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ASR Acourate users

labdoc

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There are a number of threads about Acourate on ASR, but there does not seem to be an "Acourate user group". There is an actual Acourate User Group, but that place is pretty quiet and full of existing Acourate users. I thought I would create a thread so that we can discuss and share what we are doing and get some new ideas on how to use this powerful software, and maybe spread the word to encourage other users to try.

It is my experience that even seasoned audiophiles are blown away when they encounter a fully corrected DSP system for the first time. We all know that speakers corrupt the signal, and rooms corrupt the speaker even more. Software like Acourate allows you to remove these influences and achieve transparency, dynamics, and impact like you have never heard. Veils were lifted, wife heard it in the kitchen, etc. - all that really happens. Forget those miniscule differences in SINAD between DAC's, the objective and subjective benefit of Acourate is magnitudes above that, and I think that every ASR member who is serious about audio should have some kind of DSP in their system.

Acourate is described by the author as a "digital audio toolbox". Although it does do room correction, it is much more than that. There are other room correction products on the market, e.g. Audiolense and Dirac Live. Although I do not own licenses for AL or Dirac, I have friends who use it and I have seen what they do. AL and Dirac are very much focused on crossover generation and room correction. They do a good job, and they do it quickly and conveniently. For comparison, I saw a friend generate a set of correction filters in Audiolense, and it took him 15 minutes from start to finish, from setting up the microphone to filters ready to be loaded into the convolver. I was very much impressed by the speed and ease of use of the software.

However, Acourate is not like that. While it does have room correction macros that provide some degree of automation, performing almost any other function is entirely manual, and requires your participation and interpretation of what the software is telling you. It really is a toolbox, in that you have a bunch of various tools that you have to learn to use and deploy at the right time, a bit like deciding whether you need a screw or a nail to do the job. While both will nominally hold two pieces of wood together, they have advantages and disadvantages and it is up to you to decide which is better. One example I recently encountered in Acourate was when I found that my tweeter had a 90 degree phase lag. My solution was to apply the Hilbert transform three times to rotate it back to position. Uli's solution was to invert the polarity of the tweeter and correct the measured phase lag via a reverse all pass filter.

There are other audio toolboxes as well, such as rePhase, OpenDRC, and others - I have no experience with those, and the intention of this thread is not to discuss other types of room correction software. This is for what Acourate can do, and how to use Acourate only. I politely request that discussion of other types of software should be limited to features that Acourate does not have, or better ways of doing things.

To get the full potential from Acourate, it is best to have full control of each individual driver in your speaker system. In my case, I already own the speaker. My solution was to bypass the internal passive crossovers and solder the cables from the drivers directly to the binding posts. Each driver has its own amplifier and DAC channel, and each DAC channel has an individual correction filter designed with the help of Acourate. I know of another person with a DIY cardioid speaker emulating Kii 3's with 30 drivers, 30 DAC channels, 30 amp channels, and filters for each individual driver made by Acourate. I am also aware that Acourate can work with a passive filter in place, but I do not know how well it would work since I do not have direct first-hand experience.

Here is the thing I love most about Acourate. Unlike ANY other audio upgrade, once you pay the cost of entry (a modest sum of €340 for EU countries, and €286 for everyone else), all upgrades are FREE. Since Uli introduced the first version of Acourate in 2003 (?) there has been one version change, the only time he has ever asked for more money from his users. And even then the upgrade was optional, your old copy of Acourate would continue to work. The other thing that is free are refinements to the way you use the software - because I have learnt a lot since the start of this journey, the measured and audible performance of my system has improved - all without me having to spend a cent. Improving the accuracy of your workflow, refining the way you use this software, and rethinking the way you do things WILL improve the sound objectively and subjectively, which is yet another reason Acourate users should be discussing this.

Unfortunately, for such a complex and powerful piece of software, online resources and even an explanation of what all the functions do is frustratingly lacking. I have to learn by trying out different functions to see if they work, and even then I have no idea of what side effects I may be causing. One aim of this thread is to help complete that information.

