Full Range TC9 Line Array CNC Cabinet

[fluid]

I removed the IR delay and put the soundcard impulse against the TC9 measurement and you can see where the ringing lines up and it rings for just over 500usec.






Byrtt do you have a measurement of the TC9 with the Behringer soundcard as the ringing does look to be very similar?

Another thing that was surprising is that the calibration of an onboard soundcard you posted in the rephase thread also had very little ringing in comparison.

When the soundcard is calibrated the frequency response and phase are compensated in a following measurement of the same device but the impulse response looks identical.

Do you think that dividing the calibration to end up with a pulse and then using that somehow to remove the ringing from the soundcard and see just the speaker response would work?

I posted the frequency response and phase plots of the different AK4495 filters in this thread

Dual mono AK4495S with AK4137 and MCU/LCD DAC kit for $160

Also surprising that they measure quite differently but the sound is not drastically changed for any of them.

[stardust4ever]

Originally Posted by fluid

Your eye is off by a few microseconds as the measurement was done at 48K so the ringing is probably closer to the 24K half sample limit. Good deduction on the timing but not outboard DSP just the DAC filter which could be considered to be internal DSP

The latest graphs also show ringing with constructive and destructive interference ripples. If there is a 44100 <-> 48000 conversion anywhere in the chain, then ringing near half the sample rate will cause moire patterns. Or possibly 22050Hz and 24000Hz waves interfering will do this causing amplitude modulation. Any two notes of equal magnitude and close in frequency will constructively and destructively interact.

[fluid]

Originally Posted by stardust4ever

The latest graphs also show ringing with constructive and destructive interference ripples. If there is a 44100 <-> 48000 conversion anywhere in the chain, then ringing near half the sample rate will cause moire patterns. Or possibly 22050Hz and 24000Hz waves interfering will do this causing amplitude modulation. Any two notes of equal magnitude and close in frequency will constructively and destructively interact.

I'm a bit confused as to what you are referring to, the image you quoted is Byrtt's with a 192kHz sample rate, that makes it a little harder to compare to mine at 48K.

Moire patterns is a video term to describe one form of aliasing, there are no 44 to 48 conversions in the loopback tests so I am a bit confused as to what you mean.

http://hometheaterhifi.com/technical...f-terminology/

This page has a pretty good explanation of aliasing and ringing, in fig 14 and the text around it it shows that the closer a filter is to a brickwall in the frequency domain the more it rings in the time domain. Even though the article is aimed at video it covers the aspects that affect audio too.

Then if anyone wants to read more about Fourier pairs, the chapter in the Scientist and Engineers guide to DSP has it
http://www.dspguide.com/ch11/1.htm

I probably should run some loopbacks on the Scarlett at different sampling rates to see how that affects the impulse response.

[BYRTT]

Originally Posted by fluid

...Byrtt do you have a measurement of the TC9 with the Behringer soundcard as the ringing does look to be very similar?

Another thing that was surprising is that the calibration of an onboard soundcard you posted in the rephase thread also had very little ringing in comparison.

When the soundcard is calibrated the frequency response and phase are compensated in a following measurement of the same device but the impulse response looks identical.

Do you think that dividing the calibration to end up with a pulse and then using that somehow to remove the ringing from the soundcard and see just the speaker response would work?

I posted the frequency response and phase plots of the different AK4495 filters in this thread

Dual mono AK4495S with AK4137 and MCU/LCD DAC kit for $160

Also surprising that they measure quite differently but the sound is not drastically changed for any of them.

Have only loopback measurements for UMC204HD and find visuals below in same scale as yours, bit surprised they exactly the same even mine is another brand and 2ch in / 4ch out where yours is a lot more, from inside inspection of mine can tell its the famous asynchronous USB Xmos chip and Cirrus Logic ADCs/DACs in this interface. It looks problem is their coded filters at 44/48kHz if using 96/192kHz ringing gets much less. Don't no about Mac OS but in Windows OS shift REW to ASIO and 96/192kHz gets available settings. First below is UMC204HD at 44+48kHz and second UMC204HD at 96+192kHz, third is talked more about below.

Yes onboard ALC892 codec from RealTek on this Ivybridge generation computer is much better, even the still avaiable old very cheap 16bit UCA222/202 from Behringer is excellent into IR and actual also sounds very good when playing native CD material and i think xrk971 support me in this opinion, he have UCA202 which is relaxed silver painted and i have UCA222 which is bit alarming red. Find it bit strange why most old interfaces is perfect filtered by the book and many new ones show such tendency.

Agree using such ringing interface if we want to see speaker IR alone its better into REW not set a calibration file for that loop and instead add from a saved mdat-file that interface loop to current measurement session and divide.

Thanks sharing those DACs data will have a look.

[wesayso]

Don't stare at these IR's too long . In a speaker measurement you won't find the ringing of a DAC after the main pulse. It will be dominated by different frequencies and/or reflections of the driver itself. You'll only see the pré ringing part. What you're looking at is a combination of the entire frequency response. Though high frequencies usually dominate it's shape.
I'm not surprised you can't hear a difference between them when most of the ringing is above 20 KHz. It's good to know what you have, but there's a lot more data behind that IR rendering which is why there are so many ways to view the same data.
I'd say get familiar with all of the different tabs in REW to get a better understanding of what you're looking at. It's all the same data as in that IR.

