KORG prologue with Dean Forrest

Cubase 13: New Sampler Track Features

By John Walden

Custom modulation envelopes can now be constructed more easily in the Sampler Track’s Pitch, Filter and Amp panels using the new shape options.

We throw some shapes with the powerful new sound‑design options in the Sampler Track.

The Sampler Track (which is a feature in the Pro, Artist and Elements editions of Cubase) is a simple but undeniably creative tool, so it’s great for experimenting with sample‑based sound design. It’s been with us a while, but in Cubase 13 Steinberg added some new features, including some intriguing spectral‑based modes for the sample playback engine. For this workshop, though, I thought I’d explore another of the new options: the shape‑based drawing tools now present in the Pitch, Filter and Amp modulation panels. These make it so much easier to design tempo‑based patterns, so that risers, pitch drops, patterned filter sweeps and rhythmic volume effects should all feel well and truly in sync with your project.

I want to keep things simple to follow, so with that in mind let’s explore some of the possibilities when using just a single sample as a starting point — in this case, I’ll be using a sustained single note from a synth pad. There are audio examples on the SOS website that you can audition (https://sosm.ag/cubase-0624) and, with the aid of a second sample (a drum loop) and a vocal hook, these include putting these Sampler Track‑designed sounds into the context of an initial sketch for a musical project.

Shaping Shifting

We’ve explored some of the Sampler Track’s established features in previous workshops so I’ll not retread that ground other than to remind you that the engine works on a single sample and provides various means of time‑stretching (designed for different use cases), loop slicing (for working with drum loops), sample looping (to create sustaining sounds), automatic mapping of the sample across a user‑defined key range, and pitch, filter and amplitude toolsets, each with envelope and LFO‑based modulation, amongst a bunch of other possibilities.

The new ‘shaper’ tools introduced in Cubase 13 mean more options when creating modulation envelopes. In the Mod panels of the Pitch, Filter and Amp windows, a new toolbar drop‑down provides access to a whole range of Factory curves, and you can save your own user curves too. Once a curve shape is selected, you can add it to the modulation envelope using either the Draw (pencil) or Paint (brush) tools. The former lets you draw a single instance of the selected curve, and you can drag as you draw to resize it. The latter lets you add multiple repeats of the selected curve and uses the panel’s Sync setting to set the beat division (length) of each repeat (so defining a rhythm for the modulation). By dragging up/down as you use the Paint tool you can control the amplitude of each repeat, and therefore the amount of modulation to be applied. And, of course, you can build custom curves by combining different shapes and/or different beat divisions.

A useful selection of factory curve shape elements are included, and you can also save your own user curve creations.

Two other options are worth noting. First, the F button enables/disables Fixed mode. When enabled, adding a new shape (with its curve nodes) replaces any curves/nodes already present in the targeted section of the overall modulation curve. When disabled, existing nodes are automatically shifted to the right along the timeline to make room when new curves/nodes are added. Fixed mode also influences what happens when you edit individual nodes and whether moving one node does (or does not) cause other nodes to the right to also change position.

Second, the Mode types in each Mod panel now include a Shaper option. You don’t have to use this mode in order to create modulation curves using the shape presets but it does allow you to create modulation curves that can cycle perfectly, as it ensures the start and end node values are linked.

One Sample, Many Sounds

In creating different sound types from a single sample, we can make use of the Sampler Track’s full feature set, but these new shape‑based modulation tools can certainly play a useful part. For my (totally arbitrary) mission of generating all my non‑drum sounds from a single source sample, there are some obvious contender sounds that I’ll need to generate. These include bass, lead, and pad/chord sounds, as well as options such as a risers or bass pitch drops that can be used to add some ear‑candy.

Let’s start with a simple case: a short (non‑sustaining) bass sound. The key settings are shown in the screenshot. In the Playback window, I’d engaged Audiowarp and selected one of the new Spectral modes, then tweaked the Formant and Key F controls to alter the tonality to taste, while in the Filter section I chose a low‑pass valve‑style filter. However, it’s the Amp and Filter modulation that then control the nature of the sound, and each is based on one of the new preset curve shapes. The Amp modulation provides a short, ramped attack, short sustain and then a rapid fade, while the filter uses an S‑shaped modulation curve to smoothly but rapidly open the filter as the sound plays to give it a little tonal movement. The Filter panel’s AMT control (far left) can be used to adjust the degree of this tonal shift, while the S‑curve nodes could be manually adjusted, along with the starting cutoff value, for further control. When played in the lower registers suitable for a bass line, the end result is a short, solid bass tone with a little sonic movement.

Even simple curve shapes become easier to generate, as seen here for our simple bass sound with short modulation curves in the Amp (on the left) and Filter (right) panels used to generate a short bass note with a little bit of tonal movement.

