewalk SONAR, the Midtown guitar patches for Rapture Pro/Rapture Session in the 2017.08 update include artificial feedback, so just follow your Rapture variant with the CA-X “Hard Rock” amp (assuming you don’t run into TH3/TH2 related issues...that’s another topic), or use something like a Marshall emulation from the Line 6 Helix, Native Instruments Guitar Rig, IK AmpliTube, etc. But also remember that you need to think like a guitar player—the pitch wheel becomes your virtual whammy bar, and use the pitch wheel for vibrato—not the mechanical mod wheel vibrato, because that’s not how guitars roll.
Happy feedback...and don’t forget the ‘60s light show!
I really like the Native Instruments’ Komplete Kontrol keyboards, so I was disappointed that when I inserted the Komplete Kontrol plug-in in Studio One, it would hang and had to be closed via Task Manager. To make matters worse, although the plug-in would load in Cakewalk by BandLab and the keyboard worked as a controller, the keyboard’s displays were basically dead—there was no browsing, scrolling, inserting instruments or effects—nothing.
For Studio One, the solution was to go into Advanced Options, and disable ReWire. Then everything works as expected. With Cakewalk by BandLab, you need to download Microsoft Cumulative Update for Windows 10 1803, KB4340917. Here’s the link to find it; choose the correct version for your operating system version.
However, don’t assume that going to Windows Update will show you need KB4340917. My computer was shown as “up to date,” so I needed to download the update to the update manually. 713 MB later, the fixes were done and Cakewalk by BandLab was restored to working correctly again with Komplete Kontrol...as it had before the 1803 Windows 10 update.
Once these fixes were done, everything worked. But it does seem like my “semi-blog” entry from July 20, “Do We Need to Be IT Professionals to Record Music?” has been answered in the affirmative!
This image shows four identical riffs. The first is played starting on low E, the second an octave above, another octave above that (i.e., starting on the open 1st string), and finally, the octave starting at the 1st string, 12th fret. The lower image shows the same file, but processed with the EQ settings described above. The level of the lowest frequency’s riff is basically unaffected, the second one is boosted somewhat, but the two higher riffs have considerably more level. This makes for a more even, sustained sound with solos played on the higher frets.
Even if they’re not exact half-steps, they go by fast enough to be perceived as non-continuous. For example, with a virtual instrument’s pitch bend set to +/-12 semitones, quantizing the bend to 1/32nd triplets will give exactly 12 steps in a one-beat octave slide up, while a 1/16th note triplet gives 12 steps over a two-beat octave slide. Make sure there’s no smoothing enabled for the pitch bend function.
For precise slides, the following table shows the amount of pitch bend change per semitone. For example if an octave is a pitch bend value of 8191, and you want to start a slide three semitones above the note where you want to “land,” start at a pitch bend value of +2048 and add equally-spaced events at +1366, +683, and just before the final note, 0. This assumes your virtual instrument has a +/-12 semitone pitch bend range, which is what I use for bass to make these kinds of slides possible.
180330 How to Adapt 5-Pin DIN Connectors to USB
180427 Transient Shaper Tips
I don’t really feel secure about MIDI-driven instrument tracks until they’re rendered (although that doesn’t mean I’ll delete the MIDI track). There are several advantages to rendering...
171222 Laptop Travel Survival Tips
180119 Killer EDM Kicks
180928 Kinder, Gentler Compression
I've been spoiled by using acoustic drum loops (by Chris McHugh) for my mostly rockish music. Even drummers think someone's actually playing drums to the music.
So for my latest EDM-oriented project, I needed electronic drums. I'd forgotten just how boring they can sound. What to do? Well, Here are a few tricks that made a huge difference.
The snare is the giveaway that you’re in robo-land because it’s going to get hit a lot. Render the snare audio to its own track, then try the following.
Vibrato plug-in. This is a fast, simple solution that adds just enough difference not to have every hit sound the same. Finding a true vibrato effect is not that easy; most of the time you'll need a chorus or flanger that can be set to one voice, with a mix for only delayed sound.
Volume shifts. This takes longer to edit, but remember that each hit on a snare will play at a slightly different level and at a different place on the drum. Alternating slight volume variations helps considerably in creating realism, even with individual hits—not just rolls.
Minor pitch shifts on different hits. We’re not talking transposition in semitones, but shifts that are more on the order of 20-40 cents.
Very short attack times. In addition to lowering the overall level, adding only a few milliseconds creates an effect that's somewhat like ghost notes.
Layer a sidestick sample with the snare for emphasis. This increases the level, and helps give the snare a more percussive feel.
It seems it's not so important that these changes duplicate what happens with a real snare, but rather, that they just keep the electronic ones from sounding all the same. That seems to satisfy the ear-brain combination.
Back when memory was expensive, cymbal samples...well...let’s just say they weren’t very good. So I got into the habit of playing real cymbals over electronic drum parts, and they added an undeniable air of authenticity. Even if you’re not a drummer, you can probably hit cymbals at the right time.
Pro drummers are able to control their timing extremely accurately to lag and lead the beat in strategic ways. While those who play drums from keyboards or controller pads often have the right intention for timing, they just don’t pull it off with the required accuracy. Try 85% quantization, which should tighten up your drum parts without strangling them. (However, I do make sure the kick is right on the beat—see if that works better for you as well.) I also like even just a little bit of swing, because it’s amazing how even a little bit of swing can make all the difference in the world when you want a drum part that grooves just a little bit better.
But Wait—There’s More!
To humanize hi-hats, check out the tip for Week 141 in the original Friday’s Tip of the Week thread in the Cakewalk SONAR forum. Also check out the tip in Week 145, “No More Boring MIDI Drum Parts.” It’s SONAR-specific, but the concept translates to many other DAWs.
Several people have commented that the keyboard bass parts in my music sound like “real” bass. Although I do play bass, most of the time I prefer keyboard bass because I’ve sampled so many basses it’s always possible to find a suitable bass sound.
Part of the realism is due to playing parts that a bassist would (and could!) play, which means including credible slides. Slides are an important bass technique—not just slides up or down a string, but over a semitone or more when transitioning between notes. For example, when going from A to C, you can extend the A MIDI note and use pitch bend to slide it up to C (remember to add a pitch bend of 0 after the note ends).
Unless you’re emulating a fretless bass, you want a stepped, not continuous, slide to emulate sliding over frets. Quantizing pitch bend slide messages so they’re stepped is one solution. In the screen shot below, the first slide goes in half-steps from tonic to an octave higher. The second one slides from G to E and steps down over three semitones. The pitch bend then returns to 0 before the next E plays.
180126 “Humanize” MIDI Parts with Timing Tweaks
This may seem like a bit of a copout in terms of giving a Tip of the Week, but if you want 31 MIDI tips, I've been posting a new MIDI tip every day during May on The MIDI Association website to celebrate "May Is MIDI Month." A few of them are updated versions of tips that have appeared here, but most are new and cover interfacing, performance tips, DAW techniques, and more. I'll be back next week with a full-blown new tip, but meanwhile, check these out if you use MIDI - I'm sure you'll find at least some that will be very useful to your musical endeavors.
