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Dirac Live Room Correction Suite: Initial impressions from free trial


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I tried the "Dirac Live Room Correction Suite".

 

This is commercial software, which sells for €487.50, which is about US$650, which is a bit more than I am willing to spend. I suppose if you think of it as replacing the need to purchase a box with an analogue equalizer, that isn't so unreasonable, but it is far more than I have ever paid for any software. Hence my observations are limited to the free trial, which gives you 2 weeks of uncrippled use.

 

There are essentially two pieces of software in this suite. The first one does room measurements, and the second is a virtual audio interface that applies the corrections.

 

I used a decent but sub-audiophile-grade Samson USB mic that I had purchased for recording podcasts, based on its flat profile between 20 Hz and 20kHz, and a favorable review in the NYT. I figured it was good enough to get an idea for how this worked.

 

The first thing to do is the room measurements. The software leads you through it, and it is pretty much idiot-proof, unlike the REW, which I still have yet to grok.

 

Here is what my room + speakers look like. Try not to laugh:

 

Screen Shot 2013-02-14 at 6.07.26 PM.png

 

The light blue lines are the L and R averages of several measurement positions (individual measurements are the darker blue lines). These measurements are all from slightly different positions on the listening chair. The software guides you through it. Roughly speaking, this gives you some idea of the error of measurement with respect to positioning the mic.

 

Next, the software allows you to provide a correction. You can correct the entire spectrum, or just a part of it (say below 150 Hz). I did both. I made a bunch of filters, including a perfectly flat one, and one that slightly descends as the frequency gets higher.

 

The software corrects both amplitude and phase, which makes it a bit different from a parametric equalizer correction (which some people prefer).

 

Here is an example of one such correction:

 

Screen Shot 2013-01-19 at 12.59.26 PM.png

 

You then can save the filter, and apply it. The filters you design are specific to your favorite audio interface (DAC, SPDIF converter or whatever) and you choose any and all sample frequencies available. It makes one filter for each of these.

 

The second piece of software functions as a virtual audio interface, soft of like Soundflower, except it isn't (thankfully) a kernel extension. So to use the Dirac filter, you select the Dirac audio interface in your playback software, and the filter has the real audio interface programmed into it. So all you have to do is start the Dirac software at login (or whenever you want it) and start playback.

 

One thing I immediately noticed is Audirvana only works with this when in stand-alone playback mode. It also is incompatible with exclusive access and direct mode playback, not surprisingly. I tried it with both Audirvana and iTunes.

 

Here is what the Dirac playback software looks like when running:

 

streamb_361x221.jpg

 

This gives you the ability to switch between up to four separate filters, and to turn the whole thing on and off.

 

I could not hear any difference between on and off until I added in a second filter. Then I could readily hear differences between my low-frequency and full-range filters, and hear differences between on and off for all the filters. I am not sure whether this is user error or a bug, but I suspect it is a bug. At first I dismissed it completely as inaudible, so it is kind of a bad bug for them to have in their software for those demoing it.

 

The difference with the low-frequency filter I made on/off is quite subtle. I guess I don't have an overwhelming bass problem to begin with (although the measurements sure aren't inspiring).

 

With the full-range filter, it definitely sounds different, and arguably better, with it on vs. off. However, I could easily replicate most of the audible difference, using the AU parametric equalizer available for free in Vox, just guessing the frequency and Q-value from inspection of the measurement plot.

 

There is also a discussion of this and other software here: The Well-Tempered Computer

 

It was also interesting to read the comments about the limitations of this approach by the guy who invented it: The Well-Tempered Computer

 

I would say that about 20% to 30% gets corrected. BTW it is still no more than 1/3 of the actual room problems.

 

So, I'm not sure what to do now. I might play around a bit more with REW and try to replicate (or not) my results, and perhaps make some filters for a future version of Audirvana, if that becomes a possibility.

 

This is a nice neat self-contained solution if you can live within the constraints imposed by the software.

