TEST: SPL Charisma 9733

 OK, guys, let's get straight to the point. Who among you has ever had the pleasure of recording an audio cassette on a tape deck?

Yes, the ranks thin considerably among the interested readers. Why do I start with such an analog introduction in the age of bits and bytes?

Well, unlike digital recording, where pushing the levels towards the 0 dB mark aims for maximum transparency and information density, analog recording has a highly extravagant life of its own beyond the (supposed) maximum level.

As an annoying teenager, I discovered more by accident that my vinyl to cassette transfers sounded significantly "better" when the first 1-2 red bars of the (then brand-new) LED meter lit up during loud passages (back when productions weren't yet dynamically over-compressed).

The saturation of the magnetic tape at the time led to an increase in loudness and a thickening of the signal, and young Axel had initiated his first homemade "mini-mastering process," not knowing what he was doing or why the result was so pleasing to the human ear.

Based on this analog sound processing and the associated signal alteration, recording technology has been trying for decades to achieve a fusion of the two preservation methods.

No one wants (or can!) do without the editing capabilities of digital technology nowadays, but any serious sound engineer moves close to audiophile euphoria as soon as they get to work on a valuable signal with correspondingly high-quality analog hardware, at least during the recording phase.

One product from this category is the Charisma from SPL, which generates the analog tape saturation effects and harmonic distortion behavior of tubes to breathe life into naturally sterile digital signals.

#### Construction

The Charisma 2 comes in a 1U rack format, with a depth of 23.7 cm and a weight of 3.5 kg.

Peering through the side vents, one can see that the Charisma doesn't rely on massive CPU logarithms for its processing function. Instead, two genuine preamp tubes (apparently of the 12AX7 type) do the work inside.

SPL secured the services of tube specialist Manfred Reckmeyer for this processor, who has earned an excellent reputation as an amp tuner among guitarists. He even completely rebuilt a '69 Marshall head for me in 1992.

Apart from the characteristically brass-colored front panel, the processor appears quite unspectacular visually, with each channel featuring only three knobs, two LEDs, and a bypass switch.

The functionality of the knobs in detail:

- **Drive:** Adjusts the drive of the tube electronics. The higher the value, the greater the number of harmonic overtones and the stronger the limiting effect.
- **Charisma:** Varies the saturation behavior of the tube. Adjustable from "Soft" (soft, slow, slightly damped peaking of harmonic structures) to "Hard" (rather late onset of harmonic distortion, but very direct at the level similar to a hard limiter).
- **Output:** Controls the output volume and serves to match the following device.

- **Process LED:** Lights up as soon as the device produces a certain percentage of harmonics.
- **Max LED:** Indicates the upper limit of processing.

On the back of the device, each channel features balanced inputs and outputs in the form of jack and XLR connectors. Additionally, there is a ground lift switch and an IEC power socket.

#### Practical Application

If you want to maximize the sound yield, you must accept that the "gods of saturation" have cursed you with the gift of a keen ear and intense training.

Especially sound engineers who haven't yet dealt with the analog life of their signals should take the time to familiarize themselves with the admittedly straightforward three knobs and the available output material.

You wouldn't believe how much the sound result varies only due to the increasing or decreasing harmonic waves and the set saturation behavior.

Moreover, the compression behavior changes radically. While the increase in the drive range can be compared to a (still dynamic) limiting, low charisma values produce noticeable compression effects that can generate characteristic pumping effects depending on the source material.

Depending on the dynamic behavior of the source material, you must decide whether you want to thicken the signal, compress it, or add distortion. It's essential to find the right balance here, keeping in mind how the signal should be placed in the mix later on.

Synthetic sounds from the keyboard sector benefit immensely from a high portion of archaic harmonic behavior. A "pig organ" in the old Deep Purple tradition really comes into its own here.

The Charisma sounds so fantastic that you might be tempted to compress every signal to the max, so that the material sounds excellent on its own but must be significantly lowered in volume in the mix due to its high sound density to avoid drowning out all other signals.

More than usual, a good producer is needed here, who has the necessary foresight of the final sound at the decisive moment and ensures a well-dosed, fine-tuning within the arrangement.

#### Extended Details

The Charisma processor is not just a simple tool but a versatile device that significantly enhances the sound quality of various instruments and recordings. Its ability to add warmth, character, and richness to digital recordings makes it an indispensable tool in modern music production. Here are some additional aspects and tips on how to get the best out of this device.

##### Integration in the Studio

Integrating the Charisma into your studio setup can be done in multiple ways. For example, it can be used on individual tracks during the recording phase, on buses for group processing, or even on the master bus for overall mix enhancement. Each approach has its own benefits:

- **Individual Tracks:** Applying the Charisma to individual tracks allows for precise control over each instrument's saturation and harmonic content. This can be particularly useful for vocals, guitars, and other lead instruments where character and presence are crucial.
- **Group Buses:** Using the Charisma on group buses, such as drum buses or background vocals, can glue the elements together, providing a cohesive and harmonically rich sound. This method helps in creating a unified sound for similar instruments or vocal tracks.
- **Master Bus:** Applying the Charisma to the master bus can add a final touch of analog warmth and cohesion to the entire mix. It’s a subtle but effective way to enhance the overall sonic quality of the mix, giving it a professional and polished feel.

##### Detailed Control of Harmonics and Saturation

The Charisma's ability to control the amount and type of harmonics and saturation is a key feature that sets it apart from other processors. Understanding how to use these controls effectively can greatly impact the final sound:

- **Drive Control:** Increasing the drive adds more harmonic distortion, which can make a track sound fuller and more vibrant. However, it’s essential to balance this to avoid unwanted distortion, especially on tracks that already have a lot of harmonic content.
- **Charisma Control:** This control adjusts the type of saturation from soft to hard. A softer setting can add a gentle warmth and roundness, ideal for subtle enhancements. In contrast, a harder setting can introduce a more pronounced and aggressive saturation, suitable for tracks that need more edge and presence.
- **Output Control:** Managing the output level is crucial to ensure that the processed signal matches the level of other tracks in the mix. This helps in maintaining a balanced and cohesive overall sound without having to make significant adjustments in the mix.

##### Practical Tips for Using Charisma

- **Vocals:** Adding a touch of Charisma to vocals can bring them to life, making them stand out in the mix. Start with a moderate drive and a soft saturation setting, then adjust to taste. This can help in achieving a warm and intimate vocal sound.
- **Guitars:** Electric guitars benefit greatly from Charisma's harmonic enhancement. For rhythm guitars, a higher drive and harder saturation can add punch and clarity. For lead guitars, a softer saturation can add sustain and smoothness.
- **Bass:** Applying Charisma to bass tracks can add depth and definition. A moderate drive with soft saturation helps in making the bass more prominent without overpowering the mix.
- **Drums:** On drum tracks, Charisma can add punch and cohesion. Use it on the drum bus with a moderate drive and hard saturation to glue the elements together, giving the drums a solid and unified sound.
- **Synths and Keys:** For synthetic sounds, Charisma can add a layer of analog warmth and complexity. Experiment with different settings to find the right balance that complements the electronic nature of the sounds while adding a touch of organic richness.

##### Advanced Uses and Creative Applications

The Charisma is not limited to conventional audio processing; it also opens up creative possibilities:

- **Parallel Processing:** Use Charisma in a parallel processing chain to blend the saturated signal with the dry signal. This technique allows for a subtle enhancement without losing the original character of the track.
- **Sound Design:** For sound designers, Charisma can be a powerful tool for creating unique textures and timbres. Experiment with extreme settings to discover new and interesting soundscapes.
- **Reamping:** Reamp tracks through Charisma to add a new dimension to previously recorded material. This can be particularly effective for tracks that were recorded digitally and need an infusion of analog warmth and character.

##### Conclusion

The Charisma is a versatile and indispensable tool for any serious producer or sound engineer. Its ability to transform digital recordings with analog warmth and character makes it a must-have in any modern studio. From individual tracks to group buses and the master bus, Charisma enhances the overall sonic quality, bringing life and richness to your recordings.

In summary, the Charisma from SPL is more than just an audio processor; it's a gateway to a richer, warmer, and more vibrant sound. Its intuitive controls and high-quality construction make it an essential addition to any studio, capable of elevating your productions to a new level of excellence. Whether you're working on vocals, guitars, bass, drums, or synths, Charisma offers the tools you need to achieve a professional and polishedsound. Let’s delve even deeper into the various aspects of the Charisma and explore how it can revolutionize your audio production workflow.

### Detailed Exploration of Features

#### Drive Control: Understanding Harmonic Distortion

The Drive control on the Charisma is pivotal in determining the amount of harmonic distortion added to your signal. This feature is particularly useful for adding warmth and depth to otherwise flat or sterile digital recordings. Here’s a breakdown of how to use it effectively:

- **Low Drive Settings:** Ideal for subtle enhancements, low drive settings introduce a gentle harmonic enrichment. This is perfect for acoustic instruments, vocals, and any track where you want to preserve the natural tone while adding a touch of warmth.
- **Moderate Drive Settings:** At moderate levels, the Drive control begins to add noticeable harmonic content, making the signal richer and more vibrant. This setting is excellent for electric guitars, bass, and drums, where a bit of edge can enhance the overall sound.
- **High Drive Settings:** High drive settings push the harmonic distortion to the forefront, creating a pronounced effect that can dramatically alter the character of the sound. This is useful for creative sound design, aggressive guitar tones, or any situation where you want to create a bold, impactful sound.

#### Charisma Control: Tailoring Saturation

The Charisma control allows you to fine-tune the saturation characteristics, providing a range of options from soft, smooth saturation to hard, aggressive distortion. Understanding how to manipulate this control is essential for achieving the desired sound:

- **Soft Saturation:** This setting is akin to the gentle compression and warmth of vintage tape machines. It is ideal for adding a touch of analog smoothness to vocals, strings, and other melodic instruments. The soft saturation ensures that the sound remains natural and pleasing while gaining subtle harmonic richness.
- **Medium Saturation:** With medium saturation, the Charisma starts to introduce more pronounced harmonic content. This is perfect for adding body and presence to instruments that need to stand out in the mix without being overpowering. Think of this setting as adding a bit of “glue” to your tracks, making them sit together more cohesively.
- **Hard Saturation:** At the hard end of the spectrum, the Charisma introduces a much more aggressive form of saturation. This setting is excellent for instruments and sounds that need to cut through the mix with authority. Use this for heavy guitars, bass, or any element where you want a strong, distinct presence.

#### Output Control: Balancing Levels

The Output control is crucial for maintaining the appropriate levels after processing. It ensures that the enhanced signal can be properly integrated into your mix without causing clipping or imbalance. Here’s how to use it effectively:

- **Matching Levels:** After adjusting the Drive and Charisma controls, use the Output control to match the processed signal level to the original. This helps in maintaining a balanced mix and prevents any unwanted jumps in volume.
- **Gain Staging:** Proper gain staging is essential to avoid distortion and maintain headroom in your mix. Adjust the Output control to ensure that the signal remains within an optimal range, preserving the integrity of the sound.
- **Creative Adjustments:** Sometimes, boosting or cutting the output level can be a creative decision. For instance, slightly lowering the output level after heavy saturation can help blend the processed signal more naturally into the mix.