These are the corrections I have implemented in my current system:

- Crossovers. Acourate allows the creation of a number of crossover types, including Linkwitz-Riley, Neville-Thiele, Butterworth, and some exotic types like Horbach-Keele and some of Uli's new crossovers which I have never heard of. I have experimented with a number of these crossovers, and I would be keen to discuss with other users the pros and cons of different types, as well as your subjective listening impressions of these crossovers.

- Driver linearization. Uli has a new method of driver linearization which involves convolution of a reverse all pass filter which to my knowledge has not been described anywhere else. I will post the method shortly.

- Time alignment. Although the time alignment procedure in Acourate is well described, there are several insights which I learnt recently that I will share in an upcoming post.

- Virtual Bass Array. I have done some experiments with this, which has been discussed in a separate thread on ASR.

- Bass phased array. I am currently experimenting with this. Discussion to follow once I have results.

- Room Correction with different target curves and settings. There are different strategies for room correction (e.g. up to Schroder, or full range, with harder or softer settings, etc). I have my own thoughts on target curves and I would be keen to hear other opinions. I am still experimenting with "hard" and "soft" settings (e.g. FDW 30/30 vs. FDW 5/5) and my opinion of the subjective sound quality is still out. I am undecided. Let us discuss.

- Interchannel Phase Alignment (ICPA) - a new feature introduced in Acourate v2.0 a few years ago. This aligns the phase of the left and right channels. Although I can see measured improvements, I am not sure I can hear a difference. Keen to hear thoughts on this.

Acourate resources:
- Accurate Sound Reproduction using DSP (Kindle, paperback) by Mitch Barnett (@mitchco on ASR). This is essential reading for every Acourate user.
- Acourate Digital Room and Loudspeaker Correction Software Walkthrough by @mitchco on Audiophile Style, 2013.
- Advanced Acourate Digital XO Time Alignment Driver Linearization Walkthrough by @mitchco on Audiophile Style, 2013.
- Understanding the "state of the art" of Digital Room Correction video on Youtube by @mitchco 2021, with ASR discussion thread (sadly closed)
- Some basic articles and white papers hosted on Acourate.com by Dr. Uli Brueggemann, author of Acourate (@UliBru on ASR)
- Archimago's Musings in 2015 and 2019 update, by @Archimago
- Digital Room Correction HK by ACK Cheng
- Acourate User Group at the Audio Vero website.
- Acourate Wiki (sadly very incomplete)
@Keith_W - Thanks for doing this. If you haven't done so already, consider an "announcement' on the Acourate user forum.
Also, please describe your equipment and listening environment
 
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Keith_W

Keith_W

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It would help if you mentioned this in post 1 or 2. I was thinking you must be full active to get the benefits you describe, otherwise you are just "fighting" the passive crossover(s).

Yeah, all DSP software I am aware of requires an active system for best results. Acourate is the same. You are right, I should have mentioned it, but to me it seemed pretty self-evident.

It seems quite capable, but it is very "proper" and expensive.

By proper I mean the way it works in samples and with functions, whereas something like REW turns samples in to time for convenience, and functions are just things you click/do (tick invert, set time offset rather than do "rotations", do arithmetic, etc).

I agree. You do have to do some maths if you want to make sense of the output. For example, before I do my subwoofer time alignment, I do a quick calculation to see if the result Acourate is reporting is feasible or not. This involves:

- Measure distance from tweeter to subwoofer = 1.2 meters
- Convert distance into time (assuming speed of sound 343m/s): 1.2 / 343 = 0.0035ms
- Convert time into samples (48kHz): (0.0035 * 48000) = 168 samples

Actually, it is a good idea to do this calculation before doing any time alignment using any software. With Acourate, there is only one additional maths step to convert time into samples, and Acourate does report the time, only that it does it in a not so obvious way.

