To talk about SPL (Sound Performance Lab) is to talk about professional audio products that have often broken new ground in analog signal processing. Transient Designer, Charisma and the Vitalizer series are just some of the names that carry the SPL badge, and users such as Bob Ludwig, Joe Chicarelli and Ed Cherney vouch for the quality of this company’s offerings.
SPL’s product catalog has room for mastering consoles, equalizers, dynamics processors, mic preamps, analog summing mixers and monitor controllers (the MTC 2381 was reviewed in issue 88 of ISP).
Combining much of SPL’s experience in two rack units, the Channel One is a mic preamp that incorporates a De-Esser, Compressor/Limiter, noise gate, EQ section and headphone amp.
Channel strip
This configuration is common in so-called channel strips, generally conceived for vocals but which often see any type of signal pass through them.
Proof of this is that, alongside the XLR mic input, the Channel One incorporates a TRS input for line and another for instrument (balanced). The mic preamp has a knob that controls the 60 dB of gain available; the line and instrument inputs share another rotary control to govern their respective 40 dB.
The input section is rounded out by buttons to activate 48V phantom power, a mic/line selector, a phase inverter and a high-pass filter (set at 50 Hz, 12 dB per octave).
The grille that lets you see the preamp’s 12AX7-LPS tube leads into the next section of the Channel One: the De-Esser.
De-Esser
The De-Esser doesn’t work in the usual way, based on a compressor “tuned” to the sibilance frequencies. Instead, SPL introduces its Auto-Dynamic De-Esser, whose circuitry listens to the sibilant frequencies and compares their level to that of the rest of the signal. If it is higher, a copy of the sibilant frequencies is phase-inverted and mixed with the original signal, to cancel only those frequencies.
This has two advantages. First, only a narrow portion of the spectrum is affected, reducing the side effects of a traditional de-esser. Second, operation is really simple, since the “auto-threshold” function detects the overall signal level at all times and applies the selected reduction regardless of the level the signal is at.
At the De-Esser’s output there is an insert point, where the signal can be picked off and sent to other devices. Output and input are via two balanced TRS jacks on the rear panel. This opens the possibility of using the Preamp/De-Esser section and the Compressor/Gate/EQ section independently, as if they were two separate units.
To make it easier to integrate the unit into a patch, the insert point can be enabled or disabled with a button located in the De-esser section.
Compressor/Limiter
Following the philosophy of intelligent circuitry, this new section has just two controls: amount of reduction and output gain (Make-Up). The remaining parameters usual on a compressor (attack time, hold, release, ratio...) are selected automatically depending on the signal.
Thus, explosive sounds are processed with faster times than other, longer-lasting sounds, whose time constants are lengthened to avoid distortion.
As for ratio, the compressor is the soft-knee type, and the input/output relationship also varies with signal level, applying low ratios to low-level signals and raising it to a maximum of 1:25 for the highest levels.
To carry out the compression, a dual topology called Double-VCA-Drive is used, employing two voltage-controlled amplifiers (type 2181), dedicated respectively to the positive and negative halves of the signal.
This results in lower distortion, since each VCA can work focused on a narrower range, and there is no risk of them going into saturation.
A button lets you switch the compressor to limiter, in which case the gain reduction knob regulates the limiter’s threshold. The manufacturer states that it is not an absolute limiter, so it is advisable to leave some headroom (2-4 dB) to accommodate a few fast peaks that can get through the processor.
Figure 1
The compressor and limiter curves are shown in Figure 1, with graph A being the reference, B the compressor and C the limiter.
This section is rounded out by a noise gate whose time constants, as it could hardly be otherwise on the Channel One, are also automatic. The knob labeled Noise Gate controls the threshold (between -100 dB and +18 dB), and there is a 6 dB offset between the opening and closing thresholds to keep the gate from triggering falsely.
EQ
The equalization section has four components. The first two are semi-parametric equalizers, whose ranges run from 33 Hz to 720 Hz (called the Low Band) and from 740 Hz to 15 kHz (this is the Mid-Hi Band), offering adjustments of +/-14 dB and +/-12 dB respectively.
The so-called Air Band lets you boost or cut frequencies above 17.5 kHz by up to 10 dB.
Finally, there is a knob labeled Distortion, which drives a FET transistor. The idea is to provide a certain warm distortion, which can range from the most subtle to clearly distinct harmonics. Although it is the last knob in the EQ section, this stage is placed right before it, so the EQ can act on the result of the distortion if necessary.
