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#11
FIR filter designs / Re: Group-delay correction of ...
Last post by Tranquility Bass - January 18, 2025, 01:51:46 PMBut what about the effects of the low-frequency box alignment on the frequency response ? We can do this on our Audioweaver test bench by switching in a high pass filter which mimics the effect of the woofer and box. Let's say we tune the woofer and box for a Butterworth alignment (ie Qt=0.7071) and a box frequency of fc=20Hz. How does this affect the frequency response? We can do this by enabling the high pass filter in the signal path and as expected we get the following frequency response where the response is 3dB down and phase is +90 degrees at 20Hz !
As expected The square wave response shows the typical drooping due to the additional group delay at the lower frequencies and at 20Hz and 10Hz is very much exaggerated !
Can we correct for this? Let's bring up RePhase again and include the effects of the low-frequency box alignment in the correction filter. We do this by including the box parameters and recalculating the filter coefficients or taps in this case.
We load these coefficients into the bottom FIR filter block in Audioweaver and set the appropriate switches so the high pass filter and new correction filter is now in the signal path. When we run a frequency response sweep we still get the same magnitude response because the correction filter does not attenuate any frequencies at all.
But the phase response is similar to the corrected phase response as before due to the large amount of delay in this type of filter.
To make more sense of this phase plot we switch in a delay in the reference channel which should match the delay of the filter. If everything is ok then the phase response of the two channels should be identical linear phase responses and the difference in phase should result in a flat or straight line with minimal gradient. Lets see what happens.
As expected the phase characteristics of both channels cancel each other out with no deviation at all and shows that the filter under test is producing a linear phase response.
And the corrected square wave responses are totally corrected except for a time delay of course !!
You are probably interested in what the response looks like at the cutoff frequency of 20 Hz and an octave below it at 10Hz where the actual response starts falling off thus having a direct effect on the fundamental frequency component of the square wave. Lets see !
For comparison if we switch in the time delay back into the reference channel to equalize the phase response between the two channels we can now see below both waveforms from both channels are in lock-step and now line up with each other no matter what frequency is used.
Note the reduction in amplitude of the sinewave output at the box cutoff frequency of -3dB whilst still being in perfect phase whereas the uncorrected response would be out of phase by +90 degrees !!
As expected The square wave response shows the typical drooping due to the additional group delay at the lower frequencies and at 20Hz and 10Hz is very much exaggerated !
Can we correct for this? Let's bring up RePhase again and include the effects of the low-frequency box alignment in the correction filter. We do this by including the box parameters and recalculating the filter coefficients or taps in this case.
We load these coefficients into the bottom FIR filter block in Audioweaver and set the appropriate switches so the high pass filter and new correction filter is now in the signal path. When we run a frequency response sweep we still get the same magnitude response because the correction filter does not attenuate any frequencies at all.
But the phase response is similar to the corrected phase response as before due to the large amount of delay in this type of filter.
To make more sense of this phase plot we switch in a delay in the reference channel which should match the delay of the filter. If everything is ok then the phase response of the two channels should be identical linear phase responses and the difference in phase should result in a flat or straight line with minimal gradient. Lets see what happens.
As expected the phase characteristics of both channels cancel each other out with no deviation at all and shows that the filter under test is producing a linear phase response.
And the corrected square wave responses are totally corrected except for a time delay of course !!
You are probably interested in what the response looks like at the cutoff frequency of 20 Hz and an octave below it at 10Hz where the actual response starts falling off thus having a direct effect on the fundamental frequency component of the square wave. Lets see !
For comparison if we switch in the time delay back into the reference channel to equalize the phase response between the two channels we can now see below both waveforms from both channels are in lock-step and now line up with each other no matter what frequency is used.
Note the reduction in amplitude of the sinewave output at the box cutoff frequency of -3dB whilst still being in perfect phase whereas the uncorrected response would be out of phase by +90 degrees !!
#12
FIR filter designs / Re: Group-delay correction of ...