And by being expensive it has probably hampered its own reach (adoption within the DIY community). Between REW and RePhase, it seems many of the same things can be achieved for free now?

I don't think it is expensive at all! It is cheaper than Dirac and Audiolense, and it does more. I have posted elsewhere about the strengths of Acourate vs. Audiolense. I suppose it is more expensive than REW or Rephase (both of which are free!). To be honest I have not looked closely to see what REW and Rephase can do in terms of correction, but I am sure that all the major boxes are ticked.

I am keen to see that info.

Sure, that will involve firing up my other PC, launching Acourate, and taking some screenshots. I will do that later :)
 
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Keith_W

Keith_W

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Are you using this on a stereo or multichannel setup?

Multichannel. See below.

@Keith_W - Thanks for doing this. If you haven't done so already, consider an "announcement' on the Acourate user forum.
Also, please describe your equipment and listening environment

If I wanted a system thread on ASR I would have created one ;) I don't want this thread to be about me, I wanted it to be about Acourate. But since you asked:

Speakers: Acapella High Violon (3 way speaker, with tweeter and midrange horns and bass cabinet) + subwoofer. These speakers have been converted to active with the passive crossover bypassed. So, four drivers per side for a total of 8 drivers for a 2 channel setup.

Amplifiers: a mixture of solid state Class A/B, valve, and Class D amps. One amp channel per driver.

DAC: RME Fireface UC, Merging NADAC. Also on the shelf: Focusrite 2i2, Presonus Audiobox USB.

Microphone: Earthworks M30, Behringer ECM8000 (x2 ... I like to have backups!)

Software: Acourate for creating filters, REW for measuring and providing a second opinion. Playback is via JRiver and Acourate Convolver.

Room: 7m length x 6m wide x 3m high general purpose living room, although I have configured the room to be dedicated to the speakers. Minimal room treatment.
 

Philbo King

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There are a number of threads about Acourate on ASR, but there does not seem to be an "Acourate user group". There is an actual Acourate User Group, but that place is pretty quiet and full of existing Acourate users. I thought I would create a thread so that we can discuss and share what we are doing and get some new ideas on how to use this powerful software, and maybe spread the word to encourage other users to try.

It is my experience that even seasoned audiophiles are blown away when they encounter a fully corrected DSP system for the first time. We all know that speakers corrupt the signal, and rooms corrupt the speaker even more. Software like Acourate allows you to remove these influences and achieve transparency, dynamics, and impact like you have never heard. Veils were lifted, wife heard it in the kitchen, etc. - all that really happens. Forget those miniscule differences in SINAD between DAC's, the objective and subjective benefit of Acourate is magnitudes above that, and I think that every ASR member who is serious about audio should have some kind of DSP in their system.

Acourate is described by the author as a "digital audio toolbox". Although it does do room correction, it is much more than that. There are other room correction products on the market, e.g. Audiolense and Dirac Live. Although I do not own licenses for AL or Dirac, I have friends who use it and I have seen what they do. AL and Dirac are very much focused on crossover generation and room correction. They do a good job, and they do it quickly and conveniently. For comparison, I saw a friend generate a set of correction filters in Audiolense, and it took him 15 minutes from start to finish, from setting up the microphone to filters ready to be loaded into the convolver. I was very much impressed by the speed and ease of use of the software.

However, Acourate is not like that. While it does have room correction macros that provide some degree of automation, performing almost any other function is entirely manual, and requires your participation and interpretation of what the software is telling you. It really is a toolbox, in that you have a bunch of various tools that you have to learn to use and deploy at the right time, a bit like deciding whether you need a screw or a nail to do the job. While both will nominally hold two pieces of wood together, they have advantages and disadvantages and it is up to you to decide which is better. One example I recently encountered in Acourate was when I found that my tweeter had a 90 degree phase lag. My solution was to apply the Hilbert transform three times to rotate it back to position. Uli's solution was to invert the polarity of the tweeter and correct the measured phase lag via a reverse all pass filter.

