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.

TEST: SPL Dynamxx 9735

 Let's start with the question of the day. Apart from a fully parametric equalizer, which processor component requires the most learning and familiarization time to achieve practical results? That's right, we're talking about the compressor!

I vividly recall how overwhelmed I was at the beginning of my career when it came to correctly handling an ADSR compressor. The time components - Attack, Decay, Sustain, and Release - were a complete mystery to me in their complexity. In my early days, this often led to significant setting errors, punishing the production with massive pumping or the breakthrough of fast transients.

A perfectly adjusted hardware compressor is a powerful tool and indispensable in our over-compressed music landscape. However, when equipped with incorrect time components, it can mutilate a transparent mix into an amputated sound experience, robbing each individual track of its identity and assertiveness.

What could be more logical than providing the ambitious user with a tool that combines the aforementioned ADSR technique into an automatism with practical values, without having to deal with presets as used in the software sector with varying degrees of quality?

SPL has developed such a hardware solution with the DynaMaxx and is setting out to master the complex mechanism with a minimum of adjustment options.

## Construction and Design

The DynaMaxx comes in a 1U rack unit with a depth of 23.7 cm and weighs 3.4 kg. It's the first compressor to work with two THAT 2181 VCAs in parallel, known for their particularly neutral and low-distortion sound.

Visually, the device is housed in an attractive blue-black finish. This stereo unit manages to handle not only the ADSR functions but also a noise gate with just three rotary controls and four push buttons per channel.

The noise gate features an ARS (Auto-Release-Circuit) and is automated like the compressor. It's designed to provide a particularly smooth fade-out. The ARS circuit "remembers" the level jump between the music signal level and the threshold set via the noise gate control.

If the difference is large, a fast release time is set; if it's small, slower release times are used.

Furthermore, the DynaMaxx features a de-compression circuit to invert the compressor's function. This function gives highly compressed sampler sounds, for example, a more dynamic sound image by working back their loudness amplitude.

## Control Panel Layout

The individual controls and push buttons on the control panel are as follows:

1. Compress: This control simultaneously adjusts threshold and ratio, based on a soft-knee characteristic. An LED placed above and to the left of the control provides information about the operation.

2. Gain: This control compensates for the loss of volume that accompanies the compression process.

3. Noise Gate: Adjusts the activation point of the noise gate.

4. Soft Limit: In this function, the DynaMaxx only processes the peak levels. Unlike the compressor function, it doesn't boost quiet signal components.

5. Effect Compression: Deactivates the automation of the release time and sets a very short value of 60 milliseconds.

6. De-Compression: Activates the de-compression circuit.

7. Active: Switches off the DynaMaxx via a relay hard bypass circuit.

8. LED Display: Two 20-segment LED chains provide information about the set compression value and the associated level reduction in 1 dB resolution.

A stereo couple switch is placed in the center, allowing the device to receive the same control voltage in stereo operation or to operate as 2x mono.

On the rear, the DynaMaxx features balanced inputs and outputs in XLR and jack format. Additionally, there's a side chain input implemented as a stereo jack socket, where filtered or triggered signals can be looped in to externally trigger the DynaMaxx.

## Practical Application

In today's audio landscape, compressors must assert themselves in both instrumental recording and sum processing, which is made more difficult by the fact that both output signals have completely different attack and decay phases.

While the usually highly condensed sum signal of a mix particularly demands uniform smoothing of the peak levels and must not suffer in terms of transparency when loudness is added, a sensitive individual recording requires particularly individual attack and release processing to prevent the notorious pumping, which is still the main problem of correct compressor setting.

Indeed, during the first configuration (single instrument / electric bass), I only had to find a tastefully sensible starting point using the Compress control; everything else actually happened as if by magic all by itself.

How often have I needed 10-15 minutes alone to perfectly adjust a compressor to the respective playing style of the artist? All work that can now be done within a minute.