180601 In Search of a Duller Guitar
180323 Optimizing Pickup Height for Sustain
quantization MIDI effect. The data remains as you played it, but you hear it play back quantized. Later on, when you start editing, you can remove the effect and make any desired changes permanent.
Options vary from program to program, although certain effects are common. The screen shot shows the roster for MOTU’s Digital Performer, with the Humanize effect window in the background. One of the more unusual options is DeFlam, which moves the start times of a bunch of out-of-sync notes to the average start time. This can be very handy with older MIDI hardware, where there may be a slight “spread” among notes hit simultaneously.
Ableton Live’s collection of MIDI effects includes one that generates chords from incoming MIDI data; if you’re looking for inspiration, follow the Random MIDI effect with the Scale effect. The Random effects applies weirdness to the data, while constraining to a Scale effect pulls it back into reality again. The late, great Cakewalk Sonar has real-time versions of standard functions too, but also includes a chord analyzer and a MIDI Event Filter as a plug-in, not just a menu items. This lets you filter out various notes, velocities, and other attributes, can create the equivalent of splits and layers from a master keyboard controller, and constrain notes to particular scales.
Cubase has a very comprehensive collection of MIDI effects—I call their Density plug-in the “Mozart” plug-in, as it removes or adds notes (remember in the movie where Mozart was told his composition had too many notes? This plug-in would have been the ticket). There’s also a microtuning plug-in that can alter the tuning of individual notes, and a MIDI control panel that lets you create and vary up to eight continuous controller signals. Studio One added MIDI Note FX in version 3, and Apple added a suite of MIDI FX plug-ins to Logic Pro X...but this isn’t intended to be a comprehensive list, just a wake-up call in case you haven’t taken the time to find out what the MIDI effects in your host sequencer can offer. Personally, I love ’em and wish there was a standard!
Compression and distortion are arguably the two most popular effects for guitar. However, guitars often don’t produce as much output with higher strings as lower strings, particularly when playing solos high on the neck. This means that for solos, the distortion is less distorted, and the compression is less compressed. You need a way to increase levels at these higher frequencies.
A stage of parametric EQ is the solution. To start, set the frequency to around 3 kHz or slightly higher, boost to maximum, and Q to minimum (sharpest). This will increase the level of the upper strings, as shown in the following screenshot.
Admit it: You’re dependent on your computer, and don’t want downtime. So follow these tips, and keep it happy! Most of this applies to desktop computers, but there are some laptop tips as well.
Follow these tips, and your computer will thank you for it. May your machine never go down in the middle of a crucial session!
I’m often surprised by how many people don’t use MIDI effects. Hopefully, they’ll read this tip, and realize how beneficial they can be for any type of MIDI project.
MIDI effects work not by processing audio, but by processing the MIDI data itself. These effects run the gamut from the utilitarian (isolate note ranges or velocities, change duration or velocity, and the like) to more “artistic” effects like delays, arpeggiatiors, and step sequencers. Although there was somewhat of a “standard” for MIDI effects early on, unfortunately that’s no longer the case. However, search the internet for MIDI plug-ins, and you may be able to find ones that are compatible with your host.
MIDI plug-ins work in real time, so you can apply them to a track temporarily, but later make them permanent. For example, suppose you lay down a quick drum part but the timing is a little shaky. Rather than edit it, or apply permanent quantization, you may be able to apply a
High-impedance guitar and bass inputs were a welcome addition to audio interfaces—you could plug directly into your computer without loading down your guitar’s pickups, which can dull the tone. But what if you want to dull the tone? Some guitarists prefer the loading effects of a low impedance input, because they find a high impedance just too darn crispy. Reduced high frequencies can contribute to a smoother, rounder sound when feeding distortion and overdrive modules in amp sims.
To regain your duller tone, just add a resistor from the hot input to ground. Since you probably don’t want to modify your interface, there are two simple options.
180302 5 Ways to Save Time in the Studio
So was my custom cabinet better than the stock one? You can judge for yourself from the audio example; the first four measures use the constructed cab, the second four measures are the stock Gratifier cab, and both guitar parts are normalized to the same peak levels. But “better” or “worse” isn’t the point: the real point is that you can create something that’s ideal for your needs, rather than settling on something that was ideal for someone else’s needs.
180720 Envelope-Followed Flanging with...Melodyne?!?
As electronic music-making devices continue to shrink, it’s no longer possible to include a 5-pin DIN connector in something that may not be much larger than a smartphone. A lot of companies have addressed the problem by using 2.5 mm or 3.5 mm TRS (tip-ring-sleeve) connectors because MIDI requires only three leads, but there was no standard so there was no guarantee of compatibility with devices using TRS connectors.
Thankfully, the MIDI Manufacturers Association has adopted and released a specification that defines MIDI over TRS connectors. The tip connects to pin 5 on a 5-pin DIN connector, the ring connects to pin 4, and the sleeve (ground) connects to pin 2. This means you can create adapter cables easily to connect devices with 5-pin DIN connectors (like audio interfaces and controllers) to the new breed of mini-devices that use TRS connectors. As with standard MIDI cables, shielded, twisted-pair wires are specified as the conductors.
For more details, go to the blog post about the new specification. To download the specification for free, sign up to The MIDI Association (it’s free, they don't spam you, and there’s a lot of good info).
I have an M-Audio Fast Track Ultra whose most recent drivers work with Windows 8; Avid says that the interface isn't compatible with Windows 10. But it is! I've run into this situation before, and here's how to solve it.
1. Download the most recent drivers for your hardware (Windows 7, 8, or 8.1) - for example, Windows 8.
2. After the download, right-click on the executable .exe file, and choose Troubleshoot Compatibility.
3. In the dialog box that appears, choose Troubleshoot Program.
4. In the next dialog box, check The program worked in earlier version of Windows but won’t install or run now, then click Next.
5. In the next screen, choose Windows 8 (or the last OS with which the driver is compatible), and then click Next.
6.Test the program. If you don’t see the splash screen that asks if you want to allow the program to make changes to your computer, bring it to the front and follow the instructions.
7. If the program works, great! But we’re not done yet. Return to the Program Compatibility Troubleshooter, click on Next, and then choose Yes, save these settings for this program.
8. Finally, you’ll be presented with the Troubleshooting has completed window with a green check mark and the word “fixed.” Cool!
I used Cakewalk Sonar’s QuadCurve EQ to create the cabinet. Of course other EQs will work, but this one has built-in high- and lowpass filters with sharp cutoffs that are well-suited to virtual cabinet-making. The main modifications are a high-frequency rolloff to shave off the top end and amp sim harmonics, cutting the mids around 1.3 kHz to reduce the upper mids and make the 200-500 Hz range more prominent, and a steep, deep notch to minimize a buzzy resonance at 2.9 kHz. A mild bass rolloff with the highpass filter provided the open back effect, while adding a high frequency boost with a shelving EQ regained the perceived loss of highs from the notch. With my cabinet complete (fortunately virtual glue dries immediately!), I was ready to record.