Last post by Tranquility Bass - January 18, 2025, 11:40:31 AMTo generate a group-delay correction filter we need a means to build one. As expected the filter we need will be a Finite Impulse Response (FIR) digital filter with many taps (multiply accumulates). Because the crossover runs at 192kHz we will also build the correction filter to run at 192kHz which is a big ask for any DSP let alone two of them used for stereo but our test PC is more than capable being an Intel i5 4590 with 4 cores running at 3.3GHz and 3 levels of cache. We used a program called rePhase to build our filter which is a freeware program available from https://rephase.org/. Here is the filter setup in RePhase. Later on, we will include compensation for the low-frequency roll-off caused by the woofer and box but for now, let's just focus on the crossover itself.
We set optimization to "Moderate" and hit the "Generate" key and then we created the following correction filter ! Note that rePhase always creates a correction filter with a net 0-phase response! This results in a non-causal impulse response that occurs before the impulse which whilst mathematical succint is physically unrealizable. To overcome this, rePhase allows you to realign the impulse response by delaying it and windowing it so there is no response before the impulse. In this case, this was done by delaying and centering the impulse response at the halfway point (i.e., half tap count) and using a Hanning window to mask out anything before the impulse. Since it's a txt file we can open it with Notepad just by double clicking on the file !
We load these coefficients into the top FIR filter block in our Audioweaver test bench above. The bottom FIR filter block is used when we also include the effects of the box frequency response which we will address later on. We run Audioweaver again and switch in the correction filter so that it is now feeding the crossover and do a frequency response sweep again. As expected the response is ruler flat since the correction filter does not alter the magnitude but only the phase !
However, the phase response is hardly surprising with so much delay in the filter so it's hard to tell the shape of the curve is linear.
To make more sense of this phase plot we switch in a delay in the reference channel which should match the delay of the filter. If everything is ok then the phase response of the two channels should be identical linear phase responses and the difference in phase should result in a flat or staright line with minimsl gradient. Lets see what happens.
As expected the phase characteristics of both channels cancel each other out with no deviation at all and shows that the filter under test is producing a linear phase response.
But the real test is the square wave response because even though the filter still measures flat the phase response appears to be linear and we can confirm this on the scope. Firstly at 1kHz and then at 100Hz the square wave is preserved so we have successfully phase corrected the non-minimum phase crossover !!
We set optimization to "Moderate" and hit the "Generate" key and then we created the following correction filter ! Note that rePhase always creates a correction filter with a net 0-phase response! This results in a non-causal impulse response that occurs before the impulse which whilst mathematical succint is physically unrealizable. To overcome this, rePhase allows you to realign the impulse response by delaying it and windowing it so there is no response before the impulse. In this case, this was done by delaying and centering the impulse response at the halfway point (i.e., half tap count) and using a Hanning window to mask out anything before the impulse. Since it's a txt file we can open it with Notepad just by double clicking on the file !
We load these coefficients into the top FIR filter block in our Audioweaver test bench above. The bottom FIR filter block is used when we also include the effects of the box frequency response which we will address later on. We run Audioweaver again and switch in the correction filter so that it is now feeding the crossover and do a frequency response sweep again. As expected the response is ruler flat since the correction filter does not alter the magnitude but only the phase !
However, the phase response is hardly surprising with so much delay in the filter so it's hard to tell the shape of the curve is linear.
To make more sense of this phase plot we switch in a delay in the reference channel which should match the delay of the filter. If everything is ok then the phase response of the two channels should be identical linear phase responses and the difference in phase should result in a flat or staright line with minimsl gradient. Lets see what happens.
As expected the phase characteristics of both channels cancel each other out with no deviation at all and shows that the filter under test is producing a linear phase response.
But the real test is the square wave response because even though the filter still measures flat the phase response appears to be linear and we can confirm this on the scope. Firstly at 1kHz and then at 100Hz the square wave is preserved so we have successfully phase corrected the non-minimum phase crossover !!
#13
FIR filter designs / Group-delay correction of a cr...
