Anniversary 10

We know this is an unusual choice for speakers. So why do it? Well, it's the real deal! It simply looks and feels way superior to a veneered or painted MDF cabinet. It's more eco-friendly and is just a way higher quality cabinet overall, that can last forever if you treat it well. As soon as you get them in your hands, you will understand what we mean.

We are super proud to say that all of our solid wood cabinets are hand built at our own factory in the heart of Denmark, with wood sourced from a local FSC certified wood supplier. FSC ensures that for each tree used, a new one is planted. Instead of MDF, which is wood dust compressed, using a lot of glue. Solid wood speakers require next to no glue in comparison, which is desirable if you wish to build a more eco-friendly product. Having our own production of cabinets is a big benefit, as we are in full control of the entire process. Like selecting a high quality furniture grade wood that is properly dried before it is treated. This is to ensure it will not contract or expand much in different climates. We are hands-on for the entire process to make sure the quality is just as we want it.

It's rare that we see speakers made from solid wood, as it is a lot more expensive to produce, which is the main reason why most companies choose another solution. Higher requirements for production are also needed, to make sure the cabinets can withstand the test of time, and different climates around the world. One thing to mention here is the way cabinets are put together, where we have a special tool to cut the wood, so the sides are locked together, not only glued like you normally have. We also build the cabinets a bit thicker than we usually do with MDF, and we dampen and brace them differently as well. It's a cabinet that does not provide any more coloration to the sound than an MDF/HDF cabinet would, controversial to what many might believe. From multiple testings, there has actually been a small improvement on resonance and decay control in comparison to MDF/HDF cabinets. To internally dampen the speakers, we use a much more effective and sadly more expensive material than the typical polyfill, named Basotect®. Besides excellent sound absorption properties, this also has the benefit of being fireproof and
OEKO-TEX® certified.

The longevity of cabinets was an aspect we wished to improve on using solid wood. With over 10 years in the business, one thing that is hard to fix on veneered or painted cabinets is the wear and tear cabinets can be exposed to over the years of owning them. Being able to fix scratches or dents on the cabinets is a huge benefit of having solid wood speakers. If you desire, you could even change the appearance of them completely. It's simply a much more forgiving and robust construction in comparison to your traditional speaker cabinets.

The PTT6.5X04-NFA-01 is a 6.5” driver that truly cracks the extended-stroke code. PURIFI’s research has identified the parameters that have so far prevented extended stroke drivers from breaking through in truly high performance audio. The distortion of the acoustic output of a driver is a combination of several separate distortion mechanisms in the motor, cone and suspension. When testing a complete driver with sine waves, a situation often arises where a small tweak seemingly improves harmonic distortion (HD) by letting two mechanisms counteract each other. This always leads to a clear worsening of intermodulation distortion (IMD) that becomes obvious when testing the unit with a more complex signal. This is why traditional HD tests fail to predict subjective sound quality.

PURIFI takes care to optimize the various distortion mechanisms separately, thus guaranteeing that any measured improvement truly reflects a real audible improvement no matter what the signal is. Accurate mathematical models are developed for the motor, suspension and the vibroacoustics of the dome and surround. These models clearly explain several distortion mechanisms in each domain, which provides fundamental insights into better ways of constructing the motor, cone and suspension. Finally, the same models are used to numerically fine-tune the geometry. This way of working reliably reduces multiple distortions mechanisms which secures low IMD for complex signals.

Low Force Factor Modulation

- Prevents voice coil current from modulating Force Factor (Bl). The PURIFI motor completely avoids this classical Achilles’ heel of long stroke drivers. This translates into low intermodulation (IMD): clean, undistorted midrange even in the presence of a heavy bass.

- Equates to low impedance modulation, meaning that drive current is not distorted by cone motion.

Very Constant Force Factor over Excursion

- Prevents voice coil position from modulating the force factor. This is the classical cause of “burbling” i.e. amplitude modulation of the midrange by large low-frequency cone excursions.

Low Surround Radiation Distortion

- The surround contributes to sound output. Conventional surrounds produce distortion as they deform. PURIFI’s Neutral Surround geometry avoids this mechanism without constraining motion. This reduces harmonic distortion as well as intermodulation distortion.