180105 Regain Control of Your Locked Hard Drive
181005 Do Orange Drop Capacitors Really Make a Difference with Guitar Tone?
180914 Optimizing Guitar Tracks for Amp Sims
Although this signal chain in Overloud’s TH3 isn’t normal, who cares? The tone sounds great. The additional post-cabs EQ compensates for the loss of highs and increased lows caused by using two cabs.
Reverb > Chorus. Conventional wisdom says time-based effects follow modulation effects, but chorusing already diffuses the sound, and putting reverb afterward creates even more diffusion because the chorus effect now extends in time. Chorusing the reverb sounds tighter, because there’s nothing after the chorus to diffuse it further.
Tremolo > Overdrive. Tremolo after overdrive gives the most dramatic “slicing” effect. But I prefer placing it before overdrive causes the higher levels from the tremolo to distort more, and the lower levels to distort less, giving a subtler, more nuanced sound. The only drawback is you can’t use a lot of distortion, because that nukes the tremolo effect.
Try these tips for better beats.
ALWAYS PROGRAM YOUR BEATS WHILE OTHER INSTRUMENTS ARE PLAYING
You don’t need much: a bassline, a percussive synth part, and maybe a pad. Playing along with other instruments keeps your beats from getting off on some mutant tangent, and lets them play well with others.
The bass part is especially important. Program the bass first, and if it’s a line that makes you want to move, the drums will fall together perfectly. And when creating beats, be honest with yourself. If you don’t start moving around like a gerbil in heat when your drum loop plays back, the people on the dance floor won’t either. Don’t waste time fixing something that doesn’t work: start over from scratch, and remember you’re there to have fun.
Most loops are either one, two, or four measures. Each kind has a different personality, so get to know your beats, and use them for what they do best.
One measure: Aside from daytime television, there are few things more boring than a one-measure loop repeated by itself over and over and over again. So, a one-measure loop’s mission in life is to provide a background for other beats, percussion parts, or goofy sounds, so you can put together layers that work together. The best one-measure loops are plain and normal. Clever syncopations, if played over and over, are like a house guest who just won’t leave. For one-measure beats, simple = good.
Two measures: Two-measure loops are cool because they’re like aerobics – one measure breathes in, the next breathes out. The structure I use for two-measure beats is “plop-float-tease.”
Plop means a heavy downbeat. Make the velocity on the kick drum a little higher, increase the kick treble a bit so it hits harder, layer a low tom hit with the kick...anything that makes the sound plop. You want people to feel, not just hear, the downbeat.
Float is the middle section. This is more like the one-measure concept, you want something that’s fairly neutral and keeps the beat progressing, without calling a lot of attention to itself.
Tease disrupts that normal flow and sets you up for the next plop. This can be some tom hits, removing the kick and hats for a couple of beats while you slip in something else, a breakbeat, whatever. If you apply beatus interruptus, when the beginning of the loop hits again, you have a strong downbeat that “re-syncs” the dancer’s butts/brains.
Check out the screen shot and listen to the audio example.
You now officially have 4,372 things with Li-On batteries that need recharging, and keeping on top of those batteries is a hassle. What’s more, how you charge and discharge those batteries greatly influences their life. Although most chargers are smart and will turn off when a battery is fully charged, there have been battery recalls due to overheating and even explosions. So, it can’t hurt not to keep something plugged in all the time. But you also do not want to run Li-On batteries all the way down and then charge them all the way up; what makes batteries happiest is small discharges followed by small charges.
To deal with this, I’ve created a “charging station” that cascades several AC barrier strips to provide outlets for all my devices that need recharging. You usually can’t fit too many transformers on a strip, but short extension cords solve that problem.
Sometimes I like big, nasty kicks...and sometimes I like everything about an analog drum kit except the kick, so let’s take a wimpy analog kick and turn it into a powerhouse. The secret to this technique is sending the kick to a bus or other track, then in that bus or track, inserting saturation-based processing followed by a sharp low pass filter (48 dB/octave is good), set to a very low frequency. This particular example uses Sonar’s Tube processor and QuadCurve EQ (set to compact mode), but similar processors in other host programs work just as well. Mix this distorted/lowpass-filtered sound behind the main kick.
The audio demo plays two measures of a “stock” TR-808 drum loop, then two measures of the same loop with the killer kick processing, then repeats the same four measures for comparison. Of course, you need to hear this on a system that can reproduce bass adequately—airplane earbuds are definitely not recommended! (Come to think of it, they should probably never be recommended for anything...but I digress.)
You might wonder if the main reason for the increase in bass is the lowpass filtering, but after the initial set of 8 measures, there are two measures of the kick through lowpass filtering and then two measures of the kick with the Tube distortion added in—you’ll hear that the distortion is definitely the “secret ingredient.” And now, you can really get your...uh...kicks.
171229 Why MIDI Effects Are Totally Cool
180216 How to Beat Jet Lag
Did you ever experience some weird computer problem, and then solved it - so you continued on with the task at hand? Before you do, type up a note about how you solved the problem, and put it in a "useful information" folder. Mine tells what to do when Windows won't recognize my Blu-Ray drive after a Windows update, a way to turn Windows driver signing on and off from the command prompt, how to recover a hard drive that decided to be read-only, a file Waves tech support sent that describes which files need to be deleted for a clean install, and more. This type of information can be invaluable the next time you encounter the problem you solved - and you surely will!
180831 Yes! Ableton Live CAN Scan More than One Custom VST Folder - and Load VST3 Plug-Ins
If you have the right amp sim for your needs, but can’t find the right cabinet...then make your own! For example, I like the Gratifier Amp’s “Modern” setting in Guitar Rig, but the standard cab sounded more brittle than I wanted. Their “Citrus” amp cab often does what I need, but I needed a beefier lower midrange, and more of an open back sound to give the bass more space.
180525 Make Sample Playback Synths More Expressive
This is a follow-up to the previous tip, because people who listened to the song (see above, click, and of course, like!) wanted to know how I got that delicious feedback guitar sound. Well...it wasn’t feedback guitar, it was synthesizer—and you can get the same kind of effect. It’s really quite simple; there are three elements:
The latter is the key to this technique. Simply layer two sine wave to produce the “faux feedback.” I transpose one 19 semitones (octave+fifth) above the fundamental, and the other 31 semitones (2 octaves+fifth). The low-pitched one has an envelope that builds up to a peak in about two seconds (see the screen shot), and then decays while the higher-pitched feedback appears over about 3 seconds. It’s important to set these envelopes for a believable attack time—long enough to that the feedback shows up with a long, sustained note; not so long that it never shows up; and not so short that it shows up all the time. Having done a lot feedback guitar in my time (ahem), it doesn’t happen instantly—you have to coax it into happening. The attack time represents the time needed to coax it into happening, and to jump from one harmonic to a higher-pitched one.
