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Article: Advanced Acourate Digital XO Time Alignment Driver Linearization Walkthrough

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Very impressive work - worth registering for as a new member! It strikes me that the many steps you went through are straightforward but rather complicated, and lend themselves to a rules-based approach to perhaps automate a bit that would help the engineering-impaired. Perhaps Uli (with help from the likes of you) could develop something within Acourate to do that. I don't propose that the result would be as good as what you achieved, but what if Acourate had a "novice" option to use a more rules-driven, automated approach, and an "expert" option that is essentially manual like your experience?

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Very interesting article. The price of the software was holding me back until now, but this brings a whole new angle to the package.

Would it be possible to send the 6 xo channels as a 6 channel pcm signal over S/PDIF from J River to a Surround Amp/Processor/Pre Amp?

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Mitchco, on a note related to justM, when you get a chance could you add a diagram of exactly how you connected all of your components to do what you did? I did look up the Hilo user guide and I see how that can be configured for 6 channels (impressive), but figuring out exactly how everything was wired is a struggle for me. Thanks!

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Mitchco, on a note related to justM, when you get a chance could you add a diagram of exactly how you connected all of your components to do what you did? I did look up the Hilo user guide and I see how that can be configured for 6 channels (impressive), but figuring out exactly how everything was wired is a struggle for me. Thanks!

 

 

I think I figured most of it out. You connected a digital stereo out from your PC to the Hilo, and then from the Hilo to each of 3 stereo (or 6 mono) amps, and then from the amps to each of the drivers in each speaker. I'm not clear on how you exactly connected the microphone to ensure accurate loopback capability for timing measurements.

 

Also, please correct me if I'm wrong, but it would seem that one could not do accurate time measurements with a USB microphone like the UMM-6 because it has an A/D converter embedded in the microphone that would be clocked differently than the digital signal output from the PC.

Edited by lazy

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Hello justM, have a look at the link in the article, just before the Conclusion, that points to Tony Knight's system. There is a diagram that is somewhat different than what you are asking for, but is conceptually similar.

 

Hello lazy, yes, the Hilo is connected via USB to the computer and Hilo's 6 analog outputs are input to the 6 amplifiers. Internally, the Hilo has a 32 x 32 channel mixer that allows one to route any input to any output, including digital and analog loopback. However, Acourate does not require patching manual loopback, but does require ASIO. With respect to measurement mic requirements, and how to hook up, please refer to the intro article on Acourate.

 

Hello Archimago, thanks for the kind words. Keep up the great writings!

 

Best regards, Mitch

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Jonathan Valin published a review for a $140,000 pair of speakers in the October 2013 TAS. He went to the speaker factory and observed the driver nonlinearity on the "scope." In the end, he concluded "DSP Can NEVER work."(emphasis not added). I don't even want to mention the openly defensive and hostile response in last month's letter to the editor to a reader who challenged Mr. Vain's conclusion.

 

Mitch, I think you've demonstrated otherwise with slightly less expensive gear. :-) The problem with DSP is getting folks to accept that all rooms and speakers are imperfect. The challenge is to carefully use the available technology to overcome. No matter how good or bad the system, there are huge benefits for everyone. That scares some speaker manufacturers a little, I guess.


THINK OUTSIDE THE BOX

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Great work and an inspiration Mitch! I am curious what sort of linearization can be done with acourate for a passively crossed-over speaker in addition to RC functionality. I gather that a near field measurement of say a 2-way can be linearized in the same manner as individual drivers given a certain bandwidth parameter, then RC correction filter at far-field? I can't wait to start playing with acourate once I have the extra bones to pull the trigger. Hopefully within the next couple months. Your articles are going to be incredible resources for all of us, so thank you kindly for not only the effort to optimize your own system by tlearning the steep curve here, but also for memorializing the process for the rest of us. You rock!

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Hi Michael, thanks! While I was working at the state of the art studio, I gave up on my home system, which at the time were Magneplanar 2.6R's (with real ribbons) paired with a Classe Audio DR2 Class A amp. Nothing wrong with the gear, but the sound quality could not compare to the purpose-built acoustically designed facility. And for good reason, that state of the art facility cost several hundred thousand to construct (without gear).

 

I got back into it not too long ago when I came across high resolution DSP software designed to optimize ones speakers and room. At the time I had a very lo-fi setup (Logitech G51 plastic computer speakers :-) and to my surprise, I heard an inkling of what may be possible. The full realization of which is now contained in this article.

 

Hi scintilla, thanks for the kind words. Yes, Uli confirms that one can measure a passive speaker in the near-field (best in free-field to avoid as much reflections as possible) and derive an overall linearization filter. The filter can also be used as pre-filter in the LogSweep Recorder. The final room correction filter will contain both the speaker correction and room correction. Have fun!

 

Best regards, Mitch

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Hi Mitch,

 

Been a bit of a voyeur when it comes to Acourate, but your post has come at the right time and has persuaded me to push the button on the software and mic.

