Montag, 20. Mai 2024

TEST: Mackie SRM350V2

 Never change a winning team or you can expand your product range. OK, the comparison seems a bit brittle, but the product developers at Mackie must have felt something similar when it came to the SRM series.

Similar to the instrument amplifier sector, there are relatively identical concepts here, which only vary upwards or downwards depending on the type in terms of speaker configuration or power output. For example, the SRM350v2 active box I was testing has its sister model SRM450v2, a product that was supposed to do very well in one of the last tests.

With the 350 version, the priorities have now been placed on better transportability and the product has been significantly reduced in terms of dimensions. Let's see to what extent the little brother can keep up in terms of sound, workmanship and practicality.


Construction:

The SRM350v2 once again uses polypropylene as a construction material, a robust material that has the right mix of hardness (stable construction) and softness (risk of breakage in the event of strong impacts). The dimensions of the construction are height 53 centimeters, width 34 centimeters, depth 31 centimeters, with a weight of 12 kilograms, ensuring that the box can also be easily transported in a small car.

The shape of the box is an asymmetrical trapezoid, which means the box can be used as a stand-alone or satellite solution for P.A., sidefill or as an approx. 45 degree wedge.

A carrying handle on the side of the housing, equipped with an ergonomically placed core, ensures that the box can be easily lifted onto a high stand. A corresponding flange is attached to the underside of the housing.

In general, the box can be described as very handy, as its center of gravity when approached on this handle can be described as very successful. Thanks to an additional recessed grip on the top of the housing, the box can also be quickly maneuvered vertically. 4 strong plastic feet secure the construction against slipping, but there are no recesses on the top for stacking.

You can see that the SRM350v2 can be flown, and appropriate screws can be attached to the top, but the primary focus is on small standing operations. Compared to the 8 rigging points of the 450 version, the 350 has focused on the essentials.



The SRM350v2 is an active biamping system, which consists of a 1.4 inch titanium tweeter horn with a load capacity of 75 watts and a 10 inch woofer speaker with a load of 250 watts. The speakers are powered by a Class D 165 watt woofer amplifier and a 30 watt Class AB tweeter amplifier.

The woofer amplifier is factory-equipped with a dynamic bass boost, which ensures that the frequencies below 70 Hertz are reduced as the volume increases. This not only ensures that the power-consuming frequencies in the high-load range absorb the lion's share of the power, but it also takes into account the fact that the human ear reacts more sensitively to the low-frequency range at high volumes.

At maximum volume, a limiter regulates the power in the soft knee setting at the first distortion; the use of the limiter is indicated by an LED on the back. In collaboration with additional subwoofers, two SRMv2s can be turned into a small P.A. in no time. put together, which also accounts for small club sound systems. It should be noted that the boxes are not suitable for unconditional outdoor use. When used in public, the components must be protected from rainwater or moisture.

The horizontal radiation is given as 90 degrees. The tweeter horn is recessed approx. 10 centimeters into the housing to protect it from damage during transport. A relatively finely perforated, black steel grille, which was secured with four screws, is used to protect the woofer installed on the rear.

The amplifier unit is located on the back of the box. The central power switch is located next to the cold appliance power socket. The input plug is the well-known variant of an XLR female and a 6.35 jack plug, both of which can be locked.

Using a switch, the input sensitivity can be switched between microphone and line level. A simple input controller is used to determine the input sensitivity for the signal applied. Due to the gain of up to +40 db (microphone) or + 5 db (line), dynamic microphones can also be connected directly.

Phantom power is not provided, which means condenser microphones must be operated with an internal battery. When it comes to sound control, the low frequencies below 100 Hertz and the high frequencies above 12 kilohertz can be increased by three dB using a contour switch.

There is an XLR male above the input plug for daisy chain cascading. Additional passive speakers cannot be connected to the amplifier unit. 19 massive black cooling fins ensure sufficient cooling of the unit.


Practice:

When you press the power switch, a bright blue LED on the front of the device reports the operating status. In general, the SRM350v2 can be said to have the same sound attributes as its big 450 sister. Here too you will encounter a soft, three-dimensional reproduction with the “velvety” sound already mentioned. Despite the lack of internal damping, the plastic housing has no resonance frequencies or standing waves.

The spatial reproduction can be described as successful, although of course you always have to take the acoustic conditions of the area of use into account. Due to the good radiation behavior of the tweeter unit, speech intelligibility remains unusually good even with unfavorable reflections and ensures that everyone's faces are happy.

In addition, the box sounds very pleasant even at low volumes; the voice coils do not require a large stroke as with other products to get a voluminous sound from the speakers.

The only criticism of the design is the fact that an angled power plug should be used when operating as a floor monitor, as a straight plug would be bent relatively sharply due to the angle and positioning of the socket.



Conclusion:

As with its big sister, the SRM450v2, the smaller 350 version ensures a harmonious and all-round successful appearance. For use in moderate volume operation, you will find a handy partner that is unobtrusive, can be set up and ready for use in a very short time, with just a few simple steps and extremely clear cable requirements.

Tons of dance and cover bands will appreciate exactly these features, plus you can start with a couple and then upgrade as needed, both in the monitor and in the P.A. Area.

A coherent system with high practical suitability.

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Additional Informations:

## The History of Multi-way Loudspeakers in the P.A. Industry

Multi-way loudspeakers have become a staple in the P.A. industry, providing superior sound quality and versatility compared to traditional full-range speakers. Their ability to reproduce a wide range of frequencies with clarity and detail has made them essential for a variety of applications, from live concerts and theatrical performances to houses of worship and corporate events.

The history of multi-way loudspeakers can be traced back to the early days of sound reinforcement, when engineers began experimenting with dividing the audio spectrum into different frequency bands and reproducing each band with a dedicated driver. This approach offered several advantages over full-range speakers, which were often limited in their ability to reproduce both high and low frequencies accurately.

One of the pioneers of multi-way loudspeaker design was Paul Klipsch, who founded the Klipsch company in 1946. Klipsch was a firm believer in the benefits of multi-way design, and he developed a number of innovative loudspeaker systems that helped to popularize the technology.

In the 1960s and 1970s, multi-way loudspeakers continued to evolve, as engineers developed new materials and techniques for driver design. This led to the development of more efficient and powerful loudspeakers that could handle a wider range of frequencies.

The 1980s saw the introduction of a number of new technologies that further improved the performance of multi-way loudspeakers. These included the use of computer-aided design (CAD) software to optimize driver design, the development of new materials such as titanium and kevlar, and the use of advanced crossover networks to divide the audio spectrum more precisely.

Today, multi-way loudspeakers are available in a wide variety of configurations, from simple two-way systems to complex line arrays with dozens of drivers. They are used in a wide range of applications, from small clubs and bars to large stadiums and arenas.

Here are some of the key milestones in the history of multi-way loudspeakers:

* **1946:** Paul Klipsch founds the Klipsch company and begins developing multi-way loudspeaker systems.
* **1960s and 1970s:** New materials and techniques for driver design lead to more efficient and powerful multi-way loudspeakers.
* **1980s:** Introduction of CAD software, new materials, and advanced crossover networks further improves the performance of multi-way loudspeakers.
* **Today:** Multi-way loudspeakers are available in a wide variety of configurations and are used in a wide range of applications.

Multi-way loudspeakers have played a significant role in the evolution of sound reinforcement. Their superior sound quality and versatility have made them an essential tool for sound engineers and musicians alike. As technology continues to advance, we can expect to see even more innovative and powerful multi-way loudspeakers in the years to come.

## Paul Klipsch and the Pioneering Spirit of Multi-way Loudspeakers

Paul W. Klipsch, a renowned American audio engineer and founder of the Klipsch company, stands as a pivotal figure in the development of multi-way loudspeaker systems. His innovative designs and unwavering belief in the benefits of separating the audio spectrum played a crucial role in shaping the P.A. industry and the way we experience sound today.

**Early Inspiration and the Klipsch Horn:**

Klipsch's fascination with acoustics began during his studies at the University of Arkansas, where he conducted research on horns for phonographs. He recognized the limitations of traditional full-range speakers, particularly their struggles with reproducing both high and low frequencies efficiently. Inspired by the natural amplification properties of horns, Klipsch began experimenting with horn designs that could improve speaker performance.

**The Klipschorn: A Cornerstone of Multi-way Design:**

In 1936, Klipsch unveiled his revolutionary creation - the Klipschorn. This horn-loaded loudspeaker system represented a significant leap forward in loudspeaker technology. The Klipschorn utilized a folded bass horn design to efficiently reproduce low frequencies. This freed up the other drivers in the system to focus on reproducing midrange and high frequencies with greater clarity and detail. It wasn't just a two-way system though.  The Klipschorn incorporated a tweeter and a squawker (midrange driver) alongside the bass horn, effectively creating a three-way design. This approach not only improved sound quality but also allowed for increased efficiency, meaning the Klipschorn could produce louder sound with lower amplifier power compared to traditional full-range speakers.

**Beyond the Klipschorn:**

The success of the Klipschorn solidified Klipsch's reputation as a pioneer in loudspeaker design. He continued to refine and develop multi-way concepts, creating new horn designs and exploring different driver configurations. He advocated for the benefits of high efficiency and accurate frequency response, challenging the prevailing industry norms. His work inspired other engineers and companies to explore the potential of multi-way loudspeakers, leading to further advancements in the field.