There are other audio toolboxes as well, such as rePhase, OpenDRC, and others - I have no experience with those, and the intention of this thread is not to discuss other types of room correction software. This is for what Acourate can do, and how to use Acourate only. I politely request that discussion of other types of software should be limited to features that Acourate does not have, or better ways of doing things.

To get the full potential from Acourate, it is best to have full control of each individual driver in your speaker system. In my case, I already own the speaker. My solution was to bypass the internal passive crossovers and solder the cables from the drivers directly to the binding posts. Each driver has its own amplifier and DAC channel, and each DAC channel has an individual correction filter designed with the help of Acourate. I know of another person with a DIY cardioid speaker emulating Kii 3's with 30 drivers, 30 DAC channels, 30 amp channels, and filters for each individual driver made by Acourate. I am also aware that Acourate can work with a passive filter in place, but I do not know how well it would work since I do not have direct first-hand experience.

Here is the thing I love most about Acourate. Unlike ANY other audio upgrade, once you pay the cost of entry (a modest sum of €340 for EU countries, and €286 for everyone else), all upgrades are FREE. Since Uli introduced the first version of Acourate in 2003 (?) there has been one version change, the only time he has ever asked for more money from his users. And even then the upgrade was optional, your old copy of Acourate would continue to work. The other thing that is free are refinements to the way you use the software - because I have learnt a lot since the start of this journey, the measured and audible performance of my system has improved - all without me having to spend a cent. Improving the accuracy of your workflow, refining the way you use this software, and rethinking the way you do things WILL improve the sound objectively and subjectively, which is yet another reason Acourate users should be discussing this.

Unfortunately, for such a complex and powerful piece of software, online resources and even an explanation of what all the functions do is frustratingly lacking. I have to learn by trying out different functions to see if they work, and even then I have no idea of what side effects I may be causing. One aim of this thread is to help complete that information.

These are the corrections I have implemented in my current system:

- Crossovers. Acourate allows the creation of a number of crossover types, including Linkwitz-Riley, Neville-Thiele, Butterworth, and some exotic types like Horbach-Keele and some of Uli's new crossovers which I have never heard of. I have experimented with a number of these crossovers, and I would be keen to discuss with other users the pros and cons of different types, as well as your subjective listening impressions of these crossovers.

- Driver linearization. Uli has a new method of driver linearization which involves convolution of a reverse all pass filter which to my knowledge has not been described anywhere else. I will post the method shortly.

- Time alignment. Although the time alignment procedure in Acourate is well described, there are several insights which I learnt recently that I will share in an upcoming post.

- Virtual Bass Array. I have done some experiments with this, which has been discussed in a separate thread on ASR.

- Bass phased array. I am currently experimenting with this. Discussion to follow once I have results.

- Room Correction with different target curves and settings. There are different strategies for room correction (e.g. up to Schroder, or full range, with harder or softer settings, etc). I have my own thoughts on target curves and I would be keen to hear other opinions. I am still experimenting with "hard" and "soft" settings (e.g. FDW 30/30 vs. FDW 5/5) and my opinion of the subjective sound quality is still out. I am undecided. Let us discuss.

- Interchannel Phase Alignment (ICPA) - a new feature introduced in Acourate v2.0 a few years ago. This aligns the phase of the left and right channels. Although I can see measured improvements, I am not sure I can hear a difference. Keen to hear thoughts on this.

Acourate resources:
- Accurate Sound Reproduction using DSP (Kindle, paperback) by Mitch Barnett (@mitchco on ASR). This is essential reading for every Acourate user.
- Acourate Digital Room and Loudspeaker Correction Software Walkthrough by @mitchco on Audiophile Style, 2013.
- Advanced Acourate Digital XO Time Alignment Driver Linearization Walkthrough by @mitchco on Audiophile Style, 2013.
- Understanding the "state of the art" of Digital Room Correction video on Youtube by @mitchco 2021, with ASR discussion thread (sadly closed)
- Some basic articles and white papers hosted on Acourate.com by Dr. Uli Brueggemann, author of Acourate (@UliBru on ASR)
- Archimago's Musings in 2015 and 2019 update, by @Archimago
- Digital Room Correction HK by ACK Cheng
- Acourate User Group at the Audio Vero website.
- Acourate Wiki (sadly very incomplete)
Nice summary and tutorial! I don't use Accourate, but someone I have huge respect for does: Bob Katz (@ digido.com)
 

Daverz

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I think it's nonsense, I used it for years in a passive setup and got great results.