### Practical Applications in Different Scenarios

#### Vocal Processing

Vocals are often the focal point of a mix, and using the Charisma on vocal tracks can make a significant difference. Here’s how to approach vocal processing with the Charisma:

- **Subtle Enhancement:** For a natural and intimate vocal sound, start with a low Drive setting and soft Charisma. This adds warmth and presence without altering the vocal’s natural character.
- **Modern Pop Vocals:** For contemporary pop vocals, where clarity and sheen are crucial, use a moderate Drive setting with medium Charisma. This adds just the right amount of harmonic content to make the vocals shine.
- **Rock Vocals:** For rock or aggressive vocal styles, push the Drive and Charisma controls higher. This introduces more harmonic distortion, adding grit and edge that complement the genre.

#### Guitar and Bass Processing

Guitars and bass are integral to many genres, and the Charisma can enhance their tones significantly:

- **Clean Guitars:** For clean guitar tones, use a low Drive setting and soft Charisma to add warmth and body. This makes clean parts sound fuller and more engaging.
- **Overdriven Guitars:** For overdriven or distorted guitars, increase the Drive and Charisma to medium settings. This enhances the natural distortion and adds richness without making the sound overly harsh.
- **Bass Guitars:** Bass guitars benefit from the Charisma’s ability to add depth and definition. Use a moderate Drive with soft Charisma to make the bass punchy and clear in the mix. For a more aggressive bass tone, increase the Drive and Charisma settings accordingly.

#### Drum Processing

Drums are the backbone of any track, and the Charisma can bring out their full potential:

- **Kick Drum:** Enhance the kick drum with a moderate Drive setting and soft Charisma. This adds punch and depth, making the kick more impactful.
- **Snare Drum:** For snare drums, use a medium Drive and Charisma to add body and snap. This helps the snare cut through the mix while retaining its character.
- **Drum Bus:** Processing the entire drum bus with Charisma can glue the elements together. Use a moderate Drive with soft to medium Charisma settings to add cohesion and warmth to the drum kit.

### Advanced Techniques and Creative Uses

#### Parallel Saturation

Parallel saturation is a powerful technique that combines the processed signal with the original, unprocessed signal. This approach allows for a more subtle enhancement while retaining the original dynamics and character:

- **Setting Up Parallel Chains:** Create a duplicate of the track you want to process and apply Charisma to the duplicate. Blend the processed signal with the original to taste.
- **Balancing Wet and Dry Signals:** Adjust the levels of the processed (wet) and unprocessed (dry) signals to achieve the desired balance. This technique is particularly effective for vocals and drums, where you want to enhance the sound without losing the original dynamics.

#### Creative Sound Design

The Charisma is not just for traditional audio processing; it can also be a tool for creative sound design:

- **Experimental Sounds:** Push the Drive and Charisma controls to their limits to create unique and experimental sounds. This can be useful for electronic music, film scoring, or any project that requires unconventional sounds.
- **Layering Textures:** Use the Charisma to add layers of texture to your tracks. For instance, apply heavy saturation to a copy of a synth pad and blend it with the original to create a rich, complex sound.

#### Reamping Techniques

Reamping is a technique where recorded tracks are played back through an amplifier or processor to add new characteristics. The Charisma is excellent for reamping digital tracks to infuse them with analog warmth:

- **Setup for Reamping:** Route your recorded track through the Charisma and back into your DAW. Adjust the Drive and Charisma settings to taste.
- **Enhancing Recorded Tracks:** This technique is particularly useful for tracks that were initially recorded digitally and lack the warmth and depth of analog recordings. Reamping through the Charisma can transform these tracks, adding a new dimension to the sound.

### Conclusion and Final Thoughts

The SPL Charisma is a remarkable tool that bridges the gap between digital precision and analog warmth. Its intuitive controls, robust construction, and exceptional sound quality make it an invaluable addition to any studio. Whether you are enhancing vocals, guitars, bass, drums, or experimenting with sound design, the Charisma provides the tools to elevate your recordings to new heights.

Investing time in learning and experimenting with the Charisma will undoubtedly pay off, as it becomes an integral part of your production workflow. The ability to add harmonic richness, warmth, and character to digital recordings cannot be overstated, and the Charisma delivers this with finesse and reliability.

In a world where digital clarity often comes at the expense of warmth and character, the Charisma stands out as a beacon of analog goodness. Its subtle yet profound impact on the sound quality makes it a must-have for any producer or sound engineer looking to achieve a professional and polished sound.

TEST: Fischer Amps In Ear Amp 2

 ### Concept of the Fischer Amps In Ear Amp 2

What was once considered a luxury item for personal monitoring has evolved into a standard piece of equipment, even in the most amateur settings. We're talking about in-ear monitoring systems, where earphones are either placed inside or on the ear, allowing musicians to consistently hear the same monitor sound regardless of their position.

While 99% of concertgoers will never understand why a monitoring system is necessary, even fewer will grasp why a drummer—whose instrument is inherently loud—would need one for their drum set. However, this system is designed precisely for these musicians, as well as others whose movement on stage is voluntarily or involuntarily limited, such as keyboardists. The Fischer Amps In Ear Amp 2 is an updated version of the original model, with improved dynamics and enhanced sound and balance controls.

The Fischer Amps In Ear Amp 2 can also be used in a studio setting, though its primary design focus is clearly for live performance. Manufactured in Germany, the amplifier's construction features heavy-duty steel, contributing to its relatively high weight of over 2 kg for a 9.5-inch product. It comes with extension wings for rack mounting in a 19-inch setup.

Despite the high-quality components, the amplifier's layout and readability are not its strongest points. The front panel's silver sparkle finish makes the small letters and numbers difficult to read. Additionally, some buttons are located very close to certain potentiometers, or in pairs very close together, increasing the risk of incorrect operation. This sometimes leads to a desire for a 19-inch housing for better accessibility.

Nonetheless, the components are very practical in their tactile response. All rotary controls are quite resistant, which is beneficial for live settings. The buttons clearly indicate whether they are engaged or not.

### Design and Construction of the Fischer Amps In Ear Amp 2

The professional intent of the Fischer Amps In Ear Amp 2 becomes evident upon examining the front panel. Starting from the left, there is an on-off switch with an LED indicator below it. Next is a unique feature in the form of two different jacks: a traditional 6.25 mm stereo jack and an XLR connector for the Fischer Amps Guitar In Ear Cable or an XLR male for connecting a standard balanced microphone line with a mini-XLR bodypack. This allows musicians far from the in-ear amp to be connected using microphone cables available on any stage, making headphone signal transmission via a multicore or sub-multicore straightforward and adapter-free.

Following this, there is a small button that requires careful operation with the pinky finger. Upon closer inspection, the tricky operation makes sense to prevent accidental activation. This button adjusts the output power in two levels, suitable for high or low impedance headphones.

The dual power levels are advantageous as they allow high impedance headphones, up to 600 ohms, to be used effectively with this amplifier. However, if high power is mistakenly applied to low impedance headphones, it could cause serious hearing damage. Thus, it's always advisable to start with the low power setting and increase if necessary. For in-ear systems, the manufacturer explicitly recommends only using the low power setting.

Next, there is the master volume control with a white marker, followed by a balance control with a red marker. This section concludes with a button to switch between stereo and mono mix signals.

The pre-amp section of the amplifier follows, managed by a blue-marked control. The Fischer Amps In Ear Amp 2 can attenuate or boost the input signal by up to ±15 dB. The optimal level is reached when the 0 dB indicators on the three-digit LED chains light up consistently and the red peak indicator flashes during brief peaks. This triggers the built-in limiter, protecting the listener's hearing.

To the right, there are two more features: an adjustable AUX-in and an additional daisy-chain output. The AUX-in allows users to directly input a personal in-ear signal without going through the sum mix at the back of the unit. The signal can be mono by using only the upper jack.

A unique aspect is the ability to pass the signal through to a shaker connected to the rear of the unit. This allows for a click track or count-in to be felt physically rather than heard, which can be particularly useful for drummers.

### Rear Panel of the Fischer Amps In Ear Amp 2

The high-quality construction continues on the rear panel. It exclusively features Neutrik XLR input and output connectors, with locking inputs. There is also a line-out for the shaker amp in the form of a jack. The amplifier offers a full-range mode or an 80 Hz high-pass filter with 24 dB/octave via a switch. Additional features include a ground lift, a voltage selector, and a cold-device power socket.

Included with the Fischer Amps In Ear Amp 2 are a power cable, a 2-meter headphone extension, rack adapter wings, and a 19-inch plate for mounting two 9.5-inch units into a single 19-inch rack.

### Practical Use of the Fischer Amps In Ear Amp 2

The practical use of the Fischer Amps In Ear Amp 2 is straightforward—it performs exactly as expected. Upon powering on, a red LED lights up, switching to green after a few seconds of standby. The amplifier is highly dynamic, offers substantial power reserves, sounds entirely neutral, and provides all the features needed for in-ear amplification on stage.

Despite the poor readability of the front panel text, the product delivers everything one might want from a live headphone amplifier. Particularly noteworthy are the resistant potentiometers, reducing the likelihood of accidental volume changes—a crucial feature to prevent unexpected dynamic spikes or jumps that could disrupt performance.

### Conclusion

With the Fischer Amps In Ear Amp 2, the German manufacturer has introduced an excellent product to its lineup. Thanks to its robust construction, flexible adjustment options, and practical design solutions, it is highly recommended for stationary live musicians. The amplifier's performance and build quality make it a reliable choice for any musician needing consistent in-ear monitoring on stage.

### Additional Information and Insights

#### Technical Specifications

The Fischer Amps In Ear Amp 2 boasts a comprehensive range of technical specifications designed to meet the needs of professional musicians. The amplifier operates within a frequency range of 20 Hz to 20 kHz, ensuring full audio spectrum coverage. It has a maximum output power of 500 mW per channel, providing ample headroom for even the most demanding in-ear monitors. The signal-to-noise ratio is rated at over 100 dB, minimizing background noise and ensuring a clean audio signal.

The unit's input impedance is 10 kOhms, suitable for various input sources. It also features a built-in limiter with a fixed threshold to protect users' hearing and prevent signal clipping. The amplifier's THD+N (total harmonic distortion plus noise) is less than 0.01%, ensuring high-fidelity audio reproduction.

#### User Experience and Reliability

Users of the Fischer Amps In Ear Amp 2 have consistently praised its reliability and performance. The amplifier's robust construction and high-quality components contribute to its durability, making it suitable for the rigors of live performance. Many musicians have highlighted the unit's consistent audio quality and its ability to deliver a reliable monitoring experience across different venues and setups.