**Klipsch's Legacy:**

Klipsch's dedication to multi-way loudspeaker technology laid the foundation for the high-fidelity sound reproduction we experience today. His principles of efficient horn design, accurate frequency response, and meticulous crossover networks have become cornerstones of modern loudspeaker construction. While technology and materials have evolved significantly since his time, the core concepts he championed remain relevant. Today, Klipsch continues to be a leading manufacturer of high-performance multi-way loudspeakers for a variety of applications, from home audio to professional sound reinforcement systems.

**Further Exploration:**

Here are some additional points to consider when exploring Klipsch's work with multi-way loudspeakers:

* **The Importance of Crossover Networks:** Klipsch recognized that simply separating the audio spectrum wouldn't be enough. He developed sophisticated crossover networks to ensure each driver received the correct frequencies and prevent overlap between them.
* **The Klipsch Synergy:** Klipsch believed in a holistic approach to loudspeaker design. He emphasized the importance of integrating the drivers, crossover networks, and enclosure design to achieve optimal performance.
* **The Debate Over Horn Design:** Not everyone embraced Klipsch's horn-loaded designs. Some audiophiles argued that horns could introduce coloration to the sound. However, Klipsch constantly improved his designs to minimize this effect.

By delving deeper into these aspects,  you can gain a richer understanding of Paul Klipsch's groundbreaking contributions to the world of multi-way loudspeakers and their lasting impact on the P.A. industry.

TEST: Mackie S408

 In addition to its main area of application, the production of mixing consoles, Mackie has increasingly invested in the area of sound reinforcement in terms of know-how and implementation in recent years. Little by little, without much fanfare, the entire area from the smallest monitor for keyboard players to the medium-sized club P.A. processed. It is not surprising that Mackie quickly gained a foothold in this segment and, like its consoles, was able to gain a good reputation within a short time.

In addition to the increasingly popular area of active speakers, Mackie also serves the classic passive area. A promising candidate from this segment is the S408 I have, a two-way speaker which, due to its asymmetrical trapezoidal shape, can handle both PA sound reinforcement and be used as a floor wedge.

Construction:

The S408 is a component that is already in the P.A. segment is to be classified. Both the load information and the dimensions go beyond the “singing system use” without negating their use in this area.

600 watts RMS and 2400 watts peak speak for themselves, and you also need at least 2 strong men to adequately move the 81 cm high, 51 cm wide and 46 cm deep, along with its 33 kg weight, or balance it on a high stand.

The case of the S408 is made of 18 mm multi-layered plywood and makes a solid and flawlessly manufactured impression. A PVC layer is used as the outer layer to provide moisture resistance.

A total of 3 handles were installed on the S408, one on each side plus a handle on the top of the case. 4 stacking corners ensure a stable hold in the upright position, its side parts are provided with small rubber feet to also prevent it from slipping when used as a wedge. The box has sufficient rigging points to allow the system to be operated sideways or upright if necessary.

The S408 is a passive 2-way system equipped with 5 speakers, which has 2 Speakon sockets (Input / Thru) as the only connections on the back, through which the systems can be cascaded if necessary. In addition to a high-midrange horn separated at 2500 Hertz for the treble range, Mackie relies on 4 8-inch ferrite speakers for the bass range. The acoustic reason is obvious and is similar to the use of 4x10" bass speakers in the instrumental area.

A larger number of smaller loudspeakers have approximately the same membrane area as a single large loudspeaker and can therefore move the same amount of air, but are significantly more faithful to impulses, have a much faster response and the spatial arrangement allows a much more balanced sound to be generated. That's what happened here too. To ensure more even radiation, two 8-inch speakers were installed slightly to the right and left, respectively, above and below the high-midrange horn.

The box is not insulated on the inside and has bass reflex openings on the back both below and above the connection panel. Even if it is generally known that the box should never be placed in the corner of a room, uncontrolled drone frequencies would undoubtedly be the logical consequence, especially with a rear reflex construction as described here.

According to the frequency diagram, the S408 has a fairly even frequency response between 100 Hertz and up to 10 kilohertz, with only a small dip at 500 Hertz and an over-presence at 6 kilohertz being noted. Below or above these marks, the playability drops significantly.

Thanks to the trapezoidal shape (radius 37.5 degrees), you can cover a vertex radius of 75 degrees when using two S408s next to each other, which promotes very good coverage of the auditorium.

Practice:

The S408 can be used in a variety of ways. Due to its neutral orientation, it can be used as a stand-alone solution, as a satellite solution on a subwoofer, as a sidefill or as a wedge.

When you start using it, you immediately notice the radiation behavior of the box. In fact, you can easily move up to 30 degrees away from the center of the box and still have no loss in frequency range. Both the intelligibility of speech transmission and the fidelity of impulses in music are retained.

The second unusual point is the sound of the box in the bass range. What I have known for a long time from the backline in the instrumental area is in the P.A. Area a bit unusual at first. Anyone who has the indirect, always slightly spongy impulse of a 15-inch or even...
Anyone who is used to the flabby 18-inch model, which has a lot of air but has a catastrophic response due to the design, will need a moment to be able to properly assess and appreciate the fast 8-inch model.
The rapid response initially confuses your hearing, but once you get used to the impulse you won't want to part with it. How sluggish your beloved 15 suddenly seems...

Don't misunderstand us, for example, in order to boost a kick at 70 Hertz for the famous "stomach effect", a whole armada of 408s will not be enough; at least a corresponding number of 10" or 12" in the subwoofer must be involved in the sound formation intervention. However, if you are looking for a very powerful satellite solution or a powerful stand-alone solution for small events, you are on the safe side with the S408.

In general, the box is very powerful in the mids, with good speech intelligibility and pleasing vocal reproduction, both from female and male voices. The box does not have a “whitewashing effect”, which means that the “silky” finish that you know from Mackie's active boxes is not present here. On the other hand, the applied signal is processed in a rather “bony” way, but this does not mean that detailed playback is not possible.

All frequencies that are important for the live range are reproduced evenly without any noticeable dips; the deflections recorded by the diagram are absolutely negligible live.

Conclusion:

With its unusual loudspeaker design consisting of a high-midrange horn and four 8-inch bass loudspeakers, the S408 from Mackie takes a path that is not entirely new but rarely used. Wrongfully so, as the test has shown.

The unusually fast impulse fidelity in the bass range and the excellent radiation characteristics make the S408 an all-purpose weapon in the fight against uneven sound reinforcement. The passive two-way system was particularly able to shine in voice transmission, but without disappointing in music reproduction.

If there was one point at all about the S408 that could be criticized, it was the not excessive, but still high weight, which on the one hand supports the impulse fidelity through minimal housing vibrations, but negates the ability to be transported by just one person.

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Additional Informations:

# The Evolution of Multi-Way Loudspeakers in Professional Audio (P.A.) Systems

Multi-way loudspeakers, or speaker systems that use multiple drivers to cover different frequency ranges, are a cornerstone of modern professional audio (P.A.) systems. They provide superior sound quality and power handling compared to single-driver speakers, making them essential for live sound reinforcement, large venues, and public address applications. This article explores the history and development of multi-way loudspeakers in the P.A. domain, highlighting key innovations and technological advancements that have shaped their evolution.

## Early Beginnings: The Birth of Loudspeaker Technology

### The Mono Loudspeaker Era

The development of loudspeakers began in the late 19th and early 20th centuries. The first practical loudspeakers were mono, full-range units that attempted to reproduce the entire audio spectrum using a single driver. These early loudspeakers were rudimentary by today's standards, with limited frequency response and power handling capabilities.

**1920s: The First Electrodynamic Loudspeakers**

The invention of the electrodynamic loudspeaker by Chester W. Rice and Edward W. Kellogg in 1925 marked a significant milestone. This design, which used a voice coil and diaphragm driven by an electromagnet, became the foundation for modern loudspeakers. However, these early speakers still struggled to reproduce the full range of audio frequencies accurately.

### The Advent of Multi-Way Systems

**1930s: The Introduction of Two-Way Systems**

As audio technology progressed, the limitations of single-driver loudspeakers became increasingly apparent. The need for better sound quality, particularly in terms of frequency response and clarity, led to the development of two-way systems in the 1930s. These systems used two separate drivers: a woofer for low frequencies and a tweeter for high frequencies. This division of labor allowed each driver to operate more efficiently within its designated frequency range, resulting in improved overall sound quality.

**1940s: The Emergence of Three-Way Systems**

The 1940s saw the introduction of three-way speaker systems, which added a midrange driver to the woofer and tweeter configuration. This further refined the audio reproduction by dedicating specific drivers to low, mid, and high frequencies, allowing for even greater clarity and accuracy. These advancements were crucial for high-fidelity audio reproduction in both home and professional settings.

## The Rise of Professional Audio Systems

### Post-War Innovations

**1950s: The Birth of P.A. Systems**

The post-World War II era brought about significant advancements in electronic technology and an increasing demand for public address systems. P.A. systems required loudspeakers capable of delivering high sound pressure levels (SPL) over large areas without compromising on sound quality. This need drove further innovation in multi-way speaker design.