I agree; I get good results with two very different passive speaker sets, the Buchardt S400s and Vandersteen cloth Quatros. I have used it to integrate an SVS SB1000 Pro subwoofer with the Buchardts, but that didn't require any quasi-anechoic measurements.

I can very much understand the frustration, though.
 

jwdevos

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Hello ASR,

My plan is to learn Accurate and CamillaDSP for room correction and eventually 2-way DIY speakers with active crossovers. I bought the book Acourate Sound Production Using DSP. Under the hardware and software requirements, the following is mentioned.
1704397972381.png


So, no USB mic, separate mic and separate PSU/ADC, clear so far. I'm having trouble understanding the implication about the DAC clocks though. Say I have a PC as a streamer, connected to a DAC like SMSL D6-s (2 channels) or Okto DAC8 stereo (4 channels). Now let's say I want to grab a separate laptop (separate from the streamer PC which implements any DSP) for performing measurements. I might buy a nice Motu interface, connect a nice calibrated mic to that, then connect the Motu to the laptop. I suppose the streamer PC needs to play some kind of test signal.
Do I need to somehow sync the clocks of the Motu on the laptop and the main system DAC on the streamer PC?

Kind regards,
Jaap
 
Last edited:

Daverz

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Hello ASR,

My plan is to learn Acourate and CamillaDSP for room correction and eventually 2-way DIY speakers with active crossovers. I bought the book Acourate Sound Production Using DSP. Under the hardware and software requirements, the following is mentioned.
View attachment 339711

So, no USB mic, separate mic and separate PSU/ADC, clear so far. I'm having trouble understanding the implication about the DAC clocks though. Say I have a PC as a streamer, connected to a DAC like SMSL D6-s (2 channels) or Okto DAC8 stereo (4 channels). Now let's say I want to grab a separate laptop (separate from the streamer PC which implements any DSP) for performing measurements. I might buy a nice Motu interface, connect a nice calibrated mic to that, then connect the Motu to the laptop. I suppose the streamer PC needs to play some kind of test signal.
Do I need to somehow sync the clocks of the Motu on the laptop and the main system DAC on the streamer PC?

Kind regards,
Jaap

You want the same clock for both playback and recording. So you would use the Motu (or similar device) for both. The sweep signal is played by Acourate running on the computer connected to the Motu, an analog mic (with a calibration file) is connected to the microphone input of the Motu, and the analog outputs of Motu are connected to the amp/speaker system. This makes it possible to automate sweeps over all speakers and drivers in one go.

I suggest asking Uli on the AudioVero forum before purchasing any equipment for using Acourate:

 

jwdevos

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Thanks @Daverz. I'll make sure to ask at AudioVero. Any thoughts on different DAC behavior? Say I make measurements and design filters with a Motu, but then switch to an Okto DAC which has much higher SINAD performance. Will this impact my filters in any way? No way to connect a mic on a regular DAC like the Okto as far as I know.
 

Daverz

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Thanks @Daverz. I'll make sure to ask at AudioVero. Any thoughts on different DAC behavior? Say I make measurements and design filters with a Motu, but then switch to an Okto DAC which has much higher SINAD performance. Will this impact my filters in any way? No way to connect a mic on a regular DAC like the Okto as far as I know.

I don't think SINAD matters much for recording since ambient room noise is so much higher. So you might get away with something cheaper than a Motu if you aren't using it for listening (e.g. Behringer UMC204HD).

