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#1
Crown Prince (PCL-1000) / Re: Model Information
Last post by dtximages - May 25, 2025, 11:01:13 AMHey I'm interested in your thoughts on this speaker. I just picked up a pair and found them VERY lacking in the bass region. Drivers all all moving but I feel there should be more on the bottom end. My Crown Princes don't hardly need a subwoofer, but the Prince sounds is nothing like that.
Could I need a woofer replacement? If so which ones?
Could the crossover be bad? How do I figure that out.
60hz and up, they sound AMAZING like all Duntechs but gosh this whimpy bass is giving me fits.
Could I need a woofer replacement? If so which ones?
Could the crossover be bad? How do I figure that out.
60hz and up, they sound AMAZING like all Duntechs but gosh this whimpy bass is giving me fits.
#2
News Updates / Re: Is Digital Audio Transmiss...
Last post by Tranquility Bass - May 22, 2025, 02:31:29 PMThis comment from another forum is very convenient if your main objective is to profit from promoting snake oil products to the punters. Usually, these products would not stand up to scrutiny in a properly conducted blind test. Not only that, but limiting equipment evaluation strictly to subjective criteria avoids proper scrutiny by using objective measurements, which could highlight problems with a product that listening tests couldn't. This could severely impact the reputation of a particular vendor's product, thus affecting the advertising revenue the forum operator could extract from that vendor.
#3
News Updates / $500,000 vs $78 Turntable Blin...
Last post by Tranquility Bass - May 06, 2025, 12:15:32 PMWhy vinyl may have made a comeback is more to do with what it used to sound like rather than what people who build expensive turntables want it to sound like. 
#4
News Updates / Re: Group-delay correction of ...
Last post by Tranquility Bass - May 04, 2025, 02:21:59 PMFor those who have asked me whether this type of phase correction filter can be implemented on the SHARC DSP in our preamp then lets take a look if it is possible.
Firstly, we will have to throttle back the sampling rate to 48kHz to get close to the ball-park and that's assuming we want to run two stereo channels on the one DSP. Unlike the PC example above we don't really have an infinite amount of resources to play with. This can be done by selecting the base sampling rate for the board of 48 kHz by setting the appropriate DIP switches. To find out how to do this refer to the following application note on our forum https://analog-precision.com/forum/wiki-and-qa/how-to-change-the-native-sample-rate-of-the-preamp/
The theoretical maximum number of taps for the SHARC running at 392 MHz on our board is 786.432 MMACS divided by 48 kHz, which is 16384 taps for both channels running. This limits it to 8192 taps per channel which is the maximum theoretical limit. Of course, we don't want to aim for the maximum number of taps otherwise there won't be enough DSP capacity to run other tasks like the Preamp and the LR crossover filter itself.
Let's bring up Rephase, our trusty freeware FIR filter calculator, and have a play around. Let's try 2048 taps for starters. As one can see below there is too much amplitude and phase deviation in the lower frequencies so this FIR filter is inadequate.
Let's double the tap count to 4096 taps per channel. We get better phase matching but still a lot of amplitude deviation in the lower frequencies.
At 6144 taps, things look much better which gives us some spare DSP capacity for other tasks such as the LR crossover itself
Of course, if you are running only one DSP module per speaker, then there is the potential to double the tap count or sampling rate. But bear in mind that each doubling of the sample rate requires twice the number of taps to yield the exact frequency resolution, which in turn requires 4 times the DSP capability, which is why we had to throttle the DSP back down to 48 kHz but is still now well within the Preamps' capability for the stereo version.
Firstly, we will have to throttle back the sampling rate to 48kHz to get close to the ball-park and that's assuming we want to run two stereo channels on the one DSP. Unlike the PC example above we don't really have an infinite amount of resources to play with. This can be done by selecting the base sampling rate for the board of 48 kHz by setting the appropriate DIP switches. To find out how to do this refer to the following application note on our forum https://analog-precision.com/forum/wiki-and-qa/how-to-change-the-native-sample-rate-of-the-preamp/
The theoretical maximum number of taps for the SHARC running at 392 MHz on our board is 786.432 MMACS divided by 48 kHz, which is 16384 taps for both channels running. This limits it to 8192 taps per channel which is the maximum theoretical limit. Of course, we don't want to aim for the maximum number of taps otherwise there won't be enough DSP capacity to run other tasks like the Preamp and the LR crossover filter itself.
Let's bring up Rephase, our trusty freeware FIR filter calculator, and have a play around. Let's try 2048 taps for starters. As one can see below there is too much amplitude and phase deviation in the lower frequencies so this FIR filter is inadequate.
Let's double the tap count to 4096 taps per channel. We get better phase matching but still a lot of amplitude deviation in the lower frequencies.
