TEST: SPL Phonitor

 When I first encountered the product description for the SPL Phonitor, I was immediately struck by a mixture of great hope and perhaps even greater skepticism. Why am I making such a fuss about this product? Well, if SPL manages to achieve their stated goal with the Phonitor, we'd be dealing with a revolutionary invention that a legion of producers, sound engineers, and audio technicians have been waiting for.

Enough with the cliffhanger - what exactly is SPL's goal? Anyone who has ever had the pleasure (and responsibility) of occupying a professional producer's chair knows the problem all too well. While you can confidently determine the frequency relationships and spatial imaging of a signal on your own studio monitors, you face significant challenges when working with unfamiliar monitoring setups in other studios.

The stereo image of an unfamiliar setup is positioned completely differently, frequencies resonate differently or are over/under-emphasized. The room acoustics of the control room do their part to turn the workspace into a natural adversary. Your personal sense of sound goes out the window, and creating the ultimate mix becomes an audio Russian roulette with a maximum 70% hit rate. This is why many producers, myself included, always bring their own reference monitors (not to be confused with "particularly good-sounding" monitors) when working on productions in other studios.

Like many colleagues, I would have loved to create a mix on a proper set of headphones, especially for late-night work at home. However, the sometimes unbalanced sound image and, above all, the complete helplessness regarding spatial localization due to the 180-degree stereo width ("through the head") of the signal has always been a major obstacle.

This is where SPL comes in, promising a device that - to quote - "generates music as it was produced, namely for playback over speakers." I must say, I'm quite intrigued!

Construction:

The Phonitor comes in a half-width 19" rack format (9.5 inches) with a height of 2U. The overall build quality is excellent, with SPL fortunately still adhering to the once highly esteemed "Made in Germany" trademark across the board.

A notable feature is that the Phonitor, like all products in SPL's Mastering Series, is based on 120V technology. A total of nine SUPRA-OPs do their duty in the Phonitor, offering a signal-to-noise ratio of 116 dB and an overload tolerance of 34 dB, resulting in a respectable dynamic range of 150 dB.

At first glance, it's clear that this device doesn't fall into the typical "headphone amplifier" category. Too many switches and controls allow for sound shaping possibilities, in stark contrast to the usually spartanly equipped "normal" headphone amplifiers.

On the front panel, we're greeted by an oversized volume control for the headphones. A Crossfeed control, adjustable in six steps, allows for level-dependent, frequency-dependent simulation of crosstalk between both channels. Changing this value is comparable to altering the room size while maintaining the same speaker setup.

The Speaker Angle control adjusts the frequency-dependent simulation of the stereo base width in terms of timing. Here, an angle between 25 and 75 degrees can be set. The Center Level control adjusts the intensity of the center signals in relation to the Crossfeed and Speaker Angle settings of the stereo signal.

In normal headphone listening experiences, the center signal (phantom center) is usually quieter than the side signals, which appear louder due to the super-stereo effect. The narrowing of the base width through Crossfeed and Speaker Angle can lead to an intensification of the phantom center, which can be very finely reduced using this control.

Two toggle switches allow you to deactivate the Crossfeed/Speaker Angle combination and Center Level, providing a good A/B comparison to a regular headphone amp.

Two large VU meters serve as the visual centerpiece of the Phonitor. The display range extends from -20 dB to +5 dB. If needed, the sensitivity can be reduced by 6 dB, extending the display range to +11 dB. In addition to their needles, the "portholes" also feature a signal LED that indicates the presence of a signal above -22 dB, and an overload LED that activates at +21 dB.

The VU meters can switch between two modes: VU and PPM. In VU mode, the needles show an average level and operate with a rise time of about 300 ms. In PPM (Peak Program Meter) mode, this value is reduced to about 2 ms.

The operation mode of the VU meters and meter calibration are also set using two small switches on the front panel. A DIM switch reduces the monitoring volume by 20 dB. This function is typically used in practice to switch between two defined volumes during mixing.

A Solo switch allows you to listen to the right or left channel individually in Solo-In-Place mode. If you want to listen to the respective channel in Solo-To-Center mode, you must also activate the corresponding Mono switches.

A phase switch allows you to activate the crucial phase inversion, which can flip the phase of either the left or right channel. When the Mono switch is activated simultaneously, a difference can be formed between the two audio channels. What remains is what's present only in the right or left of the stereo image.

On the rear panel, the Phonitor features XLR Left/Right inputs and outputs. The input signal passes unchanged to the output sockets, so no monitor output is lost. Additionally, there's a ground lift switch, a voltage selector switch, and an IEC power connector.

In Practice:

After powering on the Phonitor, the VU meters light up, indicating its operational state. Following the manual, I carefully work my way through the parameters, constantly comparing the headphone signal with the "real" monitors.

You wouldn't believe it, but with meticulous adjustment of the parameters, depending on your own monitoring setup, you can actually generate a true-to-life representation of your speakers. I'm completely thrilled! Admittedly, you should set aside several hours and initially focus on understanding the effects of the controls with full concentration. After all, the goal is to "replicate" your own monitoring environment, which can be a real challenge for the uninitiated.

Once you've grasped the workings of the parameters, you can literally build your own environment. All switches provide practical functionality and offer manifold possibilities for tonal control. Only the DIM switch, with its -20 dB reduction, feels a bit too strong for my personal taste, but this is subjective and doesn't detract from the overall impression.

Even with critical listening, I couldn't detect any tonal alterations or losses despite SPL's massive intervention in the phase landscape of the signal.

Expanding on the Phonitor's Capabilities:

While the Phonitor excels at recreating a speaker-like experience through headphones, it's worth delving deeper into its potential applications and how it fits into various workflow scenarios.

For mixing engineers, the Phonitor opens up new possibilities for creating balanced mixes even in less-than-ideal acoustic environments. By simulating a well-tuned control room, it allows for critical decisions on stereo placement, depth, and overall balance without being at the mercy of room acoustics. This can be particularly valuable for those working in home studios or when traveling.

Mastering engineers might find the Phonitor useful as a secondary reference point. While it's not meant to replace high-end speakers in a treated room, it can provide valuable insights into how a mix might translate to headphone listeners. The ability to switch between different speaker angle simulations could help in assessing how a master might sound on various playback systems.

For recording engineers working on location, the Phonitor could serve as a reliable reference when setting up temporary control rooms. Its ability to simulate different speaker setups could help in achieving consistent results across various recording environments.

It's important to note that while the Phonitor excels at creating a speaker-like experience, it doesn't address the frequency response characteristics of the headphones themselves. Users will still need to be intimately familiar with their headphones' sonic signature to make accurate judgments. Some engineers might choose to pair the Phonitor with headphone correction software for a more complete solution.

The Phonitor's VU meters deserve further exploration. In addition to their aesthetic appeal, they serve a practical purpose in gain staging and maintaining consistent levels. The ability to switch between VU and PPM modes allows engineers to monitor both average and peak levels, which can be crucial in maintaining appropriate headroom throughout a mix.

The phase inversion and mono summing capabilities of the Phonitor also warrant attention. These features, while simple in concept, can be powerful diagnostic tools. The ability to flip the phase of one channel and sum to mono can quickly reveal phase issues in a mix, helping to identify problematic frequency cancellations or enhancements.

For those working with binaural or immersive audio content, the Phonitor's crossfeed and speaker angle controls could prove invaluable. By adjusting these parameters, engineers can fine-tune the perceived spaciousness of binaural recordings or assess how well spatial audio content might translate to standard stereo playback.

It's worth considering how the Phonitor fits into the broader ecosystem of headphone-based mixing solutions. While some software plugins attempt to simulate speaker playback through headphones, the Phonitor's hardware approach offers several advantages. There's no latency to contend with, no CPU overhead, and the tactile control over parameters can lead to a more intuitive workflow.

However, the Phonitor is not without its limitations. Its fixed crossfeed curve might not be ideal for all types of program material or headphones. Some users might prefer systems that allow for more customization of the HRTF (Head-Related Transfer Function) used in the crossfeed simulation.

The lack of a built-in DAC might be seen as a drawback by some potential users, especially given the Phonitor's premium positioning. While this allows for greater flexibility in choosing a DAC, it also means an additional component in the signal chain.

For professionals considering the Phonitor, it's important to view it as part of a larger monitoring strategy rather than a complete solution in itself. It complements rather than replaces traditional speakers, and its true value becomes apparent when used in conjunction with well-understood headphones and in comparison with other monitoring options.

The Phonitor's potential impact on workflow shouldn't be underestimated. By providing a consistent, speaker-like reference point, it could allow for more efficient use of time when working away from a main studio. This could be particularly valuable for producers and engineers who split their time between multiple workspaces or who often need to make critical decisions outside of their primary environment.

In educational settings, the Phonitor could serve as a valuable tool for teaching critical listening skills. Its ability to switch between standard headphone playback and simulated speaker playback could help students understand the differences between these two monitoring paradigms and develop a more nuanced understanding of stereo imaging and mix translation.

Conclusion:

Well, I'll be damned! I wouldn't have thought it possible, but I'm actually hearing the stereo image of my monitors through my headphones! Finally, a way to precisely position instruments in the stereo field, even when only headphones are available. SPL has truly created an outstanding product here!

