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Keith_W

My honest review of Acourate

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2 minutes ago, Keith_W said:

(e.g. did you know that Dirac ONLY generates filters in its own proprietary format which can not be exported, so the ONLY convolution engine you can use with Dirac is its own!)

Thanks for all the interesting commentary on Acourate but I have to ask about this minor point.  Dirac can/does generate VST-compatible filters which I run with JRiver.


Kal Rubinson

Music in the Round

Senior Contributing Editor, Stereophile

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I am also using Dirac's vst plugin with Jriver. I wish Dirac had a crossover capability. I am staying away from Acourate because the reasons you wrote about.


HTPC/ Jriver - Minidsp nanoAVR HDA dac - MSB-MVC-1 volume control - Cinepro 2k6 amp - Geddes Abbey speakers plus 4 x 10" Aurasound subs

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Thank you, Keith_W for your succinct summary and review of Acourate. Great suggesting improvements which could broaden the appeal of a technically superior product. Because of its complexity, and the user being dependent on the creator to have the full benefit, the product is extremely vulnerable. For example should the creator fall ill, where does that leave the user?

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6 minutes ago, mitchco said:

Finally, there will absolutely be a 2nd edition of my book expanding on what currently is there, plus the addition of other DSP software that I feel is on par with Acourate from a SQ perspective. 

 

 

 

Great Mitch! Any timeline on that?

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Hey Gadgetry, not yet. I am still evaluating which DSP tools will make the next Edition. Also, I have been collecting feedback from the community to see which DSP features provide the most interest for the majority of folks. I have a list from Keith, but also looking for common goals. For example, subwoofer integration is high on the list for many folks...

 

Is there something specific you are looking for?

 

Kind regards, Mitch

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Hi Mitch, thanks for posting in this thread. Do you know of any good resources to learn more about DSP besides the book that you wrote? 

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I don't understand how you get confused at this point.

 

Align microphone with acourate tool require seconds only.

Generate filters (4 way) is less than 1 min.

Mesurement is 60 sec

Run the 4 macro is less than 60 sec

 

 

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On ‎2‎/‎04‎/‎2018 at 8:19 PM, Keith_W said:

For this reason I am going to move away from Acourate and try something else. 

 

How are the investigations of other options proceeding?

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I find Acourate great: works very well, very flexible, and not very expensive. Sure it requires you spend time to learn and ask questions to figure things out...but isn't that discovery process part of the fun? It is to me. Just like learning about building speakers.

 

Per Mitch's article I looked into Audiolense, but it's a lot more expensive.

 

I have found Uli and the Acourate forum members very helpful. And I certainly have no DSP school training.


Roon Server: Gigabyte GA-Z170X UD5, i5 6600, Crucial MX200 SSD, HyperX 1x16GB RAM, Seasonic 520W 80+ PSU, 3 HDD for media, Zalman Z3 Plus case, Win 10 Pro, convolving in Roon digital xo + DRC + upsampling. Output through ethernet.

Roon Client AudioPC: Intel S1200KPR, Xeon E3 1265lv2, Crucial V4 SSD for OS, 8GB ECC RAM, PPA USB card v2, wide range picoPSU, PPA SATA cable for SSD, Streacom FC8 Evo case, lab linear PSU for picoPSU, Windows Server 2012 R2, AudioPhil's Optimizer, Acourate (digital crossovers, room correction )

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I wish Acourate had a batch mode like Matlab, where you could replay the Acourate history file, for example, after re-editing it. In fact I am starting to use Matlab instead to create filters for xovers. Its been 30 years since I used Matlab, so I am a bit rusty.

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Thank you for the balanced review Keith. Do you have any plans to try Audiolense? I have tried finding pros and cons between the two software but have not been able to find much.

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3 minutes ago, StreamFidelity said:

According to my previous experience, the success of the room correction stands and falls. Who just thinks to use the default setting (linear frequency response) has actually lost already. I have recreated the sounding of my speakers.

 

Hi, do you limit room EQ to schroeder frequency (200/300/whatever Hz) or apply room EQ to the full audio bandwidth, 20-20kHz?

 

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Time optimization by phase correction is always performed over the entire frequency response 20 Hz - 24 kHz. Currently I'm trying an amplitude correction only up to 1.2 kHz. It sounds great.