Amp sims are critically dependent on input levels - especially with distortion-oriented presets. Because your guitar tracks are always recorded dry, they retain the naturally percussive nature of guitar strings. So, unless you guitar levels are consistent, you'll find yourself having to adjust the input Gain or Drive control constantly to compensate for changes in level.
That is...unless you normalize your dry guitar tracks so that the levels are at least somewhat consistent. Before mixing a guitar part, I'll normalize the guitar track and then recall the desired amp sim preset. The sound will be predictable, because the input level will be predictable. Often, it's not necessary to make any tweaks; when it is, a quick edit on the input Drive or Gain will be all that's needed to optimize the guitar sound for the track.
181116 Dealing with Rechargeable Batteries
And since you’re probably curious as to what this sounds like, my latest song showcases it. Listen to the following song—a dance remix of “To Say 'No' Would be a Crime” from my “Simplicity” album (click on the Music tab)—and listen carefully around 1:48 and 3:03, where I’ve inserted "time traps" to draw things out a bit. Fun stuff!
Screenshot: Exporting individual Cakewalk by BandLab tracks prior to loading them into Studio One for mastering. This same technique works for other programs as well.
As mentioned over the years, I use several different DAWs because each has its own attributes. Of these, Studio One has a unique integration between its multitrack Song page and mastering-oriented Project page. If only other programs had the same feature...well it turns out that even if they don’t, they can make friends with the one that does.
In Studio One you can mix your song down to stereo, and directly into the Project page for mastering. The unique aspect is if you’re putting together a collection of songs and find that, for example, the vocal on one of the songs is a little soft compared to the others, you can switch over to it in the Song page, make the change, then remix it automatically back into its place in the Project page. The Project page then contains the new mix.
It’s easy to assume that to take advantage of this, you need to start your song in Studio One. But that’s not quite the case. I often taken songs created in Sonar or Ableton Live, not just Studio One, and export all the song tracks as individual files (“stems”). I then import them into a new Studio One Song, verify that it sounds the same as it did in the original program, and mix the stems down to the Project page.
Now I can proceed with the mastering. But if a tweak is needed, it’s easy to make any changes to the individual stems—change levels, EQ, even add processing if needed. And if the stem has some horrible flaw that the Song page can’t fix, it’s always possible to go back to the original program, make the fix, and re-export the stem.
Oh, and if you’re thinking “but then I have to learn another DAW”—what I’ve described uses only a fraction of Studio One’s feature set. I figured out how to do it a couple hours, so I’m sure you can too. It’s definitely a “best of both worlds” situation.
In a probably not unexpected trend, fewer audio interfaces are including the five-pin DIN connectors that were part of the original MIDI protocol (your trivia for today: DIN stands for Deutsche Institut für Normung, a German standards organization). These transfer data into the computer via the MIDI in, and transmit computer data to a MIDI-savvy sound generator or other MIDI gear via the MIDI output.
These days, it’s more likely that MIDI data will be carried over USB, so many newer MIDI devices connect to computers with USB rather than 5-pin DIN-compatible cables. Fortunately some controllers have both options, but the handwriting’s on the wall: MIDI’s future is not 5-pin DIN connectors.
But MIDI’s past is, and fortunately, the MIDI spec has always paid attention to making sure older MIDI gear doesn’t become obsolete. If your audio interface doesn’t have 5-pin DIN connectors but you need them, it’s not a problem. MIDI interfaces are available that connect with a computer’s USB port and provide MIDI in and out with 5-pin DIN connectors. Some of these interfaces are simply cables, which makes them very convenient if you have an instrument or two that you need to connect to your computer via USB. The photo shows ESI’s MIDIMATE eX, which provides DIN-to-USB and USB-to-DIN conversion. What impresses me the most about this particular converter is being able to take the output from an Ensoniq TS-10 with polyphonic aftertouch, turn off local control, echo the data through the computer, and feed it back to the TS-10’s sound generators—without any loss of data. Pretty cool.
180420 What Do Those Spectrum Response Parameters Mean, Anyway?
Spectrum analyzers can definitely help during the mixdown process by either providing visual confirmation for what you think you hear, or providing a visual “early warning system” for issues you haven’t heard yet. Spectrum analyzers range from simple eye candy toys to sophisticated test equipment, but even some of the more modest ones have adjustable parameters so you can customer their response. However, it’s not always obvious what these parameters do. So, here are the most common parameters, and what they mean.
CUSTOMIZING SPECTRUM ANALYSIS RESPONSE
Spectrum analyzers vary greatly in terms of their adjustable parameters, from simple – you can’t adjust anything – to multiple parameters that let you customize the analysis and display process. Here are some of the most common parameters.
180518 31 MIDI Tips!
181026 A Different Kind of Gated Reverb Effect
I’ve covered how some virtual instruments sound better in 96 kHz projects because this eliminates foldover distortion (aliasing), and mentioned some ways to obtain the benefits of recording at 96 kHz in 44.1 kHz projects. Most of these involve exporting a MIDI track, opening it in a 96 kHz project, opening the virtual instrument, rendering, and then sample-rate converting back down to 44.1 kHz.
However, if a project consists of only virtual instruments (i.e., no audio tracks) there’s a simpler way—change the project sample rate to 96 kHz (or higher, if you want), render the instruments to audio, and then change the project sample rate back down to 44.1 or 48 kHz. Most programs will sample rate convert automatically when you change the project sample rate, and now you’ll have rendered audio files free of aliasing.
If you do have a project with audio files, all is not lost. Save As... a new project, strip out everything except the virtual instruments, bump up the sample rate, render each instrument as audio starting from the project start, and then export the rendered files. Import them into the original project; with some programs, you can open up two instances, and just drag the rendered files from the high-sample-rate project over to the original one.
171103 Better MIDI Bass Parts
180316 Why It's a Good Idea to Render Instrument Tracks
171201 Out-of-Phase Pickup Tone Emulation with EQ
180921 Fixing MOTU Digital Performer Problems with Windows
Why be normal? With some basic editing, you can create your own signature sound from sample-based instruments. Sometimes even a simple parameter change or two is all you really need to customize a sound to fit your needs. Like what, you say? Well, like...
LFO WAVEFORM CROSSFADES
Using an LFO to crossfade between two waves (each must be followed by its own DCA) provides a less static, more animated sound if you choose related waveforms (e.g., two different organ sounds, 5% and 50% pulse waves, two different basses, etc.). Note that you may need to tweak the oscillator levels a bit so that there’s no significant level variation between the two.
A PEAK EXPERIENCE
Synths can often generate strong peaks, and unless you tame them, they may create havoc when recording. Proper synth programming can help; for example, even though detuned (chorused) oscillators sound fat, there’s a substantial output boost when the chorused waveform peaks occur simultaneously. To reduce this, drop one oscillator’s level about 30% – 50% compared to the other. The sound will remain fat, yet the peaks won’t be as drastic.