This opens up entirely new avenues for production. For example, you could set up sounds and frequencies in the studio during the day, take the production home on a hard drive, and finish the mix in terms of soundstage, etc., in your home office in the evening if necessary. Words like flexibility and independence come to mind. It's a real boon for anyone who still makes the effort to deliver high-quality productions in the age of "no-budget-over-virtual-knock-me-dead-simulator-cobbled-together-and-mixed-in-the-bedroom" subpar productions.

However, if you now believe you can set up EVERYTHING using the Phonitor, please keep in mind that to work exclusively with the Phonitor, you must know your headphones PERFECTLY in terms of frequency response, etc. The Phonitor only handles the stereo image; you're still listening through headphones, which have their own sound!

By the way, the extent to which music fans will be able to aurally grasp and appreciate this extra effort remains to be seen. I recently saw kids strolling through town with mono headphones that had the frequency response of a megaphone...

In the broader context of audio technology, the Phonitor represents an interesting bridge between traditional speaker-based monitoring and the increasing prevalence of headphone use in professional audio. As more content is consumed via headphones, tools like the Phonitor may become increasingly relevant in ensuring that mixes translate well across various playback systems.

It's also worth considering the Phonitor in light of emerging immersive audio formats. As technologies like Dolby Atmos and Sony 360 Reality Audio gain traction, tools that can accurately represent spatial audio information over headphones may become increasingly valuable. While the Phonitor is primarily designed for stereo content, its underlying technology could potentially be adapted or expanded to address these new formats in the future.

Ultimately, the SPL Phonitor is a specialized tool that addresses a specific need in the audio production world. Its value will depend largely on individual workflows and requirements. For those who frequently work on headphones or need a reliable reference when away from their main monitoring setup, the Phonitor could be a game-changing addition to their toolkit. For others, it may be an interesting but non-essential luxury.

As with any audio tool, the true test of the Phonitor will be in its long-term use across a variety of projects. Initial impressions are certainly positive, but its real worth will be determined by how it influences the quality and efficiency of users' work over time. For now, it stands as an intriguing and innovative approach to a long-standing challenge in audio production.

TEST: SPL Passeq

 In today's audio landscape, very few products manage to elicit a respectful "Ohhh" from even the most experienced and jaded producers. While many former giants of studio technology now limit themselves to manufacturing budget interfaces, or lie gutted in the business corner after being bought out and stripped of their expertise, there are still a select few who hold high the flag of high-end outboard gear. They refuse to be deterred from their path despite massive attacks from the plugin market. One such company is undoubtedly SPL from Niederkrüchten, Germany, which has now launched the second generation of the "king of passive equalizers" with the SPL PassEQ.

A Fundamental Question...

Whenever I've tested a product from SPL's high-end hardware portfolio, one could be certain of a particular type of comment. There was always an entry along the lines of "Way too expensive, my XYZ plugin does the same thing and is included in the package." Undoubtedly, the vast majority of productions today are recorded digitally and never leave the digital realm until they're made audible again through analog speakers. Undoubtedly, very good productions can be created this way.

It's also undisputed that for the majority of music listeners, it's completely irrelevant how well a production is mixed and mastered. Issues like the color of their iPhone or "where is there even room for a speaker" are far more important than the charging behavior of capacitors and the saturation characteristics of coils. And you don't sell a single extra unit of CDs or MP3s by switching from "very good" to "phenomenal" sound quality. So why all the fuss?

But when you turn to the small segment of audiophile connoisseurs who not only enjoy the perfection of a flawless production but can also reflect back to a producer/engineer that it's the maximum of what technology can deliver, it's a completely different approach in terms of personal satisfaction. Whether it's Hilti for craftsmen, Tesla in electric vehicle manufacturing, or SPL in the hardware sector, the air gets very thin in this region, and thanks to "Made in Germany," the price tags correspondingly multi-digit. After all, no one is forcing anyone to buy.

The Inner Workings of the SPL PassEQ

The SPL PassEQ Model 1650 / 1654 is the successor to Model 2595. The predecessor was already considered the non plus ultra in the passive filter realm, so its successor will have a hard time raising the bar even further. Like its predecessor, the current SPL PassEQ works with 120V technology, which all SPL products in this price category use. This technology has proven to be significantly superior to the competition, which mostly works with 40V, in terms of dynamic range, distortion threshold, and signal-to-noise ratio.

The SPL PassEQ itself works with 3 filter bands per side for the boost range and 3 bands for the cut range. With 12 switchable frequencies per band, this makes 36 bands for the boost range and 36 bands for the cut range. Compared to its predecessor, the frequencies have been revised once again, and the quality of the frequencies in terms of capacitors, coils, and resistors has been recalibrated. The term "passive" of course only refers to the filter network; the SPL PassEQ naturally also has active components such as make-up amplifiers and the 120V technology.

The Housing of the SPL PassEQ

This product is clearly not built for mobile operation. You wouldn't even want to move this intimidating behemoth with its 10 kg weight and 4U height in the studio, which probably means that you install the SPL PassEQ in the rack once and then never touch it again. It's very convenient that on the rear panel there are only the balanced XLR outputs, the main fuse, a ground lift, and unfortunately also the power switch. This means using a power strip switch or crawling behind the 19" rack before starting production. Depending on which color the customer wants, the product designation varies between 1650 (black) and 1654 (red), a product variation that was also available with the "Iron" compressor.

Conceptually, the SPL PassEQ is a dual mono EQ, with both sides capable of being operated independently. You won't find a stereo link on this product, as it wouldn't be easily implementable with this design. However, since all controls are stepped, stereo operation is very easy, eliminating the fiddly search for identical levels.

Visually, the 14 controls per side are arranged in a star shape. According to the manual, the Q factor has been readjusted for each frequency range, with only the high-frequency range having an additional 41-step variable Q value. Even though the SPL PassEQ should generally only be operated by ear, here are the key data of the frequency bands again:

LF- : Low Frequency Cut, Shelving with approx. 6 dB/oct, 30 Hz to 600 Hz, max. 22 dB.
LF+ : Low Frequency Boost, Shelving with approx. 6 dB/oct, 10 Hz to 550 Hz max. 17 dB.
MF- : Mid Frequency Cut, Bell, 200 Hz to 6 kHz, max. 11.5 dB.
MF+ : Mid Frequency Boost, Bell, 220 Hz to 4.8 kHz, max 10 dB.
HF- : High Frequency Cut, Shelving, 580 Hz to 22 kHz, max. 14.5 dB.
HF+: High Frequency Boost, Bell, 5 kHz to 35 kHz, max. 12.5 dB.

As expected, the frequency bands of the adjacent frequency controls overlap, which should be considered in the phase response, but sometimes also opens up interesting sonic aspects.

Two illuminated bypass switches, one for each channel, serve as On/Off switches, plus the well-known SPL feature of Auto Bypass. To keep the ears fresh, when this function is activated, the sound processing is deactivated and reactivated at an adjustable interval to show the engineer the sonic effect compared to the original, without having to leave the sweet spot of the listening position.

The SPL PassEQ comes with a very well-written bilingual operating manual, which not only explains the pure functionality of the product but also generally addresses the different philosophies of filter use, especially the interaction between cutting and boosting frequencies.

The SPL PassEQ in Practice

Although the SPL PassEQ will probably find its main area of application in high-end mastering studios, it can of course also be placed classically in the signal path before conversion to the DAW, put on subgroups, or placed in the mix via AUX paths.

Especially when it comes to subtle modification of the source material, the SPL PassEQ scores across the board. It's hard to describe, but you get the impression that every signal experiences a sonic enhancement as soon as it has passed through the EQ. Be it a subtle tightening of the bass, an elegant peak in the midrange, or some shine in the uppermost treble range, the SPL PassEQ creates an alarming addiction factor that old hands might still remember from using the Aphex Aural Exciter in the eighties. Before you know it, you're sending virtually every signal through the product, and everything that sounded very good untreated until now suddenly seems rather dull and lackluster.

This also shows the strength of the product, which subtly supports strengths in the source material with an unpretentious basic orientation and elegantly conceals flaws. The parallel operation of the 6 filters is truly a powerful tool, although you would have to start with a massively distorted signal to activate all 6 filters simultaneously. Even if you start with a perfect signal, you still want to add a touch of shine in the extreme high-frequency range. It really takes a strong will not to let the SPL PassEQ run always and everywhere.

Expanding on the Technical Aspects

The SPL PassEQ's 120V rail technology is a key factor in its exceptional performance. This high-voltage design allows for significantly more headroom compared to standard 30V or 40V designs. The increased voltage translates directly into improved dynamic range and lower noise floor. In practical terms, this means the PassEQ can handle even the hottest input signals without breaking a sweat, maintaining clarity and definition even during complex processing tasks.

The passive filter networks employed in the PassEQ are another cornerstone of its design philosophy. Unlike active EQ circuits that rely on op-amps for gain, passive EQs use inductors, capacitors, and resistors to shape the frequency response. This approach is often praised for its musical and natural-sounding results, as it avoids potential coloration or distortion introduced by active components.

However, the PassEQ isn't purely passive. It incorporates active make-up gain stages to compensate for the inherent signal loss in passive networks. These gain stages are designed with the same attention to detail as the rest of the unit, ensuring they maintain the clarity and low noise floor of the passive section.

The stepped controls on the PassEQ serve multiple purposes. Beyond facilitating easy recall of settings, they also ensure precise matching between the left and right channels for stereo operation. Each step corresponds to a specific combination of components in the filter network, guaranteeing consistency and repeatability.