A similar filter modulation approach was used to create a lead sound but for this I used a high‑pass filter. In this case, the Mono and Legato options were enabled in the Sampler Track’s main Toolbar area so that the Pitch panel’s Glide control could then be used to provide a pitch slide between overlapping notes.

For this exercise, I created a couple of different pad sounds, both of which involved defining a loop region in the main waveform display (enabling Snap To Zero Crossings and adding a crossfade in/out to the loop region help with creating a smooth looping experience). For the first of these pads, I used a simple curved shape to add a very gentle filter sweep that created a subtle tonal shift as the sound sustained. However, for the second pad, I used the shaper‑based tools to create a custom Amp modulation (the main screenshot shows a similar custom modulation curve). When played, this creates a rhythmic effect that’s automatically sync’ed to tempo. And, with a further custom modulation curve added to the Filter, I was able to add a tonal element.

The various curve shape options were also useful in creating some ear candy elements. For example, for a bass ‘hit and drop’, I duplicated my bass Sampler Track and, as well as making tweaks to the filter settings, I added a short pitch modulation curve that smoothly dropped the pitch by an octave. By using the beat divisions in the display, you can adjust the curve shape as you draw it, so its length locks in sync with the project tempo. Similar curves in the Amp and Filter modulation panels can help emphasise the effect created.

I created a number of different risers. Obviously, the initial element required for this is a volume rise that lasts the required time. Again, the beat grid in the modulation panels lets you draw a suitable curve to exactly the length that’s required, and I created one‑beat, two‑bar and four‑bar versions using this approach. The rise effect can then be enhanced by using a similar modulation curve for the filter, and I used a band‑pass filter this time, to sweep upwards as the riser was playing. In a second version of the four‑bar riser, I added a further element by superimposing a small amount of LFO based modulation (based upon a pulse waveform set to 16ths) on top of the shaper‑based curve to give the riser an additional rhythmic pulse effect.

To keep this workshop easy to follow, I confined my experiments to a single source sample — this is a great way to learn what’s possible too!

Lock Then Load

To keep this workshop easy to follow, I confined my experiments to a single source sample — this is a great way to learn what’s possible too! — but in the real world you can use a selection of contrasting source samples to give you a broader range of tonal properties to play with. Also, note that if your modulation curve designs result in something you like, but you’d still like to experiment further, you can try using the same settings with a different underlying sample to produce a variation. To do this, just create a duplicate of your Sampler Track and enable the Lock button (the padlock icon) on the main Sampler Control toolbar strip for the duplicate. You can then load (or drag and drop) an alternative sample into the Sampler Control panel. This replaces the original sample in this instance but retains (locks) all the other settings — including your carefully constructed modulation curves!

As mentioned above, you can hear the results of my shaper‑based modulation experiments in the audio examples on the SOS website, both in isolation and in a short sketch. It’s great that Steinberg are continuing to nudge the Sampler Track feature set forward. No, it’s not a fully blown HALion or Kontakt‑style sampler, but it’s quick and easy to get to grips with, full of creative potential, and well worth digging into if you like to design your own sounds. 

Published June 2024

volca modular Patch of the Week 4: 4-bar Dynamic Synth bass

Cubase 13: Stock Vocal Compressors Compared

By John Walden

What’s your pick for routine vocal compression duties? Vox Comp, Black Valve or Compressor?

Cubase now boasts a number of decent stock compressor plug‑ins: Vox Comp, Black Valve and Compressor. Which should you choose for vocals?

When it comes to modern vocal production, compression is pretty much a ubiquitous part of the processing chain. In most cases, the primary aim will be to manipulate the dynamic range so that, when placed within the final mix, every part/phrase/word of the vocal comes across clearly. Having achieved this, a secondary consideration might be to impart some character or attitude to further enhance the vocal, either tonally, or in terms of finessing the attack or sustain elements of the individual words.

Cubase now offers a number of stock compressors, including two new choices added in v13: Vox Comp (for Pro and Artist users) and Black Valve (Pro only). So how do you decide which of these compressors to use on your vocals? To help you choose, I grabbed a suitable vocal track and put the two ‘new for v13’ offerings up against the long‑standing standard Compressor plug‑in, to see if I could identify the role that each might play in achieving these typical compression aims. You’ll also find, on the SOS website, some audio examples that demonstrate what I’m talking about: https://sosm.ag/cubase-0724.

One Knob Wonder

As can be seen in the main screenshot, Steinberg have taken the keep‑it‑simple approach to Vox Comp’s control set. The documentation suggests that it features a ‘highly adaptive’ algorithm that’s designed for processing vocals. Given that there is a specific option to engage a low‑latency ‘live’ mode, I assume that there’s usually a ‘look‑ahead’ stage of a few milliseconds that lets the plug‑in see what’s coming and adjust the processing to suit.