High-resonance filter settings are also a problem if you hit a note at the filter’s resonant frequency. Try adding a limiter at the output to cut peaks down to size (use as fast an attack as possible).
Having an LFO pan an instrument sound back and forth is usually pretty gimmicky (although this can work with short percussive sounds, as you don’t hear them long enough to detect an audible sweep). However, one panning technique can sound quite natural: Modulate panning with velocity. When you first hit a note its stereo position will depend on the velocity, but as it sustains, it will retain its location in the stereo field until replayed.
WHY SINE AND TRIANGLE WAVES DESERVE RESPECT
Do you think sine and triangle waves are the most boring waveforms in the world? They actually have many uses. For a fuller acoustic guitar or piano sound, layer a sine wave along with the lower notes. To attenuate the sine wave at higher notes, modulate the wave’s amplitude negatively according to keyboard note position (i.e., the higher you play on the keyboard, the lower the level). Also keep the overall level low — just enough to provide a subtle psycho-acoustic boost.
In fact, sine and triangle waves can add more depth to almost any sample because digitally-generated waveforms can have more presence than digitally-recorded sounds. For example, harp samples may lack a bit of “you are there” presence due to mic limitations, room acoustics, etc. Layer a triangle wave with the harp (adjust the triangle’s amplitude envelope so that it “tracks” the harp); the triangle wave provides presence, while the sample provides detail and realism. Initially set the triangle wave level to 0, then bring it up slowly to taste. Keep it subtle — we’re talking background reinforcement, not something obvious.
Here’s another triangle trick: To add some male voices to an ethereal female choir, layer a triangle wave tuned an octave lower. This gives a powerful bottom end that sounds like guys singing along. To maintain the ethereal quality in the upper registers, consider modulating the triangle wave amplitude according to keyboard position so that the triangle wave is not apparent on higher notes.
MORE RESPONSIVE PARAMETERS
“Doubling” modulation routings can make a parameter more responsive. For example, most keyboards have a global pressure control, adjustable for heavy, light, or moderate action. I usually choose moderate, but occasionally need a patch to have a lighter, more responsive feel. Assigning pressure twice to the same parameter (such as overall level or filter cutoff; most parameters can accept more than one modulation source) increases the sensitivity for just that parameter. The controllers will sum together, thus creating more change for a given amount of pressure. This same trick works for velocity.
To strengthen an instrument’s attack, take advantage of the fact that bass sounds (slap bass, synth bass, plucked acoustic bass, etc.) tend to have fairly complex attacks. Transpose the bass wave up an octave, and layer it behind the primary sound. You’ll probably want to add a fairly rapid decay to the bass so that its sustain doesn’t become a major part of the sound.
BETTER STRINGS THROUGH LAYERING
String synthesizers of the 70s, based on sawtooth or pulse waves, created rich, syrupy string sounds that weren’t super-lifelike, but nonetheless sounded pretty cool. Sampled strings may sound more realistic, but often lack the smoothness of analog simulations. For the best of both worlds, dial up a sawtooth or pulse wave, and adjust its envelope for as realistic a string sound as possible. Now layer it behind a string section sample, and the synthesized waveform will “fill in the cracks” in the digital waveform.
PITCHED PERCUSSIVE TRANSIENTS
Percussion instruments, when played across a keyboard, acquire a sense of pitch. Layering these with conventional melodic samples can yield hybrid sounds that are melodic, but have complex and interesting transients. Cowbell is one of my favorite samples for this application. Claves, triangle dropped down an octave, struck metal, and just about any other pitchable percussion can also give good results.
The above suggestions are just the tip of the iceberg. Sample playback synths can be a rich source of sounds that exceed your expectations, but you have to get in there and do some parameter value tweaking. Go ahead and mess around — you have nothing to lose but sounds that are like everybody else’s.
181123 The "Useful Information" Folder
181109 Get Sustain with Amp Sims
I have a lot of plug-ins. As in, a lot, and some of them are old or haven't been used much, so they hadn't been updated. Recently, I installed the latest version of Vegas Pro 16 (an excellent program, by the way) but when doing the initial scan, it would get waylaid by certain plug-ins, requiring me to start the scan over again.
The solution: I renamed all folders containing VSTs so Vegas wouldn't recognize any of the plug-ins, and of course, it opened right up. Then I could rename one folder, open up the VST folder preferences, add that one folder to the scan, close, and re-open to find out if that folder had the problematic plug-in.
Eventually, it turned out to be some plug-ins that were locked to a particular program, and apparently threw a hissy fit when another program tried to open them. I isolated them to their own folder, placed that in only their host's scan path, and all was well.
If you think a keyboard is only for playing notes, four or five octaves may be sufficient. However, many virtual instruments (e.g., FXpansion Geist, Native Instruments Kontakt, IK Multimedia SampleTank, EastWest’s Play engine, etc.) use MIDI keys not only to play specific notes but also to trigger articulations or variations on a basic sound.
If your main USB MIDI controller doesn’t have enough notes, no worries—trade it in for that deluxe 88-note weighted keyboard you’ve always wanted (hey, you only live once). But if you lack the space or finances, add a second USB MIDI controller for doing switching—even if it’s just something like a little Korg plastic keyboard designed for mobile applications. Your sequencer probably won’t be able to merge incoming MIDI streams, but no worries there either: MIDI Solutions’s Merger, which costs about $70, will merge two data streams to a single output. There are also several DIY circuits for MIDI mergers on the web.
180608 The DAW/Mastering Connection
171117 Synthesizer Meets Feedback Guitar
180406 Why You Need a Keyboard Controller with More Octaves
Recording music is supposed to be a joyful experience, not a stress-inducing one. So, here are some tips on how to save both time and stress in the studio.
1. Use the plug-ins and virtual instruments bundled with your host as much as possible. Most hosts now include a decent assortment of instruments and processors, which has several advantages:
Granted, bundled instruments won’t do everything. But keep your collection of instruments manageable: avoid the temptation to download a zillion shareware plug-ins “just because you can.” It’s more to learn, more to maintain, and more that can go wrong.
2. Choose an audio interface with lots of inputs. The more inputs you have, the more instruments and mics you can keep patched in permanently: mics, guitars, keyboards, etc. You don’t want to re-patch; it’s great to have everything ready to go, so all you need to do is record-enable a track to make music.
3. Manage your updates. Schedule a time for updates (e.g., once every month or so), then check for updates to your plug-ins, host, operating system, graphics card, etc. Mass upgrading can be more efficient than doing one update at a time.
4. Print a list of keyboard shortcuts. Refer to it often; after a few weeks, you’ll have the list memorized—and keyboard shortcuts save serious amounts of time.
5. Learn to cut your losses. Sometimes a performance or a song just isn’t happening. You try some EQ, some effects, some mix changes, maybe an overdub or two...nope. You’ve written music before, and you’ll write music again. If something isn’t flowing right, don’t complicate your life: Cut your losses and move on!