**1960s: The Evolution of Sound Reinforcement**

The 1960s marked the beginning of modern sound reinforcement systems. The rise of rock and roll and large-scale concerts necessitated powerful and reliable P.A. systems. Companies like JBL, Altec Lansing, and Electro-Voice began producing robust multi-way loudspeakers specifically designed for live sound applications. These systems often featured horn-loaded designs to increase efficiency and directivity, making them ideal for projecting sound over long distances in large venues.

### The Transition to Modular and Scalable Systems

**1970s: The Advent of Modular P.A. Systems**

The 1970s saw the introduction of modular P.A. systems, which allowed for greater flexibility and scalability. Modular systems consisted of separate components—such as subwoofers, midrange cabinets, and high-frequency horns—that could be combined and configured to suit the specific requirements of different venues and events. This approach provided sound engineers with the ability to tailor the sound system to the acoustics of the performance space, resulting in better sound quality and coverage.

**1980s: Advances in Speaker Technology**

The 1980s brought about significant advancements in speaker technology. The development of new materials for driver diaphragms, such as Kevlar and polypropylene, improved the performance and durability of loudspeakers. Additionally, advancements in crossover design and digital signal processing (DSP) allowed for more precise control over the frequency response and phase alignment of multi-way systems.

## The Digital Revolution: Modern Multi-Way P.A. Systems

### The Impact of Digital Technology

**1990s: The Introduction of Digital Signal Processing**

The 1990s witnessed the widespread adoption of digital technology in audio systems. Digital signal processing (DSP) revolutionized the design and functionality of P.A. systems. DSP allowed for sophisticated crossover networks, equalization, and time alignment to be implemented with unprecedented precision. This resulted in multi-way loudspeakers that could deliver exceptionally accurate and consistent sound quality across a wide range of frequencies and SPLs.

**2000s: The Rise of Line Array Systems**

The early 2000s saw the emergence of line array systems, which represented a significant leap forward in P.A. technology. Line arrays consist of multiple speaker modules stacked vertically to create a cohesive and highly directional sound field. This design offers several advantages, including improved sound coverage, reduced feedback, and greater control over dispersion patterns. Line array systems quickly became the standard for large-scale concerts and events, with manufacturers like L-Acoustics, Meyer Sound, and d&b audiotechnik leading the way.

### Modern Innovations and Trends

**2010s: Compact and Portable P.A. Systems**

The 2010s brought a focus on portability and ease of use without sacrificing sound quality. Advances in lightweight materials and compact design allowed manufacturers to create powerful multi-way P.A. systems that were easy to transport and set up. These systems often featured built-in amplification and DSP, providing an all-in-one solution for small to medium-sized events.

**2020s: Networked Audio and Smart P.A. Systems**

The current decade has seen the integration of networked audio technologies and smart features into P.A. systems. Ethernet-based audio protocols, such as Dante and AVB, enable seamless digital audio distribution and control over large networks. Smart P.A. systems leverage AI and machine learning to optimize sound quality and performance in real-time, adapting to changing acoustic conditions and user preferences.

## Conclusion

The evolution of multi-way loudspeakers in professional audio systems is a testament to the relentless pursuit of better sound quality, reliability, and flexibility. From the early days of rudimentary mono loudspeakers to the sophisticated, digitally-controlled multi-way systems of today, each advancement has brought about significant improvements in the way sound is reproduced and experienced.

Modern multi-way P.A. systems offer unparalleled performance, capable of delivering clear, powerful, and immersive sound in any environment. As technology continues to advance, the future of P.A. systems promises even greater innovations, ensuring that audiences worldwide can continue to enjoy exceptional audio experiences.

The journey of multi-way loudspeakers is a fascinating example of how engineering and technology can work together to meet the ever-evolving demands of the professional audio industry. From the humble beginnings of the electrodynamic loudspeaker to the cutting-edge line array systems of today, the evolution of multi-way P.A. loudspeakers is a story of innovation, adaptation, and excellence in sound reproduction.

TEST: Mackie ProFX12

 Compact mixer. There's something subconsciously derogatory about the expression alone, don't you think? For market-psychological reasons, the adjective “small” may not be used at all (“Man, he’s small...”), so you decide on an expression that reflects the essence in its basic statement. However, everything is given a subtle upgrade so that nothing seems as it is. You can't imagine how these sales strategies annoy me...

It is not really surprising that an omnipresent company like Mackie also offers some of its products in the small and small class, but it is more surprising that the manufacturer has also been offering some of its products in the budget range for some time. Oh dear, just the word makes experienced engineers cringe. Many people pale as chalk as they remember the merciless cannibalization mistakes of the competition, which relegated their big name to the abyss of second and third class out of pure greed for profit maximization and partly immature Asian production.

A product like Mackie in particular has to be very careful not to mess up the specifications from the former USA production and not to let the workmanship and sound fall below a certain minimum level of quality, even with cheap production in China. The good reputation of the company and the loyalty of the customer will thank you.


construction

The great thing about a Mackie is that if you've ever used a Mackie console, you'll find your way around it straight away. Regardless of whether it is a large F.O.H. console, rack mixer, submixer or project studio, all functional elements are almost always arranged in the same way and are also self-explanatory. With its dimensions of 37 centimeters deep, 35.7 centimeters wide and approx. 9.1 centimeters high, the model can still be classified as a small mixer.

At first glance it is immediately clear that when it comes to design, nothing is left to chance at Mackie. All jack sockets are clearly arranged, adequately labeled and their functionality is self-explanatory. As usual, the housing is made of silver/black anodized and folded sheet metal, impeccably manufactured, but plastic covers have been installed on the side for cost reasons. There are four pleasantly soft rubber feet on the underside of the housing, which offer the product high slip resistance even on smooth surfaces.


The ProFX12 is a 12-channel or 14-channel (channels 5/6 - 7/8 each share a stereo input plus 1 XLR socket) combination of microphone and line mixer, whereby the sound engineer has 6 microphones and 4 Stereo signal paths are available for processing. Channels one to six have XLR female input sockets, which, as is usual with Mackie in this price range, are not equipped with a lock. However, thanks to their upward guidance, the risk of the plugs being unintentionally removed is very low. In addition, the plugs have sufficient support thanks to the slight internal wedging, even without the missing locking mechanism. If necessary, channels 1 - 6 can be supplied with phantom power.

A short stroll through the canals. Below the XLR sockets, which are used to connect dynamic or condenser microphones, there is another input jack socket, which can be connected either symmetrically or unbalanced. Immediately below is the insert path, through which a standard compressor or comparable dynamics processor can be inserted via a Y signal routing. Channel 1 also has an additional selector switch for a directly applied high-impedance signal such as an electric bass. That no one would come up with the idea of connecting a guitar directly here and then complain about the bad, distorted sound ;-)

A low-cut sound filter at 100 Hertz and the gain control are the next steps in the train, followed by a three-band tone control with the center frequencies 12 kilohertz, 2.5 kilohertz and 80 hertz. A panorama control, a mute switch, aux send, FX send and an overload display complete the short description. In the master section, the ProFX12 has the usual Mackie comprehensive features. In addition to the main outs in XLR and jack, we have a monitor out, an FX out, a stereo return, a headphone and a footswitch connection for the internally installed effects device.

Furthermore, a seven-band summing graphic EQ and a USB connection are offered, which ensures that a computer signal can be fed in. There are also four RCA sockets on the panel, which ensure tape-in and tape-out. Two chains, each consisting of twelve LEDs, provide information about the respective water levels.




Practice

In addition to the comprehensive range of connections, Mackie has always been one of the spearheads when it comes to preamplifiers and filters. The first VLZ series was downright legendary, which achieved excellent values in terms of sound culture in relation to its selling price at the time. Even if the selling price of the ProFx12 already suggests that a direct comparison would be made under unfair conditions, unfortunately every Mackie desk still has to be measured against its protagonists.

To put it bluntly, the console performs satisfactorily, although the preamp and EQ have to admit defeat more or less in a direct comparison. Unexpectedly, the preamplifier cuts a consistently good figure. The well-known saturation immediately before the onset of the first clippings could easily be saved in the budget class.

The filters sound more appropriate to the selling price, quite valuable in terms of price, but still rather unspectacular and a bit flat. Although the treble control, which is generally considered difficult, fortunately avoids harsh and scratchy highs, the mids come across as a bit inconsequential without the necessary character of their own. The bass filter also lacks a little volume, but this can be attributed to the budget range used. All in all, the section offers a good cut, which will be sufficient for home recording and live shows, but cannot meet professional standards. However, this was never the approach of the product.

The same applies to the FX section as to the filter area. What is on offer can certainly be described as successful for the selling price, but in detail it can only keep up with the midfield. In terms of processor technology, the spatial depth and resolution of the reverb tails must be able to cope with the algorithms that the manufacturer has given you. These make a passable impression, but nothing more. Once again quite satisfactory for the live area, less so for the ambitious studio area.



Conclusion

Mackie does well, even within the budget range. Despite the overpowering legacy of some of the protagonists of the small mixer scene, the ProFX12 impresses with good workmanship, clear management, good preamps and appropriate filters. In addition to sufficient connection peripherals, the USB connection also makes live recording or feeding in a PC signal child's play.

