There are two kinds of equalizers – graphic equalizers, and parametric equalizers.

Graphic EQs only allow you to change the gain of each filter – but they do also let you draw a happy face.

The center frequency and the bandwidth for each filter, or “band”, are decided by the product designer. If the graphic EQ is a 31-band graphic EQ, the bands are at ISO standard center frequencies with a standard bandwidth (Q = 4.4). This reduces interaction between adjacent EQ bands.

Parametric EQs let the user define the gain, bandwidth, and center frequency parameters for each filter. For this reason, parametric EQs are the choice of experts — because we can choose the center frequency and the bandwidth needed to solve each problem!

Our first two Audison bit DSPs had 31-band graphic EQs on every output channel. One of those DSPs, the bit Ten, is still in production today.

Our latest, fourth-generation bit DSP amplifiers have 15 EQ bands on each output channel, with parametric control — in addition to 12 input EQs (12 bands each) and a global EQ with 70 effective EQ bands!

Is it really necessary to have 31 EQ bands?

It is indisputable that a 15-band parametric EQ is more powerful a tool for tuning than a 31-band graphic. It’s not even a question. But we still get this request.

Why? Well, our first expectation was that this person must have been running out of EQ bands – but as any experienced vehicle tuner already knows, it’s hard to use 15 parametric bands, especially in fully-active systems. Needing 31 bands per channel in active systems is just unimaginable! Neither the tweeter, the midrange, the midwoofer, nor the subwoofer channels will cover enough notes to need that much work.

As experienced tuners also know, EQing narrow dips in the response above 1500 Hz is a waste of time – especially if they were detected with a single-mic stationary measurement. An inexperienced tuner might feel that they want to do a really good job and get a ruler-flat response by chasing those narrow dips above 1500 Hz, but these dips are really inaudible to us (since they change frequency as our heads or mics move slightly, our brains already know to ignore them).

So why do people ask for 31 bands per channel?

We have wondered this for years, and I used to think that people couldn’t handle parametric EQ, and needed the simplicity of graphic EQ. That’s why we had the Graphic EQ mode in bit Drive – but we still got the complaint.

We think we finally have this sorted. It turns out, these users aren’t really asking for 31 bands, and they aren’t scared of parametric EQ controls. They want the EQ bands to already be in the right locations, and the easiest way to do it is to place them on the familiar 1/3 octave center frequencies, so they don’t have to be moved around! The benefit of a parametric EQ is the opportunity to choose the center frequency, but these users are asking to not have to make that choice (they may or may not know it, but they want that 4.4 bandwidth value as well).

They want it ready to go!

The ISO standard frequencies are exactly what you see in the user interface for the original bit One EQ: 20, 25, 31, 40, 50, 63, 80, 100, etc, etc. 

So, shouldn’t we just “listen to the market” here?

Our hardware limitation here is that we have already used over 100 biquad digital filters for our OEM-correction input EQ (12 bands for each of 12 input channels), and nearly another 100 for our global tuning EQ (5 controls, replicated on each of 14 output channels).

That doesn’t leave enough DSP resources to deliver 31 bands that we don’t really need, on each of 14 output channels. In the image to the right, you can see how the default center frequencies aren’t that helpful with fully-active systems.

What do we do? Do we install 31 bands, even if it means we take capabilities away from our industry-leading OEM correction EQ tools? Not an option – in that area, 12 is barely enough. Eliminate our very popular and useful global EQ? Not doing that, either – users switching from products without a dedicated global EQ tell us how much they love ours.What do we do? Do we install 31 bands, even if it means we take capabilities away from our industry-leading OEM correction EQ tools? Not an option – in that area, 12 is barely enough. Eliminate our very popular and useful global EQ? Not doing that, either – users switching from products without a dedicated global EQ tell us how much they love ours.

What we have done is introduce two new features

Full Time Graphic Mode

With bit Drive prior to 2.1, in Parametric mode, each output EQ filter was visible across the bottom of the screen, and each was controlled by a single interface element box with three controls: Vertical gain slider, rotary Q/bandwidth knob, and rotary center-frequency knob. To use the EQ band, you selected the desired band by clicking on it, then you used the controls within the box. The default bandwidth was 2.2. If you wanted to change the band’s function from EQ peak (the default) to shelf, notch, or all-pass instead, you did that by clicking on the button for that band, and selecting the desired filter type. Then the 3-element control box would change to present the appropriate controls (notch filters and all-pass filters don’t need a gain slider, first-order all-pass filters don’t need a “Q” adjustment, etc).

If you changed the EQ to Graphic mode, now you were presented with 15 gain sliders, with fixed center frequencies and fixed “Q” bandwidth. Now you had no access to the other filter types listed above.

With bit Drive 2.1, we now show 15 gain sliders at all times (see the image below).The three-element control box now loses its gain slider, and now just has two rotary knobs.

We have removed the Graphic/Parametric toggle button, and the default bandwidth is still 2.2. This makes the EQ easier to use, especially if the center frequency of the EQ band happens to be at the desired frequency.

But with 15 bands spread out over the 20-20k range, what are the odds of that? Read on!

Dynamic Band Assignment

We replaced the Graphic/Parametric setting with the Dynamic Band Assignment button. When you click this button and confirm the action, bit Drive checks the crossover frequency for the selected channel. Based on the crossover frequency, DBA will change the center frequencies of all 15 bands to cover the range required (this focuses on the passband, but is not solely limited to the passband – sometimes stop band EQ is required, of course).

For fully-active 4-way systems, DBA will cover all needed frequencies within each channel (tweeter, midrange, midbass, and subwoofer). For active 3-way systems, with a 6.5 playing from 80 to 2500 (for example) DBA can’t quite reach 1/3 spacing on the woofer, but it’s close.