Circuit Design and Component Selection

SPL's approach to circuit design and component selection for the PassEQ is meticulous. Every resistor, capacitor, and inductor is chosen not just for its electrical properties, but for its sonic characteristics as well. This level of attention extends to the circuit board layout, where signal path length is minimized and potential sources of interference are carefully isolated.

The capacitors used in the PassEQ's filter networks are a mix of high-grade film and electrolytic types, each chosen for optimal performance in its specific application. The inductors are custom-wound to SPL's exacting specifications, ensuring consistency and minimizing potential sources of distortion.

Even the potentiometers receive special treatment. SPL uses conductive plastic types for their smooth operation and long-term reliability. The stepped nature of the controls is achieved through a combination of the potentiometer itself and a precision resistor network, providing both the tactile feel of a stepped control and the electrical precision required for repeatable settings.

Power Supply Considerations

The power supply of the SPL PassEQ is as carefully engineered as the audio circuitry it supports. A robust toroidal transformer forms the heart of the power supply, chosen for its efficiency and low electromagnetic emissions. Multiple stages of voltage regulation ensure that the 120V rails remain stable under all operating conditions.

Extensive filtering is employed throughout the power supply to minimize any potential noise or ripple that could find its way into the audio path. This includes both traditional passive filtering and active regulation stages.

The ground design of the PassEQ is also noteworthy. A star-grounding scheme is employed to minimize ground loops and reduce the possibility of noise entering the system through shared ground paths. The inclusion of a ground lift switch on the rear panel provides additional flexibility when integrating the PassEQ into complex studio setups.

User Interface and Ergonomics

While the technical performance of the PassEQ is paramount, SPL has not neglected the user experience. The front panel layout, with its star-shaped arrangement of controls, is not just visually striking but also ergonomically efficient. This design allows for quick access to all parameters without the need to hunt through menus or switch between layers.

The illuminated bypass switches serve a dual purpose. Beyond their primary function of engaging or disengaging the EQ, they also provide a clear visual indicator of the unit's status, visible even in dimly lit studio environments.

The Auto Bypass feature is a thoughtful inclusion that speaks to SPL's understanding of the creative process. By automatically toggling the EQ in and out of the signal path at user-defined intervals, it allows for objective comparison between processed and unprocessed audio. This can be invaluable in making fine adjustments and avoiding the pitfall of "EQ creep," where small, incremental changes can lead to over-processing.

Application in Modern Workflows

While the SPL PassEQ is undoubtedly a high-end piece of analog hardware, it's designed with an understanding of modern, hybrid workflows. Its balanced XLR inputs and outputs allow for easy integration with professional audio interfaces, allowing it to be inserted into a DAW session via hardware inserts.

For those working primarily in the box, the PassEQ can serve as a high-quality front end for recording, imparting its character to signals before they enter the digital domain. Alternatively, it can be used as part of a master bus chain, providing that final touch of analog magic to a mix before it's printed.

In mastering applications, the PassEQ truly shines. Its ability to make broad, musical adjustments without introducing artifacts is invaluable when working with full mixes. The overlap between adjacent frequency bands allows for complex tone shaping that can be difficult to achieve with traditional parametric EQs.

The PassEQ in Context

It's worth considering the SPL PassEQ in the broader context of the current audio equipment market. While software plugins have undoubtedly democratized access to high-quality processing, there remains a dedicated segment of the industry that values the unique characteristics of premium analog gear.

The PassEQ represents a significant investment, and it's natural to question whether its performance justifies its price tag, especially in an era where powerful EQ plugins are available at a fraction of the cost. However, it's important to recognize that products like the PassEQ aren't just about achieving a particular frequency response – they're about the entire signal path, the quality of the components, the headroom provided by the 120V design, and the subtle but cumulative effects these factors have on the final sound.

For studios and engineers operating at the highest levels of the industry, where even marginal improvements in sound quality can make a difference, the PassEQ represents a tool that can provide that extra edge. It's not about replacing digital workflows entirely, but about having options and being able to choose the right tool for each specific task.

Conclusion

The SPL PassEQ represents the culmination of decades of analog audio engineering expertise. It's a product that doesn't try to reinvent the wheel but instead refines every aspect of its design to achieve a level of performance that stands out even in the rarefied air of high-end audio equipment.

Its strengths lie in its ability to shape sound in a way that feels organic and musical, free from the artifacts that can sometimes plague digital processing. The PassEQ excels at making broad, sweeping tonal adjustments that enhance the natural character of a sound source, rather than imposing an obvious effect.

However, it's important to approach the PassEQ with realistic expectations. It won't magically transform a poor recording into a masterpiece, nor will it necessarily provide better results than well-designed software in every situation. Its value lies in its unique character, its exceptional build quality, and the subtle but meaningful improvements it can bring to already high-quality audio.

For those operating at the highest levels of audio production, where every decibel counts and the quest for sonic perfection is never-ending, the SPL PassEQ represents a powerful tool in the arsenal. It's a reminder that in the world of audio, there's still room for meticulously crafted hardware alongside cutting-edge digital technology.

Ultimately, the SPL PassEQ is a testament to the enduring appeal of premium analog gear in an increasingly digital world. It's not for everyone, and it doesn't need to be. For those who can appreciate and utilize its capabilities, it offers a level of performance and a particular flavor of sound shaping that stands apart in today's audio landscape.

TEST: SPL Masterbay S

 It's an interesting fact. Although our current DAWs are plastered with mastering software of all kinds, you still find predominantly hardware processors of all types in high-end mastering studios worldwide, which need to be managed. I won't start a new thread about hardware versus software here, but deep down we all know that in the high-end sector, hardware simply sounds better. We only use software solutions out of convenience and budget reasons. The extent to which the consumer hears or even appreciates this is another matter entirely. Germany's number one in professional hardware, the company SPL, set out some time ago to bring its mastering battleship MasterBay to market in a consumer-friendly version under the name SPL MasterBay S. Here now is the long-overdue test report.

The Concept of the SPL MasterBay S

As we all know, the well-known patchbay represents the central interface in every ambitious recording studio when it comes to integrating hardware processors. If you work on a project-by-project basis, you configure your current setup thoroughly once, complete the project, and start the next one with a correspondingly varied setup. So far, so good, but what if you want to compare different processors in an A/B comparison? The personal memorization of the last sound impression is quickly lost during the manual patching process, not to mention the difference in loudness.

What is needed here is a switchable patchbay, which primarily switches various hardware processors as stand-alone or in multiple loop operation during mixdown or as an ambitious AUX extension in console operation, while focusing on the most neutral sound possible and compensating for any level differences. This is where the SPL MasterBay S comes into play, with the letter "S" in this case standing for "Small". In contrast to the battleship MMC2, SPL adapts to the budget range here and offers a very comprehensive loop operation for up to 4 loops, in which individual processors or entire effect chains can be operated.

The Individual Sections of the SPL MasterBay S

Let's start with the corresponding connections on the rear of the product. As befits a professional device, all connections are stereo balanced in XLR. For simplicity, here's a list in table form:

- Main In: Main input, maximum input level +23dBu.
- Rec Out: Output of the processed signal, usually where an interface or converter is placed.
- Monitor Out: Connection for the monitor controller.
- DAW Return: Return of the signal to the DAW.
- Insert Send and Return 1-4: This is where the processors are looped in.
- Metering: Executed in TRS, you can connect a VU or PPM meter to monitor the level values in the recording signal.

The front side of the SPL MasterBay S is no less impressive, but additionally contains some very interesting features that greatly increase the practical value of the device.

Let's start with the trim potentiometer, which ranges from -6 to +6dB. An RK27 from ALPS, nicknamed "Big Blue", known for its high overload resistance and high chamber synchronization, is used. The trim switch above it can also completely remove the potentiometer from the signal path.

The following Insert switches 1-4 switch the corresponding loops. It is recommended to use processors for error correction such as de-essers, de-clickers, or noise reduction in Insert 1. Insert 2 is ideal for dynamic processors such as compressors or limiters. Insert 3 is perfect for equalizers, while Insert 4 is suitable for special effects such as exciters or stereo wideners.

The Bypass switch allows you to compare the original signal with the processed signal at any time. The Input switch allows you to choose between the Main In and DAW Return inputs. This is particularly useful when you want to compare the analog signal with the digital signal from your DAW.

The Monitor switch allows you to listen to either the input signal or the processed signal. This is particularly useful when you want to hear the effect of your processing in real-time.

The Output switch allows you to choose between the Rec Out and Monitor Out outputs. This is particularly useful when you want to send different signals to your recording device and your monitoring system.

The SPL MasterBay S in Practice

The SPL MasterBay S impresses with its clear and intuitive operation. The switches and potentiometers have a high-quality feel and operate precisely. The sound quality is, as expected from SPL, excellent. The device maintains a neutral sound character and does not color the signal noticeably.

One of the great strengths of the MasterBay S is its flexibility. You can easily create complex signal chains and switch between them at the push of a button. This allows for quick A/B comparisons, which is invaluable in the mastering process.

The trim potentiometer proves to be very useful in practice. It allows you to compensate for level differences between different processors or to adjust the overall level of your signal chain. The ability to bypass the trim potentiometer completely is also a nice touch, allowing you to maintain an absolutely clean signal path when no level adjustment is needed.

The metering output is another practical feature. It allows you to connect an external meter to monitor your levels accurately. This is particularly useful in mastering situations where precise level control is crucial.