Once inserted, you simply dial in Vox Comp’s Threshold knob to achieve the required amount of compression (there is no user control over the attack/release or ratio settings; presumably, these are built into that adaptive algorithm). If you need to level‑match the processed signal to the original, you can adjust the Output knob. I’ve not used it here, but you can also use Vox Comp for parallel compression without setting up any routing in Cubase, thanks to the Dry/Wet knob.

As is easily demonstrated by the audio examples that accompany this workshop, the results can be very good indeed. My example vocal features noticeable level differences between the verse and chorus sections, and several words within the verse section that are quieter than some others. Using the settings shown within the first screenshot, Vox Comp’s one‑knob magic does a very good job of levelling out these quite marked differences (the loudness range across the example goes from 13.1 LUFS unprocessed to 7.8 LUFS after processing) while still maintaining a fairly natural sound.

These changes are reflected in the vocal’s waveform, as shown in the second screenshot. Sonically, it’s impressive stuff: a much more even vocal performance is achieved effortlessly, the words and individual syllables can all be heard clearly, and it still sounds fairly natural. If in doubt about your vocal compression choice, then, Vox Comp should be a very good place to start, regardless of your confidence level and/or experience with compressors featuring fuller control sets.

Changes to the vocal waveform as a result of the compression settings shown in the first screenshot. Each panel shows a short example from the same vocal performance — the top and middle panels are from the quieter verse section, while the bottom panel is from a louder chorus section. The top panel compares the unprocessed vocal (pink) with the Vox Comp version (pale blue). The middle panel shows the three different compressors: Vox Comp (blue), Black Valve (pale red) and Compressor (pale green). The bottom panel shows three different configurations for Compressor, using the settings shown in the other screenshot (in pale green), with the same settings but an 8:1 Ratio (in pink), and with slower attack and release settings (in pale blue).

Added Character

Users of Cubase Pro 13, though, have the option of using the Black Valve plug‑in, which emulates a classic tube‑style compressor. There are more knobs here (five, compared with Vox Comp’s three) and it’s not designed solely with vocals in mind, but it still offers ease of use and is certainly useful in this context. The plug‑in emulates two important characteristics found in Black Valve’s hardware inspirations: a relatively slow attack time and the option to add some preamp‑based saturation to the sound.

The Dry/Wet and Output level controls function as they do in Vox Comp and, again, there are no individual attack, release or ratio controls. But as on lots of hardware valve and optical compressors, the Peak Reduction knob acts as a sort of ‘macro’ control, influencing both threshold and ratio parameters, with higher values providing more overall compression. The Drive knob controls the signal level going into the modelled tube preamp and, pushed suitably hard, adds some saturation to the sound. The gain control then provides a post‑preamp gain‑staging option before your signal hits the compressor.

As shown in the screenshot, I set a pretty high Drive level to make the tube saturation sound a little more obvious in the audio examples (it’s particularly noticeable on the harder consonants), but note that if you wanted to exaggerate this further then you could apply a few extra dB of gain prior to the signal reaching the plug‑in — this would drive the tube stage even harder. I didn’t think it was appropriate for this particular vocal part, but the modelled tube does break up in a very pleasing fashion so it’s an option to bear in mind in a different context.

My Black Valve settings produced a slightly greater increase in the loudness of the quieter sections, but more noteworthy is the slightly higher levels of some note onsets.

The combination of saturation and compression resulted in a shift from the original 13.1 LUFS to 6.2 LUFS, and the much more even nature of the processed performance is clearly audible. Looking at the resulting waveform display, my Black Valve settings produced a slightly greater increase in the loudness of the quieter sections, but more noteworthy is the slightly higher levels of some note onsets. This is not a surprise given the somewhat slower attack time, and although it’s a small difference, to my ears it results in the starts of words having a slightly stronger emphasis. That might or might not be a desirable result, depending on the captured performance and the musical context, but it’s a useful option to know about. So if you’re looking for a compressor that might impart a little additional character to your sound, while still being very easy to use, Black Comp is a great candidate — I’d certainly check it out before choosing to splash the cash on a new third‑party compressor plug‑in.

Clinical Precision

Cubase’s standard compressor, called simply Compressor, is designed for those occasions on which you want precise control over all the elements of the compression process. It requires a little more user input (and probably experience), but the results are clean and controllable. As shown in the screenshots, for the main example used here I used a combination of a modest (4:1) ratio and fast attack and release times. I then just adjusted the threshold to taste until I felt the vocal was sitting at a more consistent level. Analysis subsequently indicated this resulted is a loudness range of 8.0 LUFS.

As shown in the short waveform, the most obvious feature of the Compressor processed version is that the loudest parts of the vocal are not compressed as much as with Vox Comp and Black Valve. This is most likely because the fixed 4:1 ratio is lower than the adaptive ones of Vox Comp and Black Valve are on average — a little experimentation revealed that an 8:1 ratio in Compressor produced something closer to the results of Vox Comp. Keeping the ratio and threshold fixed and varying Compressor’s attack and release produced exactly the sorts of changes you might expect in the waveform, making it possible to experiment with the balance between note attacks and note sustains with a level of control that’s not really possible with the simplified controls offered by Vox Comp and Black Valve.