The default for DBA is 1/3 octave spacing – the same spacing used by bit One and bit Ten 31-band EQs. The default “Q” changes from 2.21 to 4.4 (the default “Q” for 1/3-octave EQs) to reduce filter interaction.

At this time, DBA is a channel-by-channel selection. There is no global setting for all 14 output channels but we will add one soon!

If you select DBA accidentally, you can reset the EQ bands to their default values with the reset EQ button for that channel.

It actually does save time!

I’ve now used Full-time Graphic Mode and DBA while tuning several cars.

I have found that those requesting 1/3-octave spacing actually have a point – in modern multichannel systems, it does save me time to have all the bands I need already be in the passband for each channel.

Otherwise, I’m spending time on each channel moving a lot of EQ bands into the right spot.

I’ve never needed close to all 15 bands in any situation, but I’m certainly glad to have had the tools I needed already in position – without losing any crucial capabilities in our Input EQ or Global EQ.

Give Dynamic Band Assignment a try next time you tune with bit Drive – I think you’ll be glad you did!

 

The post How many eq bands do you need? first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

This amplifier delivers 600W RMS @ 4Ω, 1000W @ 2Ω, and 1200W @ 1Ω, based on the proven AF M1D design. It features Audison’s proprietary subwoofer-specific DSP processing, including EQ, crossovers, delay, Dynamic Bass EQ, and Bass Magnification Processing, ensuring a rich musical experience at any volume level. The AF M1.7 bit supports up to 12 input channels, with flexible routing and mixing capabilities through bit Drive software, and includes Universal Speaker Simulator (USS) circuitry for seamless integration with factory sound systems. Additionally, it supports the B-CON High-Resolution Bluetooth streaming receiver, allowing for smartphone control via the B-Con Go app, and is compatible with optional DRC system controllers. The AF M1.7 bit is ideal for car audio enthusiasts seeking exceptional sound quality and advanced control features.

The post The Audison Forza AF M1.7 bit first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

The post New 1.2 bit Drive release first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

Sound waves are pressure changes that travel through air at a given speed (the “speed of sound”). The traditional “squiggly line” is a graph of air pressure at one point in space, as the pressure changes move past a point in space, the air pressure at that point rises and falls. These air pressure cycles make up a wave.

Figure 1: Rising and Falling Air Pressure

The difference between the highest and the lowest peaks of air pressure is the amplitude of the wave. That’s basically how loud it is.

Figure 2: The height of the peaks on the graph indicates the Amplitude

When we talk about sound, we talk about cycles of air pressure rising and falling. Our ears hear this repeated cycle of rising and falling air pressure.

Figure 3: The physical distance between identical pressure values on the wave is the wave length

There is a unit of measurement – the Hertz – which signifies how many “cycles per second” comprise the wave. This is called the frequency.

Each air-pressure cycle has a physical length. This is the distance between the beginning of a cycle, and the end of that cycle and the beginning of the next cycle. That is called a wavelength. The length of a wave varies with its frequency. A 20 Hz wave length is 675 inches, and a 20kHz wavelength is 0.675 inches. A 440 Hz wavelength is 31 inches. (This image is not to scale.)

Figure 5: Wavelengths in the audible range vary significantly

We describe the cycles as if they are circles, since they circle from ambient air pressure, up, then down, and then back to ambient air pressure. We start at 0 degrees, we go through 90 degrees to our peak air pressure, we go through another 90 degrees back to ambient, then another 90 degrees to our lowest point of air pressure at 270 degrees, and then we finally climb back to 360 degrees and ambient air pressure again.

Figure 6: Phase describes a point in the cycle

Figure 7: A cycle is a circle.

Regardless of whether the cycle takes place 20 times a second, or 20,000 times a second, this manner of measuring the cycle in degrees works the same way.  We call this phase. The concept of phase allows us to define where in the cycle we are at any point in time. In car audio, we often discuss phase and polarity interchangeably, but this confuses matters. Polarity is binary, but phase exists on a spectrum.

If we do something to change the phase – if we cause phase shift – we are changing where the pressure wave is at a given point in space, without changing the frequency. A wave can be phase shifted in many ways.

One way is to reverse the (+) and (-) wires – the polarity. This shifts phase 180 degrees (for steady-state signals).

Figure 8: Polarity is binary.

When you have one sound wave, phase changes are really hard to hear. If you play music over a single speaker, and you reverse polarity (which forces a 180-degree change in phase at every frequency), no one can determine which connection is correct and which one is reversed. There is no absolute polarity.

Similarly, researchers have manipulated the phase of signals at specific frequencies, and humans have been unable to hear the difference when listening to music over a single speaker. This surprised the researchers, who expected to measure how phase distortion affected music playback, and could not provide evidence that it did! (It is true that with test tones over headphones, humans can hear phase manipulations – but not with music over speakers).

However, when we have multiple sound waves at the same frequency – such as when more than one speaker plays the same sound – things get more complex.  The two pressure waves add together at our ears. The intuitive expectation is that two speakers are louder than one speaker, but anyone who’s put two subwoofers in a box and inadvertently wired one incorrectly knows that this is not always the case.

Figure 9: Two subwoofers, wired out of polarity.

When two speakers play the same note, and they are the same distance from us (such as the two subwoofers above), the waves arrive at the same time, AND they are aligned with each other. (When sound waves travel the same distance since they both travel at the same speed, it takes them the same time to arrive at their destination.) This means they are aligned in phase.

Figure 10: Two waves aligned in phase.

When the waves arrive aligned, they add together to a larger wave, with taller peaks of air pressure and deeper dips of air pressure (greater amplitude).

Figure 11: Two waves, aligned in phase, summed.