Low Magnetic Hysteresis Distortion

- Hysteresis means that magnetic domains in iron retain traces of previous magnetization / demagnetization cycles which cause distortion when magnetized or demagnetized again. This distortion masquerades as benign harmonic distortion when tested with a sine wave but takes on a crackling or noisy character with more complex signals.

- Removing hysteresis distortion translates into a very “fluid” presentation with excellent front/back separation in the stereo image and a perfectly black background between instruments.

Cleaner than large-cone short-stroke drivers.

- The combination of the above characteristics results in a compact long-stroke driver that delivers complex sound with a clarity and lack of effort that was previously the exclusive province of large-cone short-stroke drivers. Additionally, thanks to the small cone size, the driver has excellent midrange reproduction. It is highly suited for two-way systems.

Tweeter material is often a hot topic in the audiophile community. Soft domes means tweeter diaphragm made from soft material like textile, where hard domes mean hard materials like aluminum. Since the birth of Buchardt Audio we have always used soft dome tweeters for the smooth reproduction, which makes them easy to listen to for hours without any fatigue, something we have become very well-known for. Soft domes are typically easy and “soft” to listen to, but are not the best at retrieving clarity and detail. Hard domes are usually associated with clarity and detail, but often come across as “harsh” and unforgiving. One of the big reasons why a tweeter can be harsh and give listening fatigue, is that hard dome tweeters tend to struggle with very pronounced resonance problems where you would have a big spike in the high frequency response. What is the solution to solve this? Beryllium is one hard material where its resonance frequency is much higher than other hard dome materials. But Beryllium comes with two huge drawbacks. It's very expensive to produce, which results in a high budget allocated to the tweeter alone. It is also an extremely toxic material to work with, and if the dome cracks, then the fumes are lethal to inhale. This also means that it's a requirement that a protected mesh is used in front of the dome, which is not optimal for the sound.

The Buchardt RFA19 (Resonance Free Alu 19 mm) is developed using an aluminum dome that solves the issues associated with hard dome tweeters. It has the benefits of Beryllium, without its drawbacks. It's built on an upgraded motor structure in comparison to our 19 mm soft dome tweeter and uses a very soft radius dome. The combination of its small 19 mm size and dome radius, means that we are able to push the resonance peak all the way up to 30.000hz, which is even higher than most Beryllium dome tweeters. Since we have a tiny 19 mm dome, it's super light and fast as well, which is a huge benefit for reproducing high frequencies with speed and accuracy, without losing power off axis as larger domes tend to do. The result is a tweeter that sounds very effortless, detailed and clear no matter where you are in the room. It does so without ever being harsh or sharp to listen to, which is a part that is very important to us!

Combining this with our perfectly matched waveguide, and you have a match made in heaven! The waveguide contributes to more than what most might think.

1. Ensures that the response and power from the tweeter is the same off axis as on axis.

2. Ensures that the tweeter will not struggle with cabinet diffraction, which helps greatly to make the speakers “disappear” and also benefit to the ease of listening and reduce listening fatigue over time.

3. One unique aspect of our waveguide design is that it's designed to match the contour plot of the woofer down to 1000hz, making the tweeter and woofer work as one driver at the crossover point.   

4. The output of the demanding lower frequencies of the tweeter's response, is significantly boosted acoustically by the waveguide, which allows us to utilize the benefits of such small tweeter domes. Our waveguide benefits more from this than most, as it's very deep.

5. It helps us to phase-align the tweeter and woofer. 

Our goal was to combine the strengths of both the soft dome and hard domes, without any of the negatives. We are super proud to say that this has been achieved to the fullest!

So the good news is that most present-day loudspeakers tend to sound pretty good on-axis. What's also true is that there are still many loudspeakers that perform rather poorly as soon as you move off-axis. Since most research suggests that what we hear in-room is estimated to be around 12% direct sound (from your speakers), 44% Early Reflections (from your room), and 44% Sound Power (how the sound loads your room), we strive to create solutions that'll deliver predictable, consistent, and excellent results both on and off axis – regardless of the living space!

The A10 keeps and controls the directivity from 1000 hz and all the way up.

What this translates to is a non-distorted, evenly distributed in-room frequency response that allows the A10's to sound balanced even at extreme angles, which in turn allows the enthusiast to enjoy sundry benefits like improved imaging, a bigger soundstage, and better transparency. This excellent off-axis response will also result in a drastically widened 'sweet spot' and most importantly, a character that will remain steadfast even as you move around your home as the music plays.