It's quite astonishing how practically the device manages even complex dynamic processes. Even through high volume fluctuations and "undisciplined" playing, the DynaMaxx doesn't get confused and stays dynamically on course. Just quickly find a suitable value for the noise gate, and you're done!

When it comes to sum processing, things are similarly quick. Despite completely different source material, the compressor leaves a very balanced impression and shines with the simplest handling.

Indeed, the VCAs used deliver a highly transparent sound that at no point tonally alienated the source material or provided it with other artifacts. A well-rounded package!

## In-Depth Analysis of DynaMaxx's Features

### Automatic Time Constant Circuit

One of the standout features of the DynaMaxx is its automatic time constant circuit. This innovative technology continuously analyzes the input signal and adjusts the attack and release times accordingly. This results in a more natural and musical compression that adapts to the source material in real-time.

The automatic circuit is particularly beneficial when working with complex material that has varying dynamics throughout. For instance, when compressing a full mix or a drum bus, the DynaMaxx can seamlessly transition between fast attack times for transient-rich sections and slower attack times for sustained passages, all without user intervention.

### Parallel VCA Design

The use of two THAT 2181 VCAs in parallel is a unique approach that contributes significantly to the DynaMaxx's clean and transparent sound. This design allows for more headroom and lower distortion compared to single VCA designs.

The parallel VCA configuration also enables the DynaMaxx to handle a wide range of input levels without introducing unwanted coloration or artifacts. This makes it equally suitable for subtle compression on delicate sources like vocals or acoustic instruments, as well as more aggressive compression on high-energy sources like drums or distorted guitars.

### De-Compression Function

The de-compression function is an intriguing feature that sets the DynaMaxx apart from many other hardware compressors. This function essentially inverts the compressor's action, expanding the dynamic range instead of reducing it.

This can be particularly useful in several scenarios:

1. Restoring dynamics to over-compressed material: In the age of the loudness war, many recordings suffer from excessive compression. The de-compression function can help breathe life back into these recordings by expanding their dynamic range.

2. Creating space in a dense mix: By selectively expanding certain elements in a mix, you can create more separation and depth, allowing each instrument to occupy its own space more clearly.

3. Creative sound design: The de-compression function can be used to create unique envelope effects, particularly on percussive sounds or synthesizers.

### Soft Limiting

The soft limiting function of the DynaMaxx provides a gentle way to tame peaks without affecting the overall dynamics of the signal. This is achieved through a gradually increasing ratio as the input signal approaches the threshold, rather than a hard knee that abruptly applies full compression.

This soft limiting can be particularly useful in mastering situations or when processing the main mix bus. It allows for increased perceived loudness without the harshness often associated with hard limiting.

## Performance in Various Scenarios

### Vocal Processing

When used on vocals, the DynaMaxx shines in its ability to maintain consistency without robbing the performance of its natural dynamics. The automatic time constant circuit proves particularly useful here, adapting to the varying intensities of a vocal performance without requiring constant tweaking.

The soft knee characteristic of the compression curve allows for a smooth transition into compression, which is ideal for vocals as it helps maintain a natural sound even when applying significant gain reduction.

### Drum Bus Processing

On a drum bus, the DynaMaxx demonstrates its versatility. The fast attack times available allow for transient shaping, tightening up the impact of kicks and snares. Meanwhile, the automatic release times prevent pumping artifacts that can often occur when compressing complex drum patterns.

The parallel VCA design comes into its own here, maintaining the clarity and punch of the drums even under heavy compression. The soft limiting function can be particularly useful for taming occasional peaks without squashing the overall drum dynamics.

### Bass Guitar

Bass guitar is notoriously difficult to compress due to its wide dynamic range and the importance of maintaining both low-end weight and upper-mid definition. The DynaMaxx handles this challenge admirably, thanks to its clean VCA design and intelligent auto-release function.

The compression maintains the body of the bass notes while evening out the performance, and the de-compression function can be creatively employed to enhance the attack of plucked notes for added definition in a mix.