Again, to my ears, the results in this example sound very clean. The overall level of compression achieved in this case was in the same ballpark but, of course, the user has much more precise control over how that’s achieved. If you’re more of a control freak, Compressor is a great choice — and note that if you want a less clinical‑sounding result, you could always use something else to add character: you might insert an instance of something like DaTube into your signal chain in front of Compressor, for example.

Is All Choice Good?

For me, the key lesson from this little experiment is that the differences between the three compressed versions of my example vocal are relatively small when compared with the differences between any one of them and the uncompressed version. Some of those differences between the compressed versions are where that secondary aim (discussed earlier) comes into play, whether it’s adding character or wanting more detailed control. If those things are important to you, Cubase clearly has stock plug‑ins that let you explore that.

But if you’re simply focused on that primary aim of getting your vocal broadly in the right dynamic range, then whichever compressor you go with, be it Vox Comp, Black Valve or Compressor, it can deliver the goods and the ‘right’ choice will probably come down to how quickly, or how easily, you can get to that good result. Yes, you can deep dive on your vocal compression if you wish, but you can still achieve impressive results by turning a single knob. How cool is that? 

Published July 2024

Patch of the Week 62: Korg volca keys Sequencer Experimentation

Cubase 13: Capturing Audio From Virtual Instruments

By John Walden

As shown here for Retrologue, Cubase Pro 13 users can now route the output of a VSTi to the input of an audio track.

New routing options in Cubase Pro 13 make it easy to archive virtual instrument parts in Cubase.

There are a number of pragmatic reasons why you might want to capture the output from the virtual instruments used in your Cubase projects to their own audio tracks. Here are two typical scenarios. First, you might need to move the project to a different computer (for example, to a collaborator or mix engineer) that doesn’t have the same VST instruments available. Second, virtual instruments can fall victim to the upgrade cycles of your DAW or OS, and preserving your virtual instrument parts as audio ensures you can revisit projects (for example, for remixes or edits) in the future.

Depending on which version of Cubase you’re using, there are various ways you can approach this audio capture, including the Render In Place function that’s available to both Pro and Artist users. For Pro users, Cubase 13 has a further option that’s both conceptually and practically very straightforward: you can select the audio output of any virtual instrument as the audio input of a separate audio track.

Alongside your audio interface’s audio input sources, you’ll see the output from any virtual instruments in the project.

Find The Source

Let’s start with a simple example: capturing the stereo output of a Retrologue synth to a stereo audio track. Having added an instance of Retrologue and an empty audio track to your project (as shown in the first screenshot), click the input selection option in the Routing section of the audio track’s Inspector panel. This displays a list of the available input sources. Alongside your audio interface’s audio input sources (that you configure in the Routing window), you’ll see the output from any virtual instruments in the project. By the way, you can do this in the MixConsole if you prefer.

As a brief aside, if you record‑enable both the instrument and audio tracks, you can capture both MIDI and audio at the same time. Enabling the Monitor (rather than Record) button on the instrument track, on the other hand, lets you record just the audio as you play the virtual instrument. This might or might not be useful in a real‑life context, but the option is there if you need it!

A couple of additional points are worth noting. First, if your audio track is mono, only mono virtual instrument outputs will be visible in the selection panel; if the audio track is stereo, both mono and stereo virtual instrument outputs will be visible and available for selection. Second, when you record the audio from a virtual instrument using this routing‑based approach, the captured audio will include the result of any active insert effects that are used on the virtual instrument channel (but not any send effects).

Make Mine A Multi

When activated, additional outputs appear in the Rack Instruments section of the input selection dialogue, using the track names assigned to them in the main MixConsole.There are several common situations where a single virtual instrument can generate multiple sounds, and it might be useful to send each one to its own audio track. Virtual drum instruments such as Steinberg’s Groove Agent or Toontrack’s EZdrummer are classic examples, but it’s also true of multitimbral sources such as HALion Sonic, NI’s Kontakt or EastWest’s Opus, or any instrument that might create a composite sound by blending two or more sound layers, where the instrument’s GUI lets you assign those layers to separate outputs.

To send each individual sound or layer to its own audio track, an additional configuration step is required — thankfully, it’s generally very straightforward. In the virtual instrument itself, you need to activate multiple outputs (use the drop‑down menu located top‑right of the standard Cubase plug‑in windows) and then assign each of the sounds/sound layers to a specific output. For your target audio tracks, these new additional instrument output options will show up at the bottom of the input selection panel (the additional inputs beyond the standard stereo output appear in their own sub‑section of Rack Instruments outputs) in both the Inspector and MixConsole.