At 6144 taps, things look much better which gives us some spare DSP capacity for other tasks such as the LR crossover itself
Of course, if you are running only one DSP module per speaker, then there is the potential to double the tap count or sampling rate. But bear in mind that each doubling of the sample rate requires twice the number of taps to yield the exact frequency resolution, which in turn requires 4 times the DSP capability, which is why we had to throttle the DSP back down to 48 kHz but is still now well within the Preamps' capability for the stereo version.
#5
News Updates / Re: The DAC Scam - Almost ever...
Last post by Tranquility Bass - May 03, 2025, 10:29:11 AMExpensive DAC's, Amps, Preamps, Streamers, cables etc are just a distraction. The real challenge is fixing up problems with loudspeakers and/or rooms In the words of the late John Dunlavy, when he was offered payment to endorse a particular well-known cable manufacturer:-
In the words of the late John Dunlavy, when he was offered payment to endorse a particular well-known cable manufacturer:-Quote"I declined their offer because I could not endorse what I knew to be useless...."
#6
News Updates / The DAC Scam - Almost everyone...
Last post by Tranquility Bass - May 02, 2025, 11:47:54 AMThis is going to create controversy.
And the follow-up video
LOL
And the follow-up video
LOL #7
News Updates / Re: Is Digital Audio Transmiss...
Last post by Tranquility Bass - March 02, 2025, 01:38:48 PMFor those who believe in USB reclockers may not after watching this
#8
News Updates / Re: Group-delay correction of ...
Last post by Tranquility Bass - January 18, 2025, 04:13:39 PMSo now we have established the effectiveness of global group-delay compensation filters as a means to linearize the phase response of a non-minimum phase crossover and proved that you do not need to use linear-phase filters we can go ahead and complete the mission. The client divided his DSP duties as follows.
The following 4-way 8th order non-minimum phase LR crossover was uploaded to the Ultimate Preamp as follows.
Whilst the stereo phase correction filter comprising of two 65536 tap FIR filters running at 192kHz sampling rate was installed on the PC as follows.
Alternatively, the two designs could have been combined into one design and run exclusively on the PC whilst seamlessly integrating with the Ultimate Preamp.
As an indication of the CPU resources used, I grabbed a screenshot of the Audioweaver Server that presents the resource usage in realtime. To get an appreciation of the amount of computations performed each second we can do a back of the napkin calculation. For each 64k tap FIR filter, the number of multiply-accumulates (MACs) can be calculated by multiplying the tap count of 65336 by the sample rate of 192,000. This is 12.6 GMACs for one filter or 25.2 G-MACs for two filters simultaneous. That is certainly some decent DSP processing capability the PC is providing. To put it all into context for every audio sample 65536 multiply-accumulates have to be performed to calculate the convolution filter and there are 192,000 of these same calculations that have to be carried out per second !! If Audioweaver is leveraging on the Pentium SSE instruction set then it is more than likely using the fused-multiply-add or FMA instruction which supports SIMD (Single Instruction Multiple Data) so can perform multiple MAC's per instruction cycle! Coupled with the 6MB of Intel smart cache and a 3.3GHz clock frequency this thing is really hauling the mail without breaking a sweat
This is why we are not too keen to upgrade the onboard DSP when the current DSP board can leverage so much external DSP capability whilst maintaining the same outstanding audio performance metrics and anyone with an Ultimate Preamp Plus can achieve this !!
The following 4-way 8th order non-minimum phase LR crossover was uploaded to the Ultimate Preamp as follows.
Whilst the stereo phase correction filter comprising of two 65536 tap FIR filters running at 192kHz sampling rate was installed on the PC as follows.
Alternatively, the two designs could have been combined into one design and run exclusively on the PC whilst seamlessly integrating with the Ultimate Preamp.
As an indication of the CPU resources used, I grabbed a screenshot of the Audioweaver Server that presents the resource usage in realtime. To get an appreciation of the amount of computations performed each second we can do a back of the napkin calculation. For each 64k tap FIR filter, the number of multiply-accumulates (MACs) can be calculated by multiplying the tap count of 65336 by the sample rate of 192,000. This is 12.6 GMACs for one filter or 25.2 G-MACs for two filters simultaneous. That is certainly some decent DSP processing capability the PC is providing. To put it all into context for every audio sample 65536 multiply-accumulates have to be performed to calculate the convolution filter and there are 192,000 of these same calculations that have to be carried out per second !! If Audioweaver is leveraging on the Pentium SSE instruction set then it is more than likely using the fused-multiply-add or FMA instruction which supports SIMD (Single Instruction Multiple Data) so can perform multiple MAC's per instruction cycle! Coupled with the 6MB of Intel smart cache and a 3.3GHz clock frequency this thing is really hauling the mail without breaking a sweat
This is why we are not too keen to upgrade the onboard DSP when the current DSP board can leverage so much external DSP capability whilst maintaining the same outstanding audio performance metrics and anyone with an Ultimate Preamp Plus can achieve this !!
#9
News Updates / 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 !!
#10
News Updates / 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 !!