 

I think the idea of using the computer to do room corrections for computer audio is an ideal goal, especially for those of us who use the computer as our only audio source component. I'm just not persuaded this is the realization of that promise.

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I'm worried that this might have come off a bit more negative than I intended. I would encourage anyone interested to download the trial and play with it for a couple of weeks. At the very least, you will learn something about how your room/speakers measure, and what a totally flat/corrected response might sound like (whether applied just to the base or the whole of the audible spectrum). You may very well find it to be the perfect solution.

 

One of the side effects is that for the first time I managed to phase my sub correctly.

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Hi Bill, nice review. I have not tried Dirac, but I am using Audiolense and Acourate. I would have to take a closer look at Dirac to see if it is using similar technology.

 

Re: your room – don’t laugh. Hey man, that’s what everyone’s room measures like (more or less). I have measured a lot of speaker/room combos over 30 years. It's a far cry from 10 to 100 kHz +- 0.25 dB which seems the norm for most electronics that we, as audiophiles, obsess over as opposed to the speakers and room where almost all the differences in SQ really are.

 

This is what my speaker/room measures before (red and blue) and after (green and yellow) using Audiolense:

 

stereo fr.jpg

 

A good starting point for a target is go as low as your speakers will allow, with a flat line to 1 kHz and then another straight line from 1 kHz to -6 dB at 20 kHz. Similar to what you have.

 

Re: With the full-range filter, it definitely sounds different, and arguably better, with it on vs. off. However, I could easily replicate most of the audible difference, using the AU parametric equalizer available for free in Vox, just guessing the frequency and Q-value from inspection of the measurement plot.

 

Very interesting. Looking at your frequency chart it shows a +10 dB peak at 60 Hz and -10 dB dip at 160 Hz. That’s about a 20 dB swing. In terms of loudness, frequencies around 50 Hz will sound 4 times as loud as those frequencies around 160 Hz. Albeit those are narrow bands and it’s hard for our ears to discriminate 1/6 octave spacing or less. Still, the bass must sound smoother with correction, no?

 

Also, that peak between 5 to 7 kHz must have got smoothed, so should sound smoother on the top, no?

 

Bob Katz, a Grammy award winning mastering engineer had this to say about Audiolense,

 

“Let me start by saying that the correction that I have gotten is the best sounding room correction I've EVER heard, analog OR digital, in my 43 years of professional listening! Which means it is now one of the best-sounding stereo systems I've ever heard!”

 

Btw, Bob's book on Mastering goes into detail on his audiophile mastering system. I really like his pragmatic approach to the art and science of being an audiophile mastering engineer.

 

Digital Room Correction (in my acoustically treated room) for me is the final step as nowhere else in the signal chain do we need to consider +-10 dB deviations in frequency response (or timing issues in milliseconds for that matter).

 

In addition to Dirac, I encourage folks to try Audiolense (90 seconds of correction for the trial) and Acourate. Acourate’s demo approach is interesting. Downloading a free log sweep program, you measure the room, then send the measurements to Uli, along with 2 songs of your choosing. Uli will work out the correction and convolve the 2 songs and send them back in which you can play like regular songs, but now with room correction. I dare say most people will have the same reaction that Bob Katz did. I know I did.

 

Thanks again Bill for the review.

 

Cheers!

 

Mitch

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Thanks for the informative reply. Audiolense certainly has a more sensible price tag ($218 today). The interface looks a bit more polished too. Am I out of luck if I use a Mac instead of a PC?

 

Tis a bummer man... I have heard of folks borrowing a Windows laptop and using Audiolense to measure and design/generate the correction filters. Then host the generated filters in a convolution engine that runs on the Mac. I can't vouch for the approach as I have not tried it, but I do have a Mac and if I find some time, I will give it a try.

 

Continuing with the informative...