Although the order of the sections is as indicated so far, the switch labeled Pre Comp. lets you place the EQ section before the compressor, something useful if you need to correct the tonal balance before dynamics processing, to avoid an exaggerated compressor response to certain frequencies (or to provoke it!).
Output, meters and headphone amp
The output section lets you set the signal level at the right point before it leaves the unit, with variations between -20 and +6 dB, and it also includes a Mute switch.
The monitoring section has LEDs indicating De-Esser operation, clip, signal and initial warm-up, and two ten-segment meters for compressor reduction and output level (PPM).
As for the headphone module, it provides the ability to create a zero-latency headphone mix for the musician if you are working with a DAW.
For this it has two connectors (L and R) on the back where you feed in the reference mix. This signal is combined with the one coming from the musician, through the preamp, via the Playback control, and the headphone output level is regulated with a knob independent of the main output level.
The unit’s analog output can be taken on either balanced TRS or XLR, and there is a switch on the back to lift the ground.
Other options
If desired, Lundahl transformers can be installed at the input, which increases the input gain by +14 dB. Many users associate the warmth of vintage gear with their transformers, and SPL offers them as an option on several of its products.
It is also possible to install an AD converter, which provides a digital stereo output over S/PDIF and Toslink optical cable. Since the Channel One offers a single channel, the free one is accessible for conversion through an additional connector on the rear panel.
In action
The Channel One tests were carried out together with those of the Red 5 Audio RV15 and RV10 mics, reviewed in this same issue of ISP. Multiple vocal recordings were made, and to illustrate how the gear behaves, seven audio samples have been prepared.
All are aligned to zero, so they can be imported into any DAW without trouble. The first is the backing track (Libens_Base_Musical.mp3), and it is provided separately so you have the flexibility to lower its level or mute it, and better appreciate the details of the vocal samples.
As for the latter, the first two (Interfaz_RV15.mp3 and Interfaz_RV10.mp3) show the response of the two mics through the preamps of a well-known audio interface.
The next two (CH1_RV15.mp3 and CH1_RV10.mp3) were made through the Channel One, but using only the preamp section. In them you can hear the body the unit imparts, giving the voice a special color.
Finally, the files Proc_CH1_RV15.mp3 and Proc_CH1_Rv10.mp3 show the result after using the remaining sections of the vocal channel.
The goal was to achieve a very intimate, upfront voice that accentuated the intent of the lyrics. To do so, a light de-esser was applied, which removed a few annoying “s” sounds without being very obvious.
Next, in the compression section, reductions of 14-16 dB were used to bring the smallest details to the front and create a sense of closeness.
The EQ section was used differently on each mic to try to achieve a similar tone. On the RV15 a good dose of “air” was applied, and the low parametric was used to slightly reduce the proximity effect (increased by the compressor) and the high one to accentuate around 5 kHz.
On the RV10 the amount of air was somewhat lower, since the result was slightly more aggressive. The low parametric was used to add a little body, and the high one was used to accentuate the same 5 kHz region, again more gently.
Note: The vocal samples carry no processing of any kind (EQ, compressor, limiter, etc.), except as noted for the last two. This was done so that the differences among them are due solely to the gear used.
Conclusions
As you can hear in the audio samples, the Channel One gives the voice a special touch. In the last two examples the processing is slightly exaggerated, but it was chosen that way to show what this channel strip is capable of when pushed a little.
The preamp has a warm, full-bodied response, and the other modules respond really well, especially considering that most of their parameters adjust automatically.
This makes the unit certainly simple to operate and allows it to deliver productive results in very little time, one of SPL’s original goals.
Some users may miss a bit more control, especially in the compressor section, since access to more parameters would let you vary the processor’s character.
However, it should be noted that the original goal has been more than achieved, since the compressor works exceptionally well with the program material it was designed for (voice). Only in situations with very high gain reduction do effects begin to appear where you would wish you could reach for another parameter.
As for the design, it must be said it is really good, both physically and functionally, with a large number of details that show off the manufacturer’s know-how.
The knobs have a smooth, precise travel, with markings where they are needed, such as at the 0 dB of EQ cut/boost or at the various frequencies, which make it easy to quickly recall different settings.
Each section can be activated with a button that lights up to indicate its status, the insert point can be switched on or off, there is a headphone amp, the possibility of fitting transformers and an AD converter...
In short, a truly complete vocal channel, very well built and easy to use, with a great sound that will surely help take your vocal productions to another level.