We all know that open back amps have less bass response than closed-back amps. In the studio, it's not too hard to convert a closed-based amp sound to an open-back sound - just use some EQ to thin out the bass end a bit. It's a little harder to do the reverse, because a closed-back amp sort of "compresses" the speaker so the sound is tighter, not just bassier. So here's a simple solution: place the open back amp so the back faces down on a rug, and point a mic down at it. There's only one caution: if the amp has tubes in it, you don't want to block the ventilation any more than needed.
2 Mic placement “flight simulator.” Most amp sims lets you move “virtual mics” around in relation to the virtual amp. The results parallel what you’d hear in the “real world,” and you can learn a lot about how mic placement affects the overall sound; IK Multimedia’s AmpliTube (above) is particularly good in this respect.
3 Pads matter. Many mics have switchable attenuator switches (called “pads”) to lower the sound level, for example by -10dB. With loud amps, engage this to avoid distortion.
4 Mic Placement. Start off with the mic an inch or two back from the cone, perpendicular to the speaker, and about half to two-thirds of the way toward the speaker’s edge. To capture more of the cabinet’s influence on the sound (as well as some room sound), try moving the mic a few inches further back from the speaker.
Moving the mic closer to the speaker’s center tends to give a brighter sound, while angling the mic toward the speaker or moving it further away provides a tighter, warmer sound. Also, the amp interacts with the room: Placing the amp in a corner or against a wall increases bass. Raising it off the floor also changes the sound. Also note that each speaker in a cab should sound the same, but that’s not always true; mic each one and listen for any significant differences.
When you see “Disk attributes cleared successfully,” you’re done! Close the command prompt box, and now your formerly locked drive will be ready to accept your precious data.
In theory, Live allows one custom VST plug-in folder - and you'll see people on the web grumbling about this limitation. But if you’re like me, you probably have several folders for VST plug-ins where plug-ins have been installed over the years.
No problem: create shortcuts for your custom VST folders, and then drop the shortcuts into Ableton’s VST plug-in folder. You can have as many shortcuts as you want, as well as nest folders within the VST folders.
Also note that with DDMF's Metaplugin, you can run VST3 effects and instruments inside of Metaplugin, which can load into Live as a VST 2.4 plug-in. There's a demo to make sure it works with your plug-ins. Problem solved :)
Sometimes I’ll want to listen to what’s happening on a DAW without being anchored to a monitoring position—like listening to a song loop in the background as I move around the house to hear if anything annoys me. Fortunately, with Windows you can stream your computer’s audio over Bluetooth. To do so, you need to use Windows drivers, not ASIO. MME is the fail-safe option, but I’ve had good luck with WASAPI as well. Here’s what you need to do.
And now, you can stream audio to your Bluetooth audio portable speaker—enjoy!
171124 Cooler Sampler Presets by Augmenting Samples with Waveforms
180202 5 Cabinet Miking Tips
180817 How to Hear the Results of Dithering
There are standard ways to arrange the order of effects—like compressor before distortion. But you already know that, so let’s look at how to improve your amp sim sound by doing what we’re not supposed to do.
Cabinet > Crunchy/Distorted Amp > Cabinet. You still want a cabinet after the amp, but putting a cabinet before the amp can “focus” the guitar’s sound by taking off some of the highs and beefing up the mids and lows. Bypass the amp and post-amp cabinet, then try different pre-amp cabs until you find one that “fattens” the guitar sound by itself. Enable the amp and post-amp speaker, and you’ll likely hear a more focused sound.
Distortion Stomp Box > Cab. No law says you must include an amp. Use a fuzz, like a Muff Pi or Rat emulation, and try different cabs. You’ll hear a different, and sometimes sweeter, distortion sound compared to distortion > amp > cabinet.
Amp > Cab > Cab. It’s tough to design a cab to sound right, and you’ll often hear a sort of plastic, “filtered” tone. Try two different cabs, like a 1 x 12 followed by a 4 x 12. With the right cabs, you can end up with a fatter, smoother, and more even-sounding tone. However, the two cabs in series will tend to boost the bass and diminish highs, so add EQ afterward. Try a broad boost in the 2-8 kHz range, and and then roll off gently in the bass range. Tweak the EQ as needed until the resulting sound sits better in a mix.
I leave the charging station off. When I take an item from it, after replacing it I turn the barrier strips on and charge everything for a while. This restores the charge on batteries that have lost some of their charge while unused. If I use a device while it’s plugged in (e.g., a Bluetooth speaker or whatever), then I also turn on the barrier strips. This way devices don’t discharge too far before getting charged up again, and they’re not left on all the time.
Okay, this isn’t your standard tip of the week...but it’s particularly important for musicians who have to fly somewhere and need to be at the top of their game.
I used to have a terrible time with jet lag. If I had a gig in Europe, I either had to arrive a couple days beforehand, or just try to tough it out at a sub-optimal level. Over a period of years, I tried different techniques and eventually found what worked. Although every person is different and has different reactions to jet lag (for example, it’s documented that some people have a harder time traveling from east to west or west to east), several other people have tried my recommendations and reported success. So…here’s the scoop.
1.Reset your watch to destination time as soon as you step on the plane. Even better, start thinking in “destination time” before you leave. For example, if you’re having breakfast in California and traveling to New York (3 hours ahead), eat lunch-type food and think “lunch.” As you travel, don’t think about the time back home—some jet lag is “all in the mind.”
2. If your flight occurs during destination sleep hours, learn to sleep on the plane. Here’s how:
3.Traveling west extends your day. Consider staying up later than usual and sleeping in as late as possible the day of the flight.
4. Diet greatly affects how your body copes with jet lag. Even if you can’t sleep on planes, following the right diet will cut the effects of jet lag. Eat high-protein breakfasts and lunches for long-lasting energy during the day, and high-carbohydrate dinners, which give a quick burst of energy, after which you get drowsy (e.g., if you get on the plane and need to sleep, eat the pasta instead of the chicken). Carry this regimen through the first several days of your trip. The worst thing you can do when you arrive at your destination is have a steak for dinner; it will make you drowsy initially, but within a few hours, wake you up as it metabolizes.
5. Caffeine can both reset and screw up your body clock, depending on how you use it. I find that after arriving at my destination, if it’s before noon I’ll drink a couple cups of coffee to make step 6 (see next) easier. Don’t have coffee (or tea) past noon, unless you’re heading west and want to extend your day.
6. Don’t take naps, as these can really disrupt your body clock. For example, I usually arrive in Europe between 9AM and noon, after getting 6 or so hours of sleep on the plane. I force myself to stay awake until an early dinner, then it’s off to sleep around 8 or 9 PM. 10 hours of sleep, and I’m at 100% the next day. Afternoon naps can promote waking up in the middle of the night and not getting back to sleep but if you must have a nap (e.g., you have an important dinner meeting or evening gig and don’t want to fall over), set your alarm and sleep for no more than 20-30 minutes.
