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New loudspeakers & room digital correction service


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Hello,

 

After some years of development we are now launching a digital room correction service. The idea is to process room measurements and propose the best filters to improve the listening experience. Details are available on our website: www.homeaudiofidelity.com

As there is no need to insert new hardware or software in the reproduction chain you can take 100% benefit of your audio system.

Compared to existing solutions (Dirac, Trinnov, DEQx, etc...) we have developped specific algorithms focused on the spatial dimension of the stereo reproduction (directivity issues, crosstalk reduction).

As soon as you have available measurements you can test our solution for free.

Don't hesitate to contact us !

 

Thierry

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I am surprised because must of your questions do have answers in the website (have you browsed the sub menus ?) Here are the answers:

The measurements have to be done by yourself using a mic and a measurement SW as explained in the "requirements" section.

The output is mainly the filters but we can join complementary info as you suggest (plots in temporal & frequency domain with before/after comparison)

The filters are generated by a proprietary solution but you can use them in almost any player having convolution capability. This is a one time payment per filters set but with discount policy if you make modifications in your installation and you need a new iteration.

 

Cheers,

Thierry

 

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Hello Kal,

 

You can't change the radiation pattern of any loudspeaker with signal processing but you can change what is actually heard at the ears. Without going into details the idea is to determine what would be the response at the ears with an ideal loudpseaker radiation pattern, and determine which signal correction is needed with the "non ideal" loudspeakers to come close to this ideal case.

 

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@Bob : thanks for the alert, I will check what we can do with the web site provider

 

@jiminlogansquare : to my knowledge you can't export measurements made with Dirac. As you have a mic you can easily perform new measurements using REW for instance.

 

@Kal : the effect of loudspeaker directivity is more pregnant in the reveberant tail of the room transfer function. Traditional equalization procedures either ignore this reflected energy with time windowing or consider it as a whole with the direct sound (power response with spectral smoothing) In all cases short filters are used which concentrate the energy at the onset of the signal and by consequence do not respect optimal time/frequency distribution.

This is visible on a time/frequency map comparison for the example of a full range driver which has directivity issues above 5kHz (with the classic equalization scheme there is still a lack of energy in the reverberant tail above 5kHz)

The difference in the approach is also visible on the typical shape of filters impulse response

Standard_filters.thumb.png.3485b6975a831943a5ef11f79ea244bb.pngHAF_filters.thumb.png.0b1410130950e5ace04dc83e0347b4dc.png

impulse_short.thumb.png.1ad05c93c1b15b15794acfcc03ece66b.pngimpulse_HAF.thumb.png.15de569209a19f387660d69613988dd2.png

 

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I understand your concern and I have updated the pricing policy to make things clearer.

Some comments from my side : you can freely test the solution and decide wether or not the improvement (like cross talk reduction for instance) is worth the cost.

I also launch this service because I do think that digital correction brings major improvements and must be affordable for anyone, not only from the cost perspective but as well from the invested time. What I mean is that when using a software like Dirac you are in some way "trapped" in an endless iteration procedure because you can play with the target curve and you might think that you can find an even better tuning each time... 

New measurements sets are really needed if you change your loudspeakers or the listening geometrical configuration. Note that you can use a software like REW to pre determined the optimal geometrical configuration if you have some degrees of freedom

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@jminlogansquare: thanks for your encouragement & for the Ron Popeil story (who has not crossed the ocean as you thought...) You perfectly explained the concept and I just want to emphasis that there are some added features vs a tool like Dirac in regards to signal processing : for instance the cross talk reduction brings stereo imaging to a much higher level and this is really something addictive once you have tested it !

 

@wgscott: the solution is working with following players : Roon, Jriver, HQplayer. Globally the Mac world is poor of multi channels convolution engines. There is one AU plugin that Yucca06 tested (Butterfly plugin) but it is quite expensive

 

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Those convolvers are not compatible with our technology because there are of "one to one" type : this means that left channel is convolved with a filter and send to left output. Same goes for the right channel. What we need is left channel convolved with 2 different filters with respective output sent to left & right channel. This is why I mentioned the butterfly plugin which offer such flexibility (a filter set = 4 different filters)

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@pkane2001 Yes HQPlayer is fully compatible as it allows multiple filters with flexible channels mapping.

For the free trial you can use the contact form Contact form and I will apply the filters on one or more test tracks of your choice. As the filters are long non minimum phase FIR filters they can't be converted into REW EQ files (biquad filters)

 

@Bob SternHere is the answer I got from the website provider support team "To be honest with you that's the first time I've heard of this issue. I've checked here in the office with a few different Mac devices using Safari and didn't have any problems visiting your website. The problem seems to stem from the user side. I'd advise them to do the following:

  1. Check if the issue occurs in other browsers (we recommend Google Chrome).
  2. Ensure Safari has been updated to the latest version.
  3. Make sure that their OS is updated to the latest possible version for their device."

So I am little bit puzzled...