### Mix Bus Application

When used across a full mix, the DynaMaxx proves to be a powerful yet subtle tool for adding cohesion and enhancing the overall energy of a track. The automatic time constant circuit really proves its worth here, handling the complex dynamics of a full mix with ease.

The soft limiting function is particularly useful in this context, allowing for a final stage of peak control without introducing obvious compression artifacts. This can help achieve commercial loudness levels while maintaining the mix's dynamic integrity.

## Comparison with Software Emulations

While software plugin emulations of classic compressors have come a long way in recent years, the DynaMaxx offers several advantages over its digital counterparts:

1. Zero latency: As a hardware unit, the DynaMaxx processes audio in real-time without introducing any latency. This is particularly beneficial in live sound applications or when tracking.

2. Analog warmth: Despite its clean and transparent design, the DynaMaxx imparts a subtle analog character to the signal that many find pleasing. This is often described as adding "glue" to a mix.

3. Tactile control: The physical knobs and buttons of the DynaMaxx offer a level of intuitive control that's hard to replicate with a mouse and screen. This can lead to faster, more musical decisions during mixing.

4. Reliability: Hardware units like the DynaMaxx are not subject to software crashes, compatibility issues, or the need for constant updates.

## Integration into Modern Workflows

Despite being a hardware unit, the DynaMaxx integrates seamlessly into modern, hybrid mixing setups. Its balanced XLR and TRS connections allow for easy integration with professional audio interfaces.

For those working primarily "in the box," the DynaMaxx can be used as an outboard processor via the aux sends of a DAW. This allows for the best of both worlds - the flexibility of digital mixing combined with the sound quality and hands-on control of high-end analog hardware.

The side-chain input also opens up creative possibilities when integrated with a DAW. For instance, you could send a filtered version of a kick drum to the side-chain input to create ducking effects on a bass track, all while benefiting from the DynaMaxx's superior analog processing.

## Conclusion

The SPL DynaMaxx proves to be a true dynamic all-purpose weapon. It provides excellent service, especially when the interested user still lacks the necessary experience or knowledge in the difficult-to-penetrate ADSR jungle, or when things need to move very quickly, for example in the live sector.

Never before have I managed to add loudness gain to even dynamically highly complex material in such a short time. All controls are practically self-explanatory and require no extensive familiarization time.

Due to its sophisticated components and very good coordination with each other, one would have to go to great lengths to coax a dynamically unbalanced sound out of the Model 9735. I certainly didn't succeed during the test operation.

The handling is very clear and easy to learn even for the inexperienced user, so that after completing "dynamic" basic training, one can gradually dare to tackle manual time component management.

A high-quality product with very high practical value! Highly recommended.

TEST: Fischer Amps FA 666 XB

 The term "In-Ear Monitoring System" is likely familiar to many readers. I would even go so far as to say that the term "Wedges" might eventually become less common as in-ear monitoring gains more popularity.

In recent years, what was once reserved for the high-end segment is now available across various price and quality ranges. However, when it comes to professional use, two German companies are frequently mentioned, much like in the wireless microphone sector.

Apart from the company Inear Monitoring from Dieburg, the name Fischer Amps often comes up. Like their competitors, Fischer Amps develops and manufactures exclusively in Germany, which is reflected in both the quality and the price. The Fischer Amps FA 666 XB, which we have for review, is priced at 689 euros. Some might recognize the name from their distribution of Ultimate Ears earphones.

In the in-ear segment, we generally distinguish between two different systems. One is universal earphones, which are based on an average fit designed to accommodate a wide range of ear shapes. The second option is a custom-fit system, where one can have an impression of their ear made either at the manufacturer or a hearing aid specialist, resulting in earphones perfectly tailored to the individual's ear. Very cheap in-ear systems might simply have a regular plug that is inserted into the ear with a rubber tip.