All in all, neat!

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Additional Informations:

## The History of Compact Audio Mixers

Compact audio mixers have revolutionized the way sound is mixed and recorded, providing a versatile and portable solution for a wide range of applications. Their compact size and comprehensive features have made them indispensable for musicians, sound engineers, and broadcasters alike.

The history of compact audio mixers can be traced back to the late 1970s, when the demand for portable and affordable mixing consoles began to grow. At that time, traditional mixing consoles were large, expensive, and often required specialized training to operate. This made them impractical for many musicians and sound engineers who needed a more portable and user-friendly solution.

In response to this demand, several companies began to develop smaller and more affordable mixing consoles. One of the first companies to introduce a compact mixer was Mackie Designs, which released the CR-1604 in 1978. The CR-1604 was a 16-channel mixer that featured a compact design, affordable price, and easy-to-use controls. It quickly became a popular choice for small bands and home studios.

Other companies soon followed suit, and by the early 1980s, the market for compact audio mixers was booming. Some of the most popular brands of the era included Yamaha, Tascam, and Behringer. These mixers offered a variety of features and price points, making them accessible to a wide range of users.

The development of compact audio mixers was further fueled by the advancement of semiconductor technology. In the 1980s, new integrated circuits made it possible to create smaller and more powerful mixing consoles. This led to the development of even more compact and feature-rich mixers, such as the Mackie 8-Bus Series, which was released in 1988.

The 1990s saw the introduction of digital audio technology, which had a profound impact on the development of compact audio mixers. Digital mixers offered several advantages over analog mixers, including superior sound quality, increased flexibility, and greater ease of use. As digital technology became more affordable, it quickly became the standard for compact audio mixers.

Today, compact audio mixers are available in a wide variety of sizes and configurations, from small 4-channel mixers to large 64-channel consoles. They are used in a wide range of applications, including live sound, recording, broadcasting, and post-production.

Here are some of the key milestones in the history of compact audio mixers:

* **1978:** Mackie Designs releases the CR-1604, one of the first compact audio mixers.
* **1980s:** The market for compact audio mixers booms, with companies like Yamaha, Tascam, and Behringer introducing popular models.
* **1988:** Mackie releases the 8-Bus Series, a compact mixer with advanced features.
* **1990s:** Digital audio technology revolutionizes compact audio mixers, offering superior sound quality and flexibility.
* **Today:** Compact audio mixers are available in a wide variety of sizes and configurations, used in a wide range of applications.

Compact audio mixers have played a significant role in the evolution of music and sound production. Their compact size, affordable price, and comprehensive features have made them an essential tool for musicians, sound engineers, and broadcasters alike. As technology continues to advance, we can expect to see even more innovative and powerful compact audio mixers in the years to come.

TEST: Mackie Onyx 1640i

Analogue is dead! “Oho,” the front rows will shout angrily, “what kind of striking and at the same time false core statement is the ride throwing out there?” And they're right, at least if you don't forget that even with the exorbitant digital craze of the last few years, our hearing functions exclusively in the analogue sector. This means that whatever you force through the I/O level, it always has to be converted twice before we can enjoy the result or fight against the urge to gag out of sheer disgust.

However, what is definitely dead (in the end, not just "smelling strong") is the classic summation of the individual tracks on 2 inch analog tape, no matter how "fat" it was in terms of sound. High operating and maintenance costs, a lack of editing options and massive space requirements have (unfortunately) sent these melodious giants to the museum, where only first-generation musicians like the writer of this article mourn after them.

Aware of this, mixing console manufacturers are also adapting to the appropriate archiving method and are increasingly combining high-quality analog filter technology with a direct converter connection to the computer, so that an external interface is no longer necessary. This is what happened with the Mackie Onyx 1640i I have.




construction

The product, once again developed in the USA and assembled in China, is a 16-channel, or 4 subgroup mixer, which can also be connected directly to the computer via 2 Firewire interfaces if necessary and manages the above-mentioned interface internally . Separately from this, commercially available outputs in the form of XLR and jack, as well as two DB-25 Tascam standard connectors are of course also available.

In terms of approach, the Onyx 1640i will primarily be used at live events, with the option of also being used in project studios due to its fast computer integration. The necessary Windows drivers for Firewire use are supplied on a CD-ROM; as usual, no driver is necessary for OSX because the system uses Apple's internal engine.

When it comes to system construction, you can choose whether to use the mixer as a desk version (front side cabling) or as a flat bed (on top cabling) via the variable use of the connection panel (Rotopad). It is also possible to screw the mixer into the rack using the rack rails provided. All channels have gain-connected inserts, with channels 1 and 2 also having high-impedance input circuitry so that, for example, a passive bass can be fed directly into the console.

In addition to 48 V phantom power and an impact sound filter with 18 dB at 75 Hertz, all input channels have a selector switch for the applied signal, which can be fed either via Firewire 1-16 or line. The following gain control allows a pre-amplification of - 20 dB to + 40 dB, followed by a push switch that allows the pre- or post-equalizer signal to be tapped.

When it comes to filters, the Onyx 1640i has two switchable shelving filters with the classic Mackie operating frequencies of 12 Khz and 80 Hz for treble and bass plus two semi-parametric mids (100 Hz - 2 kHz and 400 Hz - 8 kHz), each +/ - 15 dB adjustable. The product also has 6 aux sends, all pre/post, solo and switchable to Firewire 9-14. The return paths are four times stereo, each with its own signal control.

As talkback you can choose an internal microphone or an external product via an XLR input. The 65 mm faders run a little sloppily and have relatively strong lateral play, but otherwise do their job perfectly.

Practice

When used with Logic 8 on a 24-inch iMac with 2GB of RAM, there's nothing to say, which is ultimately high praise. The forward and return routing of the signal worked as desired and never caused any problems with latency or anything similar. So let's get straight to the most important point of a console, the sound, whereby the collaboration between pre-amplification and filter is particularly important here.

To put it bluntly, the built-in equalizers do a decent job, but they can't keep up with Mackie's VLZ series when it comes to sound quality. The high cow tail still makes a pretty good impression and doesn't let the feared sharpness appear at any time, a real workhorse. The bass range is quite unspectacular, which on the one hand means a practical reference, but is also a little pale in terms of depth and warmth.

I was a little disappointed by the parametric mids. Definitely sufficient for live use, but the filters lack a bit of “silkiness” in studio use. The frequencies are picked up too coarsely if you have screwed the fine fuse into your ear. In order to avoid any misunderstandings, the performance of the equalizers is always very good for the price asked, but anyone hoping to achieve the legendary Mackie sound of the next higher product group may be a little disappointed.

The preamplifier does its job satisfactorily, although the transition to clipping could be a little smoother. What I still appreciate about the VLZ series is the smooth transition into clipping, which is accompanied by very high-quality compression immediately before the first distortion is reached. This wasn't quite as easy to achieve with the Onyx product.

An interesting function in terms of acoustic level adjustment could be found using the solo switch on the headphone input. If you take the signal from the main mix and only listen to it via the solo circuit, the clippings on the gain control show through with a short click, so that you can easily control the maximum level without visual control without the signal being overdriven too much destroy.



Conclusion

With the Onyx 1640i, Mackie delivers a practical mixer that offers a high level of flexibility, good signal routing and, thanks to the built-in Firewire interface, can be used both in the live area and in the project studio. The product impresses with Mackie's well-known clarity and experience has shown that it will meet with open ears from the responsible sound engineers.

Whether live recordings, studio work or whatever, the Onyx 1640i's design means it can be used in almost all production areas, even up to DVD production. However, anyone hoping to purchase the legendary Mackie sound culture of the professional league for 16 channels for under €2000 will unfortunately have to be put off. Both pre-amp and filter do a good job, nothing more, nothing less.

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Additional Informations:

Audio mixers, often called mixing consoles or soundboards, are devices that combine multiple audio signals, blend them, and output the result to various destinations. They are pivotal in sound recording, live performances, broadcasting, and more. The development of audio mixers has a rich history, evolving significantly from large analog consoles to the compact digital mixers we see today. This article delves into the journey of audio mixers, focusing on the innovations and technological advancements that shaped the modern compact mixer.

## Early Beginnings: The Birth of Audio Mixing

### The Analog Era

The origins of audio mixers can be traced back to the early 20th century with the advent of radio broadcasting and the growing need to manage multiple audio sources. The earliest mixers were rudimentary, often homemade by engineers, featuring basic components such as resistors, transformers, and capacitors to blend signals.

**1930s and 1940s: The Pioneers**

During the 1930s and 1940s, radio stations began using more sophisticated audio consoles. These early mixers were entirely analog, using vacuum tubes (valves) to amplify audio signals. They were bulky and required significant manual operation. RCA and Western Electric were among the first companies to produce commercial audio mixers, which were primarily used in radio broadcasting and early film sound production.

**1950s: The Rise of Multitrack Recording**

The 1950s marked a significant leap in audio technology with the introduction of multitrack recording. Pioneered by innovators like Les Paul, this technology allowed multiple audio tracks to be recorded separately and then mixed together. This development necessitated more complex mixers with additional channels and features like EQ (equalization) and auxiliary sends.

