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Category Archives: Friday Tip of the Week


Friday Tip of the Week: Upsampling in Studio One, Part 1

The controversy about whether people can tell the difference on playback between audio recorded at 96 kHz that’s played back at 44.1 kHz or a higher sample rate (such as 96 kHz) has never really been resolved. However, under some circumstances, recording at a higher sample rate can give an obvious, audible improvement in sound quality. In this week’s tip we’ll investigate why this happens, and in next week’s tip, tell how to obtain the benefits of recording at a higher sample rate in Studio One with 44.1 and 48 kHz projects.

 

REALLY? CONVINCE ME!

A Song’s sample rate can make a difference with sounds generated “in the box,” for instance using a virtual instrument plug-in that synthesizes a sound, or distortion created by an amp simulator. Any improvement heard with high sample rates comes from eliminating foldover distortion, also known as aliasing.

Theory time: A digital system can accurately represent audio at frequencies lower than half the sampling rate (e.g., 22.05 kHz in a 44.1 kHz project). If an algorithm within a plug-in generates harmonic content above this Nyquist limit—say, at 40 kHz—then you won’t hear a 40 kHz tone, but you will hear the aliasing created when this tone “folds down” below the Nyquist limit (to 4.1 kHz, in this case). Aliasing thus appears within the audible range, but is harmonically unrelated to the original signal, and generally sounds pretty ugly.

Foldover distortion can happen with synthesized waveforms that are rich in harmonics, like pulse waves with sharp rise and fall times. (Amp sims can also be problematic; although their harmonics may be weak, if you’re applying 60 dB of gain to create overdrive or distortion, the harmonics can be strong enough to cause audible aliasing).

SO IS IT A PROBLEM, OR NOT?

Not all plug-ins will exhibit these problems, for one of four reasons:

  • The audio isn’t rich enough in harmonics to cause audible aliasing.
  • The plug-in itself can oversample, which means that as far as the plug-in is concerned, the sample rate is higher than that of the Song. So, any foldover distortion occurs outside the audio range.
  • The project sample rate is high enough to provide the same kind of environment as oversampling.
  • The plug-in designers have built appropriate anti-alias filtering in to the algorithms.

Many modern virtual instruments and amp sims oversample, and DAWs can handle higher sample rates, so you’d think that might be the end of it. Unfortunately, there can be limitations with oversampling and higher project sample rates.

  • Recording an entire project at a higher sample rate stresses out your computer more, reduces the number of audio channels you can stream, and won’t allow you to run as many plug-ins.
  • Oversampling requires more CPU power, so even if all your instruments are oversampling internally, you may not be able to use as many instances of them.
  • Although some instruments may perform 2x oversampling, that still might not be sufficient to eliminate aliasing on harmonically rich sources—so oversampling an oversampled instrument can still make a difference.

Furthermore, with plug-ins that oversample, the sound quality will be influenced by the quality of the sample-rate conversion algorithms. It’s not necessarily easy to perform high-quality sample-rate conversion: check out comparisons for various DAWs at http://src.infinitewave.ca (where, incidentally, Studio One rates as one of the best), and remember that the conversion algorithms for a plug-in might be more “relaxed” than what’s used in a DAW.

So what’s a musician to do? In next week’s Friday Tip of the Week, we’ll cover how to do upsampling in Studio One to reap the benefits of high-sample-rate recording at lower sample rates. Meanwhile, if you still need to be convinced recording at different sample rates makes a difference, check out this audio example of a synthesizer recorded in Studio One first at 44.1 kHz, and then at 96 kHz:

Friday Tip of the Week: A Sweeter, Beefier Ampire

A Sweeter, Beefier Ampire

Let’s transform Ampire’s Crunch American from a motor scooter into a Harley. Here’s our point of departure:


Insert the Multiband Dynamics before Ampire. The default patch is fine, but drag the High Mid and High gain and ratio settings down all the way. The goal here is to add a bit of compression to give more even distortion in the mids and lower mids but also, to get rid of high frequencies that, when distorted, create harsh harmonics.

 


After Ampire, insert the Pro EQ. The steep notch around 8 kHz gets rid of the whistling sound you’ll really notice in the before-and-after audio example, while the high-frequency shelf adds brightness to offset the reduced high frequencies going into Ampire. But this time, we’re increasing the “good,” post-distortion high frequencies instead of the nasty pre-distortion ones.


Those two processors alone make a big difference, but let’s face it—people don’t listen to an amp with their ear a couple inches from the speaker, but in a room. So, let’s create a room and give the sound a stereo image with the Open Air convolution reverb. I’ve loaded one of my custom, synthetic IR responses; these are my go-to impulses for pretty much everything I do involving convolution reverb, and may be available in the PreSonus shop someday.  Meanwhile, feel free to use your own favorite impulses.


Of course, you can take this concept a lot further with the Channel Editor if you want to tweak specific parameters to optimize the sound for your particular playing style, choice of pickups, pickup type, and the like…hmmm, seems like that might be a good topic for a future tip.

That’s it! Now all that’s left is to compare the before and after example below. Hopefully you’ll agree that the “after” is a lot more like a Harley than a motor scooter.


 

 

 

 

Friday Tip of the Week: Create Virtual Room Mics in Studio One

Create Virtual “Room Mics”

Room mics can add ambiance and enhance the stereo image, but with close-miking and direct injection recording, we lose that sense of space. The lack of room mics is particularly noticeable with an instrument recorded direct, when it’s mixed with miked acoustic or electric tracks; the direct track just won’t seem to mesh quite right with the other sounds.

Room mics add short, discrete echos. Splitting the audio into four Analog Delay processors as parallel inserts does a fine job of emulating room ambiance.

First, set up the Splitter for four splits, and choose Channel Split for the Split Mode.

Set the controls on the four delays identically except for the four time parameters; turn off Sync and choose 11, 13, 17, and 23 ms. These are prime numbers so they don’t create resonances with each other.

Now use the Channel Editor to create macro knobs for controlling the Mix, Feedback, and High Cut. Assign Knob 1 to the Mix controls on all four delays. Set this to go from minimum to maximum so that if you use this FX Chain as a bus effect, you can set Mix for maximum (no dry signal). Otherwise, when used as an Insert, you’ll likely keep the Mix at 50% or below.

The second knob controls Feedback level. I’ve limited the maximum amount to around 60% for each of the delays. Experiment with this knob depending on the audio source; more feedback gives more diffusion. With percussive instruments like drums, you’ll want more feedback than with sustained instruments.

The third knob controls the amount of High Cut for the delays. Set this so the High Cut doesn’t go much lower than 2.5 kHz.

To hear what this FX Chain can do, load a mono AudioLoop like Guitar > Pop > Dry > 01a Basement Jam E min. Set Mix and Feedback to around 40%, and High Cut to 7 kHz or so. You’ll hear the guitar playing in a room, with a lifelike stereo image.

And don’t forget to save the FX Chain—you’ll likely want to use it again!