Sound Quality and Technical Specifications

The SPL MasterBay S boasts impressive technical specifications that contribute to its excellent sound quality:

- Frequency response: 10 Hz - 100 kHz
- Signal-to-noise ratio: > 110 dB
- THD+N: < 0.001%
- Crosstalk: < -90 dB

These specifications ensure that the MasterBay S maintains the integrity of your audio signal throughout the processing chain. The wide frequency response ensures that even the subtlest details in your mix are preserved, while the low noise floor and minimal distortion keep your signal clean and clear.

Comparison with Other Products

When compared to other mastering-grade switching systems, the SPL MasterBay S holds its own. While it may not have all the features of its bigger brother, the MMC2, it offers a more affordable option for those who don't need the full feature set of a high-end mastering console.

Compared to simpler patchbays or manual patching systems, the MasterBay S offers significant advantages in terms of workflow efficiency and sound quality. The ability to switch between different signal chains instantly and perform accurate A/B comparisons is invaluable in a mastering context.

Integration into the Studio Environment

The SPL MasterBay S integrates seamlessly into both professional and home studio environments. Its standard XLR connections make it compatible with a wide range of professional audio equipment. The ability to switch between DAW and analog inputs also makes it a versatile tool for hybrid analog/digital setups.

For smaller studios or project studios, the MasterBay S can serve as the centerpiece of a hardware-based mastering setup. It allows you to incorporate a few choice pieces of outboard gear into your mastering chain without the need for a complex and expensive mastering console.

Potential Improvements and Considerations

While the SPL MasterBay S is an excellent piece of equipment, there are a few areas where improvements could be made:

1. Digital connectivity: In an increasingly digital world, the addition of digital I/O could enhance the unit's versatility.

2. More insert points: While four insert points are sufficient for many users, some may wish for more.

3. Recall functionality: The ability to save and recall routing configurations could be beneficial for users working on multiple projects.

It's also worth noting that to fully utilize the MasterBay S, you need to have quality outboard gear to connect to it. This could represent a significant additional investment for those just starting to build their hardware collection.

The SPL Marc One is a high-end monitor and recording controller that has garnered attention for its excellent build quality, top-tier hardware components, and impressive audio performance. However, like any piece of equipment, it has its known issues and potential weaknesses. Here are some of the notable points:

### Known Issues and Weaknesses

1. **External Power Supply**: One of the primary criticisms of the SPL Marc One is its reliance on an external multi-voltage power supply. This design choice, likely made to save space, means that the device requires an additional power outlet. While the transformer is located directly on the plug (rather than halfway along the power cable), it still occupies a socket on the power strip, which can be inconvenient in setups with limited power outlets[1].

2. **Limited I/O Channels**: The built-in interface of the Marc One features only two inputs and two outputs. While this might be sufficient for many users, those requiring more extensive I/O capabilities might find this limitation restrictive. The focus on quality over quantity is evident, but it does place the Marc One in a different category compared to more feature-rich alternatives[1].

3. **USB-B Port**: The use of a USB-B port instead of the more modern USB-C might be seen as a drawback by some users. As the industry gradually shifts towards USB-C for its enhanced capabilities and compatibility, the absence of this port on the Marc One could be a point of contention. However, the USB-B port does offer greater compatibility with existing setups, and USB-C is backward compatible with appropriate adapters[1].

4. **Solo Operation Design**: The SPL Marc One is primarily designed for solo operation. This is evident from the solitary headphone output and the fact that both the monitors and the headphone output can only be fed with the same signal. This design choice makes it ideal for individual producers who prioritize high fidelity but might not be suitable for collaborative environments where independent mixes are required for multiple listeners[1].

5. **No Rack Mounting Option**: The Marc One is designed exclusively for desktop use, with no provision for rack mounting. This could be a limitation for users who prefer or require rack-mounted equipment to save desk space or for better integration into their existing studio setups[1].

### Additional Considerations

1. **Component Quality**: The SPL Marc One uses high-quality components, such as the AKM AK4490 chip for its DAC, which supports 32-bit at 768 kHz and DSD4 and DSD256. This ensures excellent sound quality but also means that the device is best paired with equally high-quality monitors and headphones to fully utilize its capabilities. Investing in high-end speakers and headphones is recommended to avoid bottlenecks in the signal chain[1].

2. **Trim Potentiometer**: The trim potentiometer, which ranges from -6 to +6 dB, is a useful feature for adjusting input levels. However, some users might find the need to manually adjust this setting a minor inconvenience, especially in dynamic recording environments where levels can change frequently[1].

3. **Crossfeed Control**: The crossfeed control, borrowed from SPL's Phonitor series, is designed to simulate the natural crosstalk of speakers when using headphones. While this feature adds flexibility and enhances the listening experience, it might not be necessary for all users, particularly those who primarily use speakers for monitoring[1].


Conclusion

The SPL MasterBay S is a well-designed and high-quality tool that brings professional-grade mastering capabilities to a wider audience. It offers a level of flexibility and control that is typically only found in much more expensive mastering consoles.

For project studio owners looking to incorporate hardware into their mastering chain, or for professional mastering engineers seeking a more affordable alternative to high-end mastering consoles, the MasterBay S represents excellent value. It allows for complex routing and instant comparisons, which can significantly streamline the mastering workflow.

The build quality is excellent, as we've come to expect from SPL, and the sound quality is transparent and clean. While there are a few areas where improvements could be made, these are minor considerations in the context of the overall package.

In summary, the SPL MasterBay S is a powerful and flexible tool that can significantly enhance your mastering capabilities. It bridges the gap between software-based mastering and high-end hardware solutions, offering a taste of professional-grade mastering workflows at a more accessible price point. For those serious about improving their mastering setup, the MasterBay S is certainly worth considering.

TEST: SPL Marc One

 Monitor and recording controllers are essential components of every project and professional studio, resulting in a vast selection of products on the market. However, when focusing on sound quality, the overwhelming number of options quickly narrows down to a more manageable pool. The German flagship company SPL once again aims to make its mark in this segment with the SPL Marc One, as this review will demonstrate. The One series focuses on providing a cost-effective alternative to the company's high-end products without compromising on the "Made in Germany" quality.

A Subtle Difference

At first glance, it's evident that the SPL Marc One shares many similarities with its smaller sibling, the Control One. The elegant form in a black layout, with the interior packaged in a solid steel housing featuring a slightly textured surface, is immediately recognizable. If this product weren't primarily intended for studio environments, the controller would even pass the scrutiny of the most discerning interior design critic, thanks to its attractive rounded housing.

The dimensions of 210 mm x 49.6 mm x 220 mm are compact, allowing for multiple placement options. Weighing just under 1.5 kg and equipped with high-quality rubber feet, the SPL Marc One maintains a secure position even on glass surfaces, providing sufficient grip for stable operation. The product is designed exclusively for desktop use, with no provision for rack mounting.

Unlike the Control One, the Marc One features a built-in interface, which sets the two products apart in different categories. This interface doesn't boast an extensive number of I/O channels, offering only two inputs and two outputs. Instead, it shines with the AKM AK4490 chip, which supports 32-bit at 768 kHz, as well as DSD4 and DSD256.

The Structure of the SPL Marc One

A brief touch of the product's controls and switches hints at why the SPL Marc One comes with a retail price of €699. As with the converter section, only the finest components are used. Firmly gripping, defined switches, tactile potentiometers, and solid sockets provide the desired haptic experience. Unfortunately, due to space constraints, SPL opts for an external multi-voltage power supply instead of an IEC connector. This once again blocks a socket on the power strip, but at least the transformer is located directly on the plug and not halfway along the power cable as with many other products.

All controls of the SPL Marc One are located on the front of the unit. Starting from the left is a triple mini-switch that toggles between the two selectable monitor paths or mutes both outputs, followed by the product's volume control. If the input level is very high, the input sensitivity can be reduced by -10 dB using a DIP switch on the rear panel. Another mini-switch allows for standard stereo playback, mono summed, or left/right swapped.

Three small red LEDs follow, indicating the product's status and potential overloads of the two channels in the interface. Next are three small knobs, beginning with the monitor control, which determines the mix ratio between input and USB signals due to latency-free playback. This is followed by the headphone volume control and a crossfeed control, similar to the one found in the company's Phonitor Matrix, which simulates the "crosstalk" of speakers in regular monitor playback to enable a more spatial representation.

The Rear Panel of the SPL Marc One

For primary I/Os, the Marc One uses balanced TRS connections, while secondary I/Os use unbalanced TS jacks for signal routing, primarily to keep the space requirements low, which would not be possible with XLR connectors. The second DIP switch on the rear panel allows for selection in recording management. When switched up, it is in the "Off" mode and enables recording of Line Input 1 via USB. In the down position (On mode), inputs 1 and 2 are summed and recorded via USB.

The controller offers 2 and 2.1 monitor functionality, with the subwoofer for Monitor A receiving a separate output. An additional Line Out provides the mix of Line inputs and USB playback, has unity gain, and is independent of the volume control. The specified values of the SPL Marc One truly suggest very high quality. For the number enthusiasts, here are some specifications:

- Frequency response: 10 Hz to 200 kHz at -3 dB
- Dynamic range of 120 dB
- A-weighted noise of -99 dB at 600 Ohms
- Total harmonic distortion of 0.002% (0 dBu, 10 Hz–22 kHz, 600 Ω)
- Crosstalk below 75 dB @ 1 kHz

Headphone amplifier:
- 0.013% THD at maximum output power
- Power: 2 x 190 mW (1 kHz, 600 Ω), 2 x 330 mW (1 kHz, 250 Ω), and 2 x 400 mW (1 kHz, 47 Ω)

Whether a USB-B port is still appropriate nowadays is ultimately for the consumer to decide. As the entire industry is slowly moving towards USB-C, some users might miss a corresponding port. On the other hand, this port allows for greater compatibility, especially since the USB-C realm is backward compatible via appropriate adapters.