Here’s a simple example using Groove Agent to illustrate the process, but the same principles will apply for any virtual instrument. As shown in the screenshot, having activated a number of additional outputs in Groove Agent, I gave the channels suitable names in the MixConsole (Kick, Snare, Hi‑hat...). Although those names don’t appear in Groove Agent’s own Mixer page (but do in the Activate Outputs pop‑up dialogue), it’s easy enough in GA to select the appropriate output channel for each of the drum sounds. Then, back in the Project window, I added a suitable number of audio tracks. I named these accordingly, and set the input source for each to the appropriate GA output.

You can activate additional outputs in multitimbral instruments such as Groove Agent, and with a little work in the GA Mixer, each sound can then be routed to its own audio track.

With the audio channels record‑enabled, when GA is triggered the audio for each drum sound is routed to the correct audio track and, when you are ready, can easily be recorded. What’s more, you can record all the record‑enabled tracks in a single pass. When you’re ready to archive your finished project, it’s super‑easy to ‘print’ all your virtual instrument sounds as audio as a single task.

The same principles can be applied to multiple sounds in a single instance of HALion Sonic 7, letting you capture each of HS7’s multitimbral sound elements to their own audio tracks. In HS7, the necessary internal routing is also configured in the instrument’s Mixer page. Things are also quite straightforward in many third‑party multitimbral instruments. For example, in EastWest’s Opus, simply activate additional outputs and you can then assign instruments in Opus’ Instrument Rack or Mix Console pages to these various outputs. Toontrack’s EZdrummer or Superior Drummer are equally easy to configure in their respective mixer pages.

If you want to handle multiple instruments in a single instance of NI’s popular Kontakt sampler, the process is a little more complex. This is purely because Kontakt offers an abundance of internal audio routing options — that necessitates some work in Kontakt’s Outputs panel (and quite possibly a dip into Kontakt’s PDF manual if you’re not familiar with the process!).

The Bus Route

Pro 13’s new routing options also apply to Group channels and FX channels, providing a very straightforward means for stem creation.If you routinely use groups to create buses for instruments in your mixes (drums, bass, guitars, keyboards, vocals and so forth), then the new audio routing options in Cubase Pro 13 open up a further option: easy stem generation. Stem generation can be useful in a number of contexts, whether it’s as an alternative form of project archival, to send to a collaborator, or in a music‑to‑picture context, to give a sound mix engineer more flexibility when adding music into the overall balance.

The same principle as outlined above for virtual instruments can also be used to capture audio from Cubase’s Group or FX tracks. For example, via the Routing Input source selection drop‑down for any audio track, you can select the output from any Group or FX Channel. As before, if you then record‑enable such an audio track, it will record the audio output from that Group or FX track. If you configure a suitable stereo audio track for each of the Group tracks (instrument bus) and FX tracks in your project, you can then generate a full set of stems (including effects stems) in a single pass.

Trickle Down Effect?

As mentioned earlier, Cubase offers a number of options for capturing the output of virtual instruments or Group tracks as audio, and the new audio routing possibilities outlined here provide a further method to achieve that. At present, it’s an option that only users of Cubase Pro 13 have available. It will be interesting to see if Steinberg do eventually trickle this possibility down to the Artist or Elements versions. Given just how simple it can be to configure even with most third‑party virtual instruments, I’m sure users of both versions would find the process an attractive prospect. 

Published August 2024

Patch of the Week 61: Benny Benassi Bass on Korg volca keys

Cubase 13: Optimising Loudness

By John Walden

Among other things, the Loudness Curve module’s settings can be customised to adjust the Duration (length of time displayed on the horizontal axis) and to show a target Reference Integrated loudness level.

Lower a mix’s integrated loudness with Squasher.

The ‘LUFS’ measurements that are used by streaming services to normalise loudness on playback, and which are now catered for by so many meters and limiters, aren’t just weighted (with an EQ curve that accounts for the sensitivity of human hearing to different frequencies) but they’re also ‘gated’: any portion of a signal where the momentary loudness falls below a defined level (‑10LU below the integrated loudness) won’t count towards the final integrated loudness.

When undertaking DIY mastering duties, the presence of this ‘floating gate’ raises two significant questions. First, for any given track, are there significant portions of time where the momentary loudness falls below the gate’s threshold? Second, if the answer is yes, might there an opportunity to raise some of those quieter sections just above the level of the gate, both to add energy or clarity, or to lower the integrated loudness — for example, to reduce the amount a track might be turned down — without negatively affecting the louder sections?

When I recently (SOS July 2024) reviewed iZotope’s RX 11, I was particularly impressed by its Loudness Optimize feature, which was conceived to address precisely those questions. As a Cubase user, that got me wondering... could I achieve something similar using only the tools that come bundled with Cubase?