I do wonder about different DAC filters (e.g linear phase vs. minimum phase). But you also typically cutoff the response at 23000 Hz in the first room correction step anyway (when recording at 48 kHz sample rate) to avoid the brickwall filter.
 
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Keith_W

Keith_W

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Hello ASR,

My plan is to learn Acourate and CamillaDSP for room correction and eventually 2-way DIY speakers with active crossovers. I bought the book Acourate Sound Production Using DSP. Under the hardware and software requirements, the following is mentioned.
View attachment 339711

So, no USB mic, separate mic and separate PSU/ADC, clear so far. I'm having trouble understanding the implication about the DAC clocks though. Say I have a PC as a streamer, connected to a DAC like SMSL D6-s (2 channels) or Okto DAC8 stereo (4 channels). Now let's say I want to grab a separate laptop (separate from the streamer PC which implements any DSP) for performing measurements. I might buy a nice Motu interface, connect a nice calibrated mic to that, then connect the Motu to the laptop. I suppose the streamer PC needs to play some kind of test signal.
Do I need to somehow sync the clocks of the Motu on the laptop and the main system DAC on the streamer PC?

Kind regards,
Jaap

The other issue which is far more serious is that Acourate communicates via ASIO only. ASIO can only access one device at a time, so it expects to see the microphone and DAC on the same device. A USB mic + separate DAC are two devices, if you set Acourate up to output to the DAC you won't even see the USB Mic appear on your drop down list and vice-versa. It may be possible to gang together an unholy marriage between the two devices using software like ASIO4ALL, but I wouldn't bet on it working, and ASIO4ALL has its own issues.

Uli recommends that you use a multichannel interface with as many DAC output channels as you need. For my system, which has 8 channels, he recommended an RME Fireface UC, but I am sure that a Motu would work just as well. If you need more than 8 channels, then those types of interfaces are rare and you will have to cobble your own together with Dante or Ravenna devices.

If you plan to measure like this:

Motu (mic input) --> Laptop (with Acourate) --> Streamer PC --> Okto DAC8

There is NO WAY to synchronize the clocks between the Motu and the DAC, so this approach is not recommended! Furthermore, goodness knows what your streamer PC is doing to the signal, you might introduce additional latency and this would throw off any timing measurements done with Acourate. You would run into the same problems as using a USB Mic. Instead, your measurement setup should look like this:

Motu (mic input) --> Laptop (with Acourate) --> Motu (DAC output)

If you want to test the performance of the convolver on your streamer PC, setup is a little more complicated and involves use of more third party software.
 

jwdevos

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Thanks for the directions everyone.

Motu (mic input) --> Laptop (with Acourate) --> Motu (DAC output)

This will be the approach, thanks for pointing me in the right direction. Most concerns I had are addressed. The only downside is having to implement the filters in CamillaDSP after finishing the Acourate process, and then having no proper way to verify the results. Could I play an external sweep file on the streamer, then pick that up with Acourate to get a broad sense of the CamillaDSP filter implementation?
Eventually I'd use the laptop, Motu, mic and Acourate to check out friends systems too. I might encounter different convolvers or just PEQ's, like the built-in Roon options for example.
 
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Keith_W

Keith_W

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This will be the approach, thanks for pointing me in the right direction. Most concerns I had are addressed. The only downside is having to implement the filters in CamillaDSP after finishing the Acourate process, and then having no proper way to verify the results. Could I play an external sweep file on the streamer, then pick that up with Acourate to get a broad sense of the CamillaDSP filter implementation?
Eventually I'd use the laptop, Motu, mic and Acourate to check out friends systems too. I might encounter different convolvers or just PEQ's, like the built-in Roon options for example.

Acourate generates a multichannel stereo .WAV that it loads into its logsweep recorder so you can verify the measurements from Acourate directly without using an external convolver.