The in-ear amplifier's user-friendly design, despite minor readability issues, allows for quick adjustments and seamless integration into existing audio setups. The inclusion of both traditional and XLR inputs provides versatility for various signal sources, catering to different musicians' needs.

#### Comparison with Competitors

When compared to other in-ear amplifiers on the market, the Fischer Amps In Ear Amp 2 stands out for its build quality, dynamic range, and user-focused features. While there are other reputable brands and models available, the Fischer Amps In Ear Amp 2 offers a combination of features that cater specifically to live performance needs.

For example, some competing models may offer similar audio quality but lack the robust construction or the flexible input options provided by the Fischer Amps unit. Additionally, the thoughtful inclusion of features such as the AUX-in and shaker output further differentiates it from other products, making it a comprehensive solution for in-ear monitoring.

#### Maintenance and Support

Maintaining the Fischer Amps In Ear Amp 2 is straightforward, thanks to its durable design and high-quality components. Regular cleaning of the connectors and occasional inspections to ensure all connections are secure will help maintain optimal performance. The manufacturer offers support and service options, ensuring that any issues can be promptly addressed.

Fischer Amps also provides detailed user manuals and documentation, guiding users through setup, operation, and troubleshooting. This support network further enhances the user experience, making the Fischer Amps In Ear Amp 2 a reliable choice for professional musicians.

### Expanding Your Setup

For musicians looking to expand their monitoring setup, Fischer Amps offers a range of complementary products. These include additional in-ear monitors, bodypacks, and accessories designed to integrate seamlessly with the In Ear Amp 2. By combining these products, users can create a comprehensive and customizable monitoring system tailored to their specific needs.

For instance, the Fischer Amps Guitar In Ear Cable allows guitarists to connect directly to the in-ear amplifier, ensuring a consistent and reliable signal. Additionally, the Fischer Amps Mini Bodypack provides a portable solution for musicians who need to move around on stage while maintaining their in-ear monitoring.

### Conclusion: A Comprehensive Solution for Live Musicians

In conclusion, the Fischer Amps In Ear Amp 2 is a standout product in the realm of in-ear monitoring systems. Its robust construction, high-quality audio performance, and user-friendly features make it a valuable asset for any live musician. The thoughtful design and practical solutions provided by Fischer Amps ensure that the In Ear Amp 2 meets the demands of professional performance, making it a highly recommended choice for those seeking reliable and high-quality in-ear monitoring.

By choosing the Fischer Amps In Ear Amp 2,

 musicians can enjoy a consistent and reliable monitoring experience, enhancing their performance and ensuring they can focus on their music without worrying about technical issues. The amplifier's versatility and comprehensive feature set make it a worthy investment for both amateur and professional musicians alike.

### In-Depth Features of the Fischer Amps In Ear Amp 2

#### Input and Output Versatility

One of the standout features of the Fischer Amps In Ear Amp 2 is its versatile input and output options. The unit is equipped with both 6.25 mm stereo jacks and XLR connectors, catering to a variety of professional and semi-professional audio setups. This flexibility allows users to integrate the amplifier seamlessly into different monitoring environments, whether on a small stage, in a large concert hall, or within a studio setting.

The XLR input options are particularly noteworthy, providing the ability to use balanced microphone lines and the Fischer Amps Guitar In Ear Cable. This ensures that musicians can achieve a clean and interference-free signal path, crucial for maintaining high audio fidelity. The inclusion of a mini-XLR bodypack compatibility extends this flexibility, enabling easy connection for musicians who need to move freely on stage.

#### Advanced Signal Processing

The Fischer Amps In Ear Amp 2 features advanced signal processing capabilities, ensuring optimal sound quality and protection for the user's hearing. The amplifier includes a built-in limiter, which activates to prevent signal clipping and distortion, safeguarding against sudden volume spikes that could cause hearing damage. This is particularly important for in-ear monitoring systems, where the proximity of the audio source to the ear can pose risks.

Additionally, the pre-amp section allows for precise signal adjustment, with up to ±15 dB of boost or attenuation available. This level of control ensures that the input signal can be finely tuned to suit the specific needs of the performance, whether boosting a quiet signal or attenuating a loud one to avoid distortion. The dual power level settings further enhance this flexibility, accommodating both high and low impedance headphones.

#### Build Quality and Durability

The construction of the Fischer Amps In Ear Amp 2 is a testament to its durability and suitability for demanding live performance environments. The heavy-duty steel chassis not only contributes to the unit's weight but also ensures it can withstand the rigors of touring and frequent handling. This robust build quality is complemented by high-quality Neutrik connectors, known for their reliability and secure connections.

The design also considers practical aspects such as heat dissipation and component protection. The solid construction and careful layout of internal components minimize the risk of damage from impacts or excessive heat, ensuring long-term reliability. This focus on durability makes the Fischer Amps In Ear Amp 2 a reliable choice for professional musicians who need equipment they can depend on night after night.

#### Practical Adjustments and Controls

In practical use, the Fischer Amps In Ear Amp 2 offers a range of controls that allow for precise adjustments. The master volume and balance controls are clearly marked and easy to operate, even in the low-light conditions often found on stage. The sturdy construction of the potentiometers ensures they remain in place once set, reducing the risk of accidental adjustments during a performance.

The additional AUX-in and daisy-chain outputs provide further flexibility for personal monitoring setups. The AUX-in allows musicians to add their own mix or additional audio sources directly into their in-ear feed, bypassing the main mix if desired. The daisy-chain output can be used to connect multiple amplifiers or extend the monitoring system, making it ideal for larger setups or complex stage configurations.

### User Feedback and Testimonials

#### Positive Experiences

Musicians who have used the Fischer Amps In Ear Amp 2 often highlight its reliable performance and high-quality audio output. Drummers, in particular, appreciate the clear and powerful monitoring it provides, enabling them to hear every detail of their performance amidst the loud stage environment. The robust build quality and secure connections are frequently mentioned as key benefits, ensuring the unit can withstand the physical demands of live performance.

#### Constructive Criticism

While the Fischer Amps In Ear Amp 2 receives high praise for its functionality and reliability, some users have noted areas for improvement. The readability of the front panel has been a common point of criticism, with the silver sparkle finish making it difficult to see the labels in low light. Some users have also suggested the addition of more intuitive control layouts to further enhance ease of use during live performances.

### Tips for Optimal Use

To get the best performance from the Fischer Amps In Ear Amp 2, users should consider the following tips:

1. **Initial Setup**: Ensure all connections are secure and double-check the input and output configurations. Start with the low power setting to avoid any potential hearing damage, especially when using low impedance headphones.
   
2. **Signal Adjustment**: Use the pre-amp controls to fine-tune the input signal. Aim for the 0 dB indicators to light up consistently, indicating an optimal signal level without distortion. Utilize the limiter to protect against sudden volume spikes.

3. **Heat Management**: Place the amplifier in a well-ventilated area to prevent overheating. Avoid stacking other equipment directly on top of the unit.

4. **Regular Maintenance**: Clean the connectors regularly to ensure a solid connection. Inspect the unit periodically for any signs of wear or damage, particularly the connectors and switches.

5. **Custom Settings**: Take advantage of the AUX-in and daisy-chain outputs to customize your monitoring setup. Experiment with different configurations to find what works best for your performance needs.

### Expanding Your Audio Setup with Fischer Amps

For musicians looking to expand their audio setup, Fischer Amps offers a range of complementary products that integrate seamlessly with the In Ear Amp 2. These products include in-ear monitors, additional amplifiers, and accessories designed to enhance the overall monitoring experience.

#### In-Ear Monitors

Fischer Amps' in-ear monitors are designed to deliver high-quality audio with excellent isolation from external noise. These monitors are available in various models to suit different preferences and budgets, offering options for both casual users and professional musicians. The ergonomic design ensures a comfortable fit, essential for long performances or rehearsals.

#### Additional Amplifiers

For larger setups or more complex stage configurations, additional Fischer Amps in-ear amplifiers can be added to the system. These amplifiers maintain the same high standards of build quality and audio performance, ensuring a consistent and reliable monitoring experience across the entire setup. The ability to daisy-chain multiple amplifiers allows for easy expansion without compromising audio quality.

#### Accessories and Bodypacks

Fischer Amps also offers a range of accessories to complement their in-ear monitoring systems. These include bodypacks for wireless monitoring, cables, and custom-fit ear molds for a personalized fit. The bodypacks provide musicians with the freedom to move around the stage without losing their monitor signal, while the custom ear molds ensure maximum comfort and sound isolation.

### Conclusion: Fischer Amps In Ear Amp 2 - A Comprehensive Monitoring Solution

The Fischer Amps In Ear Amp 2 stands out as a top choice for professional musicians seeking a reliable and high-quality in-ear monitoring solution. Its robust construction, versatile input options, and advanced signal processing capabilities make it suitable for a wide range of performance environments. Despite minor readability issues on the front panel, the amplifier delivers exceptional audio performance and reliability.

By choosing the Fischer Amps In Ear Amp 2, musicians can ensure a consistent and high-quality monitoring experience, allowing them to focus on their performance with confidence. The amplifier's versatility and comprehensive feature set make it a valuable addition to any live sound setup, whether for a solo performer or a full band. With the support of additional Fischer Amps products, users can create a fully integrated and customizable monitoring system tailored to their specific needs.

TEST: SPL Auditor

 A Headphone Amplifier!

At first glance, many readers might shrug their shoulders and think, "So what? Every mixing console has a built-in headphone amp." While this is true, it's worth noting that every mixing console also has built-in equalizers, yet you'll find multiple outboard EQ units in any ambitious studio. So why has SPL introduced a standalone headphone amplifier like the Auditor, which not only features a single headphone output but also comes with a hefty price tag of nearly 900 euros? It's certainly not due to product suicide or boredom in the development department.

Construction:

Let's address this upfront - the Auditor has almost no relation to its high-end counterpart, the Phonitor, which generates a spatial listening situation through complex phase management. The Auditor is indeed "just" a pure headphone amplifier, but one with an extremely high-quality build.

As with many products from SPL, the Auditor employs 120-volt technology. The house-made discrete SPL-SUPRA operational amplifiers operate at 120 volts, approximately double the operating voltage used in competing products. These voltage values are particularly beneficial in terms of dynamic range and overload resistance. The Supra op-amps boast a signal-to-noise ratio of 116 dB and offer an additional 34 dB of overload resistance, resulting in a dynamic range of about 150 dB.

The amplifier is suitable for all headphones with an impedance of 30 ohms or higher, provided they use a standard stereo jack plug. The only switching option on the front panel is a mono switch, which allows for checking the mono compatibility of the phase position. If the signal becomes quieter or frequency cancellations occur when this switch is activated, the signals are not mono-compatible.

A blue LED on the front indicates the operating status. Weighing 2.65 kilograms and measuring 95 x 210 x 315 mm (H x W x D), the product can be considered relatively compact. The internal components are housed in brushed aluminum.