### The Transition to Transistors

**1960s: The Advent of Transistor Technology**

The invention of the transistor in the late 1940s eventually revolutionized audio equipment. By the 1960s, transistors replaced vacuum tubes in mixers, leading to more reliable, less bulky, and less power-consuming designs. Companies like Neve, EMI, and SSL (Solid State Logic) emerged, producing iconic analog mixing consoles. These mixers provided superior sound quality and greater flexibility, becoming staples in recording studios worldwide.

**1970s: Modular Designs and Increased Functionality**

The 1970s saw further advancements with the introduction of modular mixing console designs. Engineers could now customize and expand their mixers with different modules for specific functions like EQ, dynamics processing, and effects. This decade also witnessed the rise of live sound reinforcement, with companies like Midas and Yamaha producing mixers tailored for live performances.

## The Digital Revolution: Compact Mixers Emerge

### Early Digital Innovations

**1980s: The First Digital Mixers**

The digital revolution of the 1980s brought about the first digital audio mixers. Companies like Yamaha and Sony were at the forefront, with Yamaha's DMP7 (1987) being one of the first commercially successful digital mixers. These early digital mixers introduced features such as digital signal processing (DSP), automation, and recallable settings, which were groundbreaking for their time.

**1990s: Digital Mixing Goes Mainstream**

By the 1990s, digital mixing technology had advanced significantly. Digital mixers became more affordable and accessible, with models like the Yamaha O2R (1995) gaining widespread popularity. These mixers provided numerous channels, onboard effects, and full automation, making them ideal for both studio and live applications.

### Compact Digital Mixers

**2000s: The Rise of Compact Mixers**

The early 2000s saw a growing demand for more portable and compact mixers, particularly for live sound applications. Digital technology allowed manufacturers to shrink the size of mixers without sacrificing functionality. Compact mixers like the Mackie Onyx series and the Yamaha MG series became popular choices for small venues, home studios, and portable recording setups.

**2010s: Integration and Connectivity**

The 2010s brought further advancements in compact mixer technology, emphasizing integration and connectivity. Mixers now featured USB and FireWire interfaces, allowing direct connection to computers for recording and playback. Additionally, wireless control via smartphones and tablets became a standard feature, offering greater flexibility for remote mixing and control.

**Modern Compact Mixers**

Today's compact mixers, such as the Behringer X32 and the Soundcraft Ui series, offer an array of advanced features. They provide multitrack recording capabilities, extensive DSP, wireless control, and even integration with digital audio workstations (DAWs). The focus is on delivering professional-grade sound quality in a small, portable format suitable for various applications from live sound to podcasting and home recording.

## The Impact of Technology on Mixer Design

### Advancements in DSP and Software

One of the most significant technological advancements in the evolution of audio mixers is the development of digital signal processing (DSP). DSP has allowed mixers to incorporate a wide range of effects and processing options that were previously only available as outboard gear. This includes EQ, compression, reverb, and delay, all integrated into the mixer.

Software advancements have also played a crucial role. Modern mixers often come with companion software that enhances their functionality, providing detailed control over all parameters, scene management, and integration with DAWs.

### Wireless and Remote Control

The introduction of wireless control has revolutionized the way mixers are used, particularly in live sound environments. Mixers can now be controlled remotely using apps on smartphones and tablets. This allows sound engineers to move freely around a venue, making adjustments on the fly to ensure optimal sound quality throughout the space.

### Integration with Digital Ecosystems

Modern compact mixers are designed to be part of a larger digital ecosystem. They often feature USB, Ethernet, and other digital connectivity options that allow them to interface seamlessly with computers, digital recording systems, and other digital audio equipment. This integration facilitates tasks such as multitrack recording, playback, and live streaming.

## Conclusion

The evolution of audio mixers from large, analog consoles to modern compact digital designs is a testament to the rapid advancements in technology over the past century. Early mixers were simple, analog devices, but the introduction of transistors, digital technology, and DSP has transformed them into powerful, versatile tools that are essential in both recording and live sound applications.

Today's compact mixers offer a wealth of features in a small footprint, making professional audio mixing accessible to a broader audience. As technology continues to advance, we can expect future mixers to become even more versatile, integrating seamlessly with emerging digital audio technologies and offering new levels of control and convenience.

The journey of audio mixers is a fascinating example of how technology can evolve to meet the changing needs of users, continuously improving and adapting to deliver better performance and functionality. From the early days of radio broadcasting to the sophisticated digital systems of today, audio mixers have come a long way, and their evolution is far from over.

Sonntag, 19. Mai 2024

TEST: Mackie MR8

 It's the time of near-field monitors!

I'd go so far as to say that you can now get the famous A-wireless device, which almost two decades ago had a massive wall surround, a weight of close to 100 kg and a unit price in the five-digit euro range (mind you, everything PRO box!) attention, can only be seen in a few high-end studios and then only if they can assert themselves due to being part of a music production complex or a corresponding record company.

In the rudimentary rental studio, however, this category of plus/plus monitoring has almost become extinct in terms of its economic viability due to the increasing number of “low-budget” or downright inflationary “no-budget” productions.

But what if you want to do the final mix via a near-field monitor, but the realistic assessment, especially in the bass range, degenerates into a sound-technique game due to the lack of powerful membrane strokes from the home 4-inch workstation PC game monitoring below 150 Hertz and you don't want to use a subwoofer that sometimes sounds undifferentiated?

For this target group, there is a product group that ranges between the “big pants” solution mentioned above and the classic near-field range.

Said monitors are almost always active, have a power range of 100 - 150 watts and, thanks to a sophisticated bass reflex solution and a woofer that is between 8 and 10 inches, can also move enough air in the bass range to generate a qualitatively balanced mix.

The Mackie MR8 monitors I was testing belong to this product group.


Construction:

The individual MR8 box weighs 12.5 kg and measures 276 mm x 408 mm x 346 mm (W x H x D), which suggests a corresponding performance quota for visual reasons alone.

The box has a voluminous bass reflex channel on the back of the housing, which needs to be taken into account when placing the box. In particular, a close-up positioning against a wall or, worse still, in a corner of the room would inevitably lead to a booming frequency or at least generate an unbalanced bass response.

Technically, the system is active two-way, divided into an 8 inch bass speaker with 100 watts of power and a 1 inch dome tweeter with 50 watts of power, both designed for 4 ohms with a maximum SPL of 116 db per pair of speakers.

Due to the active two-way solution, the sound and performance losses, for example due to cables or other passive components, are reduced to a minimum.


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The housing of the MR8 is made of MDF and is painted matt black. A little caution is advised here, as the surface, which is really elegant to look at, reacts very sensitively to scratches or abrasions.

In addition to a cold appliance plug and a slow-blow microfuse, the system's on/off switch is also located on the back. As with the MR8's little sister, the MR5, I would prefer the power switch on the front of the housing so that I can deactivate the product by hand without major contortions and not just use a power strip for this.

Three different standards are available to the MR8 for feeding the signal. In addition to the professional, balanced inputs with XLR female and 6.25 mm jack (TRS), the system also has an unbalanced RCA (RCA) input.

This is very helpful if you are only able to connect the MR8 directly to a standard sound card, which usually only has a signal routing with RCA plugs.

The housing is magnetically shielded and therefore allows it to be used close to a PC screen or TV. The device is also equipped with overheating protection and switches off the power amplifier if the room temperature is too high.

The MR8 was developed in the USA and the product is manufactured in China.


In the settings section you can use two filters to influence the high and low frequency range and configure the final volume. The high frequency filter allows an increase/decrease of +/- 2 db at 5 KHz, the bass range can be increased by 2 db or even 4 db at 100 Hz. This means you can adapt the basic sound of the monitoring to your personal taste, or generally give the mix a higher-pitched or bass-heavy tendency.



Next to the filter settings there is a tiny knurled screw with which the power output of the power amplifier can be configured, either with a Phillips screwdriver or, with a little fiddling, by hand.

The only guide for this work is a slight lock at 12 o'clock; everything else has to be adjusted by ear or, if accessed from the rear, by eye.

In order to set a pair of speakers to exactly the same power output, a classic potentiometer knob would have been better due to its increased readability.


Practice:

When you press the power switch, a small blue LED on the front of the housing lights up and indicates the operating status. The MR8 has no noise of its own, such as background noise or even mains hum, and even with the volume control turned all the way up, the box remains absolutely silent in terms of any background noise.

First of all, let's switch the tone control to neutral playback, i.e. no increases or decreases through the mini switches on the back.

The first sounds that can be heard from the MR8 are characterized by a very neutral reproduction. In fact, we are not dealing with an over-presence of certain frequencies in the sound spectrum; at first glance, the entire reproduction range appears to be very balanced.

Due to our non-linear hearing spectrum with the strong mid-range overemphasis, this area is of course always the crux of the matter by which monitoring must be measured. In addition, this is where the user's personal taste comes into play and can only be evaluated with a high degree of subjectivity.

In my opinion, the MR8 holds its own very well here. Normally, products of a similar design have some problems in the overtone range, especially because of the dome tweeter, but with the MR8 this is more limited than I would have expected.

The Mackie monitors also completed the infamous test with a heavily distorted guitar, which is known for its harsh reproduction in the high-mid range, with an unusually good performance.