We've found that this trait best suits most people’s rooms and lifestyles. Especially those with untreated rooms. The bottom line? The A10 is capable of delivering exceptional in-room performance in any room.

The Anniversary 10 features the best wireless audio technology available. They can support wireless transfer from any WiSA transmitter with a lossless resolution of 24bits/96kHz and with a synchronization error between the speakers of less than 1 sample with the shortest available latency.

This means that we can ensure that the package of bits stored in the mastering studio where the music is recorded can be transported without any loss directly into the heart of the speakers, where our perfectly matched signal path takes over the processing of the data. The benefit of the WiSA technology is, that not only is the signal transfer lossless, it is also controlled in the receiver side, such that the actually received data corresponds 100% with the data sent out. If not, it is sent again. Therefore, we are 100% sure, our speakers are provided with the exact same signal as input to the system. No coloration or degradation. This pure digital signal (PDS) is fed directly into our quad-core DSP, where we do state-of-the-art processing, to optimize the signals for each individual driver to perfection.

A DSP itself does not do anything. Used carelessly, a DSP will not help you a lot. But used with proper care and deep analysis, a DSP can bring you very far. But – our DSP is much more than just an advanced crossover network between the audio signal and each of the drivers.

DSP equalization perfected

We have 2700 measurements available for each driver, When run through our own advanced analysis tool, it enables us to distinguish the different acoustic phenomena to the degree that we can also judge which nonlinearities to compensate and precisely how, and which not to touch. We will analyze the effect of our EQ on each of the 2700 measurement points around the speaker, and evaluate the impacts in each point, to judge the acoustic effect of a given correction. With properly set targets, the effect of this EQ approach is an improvement so far unheard. With this technology we can very accurately eliminate system resonances, because they will appear equally bad to all directions, whereas, for example some diffractions will only be seen in specific directions and need different compensation. All of this is done with the aspect in mind to make a speaker as linear as possible, with as little compromise as possible to the off-axis response. Therefore, the fundamentally good acoustic configuration helps us a lot to make as little compromise to either as possible. The bottom line is, the A-series audio reproduction is a huge step forward in quality.

Bass extension

Another benefit from the use of active systems is the ability to extend the bass reproduction. We tune our systems to play as deep as we can justify given the amount of power available and which the cone/displacement allows us to do, without any distortion to the system. Our DSP will monitor the excursion and health state of each driver, and will adapt the processing dynamically, to ensure the output from each driver is always non-distorted and trimmed to perfection, regardless of the conditions.

Our use of closed cabinets also effectively eliminates the problems occurring in ported systems, where the port at high SPL’s in many cases will become non-linear and unpredictable in terms of output and behavior. Because of our driver monitoring technology, we can extend the performance way beyond what can normally be achieved with a speaker of this size and structure, compared to more traditional designs. Therefore, do not be fooled by the compact sizes of these babies – they will hit way beyond their size.

In traditional HiFi systems, a lot of effort is often spent finding the right combination of components to meet the optimum performance. In most cases this is done by ear in private homes - and takes up a lot of time and is quite costly with all the component exchange needed. At Buchardt, we decided to do this component matching for the system by scientific measures and systematic evaluation in our lab and listening room, to ensure we made the right choices.

The Amp/DAC used in the A10

Most amplifiers only accept an analog input signal, which means you require a dedicated DAC (Digital Analog Converter) before this stage. The A10s amplifiers work differently, as they accept a digital input, eliminating the need for a dedicated DAC stage. In short, this means you have both DAC (Burr-Brown®) and amplification within the same chip/stage in the A10, effectively shorting the signal path and overall simplifying the system. 

One of the things we were also looking at improving with the A10 was efficiency, heat and noise. You can expect a speaker that does not draw much power at arrange SPL levels or in idle. It also stays cool and generate very little noise. 
 

Automatic ISO 226:2003 Compensation = Low Level Enhancement - LLE

It is well known and documented in research, that the perceived sound balance is dependent on the level at which sound is reproduced. Best example is that with traditional loudspeakers, when listening at quieter levels, there is very little experienced bass and treble reproduction, although at louder level the bass and treble is clearly present.

Since we are in control of the entire sound-reproduction chain of the speakers, we can predict the exposed level of sound at the listeners ears, which means we are able to compensate for this effect completely and very accurately. However, we will only do this compensation at low levels (<70dB), where we know most listeners will not perform their critical listening.