The Auditor rests on four sturdy, screwed-on feet with rubber pads on the underside to prevent slipping. Two foldable front feet allow the angle of the device to be adjusted upwards by a few degrees. The front panel is dominated by an oversized volume control. This knob can attenuate the input signal by -80 dB or boost it by 10 dB. The 0 dB position is at 2 o'clock.

On the rear panel, the connection options consist exclusively of lockable and balanced XLR male and female connectors. As always with SPL, the labels are readable from both above and below - a simple feature that cannot be overvalued in studio operations. How often have I crawled around in a tangle of cables with a flashlight and hand mirror because a cable needed to be swapped? An IEC connector with a fuse and a voltage selector switch complete the rear panel layout.

Practice:

Now, why use a headphone amplifier at all, when conventional wisdom suggests that you can't really mix a proper production using the extreme stereo function of a regular headphone, and that one should avoid falling into this trap? While this is true, headphones are still unbeatable when it comes to control functions.

As a complementary alternative to loudspeaker monitoring, the extremely high precision of headphones in terms of detail resolution can reveal significantly more than comparable speakers. Excluding room influences, acoustic pitfalls such as noise, clicks, or pops, which often occur in edits, are placed under an acoustic magnifying glass and are therefore easier to detect.

From the first notes, the high impulse fidelity of the Auditor is noticeable. There's no compression to speak of, let alone clipping that cuts off the half-waves. Even high dynamic jumps are cleanly transmitted without any clouding of the sound impression. Due to the neutral design, the individual tracks sometimes sound a touch more centered than with corresponding loudspeaker reproduction, but these are really only nuances that a trained ear can adjust to within a short time.

Looking at the frequency responses, phase response, and distortion factor of the Auditor, one can confidently speak of linear reproduction, with the first two diagrams seeming to be drawn with a ruler. According to the manufacturer, the product internally processes everything between 5 Hz up to 200 kilohertz. By the way, the headphones should be unplugged before switching off the device, as SPL has dispensed with any discharge protection circuits for sonic reasons.

Conclusion:

Admittedly, the Auditor's almost brash suggested retail price might make even ambitious sound engineers flinch briefly, but only until they can test the product in practice. Developed and built in Germany, the Auditor is sonically extremely neutral and high-resolution, very resistant to overload, and almost fatigue-free in continuous operation. In conjunction with loudspeaker monitoring, it offers a real enrichment for studio operations.

In times when headphone operation is becoming increasingly common due to sometimes acoustically questionable room constructions, the Auditor offers more security in terms of sound assessment, especially in the detailed editing area. Personally, I would like to see a 19-inch rack version of the product, as the standalone solution seems to struggle to find the right place in studio operations.

Technical Specifications:

- Frequency response: 5 Hz - 200 kHz (+0/-3 dB)
- THD+N: < 0.001% (1 kHz, +4 dBu)
- Dynamic range: > 150 dB
- Signal-to-noise ratio: 116 dB
- Crosstalk attenuation: > 90 dB
- Input impedance: 20 kΩ balanced
- Maximum input level: +24 dBu
- Gain range: -80 dB to +10 dB
- Output impedance: < 0.5 Ω
- Maximum output level: +24 dBu
- Maximum output power: 2 x 1.25 W into 32 Ω

Power Supply:
- Operating voltage: 230 V AC / 50 Hz (115 V AC / 60 Hz)
- Power consumption: max. 15 W
- Fuse: 230 V: 315 mA slow-blow, 115 V: 630 mA slow-blow

Dimensions and Weight:
- Width: 210 mm (8.27")
- Height: 95 mm (3.74")
- Depth: 315 mm (12.4")
- Weight: 2.65 kg (5.84 lbs)

The Auditor's 120-volt technology is a key feature that sets it apart from many other headphone amplifiers on the market. This high-voltage design allows for an exceptionally wide dynamic range and headroom, which is crucial for accurate monitoring in professional audio environments. The discrete SUPRA operational amplifiers, designed and manufactured by SPL, are at the heart of this technology.

These op-amps operate at voltages significantly higher than those found in typical audio equipment, which usually run at around 30 to 60 volts. The increased voltage allows for several benefits:

1. Increased Headroom: The higher voltage provides more room for signal peaks before clipping occurs, resulting in cleaner, more transparent audio reproduction.

2. Lower Noise Floor: The 120-volt design contributes to the Auditor's impressive signal-to-noise ratio of 116 dB, ensuring that even the quietest passages in a mix are reproduced accurately without being masked by system noise.

3. Improved Transient Response: The high-voltage design allows for faster slew rates, meaning the amplifier can respond more quickly to rapid changes in the audio signal. This results in more accurate reproduction of transients and improved overall clarity.

4. Enhanced Dynamic Range: With a dynamic range exceeding 150 dB, the Auditor can faithfully reproduce both the softest whispers and the loudest crescendos in a recording without compression or distortion.

The Auditor's mono switch is a valuable tool for checking phase coherence and mono compatibility of stereo mixes. When engaged, it sums the left and right channels, allowing the engineer to identify any phase cancellation issues that might occur when the mix is played back in mono. This is particularly important for ensuring that a mix translates well across various playback systems, including those that sum stereo to mono.

The oversized volume control on the front panel is not just for show - it's a high-precision potentiometer that allows for fine adjustments to the listening level. The wide range of adjustment (-80 dB to +10 dB) provides flexibility for different headphone sensitivities and personal listening preferences.

The balanced XLR inputs and outputs on the rear panel are another nod to professional use. Balanced connections offer better noise rejection over long cable runs, which is crucial in studio environments where the Auditor might be located some distance from the mixing console or audio interface.

In terms of build quality, the Auditor's all-metal chassis provides excellent shielding against electromagnetic interference, which is crucial for maintaining the unit's low noise floor. The adjustable front feet allow for ergonomic positioning, which can be particularly useful in cramped studio environments where space is at a premium.

While the Auditor doesn't offer the spatial simulation features of its more expensive sibling, the Phonitor, its focus on pure signal quality makes it an excellent choice for critical listening tasks. Its ability to drive a wide range of headphones, from low-impedance consumer models to high-impedance professional units, adds to its versatility in the studio.

The lack of a protective circuit for headphone disconnection, while potentially risky, is a deliberate choice by SPL to maintain the purest possible signal path. This decision underscores the company's commitment to audio quality above all else, even if it requires more careful operation by the user.

The dynamic range of 150 dB, as specified for the SPL Auditor headphone amplifier, is an exceptionally high value that manifests itself in several aspects in everyday use:

1. Noise-free operation: Such a high dynamic range means that even during very quiet passages, practically no inherent noise from the amplifier is perceptible. This allows for extremely detailed reproduction even at low volumes.

2. Rich in detail: The large dynamic range allows for the perception of the finest nuances in music, from the quietest to the loudest parts. This is particularly important for classical music or audiophile recordings that often have a very wide dynamic range.

3. Overload resistance: With a 150 dB dynamic range, the amplifier can process very loud signal peaks without distortion, avoiding clipping or compression.

4. Realistic reproduction: The human auditory sense has a dynamic range of about 120-130 dB. At 150 dB, the Auditor significantly exceeds this range, allowing for a true-to-life reproduction of the entire audible spectrum.

5. Flexibility: The large dynamic range allows for optimal driving of various types of headphones with different sensitivities, without having to compromise on sound quality.

6. Precise control: For sound engineers and producers, this high dynamic range means they can hear even the smallest errors or artifacts in a recording, which is essential for precise mixing and mastering.

7. Future-proofing: Although current recordings rarely utilize the full 150 dB dynamic range, the amplifier is prepared for future high-resolution audio formats that might potentially use a larger dynamic range.

It's important to note that the full benefit of this enormous dynamic range only comes into play in conjunction with high-quality headphones and high-quality audio recordings. In everyday use with average recordings and headphones, one probably won't be able to exploit the full range of these 150 dB, but the reserves ensure consistently clean and distortion-free reproduction.

In conclusion, the SPL Auditor represents a no-compromise approach to headphone amplification. Its high-voltage design, meticulous attention to signal integrity, and robust construction make it a valuable tool for audio professionals who demand the utmost accuracy in their monitoring chain. While its price point may put it out of reach for some, those who invest in the Auditor will find a device capable of revealing nuances in their mixes that might otherwise go unnoticed, potentially leading to better-sounding final products.

TEST: Soundcraft MFXi 12

 Although I have been involved with music technology of all kinds for several decades now, certain facts still give me pause for thought. Recently, it struck me again how dominant British consoles have become in terms of sound shaping. Especially in the early days of studio and live sound reinforcement technology, one could almost speak of a monopoly position.

Particularly when it comes to filters, "British standards" still apply today, from simple rehearsal room setups to the upper echelons of the few remaining high-end studios with corresponding facilities. And of course, one should not forget the characteristic preamplifier.

One of the big names from this era is Soundcraft. Active for over three decades now, the company, which now operates under the aegis of Harman International, is also opening up to smaller budgets without having to make too many concessions in the aforementioned references. As expected, this unfortunately no longer goes hand in hand with European manufacturing, but must be produced on the other side of the globe, in China.

With the MFXi series in the 12-channel version, I now have a small console for testing that sets out to follow in big footsteps in terms of tradition and sound.

## Construction

Even if one didn't know which country Soundcraft originated from, the "classic" British surface of the small console jumps out at you immediately after unpacking the test subject. Villa Kunterbunt at its best! Fully committed to British tradition, the optics indulge in the classic "color orgy", once again ruining the rather discreetly oriented sense of taste of the continental European ;-)

Potentiometer knobs vie for supremacy in the "colorful-as-colorful" contest with the colors yellow, red, pastel green, purple, light blue, black and white, all presented on a dark blue frame. Yes, that's how the British are, just as isolated in terms of taste as their currency and their little island on which they roam ;-).

The MFXi 12 is a 12-channel mixer that comes in an extremely compact package with dimensions of 482 mm x 94 mm x 405 mm and a weight of 6.7 kg. In addition to the twelve mono inputs, there are two additional stereo channels available for management. As a special feature, the product has an integrated Lexicon FX device, which has 32 memory slots. If needed, the mixer can also be rack-mounted using two side wings.

A mono channel strip is structured as follows:

- Mic: lockable XLR input for microphones or similarly low-level signals, all of which can be supplied with 48V phantom power if needed
- Line: balanced jack input for high-level signals such as drum machines, keyboards, etc.
- Insert: insertion point before the gain control in Y-configuration, typically used for compressor or limiter
- Gain: preamplifier control from 5 dB to 60 dB
- High-Pass Filter: low cut filter at 100 Hz, often used to reduce noise from wooden stages or finger-drumming noises of hyperactive singers
- HF: Shelving filter at 12 kHz with +/- 15 dB
- MF: Semi-parametric filter from 150 Hz to 3.5 kHz with +/- 15 dB at a Q factor of 1.5
- LF: Shelving filter at 80 Hz with +/- 15 dB
- Aux 1: fixed pre-fade, thus suitable as a monitor path
- Aux 2: switchable pre- or post-fade
- Fx: Level control for Lexicon FX
- Pan: Panorama control
- Mute: Mute switch with control LED
- PFL: Pre-Fade-Listening, signal is routed to headphones, control room outs and signal LEDs for pre-listening
- Mix/Sub: optional routing of the signal to main out or subgroup left/right
- PK: red peak LED, connected to pre-EQ, post-EQ and post-fade
- SP: Signal LED, indicates if a signal is present, pre-EQ switched

As expected, the stereo input channels lack XLR inputs, and there is no insertion point. They also feature a fixed mid-filter at 720 Hz, also adjustable by +/- 15 dB, with a Q factor of 0.8.