However, if you wish to send the signal through CamillaDSP and measure with Acourate, this is more complicated. It will go something like this:

Acourate 2 channels out --> Motu (loopback) --> CamillaDSP (8 channels out) --> Motu for DAC --> SPEAKERS --> Motu (Mic in) --> Acourate
 

3ll3d00d

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Could I play an external sweep file on the streamer, then pick that up with Acourate to get a broad sense of the CamillaDSP filter implementation?
It would be simpler to use rew to verify the output as then you can use it's external sweep feature
 

Dathzo

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Good day and thanks @Keith_W for the great info sharing.

A pretty basic question. I have an AVC (Denon X3800) that allows having up to 4 independent sub signals. I have a 2.2 system at the moment. I have the AVC connected to my PC. Would it be possible to use Acourate with this setup? Would the software see 4 independent channels?, i.e., 2 mains + 2 subs? Or is it a multichannel sound card like RME, Motu a must?

Many thanks
 
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Keith_W

Keith_W

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Good day and thanks @Keith_W for the great info sharing.

A pretty basic question. I have an AVC (Denon X3800) that allows having up to 4 independent sub signals. I have a 2.2 system at the moment. I have the AVC connected to my PC. Would it be possible to use Acourate with this setup? Would the software see 4 independent channels?, i.e., 2 mains + 2 subs? Or is it a multichannel sound card like RME, Motu a must?

Many thanks

I will split your question into two answers - recording and playback.

1. For recording sweeps: your Denon X3800 MUST have an ASIO driver and microphone input capable of supporting a calibrated 48V Phantom Power microphone. You could use whatever mic came with your Denon, but it is unlikely to be calibrated, and not of high quality. I checked Denon's website, and it has no ASIO driver. So I don't think you could use your Denon to record sweeps. In this case, you will need a multichannel interface with as many DAC outputs as you need, e.g. a Motu Ultralite Mk.5 or RME Fireface UCX.

Other software (e.g. Audiolense) allows you to use a USB microphone, but there are many reports of inconsistent timing measured with USB microphones. This is because the microphone ADC is not clock synchronized to the DAC. I am starting to take the view that USB mics are good for frequency response sweeps only, and they are too inconsistent for time alignment. This is not based on first-hand experience, I formed this view from reading about complaints of USB mics on other forums.

2. For playback: in this scenario, you measure the sweeps with Acourate and playback via the Denon. In this case, you will need either:

(2a) the Denon is capable of convolution. I checked Denon's website and it does not appear to support this. The online manual suggests that the Denon only supports Audyssey. I don't know how you would even get filters generated by third party software (like Acourate) onto your device. In any case, Acourate only outputs .WAV files, and if you need it in another format, you will need more software to convert it into another format. Your Denon is likely to be incapable of 65536 tap FIR filters given that very low powered DSP chips are usually installed in AVR's. Alternatively:

(2b) you use your Denon as a multichannel DAC, and use a convolver hosted on your PC (like Acourate Convolver, Hang Loose Convolver, JRiver, Roon, etc) to do the processing and send processed signals to the DAC. In this case, your Denon needs to be recognized by Windows as an audio device, preferably via ASIO or WASAPI. Sadly for you, it does not appear to be the case for either. There is a downside to doing this however - all FIR filters introduce latency, which will cause lip sync problems if you use your AVR for video as well. You will need some way to adjust lip sync with your AVR.

Your Denon already has Audyssey built in. You could use it. The advantage of doing this is that your hardware already supports it, and it is likely to produce a "good enough" result. It won't produce ultimate quality - if that is what you want, you will need a major reconfiguration of your system and be prepared to climb the DSP learning curve. You will also lose convenience and may even lose functionality - one example is getting any convolver (not just Acourate) to process HDMI audio. For this you need two additional pieces of equipment - a HDMI splitter, and an interface card capable of internal routing. Audio goes into the HDMI splitter, which extracts the audio and sends it to the interface card. The card then routes audio from the digital input into the convolver input, and the convolver outputs via the DAC on the interface card.
 
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