When the waves arrive at different times, the waves may be misaligned to some degree. This means they are “out of phase” to some degree (phase is a relative term, not an absolute – that’s why we measure phase in degrees).

Figure 12: Two waves, slightly misaligned in phase due to distance.

So, let’s say that the signal is a 440 Hz tone. The wavelength for 440 Hz is about 31 inches.

If the two speakers are both 31 inches away – the same distance away from us – the sounds arrive at the same time, and the waves are aligned in phase. They add together to one louder sound (+6dB louder). This explains why we expect two speakers to be louder than a single speaker.

Figure 13: Two identical sound waves emanating from the same point, in phase.

If one speaker is one wavelength farther from us than the other – if one speaker is 31 inches farther from us than the other – the two waves will still be aligned, and they will sum together to one louder sound. They will add to almost 6dB (the sound which travels farther will be ever-so-slightly attenuated by its 31-inch-longer path).

Figure 14: Two 440 Hz waves (Middle C) emanating from two different points in space, on 31 inches farther from the measurement point, arriving in phase.

The difference in these distances is called a path-length difference, and we will sometimes use this term to describe the relative positions of two speakers.

If the second speaker is farther away than the first speaker, and that path-length difference is a whole multiple of the wavelength – if one wavelength is 31 inches, then two wavelengths is 62 inches, and four wavelengths is 124 inches – then the waves are also aligned in phase, and they still get 6dB louder when they add together.

But what if the path-length difference is not a whole multiple of the wavelength? What if it is less than one wavelength?

Let’s start with the path-length difference being one-half of a wavelength. In that situation, the second wave is only halfway through its cycle, halfway past the measurement point – so it’s 180 degrees out of phase with the first wave.

This is as misaligned as they can get, so when the two waves add together, we get destructive interference and near-complete cancellation.

There’s no resulting pressure change at all!

Figure 15: Two identical waves 180° out of phase, and the resulting sum being near-complete cancellation.

This accounts for how two subwoofers cancel each other out when we inadvertently wire one backward (remember, inverting polarity forces a 180-degree phase shift).

“Well, this is good to know in theory”, you might say, “but music consists of many, many waves, at many various frequencies, all at the same time. And most of the time the second speaker isn’t the same distance, or a multiple of a wavelength, or a multiple of a half wavelength. Most of the time it’s somewhere in between.”

Yes, it is usually somewhere in between – since every frequency has a different wavelength, every speaker pair with a path length difference is aligned at some frequencies, slightly misaligned at some others, and badly misaligned at a few! This means that some frequencies reinforce each other and get louder, but some cancel each other and get quieter. Here are three tables to show what happens.

Table 1: Path-length differences and degrees of phase misalignment

Identical distances  = 0 degrees

0.125 wavelength   = 45 degrees

0.187 wavelength   = 60 degrees

0.25 wavelength     = 90 degrees

0.5 wavelength       = 180 degrees

0.75 wavelength     = 270 degrees

One wavelength     = 360 degrees/0 degrees

Table 2: Summing two identical waves together

0 degrees phase shift     = +6dB

45 degrees phase shift   = +5.65dB

60 degrees phase shift   = +5.35 dB

90 degrees phase shift   = +3dB

120 degrees phase shift = 0dB

150 degrees phase shift = -3dB

180 degrees phase shift = -30dB

210 degrees phase shift = -3dB

240 degrees phase shift = 0dB

270 degrees phase shift =  +3dB

300 degrees phase shift = +5.35db

315 degrees phase shift = +5.65 dB

360 degrees phase shift = +6dB

Testing has shown that amplitude differences of less than 3dB are not apparent as “louder” or “quieter” to the human hearing system. Those less-than 3dB differences sound like tonal changes, but are not noticeably louder or softer. Changes of less than 1dB are very subtle and difficult to discern reliably.

Two identical waves summed together, and aligned, result in +6dB increase in amplitude. That is our baseline expectation for the gain we get when summing two identical waves – or when using two speakers instead of one.

When we sum two signals, it’s often by playing two speakers, and when we use two speakers, we expect that +6dB of gain. If we don’t get it, we are probably wasting money on speakers and on amplifier power (remember, +3dB requires 2x the amplifier power, and -3dB is what we get when we lose half our amplifier power – so a 3dB swing is a very important change in our acoustic result!)

Here is what we get when path-length differences are present.

Table 3: Misalignment measured in wavelengths and its effect on the sum

Same path length  = gain of +6dB (the expected result from summing)

1/8th wavelength  = gain of 5.65dB (failed to gain an expected 0.35dB)

3/8  wavelength     = gain of 5.35dB (failed to gain an expected  0.65dB)

1/4 wavelength      = gain of 3dB (failed to gain an expected 3dB)

1/2 wavelength      = gain of -30dB (failed to gain an expected 36dB)

3/4 wavelength      = gain of 3dB (failed to gain an expected  3dB)

5/8 wavelength      = gain of 5.35dB (failed to gain an expected .65dB)

7/8 wavelength      = gain of 5.65dB (failed to gain an expected  0.35dB)

1 wavelength          = gain of +6dB (we again get the expected result)

Here are some visual examples of these misaligned waves adding together:

0 degrees phase shift = sum to +6dB

Figure 16: These sum to +6dB

Figure 17: These sum to slightly less than +6dB.

Figure 18: A 60-degree misalignment still has less than a 1dB impact.

Figure 19: A 90-degree misalignment loses us 3dB of the potential 6dB (which is half of our power!

Figure 20: At 120° of misalignment, there is no increase at all.

Figure 21: At 180° of misalignment, the signal is nearly completely canceled.

Figure 22: A complete wave length – 360* of misalignment – sums to nearly as much as no misalignment at all!

Every time the same sound arrives to our ears from more than one location, we have a potential increase in amplitude – or a potential decrease.