 

I have bought a Void Indigo speaker system, and need to use the XO function as well as the DRC of Acourate to get it up and running.

 

To that end i have bought an 8 channel ADC / DAC - the Lucid 88192 which i can find at a good price.

 

Jriver will take over the front end and volume control duties.

 

i will send digital audio out via the toslink on my computer and into the Lucid running at 48KHz (until i find a 24/192 card with a decent AES digital output.)

 

Is relying on JRivers volume control a safe option and are there any glaring problems you would forsee?

 

Thanks

 

Mike

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Hmmm - what i thought was an ADAT port on the back of my computer is actually an SPDIF via toslink.

 

Are there any cheap multichannel soundcards that output digital over AES / EBU at 24/192?

 

If not i will have to move to a lynx aes16...

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Hi Mitch,

 

Been a bit of a voyeur when it comes to Acourate, but your post has come at the right time and has persuaded me to push the button on the software and mic.

 

I have bought a Void Indigo speaker system, and need to use the XO function as well as the DRC of Acourate to get it up and running.

 

To that end i have bought an 8 channel ADC / DAC - the Lucid 88192 which i can find at a good price.

 

Jriver will take over the front end and volume control duties.

 

i will send digital audio out via the toslink on my computer and into the Lucid running at 48KHz (until i find a 24/192 card with a decent AES digital output.)

 

Is relying on JRivers volume control a safe option and are there any glaring problems you would forsee?

 

Thanks

 

Mike

 

Hi Mike, cool project!

 

With respect to 24/192 AES card, in addition to CA, you might try searching/posting at the Acourate forum and JRiver sound card forum if you haven’t already.

 

JRiver’s digital volume control has worked flawlessly for me. The volume protection works perfectly and the SQ is as good as it gets.

 

Let us know how your project progresses.

 

Best regards, Mitch

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Thanks Mitch,

 

I will post some pictures Christmas and new year, when i get a chance to get things up together. The Lucid converter was a bit of a punt, but at £400 for 8 channels of AD / DA, it has got to be worth a shot.

 

Will pick up an AES card now, so the computer can talk to the DAC.

 

Have been selling off random bits of audio equipment to fund the project - should keep me entertained for a while!

 

The last bit to get will be Acourate :-)

 

Cheers

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mitch,

 

i would love to see a walk-thru for us mac folks using e.g. PureMusic/Vinyl's built-in crossover and fuzzmeasure plus a calibrated mic. i've been trying to time-align the three drivers of my speaker using fuzz, but i'm not sure whether using impulse or step response or time aligning for phase.

 

also i wonder if i'm really the only one that does not like LR 4th order crossovers? i find they do more harm than e.g. a 3rd order butterworth.

 

christian

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Mitch,

 

Great article, thanks for the in depth look at this.

 

The amplification is Nelson Pass DIY A40 stereo Class A amplifier that drives the 97 dB @ 1w/1m sensitivity 15” 4 ohm woofers and 4 x DIY Amp Camp Amp’s (ACA) single-ended Class A Mosfet amplifiers drive the 110 dB and 114 dB @ 1w/1m sensitivity midrange and high frequency compression drivers respectively. While the ACA’s are rated at 6 watts, with the sensitivity of the compression driver/horn combos, I very rarely push 1 watt into them.

 

In college, one of my projects was electronic xover. I took a VAX11/780 (dating myself) that would measure the drivers and determine the xover points which then I could use to make the xover. But an interesting part of that study was one about amplifier usage. I was using a modified Dynaco Mark III for the base and then my stereo push pull EL34 amp (not an ST70) on the midrange and a SET triode connected EL34 for the tweeters.

 

I went back and forth and so much more and we talked about the design with a ton of people then it was realized that between 150-8KHz that you really had to use the same amplifier. The reason was that the differential in sound between the amplifiers was greater than the differential in sound of the drivers or the electronic xover.

 

The differential in sound of say a SET transistor amp and a PP transistor amp is going to be huge. Same for tube stuff...

 

@ RMAF this year we showed plasma tweeters which had their own electronic xover and driver. The same thought went into this that my papers had back in college. Some real thought has to be made here and people who think midrange starts at 1KHz are all nuts. Remember open E string on a guitar is 80Hz!! Middle C is what like 260Hz on a piano and A over middle C is 440Hz. Most of the vocal range happens way before 1Khz.

 

I think if you are looking for a more accurate system you may want to take a look at the amps.

 

Thanks,

Gordon

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Hi Mitch,

 

Excellent article.

When I read about Acourate months ago, I was skeptical about its capabilities. Software like that in your hands truly shine. Now I can get the whole picture and I just want to buy it to apply the knowledge I got by reading your *amazing* article.

 

Digital XO is the future, now.

 

Thank you again.