Enough of the tester's comments, let's get to the strengths of the speakers. As expected, the MR8 shines with its construction in the treble and bass ranges. The highs are depicted in great detail and manage the balancing act between too “silky” whitewashing and too harsh a resolution.

The bass reproduction of the design is particularly pleasing. I was expecting the classic “inflated” version with its low-frequency over-presence and the resulting sponginess that I know from similar bass reflex designs. Instead, the MR8 shines with a pleasantly tight and extremely faithful reproduction, especially in the double-digit Hertz range.


Conclusion:

Be careful, I didn't expect that! It is actually very rare to find a design that can be said to have features from the next and the next higher price range.

Like any good listening, the Mackie MR8 of course has its own sound, which first needs to be explored and internalized. But if you have done this with some of his reference productions, you will actually find an outstanding monitor in the MR8, which can do more tonally than its price suggests.

A real recommendation!

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Additional Informations:


The history of nearfield monitors is intertwined with the broader history of studio monitors and the evolution of audio technology.

## Early Beginnings

The journey of studio monitors began in the 1920s and 1930s¹. During this period, studio monitors were used primarily to check for noise interference and obvious technical problems rather than for making artistic evaluations of the performance and recording¹. The loudspeaker, which forms the basis of modern studio monitors, was first described by Werner Siemens in the 1870s². Sir Oliver Lodge patented the moving coil loudspeaker in 1898, but it was Rice and Kellogg who developed the first practical design in the early 1920s².

## The Advent of Nearfield Monitors

The concept of nearfield monitors emerged as a solution to the challenges of audio reproduction in studio environments. An unqualified reference to a monitor often refers to a near-field (compact or close-field) design¹. This is a speaker small enough to sit on a stand or desk in proximity to the listener, so that most of the sound that the listener hears is coming directly from the speaker, rather than reflecting off walls and ceilings (and thus picking up coloration and reverberation from the room)¹.

## The Altec Lansing Duplex

The first speaker to be widely adopted for critical monitoring purposes by the nascent recording studio industry was the Altec Lansing Duplex, particularly the 604 model². This Duplex driver was pervasive throughout the studios of the time and the driver itself was in continuous production from 1943 to 1998²!

## The Impact of Technology

The development of nearfield monitors has been significantly influenced by advancements in technology. The invention of the transistor in the 1950s, for example, led to the creation of smaller, more efficient amplifiers. This, in turn, allowed for the development of smaller, more portable studio monitors.

## The Yamaha NS-10

One of the most iconic nearfield monitors is the Yamaha NS-10. Introduced in 1978, the NS-10 started life as a domestic hi-fi speaker, but it was quickly adopted by the recording industry. Its popularity was due to its ability to reveal flaws in recordings, making it an invaluable tool for quality control.

## The Role of Nearfield Monitors Today

In today's digital age, nearfield monitors have become an essential tool in audio production. They are used in a variety of settings, from professional recording studios to home studios, and are valued for their ability to provide accurate sound reproduction at close listening distances.

## The Future of Nearfield Monitors

As technology continues to evolve, so too will nearfield monitors. Future developments may include advances in driver materials and designs, improved amplifier technologies, and the integration of digital signal processing (DSP) to further enhance sound quality.

In conclusion, the history of nearfield monitors is a testament to the ongoing quest for perfect sound reproduction. From the early days of the Altec Lansing Duplex to the modern nearfield monitors of today, this journey has been marked by continuous innovation and improvement. As we look to the future, it is clear that nearfield monitors will continue to play a crucial role in the world of audio production.

## Modern Nearfield Monitors

Today, near-field monitors allow the broadest range of sound frequencies to be played back to the user/sound engineer at a fairly low volume and at a close proximity while maintaining all the sound detail to be preserved and edited³. With more people than ever producing music from home (or away from purpose designed acoustic spaces), demand for high quality nearfield monitors has never been greater⁴.

In conclusion, the evolution of nearfield monitors has been a journey of technological innovation and adaptation to the changing needs of audio production. From their early beginnings to their modern incarnations, nearfield monitors have played a crucial role in shaping the sound of music and audio productions around the world.

TEST: Mackie MR6

 As with the smaller version MR5 and larger version MR8, the American manufacturer is now bringing the popular 6-inch version onto the market in its third edition, the Mackie MR6 MK3. It is no longer possible to determine today whether the former Boeing employee and namesake Greg Mackie, who achieved world fame in the 1970s with his mixing consoles, was aware that he would one day take an established position in the near-field monitor sector. The fact is, however, that especially in the consumer sector, the comparatively low selling prices due to Chinese production have allowed the customer base to grow continuously.

However, it is also a fact that, especially in recent years, other top dogs in the sound reinforcement sector have increasingly entered the market with a budget line, so Mackie really can't complain about the lack of competition. Especially in the highly competitive area up to €200 per box, Mackie has to assert himself against an entire close-range armada from Asia. It remains to be seen to what extent the field can be maintained in this segment.


construction

A single Mackie MR5 MK3 box weighs around 7.2 kilograms and has dimensions of 322 mm x 221 mm x 305 mm (HxWxD), which requires a little planning when it comes to placement on the direct PC workstation. Although the monitor is much easier to handle compared to the large MR8 variant, placing it directly next to the monitor can lead to space problems at one or another workplace. As with almost all representatives of this performance class, it is also important to ensure that there is enough distance from the nearest wall, as the system quickly tends to boom in the bass range due to the bass reflex opening at the back. Conceptually, the Mackie MR6 MK3 system is based on a 2-way system, which, divided into a 6.5 inch bass speaker and a 1 inch tweeter, has an output of 65 watts A/B with a maximum SPL of 112 dB per speaker .

The housing is made of 12 mm thick MDF, with the front panel made of 25 mm. The whole thing is laminated with an attractive, black textured paint, which leaves a good visual impression, but is comparatively sensitive to scratches. Frequently moving the speakers will inevitably leave unsightly sanding marks on the underside of the housing. However, the workmanship is impeccable and no impurities could be found.

Due to the active orientation of the system, as is often the case with the Mackie MR6 MK3, the system's on/off switch is located on the back of the housing along with a power plug and a slow-blow fuse. If you don't want to switch the monitors on and off with a power strip, you should make sure that you can easily reach behind the box from the control position or reach the desk from behind.

The transmission range of the box is specified by the factory as 46 Hz - 20 kHz, although due to the dimensions of the housing you physically have to leave the church in the village. It would be presumptuous to demand linearity below 80 Hertz for this design, so if you have a corresponding requirement in the sub-bass range, you should think about purchasing a corresponding subwoofer. Mackie offers the appropriate model MR10S, which can cover the low bass range down to 35 Hz with a 10" speaker and set the crossover between 40 - 180 Hertz.

In order to ensure maximum flexibility with regard to the signal input, Mackie also offers the unbalanced RCA (RCA) input in addition to the professional signal routing in the form of an XLR female and 6.25 mm jack (TRS). This means that the Mackie MR6 MK3 can also be used in desktop workstations, where internally installed high-quality connections are still in short supply and you usually have to be content with the built-in RCA-based sound cards.

On the back of the housing, in the settings section, you can configure the high and low frequency range to suit your personal taste using two filters. In the treble range, the manufacturer allows an increase / decrease of +/- 2 dB at 3.25 KHz, the bass range can be increased by 2 dB or even 4 dB at 100 Hz. Both filters have a shelving characteristic.

In general, Mackie has addressed and corrected the criticism of the previous models regarding the volume control in the MK3 variant of its MR series. An axis now acts as a potentiometer, which has a grid in the U position at 12 o'clock and on which a standard knob can be positioned for better visual control if necessary. This means that the important, parallel, even volume adjustment of both speakers can be guaranteed much better than with the MK2 versions of the MR5 and MR8 models.


Practice

As with its sister models, Mackie shows an affinity for the color green, as when you press the power switch, the current Mackie logo lights up in green on the front of the housing and indicates the operating status. In the neutral filter position, the Mackie pair produces a balanced basic sound. When it comes to spatial depth grading, the system leaves a good impression; the resolution of the signal can be described as successful.

The bass range is surprisingly powerful and reproduces the frequency range below the 150 Hertz mark comparatively voluminously even without a subwoofer switched on. Mackie actually manages to reproduce the most important mainstream frequencies well in the low frequency range. However, if you work in electronic music with a strong bass bias, you will quickly push the system to its limits and should therefore rely on the support mentioned above.

As is well known, the crossover area is always a bit tricky with a 2-way system. Mackie chose the 3.25 kHz range as the interface for the MR6 variant, which actually promotes a slight dip in the midrange. Although this gives the monitor a pleasant basic sound, it may tempt the technician to place too much of a good thing about midrange in this sensitive frequency range. Appropriately, Mackie also placed his shelving filter in exactly this area, which at least takes away this slight frequency hole, but at the same time increases the treble range.

Nevertheless, the Mackie MR6 MK3 sound very good in its price segment. Particularly in the hearing-sensitive range between 2-4 kHz, the signal present is captured acoustically well and does not fall into the factory-set “loudness level” of many competitors.

The frequency boost should only be treated with caution in the bass range. As a personal sound system, for example in a 5.1 setup, you can switch the boost to experimental mode, provided your lovely wife doesn't use the usual bass veto. As a classic near-field monitor in the studio, I would personally use this area with caution so as not to jeopardize the neutral mix.