Analyse This

For Cubase Pro users (v11 and above), the Loudness Curve module of the SuperVision analyser plug‑in can help with the first question. Inserted at a suitable position in your project, this provides a real‑time visual display of your audio’s loudness. The Loudness Curve module settings (first screenshot) allow you to customise the display. You can, therefore, set the time resolution of the display to suit your needs (I used a 45‑second window here) and choose which loudness parameters are shown (in this example, just the blue momentary and green integrated loudness curves are displayed).

SuperVision’s Loudness Curve module can provide very useful visual feedback on the loudness and dynamics of your audio.

As shown in the left‑hand panel of the second set of screens, these curves update in real time as the track is played. After an initial playback, the integrated curve will stabilise and give a reasonable sense of the overall loudness. You can then dip back into the Settings panel and set the Reference Integrated value to 10LU below the observed integrated loudness value. In this example, the integrated loudness is about ‑18LUFS, so I set the Reference Integrated value (it appears as a horizontal pale blue line) to ‑28LUFS. This isn’t what this parameter is really intended for (I’ll come back to that later) but it does provide you with a simple visual guide to spot sections of the track where the dynamic range may be great enough for the momentary loudness to fall more than 10LU below the integrated loudness curve.

Incidentally, for any Artist and Elements users, who won’t have SuperVision, APU Software’s Loudness Meter offers a free‑to‑download alternative to SuperVision. It provides some pretty sophisticated features that include a real‑time momentary loudness curve similar to that found in SuperVision’s Loudness Curve module.

The Squasher plug‑in, available in Cubase Pro, Artist and Elements, can apply upward compression.

Going Up

This visual feedback is a useful check to know if the gate might influence how streaming loudness normalisation is applied to a track, but it’s still your decision to do something about it or not. If you do, the focus needs to be on raising the level of the quieter sections of the track while leaving the louder parts unaffected, and there are a number of approaches that might be considered. One, for example, would be parallel compression. But the Squasher plug‑in, available in Cubase Pro, Artist and Elements, can apply upward compression, and this is super‑easy to experiment with.

The final screenshot shows some Squasher settings that might serve as a good starting point. For simplicity, the plug‑in has been set to operate in single‑band mode. Within that band, the Down knob has been set to 0% and the downward compression Threshold (the small downward arrowhead) has been set to 0dB. These settings should ensure that no conventional (downward) compression is applied, which would otherwise affect the louder sections; these should pass through the plug‑in with their dynamics intact.

Squasher is capable of upward compression, which allows you to manipulate the loudness dynamics of your track in some interesting ways.

In contrast, the Up knob, which controls the ratio of any upwards compression, has been set to 100%. With this set, you can then gradually raise the Up Threshold value (the small upward facing arrowhead) until you see the upward compression kick in, as indicated in real time by the pale bar at the base of the Threshold graphic.

At this stage, Squasher’s main Input and Output controls have been left at zero (no overall gain adjustment applied; I’ll come back to the Output control in a moment). In addition, the Drive control (in the Parameters panel) has been left at zero so no saturation is added, while Gate is set at its lowest available value (‑60dB), effectively bypassing Squasher’s internal gate. In this case, the Attack and Release are both on fast settings but, as with conventional downward compression, you can experiment with adjusting these based on the tempo of the track, letting your ears judge what produces the most transparent results.

Spot The Difference

From this starting point, you can adjust the two main Mix controls, the Up Ratio and the Up Threshold values to vary just how much (or how little) upward compression is applied. Again, your ears are the key here, both in terms of just how much louder the quieter sections of your track become but also in terms of how transparently any loudness change can be achieved.

However, SuperVision can also be a guide: place a second instance of the plug‑in immediately after Squasher and, if you use the same display settings as in the pre‑Squasher instance, you should easily be able to see the difference in the loudness curves pre and post upward compression.

The right‑hand side of the first screen shows an example using exactly this configuration. While the momentary loudness curves are the same in the louder parts of the audio, Squasher’s upward compression has bought up the loudness during the quieter sections. While I’ve perhaps overcooked the processing here for illustrative purposes (and could lower some combination of the Mix, Up Threshold and Up Ratio controls for a more subtle effect), Squasher’s potential to manipulate the dynamics of your audio, increasing the loudness of quieter sections while leaving louder sections untouched, is easy to see.

Target Practice?

I mentioned above that I’d return to the Loudness Module’s Reference Integrated setting and Squasher’s Output control. Almost all streaming platforms have now published the integrated loudness reference levels used by their normalisation algorithms. They’re not all the same, but they are similar and, as an example, Spotify uses ‑14LUFS‑I as its default — on playback, songs will be turned either up (as much as peak levels allow; there’s no limiting applied in the default mode), or down to ensure they’re played back at ‑14LUFS‑I.