The outcome from this compensation is a smooth and constantly adapting timbre to the sound, so that it will also sound dynamic and impressive at lower levels, where you can have conversations at the same time during listening. This is a very well engineered feature, primarily aimed at professorial use in studios. It allows the mastering engineers to work at lower volumes and still get a precise representation of the music. This allows for much longer work sessions without getting fatigued. For HiFi use, if you often listen at lower volumes this would be a game changer for you!

Now this is cool and unheard of in the industry! With the flexibility of such powerful DSP, we are in complete control of what we want the speaker to do. This means, that we will provide downloadable master tunings to completely change the fundamentals of the speakers sound design.

So what does this mean?

The stock Mastertuning the speakers come with is a very neutral sounding tuning with a flat and very deep response. We think this gives you a very versatile speaker that sounds great under most circumstances. 

But what if your needs are different? What if you rarely play loud and just value crazy deep bass response? Or maybe you are forced to place the speakers near wall boundaries or in a bookshelf? Maybe you just want a more warm and forgiving sound, or more bright and detailed? If you are using them as a Studio monitor in a nearfield setup? What if you want something where deep bass is not important, but SPL is? Maybe you want to use subs with them? There can be many reasons, and we aim to be able to cover all needs with our Mastertunings selections, which also evolves over time.  

You can check out the different Matertunings which can be found under our download section HERE

Here you can see a quick guide on how you change Mastertunings:

With the A10 we have put a lot of effort into optimizing it for nearfield and studio use for both professional and amateur. The A10 has a dedicated Mastertuning for studio use, which makes the transition between the drivers work better in a nearfield configuration. It also allows the speakers to play truly full range, so you can have an all-in-one monitor without a sub. It's extremely flat in its frequency response, has tight and precise bass response, which all results in a transparent speaker, perfect as a high-end studio monitor.

In comparison to our previous active speakers, the A10’s use of digital amplifiers greatly reduces the noise level when speakers are idle.

*We recommend a minimum distance of 1 meter to get the best driver integration.

A thorough engineered room correction that will surprise you in a very positive way, easy and simple to use, yet fixes what you want, without the typical artifacts we sadly see from most room corrections on the market today. Mainly it’s designed to help you with room modes (bass issues caused by the room), it will adapt the bass response to your room, and the speakers placement in the room.

Optimal speaker placement and huge bass traps is rarely an option for most, this might just be the biggest upgrade in terms of sound quality improvements you have done, no joke!

The technical explanation is a huge topic you can read here :

It is well known that the environment in which a speaker is used can change the listening experience significantly. Generally, about listening rooms, the dominating parts are room modes and boundary effects. Our optimization can be done in few minutes – and the data is collected continuously, using the built-in microphone of a smart phone. Please allow us to explain more about them.

Room modes

When music is played in a room, the boundaries of the room will cause sound pressure to be reflected at the boundaries, which is causing a phenomenon often referred to as room modes. These are resonances appearing between walls, ceiling/floor or even in several directions. They will appear at specific frequencies, depending on the room dimension.

These room modes are 3 dimensional issues, meaning that they will appear differently in the room, depending on where you are located. The figure below shows room modes in one dimension of the room e.g. room width. F1 refers to the 1st order standing wave. F2 is a higher frequency mode in the same direction, and the same for f3 and f4 – this is often referred to as the mode order. In theory there are infinite room modes orders, moving upwards in frequency – we will come back to that below

Figure 1 – room modes between two walls.

As can be seen from figure 1, left side, the f1 has a large area along the walls, where there is high sound pressure level (SPL) and in the middle of the room there is low SPL. For a listener, this particular frequency will appear with a lot of change, when walking from wall to wall in this room.

Another thing to be noted on this basis is, that when the mode order is increased, the deviation in SPL comes close and closer together, making it difficult to distinguish each mode – and only small changes in distance will cause big change in SPL. At this point, the room is entering what is often referred to as the diffuse field of propagation – and the room is no longer dominated by well-defined room modes. It will be one big mixture of standing waves in all directions. Normally this transition happens around 250-300Hz. This transition frequency is often referred to as the Schröder frequency – and is unique for each listening room. It is shown in figure 2 below, how the transition from low modal density happens at Fs and changes into a blur of standing waves – diffusion.

Figure 2 - Fs, Schröder frequency identifies where the transition from modal area to diffusion happen.