And this means that whenever two speakers play a wide audio range, some notes will get louder and some will get quieter – unless the two speakers are the same distance away from us. (This problem is not usually noticeable with subwoofers in cars, since the path length differences for subwoofer drivers are a small fraction of the wavelengths the speakers are playing – but if we let our subwoofers play midrange notes, the problem would affect them too.)

Fortunately, only the worst misalignments result in near-complete cancellation. Slight misalignments have been shown to be inaudible, and moderate misalignments – while they should be avoided – are not disastrous. It’s the worst misalignments – those which result in significant loss of total output – which must be avoided at all costs.

As a good friend says, “It’s not important that we be perfectly in phase, but it is very important that we not be perfectly out of phase”.

Here’s an animation that illustrates this wonderfully:

https://www.acs.psu.edu/drussell/Demos/superposition/interference.gif at https://www.acs.psu.edu/drussell/Demos/superposition/superposition.html

In car audio, these multiple arrivals can result from four different causes:

1. You have left and right speakers, and you play a stereo recording with content meant for the center of the stage. To accomplish this, the recording engineer puts the content in left and right channels equally, in phase.
2. You have rear speakers, and they play the same sounds as the front speakers.
3. You have a multiple-element speaker system with crossover filters, so the low-passed speaker and the high-passed speaker play the same sounds in the overlapping transition band of the crossover filter network.
4. A reflected sound in the cabin arrives later than the direct sound.

Figure 23: Stereo Path Lengths, Rear-Speaker Path Lengths, Crossover-Transition Path Lengths, and Reflections

What does this do to our sound?

A great deal of damage, it turns out. Fortunately, it turns out that we cannot hear multiple arrivals as multiple arrivals – as echoes – until the path-length differences involved are much longer than can fit inside a car. This is what lets us install multiple speakers without hearing echoes, but it has other effects.

Some notes get louder, and some get a lot quieter. Multiple arrivals create a pattern of peaks and valleys in our frequency response – the peaks due to the constructive reinforcement of some waves adding together to higher amplitudes and getting louder, and the valleys due to the destructive cancellations from other waves combining to lower amplitudes. This pattern is called a comb filter.

Here is an example. We have measured two sources of full-range pink noise, in phase, arriving at the same time. The red and blue lines measure Channel 1 and Channel 2, and the green line is the sum of 1+2.

Figure 24: The red and blue traces are two identical-response channels, and the green is the sum of the two when the two are aligned in phase and time (reflections eliminated).

In this example, the combination, or sum, is +6dB greater – at every frequency –  than either signal alone. This indicates that the two signals we summed together are in phase at all frequencies. This is a simulation – in real life, we never get this perfect a result.

And here is an example of the same two signals, summed together, after one travels 27 inches of distance relative to the other, and is delayed 2.01 mS by this path-length difference. As you can see, there is significant loss of amplitude, and the frequency response is significantly damaged.

Figure 25: The comb filter created when the same two traces are summed together, when one is delayed by 2.0mS (or, 27 inches).

Below about 100 Hz, it seems the expected 6dB increase can be seen, as well as at 500 Hz, 1000 Hz, etc. (In reality, the phase is not perfectly matched and the increase is a fraction below +6dB.) However, there is massive signal loss at 250 Hz, at 750 Hz, at 1250 Hz, etc. Above 5000 Hz, the individual cancellations are no longer visible, even on this high-resolution measurement – but a close examination reveals that the average sum is still only half of the expected +6dB.

Knowing the phase by seeing the sum

If we have two signals which have the same amplitude, summing them together can tell us how aligned the two signals are at any given frequency!

Figure 26: The sum tells us the phase offset.

In the diagram above, we know the two signals are aligned in phase if the sum is +6dB, while if the sum is -30dB, we know the two signals are 180° out of phase – and so on. All these various phase shifts are the result of one 27” path-length difference. The path length difference affects different wavelengths differently. Refer to Table Two and Three above for more information.

What do we do about this “multiple-arrival” cancellation problem?

Do we have to calculate and predict the effects of path-length differences for every wavelength for every frequency from every speaker in the system? The audible range covers at least 20 Hz to 20,000 Hz!

No. There is actually a much simpler approach. We can use distance-based delay to address these first three causes of multiple arrivals at a given listening position (that is, left/right, front/rear, and high/low interactions).

The distance-driven delay process calculates the delay to apply to each channel based on the path-length differences, and the flight times for the sound from each speaker to arrive at the listening position. Once all the absolute speaker distances are entered, the differences are calculated, the flight times are calculated based on the speed of sound, and all the channels are delayed by an appropriate amount. The signal for the speaker farthest from the listener is not delayed at all – we delay the signals for the others to align them with the signal coming from the one farthest away. It’s very straightforward, and very effective.

This does require our signals to be aligned in phase and time when we start, of course.

It also works for one listening position, but not for multiple listening positions. If we want to have great sound in multiple listening positions, we must use other approaches to manage the problem of phase cancellations resulting from path-length differences.

How precise must we be when measuring the distances?

We need to be accurate, but in terms of precision, microscopic differences in delay don’t result in massive changes in fidelity.

We do not have to be overly precise when we apply delay in the low bass frequencies, because the wavelengths involved are very long, and as we saw above, small misalignments result in very small amplitude differences and are not audible. An 80 Hz wavelength is 166 inches. One-eighth (0.125) of that wave length is 20 inches. If we look at the tables above, a 0.125 wavelength misalignment results in the sum being 5.65dB greater in amplitude, rather than 6dB. So if we made a measurement error of 20 inches – which would be a very poor measurement – we would fail to attain that potential 0.35dB!