The master section has the usual control options of a console in this series, such as separate subgroup management and FX management. It's nice that despite the cramped space on the master control panel, thanks to "Villa Kunterbunt", you can assign the individual controls to their subordinate and superordinate functions in seconds.

On the rear, the console has only a very sparse equipment, which is limited to an IEC connector, an ON/OFF switch and a Kensington anti-theft device.

## Practice

It's tight on the console surface, that much must be said. Anyone hoping to be able to turn a potentiometer of the channel strips completely by 270 degrees with two fingers, with the exception of the gain control, will unfortunately be disappointed, unless they have extremely slim fingers like a model or stopped growing at the tender age of nine. Here you inevitably always bump into the adjacent controls, so you can only change your settings in 2-3 approaches. Well, not an existential criterion, but it annoys me like hell!

On the other hand, the controls run pleasantly stiffly, so that at least you don't unintentionally change the settings of the next pot again. At least that... Of course, you have to save on width to keep the 19-inch component in mind, but a little more depth would have worked wonders here. The general workmanship, however, is very good, there's nothing to complain about here. No wobbling, no jerking, no scratching, excellent.

In terms of preamp and filters, the MPXi 12 leaves a really good impression. Not that it completely blows you away in both sound groups, as you're used to from the large Soundcraft consoles, but for the price Soundcraft is asking, the components do quite remarkable things.

Once again, the Lexicon FX processor proves to be a real highlight. The spaces generated by the integrated effects unit are of high plastic density, great liveliness and know how to convince more than just for this price, respect!

## Conclusion

Nomen est omen! That's what Soundcraft must have said to themselves when they conceived the concept of the MPXi series. The task of bringing a good-sounding console to market as inexpensively as possible while not having to cut too many trademarks has been accomplished by the British company.

With the MPXi 12, the traditional company has created a small, flexible console that doesn't disappoint in terms of sound, scores in terms of workmanship, and comes with an excellent sounding Lexicon effects unit right from the start, for which you would have had to pay far more than half the console price for a hardware version not too long ago.

If it weren't for this cramped space on the control panel, you'd have to search hard to find fault with the product.

Handy, practical, good, a real Soundcraft indeed.

## Technical Specifications

- Frequency response: 20 Hz - 20 kHz (+0/-1 dB)
- THD: <0.007% @ 1 kHz, 0 dBu
- Noise: -127 dBu (150Ω source, max gain, 20 Hz - 20 kHz)
- Crosstalk: >90 dB @ 1 kHz
- Input impedance: Mic: 2 kΩ, Line: 10 kΩ
- Maximum input level: Mic: +22 dBu, Line: +30 dBu
- Phantom power: +48V, switchable per channel
- EQ: HF: ±15 dB @ 12 kHz, MF: ±15 dB (150 Hz - 3.5 kHz), LF: ±15 dB @ 80 Hz
- High-pass filter: 100 Hz, 18 dB/octave
- Aux sends: 2 (1 pre-fade, 1 switchable pre/post-fade)
- FX send: Post-fade
- Subgroups: 2 (L/R)
- Main mix: Stereo
- Lexicon FX: 32 presets (various reverbs, delays, choruses)
- USB interface: 2-in/2-out, 24-bit/48 kHz
- Power consumption: 40W
- Dimensions (W x H x D): 482 x 94 x 405 mm
- Weight: 6.7 kg

## Additional Features

The Soundcraft MFXi 12 also includes several features that enhance its versatility:

1. GB30 Mic Preamps: These renowned preamps, designed by Graham Blyth, provide exceptional clarity and headroom.

2. Sapphyre Asymmetric EQ: This EQ design allows for more musical and natural-sounding adjustments to the audio signal.

3. dbx® Limiters: Channels 1-2 feature built-in dbx limiters to prevent signal overload.

4. 100mm Faders: Long-throw, smooth faders for precise level control.

5. Lexicon Effects Engine: 32 studio-grade effects including reverbs, delays, and modulation effects.

6. Aux Sends: Two aux sends per channel for external effects or monitor mixes.

7. Direct Outputs: Each mono input channel features a direct output for multitrack recording.

8. 4-band EQ on Mono Inputs: Allows for detailed sound shaping.

9. 3-band EQ with Swept Mids on Stereo Inputs: Provides flexibility for stereo sources.

10. Mute Groups: Allows for quick muting of multiple channels.

11. Talkback Section: Built-in talkback mic with routing options.

12. 12-segment LED Meters: Accurate visual feedback of signal levels.

13. Rack-mount Kit Included: Allows for easy integration into existing setups.

14. Robust Metal Chassis: Ensures durability for live and studio use.

15. Global Phantom Power: Switchable +48V phantom power for condenser microphones.

## Applications

The Soundcraft MFXi 12 is suitable for a wide range of applications:

1. Small to Medium Live Venues: Ideal for clubs, bars, and small theaters.

2. Houses of Worship: Perfect for managing multiple inputs in church settings.

3. Educational Facilities: Great for school auditoriums and music departments.

4. Project Studios: Offers professional features for home and project studio recording.

5. Mobile DJs: Compact size makes it suitable for mobile setups.

6. Podcasting: Multiple inputs and USB interface make it great for podcast production.

7. Small Broadcast Setups: Suitable for small radio stations or internet broadcasting.

8. Corporate Events: Handles multiple microphones and playback sources for presentations.

9. Rehearsal Spaces: Provides monitoring options for band rehearsals.

10. Small Theater Productions: Manages multiple actors' mics and sound effects.

## User Interface and Workflow

The MFXi 12's user interface is designed for intuitive operation:

1. Color-coded Controls: Different colored knobs for different functions aid quick identification.

2. Logical Layout: Channel strips follow a top-to-bottom signal flow for easy understanding.

3. Clear Labeling: All controls are clearly labeled for easy reference.

4. Illuminated Switches: Mute and PFL switches are illuminated for clear status indication.

5. Centralized Master Section: All main controls are grouped for easy access.

6. Dedicated FX Return: Separate fader for the internal Lexicon effects return.

7. Flexible Monitoring: PFL and AFL options for easy signal checking.

8. Comprehensive Metering: Main output and solo bus have 12-segment LED meters.

## Sound Quality

The MFXi 12 maintains Soundcraft's reputation for excellent sound quality:

1. Low Noise Floor: The preamps and overall circuit design result in a very low noise floor.

2. Wide Dynamic Range: Capable of handling both very quiet and very loud sources.

3. Transparent EQ: The Sapphyre Asymmetric EQ allows for musical adjustments without artifacts.

4. Clean Signal Path: High-quality components throughout maintain signal integrity.

5. Warm British Sound: The preamps impart a subtle warmth characteristic of British consoles.

6. Professional Effects: The Lexicon effects add depth and space without compromising clarity.

7. Accurate Stereo Imaging: Precise pan controls and stereo channels maintain a wide, accurate stereo field.

The Soundcraft MFXi 12 distinguishes itself from other 12-channel mixers in several aspects:

1. British Sound: The MFXi 12 offers the characteristic "British" sound that Soundcraft is known for. This is particularly evident in the preamps and EQs, which impart a subtle warmth and musicality.

2. Integrated Lexicon Effects Processor: A highlight is the built-in Lexicon FX processor with 32 memory slots. The effects, especially the reverbs, are characterized by high quality and liveliness.

3. Color-coded Control Surface: The MFXi 12 stands out with its striking, colorful design. The multi-colored potentiometer knobs facilitate operation and are typical of the British style.

4. Compact Construction: With dimensions of 482 mm x 94 mm x 405 mm and a weight of 6.7 kg, the MFXi 12 is very compact and portable.

5. Flexible Channel Configuration: In addition to the 12 mono inputs, the mixer offers two additional stereo channels.

6. High-quality Filters and EQs: The MFXi 12 features the high-quality filters and EQs typical of Soundcraft, which are considered a reference in the industry.

7. Rack-mounting Option: If needed, the mixer can be rack-mounted using two side wings.

This combination of British sound character, high-quality features, and compact design makes the Soundcraft MFXi 12 an interesting option compared to other 12-channel mixers in this price range.

Furthermore, the MFXi 12 offers some additional features that set it apart:

8. GB30 Mic Preamps: These preamps, designed by Graham Blyth, provide exceptional clarity and headroom, which is not common in this price range.

9. Sapphyre Asymmetric EQ: This unique EQ design allows for more musical and natural-sounding adjustments to the audio signal, a feature often found in higher-end consoles.

10. dbx® Limiters: Channels 1-2 feature built-in dbx limiters to prevent signal overload, which is a valuable addition for live sound applications.

11. USB Interface: The inclusion of a 2-in/2-out USB interface allows for easy integration with computers for recording or playback, which is not always present in analog mixers of this class.

12. Comprehensive Routing Options: The mixer offers flexible routing capabilities, including subgroups and multiple aux sends, providing versatility for various applications.

13. Build Quality: Despite its affordable price point, the MFXi 12 maintains Soundcraft's reputation for robust construction and reliability.

14. Legacy and Support: Being part of the Soundcraft family, users benefit from the company's long-standing reputation and support network.

These features, combined with Soundcraft's heritage in professional audio, make the MFXi 12 a unique offering in the 12-channel mixer market. It bridges the gap between budget-friendly mixers and more expensive professional-grade equipment, offering a taste of high-end features at a more accessible price point.

In conclusion, the Soundcraft MFXi 12 represents a compelling option for those seeking professional-grade mixing capabilities in a compact, affordable package. It successfully brings many of the hallmarks of larger Soundcraft consoles to a more accessible format, making it a versatile tool for a wide range of audio applications.

TEST: Sony DWZ M50

 ## Sony DWZ-M50 Wireless Microphone System: A Hidden Gem in the Audio World

It's always interesting to observe how a company name, which is among the most famous in one sector of consumer electronics, can be underappreciated in a related field despite years of continuous presence and high-quality product lines. Sony is all too familiar with this phenomenon. Despite producing high-end reverb units and the legendary Dash machines in the late eighties, the corporation is mostly associated with screens or playback devices in the minds of users.