Boundary effect

Another room-related problem is the boundary effect. This is what appears when you locate a speaker close to a boundary. From factory – most speakers are tuned to be linear; but this tuning is done in a free-field situation with no boundaries. When a boundary appears behind a speaker, the SPL hitting that boundary will be reflected and thrown back to the listener. This will cause two artefacts: bass-boost and interference

1. Since high frequencies are mainly thrown forward and bass low frequencies are distributed all the way around the loudspeaker, most of the energy from the speaker hitting the rear wall is low frequencies. This means a boundary will cause low frequencies to be reflected and thrown back to the listener, while high frequencies will not. This will cause an overweight of low-frequency content from the speaker as it moves closer to the boundary. More boundaries (corners) will cause this effect even more. This is often referred to as the boundary gain effect (BGE).

2. The other thing that happens, when you move the speaker to a boundary is, that the sound reflected at the boundary will come back to the listener, but now with a latency, compared to the original (non-reflected) signal. This means that at some frequencies they will cancel each other – and at some frequencies they will summarize. The boundary causes some interference which is not easily predicted.

Continuous Soundfield Sampling, CSS

The Buchardt room optimization method takes all the above effects into consideration when performing the calibration. When you walk around with the microphone – covering all positions in the room, we can identify each single room mode – and calculate exactly how to attack it precisely. We can identify the boundary effect caused by the loudspeaker positioning and the interference caused by the boundary. We refer to the method as Continuous Soundfield Sampling (CSS). The benefit of this method is that we have more data available than if you only perform discrete points measurements. With discrete points you have no idea if you are in a minimum or maximum of a room mode – and the risk of making wrong corrections on this basis is high.

Correction only where it makes sense

Since the room acoustics only impacts at frequencies below the Schröder frequency, our compensation will only correct frequencies below that. However, we do capture data all the way up to the high-frequency band. This data is used to align the low-frequency output with the high-frequency output. Since the corrections at low frequencies are quite significant, it can sometimes be difficult to see where the ‘natural’ response of the speaker is – for this we use the high-frequency part for aligning the two bands perfectly with each other.

The correction implementation

The mathematics and analysis tools used to analyze the CSS data is quite heavy. But the CPU of a smartphones or tablets will do the job for us. There is no need for cloud-service computers or home PC for this method. We have spent numerous hours optimizing our programming so that the optimizations can be calculated by a smartphone in only a few seconds!! This is thanks to a lot of creative thinking and some ground-breaking research into how an IIR filter can be calculated and ensured to be stable – and still do a perfect fit for our target room EQ response. Once calculated by the phone CPU, the data can be transferred to any DSP and executed with no addition of system latency.

How to do this?

To perform the room correction, you need the dedicated Zen microphone or calibrated iPhones, which is the models 6S to 14. Remember that you only need to do this once, which means, that as long as you have a friend or family member that owns an iPhone (6S - 14), you can do this. If you move the speakers, you need to redo the correction. Only iPhones (6S - 14) and Zen microphones are calibrated for the system, not iPads or Android phones! Note: With the Zen, you can use both IOS and Android 

Our room correction has been calibrated to each iPhone model since the iPhone 6S to iPhone 14. We have chosen iPhones, because they have very low tolerance regarding their quality control. This means, despite poor quality microphones in the iPhones, we can still trust them to deliver the data we need within a very acceptable tolerance for low frequencies. To bypass the noise from these microphones, we are capturing thousands of measurements doing the one minute the measurement process takes. Yeah, it’s a pretty smart team behind this, right? With the newly released Zen microphone, you do get slightly better resolution of the measurements, especially below 40hz. 

To get started, download the Buchardt App from the App store. When you start the room correction via the Buchardt App, you will hear the speakers do pink noise, and see a countdown from 60 seconds. Doing this countdown, walk around with the Zen or iPhone and try to capture as much of the rooms air as possible with large circular motions, but not closer than roughly 1 meter (40”) from the speakers.

When the time is up, you will be shown your room’s frequency response to really get an idea of where your room modes are, within the frequency band. The phone will then use its horsepower to compile all the gathered data to calculate the perfect bass filter. It's directly tailored to your room and the speakers’ placement. You can now switch between "corrected" and "uncorrected" to instantly hear the difference. For most, this will be one of those WOW situations! We are so proud of how well this actually works!