In Practice

As always, the description of a product's sound is purely subjective and does not speak for the general public. However, I am quite certain that every user of the SPL Marc One can agree on the sonic description of "excellent." The product has an extremely stable basic sound with outstanding resolution and plenty of headroom for any type of headphone. The above-mentioned values speak a clear language regarding the performance in the headphone area and, in conjunction with the crossfeed control, enable excellent flexibility, as otherwise only provided by top-class dedicated headphone amplifiers.

All sound material is resolved very finely and experiences the maximum sound quality that can come to a signal flow through the components used. The product impresses not only with outstanding audio values but also with its "unexcited" appearance and simple yet effective haptics, which make working with this product a real highlight. Whether by chance or not, the SPL Marc One fits exactly under the foot of my 27" iMac and would thus perfectly integrate into a large number of studios. Otherwise, it also fits very well beside the foot.

The discerning reader will also have noticed that the controller is primarily designed for solo operation. Both the solitary headphone output and the fact that the monitors and headphone output can only be fed with the same signal indicate that the target group is the individual producer who places great value on the highest fidelity both through the speakers and the headphones.

However, one should always keep in mind that the SPL Marc One can only fully play out its strengths in cooperation with the corresponding components. It makes no sense to invest in a high-quality controller if, on the other hand, the equivalent value of the monitors, possibly even including the headphones, only amounts to the same amount as the controller alone. As always in the signal flow, it only takes one "weak" component to make the entire construct wobble and equalize all other high-quality products. Therefore, I recommend investing at least 5 times, preferably 7 times, the amount for the acquisition of equally high-quality speaker or headphone material to be able to use the full performance range of the controller. The end result will justify the investment.

Extended Information

Integration with Studio Environments

The SPL Marc One is designed to seamlessly integrate into various studio setups. Its compact size and sleek design make it an ideal choice for both home studios and professional environments. The controller can be easily placed on a desk or mounted in a custom studio furniture setup, providing easy access to its controls.

For studios that frequently switch between different monitoring systems, the Marc One's dual monitor outputs prove particularly useful. This feature allows for quick comparisons between different speaker setups, enhancing the mixing and mastering process.

Digital-to-Analog Conversion

The heart of the SPL Marc One's digital capabilities lies in its AKM AK4490 chip. This high-performance DAC is known for its excellent sound quality and low distortion characteristics. The chip's ability to handle high-resolution audio formats, including DSD4 and DSD256, makes the Marc One future-proof for evolving audio standards.

The 32-bit/768 kHz capability ensures that even the most demanding audiophile recordings can be reproduced with utmost fidelity. This level of resolution allows for the capture of subtle nuances and micro-dynamics that might be lost with lesser converters.

Crossfeed Technology

One of the standout features of the SPL Marc One is its crossfeed control, borrowed from SPL's renowned Phonitor series. This technology aims to simulate the natural crosstalk that occurs when listening to speakers in a room, but applied to headphone listening.

The crossfeed function can be particularly beneficial for mixing engineers who need to switch between speaker and headphone monitoring. It helps create a more natural and spacious soundstage when using headphones, potentially reducing ear fatigue during long mixing sessions.

USB Interface Functionality

While the USB interface of the Marc One is limited to two inputs and two outputs, its implementation is focused on quality rather than quantity. The USB connection allows for direct recording and playback from a computer, making it an excellent choice for project studios or mobile recording setups.

The ability to sum inputs 1 and 2 for USB recording provides flexibility for capturing stereo sources or two mono sources simultaneously. This feature can be particularly useful for recording interviews, podcasts, or stereo instrument performances.

Build Quality and Longevity

SPL has a reputation for building equipment that lasts, and the Marc One is no exception. The use of high-quality components, from the potentiometers to the internal circuitry, ensures that the unit will maintain its performance over years of use.

The solid steel housing not only contributes to the product's durability but also helps in shielding the sensitive audio components from electromagnetic interference. This attention to detail in construction translates to cleaner, more accurate audio reproduction.

Comparison with Competitors

In the competitive landscape of monitor controllers, the SPL Marc One positions itself as a high-end option for discerning users. When compared to similarly priced competitors, the Marc One often stands out for its build quality and sound performance.

However, it's worth noting that some competitors in this price range may offer additional features such as talkback functionality or more extensive I/O options. The Marc One's strength lies in its focus on core functionality and audio quality rather than an extensive feature set.

User Experience and Workflow Integration

The intuitive layout of the SPL Marc One contributes significantly to its user-friendliness. The straightforward control scheme allows for quick adjustments during recording or mixing sessions without interrupting the creative flow.

For producers and engineers who frequently switch between different monitoring scenarios – such as comparing mixes on different speaker sets or transitioning from speakers to headphones – the Marc One's design facilitates these workflows efficiently.

Potential Limitations

While the SPL Marc One excels in many areas, potential buyers should be aware of its limitations. The lack of a built-in talkback microphone might be a drawback for some studio setups, particularly those involved in voice-over work or requiring frequent communication between the control room and recording space.

Additionally, the reliance on an external power supply, while common in this product category, may be seen as a minor inconvenience by some users, particularly those with limited power outlet availability in their studio space.

Conclusion

The SPL Marc One represents a high-end monitor and recording controller in the German manufacturer's portfolio. The product impresses with its excellent build quality, top-tier hardware components, aesthetic appeal, and a converter concept that extracts the maximum from the signal flow with 32-bit at 768 kHz capabilities.

The controller's operation is intuitive, self-explanatory, and characterized by a simplicity that is often desired, especially in the monitoring realm. The timeless appearance, combined with excellent internal values, is convincing across the board, particularly for the solitary producer who doesn't need to route independent mixes to additional listeners.

While the SPL Marc One comes with a premium price tag, its performance and build quality justify the investment for those seeking uncompromising audio quality in their studio setup. It stands as a testament to SPL's commitment to delivering professional-grade audio equipment that meets the exacting standards of modern music production.

TEST: Fischer Amps Body Pack XLR

 Most users, when thinking about in-ear monitoring, will likely think of a wireless system consisting of a receiver and a transmitter. However, there are situations where the physical freedom provided by a wireless system is not necessary. For example, in theater performances or for musicians who are required or instructed not to move much on stage. In such cases, it is significantly better to rely on a wired variant and avoid the issues associated with wireless systems, such as frequency selection, dropouts, battery consumption, and similar concerns.

For these situations, the Fischer Amps Body Pack XLR was developed in collaboration with the Guitar In Ear Cable. The crucial point is that you have the option to extend your in-ear system almost indefinitely with a special cable, allowing you to take up a significantly distant position from your amplifier without having to forgo in-ear monitoring.

To use the system, however, a special headphone amplifier, also from Fischer Amps, is required, which specifically supports this system with XLR cabling. The usage is as follows:

The system features a passive in-ear amplifier called the In Ear BodyPack XLR. This is attached to the belt, waistband, or similar using a clip. The Fischer Amps Body Pack XLR is then connected to the Fischer-Amps headphone amplifier via an XLR plug using an XLR cable of unlimited length. Parallel to the XLR cable, the Guitar In Ear Cable has a high-quality instrument cable of the same length attached using several heat shrink tubes. The lengths of this cable combination vary between 6 and 10 meters. This cable combination prevents cable clutter on stage, ultimately leaving you with only one cable that supplies both the in-ear system and the guitar or bass amplifier with the appropriate signal.

## Why XLR?

Some may wonder why one should opt for such a solution. In principle, it would be possible to use a regular headphone amplifier, which usually has a stereo jack output, to supply a headphone and achieve the same result with home wiring. This is essentially correct, but the XLR combination offers an invaluable advantage.

Normally, XLR cables are available in abundance on any stage or festival due to microphone setups. This means you can always extend your cable from anywhere. This is not the case with a regular jack headphone amplifier, especially since a jack extension is not nearly as stable in connection and is almost never available at any festival or club wiring. In this case, you would have to bring all the cabling yourself and would not have the option to change the length of the in-ear cable.

## The Fischer Amps Body Pack XLR in Practice

First of all, it should be noted that the Body Pack is a classic "Made in Germany" product with all its positive attributes. The product is extremely robustly built, and even the clip cannot be dislodged with significant pulling and pushing. The volume control is not only pleasantly resistant but also additionally notched, which provides a better tactile feel.

The notching, in particular, gives a pleasant feeling, so that when you change the volume, you not only notice a difference in the earphones but can also make very fine adjustments with your fingertips. Since the Fischer Amps Body Pack XLR is likely to be located on the back of the body and you cannot see where the control is, tactile impressions are very important. Overall, all screws, connectors, and sockets are of the highest quality, so there is absolutely nothing to criticize here, only to praise.

The same quality continues in the Guitar In Ear Cable, where the cables strike the right balance between rigidity and flexibility. A cable that is too flimsy is just as unpleasant as a half railway track that cannot be bent once the cable has taken its position.

## Considerations for the 10-Meter Version

When using the 10-meter version, it should be noted that if you use passive pickups on your guitar, anything over 6 meters in cable length will have some impact on the sound. Usually, this results in a high-frequency damping, which, depending on the tonal orientation of the pickup, can positively affect the overall sound. However, it should be noted that completely neutral reproduction with passive pickups is only possible up to about 6 meters, or you use active pickups, where cable lengths well over 30 meters pose no tonal problem.