To bridge this gap in perception, the Japanese conglomerate has introduced the DWZ Wireless Series to their lineup. Manufactured in Korea, the products consist of the DWZ-B30GB, designed for instrumentalists, and the DWZ-M50 system, which I have for testing. The latter comprises a microphone and its corresponding receiver. Both products operate in the 2.4 GHz band, which is exempt from the general unrest regarding carrier frequencies. The signal is transmitted digitally at 24-bit/48 kHz.

A helpful tip: on the Thomann Music Store website, you can view the frequencies used by any wireless system, regardless of manufacturer, along with their worldwide approvals and any associated restrictions. This is an invaluable tool that sheds light on the sometimes chaotic regulations of international radio traffic.

## Construction

The Sony DWZ-M50 package includes the ZTX-M01 handheld microphone, the ZRX-HR50 half-rack receiver with power supply, a microphone holder, and two antennas that are screwed onto the back of the receiver using two bayonet locks. The overall build quality of the package is impressive. Sufficiently thick sheet steel or metal is used at almost all relevant points, suggesting a long lifespan. You'll search in vain for cheap plastic latches or other inferior materials.

All connections on the receiver are located on the rear panel and consist of two unbalanced 1/4" outputs, a balanced XLR output (switchable between line and microphone level), and a USB port for maintenance work. On the front, an endless rotary encoder with push function navigates the menu, complemented by an Esc button for returning to the main screen and the On/Off switch. A clearly readable color display informs the user about all important parameters such as reception quality, transmitter battery status, the selected channel, and any use of the internal equalizer. The equalizer features five bands (60 Hz, 250 Hz, 1 kHz, 4 kHz, 12 kHz), each of which can be boosted or cut by 12 dB.

The system transmits on six different channels, with two transmission options available. In the "Wide" preset, the 2.5 MHz range is used, which leaves other frequency users virtually undisturbed. In return, you have to accept a slightly higher latency. The alternative is "Narrow," which operates with a narrower bandwidth and shorter latency but may potentially interfere with existing wireless networks. The handheld and receiver obtain their optimal channel selection through a scan mode, which in this setup is called "Clear-Channel-Scan."

The ZTX-M01 handheld is powered by two AA batteries and, according to the manufacturer, has an average operating time of 8 hours. A lockable Power/Mute switch allows the microphone to be temporarily deactivated, which is particularly useful during breaks in performance and relieves the sound engineer. The lock can only be accessed by unscrewing the middle part of the microphone. Here you'll also find a digital display for matching the channel to the receiver and a USB port for firmware updates.

The handheld comes factory-equipped with a dynamic capsule featuring a cardioid polar pattern. If needed, it can be retrofitted with a proprietary supercardioid, wide cardioid, or condenser capsule. Third-party capsules from various manufacturers such as Shure or Neumann can also be mounted on the handheld. When unscrewing the microphone capsule, there's also the option to reduce the output level of the microphone via a small PAD switch, offering attenuation of 6 or 12 dB. This allows you to control even Tom Jones-level sound pressure without internal distortion.

## In Practice

Setting up the combination is, as expected, straightforward. The handheld and receiver find each other immediately, with channel selection occurring automatically. In terms of sound, the ZTX-M01 offers a very clear and neutral reproduction of the voice with a slight emphasis on the presence frequencies. Nevertheless, you don't need to worry about constant de-esser use; the tonal design is practical and gives even more subtle voices good assertiveness.

The handheld feels good in the hand, shows good balance, and due to its moderate weight of about 300 grams including batteries, it won't become a burden even for delicate female arms during a longer performance. Signal transmission is stable; even several walls within my studio or a trip to the end of my garden property did not lead to any signal loss. Remarkably, the latency is extremely short and inaudible at just 3 milliseconds, guaranteeing truly immediate transmission. The handheld convinces in both singing and speech applications, shining with an unpretentious basic configuration.

## Sound Quality

The Sony DWZ-M50 system impresses with its audio fidelity. The microphone capsule delivers a frequency response that is well-suited for vocal applications, with a slight boost in the upper midrange that adds clarity and presence to the voice without becoming harsh or sibilant.

In testing, the system performed admirably across a range of vocal styles. From soft, intimate performances to powerful rock vocals, the DWZ-M50 maintained clarity and detail. The cardioid polar pattern effectively rejects off-axis sounds, helping to isolate the vocalist's performance and reduce the risk of feedback in live situations.

The system's digital transmission ensures that the audio quality remains consistent throughout the operating range. There's no degradation of signal quality as you move away from the receiver, which is sometimes an issue with analog wireless systems.

The built-in equalizer in the receiver is a valuable tool for fine-tuning the sound to suit different voices or to compensate for room acoustics. With its five bands, it offers enough flexibility to make meaningful adjustments without becoming overly complex.

## Range and Reliability

One of the standout features of the DWZ-M50 is its impressive range. In open-air tests, the system maintained a stable connection at distances exceeding 60 meters (about 200 feet). Even in more challenging environments with walls and other obstacles, the range remained more than adequate for most live performance scenarios.

The 2.4 GHz band, while more crowded than traditional UHF bands, proved to be surprisingly robust. The Clear-Channel-Scan feature effectively identified and locked onto the clearest available frequencies, minimizing interference even in RF-dense environments.

During extended use, the system demonstrated excellent reliability. There were no dropouts or unexpected disconnections, even when other wireless devices were in use nearby. This level of dependability is crucial for professional applications where signal loss is not an option.

## Battery Life

Sony's claim of 8 hours of battery life from two AA batteries proved to be conservative in our tests. With fresh alkaline batteries, we consistently achieved over 9 hours of continuous use before the low battery indicator appeared. This is more than enough for most performance scenarios, though for longer events or heavy users, rechargeable NiMH batteries could be a cost-effective and environmentally friendly option.

The battery life indicator on both the handheld transmitter and the receiver display was accurate and provided ample warning before power depletion, allowing for timely battery changes.

## Versatility

While the DWZ-M50 is primarily designed for vocal applications, its versatility shouldn't be overlooked. The ability to swap out the microphone capsule opens up a range of possibilities. For instance, using a supercardioid capsule could provide better isolation in noisy stage environments, while a condenser capsule might be preferred for capturing more nuanced performances in controlled settings.

The switchable output level on the receiver (between mic and line level) adds to the system's flexibility, allowing it to interface seamlessly with a wide range of audio equipment from mixing consoles to camera inputs for video production.

The inclusion of both XLR and 1/4" outputs on the receiver is a thoughtful touch, ensuring compatibility with virtually any audio setup without the need for adapters.

## User Interface and Ease of Use

Sony has done an excellent job with the user interface of the DWZ-M50 system. The color display on the receiver is bright, clear, and provides all the necessary information at a glance. The menu system, navigated by the rotary encoder, is intuitive and easy to use, even for those who might be less technically inclined.

The automatic pairing between the transmitter and receiver is a time-saver, especially in fast-paced live environments. The Clear-Channel-Scan feature, which automatically selects the best available frequency, works quickly and effectively, reducing setup time and potential for user error.

The lockable power/mute switch on the handheld transmitter is a smart feature that prevents accidental muting or power-off during performance. While the lock can only be engaged by partially disassembling the microphone, this design ensures that it won't be accidentally toggled.

## Comparison with Competitors

In its price range, the Sony DWZ-M50 competes with systems from established pro audio brands like Shure, Sennheiser, and Audio-Technica. While these brands may have more name recognition in the professional audio world, the DWZ-M50 holds its own in terms of features and performance.

Compared to similarly priced systems, the Sony offers comparable or better sound quality, and its range and reliability are on par with the best in its class. The inclusion of a 5-band EQ in the receiver is a standout feature that many competitors don't offer at this price point.

Where some competitors might have an edge is in their ecosystem of compatible products or in the availability of alternative capsules. However, Sony's openness to third-party capsules somewhat mitigates this disadvantage.

## Value for Money

Considering its feature set, build quality, and performance, the Sony DWZ-M50 represents excellent value for money. At a street price under 600 euros, it offers capabilities that are often only found in systems costing significantly more.

The robust construction suggests that this is a system built to last, which should be factored into any value calculation. Additionally, the flexibility offered by the interchangeable capsule design and the built-in EQ adds to the long-term value of the system.

## Potential Improvements

While the DWZ-M50 is an impressive system overall, there are a few areas where improvements could be made in future iterations:

1. Including a carrying case would be a welcome addition, especially for users who frequently transport their equipment.

2. The ability to use rechargeable lithium-ion batteries, perhaps with a charging dock, could appeal to heavy users and venues.

3. While the 2.4 GHz band performs well, offering a UHF option could broaden the system's appeal in markets where 2.4 GHz wireless use is more restricted.

4. Expanding the range of proprietary capsules would give users more options without needing to look to third-party manufacturers.

## Conclusion

For those willing to look beyond the big names in the transmitter segment, Sony has a real bargain on hand with the DWZ-M50 package. With a street price under 600 euros, the system sits in the mid-price range but can boast features that are generally only included in the scope of delivery from a four-digit price point upwards.

The build quality of the Japanese manufacturer's Made in Korea product is very good and suggests a long service life. In terms of sound, the handheld asserts itself in practice through its neutral to brilliant tuning and covers a correspondingly wide range of vocal applications. The PAD switch also allows you to tame very loud voices accordingly, even before the signal starts its journey to the receiver.

The Sony DWZ-M50 is a product worth testing, one that doesn't need to shy away from comparison with other providers. It offers professional-grade features and performance at a price point that makes it accessible to a wide range of users, from ambitious amateurs to working professionals.

Its combination of sound quality, reliability, and user-friendly features makes it a strong contender in the competitive wireless microphone market. While it may lack the brand recognition of some competitors in the pro audio space, the DWZ-M50 proves that Sony's expertise in consumer electronics translates well to professional audio products.

For venues, houses of worship, educational institutions, or touring performers looking for a dependable wireless system that won't break the bank, the Sony DWZ-M50 deserves serious consideration. It's a reminder that sometimes the best solutions come from unexpected places, and that it pays to look beyond the usual suspects when choosing audio equipment.

In an era where wireless spectrum is becoming increasingly crowded and regulated, the DWZ-M50's use of the 2.4 GHz band also offers a degree of future-proofing. As traditional UHF bands become more restricted, systems like this may become increasingly attractive options.

Ultimately, the Sony DWZ-M50 is a testament to the company's engineering prowess and understanding of user needs. It's a system that punches above its weight class and serves as a worthy ambassador for Sony in the professional audio world. For those in the market for a new wireless microphone system, the DWZ-M50 is not just an option to consider – it's one that could very well end up being your top choice.

TEST: Sonic Farm Xcalibur JC

 How times have changed. For decades, sound engineers of all stripes went to great lengths to eliminate overloads and distortions of all kinds from recordings. In the early days of recording technology, tube-based analog equipment induced an almost latent saturation, which, especially in the summing stage, provided a considerable compression of the material and an accompanying first "mastering stage" long before this production step was planned as an integral part of a sound recording. Today, equipment resistant to overloading is no longer a special feature; rather, one sometimes tries to give the sound material the decisive "kick" with a dedicated half-wave cut at the right place. The Canadian company Sonic Farm carries the Sonic Farm Xcalibur JC, a saturation preamplifier in its portfolio, which offers far more unique selling points than just saturation.