We do not have to be overly precise when we apply delay to the treble frequencies, either – since these wavelengths are impossibly short, making misalignments appear and disappear with every minute movement of our heads. The wave length of 5000 Hz is just under 2.7 inches. The half-wavelength – where phase would be 180 degrees different – is 1.35 inches. We move our heads that much (and more) all the time, without noticing acoustic aberrations. Our hearing systems learned long ago to ignore narrow deviations once we get above the frequency where a sound would be 180 degrees out of phase at one ear relative to the other ear (and that is around 1500 Hz).  The result is that treble misalignments are not audible as individual cancellations (they are audible as lost amplitude, however, and that amplitude can be made up for in other ways).

When it comes to achieving fidelity in a car, using delay to overcome path-length differences is an important tool, but resolution in these measurements past a certain point is certainly not a big deal. It’s far more important that we verify that our signals are in phase when we start, especially OEM signals. OEM signals are rarely aligned in phase and time any longer, so testing and correcting these signals is a vitally-important topic for another day.

Does that mean that delay “puts everything back in phase”?

While that would be great, delay doesn’t magically fix every phase misalignment.

Some in our industry have even complained that distance-based delay doesn’t work, because after it’s applied, there are still phase misalignments remaining.

Properly-applied distance-based delay eliminates the comb filter cancellations caused by path-length differences, in one listening position. That’s all it does.

Does that mean that tape measures are not useful in predicting phase cancellations? Of course not!

What else causes phase misalignments?

• If the signal you’re starting with has phase and time nonlinearities
• If you set your crossover parameters problematically
• If your OEM signal has uncorrected phase and time processing
• If you wired an signal input out of polarity in error
• If you wired a speaker output polarity in error
• If there are reflections (which there always are)
• Any change in amplitude (such as a crossover filter)

Distance-based delay won’t address these issues. If one or more of these problems exist in your system, and you apply distance-based delay, those problems will still exist in your system. That doesn’t mean delay’s not a powerful and valuable tool in achieving great sound, or course – it simply is a great tool to address the problems caused by path-length differences.

What about reflections?

“Direct sound” travels directly from a speaker to our ears, taking the shortest possible path. “Reflected sound” takes a longer path – it travels first from the speaker to a reflective surface, reflects off of that surface, and then travels to our ears. For this reason, reflected sounds arrive later than direct sounds. Once the same sound is arriving at two different times, we have multiple arrivals!

Fortunately, the farther sound travels, the more it is attenuated (that is, some of the initial amplitude is lost). So, reflected sounds are usually less powerful than the direct sound, and that means that phase cancellations are not as severe as they can be when both sounds are the exact same amplitude. For the worst cancellations to occur, the two sounds must be similar in level. This is one reason that reflections are the least crucial cause of multiple arrivals.

Reflections are a part of every listening room. We expect reflections – if we magically eliminated them, the sound would be unpleasant to our ears. For the purposes of this exercise, we will accept the effects of reflections as a cost of doing business. It turns out that we can achieve wonderful sonic results without worrying about the phase effects of reflections very much.

Best Practices

So, best practices include:

• Verify the phase and time integrity of the signal you’re starting with
• Correct the signal’s phase and time nonlinearities before tuning
• Set up the system in such a way that QC catches any wiring errors
• Configure crossovers so that phase errors are not baked into the result
• Do not go out of your way to create reflections by getting overly creative on your speaker locations. (Some complex speaker installations end up making reflections worse than the OEM locations!)

Once you follow these best practices, setting delay based on speaker distances can be a very powerful – and very simple – tool to deliver great sound in a car.

The post Waves, Distances, Phase, and Delay in Cars first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

At Audison, our commitment to delivering unparalleled sound quality and innovation in-car audio systems remains unwavering. We are thrilled to present our latest masterpiece – the Audison SR 6.600 6-Channel Amplifier with Hi-Res Audio Certification. This amplifier is engineered to redefine your in-car listening experience, setting a new standard in audio excellence.

 Unleash the Power of Hi-Res Audio

 The Audison SR 6.600 is not just an amplifier; it’s a gateway to the world of high-resolution audio. With Hi-Res Audio Certification, you can expect audio playback that preserves every intricate detail and nuance of your favorite tracks. Whether you’re a music enthusiast or a professional audiophile, this amplifier will breathe life into your music collection, allowing you to hear it as the artist intended.

 Powerful Performance, Distortion-Free Sound 

Harnessing cutting-edge technology and over four decades of expertise, the Audison SR 6.600 offers uncompromising power and precision. With six channels of amplification and two channels with High Current Capability,  it can effortlessly drive your entire car audio system, ensuring a balanced and immersive soundstage. Our advanced engineering minimizes distortion, so you can enjoy clean, distortion-free audio even at high volumes.

Versatility Meets Innovation

Versatility is at the core of the SR 6.600’s design. The Fully Bridgeable architecture lets it operate in 3, 4, 5, and 6-channel modes. The two High-current-capable channels can deliver 300W of constant RMS power. It features a flexible crossover network with a combination of high-pass, band-pass, and low-pass filters to manage your entire active speaker system. 

Built to Last

 At Audison, quality and durability are non-negotiable. The SR 6.600 is built to withstand the rigors of the road while maintaining its exceptional performance. Its robust construction ensures longevity, so you can enjoy your enhanced audio experience for years to come.

Seamless Integration

 We understand the importance of a seamless installation process. The Audison SR 6.600 is designed with ease of installation in mind. Like the other members of the SR family, the SR 6.600 uses Universal Speaker Simulator technology on all its speaker-level inputs, automatically making OEM muting protection a thing of the past. The speaker-level inputs can handle up to 40 volts of OEM signal. Make your OEM upgrade trouble-free with Audison! 