## Conclusion

With the Fischer Amps Body Pack XLR, the German manufacturer has a very practical, uncomplicated tool in its portfolio. The concept is simple, the implementation straightforward, and the workmanship excellent.

However, it should be kept in mind that a suitable headphone amplifier, such as the Fischer Amps In Ear Amp 2, is required for operation.
### Design and Build Quality

The Fischer Amps Body Pack XLR is designed with durability and user-friendliness in mind. Its robust construction ensures that it can withstand the rigors of live performances and frequent use. The metal housing provides excellent protection against physical damage, while the high-quality connectors ensure a stable and reliable connection.

The volume control, as mentioned earlier, is not only resistant but also features a notched design, allowing for precise adjustments even in low-light conditions or when the device is out of sight. This attention to detail highlights Fischer Amps' commitment to creating products that cater to the needs of professional musicians and performers.

### Compatibility and Versatility

One of the standout features of the Fischer Amps Body Pack XLR is its compatibility with a wide range of in-ear monitors and headphones. This versatility makes it an ideal choice for musicians who use different types of in-ear systems depending on the performance or rehearsal setting.

Additionally, the ability to use standard XLR cables for extending the connection means that users can easily integrate the Body Pack XLR into existing setups without the need for specialized cables. This flexibility is particularly beneficial for touring musicians who may need to adapt to different stage setups and environments.

### Sound Quality and Performance

The Fischer Amps Body Pack XLR is designed to deliver high-quality audio with minimal signal loss. The passive design ensures that the audio signal remains clean and unaltered, providing an accurate representation of the sound source. This is crucial for musicians who rely on precise audio monitoring to perform at their best.

The use of high-quality components and careful engineering ensures that the Body Pack XLR maintains a low noise floor, reducing the risk of unwanted interference or hum. This attention to detail in the design and construction of the device ensures that it meets the high standards expected by professional users.

### Practical Applications

The Fischer Amps Body Pack XLR is particularly well-suited for a variety of performance settings. In theater productions, where actors and performers may need to remain stationary or move within a confined area, the Body Pack XLR provides a reliable and unobtrusive solution for in-ear monitoring.

For musicians, especially those in bands or orchestras, the Body Pack XLR offers a practical alternative to wireless systems. By eliminating the need for batteries and reducing the risk of signal dropouts, it provides a more stable and consistent monitoring experience. This can be particularly beneficial in rehearsal settings where long sessions may be required.

### User Experience and Feedback

Users of the Fischer Amps Body Pack XLR have consistently praised its reliability and ease of use. The robust construction and high-quality components have been highlighted as key strengths, along with the device's ability to deliver clear and accurate audio.

Musicians and performers have also appreciated the flexibility offered by the XLR connection, allowing them to easily extend their setup as needed. The notched volume control has been particularly well-received, providing a tactile and precise way to adjust audio levels on the fly.

### Additional Features and Accessories

Fischer Amps offers a range of accessories to complement the Body Pack XLR, including various lengths of XLR cables and high-quality in-ear monitors. These accessories are designed to work seamlessly with the Body Pack XLR, ensuring a cohesive and reliable monitoring system.

The company also provides a range of headphone amplifiers, such as the Fischer Amps In Ear Amp 2, which are specifically designed to work with the Body Pack XLR. These amplifiers offer additional features and controls, allowing users to tailor their monitoring setup to their specific needs.

The Fischer Amps Body Pack XLR is particularly effective in several specific application areas:

1. Theater productions: It provides a reliable wired in-ear monitoring solution for actors and performers who need to remain stationary or move within a confined area on stage[1][2].

2. Orchestras and ensembles: The system allows musicians to have individual volume control while maintaining a clean stage setup using standard XLR cables[2].

3. Live music performances: For bands or musicians who are required to stay in fixed positions on stage, eliminating the need for wireless systems[1].

4. Studio recording sessions: Offers a stable monitoring solution without wireless interference concerns[2].

5. Conferences and presentations: Provides clear audio monitoring for speakers or panelists[2].

6. Situations requiring extended cable runs: The use of XLR connections allows for practically unlimited extension of the monitoring system using standard microphone cables readily available at most venues[1][2].

7. Environments with high RF interference: By using a wired solution, it avoids potential dropouts or interference issues common with wireless systems.

8. Budget-conscious setups: Eliminates the need for expensive wireless transmitters and receivers while still providing high-quality monitoring.

9. Multi-musician setups: When used with the Fischer Amps In Ear Amp 8, it can provide individual volume control for up to eight musicians from a single source[2].

10. Scenarios where battery management is problematic: Being a passive device, it doesn't require battery power, reducing maintenance and reliability concerns during long performances or rehearsals[1][3].

The Fischer Amps Body Pack XLR's versatility and reliability make it an excellent choice for these applications, especially when combined with its ability to integrate seamlessly into existing audio setups using standard XLR cabling.

The Fischer Amps Body Pack XLR can be effectively used in studio conditions as well. Here are some key points about its use in studio settings:

1. Clean signal path: The passive design of the Body Pack XLR ensures a clean, unaltered audio signal, which is crucial for accurate monitoring in studio environments[1][5].

2. Versatility: It can be used with various types of headphones and in-ear monitors, allowing studio musicians and engineers to use their preferred monitoring devices[2].

3. Individual volume control: The Body Pack XLR allows users to adjust their own monitoring volume without affecting the main mix, which is particularly useful in multi-musician recording sessions[5].

4. Cable management: The XLR connection allows for easy integration into existing studio setups and can help reduce cable clutter[1][2].

5. Durability: The robust construction (aluminum housing, stainless steel belt clip, Neutrik connectors) makes it suitable for long-term studio use[1][2].

6. No interference: Unlike wireless systems, the wired nature of the Body Pack XLR eliminates potential RF interference issues that could affect recording quality[3].

7. Compatibility: It works well with studio headphone amplifiers, including Fischer Amps' own In Ear Amp series, providing a cohesive monitoring solution[3][5].

8. Extended monitoring setups: In larger studio spaces, the ability to use standard XLR cables allows for flexible positioning of musicians while maintaining high-quality audio monitoring[1][5].

While primarily designed for live performance scenarios, these features make the Fischer Amps Body Pack XLR a valuable tool in studio environments as well, offering reliable and high-quality personal monitoring for recording sessions.

### Conclusion and Final Thoughts

The Fischer Amps Body Pack XLR is a well-designed and highly practical solution for in-ear monitoring in a variety of performance settings. Its robust construction, high-quality components, and flexible XLR connection make it an ideal choice for professional musicians and performers.

While it requires a compatible headphone amplifier for operation, the benefits of using the Body Pack XLR far outweigh this minor inconvenience. The ability to extend the monitoring setup with standard XLR cables, combined with the device's reliable performance and ease of use, make it a valuable addition to any professional audio setup.

In summary, the Fischer Amps Body Pack XLR is a testament to the company's commitment to quality and innovation. It provides a reliable and flexible solution for in-ear monitoring, ensuring that musicians and performers can focus on their performance without worrying about technical issues.

TEST: SPL Iron

 Oh, children, how much has been written about the demise of classic 19-inch outboard gear. Thanks to an inflationary flood of plug-ins, additionally equipped with an exponential number of presets, some contemporaries already envisioned the total absence of any hardware in nearly all recording studios worldwide. CPU power versus analog experience, why understand how a product works when the preset is so close at hand. The swan song echoed up to the levels of the last high-end studios, and the dismantling of 19-inch racks was already in full swing.

But then there's the countermovement. Why am I thinking of Asterix now? "All of Gaul is occupied. All of Gaul? No, one small village..." In the audio world, that village is called Niederkrüchten, and the menhir being thrown around is called SPL Iron. It's a mastering tube compressor where the term "menhir" can be taken quite literally. 4 rack units high, 11 kilograms in weight, a product name straight out of the metal scene, and a front panel that would look good in the control module of the Starship Enterprise, instilling more than just respect in the ambitious user.

Oh yes, this behemoth comes with a retail price of €4,290! Any questions? Then let's dive into the top league of the old school, the classic high-end segment.

Construction

Primarily designed as a mastering compressor, the SPL Iron doesn't shy away from showcasing its qualities as a 2-channel single compressor or in a subgroup. Developed by company founder Wolfgang Neumann, this tube compressor cites the legendary Fairchild 670 as the inspiration for its creation, although the product should be understood merely as an inspiration. Once again, SPL has further developed and optimized a successful concept. For example, 2 tubes per channel work in the signal path, which additionally influence the sound characteristics via a triple bias control.

Moreover, via the control labeled "Rectifier," various rectifier circuits can be activated to generate the bias control voltage, switching between pure germanium, silicon, or LED characteristics. As expected, SPL also uses its proprietary 120-volt technology in the Iron, guaranteeing additional headroom in the processing.

Similar to other flagships from SPL production, such as the Passeq, SPL pulls out all the stops with the Iron, offering everything the user can imagine in terms of exquisite content. The finest components, some of which are even supplied as custom-made items, are used throughout. Starting with Mu-metal iron transformers from manufacturer Lundahl, to VU meters equipped with special ballistics, to high-quality Big Blue potentiometers from ALPS. The product is available in the classic black outfit as well as in a bold red.