The Concept of the Sonic Farm Xcalibur JC

It becomes apparent during the unpacking process that the Sonic Farm Xcalibur JC is not your typical 19-inch semiconductor equipment. Weighing nearly 8 kg and with an installation depth of 34 cm, one might initially mistake the 1U high rack unit for a power amplifier rather than a typical preamp. Nevertheless, this product is a 2-channel microphone preamplifier and saturator for processing microphone, line, and instrument signals, based on a Class-A tube circuit utilizing EF86 tubes. These pentode tubes, known for their low noise and high gain characteristics, contribute significantly to the unit's sonic signature.

The "JC" suffix stands for the signature version by engineer and producer Joe Chiccarelli, who celebrated his greatest successes with artists such as Alanis Morissette, Elton John, Bee Gees, Journey, and Frank Zappa. According to Wikipedia, his name is associated with over 50 million albums sold, though this seems a conservative estimate considering that the "Saturday Night Fever" soundtrack by the Bee Gees, which he mixed, alone sold over 40 million units. Chiccarelli's influence on the unit's design is evident in its versatility and attention to detail in the saturation stages.

One of the unique features of the Sonic Farm Xcalibur JC is the cascading capability of the two channels, which can be activated by a mini-switch in the center of the front panel. This feature allows users to create a serial signal path, effectively doubling the available gain and saturation options. In this configuration, one can also inflate the two saturation stages into a formidable distortion effect, reminiscent of classic tube overdrive pedals but with far greater clarity and control.

The fact that Sonic Farm has given considerable thought to overdriving is evident not only in the use of two pentodes per channel, providing up to +68 dB gain for microphone input and +48 dB for line and instrument signals, but also in the optional use of FET transistors for distortion. FET transistors are characterized by the fact that, when overdriven beyond 0 dB, they do not convey the harsh distortion of regular transistors, but rather produce a significantly softer distortion not unlike tube overdrive. This hybrid approach allows for a wide range of tonal possibilities, from subtle warming to full-on distortion.

The output levels of the channels are separately adjustable, with the preamp utilizing Cinemag output transformers. These high-quality transformers are known for their ability to add a subtle, musical coloration to the signal, further enhancing the unit's analog warmth.

The Front Panel of the Sonic Farm Xcalibur JC

Those expecting a pure 19-inch distortion unit will be surprised by the Xcalibur JC's comprehensive feature set. A considerable number of controls and switches on the front panel indicate that the Sonic Farm Xcalibur JC is equally concerned with high-quality preamp functionality as it is with saturation effects. The first three pushbuttons on the left side offer standard features: phantom power (48V), a 15 dB PAD switch, and an instrument selector switch that routes the input to the front panel jack.

The following three rotary controls "CLN" (Clean), "DRV" (Drive), and "BLD" (Blend) form the core of the unit's saturation capabilities. These controls, in conjunction with several toggle switches, allow for precise tailoring of the distortion character. The "CLN" control adjusts the input gain of the clean signal path, while "DRV" determines the amount of signal sent to the saturation circuit. The "BLD" knob allows for a perfect balance between the clean and saturated signals, enabling anything from subtle harmonic enhancement to full-on distortion.

A two-color LED (red/green) reflects the input level, providing visual feedback on the signal strength. It's worth noting that distortion in the red range is not necessarily desirable but merely indicates that the input level is too high for clean operation. This metering system allows for precise gain staging, crucial for achieving the desired level of saturation without unwanted clipping.

Following these controls is another selector switch for the output, allowing a choice between a low-distortion solid-state IC or a transformer. This feature provides two distinct flavors of output coloration, with the solid-state option offering a more transparent sound and the transformer adding a subtle, vintage-like character to the signal. A phase switch completes this section, allowing for easy correction of phase issues that may arise when using multiple microphones.

The upper row of the Sonic Farm Xcalibur JC is dominated by a total of 7 toggle switches, all of which have 3 switching states. This array of switches provides an unprecedented level of control over the unit's tonal shaping capabilities. Starting with a Fat Shelving Preset Boost and Gain Max switch on the left side, the switch in the left position (Fat) provides a low-frequency shelving boost from 300 to 1000 Hz, depending on the setting of the Gain switch two positions to the right. Lower gain settings correspond to a higher corner frequency and consequently a stronger bass boost. This filter does not use a separate stage but utilizes the clean tube amplification stage, ensuring that the bass enhancement remains musical and natural-sounding.

The boost level can be fine-tuned with the left of the two trim potentiometers, accessible from an upper flap located between the boost switches about a centimeter from the front panel. This level of detail in control allows users to dial in precisely the right amount of low-end enhancement for any source material. The maximum boost also depends on the Gain switch; the lower the Gain switch, the more bass boost is available, up to about 12 dB with a 6dB/octave slope.

The high-frequency shelving boost, labeled as "AIR," operates from 1 to 8 kHz and also depends on the Gain switch setting. Lower gain settings correspond to a lower corner frequency and consequently a stronger treble boost. In the left position, the corner frequency is about 1.5 octaves lower than in the right position. This filter also affects the clean tube amplification stage, ensuring that the high-frequency enhancement remains smooth and musical. As with the low-frequency boost, the exact onset of the AIR effect can be adjusted via a trim potentiometer inside the housing, allowing for precise tailoring of the high-end response.

After the aforementioned GAIN switch, we come to the microphone input impedance switch labeled "IMPED." The impedance selection is a crucial feature that can significantly affect the tonal characteristics of different microphones. The lower the impedance, the higher the load the preamplifier input exerts on the microphone. The standard values are 10 kΩ for the middle position (HI), 900 Ω for the left (LO), and 2400 Ω for the right (MED).

The "PAD" switch also affects the resulting microphone input impedance value, providing even more flexibility in matching the preamp to various microphone types. A change in input impedance tends to affect the sound of dynamic microphones more significantly, to some extent ribbon microphones, and to a lesser extent or not at all condenser microphones. Lower impedance values generally roll off the highs somewhat, which can be useful for taming overly bright sources or adding a vintage-like character to modern microphones.

A subsequent high-pass filter cuts at either 160 Hz (Pos. 1) or 80 Hz (Pos. 2) with 6dB/octave slope. Only the clean signal is processed by this high-pass filter, allowing users to remove unwanted low-frequency content before it reaches the saturation stage. This can be particularly useful for cleaning up boomy sources or reducing proximity effect on close-miked vocals.

The following ODf/ODf1 switch is a high-pass filter before the overdrive stage, which can be useful if too much bass or low mids hit the OD tube and cause excessive "mud" in the distorted signal. In this case, the unwanted bass or mids can be attenuated with this 6dB/octave filter with a gentle slope. The individual positions allow for full engagement (left), attenuation of bass and low mids (center), and attenuation of only bass (right). Depending on the instrument connected, interesting effects can be achieved here, especially in the Lo-Fi range.

In the JC signature version, an additional ODf2 control is employed. This is a post-drive low-pass filter with a pre-mix of 3 positions: 5.5 kHz, 12dB/octave (left); 18 kHz (center), and 1 kHz, 6dB/octave (right). The first position can be used to suppress buzzing distortions that may occur when processing some signals, particularly useful for taming harsh high-frequency content in heavily saturated signals. The middle position is practically a bypass, allowing the full frequency range of the distorted signal to pass through. The right position is useful when you want to mix in some saturation to make something sound bigger and fatter without the distortion becoming too obvious, effectively acting as a "thickening" effect.

The rear panel of the Sonic Farm Xcalibur JC is straightforward and functional, featuring 6x XLR connectors (4 in, 2 out), L/R microphone inputs, L/R line inputs, a ground lift switch, main fuse, mains selector switch, and an IEC power socket. This layout ensures easy integration into any studio setup, whether as a front-end for recording or as an insert processor for mixing and mastering applications.

As evident from this detailed breakdown of features, the Sonic Farm Xcalibur JC offers very comprehensive signal processing capabilities, allowing for extensive sound shaping at every stage of the signal path. The unit's design philosophy seems to be centered around providing maximum flexibility and control to the user, enabling everything from subtle analog warmth to extreme distortion effects. This level of control is particularly valuable in the context of modern production techniques, where the ability to precisely shape tone and add character to digital recordings is highly prized.

In Practice

Activating the Sonic Farm Xcalibur JC with what is probably the stiffest on/off switch I've ever operated, we're greeted by a strong red glow emanating from the unit. This visual feedback, reminiscent of classic tube gear, sets the stage for the sonic experience to come.

As a guitarist, the Sonic Farm Xcalibur JC immediately brings a broad smile to my face, thanks to the excellent quality of its distortion capabilities. The unit truly shines when processing instruments where the treble component is not too pronounced, such as Hammond organ or clavinet. In these cases, the Xcalibur JC produces outstanding sounds right out of the gate, easily reaching distortion levels comparable to a high-gain guitar amp, but with significantly less unwanted noise.

The saturation characteristics of the Xcalibur JC are notably smooth and musical, even at extreme settings. This is likely due to the careful implementation of both tube and solid-state circuitry in the signal path. The ability to blend between clean and saturated signals allows for a wide range of tonal possibilities, from subtle harmonic enhancement to full-on distortion.

One particularly impressive aspect of the Xcalibur JC is its ability to maintain clarity and definition even when heavily saturated. This is where the unit's hybrid design really comes into play, combining the warmth of tubes with the precision of solid-state components. The result is a distortion character that adds richness and complexity to the source material without becoming muddy or indistinct.

The extensive EQ and filtering options prove invaluable in shaping the distorted signal. The ability to tailor the low and high-frequency content both before and after the saturation stage allows for precise control over the final tonal character. This level of control is particularly useful when working with dense mixes or when trying to achieve specific vintage-style tones.

While the Xcalibur JC excels at more extreme distortion effects, it's equally capable of providing subtle coloration and harmonic enhancement. At lower drive settings, the unit imparts a gentle warming effect that can add depth and dimension to digital recordings. This versatility makes the Xcalibur JC a valuable tool not just for tracking, but also for mixing and mastering applications.

The question of whether to add a speaker (simulation) to the signal path is left to individual preference, but even without any additional sound processing, the Sonic Farm Xcalibur JC convinces across the board. Its raw output is remarkably usable and musical, a testament to the quality of its internal components and circuit design.

Of course, the product can also provide subtle fattening in the range of tape saturation, but it's the high gain reserves that truly offer a unique selling point in the preamp segment. The extensive filtering options ensure that there's something for everyone, and any remaining sonic possibilities can be addressed with a pre- or post-EQ.

One minor criticism could be leveled at the unit's noise floor, which becomes noticeable at extreme gain settings. However, this is a common characteristic of high-gain tube circuits and is generally not problematic in real-world recording scenarios. The noise is easily overshadowed by the signal in most applications, and the richness of tone more than compensates for any slight increase in noise.