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Hi Christian, I wish I could walk through the software you listed. Unfortunately, I am not a full-time audio reviewer, perhaps someone else on CA has experience with the software you have…

 

Hi Gordon, thanks for your comments. With respect to identifying frequencies, I like the classic Carnegie Hall chart that can be ordered separately or comes with Bob Katz’s excellent book on Mastering.

 

Re: amplifiers. As a system designer, using state of the art Digital XO, I can match individual speaker components to amplifiers of choice. My choice of amplifiers are based on what the best match is for each individual speaker driver to most accurately reproduce what is stored in digital media on disk.

 

As mentioned in the article, the 15” bass driver/cab requires some power and a reasonably high damping factor to accurately track the electrical waveform being fed to it. However, the compression drivers require very little (literally milliwatts of) power and low damping factor to accurately track. From a system design perspective, I can tailor fit a high power, high damping factor amplifier to provide the bass driver with transient impact and damping control at the same time. With the ACA’s sonic signature similar to Nelson Pass’s commercial Aleph J design, I can get a very detailed, yet smooth response from the compression drivers/horn combos, which have very different electrical load requirements compared to a 15” cone bass driver loaded in a BR cab.

 

Re: “The reason was that the differential in sound between the amplifiers was greater than the differential in sound of the drivers or the electronic xover.” I would say that the sonic signatures between the two Nelson Pass amp designs are somewhat audible if directly compared to each other. However, from a system design perspective, they are synergistic when used together to match the individual electrical load characteristics of each driver, which is what I am after.

 

Take a look at any of the acoustic frequency responses, distortion measurements, and impulse/step responses in this article. Even with the corrected responses, the variability is still greater than virtually any modern amplifier would measure today. Speakers and rooms have the most variability in the audio chain and can benefit the most from modern DSP. Hence the article on Advanced Acourate and by following the steps in the article one can get closer to accurately reproducing music that is stored digitally on disk.

 

perolater, thanks for your kind words. Acourate is state of the art Digital XO with capabilities to time align and linearize all drivers, and provide room correction. It is an ear opening experience. Let us know how your project goes!

 

Best regards, Mitch

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Hi Mitch,

 

I'm sorry I'm very late to this, I'm looking for a way to connect up a Klipsch horn system in a similar way. Can you confirm that when you say - "the Hilo is connected via USB to the computer and Hilo's 6 analog outputs are input to the 6 amplifiers" - you are using the monitor and headphone outputs as well as the line outs? If this is the case have you noticed any differences, other than the cable terminations, between the three outputs? Do you think that using the Hilo is an improvement on using a PCI card? I ask because I'm getting lost in the spec sheets of the various professional audio interfaces. Many thanks.

 

David Whistance

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Hi David,

 

Yes, in my case, the line out goes to the bass amps, the monitor out goes to the mid amps, and the headphone out goes to the HF amps. I have extensively tested and listened to each of the three stereo analog outputs and cannot detect any differences. The specs are virtually identical for each set of analog outputs.

 

The monitor and headphone outs have an independent analog volume control which allows one to trim and match the levels given the (very) different driver sensitivities (e.g. woofer versus compression driver). As a side note, low noise amps on the mid and treble compression drivers will be important.

 

I am extremely happy with the SQ of the Hilo. Personally, I feel its transparency is second to none. I had a Lynx L22 PCI card and while it is no slouch, I believe the Hilo is a significant improvement. The flexibility of the Hilo allows me to do virtually anything I want and I have yet to come across a patch/routing scenario that can't be done.

 

Here is a link to a few other converters: High-quality 8-channel analog output that may be of consideration.

 

Last point, ensure that the interface/converter has a quality ASIO driver. Hilo's is one of the best and is multi-client. Other's not so much and another check point on the list.

 

Let us know how you make out!

 

Cheers, Mitch

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Hi Mitch

 

Many thanks for the really quick and helpful reply. I'll have to start start saving for a Hilo, or at least looking for a buyer for my current valve preamp. I'll also have to look for a compatible amplifier to use with my moving coil cartridges so I can try using PureVinyl rather than my phone stage. At the moment I use a set of Supratek Merlot valve amps but will probably move to solid state for tri-amping, mostly on cost grounds. I am thinking about three JLH 69's or First Watt amps but will have to see what funds and time allow.

 

David

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Mitch, thanks for this great article. Based on the article, I just started using Acourate. So far so good. Your article helped a lot.

 

I have a few questions about input sample rates and filter sample rates.

 

Q: Acourate will compute filters for multiple sample rates: 44, 48, 88 etc. by checking the boxes in macro 4. Is it a best practice to just run once (for example using a 192 input sample rate) and check all the boxes in macro 4? Or is it better to run the macros multiple times, for example the 1st time with 44 input rate and only check the box for the 44 output filter in macro 4, the 2nd time with 48 input rate and only check box for 48 output filter in macro 4, etc?

 

I guess I'm wondering if sample rate conversion has any negative effect.

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