 Elevate Your Sound Today

The Audison SR 6.600 6-Channel Amplifier with Hi-Res Audio Certification is a testament to our unwavering dedication to audio excellence. It’s more than an amplifier; it’s an invitation to embark on a sonic journey like no other. Elevate your in-car sound experience to new heights with the Audison SR 6.600.  Discover the difference. Experience the power of Hi-Res Audio. Upgrade to Audison today.

FEATURES

• High-Current Capability delivers high power into the most difficult loads (1Ω Stereo, 2Ω Bridged, for 5/6)
• Staggered power optimized for 3-way active front with higher power for woofers, or 600W of bridged power for a sub
• 5 ch output operation mode to create a five-channel system, with 2Ω subwoofer.
• Flexible crossover filter networks deliver Active 3-way, Front/Rear/Sub, or six channels of high-passed power
• ADC (Audison D-Class) Technology 2Ω stable output providing Hi-Res certified audio
  performance with superior efficiency in a compact size.
• Extruded aluminum compact design with a fanless convection cooling system. 
• Built-in USS (Universal Speakers Simulator).
• Balanced high-noise rejection inputs for Speaker-In (for OEM head units) and RCA-In (for
  Aftermarket head units).
• ART (Automatic Remote Turn-On/Off) automatically turns on/off the amplifier when the OEM head-unit turns on/off (can be enabled/disabled, with Speaker-In
  inputs).
• Accepts up to 40-volt input signal (Speaker-In).
  Bass Boost(50 Hz) adjustable up to 12 dB, to increase the subwoofer punch. 
• Lo-Pass and Hi-Pass 12 dB adjustable (50 Hz to 5000 Hz) filters, providing the ability to build a 3-way system, a front 2-way + rear system, or a front 2-way + one powerful subwoofer (2Ω).
• Optional VCR-S2/ARC 01 Remote Volume Control provides the ability to adjust the OUT 5/6
  volume level from the dashboard.

The post The SR 6.600 first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

To Mr Emidio Vagnoni, “the father of Audison”, all audio amplifiers needed to be able to make rated power on all channels simultaneously. Anything less was a compromise.

Some amplifiers today purposely “throttle back” some channels compared to others, deciding which channels need to make their rated power based on how they are being used.

Now, designing an amplifier with purpose-specific channels is one thing. Audison has produced many of these designs, such as the groundbreaking AV 5.1k 5-channel amplifier and the LRx 6.900 six-channel amplifier.

In those cases, Audison rated each channel appropriately so that the customer understood how much power was able to be counted on as available. No matter how we think amplifiers may be used, our dealers always find innovative system designs that break new ground. The best amplifier is one which is flexible.

We are surprised to learn that some “modern” amplifiers list one power rating for appearance’s sake, but deliver significantly less power on most channels, depending on the use case. We have even had dealers tell us these amplifiers make this optimistic rating on all channels at the same time – when that is simply impossible.

For a power amplifier to deliver rated power on all channels at the same time, the power supply must be able to handle the cumulative demand. In these cases, the power supply may be clever in how it manages its limited resources, but it is not able to handle all channels driven to rated power at one time.

The benefit of Audison’s All Channels Driven approach to the in-car audiophile is that – regardless of the application – clean audiophile power is guaranteed. This approach also meets the ANSI international industry standard for amplifier specifications.

The post All Channels Driven – an audiophile design philosophy first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

Take the great-sounding Audison Thesis speaker. Now, put it in a car.

Wherever we install it in a car, it’s not a perfect speaker location. It might be in a door, or it might be in a dashboard, or it might be in a trim panel. It might be on axis or off axis, but we know the sound almost certainly will be affected by the place that we installed it, and the reflections in the vehicle. DSP lets us manage the speaker’s response with equalization, so that it sounds better – in spite of the fact that we installed it in a car. 

We can safely say that this one speaker won’t be able to play all the sounds we want to play to reproduce music. One speaker won’t play all the way from the upper treble all the way down to the lowest bass. We usually need speakers of different sizes (tweeters, midranges, woofers, subwoofers) when we want to do a good job of reproducing all these different notes. 

Once we have one of these multi-element speaker systems – a two-way system or a three-way system or a four-way system – we’ve gone from one speaker to 2, or 3, or 4! The individual speaker drivers will interact with each other at any frequencies where they overlap – that is, wherever they both contribute to the sound. We want the result of this interaction to sound good, but any time you have two speakers playing the same note, they might cancel each other out to some degree (sometimes, to a great degree).

This can happen even if all the speaker drivers are installed into a cabinet (think, “home speaker”), but it’s even worse when the various speaker drivers are installed in different locations around the vehicle, at different distances from the listener. DSP helps us manage our speaker system so each speaker plays only the notes it should, and lets us ensure our multiple speakers play well together without bad cancellations.

On top of that, we are talking about car stereo. Stereo uses two channels, left and right, and when it’s set up properly we hear an illusion of musical performers between the speakers, in front of us. To do this, we need both left and right speakers in front of us, for every kind of driver we are using – tweeters, midranges, and woofers (subwoofers are exempt from this requirement). Suddenly, we need almost twice as many speakers as we needed in the previous paragraph! Again, whenever multiple speakers are playing the same sounds, they can interfere with each other, and stereo speakers play the same sounds left and right any time the performer is in the center of the stage – so that’s another big potential problem.

DSP helps us manage the left and right speakers so that everything is in alignment and works well together, and then the performer recorded in the center of the stage sounds the way they should.

Car sound systems often need to be louder, because there is a lot more background noise (and sometimes because that’s what the client prefers!) So, we often add rear speakers. Well, when we install rear speakers, we create another opportunity for speakers playing the same notes to interfere with each other! Now the rear speakers can interfere with the front speakers.

DSP helps us manage this potential problem – we can make the rears and fronts play nicely together, without ruining the sound – or the stereo illusion – up front.