### Advantages of Custom Systems

The advantages of a custom-fit system are clear. By completely sealing off the ear canal, one can effectively block out external noise if needed. The system fits very securely, does not wobble, and generally offers better sound quality, especially in the bass range. Universal systems, on the other hand, are typically less expensive since custom fitting, much like a tailored suit, comes with a higher price tag.

The Fischer Amps FA 666 XB is a universal earphone but comes with a hefty price tag of nearly 700 euros. This suggests that its sound quality and fit are likely far superior to what a typical universal earphone can offer. Let's find out.

### Construction of the Fischer Amps FA 666 XB

The Fischer Amps FA 666 XB comes in a sturdy, small case, including a cleaning tool and four pairs of silicone tips in sizes S, M, L, and double-flange. Additionally, there are three pairs of foam tips, also in sizes S, M, and L. The earphones are equipped with six balanced armature drivers: 2x Bass, 2x Mid, and 2x High, with an emphasis on the bass range, as indicated by the product name.

The earphones feature a somewhat peculiar plastic body designed to conform to the inner ear, to which the detachable cable is connected. The cable is 140 cm long and ends in a gold-plated mini stereo jack. The connectors between the cable and the earphones are slightly angled but do not have the stiff plastic guide found in other comparable models. Fischer Amps uses a highly flexible plastic guide that provides only a slight direction for the cable, enhancing comfort.

The earphones are best inserted into the ear by placing them on the ear and then rotating them into the ear canal. The cable is designed to withstand a force of 800 Newtons before breaking and is very durable in terms of kink resistance.

### Impedance and Practical Use

The impedance of the earphones is 14 ohms at 100 Hz, 16 ohms at 1 kHz, and 21 ohms at 10 kHz. These values indicate that the earphones have very low impedance, allowing them to maximize volume from any wireless receiver. However, this also means that one must be extremely cautious when adjusting the output volume to avoid hearing damage.

### The Fischer Amps FA 666 XB in Practice

When evaluating the quality of in-ear monitoring systems, two factors are crucial: sound quality and comfort. Sound quality is significantly influenced by how well the earphones fit, particularly in the bass range, as any small opening can greatly reduce the volume the earphones can produce.

I cannot definitively say how average my ear canal is shaped, but I had no trouble inserting the Fischer Amps earphones. When inserted as recommended, by rotating them into the ear from front to back, the plastic bodies naturally settle into the optimal position within the ear.

The fit is very secure, with minimal movement, only a few steps away from a custom-fit system. Even with vigorous head movements, the earphones stayed firmly in place. Other components of the system, such as the twisted, durable cable and the mini-jack plug, are of the highest quality, indicating a top-tier product.

Sound-wise, it is immediately apparent that the system is not designed for linearity but is optimized for specific instrument groups. Bassists, drummers, and DJs, who require enhanced bass, will appreciate this system for its clear bass boost.

One could roughly say that everything from the low mids to the highest highs is relatively neutral and airy, while the bass range receives a significant boost, making it overly present. Those seeking a completely linear sound might not find this system suitable, but those who prefer a rich bass will be well-served.

I had the opportunity to compare the system with another Fischer Amps model, the same system without the XB designation, indicating no bass boost. The non-XB version provides a more even and linear sound. Depending on personal preferences, one can choose whether they want a higher bass presence or not. Personally, I find the enhanced bass more enjoyable for casual listening due to its comforting character, but the linear version is better for mixing or if one is already exposed to strong bass frequencies on stage.

### Extended Information on Fischer Amps FA 666 XB

Beyond the fundamental aspects of fit and sound, it's important to consider the broader capabilities and design features that contribute to the Fischer Amps FA 666 XB's performance.

#### Durability and Build Quality

The Fischer Amps FA 666 XB is built to last, with a rugged construction that can withstand the rigors of frequent use. The reinforced cable and robust connectors ensure that the earphones remain reliable even under heavy use. The gold-plated mini-jack plug not only provides a secure connection but also contributes to the earphones' overall longevity by resisting corrosion.