I used to be wiped out for almost a week when going to Europe from the US. Now I can land in the morning and play a gig that night, or cover a trade show the next day with no problems. I still get a bit of jet lag when going west, but overall, it’s a big improvement.
181019 Create Stretchable Audio Files for Free
180713 Multi-Timbral Instrument vs. Multiple Instances—Which is Better?
Some people swear that Orange Drop capacitors sound noticeably better than standard capacitors. Others insist that if two capacitors from different manufacturers have the same value, they're going to sound the same - period. So who's right?
Both of them. If two capacitors have the same value, they will sound the same. However, just because two capacitors say they have the same value doesn't mean that they do. Ceramic capacitors in particular are subject to wide variations with temperature changes - so if your guitar's tone control uses ceramic capacitors, the tone may change as the stage lights heat up your axe.
Orange Drop capacitors were known for being stable and accurate, so the value printed on the capacitor was very close to the actual value. As a result, Orange Drop capacitors can sound "better" not because of sound quality, but because they'll provide the sound you expect to hear - not one that's been altered by temperature.
The reason compressors produce pumping, breathing, and other artifacts is because they have to work too hard. If the compressor is seesawing back and forth between no signal, too much signal, applying gain control, releasing gain control, figuring out which part of the knew to track...well, I’m getting tired just thinking about it.
If you want truly transparent compression, parallel compression can help but the most effective method I’ve found, particularly for vocals, is placing two compressors in series, with both set for light compression. When adjusted properly, the result is a significant amount of compression but the sound will be less obvious than using a single compressor to give the same amount of compression. The first stage doesn’t have to work too hard, and it “pre-conditions” the signal so that the second compressor doesn’t have to work too hard, either.
The main drawback is that unlike standard compression, where you need adjust only one set of controls, the à la carte approach requires adjusting two sets of compressor. While this might seem like a disadvantage, most of the time you’ll set them to similar settings anyway. You may even find that you can save the first compressor’s settings as a preset, and just load the same preset into the second compressor.
You might also think there would be added noise, but in practice, that doesn’t seem to be the case; because of the division of labor between the two compressors, they also divide the amount of noise they generate.
So next time you want really transparent compression on vocals, try the dual compressor approach—and it works on other signals, too.
It’s that time of the year when many of you will be flying somewhere to be with family and friends. But if you’re taking a laptop, there are some special considerations you need to consider before calling Lyft to take you to the airport—so here’s a follow up to last week’s set of tips. We’ll get back to music next week!
Actually, yes! This works not only for individual tracks, but is pretty amazing with program material. Even better, the flanging is not your basic, boring “whoosh-whoosh-whoosh-lookit-me-I’m-a-flanger!”-type flanging, but follows the amplitude envelope of the track being flanged. It’s also easy to do.
1. Duplicate the track you want to flange.
2. Set up Melodyne to edit the duplicated track.
3. In Melodyne, choose the Percussive algorithm.
4. Select all the Melodyne blobs.
5. Set the Pitch Grid to No Snap.
6. Click on one of the selected notes, and drag slightly off pitch—certainly less than half a semitone to start. You’ll hear some ultra-cool flanging.
Don't believe me? Check out the audio example!
It’s one of those age-old questions, so let’s answer it. Here’s the situation: You have a multi-timbral instrument, like IK SampleTank or NI Kontakt. You want to play back four instruments. Is it more efficient for your computer to run one instance and load it up with four instruments, or run four instances, each with its own instrument?
Testing time...I opened up Cakewalk by BandLab on a four-core, i7 machine because the program has a performance meter that shows what’s going on with the individual cores. Each core runs at 3.10 GHz.
For the first test, I loaded up SampleTank 3 with four fairly processor-intensive instruments—two different pianos, and two different guitars, each driven by an insane drum ‘n’ bass MIDI groove to make sure each instrument got a lot of exercise.
The second test used the same instruments and the same MIDI files, but each instrument was loaded into its own instance of SampleTank 3.
And here are the results. The image on the left shows that with a single multi-timbral instance, the first core is doing all the work. The image on the right shows that with multiple, single-instrument instances, the load is distributed more evenly over the four cores. However, note that running more instances uses up more RAM. So the bottom line is if your priority is conserving RAM, then use a single instance and load your instruments into it. If your priority is CPU efficiency, then use multiple instances, each with its own instrument. You can also choose to "split the difference," like have two instances, each with two instruments. There may not be a "universal answer"...but at least we have an answer!
Having problems with DP crashing and doing strange graphic things (like meters not moving) with Windows? It’s almost certainly related to a misbehaving plug-ins. Choose Edit > Preferences > General > Audio Plug-ins, and create a new plug-in Set. Check only the MAS plug-ins, and everything should work properly. Then, start adding plug-ins back in to narrow down which one (or ones) is causing the problem. Generally, you’ll find if one plug-in of a company’s suite works, the rest of them will (e.g., all of IK Multimedia’s T-RackS plug-ins work fine). Older plug-ins, wrapped plug-ins, and freebies are the ones to suspect.
180209 Baking Better Beats
180511 Improve the Sound Quality of Virtual Instruments
In the above screen shot, the gray notes are a kick that lands right on the beat. The dark blue bass notes hit at the same time as the drums, and are behind the beat just a little bit to make sure the kick gets the spotlight - this way the music feels more rhythmic than melodic. With MIDI sequencers, a track shift function will take care of moving the start time for selected notes. With hard disk recorders, you can simply grab a part on-screen and shift it, or use a “nudge” function (if available). Even a few milliseconds of shift can make a big difference.
If you want to bring just a couple instruments out from a mix, patch a distortion plug-in set for very little distortion into an aux bus during mixdown. Now you can turn up the aux send for individual channels to make them jump out from a mix.
PITCH CHANGES IN SYNTH ENVELOPES
This involves a little synth programming, but the effect can be worth it. As one example, take a choir patch with two layers (the dual layering is essential). If you want this sound to draw more attention to itself, use a pitch envelope to add a slight downward pitch bend to concert pitch on one layer, and a slight upward pitch bend to concert pitch on the other layer. The pitch difference doesn’t have to be very much to create a more animated sound—remove the pitch change, and notice how the choir sits further back in the track.
MIXING VIA EQ
EQ is a very underutilized resource for mixing. Turning the treble down instead of the volume can bring a track more into the background without having it get “smaller,” just less “present.” A lot of engineers go for really bright sounds for instruments like acoustic guitars, then turn down the volume when the vocals come in (or some other solo happens). Try turning the brightness down a tad instead. And of course, being able to automate EQ changes makes the process go a lot more easily.
Overall, when it comes to mixing you have a lot of options other than just changing levels, and implementing changes in this way can make a big difference to a mix’s “character.” Try adding some of the above tricks—or similar ones of your own making—to your mix, and you’ll add yet another dimension to your sound.
Friday's Tip of the Week
Some tips are about production, some about playing, some about music...it all depends on what seems like fun that
week.Tips are arranged with the oldest first, and newest last; scroll down to see the most recent tips.