Conclusion

With the Mackie MR6 MK3, the American company is introducing a 6-inch variant in its range for the first time, which scores with good to very good values at a comparatively low retail price.

A good spatial separation with a decent resolution means that the Mackie MR6 MK3 looks good in the project studio, in post-production or in the video sector. The system can also be used in surround applications if there is enough space to set it up.

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Additional Informations:

In the realm of audio engineering, precision and accuracy are paramount, particularly when it comes to recording, mixing, and mastering music. Among the essential tools that have revolutionized the way sound engineers achieve these sonic ideals are nearfield monitors. These compact loudspeakers have become ubiquitous in professional studios worldwide, yet their rich history and the innovative advancements that led to their widespread adoption are often overlooked. Let's delve into the captivating narrative of nearfield monitors, tracing their evolution from humble beginnings to their status as indispensable studio mainstays.

**The Early Days: Pioneering Efforts in BBC Studios**

The genesis of nearfield monitors can be traced back to the BBC's legendary Maida Vale Studios in the 1960s. As the British Broadcasting Corporation sought to enhance the quality of its radio broadcasts, engineers grappled with the challenge of accurately monitoring audio in the less-than-ideal acoustic environments of control rooms. Existing loudspeaker technology at the time, primarily designed for home listening, struggled to provide the precision and detail required for critical audio evaluation.

In response to this need, BBC engineers embarked on a series of experiments, utilizing smaller loudspeakers placed closer to the listener. This approach, termed "nearfield monitoring," offered several advantages. By reducing the influence of room acoustics and focusing on direct sound from the speakers, engineers could better discern the nuances of their recordings, leading to more precise mixing and mastering decisions.

**The Auratone 5C: A Legendary Nearfield Monitor Emerges**

Among the BBC's early nearfield monitor prototypes, one particular design stood out: the Auratone 5C. Developed by BBC engineer Arnold Groves in the 1970s, the Auratone 5C featured a 5-inch woofer and a 1-inch tweeter housed in a compact wooden cabinet. Its unremarkable appearance belied its remarkable sonic capabilities.

The Auratone 5C's reputation quickly spread beyond the confines of the BBC, garnering widespread acclaim among professional audio engineers for its ability to reveal even the subtlest flaws in recordings. Despite its limited frequency range and lack of bass extension, the Auratone 5C's unflattering honesty made it an invaluable tool for identifying and correcting audio imperfections.

**Commercialization and the Rise of Industry Standards**

The success of the Auratone 5C spurred the commercialization of nearfield monitors, leading to an influx of new designs from various manufacturers. Yamaha, JBL, and Dynaudio were among the first companies to introduce their own nearfield monitor models, each with unique characteristics and sonic signatures.

As the popularity of nearfield monitors grew, so did the demand for standardized performance criteria. In 1986, the Audio Engineering Society (AES) published its AES-4 standard for nearfield monitor specifications, establishing guidelines for frequency response, distortion levels, and dispersion characteristics. This standardization helped ensure consistency and reliability among nearfield monitor designs, further solidifying their position as essential studio tools.

**Technological Advancements and the Evolution of Nearfield Monitors**

Over the decades, nearfield monitors have undergone continuous technological advancements, driven by the pursuit of ever-greater accuracy and fidelity. The introduction of new materials, such as aluminum and composite cones, improved speaker drivers, and advancements in cabinet design have all contributed to the refinement of nearfield monitors.

Digital technology has also played a significant role in the evolution of nearfield monitors. The integration of digital signal processing (DSP) has enabled more precise control over equalization, filtering, and room correction, allowing engineers to tailor the sound of their nearfield monitors to specific acoustic environments.

**Nearfield Monitors Today: Indispensable Tools for Audio Professionals**

Today, nearfield monitors have become indispensable tools for audio professionals across various disciplines, from music production and recording to post-production and broadcast. Their compact size, precise sound reproduction, and versatility make them suitable for a wide range of applications, from intimate home studios to large professional facilities.

Modern nearfield monitors offer a remarkable level of sonic performance, capable of reproducing audio with stunning detail, clarity, and dynamic range. They have become the cornerstone of critical listening in professional audio environments, enabling engineers to make informed decisions that result in high-quality audio productions.

**Conclusion: A Legacy of Innovation and the Future of Nearfield Monitors**

The journey of nearfield monitors from their humble beginnings in BBC studios to their ubiquitous presence in professional audio environments is a testament to the power of innovation and the relentless pursuit of sonic excellence. These remarkable loudspeakers have transformed the way sound engineers approach recording, mixing, and mastering, ensuring that the music we hear is of the highest possible quality.

As technology continues to evolve, the future of nearfield monitors is undoubtedly bright. With advancements in materials, processing power, and artificial intelligence, we can expect even more sophisticated and versatile nearfield monitors to emerge, further enhancing the capabilities.

TEST: Mackie MR5

 I still remember it well. About 2.5 decades ago there were only three different monitors in almost every professional recording studio. A “big” monitor from various manufacturers, which was always turned up to full volume when the “delegates” from the record company came over to inspect their “product”, the “midrange bomb” Yamaha NS-10 (“what sounds tolerable here works on every speaker”) ) as a near field and the broadband Quaker Audax for simulating the pathetic kitchen radios or similar background sprinklers.

I don't even remember when I first noticed other near-field monitors, the entire professional sector was so focused on the NS-10. Despite their high recognition value, the near-field monitors were still only “Plan B”, as every sound engineer naturally preferred to focus on the sound of their freezer-sized A-monitors sunk into the masonry rather than the practical near-field range.

Times have changed... The segment of advance-eating high-end studios has shrunk to a minimum worldwide due to the massive budget cuts in all areas of the music industry and is now only used by a few artists in the Bundesliga who have more production costs or less irrelevant, used and paid for.

Near-field monitors are now the bread and butter components of every music/video production and sometimes even have to take on the function of A monitoring in project studios. Flexibility, impulse fidelity and frequency linearity are all the more important these days, even at higher volumes, all packed into the most compact, adequately shielded housing possible. Oh yes, and it would also be nice if it were an active system to minimize transmission losses and coordination problems...

Not exactly a small wish list, is it? Let's see whether the Mackie MR5 can meet these demands, as they are rushing into exactly this gap that has been fiercely defended by the top dog Genelec for years.


Construction:

The individual MR5 box weighs 6.5 kg and measures 19.7 cm x 29.2 cm x 26.6 cm (W x H x D), which makes it appear to be an adequate solution even when space is limited . But as we all know, the speaker shouldn't be placed too close to a rear wall, as the bass range changes dramatically, especially since the system's wide bass reflex opening radiates to the rear.

Technically, the system is two-way, divided into a 5.25 inch bass speaker with 55 watts of power and a 1 inch dome tweeter with 30 watts of power with a maximum SPL of 113 db per pair of speakers. I find the relatively small difference in the design of the power amplifier performance interesting, as I have so far increasingly encountered a split in the ratio of 1:3 in relation to the treble to bass range.

The bass speaker has a fairly deep stroke, but is suspended unexpectedly tightly in order to be able to respond to large impulses.



In addition to a cold appliance plug and a slow-blow microfuse, the system's on/off switch is also located on the back. So if you don't want to switch the monitoring on and off with a power strip, you should make sure that you can easily reach behind (!) the speaker from the control position in order to avoid contortionist-like contortions in front of the customer.

Three different standards are available to the MR5 for feeding the signal. In addition to the professional, balanced inputs with XLR female and 6.25 mm jack (TRS), the system also has an unbalanced RCA (RCA) input. Another indication that the MR5 will be used more in the desktop workstation area, where internal professional connections are always in short supply and you usually have to be content with the built-in RCA-based sound cards.

In the settings section you can use two filters to influence the high and low frequency range and configure the final volume. The high frequency filter allows an increase/decrease of +/- 2 db at 5 KHz, the bass range can be increased by 2 db or even 4 db at 100 Hz. This means you can adapt the basic sound of the monitoring to your personal taste, or generally give the mix a higher-pitched or bass-heavy tendency. Or maybe you just want to compensate for the frequency loss in your hearing from your hard “pre-Marshall stack posing” days ;-)

Next to the filter settings there is a tiny knurled screw with which the power output of the power amplifier can be configured, either with a Phillips screwdriver or, with a little fiddling, by hand. Due to the lack of visual control, precise adjustment becomes a matter of luck.

There is a slight indentation in the plastic frame, but you can neither feel it nor see it without direct light. The only way to set it safely without using the flashlight is to “completely off” or “full load”. A simple potentiometer would certainly have been more helpful here.


Practice:

When you press the power switch, a small blue LED on the front of the housing lights up and indicates the operating status. The first listening impression is quite promising, albeit independent. The MR5 is characterized by a very special sound, which is based on the direct Genelec competition in some areas, but without taking over. The well-known soft focus of the Genelec cannot be heard to the same extent on the Mackie monitor, but the indirect bass swing allows for slight tonal parallels.

The spatial depth gradation of the signal is very successful, a fact that can largely be attributed to the excellent tweeter. Detached from the stereo width, the MR5 pair produced a balanced stereo image, which did not lead to any drops in differentiation.