Last post by Tranquility Bass - January 18, 2025, 10:50:09 AMThere is a school of thought that says the only way to accomplish a linear phase crossover is to use linear phase filters which are a special class of FIR filter that sums to a unity response with a fixed time delay. So when both high pass and low pass sections are added together the response is a delayed version of what is being fed into the crossover. As I have discussed in another thread this type of filter has issues with proper impulse response cancellation due to imperfect matching between drivers which can result in severe pre-ringing leakage among other topics. The proponents of these filters will never talk about this aspect of this type of filter topology, instead pretending that it doesn't exist or can easily be minimized so that it is never a problem.
The other way of linearizing the crossover is to take an existing non-minimum phase crossover and apply a global group-delay correction filter which is just an FIR filter with a large number of taps. The advantage here is there are no FIR filters in the actual crossover to generate any pre-ringing artifacts. This is the approach taken by a few commercial designs as well as here, although the correction filter usually requires a large number of taps and has a large delay depending on the sampling rate and tap count. So this should be taken into account if you want to use these speakers with other speakers or in conjunction with a video display in say a home theater system in order to avoid lip-syncing issues.
We took a four-way 8th-order Linkwitz-Riley crossover from one of our clients who own one of our Preamps and applied a global group-delay correction filter to linearize the phase response. The LR crossover was already flat but the phase response was anything but linear so a square wave fed into it did not look like a square wave coming out of it. A linear phase response means that there is just a time delay in the signal path whilst still maintaining a flat magnitude response but there is no waveform distortion.
To measure the phase distortion of the crossover in isolation we needed to sum the outputs of the crossover before they were level adjusted and time delayed so we could focus on the crossover alone without the effects of the speaker and the room etc which we know adds its own artifacts to the response. Later on, we will look at compensating for those as well but for now we are just focussing on fixing up the crossover and making sure the group-delay correction filter is doing its job properly.
To evaluate the crossover we just looked at one channel and summed the outputs to measure the net frequency response. We also fed the input to the crossover to one of the outputs of the preamp as a reference for the analyzer to create the frequency response. The summed outputs of the crossover are fed to another channel on the preamp. We also added a few multiplexers or selector switches to enable us to switch in and out various filters quickly so we can quickly evaluate the effects of different filter configurations. Both the input to the crossover and the summed outputs are then connected to our dScope III audio analyzer. The dScope III can measure both amplitude and interchannel phase. The following Audioweaver test bench is what we used to evaluate the effectiveness of the global correction filters.
All of the DSP is done on a PC instead of the Preamp itself because the group-delay correction filters required are beyond the resources of the onboard SHARC DSP. The Ultimate Preamplifier Plus (UPP) has a unique feature in that it allows one to run DSP on an external device such as a PC as though it was running inside the Preamp. In fact, in another thread, we proved that the audio path was transparent and the performance of the Preamp was unaffected by the noisy environment of a PC ! The Ultimate Preamplifier Plus can also share DSP between its own onboard DSP as well as an external PC so for example you could build a non-minimum phase LR crossover using the onboard SHARC DSP whilst running the correction filter on an external PC and then switch the correction filter in or out from the Preamp menu. This is the approach our client is taking but for proof of concept and convenience, we ran everything on an external PC which allowed us to quickly evaluate and test everything in the one environment. The main aim was to validate the effectiveness of the global group-delay correction filters and prove that you don't need to use dedicated linear-phase filters to achieve this.
First, we measured the frequency and phase response of the crossover filter alone using a log frequency axis. We note the flatness of the filter as expected but the phase is anything but flat ! The top blue trace is the sweep of the input which is ruler flat as expected and the bottom red trace is the crossover summation of all drivers which is also ruler flat as expected.
To evaluate the phase response we reran the frequency response test using a linear frequency axis instead of the usual log axis. If the phase is linear then the phase response should look straight and not curved. Since the analyzer cannot distinguish phase greater or less than 180 and -180 degrees respectively it wraps or folds the phase so the response always is contained within a 360 degree envelope which makes it look like a saw-tooth. From the phase plot below we can see the phase is anything but linear which was expected !