[torgeirs]

It has been mentioned earlyer about overshoot and ringing.
If a low pass filter has Q over 0.5, a step response or impulse response will show this Q value as ringing. It actually is energy storage. To me it looks like the TC9 has some resonance/ energy storage before it falls off at HF.
There really is a lot of litterature on this as it is important when designing stable amplifiers and calculating phase margin.

[fluid]

Originally Posted by BYRTT

Have only loopback measurements for UMC204HD and find visuals below in same scale as yours, bit surprised they exactly the same even mine is another brand and 2ch in / 4ch out where yours is a lot more, from inside inspection of mine can tell its the famous asynchronous USB Xmos chip and Cirrus Logic ADCs/DACs in this interface. It looks problem is their coded filters at 44/48kHz if using 96/192kHz ringing gets much less. Don't no about Mac OS but in Windows OS shift REW to ASIO and 96/192kHz gets available settings. First below is UMC204HD at 44+48kHz and second UMC204HD at 96+192kHz, third is talked more about below.

Yes onboard ALC892 codec from RealTek on this Ivybridge generation computer is much better, even the still avaiable old very cheap 16bit UCA222/202 from Behringer is excellent into IR and actual also sounds very good when playing native CD material and i think xrk971 support me in this opinion, he have UCA202 which is relaxed silver painted and i have UCA222 which is bit alarming red. Find it bit strange why most old interfaces is perfect filtered by the book and many new ones show such tendency.

Agree using such ringing interface if we want to see speaker IR alone its better into REW not set a calibration file for that loop and instead add from a saved mdat-file that interface loop to current measurement session and divide.

Thanks sharing those DACs data will have a look.

Very similar probably are the same DAC chips. Thanks for the info 192 kHz may be the way to go. I will test that when I get the chance.

Originally Posted by wesayso

Don't stare at these IR's too long .

Ha Ha, very true.

In a speaker measurement you won't find the ringing of a DAC after the main pulse. It will be dominated by different frequencies and/or reflections of the driver itself. You'll only see the pré ringing part. What you're looking at is a combination of the entire frequency response. Though high frequencies usually dominate it's shape.

Maybe, but the ringing is worse that I thought it would be and due to the length of it and the fact that it is very high in frequency means that it could well have an impact on the impulse view. As you say because it is so high it doesn't really show up in the other views unless you filter for it.

It was the speaker measurements that started it all off although that was really the weirdness before the peak which has now been laid to rest.

I'm not surprised you can't hear a difference between them when most of the ringing is above 20 KHz. It's good to know what you have, but there's a lot more data behind that IR rendering which is why there are so many ways to view the same data.

Not much of a difference, I can hear some difference between the filters on the AK but not all of them. Another case where the measurements look worse than the difference in sound. The slow filters in the AK do have quite a pronounced HF rolloff that can be seen in the frequency response measurements I posted in the DAC thread.

[torgeirs]

Just a reflection on the SPL achivable when series/ parallell connecting.
Say you have 50 volt input signal. Then there is 50/5 = 10 volts on each element.
Lets say 85 dB at 2,83 volts and 97 dB at 10 volts.
Power summing of 24 element is 6 x 3 dB or 18dB gain. So max 115 dB

Think sweps are integrated over swep time to simulate pink noise in REW?
So the actual SPL when doing a sweep is about 20 dB less than the resulting SPL shown.
115 dB sin sweep is crasy loud and can damage the hearing.
130 dB sin sweep WILL damage hearing

[fluid]

Originally Posted by torgeirs

It has been mentioned earlyer about overshoot and ringing.
If a low pass filter has Q over 0.5, a step response or impulse response will show this Q value as ringing. It actually is energy storage. To me it looks like the TC9 has some resonance/ energy storage before it falls off at HF.
There really is a lot of litterature on this as it is important when designing stable amplifiers and calculating phase margin.

I'm not sure about that. The waterfall plot looks very clean to me. Here is an un-smoothed plot of an indoor measurement at 20cm.



As it could be forgotten that this is a thread about a line array build I thought I would show what I have been up to today.

I felted the other speaker and put the socks on.

This is how the speakon looks on the back with the sock in place. Pretty happy with that.




I have put them in living room in preparation for some listening, not the best photo but that's the best I can do tonight

[fluid]

Originally Posted by torgeirs

Just a reflection on the SPL achivable when series/ parallell connecting.
Say you have 50 volt input signal. Then there is 50/5 = 10 volts on each element.
Lets say 85 dB at 2,83 volts and 97 dB at 10 volts.
Power summing of 24 element is 6 x 3 dB or 18dB gain. So max 115 dB

Think sweps are integrated over swep time to simulate pink noise in REW?
So the actual SPL when doing a sweep is about 20 dB less than the resulting SPL shown.
115 dB sin sweep is crasy loud and can damage the hearing.
130 dB sin sweep WILL damage hearing

If all of that maths was due to the fact that the waterfall graph says 100dB then don't worry it was not taken at 100dB. The IR was exported and normalised before being re-imported so it could be compared to Byrtt's that had been recorded at a different level.

The max SPL depends on the EQ that is used. The more low frequency EQ is used the lower the overall max spl. Line array gain and increase in sensitivity is not across all frequencies so the overall limit before any eq is set by the ability of a single driver at the highest frequencies. Once you start EQ'ing the low frequencies you will run out of power or excursion before the single driver high frequency limit usually.