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I tend to agree with some of the interface frustration but I have had success with 3 of my systems now for basic room corrections.  I took analog filter design and DSP in college more than 20 years ago, but I don't really think any of it helped.  All I can remember from DSP (even though I still have my book, homework sets and notes) is suffering through all of the algebraic substitutions for time series and trying to grok convolution. None of that is particularly helpful for using acourate.  Like most things, I think it is simply an iterative learning  process that builds competence. I bought Mitch's e-book but haven't started it yet.  I am sure I will learn a few more things that way too. In the end, my system now sounds better than any system I have had in my life. Using Pure Vinyl with a plugin convolver Roon and HQPlayer, I can RC every source including vinyl, casette tapes, an FM tuner, CD's and digital files. Let me tell you, a casette made in the early 80's on an early 70's pre-Dolby BIC deck by my father of a 1950's record sounds a whole lot better than I thought it did once I could hear it properly reproduced.

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On 5/10/2019 at 7:01 PM, StreamFidelity said:

Time optimization by phase correction is always performed over the entire frequency response 20 Hz - 24 kHz. Currently I'm trying an amplitude correction only up to 1.2 kHz. It sounds great.

Above methods  - for 2 channel ... does this cover 2.2 , stereo + stereo subs (in my room, subs are confined to corners, ie in line of L, R front to listening position)?


elgar iso tran(2.5kVa, .0005pfd)>consonance pw-3 boards>macmini (16gb ram-ssd-clones psu-mmk)[hd(psu)]>netgear gs105 switch(uptone wallwart)>ghent js ethernet>ultraRendu(lps1.2)(js-2)>uspbc>iso regen(lps-1/naq psu)>curious link>rme adi-2 dac (js-2)>cawsey cables>naquadria sp2 passive pre> 1.naquadria lucien mkII.5 power>elac fs249be + elac 4pi plus.2> 2.perreaux9000b(mods)>2xnaq12” passive subs.

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10 hours ago, jamesg11 said:

Above methods  - for 2 channel ... does this cover 2.2 , stereo + stereo subs (in my room, subs are confined to corners, ie in line of L, R front to listening position)?

I know from other users that subwoofers can be integrated with Acourate. Some even create the crossovers for speakers. Unfortunately, I have no experience with it.

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On 5/10/2019 at 4:46 AM, StreamFidelity said:

Many things written here are correct. In a simple stereo setup, however, Acourate is no higher science.

@StreamFidelityBeen reading about Acourate and I think I may give it a go. I have a question about mics. Currently I use a UMIK-1 USB mic with REW. It looks like the author recommends a separate mic/preamp combo to run Acourate, but UMIK-1 works but maybe with some errors? Any advice here? 

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that is also my setup (UMIK-1 and Acourate). I've read about the potential clock drift and Ulrich Brüggemann explained on how to omit it (have not had the time to try it out myself though):

 

(https://groups.yahoo.com/neo/groups/acourate/conversations/messages/9557😞

The Dirac option introduces Dirac pulses before, between and behind the logsweeps. Thus it is possible to measure the number of samples between the Diracs in the recording. If the number differs to the number of samples in the original logsweep then it is possible to calculate a correction factor. A multiplication with the samplerate gives a new samplerate. So you can apply a samplerate conversion (TD-Functions - SRC - special) to the logsweep recording and finally run a logsweep convolution on the resampled recording. This procedure allows to compensate a clock drift or time stretching/compression effect caused by a USB microphone.

 

Here's a workflow some user has posted:

1.    A mono logsweep is generated, including Dirac pulse option, at 48 khz samplerate

2.    The distance between the Dirac pulses is 1080000-24000=1056000 samples in the original sweep

3.    In the rec logsweep the distance between pulses has been found as 1056021 samples (21 more)

4.    The logsweep_rec has been resampled at 47999.045 hz samplerate 

5.    Now the distance between the Dirac pulses is again 1056000 

6.    A logsweep convolution (menu Logsweep/Logsweep convolution) has been done

7.    The result, Pulse48L, has been saved

 

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5 hours ago, sledwards said:

Currently I use a UMIK-1 USB mic with REW.

 

Unfortunately, I have no experience with UMIK-1 USB Mic. I'm using the Microphone calibrated by AudioVero. The calibration file compensates the irregularities in frequency response of the mic. This is essential, otherwise the measurements are wrong.

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28 minutes ago, StreamFidelity said:

 

Unfortunately, I have no experience with UMIK-1 USB Mic. I'm using the Microphone calibrated by AudioVero. The calibration file compensates the irregularities in frequency response of the mic. This is essential, otherwise the measurements are wrong.

The UMIK-1 has calibration files (0 and 90 deg position). What I am not sure is if they are compatible with Acourate.

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