So, DSP helps us get better sound from an individual speaker (with equalization). It helps us get more output and better sound when we use multiple speaker drivers to cover the audible range (with crossover filters, level controls, and cancellation management using delay and phase processing). It also helps us when we use left speakers and right speakers to play stereo sound, or to get louder by adding speakers in the rear without interfering with the front speakers (cancellation management tools again). While the biggest cancellation management tool we have is delay, it’s not the only tool we have for managing cancellations.

What about OEM Integration?

OEM system designers have the same access to DSP processing we do – maybe even more, since they can add it to their OEM head units or amplifiers in the design phase. If they decide to use equalization, crossover filters, delay, or phase manipulation, they’ve got the capability. It seems that most OEM head units have a basic DSP functionality today – we see basic head units with all those types of processing in many mid- and entry-level cars today.

The thing is, none of that processing is intended for our new speaker system. If there are crossover filters, they are probably not the crossover filters that would make our speakers sound great. If there is equalization, or delay, or phase manipulation, none of it was done for our speaker system (often, it’s a watered-down tune averaged for both front seats,  rather than the driver-seat-optimized systems we often sell our customers). What’s worse, OEM sound processing often foils the techniques we plan to use to manage our new speaker system. If we don’t correct the OEM processing, we can run into great difficulty in getting the sound we expect from our new speakers. If we want to use delay, we need all the signals in phase with each other at all frequencies before delay will work the way we expect.

DSPs with OEM Integration capabilities have specific features intended to let us correct for OEM processing before we use DSP to manage our new speaker system. Not all DSPs have OEM Integration functionality, and most are pretty limited in this regard – but a DSP with good OEM Integration can make it much simpler to get good sound.

I thought that DSP was for high-end.

Depends on what you mean by “high-end”. If you mean, “we care enough about the resulting sound that we want it to be good”, then maybe it is for “high end”? But OEMs are using it to make base-level deck-and-four systems sound better – and get louder. It’s not just for “SQ” cars – it can make any speaker system sound significantly better!

DSP helps you get louder?

Well, when multiple speakers interact, they often lose output at various notes. Using a DSP to manage cancellations can result in a system that has more output across the board.

What do we do with a DSP to make stereo work in cars?

Basically, we make the left and right sides sound the same, and we manage the various speakers so they don’t interfere with each other too much. Once you’ve done that, stereo reproduction just happens – it really is a side effect of playing stereo recordings on systems that meet those two conditions.

Isn’t there a lot more involved?

Well, yes and no. There are indeed some things that experience helps us with, but making stereo work in a car is indeed making left and right match in level and frequency response, and putting the speakers in phase with each other using the cancellation management techniques listed above. That’s what makes stereo work in your living room. With OEM Integration jobs, the most important thing is that you start with a good signal.

Do you recommend a certain DSP processor?

Yes – the new AF Forza family of DSP amps have industry-leading OEM Integration features, very powerful equalizers, and a great set of system- and cancellation-management tools.

The post Why DSP? first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

Many premium OEM sound systems use a common architecture: the source unit relays unprocessed audio signal to an amplifier, and also communicates with that amplifier on some data network connection. When the driver changes any audio setting of the system – volume, front/rear fade, bass/treble tone controls, etc. these commands are relayed on the network, and performed in the OEM amplifier. In addition, all vehicle-specific processing – phase, delay, and EQ processing – is performed in the OEM amplifier. The OEM amplifier also handles integrating all non-entertainment sounds – handsfree audio, telematics audio, chimes, and alert tones.

As of this writing, Maestro AR products cover over 1200 vehicles, with more being added regularly. In addition, iDatalink can now reprogram hundreds of OEM head units in non-amplified systems to operate in amplified mode, with all the benefits mentioned above! (See the iDatalink application guide for specifics vehicle information.)

Some Maestro interfaces offer a Toslink output, and AF can integrate with this Toslink signal and deliver pristine sound. In applications where the signal into the Maestro device is digital, using the Toslink into AF Forza results in a fully-digital signal path – Full DA HD! With these devices, AF Forza maintains the directional nature of the safety tones – something which cannot be done with Toslink-only signals (Maestro and AF Forza work together to route a combination of digital audio and analog non-entertainment signals as needed).

Thanks to AF Forza’s high-speed native serial data port (reserved for Maestro), lightning-quick responsiveness to the OEM controls is now possible. In addition, some OEM controls can be configured to perform specific aftermarket functions!

• automatic adjustment of the AF bit amplifier’s input level
• specific AF bit functions managed by the OEM source tone controls
• use of the bass control as the subwoofer volume level control
• selection of AF bit presets using the OEM source controls

The post Maestro AR interfaces for Audison Forza amplifiers first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

There are three big areas where Forza moves the category forward: 1 Power.

Audison Forza 5-channel amp is now 1000 total watts RMS – over 2X the power! 100×4 at 4Ω, 600×1 @ 2Ω, RMS. The new AF Compact 8-channel is almost double the power of the model it replaces. The other amplifiers have similar increases. Forza lives up to its name!

2 Tuning.

Audison bit Drive software gives you more – more EQ bands per channel, more phase filter controls per channel, a global Final Tuning EQ, and more visibility into the results (with our integrated real-time analyzer user interface).

Your USB measurement microphone turns bit Drive software into a real-time analyzer, where the acoustic readout is combined with the EQ graph for the utmost in simplicity.

Most models have 6 unique processed preamp outputs in addition to the amplified channels (the AF M12.14 bit has 2 processed preamp channels on top of the 12 amplified channels). Want 8 cabin speakers plus a subwoofer? Add any 4-channel to the bit 5-channel for a 9-channel system!