And now for the money shot. What does this crossover filter do to a square wave? The square wave is important because it is not just a single tone but rather a series of odd harmonically related tones with a monotonically decreasing magnitude. If the phase is not linear, it will manifest itself as distortion in the waveform even though the magnitudes of the harmonics are still correct. So, let's look at the test results on a scope.
Firstly at 1kHz and then at 100 Hz !! This is what a non-minimum phase crossover does to a square wave !! Can we fix this ??
The other way of linearizing the crossover is to take an existing non-minimum phase crossover and apply a global group-delay correction filter which is just an FIR filter with a large number of taps. The advantage here is there are no FIR filters in the actual crossover to generate any pre-ringing artifacts. This is the approach taken by a few commercial designs as well as here, although the correction filter usually requires a large number of taps and has a large delay depending on the sampling rate and tap count. So this should be taken into account if you want to use these speakers with other speakers or in conjunction with a video display in say a home theater system in order to avoid lip-syncing issues.
We took a four-way 8th-order Linkwitz-Riley crossover from one of our clients who own one of our Preamps and applied a global group-delay correction filter to linearize the phase response. The LR crossover was already flat but the phase response was anything but linear so a square wave fed into it did not look like a square wave coming out of it. A linear phase response means that there is just a time delay in the signal path whilst still maintaining a flat magnitude response but there is no waveform distortion.
To measure the phase distortion of the crossover in isolation we needed to sum the outputs of the crossover before they were level adjusted and time delayed so we could focus on the crossover alone without the effects of the speaker and the room etc which we know adds its own artifacts to the response. Later on, we will look at compensating for those as well but for now we are just focussing on fixing up the crossover and making sure the group-delay correction filter is doing its job properly.
To evaluate the crossover we just looked at one channel and summed the outputs to measure the net frequency response. We also fed the input to the crossover to one of the outputs of the preamp as a reference for the analyzer to create the frequency response. The summed outputs of the crossover are fed to another channel on the preamp. We also added a few multiplexers or selector switches to enable us to switch in and out various filters quickly so we can quickly evaluate the effects of different filter configurations. Both the input to the crossover and the summed outputs are then connected to our dScope III audio analyzer. The dScope III can measure both amplitude and interchannel phase. The following Audioweaver test bench is what we used to evaluate the effectiveness of the global correction filters.
All of the DSP is done on a PC instead of the Preamp itself because the group-delay correction filters required are beyond the resources of the onboard SHARC DSP. The Ultimate Preamplifier Plus (UPP) has a unique feature in that it allows one to run DSP on an external device such as a PC as though it was running inside the Preamp. In fact, in another thread, we proved that the audio path was transparent and the performance of the Preamp was unaffected by the noisy environment of a PC ! The Ultimate Preamplifier Plus can also share DSP between its own onboard DSP as well as an external PC so for example you could build a non-minimum phase LR crossover using the onboard SHARC DSP whilst running the correction filter on an external PC and then switch the correction filter in or out from the Preamp menu. This is the approach our client is taking but for proof of concept and convenience, we ran everything on an external PC which allowed us to quickly evaluate and test everything in the one environment. The main aim was to validate the effectiveness of the global group-delay correction filters and prove that you don't need to use dedicated linear-phase filters to achieve this.
First, we measured the frequency and phase response of the crossover filter alone using a log frequency axis. We note the flatness of the filter as expected but the phase is anything but flat ! The top blue trace is the sweep of the input which is ruler flat as expected and the bottom red trace is the crossover summation of all drivers which is also ruler flat as expected.
To evaluate the phase response we reran the frequency response test using a linear frequency axis instead of the usual log axis. If the phase is linear then the phase response should look straight and not curved. Since the analyzer cannot distinguish phase greater or less than 180 and -180 degrees respectively it wraps or folds the phase so the response always is contained within a 360 degree envelope which makes it look like a saw-tooth. From the phase plot below we can see the phase is anything but linear which was expected !