#### Additional Accessories

The inclusion of multiple silicone and foam tips in various sizes ensures that users can find the most comfortable and secure fit for their ears. This versatility is particularly beneficial for performers who need their earphones to stay in place during energetic performances. The cleaning tool provided helps maintain the earphones, ensuring they remain hygienic and functional over time.

#### Sound Isolation

The Fischer Amps FA 666 XB excels in sound isolation, an essential feature for in-ear monitors. By effectively blocking out external noise, these earphones allow performers to hear their mix clearly without needing to increase the volume excessively. This feature is crucial for protecting hearing health, especially for musicians who are exposed to high sound levels regularly.

#### Sound Signature and Tuning

The sound signature of the Fischer Amps FA 666 XB is tailored for those who demand powerful bass response without sacrificing clarity in the mid and high frequencies. The balanced armature drivers are specifically tuned to deliver a detailed and immersive listening experience, making these earphones suitable for a wide range of musical genres and performance settings.

#### Customization and Personalization

While the Fischer Amps FA 666 XB is a universal fit model, the variety of tips provided and the ergonomic design allow for a degree of customization. For users who seek an even more personalized experience, Fischer Amps offers custom-fit options that can be molded to the exact shape of the ear, providing the ultimate in comfort and sound isolation.

#### Versatility in Use

These earphones are not only ideal for live performances but also for studio monitoring and casual listening. Their detailed sound reproduction makes them suitable for critical listening, allowing users to pick up on subtle nuances in the music. The robust build quality and reliable performance make them a dependable choice for professional musicians and audio engineers.

#### Comparison with Competitors

In the competitive market of in-ear monitors, the Fischer Amps FA 666 XB stands out for its combination of build quality, sound performance, and user comfort. Compared to other models in its price range, it offers a compelling package that balances premium features with practicality. The focus on bass enhancement makes it a unique offering, particularly appealing to performers who rely heavily on low-frequency sounds.

#### Conclusion

With the Fischer Amps FA 666 XB, the German manufacturer has introduced a highly competitive product in the universal in-ear monitor market. The system is meticulously crafted, offers outstanding sound quality, and is well-suited for a variety of live performance scenarios.

For those in search of a high-quality, durable, and sonically impressive in-ear monitor, the Fischer Amps FA 666 XB is certainly worth considering. Its robust construction, excellent sound isolation, and powerful bass response make it a versatile tool for both professional and personal use.

Testing the Fischer Amps FA 666 XB is highly recommended for anyone seeking to enhance their in-ear monitoring experience.

### Detailed Examination of the Fischer Amps FA 666 XB

#### In-Depth Sound Analysis

The Fischer Amps FA 666 XB delivers a sound profile that caters to both performers and audiophiles. Its six balanced armature drivers—two dedicated to bass, two to mids, and two to highs—provide a detailed and immersive listening experience. The bass boost, as indicated by the XB (Extended Bass) in its name, significantly enhances low-frequency performance, making it particularly suitable for bassists, drummers, and electronic music performers.

The mids are well-defined, ensuring that vocals and mid-range instruments are clearly articulated. The highs are crisp without being harsh, providing a balanced soundstage that is both wide and deep. This careful tuning allows users to hear every nuance in their performance or mix, making the FA 666 XB an invaluable tool for critical listening.

#### Comfort and Fit

One of the standout features of the Fischer Amps FA 666 XB is its ergonomic design. The earphones are shaped to fit comfortably within the ear canal, providing a secure fit that minimizes the risk of falling out during use. The variety of silicone and foam tips included in the package ensures that users can find the perfect fit for their ears, further enhancing comfort and sound isolation.

The lightweight design of the earphones, combined with the flexible plastic guide on the cable, means that users can wear them for extended periods without discomfort. This is particularly important for musicians and performers who may need to wear in-ear monitors for the duration of a concert or recording session.