Controls

Even though it has been documented countless times, smoothing dynamic peaks is among the highest classes of signal processing. No other processor is as difficult to operate as an analog compressor, no other tool can generate so much sonic gain and, in the same breath, cause so much damage. To make operation as clear as possible, the SPL Iron has been visually very generously dimensioned in all controls.

To state it upfront, you'll search in vain for a ratio control on a tube compressor. Rather, the degree of compression depends on the threshold level and the input level, with compression increasing at higher levels and lower thresholds. Not least for this reason, compressors of this design are considered particularly "musical" in their sound.

Back to the controls. The eye-catchers are the two huge threshold controls, which have a 41-step gradation. Below this control are the input control on the left and the output control on the right, with a toggle switch allowing the function to be switched between standby, amplification, or attenuation of the level. The attack and release controls are equipped with a six-step grid and are located on the left and above the threshold control, respectively. Mirrored to this is a sidechain EQ switch, which, if needed, intervenes in 4 preset frequency curves, and the aforementioned Rectifier, one of the highlights of the SPL Iron. Optionally, the sidechain can also be driven by an external sound source.

But that's not all in terms of special features. Centrally located are 2 presets that can be activated via a mini-switch if needed. In the "Airbase" position, the signal experiences a touch of loudness in the sound due to the boost of bass and treble, while in the Tape Roll-Off position, the sound behavior of old tape machines is imitated in the form of a slight lowering of the bass and a stronger lowering of the treble. As a very practical cherry on top, the SPL also impresses with its Auto Bypass circuit. To prevent ear fatigue and ensure an objective assessment of the signal at all times, the deactivation of the sound processing can be left to an automatic system. Using a centrally located rotary control, you can choose the time window between short (left stop) and long (right stop).

Finally, there's the Link switch to mention, which not only manages the left channel in the time parameters Attack and Release together, as is usually the case, but couples all controls except the Input and Output controls.

In Practice

Oh boy, this is going to be another test that's brimming with superlatives. Starting with a noise level that can still be captured metrologically in a diagram but in practice lies below the threshold of perception. But when it comes to the actual sound, it's truly difficult to give an appropriate description to the generated sound. Even in the first processing steps, the SPL Iron impresses with a very soft and unobtrusive processing, which immediately triggers an addictive character. Depending on the Rectifier setting in combination with the Tube Bias switches, you can indeed generate all the sound effects for which the archaic component in the form of a tube stubbornly persists in the higher performance class.

Depending on the Rectifier circuit, either the impulse peaks move to the foreground, while in the next setting, suddenly the room component gains significantly more volume. The saturation generated with the Bias switch also has a dramatic effect on the final sound depending on the combination with the Rectifier control. The musicality in the overall sound is always maintained. The delivered signal is compressed so roundly that it's rarely offered elsewhere. If you want to get really dirty now, throw a DAW compressor plug-in into the mix after an extensive session with the SPL Iron and compare the sounds. You won't believe your ears.

Expanding on the Technical Aspects

The SPL Iron's impressive performance is not just about its sound quality, but also its technical specifications. Let's delve deeper into some of these aspects:

1. 120V Rail Technology:
SPL's proprietary 120V operating voltage is a key factor in the Iron's performance. This higher voltage allows for greater headroom and dynamic range compared to standard 30V or 60V designs. The result is cleaner signal processing, especially when dealing with complex material or high signal levels.

2. Tube Selection and Biasing:
The Iron uses carefully selected tubes, likely ECC83/12AX7 types, known for their smooth distortion characteristics. The triple bias control allows users to fine-tune the tube's operating point, affecting harmonic content and compression behavior. This level of control is rare even among high-end tube compressors.

3. Rectifier Options:
The rectifier circuit in a compressor plays a crucial role in shaping its dynamic response. The Iron's switchable rectifier types (germanium, silicon, LED) each have unique characteristics:
   - Germanium: Slower response, softer knee, more vintage character
   - Silicon: Faster response, harder knee, more modern sound
   - LED: Very fast response, can add a slight edge to transients

4. Sidechain EQ:
The built-in sidechain EQ offers four preset curves, likely tailored for common mastering scenarios. This feature allows for frequency-dependent compression without the need for external equipment. Common applications might include de-essing (high-frequency sensitivity) or preventing low-frequency pumping.

5. VU Meters with Custom Ballistics:
The VU meters aren't just for show. Their custom ballistics are designed to provide accurate representation of the compressor's action, especially important for the often subtle moves made in mastering.

6. Transformer Design:
The use of Lundahl transformers is significant. These high-quality components contribute to the unit's overall sound character, potentially adding a slight, pleasing coloration and helping to isolate the unit from external interference.

Practical Applications

While primarily designed for mastering, the SPL Iron's flexibility makes it suitable for a variety of applications:

1. Mastering:
In mastering, the Iron excels at providing subtle cohesion to a mix. Its ability to gently round off peaks while adding harmonic richness can help bring life to digital recordings. The auto-bypass feature is particularly useful here, allowing for objective before/after comparisons.

2. Mix Bus Compression:
On a mix bus, the Iron can add "glue" to the overall mix. Its musical compression characteristics and variable tube coloration can enhance the sense of depth and width in a mix.

3. Vocal Processing:
For vocal tracks, the Iron's smooth compression and harmonic enhancement can add presence and intimacy. The sidechain EQ can be used to prevent excessive compression on sibilants.

4. Bass Guitar:
The Iron's tube circuitry and flexible compression settings make it excellent for bass. It can add warmth and consistency to bass tracks without losing low-end impact.

5. Drum Bus:
On drums, the Iron can provide anything from subtle sustain enhancement to aggressive, pumping effects. The rectifier options are particularly useful here for shaping the compressor's attack and release characteristics.

Comparison with Digital Emulations

While software emulations of classic compressors have come a long way, hardware units like the SPL Iron offer several advantages:

1. Tactile Control: The large, high-quality knobs and switches provide a level of intuitive control that's hard to replicate with a mouse and screen.

2. Real-time Processing: Hardware compressors process audio in real-time without introducing latency, which can be crucial in live or tracking scenarios.

3. Unique Analog Character: While digital emulations can sound very good, they often struggle to perfectly replicate the subtle nonlinearities and organic nature of analog circuits, especially when pushed hard.

4. Dedicated Processing: Using outboard gear like the Iron frees up CPU resources for other tasks in a DAW-based workflow.

5. Inspiration Factor: There's an intangible but real inspirational quality to using high-end hardware that can positively influence the creative process.

Integration in Modern Workflows

Despite being a piece of high-end analog outboard gear, the SPL Iron can integrate seamlessly into modern, hybrid analog/digital setups:

1. Digital Audio Workstation Integration: The Iron can be inserted into a DAW mix via high-quality A/D and D/A converters, allowing for recall of settings and automation of the bypass function.

2. Stem Mastering: In today's world of stem mastering, the Iron can be used to process individual stems before final assembly in the digital domain.

3. Parallel Processing: The unit's dual-mono capability allows for parallel compression techniques, where the compressed signal is blended with the dry signal for more subtle control.

Potential Improvements

While the SPL Iron is a top-tier product, there are always areas for potential improvement or expansion:

1. Digital Control: Some users might appreciate the option for digital recall of settings, although this would add complexity to the pure analog design.

2. Mid-Side Processing: Given its mastering focus, mid-side processing capabilities could be a valuable addition for some users.

3. Variable High-Pass Filter: While the sidechain EQ presets are useful, a fully variable high-pass filter in the sidechain could offer even more precise control.

Conclusion

With the SPL Iron, Wolfgang Neumann and his team have brought a tube compressor to the market that is truly in a class of its own. Everything that can be written in superlatives in a test for a device of this type applies to this unit.

From excellent measured values, through brilliant detailed solutions, to unmatched flexibility, to a unique sound that will convince any doubter. SPL has truly created a masterpiece with this product.

Yes, the product has its price, but with development AND complete manufacturing in Germany, coupled with these exceptional values, a retail price beyond the €4,000 mark seems like fair value for this unique sound.

Currently the best that a tube compressor has to offer worldwide!

Final Thoughts

The SPL Iron represents a pinnacle in analog compression technology. It's a testament to the enduring value of high-quality hardware in an increasingly digital audio world. While its price point puts it out of reach for many, for those operating at the highest levels of audio production, it offers a level of sound quality and control that justifies the investment.

However, it's important to note that no single piece of equipment is a magic bullet. The SPL Iron, as impressive as it is, is ultimately a tool. Its true value lies in the hands of skilled engineers who can leverage its capabilities to enhance their productions. For those with the skills to use it and the projects that demand its level of quality, the SPL Iron is an exceptional choice that can provide that elusive finishing touch to world-class audio productions.

TEST: SPL Gain Station

 Alright, I'll come clean! I, too, was once ignorant, almost foolish!

I vividly remember my early twenties when I was purely an instrumentalist. Various sound engineers would go on and on about their "outboard gear worship," which frankly got on my nerves.

Back then, there was endless chatter and debate about the appropriate input channel and the associated signal path. In my estimation at the time, all this 19-inch rack talk was completely over the top. I preferred to focus on my guitar, indulging instead in equally exaggerated discussions about the vibration characteristics of one-piece versus two-piece mahogany bodies. As I said, I had no clue...

It wasn't until years later, when I first occupied the responsible producer's chair and was often entrusted with the role of sound engineer, that I recognized the truly crucial importance of the first signal stage. The reality is, what you don't perfectly process or even mess up in the first recording stage, you won't be able to salvage even remotely in the subsequent stages with EQ, dynamics, or mixing.