3 OEM Signal Correction.

This is the big one. This is a complete revamp of Audison’s OEM analysis and correction tools, which satisfies the desire for simplicity from rookie installation technicians, but delivers the control requested by the most advanced technicians!

Audison bit processors have had de-EQ since the beginning, but until now, correcting phase and time was only possible in bit One HD Virtuoso. Now AF Forza can correct OEM phase, time, and EQ processing with only a few clicks!

We are hearing from dealers around the world who are finding and correcting OEM processing with AF Forza – often in vehicles, they hadn’t expected to! Delayed rears, phase-equalized fronts, delayed woofers – AF Forza handles them all, and this is perhaps its most important advance: If we don’t start with signals aligned in phase and time, it can be very difficult to get great sound.

AF Forza can solve problems at the front end – the input signal – and thus save you time on the back end, as you’re tuning the sound.

This suite of signal correction tools, along with the bit Drive user interface, truly sets AF Forza apart.

The post AF Forza – The Big Three first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.

No category has been more confusing, or more disappointing, to in-car audiophiles than “high-resolution Bluetooth receivers”. Different handsets and different digital media players often deliver different sound-quality results, and it can be difficult to predict those results before the installation. Now, with the Audison B-CON audiophile Bluetooth receiver, you can guarantee performance. Audison B-CON is the first (and the only, at the time of this writing) Bluetooth 5.0® player designed for automotive use that has obtained the “Hi-Res audio wireless” certification from the Japan Audio Society.

WHAT’S A CODEC?

  Audiophile performance over Bluetooth is, first and foremost, about the codec – the digital encoding format that both the sending device and receiving device use to transfer the data. Part of the “pairing” process is the two devices agreeing on which codec they will use. We can only select a codec that both devices – the sender and the receiver – both support.  

All Bluetooth devices support a codec called SBC. It’s the baseline codec in the Bluetooth specification. It’s not a lossless codec. It’s decent, but it’s not great, and under some conditions, it sounds pretty poor. But the Bluetooth specification lets you use a better codec than SBC – as long as it’s supported by both devices. Some of these other codecs promise to deliver “high resolution” support, but in reality, the connections made using those codecs may often not be established at the highest quality level – the user may not get what they’re expecting. It’s usually impossible to see – this happens invisibly at the Bluetooth wireless level.  

B-CON USES A HIGH-PERFORMANCE CODEC CALLED LDAC

LDAC is now available in many Android phone handsets, as well as many audiophile digital-media players. Like most codecs, the LDAC data connection can be implemented at various levels of quality, but at the highest level, it’s a lossless codec. Our stringent lab testing shows that over the highest-quality LDAC connection, the analog frequency response reaches 40kHz – and this is the mode that earned the B-CON the “High Resolution” certification from the JAS.  

B-CON is compatible with all audio formats and reaches maximum performance with uncompressed BT streaming (max 96kHz / 24bit, only with Android LDAC devices), both playing local files and with Apps that provide Hi-Res streaming (Tidal, Qobuz…). With B-CON, you know the level of quality the connection has achieved!

The LED indicators on the top of the device confirm the data rate for you – 48k, or 96k.    

WHAT ABOUT APPLE IOS USERS?

  B-CON also uses AAC, which is the highest-performance codec supported by iOS. (Many assume AAC is “Apple Lossless”, but Apple Lossless isn’t suitable for use with Bluetooth and no one uses it. AAC stands for Advanced Audio Codec, and it’s not a proprietary Apple thing – it’s used widely in the tech world). Even though AAC isn’t lossless, and doesn’t permit high-resolution performance above 20k, it still performs audibly better than SBC (the default Bluetooth audio codec), and that means that B-CON gives you the best sound possible from iOS devices over Bluetooth. As you can see in the figure below, the bandwidth obtained by playing a “White Noise 96 kHz / 24 bit at -3dB” audio file with LDAC codec from an Android device is 20 ÷ 48 kHz. With Apple IOS devices, bandwidth is limited to 20 ÷ 20 kHz. This limitation does not depend on the B-CON but on the type of Bluetooth codec used in the Apple device (AAC).    

OUTPUTS, INPUTS, AND THROUGHPUTS

  The B-CON has RCA analog and Toslink SPDIF optical outputs (the digital output is 96kHz / 24bit). The analog outputs are High-Resolution certified, with a signal-noise ratio of 100dBA. Please check the B-CON product page to download the manual and tech. sheet to discover all the connection possibilities.    

The B-CON also has a Toslink SPDIF input, and an internal digital switcher with some innovative logic. If you have a Toslink source – such as the output of an external preamp – and you add the B-CON, connect the B-CON Toslink out to the bit device’s Toslink input, and the preamp’s Toslink output to the input of the B-CON. Then re-connect your Bluetooth devices. Now, when listening to streaming audio from your handset, it will come through the B-CON. When you get a phone call, the B-CON will switch to passthrough mode and allow the OEM Bluetooth hands-free device to handle call audio. It works seamlessly.  

ABSOLUTE VOLUME

 In keeping with Audison tradition, during the design phase, particular attention was paid to volume management, a fundamental aspect for the search for the highest audio quality. Most Bluetooth devices apply the volume controls to the audio stream, but this compromises the potential dynamic range significantly – often below the expected performance from digital audio!

The study by the Audison R&D team focused on the audiophile use of the “Absolute Volume” function. In the B-CON this manages the Master Volume of the DSP while keeping the digital stream at full potential dynamic range – avoiding the loss of resolution that occurs when applying the handset’s volume commands to the audio stream. This requires an ADC port, present on bit Virtuoso and Forza bit DSP amps line.

The post B-CON BLUETOOTH HI-RES RECEIVER first appeared on Audison - Car Audio Amplifiers, Speakers, Processors.