And now for the money shot. What does this crossover filter do to a square wave? The square wave is important because it is not just a single tone but rather a series of odd harmonically related tones with a monotonically decreasing magnitude. If the phase is not linear, it will manifest itself as distortion in the waveform even though the magnitudes of the harmonics are still correct. So, let's look at the test results on a scope.
Firstly at 1kHz and then at 100 Hz !! This is what a non-minimum phase crossover does to a square wave !! Can we fix this ??
#14
Performance and Measurements / Re: DAC - THD + Noise @ 1kHz
Last post by Tranquility Bass - January 14, 2025, 06:19:07 PMA recent review by a well-respected online loudspeaker magazine of a prospective competitor to the Ultimate Preamp, whose product also runs Audioweaver, produced the following independent measurements on an Audio Precision APx555 audio analyzer. As can be seen the THD+N is about 1 order of magnitude (10x or 20dB )worse than what we were able to measure on our dScope III analyzer, and I dare say that we were pushing the lower limits of our analyzer at the time so I would expect to see even better results on an APx555 which is the gold standard these days 
This is a very much below-average result, and it is little wonder it didn't stay up long on ASR and was removed rather quickly. It's the sort of result you would have expected from a cheap PC motherboard sound card or external USB sound card from a decade ago. Tsk tsk tsk... it looks like someone is heading back to the drawing board again
This is a very much below-average result, and it is little wonder it didn't stay up long on ASR and was removed rather quickly. It's the sort of result you would have expected from a cheap PC motherboard sound card or external USB sound card from a decade ago. Tsk tsk tsk... it looks like someone is heading back to the drawing board again
#15
Performance and Measurements / NEW Jitter tests showing the e...
Last post by Tranquility Bass - January 14, 2025, 02:34:24 PMThe following Jitter tests illustrate the benefit of using the built-in Inter-sample Overs Guard in preventing internal numerical overflow inside the asynchronous sample rate converter of the SHARC DSP. We used the same J-test methodology that we used for our other tests.
Firstly the following shows the scope screen capture of one channel of the output of the Preamp with the J-test signal from the analyzer applied directly to the USB port of the Preamp. For all intents and purposes, it should look like a 12kHz sinewave but because of Intersample Overs, there is internal overflow in the sample rate converter which manifests itself as waveform distortion similar to clipping which is a side effect of upsampling or interpolating to a higher sampling rate.
The following shows the resultant FFT spectrum and notable artifacts can be observed.
Expanding the frequency scale of the spectrum above shows in much more detail the problems caused by the inter-sample overs.
With the Inter-samples Guard switched on we noted how many times the sample rate converter was overloading in the statistics collecting mode ! Once activated the Guard reduced the signal input level to avoid any more internal arithmetic overflows and reported that the signal had to be reduced by 3dB to achieve this.
And the signal from the output of the Preamp was reduced in amplitude by 3dB but is no longer visually distorted!
Now let's see the effect on the J-test spectrum from the analyzer. A vastly much improved and cleaner spectrum is observed below when the sample rate converter is not overloading and essentially proves how transparent the SHARC sample rate convert can be !
The expanded frequency scale of the above FFT spectrum is shown below !
Firstly the following shows the scope screen capture of one channel of the output of the Preamp with the J-test signal from the analyzer applied directly to the USB port of the Preamp. For all intents and purposes, it should look like a 12kHz sinewave but because of Intersample Overs, there is internal overflow in the sample rate converter which manifests itself as waveform distortion similar to clipping which is a side effect of upsampling or interpolating to a higher sampling rate.
The following shows the resultant FFT spectrum and notable artifacts can be observed.
Expanding the frequency scale of the spectrum above shows in much more detail the problems caused by the inter-sample overs.
With the Inter-samples Guard switched on we noted how many times the sample rate converter was overloading in the statistics collecting mode ! Once activated the Guard reduced the signal input level to avoid any more internal arithmetic overflows and reported that the signal had to be reduced by 3dB to achieve this.