Personally, I liked a linear filter setting the most in terms of sound; you may be able to activate the treble reduction if necessary. When Hohenboost was activated, the sound became too “biting” and too “sharp” for me. In my opinion, an activated bass boost only makes sense if the monitor is placed in a large open space; the risk of “low-frequency slurring” is too great, which pushes the 5.25-inch woofer to its performance limits.

Although the sub-range is transmitted confidently, the associated large strokes of the dome take away the presence of the signal and cause it to lose differentiation even at a moderate volume.

Once you get used to the sonic independence of the Mackie MR5 and adjust your subjective hearing to the monitoring using its reference productions, the product is really easy to work with.

Keyboard-heavy productions, drums and vocals in particular sound very good on the MR5, while distorted guitars suffer somewhat from the dome tweeter, which radiates frequencies very “clearly” from around 4 Khz, and a distorted guitar due to its immense overtone spectrum Very “clinical” sound missed.


Conclusion:

The Mackie MR5 offers a really good price-performance ratio. Rarely have I heard such a “grown-up” sound from such a “clear” and comparatively inexpensive construction.

Excellent spatial separation and a wide range of applications make the MR5 a real alternative in the project studio, post-production or video sector. The MR5 can also impress in surround use due to its very good radiation potential.

All in all a very good product! Recommended!

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Additional Informations:

Nearfield monitors, also known as nearfield studio monitors or simply studio monitors, are an essential tool in the music production process. They provide accurate sound reproduction, allowing engineers and producers to make precise adjustments during mixing and mastering. This article explores the history of nearfield monitors, from their origins to their current state, highlighting key developments, technologies, and influential figures in their evolution.

#### Early Audio Monitoring: The Pre-Nearfield Era

Before the advent of nearfield monitors, audio monitoring in studios relied on large-format speakers typically mounted on walls or soffits. These "main monitors" were designed to deliver high SPL (sound pressure levels) and cover wide frequency ranges, intended for playback in large control rooms. While they were capable of producing powerful sound, their accuracy was often compromised by room acoustics and reflections, making precise mixing decisions challenging.

During the 1960s and early 1970s, the focus was on improving the accuracy of these large monitors. Companies like Altec Lansing, JBL, and Tannoy dominated this era, providing high-quality loudspeakers for professional studios. However, the need for a more intimate and precise listening environment led to the development of nearfield monitors.

#### The Birth of Nearfield Monitors: 1970s

The concept of nearfield monitoring emerged in the early 1970s, driven by the need for more accurate and consistent monitoring in recording studios. The term "nearfield" refers to the placement of the speakers relatively close to the listener, typically within 3 to 5 feet. This positioning minimizes the impact of room acoustics, providing a more direct and uncolored sound.

One of the earliest and most influential nearfield monitors was the Yamaha NS-10M, introduced in 1978. Originally designed as a consumer bookshelf speaker, the NS-10M was adopted by engineers for its unique sound characteristics. Its midrange-forward response and revealing nature made it a favorite for mixing, as it exposed flaws that might be masked on more flattering speakers. The NS-10M's iconic white woofer cone and black enclosure became a staple in studios worldwide, setting a precedent for nearfield monitoring.

#### The 1980s: Growth and Standardization

The 1980s saw rapid growth in the adoption of nearfield monitors, as the benefits of close-field listening became widely recognized. Several manufacturers began developing monitors specifically designed for nearfield use, leading to significant innovations in speaker design and technology.

One of the key players during this period was Genelec, a Finnish company founded in 1978. Genelec introduced the S30, one of the first active nearfield monitors, in 1983. Active monitors, which include built-in amplifiers, offered several advantages over passive designs, including better integration between the amplifier and speaker, reduced signal loss, and greater convenience. Genelec's commitment to accurate sound reproduction and innovative engineering earned them a reputation as a leader in the field.

Another influential company was KRK Systems, founded in 1986 by engineer Keith R. Klawitter. KRK's nearfield monitors, known for their distinctive yellow woofers, gained popularity for their precise imaging and balanced frequency response. KRK monitors became a preferred choice in many studios, further cementing the importance of nearfield monitoring in professional audio production.

#### The 1990s: Digital Revolution and Enhanced Accuracy

The 1990s brought significant changes to the music industry with the advent of digital recording and production technologies. The demand for more accurate and transparent monitoring increased as producers sought to exploit the capabilities of digital audio. This era saw several advancements in nearfield monitor design, driven by the need for higher fidelity and precision.

Mackie, an American audio equipment manufacturer, made a notable impact with the introduction of the HR824 in 1996. The HR824 featured a rear-firing passive radiator, which extended the low-frequency response and improved overall accuracy. Mackie's innovative approach to speaker design and their commitment to affordability made high-quality nearfield monitoring accessible to a broader range of users.

Dynaudio, a Danish speaker manufacturer, also rose to prominence during this period. The BM series, introduced in the mid-1990s, featured advanced driver technology and meticulous craftsmanship, providing exceptional detail and clarity. Dynaudio's emphasis on high-quality components and rigorous testing ensured that their monitors delivered consistent and reliable performance.

#### The 2000s: Advancements in Technology and Customization

The early 2000s saw further advancements in nearfield monitor technology, driven by the proliferation of home studios and the increasing availability of high-quality audio production tools. Manufacturers focused on refining driver designs, cabinet construction, and electronic components to achieve even greater accuracy and performance.

One of the key innovations during this period was the use of advanced materials for driver construction. Companies like Adam Audio, founded in 1999, introduced ribbon tweeters and other novel technologies to improve high-frequency response and reduce distortion. Adam Audio's A7, released in 2007, quickly gained a reputation for its detailed and transparent sound, becoming a popular choice among professional and home studio users.

Another significant development was the integration of digital signal processing (DSP) in nearfield monitors. DSP allowed for precise control over crossover frequencies, equalization, and time alignment, resulting in improved accuracy and consistency. JBL's LSR4328P, introduced in 2006, featured built-in DSP and network connectivity, allowing users to calibrate their monitors to their specific room acoustics using software.

#### The 2010s to Present: Precision and Personalization

The past decade has seen continued advancements in nearfield monitor technology, with an emphasis on precision, customization, and user-friendly features. Modern monitors are designed to meet the demands of increasingly sophisticated audio production environments, providing unparalleled accuracy and flexibility.

One of the notable trends in recent years is the focus on room correction and acoustic optimization. Companies like Genelec and Neumann have developed monitors with built-in room calibration systems, such as Genelec's GLM (Genelec Loudspeaker Manager) and Neumann's MA 1 Automatic Alignment. These systems use microphones and software to analyze the acoustic characteristics of the listening environment and adjust the monitor's response accordingly, ensuring optimal performance in any room.

Another significant development is the rise of compact and portable nearfield monitors, catering to the needs of mobile producers and smaller studio spaces. Models like the IK Multimedia iLoud Micro Monitor, introduced in 2016, offer impressive sound quality and features in a compact form factor, making professional monitoring more accessible than ever before.

#### Key Innovations and Features

Several key innovations and features have defined the evolution of nearfield monitors:

1. **Active Monitoring**: The integration of amplifiers within the monitors themselves, pioneered by companies like Genelec, has become the standard. Active monitors provide better control over the audio signal and simplify the monitoring setup.

2. **Advanced Driver Materials**: The use of materials such as Kevlar, carbon fiber, and ribbon tweeters has improved the performance of drivers, resulting in greater accuracy and reduced distortion.

3. **DSP and Room Correction**: Digital signal processing allows for precise control over the monitor's response, while room correction systems help mitigate the impact of room acoustics, providing a more accurate listening experience.

4. **Compact and Portable Designs**: The development of smaller, high-performance monitors has made professional-quality monitoring accessible to mobile producers and those with limited studio space.

5. **User-Friendly Features**: Modern nearfield monitors often include features such as wireless connectivity, built-in equalization presets, and customizable settings, enhancing their versatility and ease of use.

#### Influential Figures and Companies

Several individuals and companies have played crucial roles in the development of nearfield monitors:

- **Yamaha**: The introduction of the NS-10M in 1978 revolutionized nearfield monitoring, setting a standard for accuracy and revealing sound.
- **Genelec**: Known for their pioneering work in active monitoring and room calibration systems, Genelec has been a leader in the field since the early 1980s.
- **KRK Systems**: With their distinctive yellow woofers and commitment to precision, KRK monitors have become a staple in many studios.
- **Mackie**: The HR824, introduced in the 1990s, showcased innovative design features that improved low-frequency response and overall accuracy.
- **Adam Audio**: Their use of ribbon tweeters and advanced materials has set new standards for high-frequency reproduction and transparency.
- **JBL**: The integration of DSP and network connectivity in models like the LSR4328P demonstrated the potential of digital technologies in nearfield monitoring.

#### Conclusion

The evolution of nearfield monitors is a testament to the relentless pursuit of accuracy and innovation in audio engineering. From their origins in the 1970s to their current state as essential tools in modern studios, nearfield monitors have undergone significant transformations. Advances in driver technology, active monitoring, DSP, and room correction have all contributed to the development of monitors that provide unparalleled precision and flexibility. As technology continues to evolve, the future promises even more exciting developments in the world of nearfield monitoring, ensuring that producers and engineers can achieve the highest levels of audio fidelity in their work.