Therefore, it indeed makes sense to acquire a preamplifier whose sole function is optimal gain boosting. The rest of the signal chain will thank you for it.

One of these purist devices is the SPL Gain Station, which, with its parallel transistor and tube concept, addresses the eternal dispute between semiconductor and vacuum tube with an additional option.

## Construction

Unlike many other preamplifiers, the Gain Station 1 doesn't come in a 19-inch rack format. Instead, the product was conceived as a mobile unit for space-saving transport in a 4.525-inch width and can be rack-mounted if needed by using four Gain Stations and an optionally available frame.

Due to its reduced size, the preamplifier can be used close to the microphone, for example, keeping cable paths short. Two sturdy carrying handles facilitate portability.

The Gain Station is a discretely built operational amplifier using Class A technology, where both output transistors always remain in a conductive state, as opposed to Class B technology, where each transistor handles a half-wave.

The tube type used is a 12AX7 LPS from the Russian tube specialist Sovtek.

SPL has placed particular emphasis on a high-quality power supply in the Gain Station, a point that is often undervalued. The transformer provides seven different voltages, all individually filtered and regulated.

## Rear Panel

The rear panel presents a spartan but functional mono setup. Besides the IEC power connector and the on/off switch, the Gain Station 1 features an XLR microphone input, a line jack input, a balanced XLR output, and a balanced jack output.

The Gain Station is also available with an integrated 24/96 AD converter module, which in this case also has additional digital optical and SPDIF inputs.

## Front Panel

The front panel of the Gain Station features 3 knobs, 6 mini-switches, and 9 LEDs, whose functions are as follows:

1. Clean Gain: Determines the pre-amplification of the Class-A transistor stage, with a control range up to +63 dB.

2. Tube Gain: Determines the pre-amplification of the tube stage. This stage is behind the transistor pre-amplifier stage, so the two gain values add up.

3. Output Level: Self-explanatory. Control range from -26 dB to +6 dB.

4. Source: Serves to select the input source (microphone or jack).

5. High Pass: Low-cut filter at 50 Hz with a slope of 12 dB (6 dB before and 6 dB after the Clean Gain stage).

6. Phase: Inverts the polarity of the microphone signal.

7. Imped.: Provides a pre-selection for the input impedance of the chosen microphone type (dynamic / condenser).

8. Phantom: Activates the 48V phantom power.

9. Limiter: Activates two different limiter types when needed (Peak: diode limiting for fast response and subtle level limiting, FET: field-effect transistors for the tube stage for compressor-like limiting).

## In Practice

Upon activating the device, the almost imperceptible level of background noise is very positive. Even with the controls turned up high, the noise level is so low that recording very quiet signal sources is not a problem.

First, the Clean Gain is subjected to a practical test. A very even pre-amplification and a soft reproduction without any coloration leave a very good impression. Even with explosive sounds or signal sources with high dynamics, the preamplifier has enough headroom to avoid "squashing" the signal or providing it with clipping.

However, the exact opposite is hoped for from a tube preamp, which ideally offers a powerful processing in terms of dynamics and saturation with a high degree of character to the signal source.

And lo and behold, Tube Gain doesn't disappoint and adds a very precisely dosable saturation to the fed signal, which will trigger real storms of joy among the followers of tube technology.

In combination with Clean Gain, almost all coloration and saturation levels are available, be it a tonally neutral booster amplifier whose use is ideally not noticeable, or a beefy, gritty tube preamp that comes close to a guitar preamp in terms of saturation and compression when fully loaded. Rarely have I heard a preamplifier that offers so much flexibility.

The tonal variety of the limiter located in the final stage is also very interesting. While the diode limiting indeed only eliminates small peaks and keeps the sound image very open, the FET switching process compresses the sum considerably.

The result actually reminds more of a compressor set to hard knee than the usually hard half-wave cut of a limiter. I really like it very much, although the factory preset might be set too tight for some users.

## Expanded Features and Applications

### Microphone Preamplifier

As a microphone preamplifier, the Gain Station 1 excels in its ability to capture the nuances of various microphone types. The selectable input impedance is particularly useful when working with different microphones, as it allows you to optimize the interaction between the mic and the preamp.

For condenser microphones, the high impedance setting ensures maximum voltage transfer, resulting in a clear and open sound. For dynamic microphones, the lower impedance setting can help to dampen resonances and provide a smoother frequency response.

The clean gain stage, with its impressive 63 dB of gain, is more than enough for even the most demanding low-output microphones, such as ribbon mics. The low noise floor ensures that even at high gain settings, you're hearing more of your source and less of the preamp.

### DI Box for Instruments

The Gain Station 1 also serves as a high-quality DI (Direct Input) box for instruments. This is particularly useful for bass players, acoustic guitarists, and keyboardists who want to send their signal directly to a mixing console or recording interface.

The tube stage can be particularly effective here, adding warmth and harmonic richness to direct sources that might otherwise sound thin or sterile. For example, a DI'd acoustic guitar can benefit from a touch of tube saturation to emulate the sound of a miked-up acoustic guitar amp.

### Reamping Tool

While not its primary function, the Gain Station 1 can also be used effectively in reamping scenarios. Reamping involves taking a recorded DI signal and sending it back through an amplifier or other processor to capture a new sound.

The Gain Station's line input can accept the recorded DI signal, and its tube stage can be used to add character before sending the signal to an amplifier. This can be particularly useful for adding analog warmth to digitally recorded tracks.

### Mastering Applications

Although primarily designed as a recording tool, the Gain Station 1 can find use in mastering applications as well. Its clean gain stage can be used for precise level adjustments, while the tube stage can add subtle harmonic enhancement to a full mix.

The limiter, particularly the FET-based option, can be used for gentle peak control during mastering. However, it's worth noting that the lack of precise threshold and ratio controls means it may not be suitable as a primary mastering limiter.

## Sound Character

The Gain Station 1's dual-path design allows for a wide range of tonal options. The clean gain path is remarkably transparent, adding gain without coloring the sound. This makes it ideal for situations where you want to capture the pure sound of your source, be it a high-end condenser microphone or a vintage instrument.

The tube path, on the other hand, can range from subtle warmth to rich, saturated tones. At lower gain settings, it adds a gentle thickness to the midrange and a slight softening of transients. As you push it harder, you'll hear more even-order harmonics, a slight compression effect, and that classic "tube sound" that can help sources sit better in a mix.

What's particularly impressive is how well these two paths integrate. You can blend them to taste, allowing for precise control over how much "character" you want to add to your signal. This flexibility makes the Gain Station 1 suitable for a wide range of sources and musical styles.

## Comparison with Other Preamps

When compared to other preamps in its class, the Gain Station 1 holds its own in terms of sound quality and flexibility. Its dual-path design sets it apart from many single-circuit preamps, offering more tonal options in a single unit.

Compared to vintage-style tube preamps, the Gain Station 1 offers cleaner, more controlled tube coloration. It's less about emulating a specific "classic" sound and more about providing a modern, versatile tool that can adapt to various recording scenarios.

Against solid-state competitors, the Gain Station 1 stands out with its tube option and its remarkably clean transistor circuit. Many solid-state preamps introduce some coloration, even in their "clean" mode, whereas the Gain Station 1's clean path is genuinely neutral.

## Integration in Modern Recording Setups

Despite being an analog device, the Gain Station 1 integrates well into modern, digital-centric recording setups. Its compact size makes it easy to keep on a desktop next to a computer and audio interface.

For those working primarily "in the box," the Gain Station 1 can serve as a high-quality front end, allowing you to capture the best possible signal before it enters the digital domain. This is particularly valuable in an era where many producers and engineers rely heavily on software processing; starting with a great analog sound can reduce the need for extensive digital treatment later.

The optional AD converter module further enhances its integration into digital workflows, allowing for a completely analog signal path up to the final digital conversion stage.

## Potential Improvements

While the Gain Station 1 is an excellent preamp, there are a few areas where improvements could be made:

1. Metering: The LED indicators provide basic level information, but a more detailed meter could be helpful for precise gain staging.

2. Output Transformer Option: Some users might appreciate the option of an output transformer for additional tonal shaping and galvanic isolation.

3. Variable High-Pass Filter: While the fixed 50 Hz high-pass filter is useful, a variable filter could provide more flexibility in dealing with low-frequency issues.

4. Stereo Linking: For stereo recording applications, the ability to link two units for matched stereo operation could be beneficial.

## Conclusion

The Gain Station 1 leaves an excellent impression! Both its concept and its sound are absolutely exemplary.

Due to the high quality of the individual components and the optional transistor or tube pre-amplification, the Gain Station offers a wide range of signal-shaping sounds.

Thus, the product can be used both as a pure microphone preamplifier and as a high-quality instrument preamp. Bassists, for example, can feed the signal directly into their power amp, while keyboardists or acoustic guitarists have a deluxe DI box with the Gain Station.

It's a highly flexible product that, due to its compact design, can fit almost anywhere. Highly recommended for both professional studios and home recording enthusiasts who want to elevate the quality of their recordings.

The SPL Gain Station 1 proves that sometimes, less is more. Its streamlined design and focus on core functionality result in a preamp that excels at its primary task: capturing and enhancing audio signals with clarity and character. Whether you're recording vocals, instruments, or full ensembles, the Gain Station 1 provides the tools to shape your sound at the source, setting the stage for a superior final product.