PLEASE NOTE: This section has been archived; I'm now doing a Friday Tip of the Week for the PreSonus Blog.
How much difference does 2 millimeters makes? A lot, actually. When I compared the output from a pickup 2 mm away from the strings to 4 mm away, the peak level went down about 8 dB. However, in the close position, the level drops off rapidly after the initial transient. In the far position, the transient is lower, and there’s a more consistent average level that leads to more sustain.
In the above image, look what happens when you normalize the two signals. In the second waveform, which is from the pickup in the far position, look past the transient: the average level is higher,and the sustain is stronger.
Even better, the reduced transient response with the pickups further away from the strings is helpful when feeding compressors, because the gain control action is smoother; large transients tend to “grab” the compressor’s gain control mechanism, which can create a “pop” as the compression kicks in. Also, amp sims generally don’t like transients as they consist more of “noise” than “tone,” so they don’t distort elegantly. Reducing transients can give a less harsh sound with a note’s attack.
So what about the lower overall level? Just turn up the input control on your interface, or the drive control in your amp or amp sim to compensate.
If you’ve set your pickups up close to your strings for more level, try moving them further away and applying more gain. You’ll like the extra sustttaaaiiinnn.
180309 The Key to Transparent Compression
If you’ve heard people talking about adding “glue” to a mix, this usually involves a bus compressor. But you can also “glue” tracks together in a subtle way by placing two standard compressors in series with high thresholds, low ratios, and no attack time. The result is dynamics control that’s so gentle, you won’t really hear that a compressor is working—but you will hear the benefits.
It’s difficult to give precise settings because the levels on tracks differ, but the level going into the first compressor is fairly high—around -3 dB—here are some suggested settings as a starting point.
Compressor 1: Threshold -6, ratio 1.3 : 1, knee 3 dB, attack 0 ms, release 200 ms
Compressor 2: Threshold -12, ratio 1.5 : 1, knee 9 dB, attack 0 ms, release 200 ms
You’ll need to adjust makeup gain based on the signal levels you’re using.
With lower input levels, lower compressor 1’s Threshold control but note that these settings are optimized to work with higher signal levels, like what you’d expect from program material. It’s not intended to work like a conventional compressor that flattens an input with a highly variable level, although you can always increase or lower the ratio or threshold for the best sound.
Now, try the following test with a finished mix:
Happy squashing—or perhaps more appropriately, happy anti-squashing.
Gating a signal with a noise gate, and controlling the noise gate from a different signal by using the gate’s sidechain input, can produce some really cool effects. One of the best known is locking bass to kick by gating the bass with a kick, but here’s something different and sort of twisted. It does require that you have kick and snare drums on different tracks.
Now the kick drum will gate the snare’s reverb, producing interesting—yet synched—rhythmic effects.
5 Direct + miked issues. Some amps offer direct feeds (sometimes with cab simulation); combining this with the miked sound can give a “big” sound. However, the miked sound will be delayed compared to the direct sound—about 1ms per foot away from the speaker. The top waveform is the direct feed, and the second one down is the miked audio. Nudging the miked sound earlier in your recording program lines up the miked and direct sounds so they are in-phase.
The big 1 and 2 are measure starts. The little numbers are beats within the measures. (By the way, Reason doesn’t look like this; I’ve colorized it to show up certain beats.)
The green note at the beginning is an extra, deep-sounding kick to give that satisfying downbeat “plop.” The middle pretty much floats, with kicks occurring on every beat (shown in yellow), although there are also a few semi-tease accents just to keep things interesting around measure 1, beat 3 and measure 2, beat 3.
The “tease” part starts just before measure 2, beat 4. The gray snare hit comes just before the beat—a big element of surprise. The kick that would normally occur at measure 2, beat 4 isn’t there; instead, it’s been shifted to the last beat to lead into the downbeat better. Finally, an extra hi-hat hit (blue) adds even more interest.
THE SWING THING
Swing lengthens the first note of an equal-valued pair of notes, and shortens the second one to compensate. Especially for hip-hop type tempos that are 100 BPM or less, injecting swing is like taking Vitamin Beat. Even a little, like 55%, will make a difference.
LOSE THE CYMBALS
Cymbals are musical one-night stands: you want them to show up, party, and leave. So make loops without cymbals, then add one-shot (single event, non-looped) cymbals on a separate track.
Don’t get fancy with the kick, snare, and hats. You need a rock-solid foundation so dancers can feel the groove. But you also need some ear candy on the top, which is percussion’s job description. But behave yourself: keep the levels sane (you want them to complement, not dominate), and use velocity a lot to vary dynamics.
AND NOW, A WORD FROM "BIG AL"
Albert Einstein once said that E = MC2, which means if you get enough mass moving fast enough, it becomes energy. That’s the whole point of beats. Get those butts moving, and you’ll create a lot of energy. More energy = more dancing = more sweating = more people going to the bar for drinks = more money for the club owner = job security for you.
I had given up on Windows 10's Groove Music player because of several issues, but didn’t like other music players as much. So eventually, I figured out how to make Groove Music behave—check it out.
I keep music files (WMA, AAC, MP3, FLAC, etc.) on a USB stick so the music doesn’t take up internal storage. Format the stick as NTFS, then right-click on it and choose “Allow files on this drive to have contents indexed in addition to file properties.” This seems to help when organizing files. If you need to make a copy for a car’s audio system, format a USB stick as FAT32 and copy over the files from the “master” music library.
Fixing “Unknown Album” and “Unknown Artist”
Any music without an album tag will end up in album called “Unknown Album.” Click on it to see a list of the music Groove Music can’t sort. Right-click on a title and choose Edit Info to see the music’s full name, but DON’T edit the info—I’ve had some songs or albums seemingly disappear from the Groove Music list after editing. You can find it via search, and then choose Edit Album Info to make changes, but I prefer to bring tagless music into a tagging program (like MP3Tag, Tigo Tago, or TagScanner). I use the VLC Media Player from videolan.org; drag the music you want to tag into it, and choose Tools > Media Information. Enter the album and artist info, then click on Save Metadata.
Adding Album Artwork
Groove Music can find databased album artwork if you right-click on the album, choose Edit Info, and then click on Find Album Info (make sure Show Advanced Options is on). If no artwork is available, check Amazon or other sites that sell CDs. Download the art, right-click on the album that needs artwork, and choose Edit Album Info. Click on the generic art, navigate to the desired artwork, then click on Open. After the artwork appears, click on Save. Groove Music can scale art, but I haven’t tested its limits.
There are three programs that can create and edit Acidized stretchable files: Magix Acid Pro, Magix Sound Forge, and Cakewalk by BandLab - the latter is a free download, although it's Windows-only. Go to BandLab.com, register, go to Apps, and download Cakewalk by BandLab. Follow the documentation for information on how to create and edit stretchable audio files. Note that most DAWs recognize these files, and will stretch them to fit tempo changes sometimes, pitch changes as well.