#### Practicality and Usability

In addition to sound quality and comfort, practicality is a key consideration for in-ear monitors. The Fischer Amps FA 666 XB excels in this area with features that make it easy to use in a variety of settings. The detachable cable is a significant advantage, allowing users to replace it easily if it becomes damaged. This extends the lifespan of the earphones and reduces the need for costly repairs or replacements.

The cable's 140 cm length is ideal for most uses, providing enough length to connect to a bodypack transmitter or a headphone amplifier without excess slack that could get in the way. The gold-plated mini-jack plug ensures a secure connection and minimizes signal loss, contributing to the overall audio quality.

#### Build Quality and Durability

Fischer Amps has a reputation for producing high-quality, durable products, and the FA 666 XB is no exception. The earphones are built to withstand the rigors of professional use, with a sturdy plastic body and a robust cable that can endure significant stress. The earphones are designed to handle a force of up to 800 Newtons, ensuring that they remain reliable even under heavy use.

The inclusion of a hard carrying case adds an extra layer of protection, making it easy to transport the earphones without fear of damage. This is particularly useful for touring musicians who need to carry their gear with them from one venue to the next.

#### Versatility Across Different Scenarios

While the Fischer Amps FA 666 XB is designed with professional use in mind, its versatility makes it suitable for a range of applications. In the studio, the detailed sound reproduction and balanced armature drivers make these earphones ideal for mixing and mastering. The enhanced bass response is particularly beneficial for genres that rely heavily on low-frequency sounds, such as hip-hop, EDM, and rock.

For live performances, the sound isolation provided by the custom-fit tips ensures that performers can hear their mix clearly, even in noisy environments. This allows for more accurate pitch and timing, leading to better overall performances.

#### Comparison with Custom-Fit Monitors

Custom-fit monitors are often seen as the gold standard in in-ear monitoring, offering a perfect fit and superior sound isolation. While the Fischer Amps FA 666 XB is a universal fit model, it comes remarkably close to the performance of custom-fit monitors. The variety of tips included allows users to achieve a fit that is almost as secure and comfortable as a custom mold.

The sound quality of the FA 666 XB is also comparable to many custom-fit models, thanks to its high-quality drivers and careful tuning. This makes it an excellent choice for those who want the benefits of custom-fit monitors without the higher price tag and the need for ear impressions.

#### User Experience and Feedback

Feedback from users of the Fischer Amps FA 666 XB has been overwhelmingly positive, with many praising the sound quality, comfort, and durability of the earphones. Musicians and audio professionals alike have noted the clear, detailed sound and the significant bass enhancement, which is particularly useful for monitoring low-frequency instruments and sounds.

The comfort and fit of the earphones have also been highlighted as major advantages, with users reporting that they can wear the FA 666 XB for extended periods without discomfort. The robust construction and reliable performance have made these earphones a favorite among touring musicians and those who require dependable gear for professional use.

#### Enhancements and Improvements

While the Fischer Amps FA 666 XB is already a top-tier product, there are always potential areas for improvement. Future iterations could benefit from wireless connectivity options, providing even greater flexibility for users. Additionally, incorporating more advanced noise-cancellation technology could further enhance the sound isolation, making the earphones even more effective in noisy environments.

Another possible enhancement could be the inclusion of more customization options, allowing users to fine-tune the sound profile to their specific preferences. This could be achieved through a companion app or inline controls that offer adjustable EQ settings.

### Conclusion

The Fischer Amps FA 666 XB represents a significant achievement in the field of in-ear monitoring. Its combination of high-quality sound, comfortable fit, and durable construction makes it an excellent choice for both professional and personal use. Whether you're a touring musician, a studio engineer, or simply an audiophile looking for top-tier earphones, the FA 666 XB delivers performance that meets the highest standards.

Testing the Fischer Amps FA 666 XB is highly recommended for anyone in search of a reliable, high-performing in-ear monitoring solution. Its robust design, outstanding sound quality, and versatility make it a standout choice in a competitive market.