This is an automatic process, so it’s not critical that the integrated loudness of your mastered track is ‑14LUFS — the whole point is that loudness normalisation plays it back at that level for you. But perhaps you’re working for a client who has specified delivery of files at a certain LUFS‑I. Whatever the reason, if you feel the urge to nudge your track towards a target level, you can.

First, use the Reference Integrated setting for its intended purpose, and set it to the required target value (‑14LUFS for Spotify, for example). Then you can simply use the visual reference this provides and adjust Squasher’s Output knob up or down until your track’s integrated loudness curve sits on or around the Reference Integrated target line. For the example used here, it would require a setting of about +4dB. Providing this doesn’t also raise any peaks close to clipping, on playback your final mastered track might give Spotify’s loudness normalisation algorithm very little to do.

And Finally...

There are a few caveats to keep in mind. First, it’s not all about the numbers: in raising the loudness of quieter sections of your audio, you’ll be changing the track’s overall dynamics. It’s always worth making loudness‑matched comparisons, so that you can judge whether the processed version sounds better or worse than the original!

...in raising the loudness of quieter sections of your audio, you’ll be changing the track’s overall dynamics. It’s always worth making loudness‑matched comparisons, so that you can judge whether the processed version sounds better or worse than the original!

Second, you’ll probably also be raising the level of low‑level noise in the source material. This may not be an issue if the original recordings were of decent quality, but do make the effort to listen out for unwanted side‑effects. If noise becomes an issue, you might be able to counter that by de‑noising your sound sources, whether with a spectral noise removal tool or strategic use of noise gates or expanders.

Finally, not all mixes and mastered tracks can benefit from this sort of loudness dynamics adjustment. If your initial SuperVision visuals don’t show the momentary loudness dropping 10LU below the integrated loudness curve for significant portions of time, further loudness manipulation might well be unnecessary or unfruitful. If it’s not broke, don’t fix it! 

Published September 2024

Patch of the Week 60: Basshunter Lead on Korg volca keys

Cubase 13: HALion Sonic’s Spectral Synth

 By John Walden

X‑Stream’s spectral engine may offer an unconventional control set but it is very easy to experiment with. The settings shown here reflect the finished DIY example discussed within the main text.

Create a complete musical cue from just a single sample with HALion Sonic’s spectral synth engine.

Steinberg recently improved the spectral synth engine in their HALion 7 soft sampler, and all Cubase users can get a taste of it through X‑Stream, a new, free‑to‑download (using Steinberg’s Download Assistant) HALion Sonic instrument. While X‑Stream is monophonic only, it is bound to appeal to those with a liking for ambient soundscapes and textures. It offers plenty of DIY sound‑design possibilities, including the ability to drop your own samples into the engine. To explore just what this makes possible, I decided to don my experimental media composer hat, and see how close X‑Stream could get me to composing a complete musical cue using just a single sample.

Blank & Drop

Once you’ve managed to stop playing with X‑Stream’s excellent collection of presets (it might take a while), it’s a good idea to wipe the slate clean by loading in the Init X‑Stream Layer, which is easy to find if you pick Layers as the Preset Type in HS’s MediaBay panel. This includes a default sample but, usefully, it resets the synth engine parameters to a generic starting point. With the engine thus ‘blanked’, you can drag and drop (from the Project window or your file browser) your sample of choice directly into X‑Stream’s spectral display panel. Almost any sample can be a contender, but I deliberately started with something very simple: a single note played on a synth. I’ll come back to some more colourful sample choices later. By the way, you can hear some audio examples on the SOS website, at https://sosm.ag/cubase-1024.

Need For Speed

It’s worth noting that if you click on the Spectral panel header, you toggle between compact and expanded control displays. The latter provides access to the full control set, including the full pitch, spectral filter, standard filter and amplitude envelope controls. Other panels can also be expanded into focus in the same way. I couldn’t hope to cover all of X‑Stream’s many controls here, though, so I’ll focus on some highlights that will provide a solid base for experimentation.

In its expanded view, the Spectral panel offers five sub‑pages accessed via the top‑left buttons. These include the amp envelope shown here, with suitably long attack and decay times.

The Speed control sets the speed at which the spectral engine scans across the underlying sample to create sound (100 percent is the same playback speed as the original). Changing the speed doesn’t change the pitch, but X‑Stream’s underlying time‑ and pitch‑stretching algorithms are very impressive, and it’s remarkable just how extreme you can get with the Speed control (0‑800 range) while still hearing totally usable sounds. I’m attempting to create a textural sound here, and a slower Speed setting (50 percent) seems a suitable choice when seeking longer notes.

For sustained sounds, you can define a loop region within the spectral display (just drag the left/right markers), and here I’ve set the Loop Mode to Alternate. This means the playback head moves forwards until it reaches the right loop marker and then reverses direction back to the left marker. This loop cycle is repeated as long as the note is held. As my sample contains a naturally fading sound, the slow cycling back and forth through a portion of it creates a smooth rise and fall in level that works pretty well for a sustained textural soundscape.