And the signal from the output of the Preamp was reduced in amplitude by 3dB but is no longer visually distorted!
Now let's see the effect on the J-test spectrum from the analyzer. A vastly much improved and cleaner spectrum is observed below when the sample rate converter is not overloading and essentially proves how transparent the SHARC sample rate convert can be !
The expanded frequency scale of the above FFT spectrum is shown below !
#16
Performance and Measurements / Measurements of THD+N/Jitter u...
Last post by Tranquility Bass - January 10, 2025, 05:25:35 PMSomeone emailed me recently after I posted the results of routing digital audio through an external PC for the purposes of expanding the DSP capability of the Ultimate-Preamp and asked whether or not using a PC adds any additional noise, distortion or jitter to the audio stream ? Well I said in theory it shouldn't due to the isolation and transparency provided by this interface but there is only one way to find out and that is to measure it.
So using the same J-test methodolgy that we used for the normal jitter tests on the various inputs to the preamp we can now apply the same test methodology whilst routing the audio through an external Dell PC used in our lab and see how it affects the jitter and noise. So we setup Audioweaver on a PC to process audio from the Preamp at 192kHz in a direct input-to-output connection as in the following:-
Then we measured the jitter from the output of the Preamp using the same J-test methodology as we used on earlier tests of jitter.
The screenshot below is the same as above but with an expanded frequency scale. As can be seen the spectrum is as clean as it is with internal DSP processing and any artifacts are way below the threshold of hearing !
We did the same for THD+Noise.
As can be seen in the test results above the PC is essentially transparent to the audio stream and does not add any noise, distortion or jitter to the incoming audio stream thanks to the hardware of the Ultimate Preamplifier Plus. This is contrary to the reputation of PC's as a noisy and hostile environment due to high-speed digital design and a switching power supply not optimized for low noise. However, the UPP mitigates these issues with ease without the use of re-clockers, network switches, fancy USB cables, add-on FIFO's etc, which would not improve anything or could indeed make things worse simply because our hardware is designed right in the first place and the above measurements are proof of that
.
So using the same J-test methodolgy that we used for the normal jitter tests on the various inputs to the preamp we can now apply the same test methodology whilst routing the audio through an external Dell PC used in our lab and see how it affects the jitter and noise. So we setup Audioweaver on a PC to process audio from the Preamp at 192kHz in a direct input-to-output connection as in the following:-
Then we measured the jitter from the output of the Preamp using the same J-test methodology as we used on earlier tests of jitter.
The screenshot below is the same as above but with an expanded frequency scale. As can be seen the spectrum is as clean as it is with internal DSP processing and any artifacts are way below the threshold of hearing !
We did the same for THD+Noise.
As can be seen in the test results above the PC is essentially transparent to the audio stream and does not add any noise, distortion or jitter to the incoming audio stream thanks to the hardware of the Ultimate Preamplifier Plus. This is contrary to the reputation of PC's as a noisy and hostile environment due to high-speed digital design and a switching power supply not optimized for low noise. However, the UPP mitigates these issues with ease without the use of re-clockers, network switches, fancy USB cables, add-on FIFO's etc, which would not improve anything or could indeed make things worse simply because our hardware is designed right in the first place and the above measurements are proof of that
.
#17
News Updates / Re: Introducing the Ultimate A...
Last post by Tranquility Bass - August 14, 2024, 02:36:07 PMSemi-populated DSPXO pcb's arrivethed to be populated and tested. These ones are earmarked for the upcoming Active Speaker system where each system requires no less than three DSPXO boards - one for each speaker and one for the central hub ! Whilst waiting for these boards to arrive we have been busy writing the new firmware that runs on these boards. The Ultimate Preamp proved that these boards are reliable, versatile and extremely high performance with more than enough DSP capability and onboard memory along with the onboard high performance DAC's and ADC's becomes the glue that binds all of the subsystems together ! Because we write the code ourselves that runs on the DSP so we get to choose how it operates and thus have infinitely more control compared to many of the off the shelf DSP systems which are basically limited to designing filters only.