Conclusion

With the Sonic Farm Xcalibur JC, the Canadian manufacturer has created an outstanding preamp with a focus on saturation, easily achieving the distortion levels of a high-quality dedicated distortion unit. The components used, along with the build quality and sonic possibilities, place this unit firmly in the top tier of studio equipment.

The Xcalibur JC enables the creation of characterful and defining sounds that can breathe life into thin or lifeless recordings. Its versatility makes it equally at home on individual tracks during recording and mixing, as well as on entire mixes during mastering.

While the price point of the Xcalibur JC places it in the professional and high-end project studio market, its performance justifies the investment for those seeking a truly versatile and high-quality saturation device. The unit's ability to impart everything from subtle analog warmth to extreme distortion effects, all while maintaining a high degree of clarity and musicality, sets it apart in a crowded market of analog-modeling devices.

In an era where digital perfection often leads to sterile and lifeless recordings, tools like the Sonic Farm Xcalibur JC provide a valuable means of reintroducing the character and imperfections that make recordings sound alive and engaging. Whether used subtly to add depth and dimension to digital tracks, or pushed to extremes for creative distortion effects, the Xcalibur JC proves to be a flexible and powerful tool in the modern recording studio.

A top-notch product that comes highly recommended for professionals and serious enthusiasts alike. The Sonic Farm Xcalibur JC represents a significant achievement in analog audio processing, successfully bridging the gap between vintage warmth and modern clarity.

TEST: Solton MF 200 A

 ## The Solton MF 200 A: A Comprehensive Review of a Multifunctional Active Speaker

In the ever-evolving landscape of audio technology, certain product lines have experienced growth rates that would make even the most audacious financial institutions blush. Among these success stories of recent years, two categories stand out: Swiss Army knives and full-range speakers, which we might aptly term "multifunctional speakers." The unique selling point of this genre lies in its ability to directly manage nearly all signals that could potentially be applied to a monitor or vocal PA speaker.

The advantages of such a system are manifold and immediately apparent. Firstly, it significantly reduces the number of components that need to be transported, wired, and purchased. More importantly, it dramatically decreases the risk of interactive misalignment between various audio components. In an era dominated by the plug-and-play generation, where even the simple task of splitting pre-amp and power amp stages within the signal chain can evolve into a time-consuming afternoon project, this streamlined approach offers an advantage that cannot be overstated.

### Rediscovering Solton: A Blast from the Past

In this review, we encounter a venerable German trademark that has been absent from the audio landscape for at least a decade: Solton. Founded in 1972, the company made significant strides in the solo entertainer keyboard market a decade later. Through its subsidiary, Craaft, Solton also achieved notable success in the bass player segment before seemingly vanishing from the radar of many audio enthusiasts.

Unfortunately, further information about the company's trajectory proved elusive, mirroring the somewhat sluggish nature of the product information available. For instance, I was unable to ascertain the country of manufacture for the MF 200 A unit under review. One might wonder if there isn't a regulation mandating the printing of the country of origin on such products.

### Construction and Technical Specifications

The Solton MF 200 A is an active 2-way speaker boasting an RMS power output of 200 watts at 4 ohms. Internally configured with an 8-ohm impedance, the unit disperses sound through a 1-inch compression driver with a 60 x 40 degree dispersion pattern for high frequencies, complemented by a Craaft 12-inch woofer for low-end reproduction. According to the manufacturer's specifications, this product delivers a sound pressure level of 97 dB (with a maximum SPL of 121 dB) and covers a frequency range from 55 Hz to 18,000 Hz. The cabinet measures 390 x 540 x 350 mm (15.4 x 21.3 x 13.8 inches) and weighs in at a substantial 20.8 kilograms (45.9 pounds). A coarse needle felt coating provides the necessary resistance against scratches and impacts, enhancing the durability of the unit.

The speaker enclosure features an asymmetrical trapezoidal design, allowing for versatile deployment as a standalone solution, a satellite in a larger PA system, a mini side-fill monitor, or even as a floor wedge at approximately 45 degrees. A carrying handle on the top of the cabinet facilitates easy vertical positioning. However, lifting the unit above chest height without readjusting one's grip – for instance, when placing it on a high stand – proves challenging. A mounting flange for stand placement is conveniently located on the bottom of the cabinet. Four robust feet ensure ample stability for the speaker, although it's worth noting that recesses for stacking multiple units are absent.

### User Manual: A Missed Opportunity

The accompanying user manual leaves much to be desired in terms of comprehensiveness and user-friendliness. It consists of a single A4 sheet containing a mere five sentences in both German and English, primarily focused on cautioning users about what not to do to avoid immediate damage to the speaker. Conspicuously absent are example settings, background information on the various input signals, and tips for optimal speaker placement. This sparse documentation suggests that either the manufacturer assumes its target audience consists solely of experienced professionals, or it adopts a somewhat cavalier "learning by doing" approach to customer education. Solton would do well to examine the average manual produced by companies like Mackie for a lesson in comprehensive user guidance.

### Connectivity and Signal Management

The mixer section's connection panel truly lives up to the multifunctional ethos, accommodating all common signal sources. Specifically, it features:

1. An XLR/TRS combo input with volume control and a 2-band EQ
2. A controllable XLR line in/out for daisy-chain cascading
3. RCA inputs
4. An instrument input

This array of connections capably manages the most important signal sources, from microphones to MP3 players, albeit in a somewhat rudimentary fashion. However, it's worth noting the absence of an adjustable microphone preamp, which could be a limitation for some users. Additionally, phantom power is not provided, meaning that condenser microphones must rely on internal battery power.

A low-cut filter at 150 Hz with a 24 dB/octave slope allows the speaker to function as a satellite, delegating bass reproduction to an active subwoofer. The power switch is located adjacent to the IEC power inlet. A Speakon connector enables the attachment of an additional passive 8-ohm speaker, which would reduce the overall impedance to 4 ohms and unlock the full 200 watts RMS capability. Without this extension, I estimate the product's output to be around 130 watts RMS.

### Practical Performance

The MF 200 A scores significant points in monitor mode, particularly regarding its connection panel design. Unlike many competitors, the sockets are angled perpendicularly on the rear of the product when the speaker is tilted at a 45-degree angle, preventing cable kinking. Upon powering up the unit, extraneous noise is minimal, although some noticeable hiss becomes apparent at higher volumes, which is generally masked by the program material.

Sonically, the initial impression is one of pronounced bass deficiency. At first, I suspected the low-cut filter might have been accidentally engaged, but this was not the case. While this characteristic prevents any muddiness in the low end, it also results in a generally pressure-less performance. The manufacturer's claimed low-end extension of 55 Hz seems more theoretical than perceptually accurate. The midrange performance can be described as average, with the upper midrange faring somewhat better. However, the high-frequency driver struggles to deliver truly "silky" highs beyond 5 kHz, leaving the top end somewhat underrepresented.

At moderate volumes, the MF 200 A handles all types of audio signals competently without the need for an additional speaker. However, as the volume increases, a slightly honky coloration becomes apparent. As suggested by the connection panel, this speaker is best suited for moderate volume applications and scenarios requiring quick, uncomplicated management of various sound sources. The dispersion characteristics of the speaker are commendable, with minimal frequency dips observed when changing listening positions.

### Conclusion

The Solton MF 200 A carves out a niche for itself in the market, provided one is willing to accept some minor sonic compromises. Its strengths lie in quick and easy operation, but this comes at the cost of limited sound-shaping capabilities and a notably bass-shy fundamental sound signature.

For solo entertainers, those seeking sound reinforcement for small venues, or users in need of an uncomplicated monitor speaker for moderate volume levels, the MF 200 A is certainly worth auditioning. Its versatility and ease of use may outweigh its sonic limitations for many potential users in these categories.

In the broader context of the active speaker market, the Solton MF 200 A represents an interesting option for those prioritizing functionality and simplicity over absolute sonic fidelity. As with any audio equipment, potential buyers are strongly encouraged to audition the speaker in person, ideally in conditions similar to their intended use case, to determine if its particular blend of features and performance characteristics align with their specific needs and expectations.

Based on the review of the Solton MF 200 A, there are several alternatives that could be considered in the same category of active multifunctional speakers. While the search results don't provide specific alternatives, I can suggest some options based on the characteristics and intended use of the Solton MF 200 A:

1. Mackie Thump series: These active speakers offer similar functionality with built-in mixers and multiple input options. They are known for their robust construction and good sound quality in the budget-friendly range.

2. JBL EON series: These speakers provide comparable features, including Bluetooth connectivity for easy control. They are popular for their clarity and power in small to medium-sized venues.

3. QSC K series: Known for their high-quality sound and durability, these speakers offer similar multifunctional capabilities with a reputation for reliability in professional settings.

4. Yamaha DXR series: These active speakers provide excellent sound quality and versatility, suitable for various applications from live performance to installed sound systems.

5. EV ZLX series: Offering good value for money, these speakers provide clear sound and multiple input options, making them suitable for similar applications as the Solton MF 200 A.

When considering alternatives, it's important to look for speakers that offer:

- Similar power output (around 200W RMS)
- Multiple input options (XLR, TRS, RCA)
- Built-in mixer functionality
- Comparable size and weight for portability
- Options for monitor wedge positioning

It's worth noting that while these alternatives may offer improved sound quality or additional features, they might come at a higher price point than the Solton MF 200 A. The choice ultimately depends on specific needs, budget constraints, and personal preferences in terms of sound signature and brand reliability.

Based on the available information, there are some notable differences between the Solton MF 200 A and the original Solton MF 200:

1. Sound Quality: According to user feedback, the original MF 200 from 1997 is reported to have superior sound quality[2]. The bass is described as deeper, and the overall sound image is considered much better in the older version.

2. Amplifier: The original MF 200 used a different amplifier that included a gain control[2]. This feature is not mentioned in the specifications of the MF 200 A.

3. Power Output: While the MF 200 A is rated at 200 watts RMS at 4 ohms (or an estimated 130 watts RMS without an extension speaker)[2], the power specifications for the original MF 200 are not provided in the search results.

4. Weight: The MF 200 A weighs 20.8 kg according to one source[2], while another lists it at 17.2 kg[3][4]. The weight of the original MF 200 is not specified in the available information.

5. Modern Features: The MF 200 A includes features like a low-cut filter at 150 Hz with 24 dB slope, which may not have been present in the original model[2].

6. Connectivity: The MF 200 A offers multiple input options including XLR/TRS combo, XLR line in/out, RCA, and instrument input[2]. The connectivity options of the original MF 200 are not detailed in the search results.

7. Year of Manufacture: The original MF 200 is mentioned as being from 1997[2], while the MF 200 A is a more recent model.

It's important to note that detailed specifications for the original MF 200 are not provided in the search results, making a comprehensive comparison challenging. The information about the original model is primarily based on user recollection and may not cover all aspects of the speaker's performance and features.

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