The Spread option, once activated, shifts the playback position between the left/right channels of your sample (it works well with mono or multi‑channel samples too), and you see two playhead cursors offset from one another when you play a note. This delivers a nice spatial enhancement even when relatively low values are used.

A Blur Of Sound

Blur Time and Blur Depth create something similar to a modulation‑based effect, such as chorus. Increasing the Blur Time gradually adds a richer quality to the sound that is very much dependent on the nature of the original sample. Blur Depth can exaggerate this, with low values remaining fairly musical and higher values moving you towards something a little more unsettling.

According to X‑Stream’s PDF manual, the Purity and Inharmonicity parameters adjust the ‘spectral purity’ of the sound. No, I’m not sure I know what that actually means either! But a quick sweep of the knobs soon gives you a sense of what’s possible. Purity seems to change the tonal character of the sound, with negative values seeming brighter and more harmonically complex, while strongly positive values seem both darker and less complex. Inharmonicity also changes the harmonic complexity, and extreme positive or negative values induce something similar to a detuning effect.

While X‑Stream is monophonic, the Stack feature provides a way to add a chord‑like element to your sound.

Well Stacked

While X‑Stream is monophonic, the Stack feature (on the right of the Spectral panel) provides a way to add a chord‑like element to your sound. Once activated, you can create a total of five layers from your core sound, with control over the relative level and pitch of each layer. Layer pitch can be specified in semitone steps within a ±36 range. You can, therefore, build a chord by specifying the required note intervals in different layers and then triggering that chord from a single note on your MIDI keyboard. And by automating either the pitch or the level of your layers you can change the voicing of the created chord.

X‑Stream offers a great selection of modulation options including the excellent step modulator, which I used to create a pulsing effect within the example sound design.I’ve done that in my examples — see the main screenshot, in which the key settings for my final patch are shown — using layers 4 and 5. Relative to the root note (pitch = 0), these layers have pitches set to +16 (a major third, one octave above the root) and +15 (a minor third, one octave above the root). By automating the level values for these two layers, you can move (or, interestingly, morph) between a more major sounding harmony to a more minor one — I did this in my example by assigning them both to the mod wheel, mapped in opposite directions, so that as one increases, the other decreases.

Mojo Filters & Mod Magic

I’ve focused on some of the more unconventional parameter options offered by X‑Stream’s spectral synth engine, but it does, of course, offer a suite of more conventional sound‑shaping tools. These include a well‑specified filter (indeed, two filters, as there’s also a spectral filter with its own envelope and presets), a multiband EQ module, and both reverb and delay modules. All can be put to good use in your DIY sound design. There’s also an amp module, and if it’s sustained soundscape/ambient sounds you are after then this is most certainly worth a visit, even if only to add some suitably slow attack and decay elements to the amplitude envelope.

As noted above, X‑Stream’s parameters can all be automated in your DAW and/or assigned for hands‑on control using a connected hardware control surface. The engine also includes two standard LFOs, an X/Y LFO and a step modulator — these can all be used to automate parameters. Right‑clicking on any X‑Stream parameter pops open a suitable dialogue box to make these assignments.

Alternatively, you can open the Matrix panel to make and/or adjust the automation settings. This provides up to 24 slots, with a huge selection of source and destination options. I used both LFOs to provide long tempo‑based sweeps on the resonance and cutoff of the main filter. The step modulator was also used to modulate the drive control within the filter. The step modulation pattern I used creates a pulse‑like effect within the sound, and I assigned a Quick Control to adjust the level of the step modulator, allowing me to blend the pulse effect in and out as required.

X‑Stream’s modulation matrix provides plenty of options for both modulation sources and destinations.

Finally for this example, as well as the stack layer automation that I mentioned earlier, I assigned the level controls of layers 2 and 3 to Cubase’s Quick Controls, allowing me to adjust the ‑24 (two octaves below) and +19 (an octave plus a fifth above) layers’ levels, to blend in these layers to change the tonal character of the overall sound in real time.

Do check out the audio examples, which illustrate just how I developed this sound from the original sample into a fully formed X‑Stream preset. I found it an interesting enough result that it prompted me to try writing a full soundscape cue (albeit a short one) using just this one sound — a totally arbitrary target, but you can hear the result in one of the examples!

One More Drop

The whole sound‑design process used here took me rather longer to explain in words than it did to actually perform. X‑Stream’s unconventional control set is, in practice, very easy to experiment with, not least because you can drag and drop your own sample into the engine. To encourage your own explorations, the last audio example includes some randomly selected samples dropped into X‑Stream and quickly tweaked based on a similar basic approach as described above. The results can be somewhat weird and unpredictable, but if you just keep dropping samples in then eventually something magical will happen... and off you’ll go exploring a new musical idea.

Published October 2024