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I am really sorry for this website issues despite the testing phase done not only in the windows world. I will not give up and chase the provider.

As we are not selling a software but only filters, this Mac/windows risk does not exist for the end product.

 

@pkane2001 : our goal is to propose the optimal filters set by default. But if you have some preference in terms of tonal balance we can adapt to different kind of house curve format (text file, picture) The process is to start by our proposal and iterate if it is really mandatory 

 

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It would be interesting to have measurements of your loudspeakers because I have never seen pure linear phase ones. Nevertheless if there is no phase issue to correct then the DSP will not change anything and you should not have any bad impact on imaging. By curiosity which type of software have you tested ?

Note that even if you have a perfect loudspeaker phase response the room response is always mixed phase and DSP can help for that as well.

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  • 1 month later...

Such in situ service (measurements / calibration / validation) would be possible in theory but certainly at a prohibitive cost to cover travel expenses. Validation with post EQ measurements is of course possible but usually the results are very similar to the simulation and this is why I considered it as optional.

 

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Danold, if you're question is about how to make required measurements please find below some instructions.

 

As an introduction to REW software, there is a good and simple tutorial available here  https://www.minidsp.com/applications/acoustic-measurements/umik-1-setup-with-rew

 

Once you have set the level (by using left loudspeaker for instance) you'll have to keep it for all the different sweep measurements.

 

From a practical point of view I suggest to follow this procedure:

- set the microphone at the sweet spot position

- set the level using the left loudpseaker

- perform the sweep measurement for this position & left loudspeaker and rename it L1

- perform measurement for this position but right loudspeaker this time, rename it R1

- move microphone to the next position & perform the measurements for right then left loudspeaker, name it R2 & L2

- repeat the procedure for the others different positions (I recommend sweet spot + 4 others as a minimum)

- save the bundle of measurements as a .mdat file

 

Hope it will help,

Thierry

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Hello John

 

It is pretty easy to make good enough measurements and it is also easy to check them. Before building a correction I always make a quality check.

Regarding your specific questions

1. It is REW that will generate the test signal for impulse response acquisition  (a sweep signal actually)

2. For a 2.1 system you have to measure left channel + sub and then right channel + sub for each location

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  • 6 months later...

Hi and thanks for your interest ! 

 

Here are my answers:

 

Having a premium mic calibration is not mandatory as I am seeking minimal correction to protect the transparency & natural character of loudspeakers: this is why having a perfect absolute values for the measurements is not a must.

 

Convolution is a pure mathematical operation and it will give the same results with all players, what can make the difference is what happens before & after this operation. The most common way of performing convolution is using FFT so there should not be CPU usage difference b/w the different options. If you want to check the CPU usage I can provide you test filters from other customer.

 

Having a dipole is usually an advantage for crosstalk reduction, otherwise the correction is mostly dependent on the room environment: it is logical to get bigger improvements with difficult cases.

 

And yes you get both versions of the filters for the free trial.

 

Best,

Thierry

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  • 2 weeks later...

Hello Peter,

 

You can certainly go for an extra DAC to manage your subs : with that option you can handle everything within Roon with one set of filters per DAC.

Regarding my service you have 2 options: you can define by yourself the crossover + time alignment filters and perform full range measurements with that configuration. I will use those measurements for the correction and merge the corresponding filters with the crossovers ones so that you get only one filters set per DAC.

The other option is to let me handling as well the crossover & time alignment definition. The rest of the process will be the same.

Regarding your last question I generate filters in wav format with double precision.

Hope this is clear enough,

 

Thierry

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Hello John

 

For the sub integration you have different alternatives : you can either make an arbritary choice for the crossover frequency (like 80 hz) or you can perform room transfer measurements of each speaker alone (sub, main left, main right) to better optimize the crossover strategy (including potential overlapping as having multiple sources in the bass region can be beneficial)

Once you have a crossover defined you need to perform the different full range room measurements.

 

Cheers

Thierry

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Hello,

 

@R1200CL

- If you want to go through Roon for the measurements you must use REW as I have not included this option in my tool

- I recommend keeping your usual Roon settings if you go through it for the measurements : if you hesitate b/w two different settings we can check if they have an impact or not on the acoustic measurement

- There are other measurement mics than UMIK and they usually come with only one calibration file, this is why I mention the 2 options for mic orientation : if you have UMIK it is easier to manage measurements by pointing the mic to the ceiling

- You can still use pink noise to set the volume, but I realized that it brought some confusion for REW newbies : using the normal listening level is a good & less complicated option

 

@Celts88

I can guide you if you'd like to fine tune your sub integration : you just need to perform acoustic measurements of each element separately at listening position and send me the files.  

 

Cheers,

Thierry

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Hi John,

 

You need to set the overall volume (either with pink noise or just using your standard listening level) and then make only one sweep measurement per loudspeaker (main left w/o sub, main right w/o sub, sub) putting the mic at the listening position.

 

Cheers,

Thierry

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