#18
News Updates / Re: Introducing the Ultimate A...
Last post by Tranquility Bass - March 07, 2024, 06:39:29 PMWhy a single amplifier driving a passive speaker is such a bad idea but nobody is talking about the elephant in the room !! Maybe if your amplifier is carved out of a solid block of aluminium billet, or you buy some uber expensive speaker cables, or use some cable lifters or even use a grounding box it may solve these fundamental issues but wishful thinking never solved any technical problems
#19
News Updates / Re: Introducing the Ultimate A...
Last post by Tranquility Bass - February 26, 2024, 05:35:27 PMSome people spend a fortune on speaker cables thinking it is the most purist form of signal delivery from amplifier to their loudspeakers !!
But unbeknownst to them the signal has to pass through a complex passive crossover full of an array of lossy and imperfect passive components before it reaches their loudspeaker drivers which is kind of defeating the purpose of having this ultra-pure interconnect cable in the first place !! Passing large amounts of current through capacitors, inductors and resistors was never a good idea. If a passive component was ever going to show its flaws was in a hostile environment like this and yet people continue to seek out the law of diminishing returns by throwing good money after bad on interconnect cables whilst ignoring the elephant in the room ! Hell, there are some speaker vendors who even go to great lengths to hide their passive crossovers by potting the components so people can't scrutinize their design choices !
With our active speaker, each driver is connected directly to its own amplifier channel with the shortest possible cable length, and even like speaker drivers do not share an amplifier. Every speaker driver has its own associated amplifier channel with its own active crossover network implemented in the most purest form inside a DSP in the form of a mathematical algorithm that more accurately mimics a passive crossover without any of its disadvantages. The advantage of doing it inside a DSP is that there are no issues with power dissipation, component tolerances and drift in values over time, component non-linearities, capacitor dielectric absorption, inductor iron losses and of course cost !! Not only that we can accurately match levels between drivers without using padding resistors that dissipate a lot of power and can drift in value over time. You might think this is overkill but when you consider that the cost of this whole speaker is less than the cost of these speaker cables alone you'd have to agree that the Ultimate Active Speaker has a hell of a lot more to offer than a traditional passive speaker system !
But unbeknownst to them the signal has to pass through a complex passive crossover full of an array of lossy and imperfect passive components before it reaches their loudspeaker drivers which is kind of defeating the purpose of having this ultra-pure interconnect cable in the first place !! Passing large amounts of current through capacitors, inductors and resistors was never a good idea. If a passive component was ever going to show its flaws was in a hostile environment like this and yet people continue to seek out the law of diminishing returns by throwing good money after bad on interconnect cables whilst ignoring the elephant in the room ! Hell, there are some speaker vendors who even go to great lengths to hide their passive crossovers by potting the components so people can't scrutinize their design choices !
With our active speaker, each driver is connected directly to its own amplifier channel with the shortest possible cable length, and even like speaker drivers do not share an amplifier. Every speaker driver has its own associated amplifier channel with its own active crossover network implemented in the most purest form inside a DSP in the form of a mathematical algorithm that more accurately mimics a passive crossover without any of its disadvantages. The advantage of doing it inside a DSP is that there are no issues with power dissipation, component tolerances and drift in values over time, component non-linearities, capacitor dielectric absorption, inductor iron losses and of course cost !! Not only that we can accurately match levels between drivers without using padding resistors that dissipate a lot of power and can drift in value over time. You might think this is overkill but when you consider that the cost of this whole speaker is less than the cost of these speaker cables alone you'd have to agree that the Ultimate Active Speaker has a hell of a lot more to offer than a traditional passive speaker system !
#20
Cantata / Technical Information/Crossove...
Last post by Tranquility Bass - January 08, 2024, 10:54:53 AMPlease feel free to add any technical material you may have on these speakers.