 

Despite speaker manufacturer's claims, folks that undertake the cost and effort to measure their speakers/room, may be surprised to find out that their tweeters rarely make it out to 20 kHz (typically -10 to -20 dB down relative to the reference level). And even fewer measure anywhere close to 30 kHz (typically -30 dB down or greater). Something to consider if interested in ultrasonic playback of hi-res material.

 

Cheers,

 

Mitch

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Despite speaker manufacturer's claims, folks that undertake the cost and effort to measure their speakers/room, may be surprised to find out that their tweeters rarely make it out to 20 kHz (typically -10 to -20 dB down relative to the reference level). And even fewer measure anywhere close to 30 kHz (typically -30 dB down or greater). Something to consider if interested in ultrasonic playback of hi-res material.

 

This is usually more a directivity issue than issue with the tweeter itself. Often you can get the full response exactly at the tweeter axis. But when you go off-axis, then the difference between speakers grow a lot, since most speakers have directivity increasing as function of frequency with more or less strange anomalies around cross-over frequency. You get good idea of the speaker with power response curve measured in echo chamber.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Tis a bummer man... I have heard of folks borrowing a Windows laptop and using Audiolense to measure and design/generate the correction filters. Then host the generated filters in a convolution engine that runs on the Mac. I can't vouch for the approach as I have not tried it, but I do have a Mac and if I find some time, I will give it a try.

 

If the filters are cross-platform and can be used with something that will run on OS X, then this really isn't a problem. In fact, one could install windows on another partition to do the measurements with the same computer, to minimize the differences. Maybe it might even work in VMWare Fusion.

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This is usually more a directivity issue than issue with the tweeter itself. Often you can get the full response exactly at the tweeter axis. But when you go off-axis, then the difference between speakers grow a lot, since most speakers have directivity increasing as function of frequency with more or less strange anomalies around cross-over frequency. You get good idea of the speaker with power response curve measured in echo chamber.

 

Hey Miska, I don't spend much time in echo chambers :-) But I like measuring in-room responses, under real world listening conditions.

 

Here is a little experiment I did rotating three different high frequency compression drivers into my system, and without any other changes, measuring the response at the listening position.

 

three tweeters and horn.jpg

 

right speaker 3 driver comparo.jpg

 

Each measurement was level matched to 85 db SPL with full range PN in REW.

 

The one that has usable response to 30 kHz cost 3X more than the other two drivers.

 

Even under DRC, where each speakers high frequency response is literally identical, each driver's sonic signature sounds quite different. One uses a titanium dome (and measured the most THD), another a phenolic diaphragm, and one a copper ring radiator (and measured the lowest THD, by almost half of the titanium dome).

 

I hear you on the Directivity Index (DI), but the tweeters themselves play a fundamental role in reaching ultrasonic frequencies as measured above. If the tweeter itself can't produce ultrasonics, the DI does not matter.

 

It has been my experience, measuring speakers in many different venues over 30 years, few tweeters have low distortion, real ultrasonic output measured at the listening position.

 

Cheers,

 

Mitch

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Mitch

 

OK, I don't know about those kind of drivers, compression drivers are not my thing. I've been mostly dealing with these types:

 

1) Conventional domes made of different materials (I have to admit hating sound of most aluminum domes and most other domes with high resonance peak at just above 20k)

2) Ribbon tweeters like Fountek Neo-series, things like Products_Fountek Electronics Co.,Ltd

3) AMT-style "ribbon tweeters" like used by Elac, Adam, etc.

4) Ring-radiators like ScanSpeak Illuminator and Revelator

5) Super-tweeters like the Tannoy ones...

 

 

For speakers, I like to see both anechoic response on tweeter axis and power response (power spectrum of total radiated sound to all directions). Plus of course compression, distortion, decay and other similar stuff, but that a different story then.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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...anyway I'm happy to measure flat tweeter response to 24k+ at listening position from my system, without EQ. Bass side needs some EQ, so I have some amount of room EQ at <500 Hz, but nothing else. And moving to a new place helped on the bass area too.

 

Another important topic is time coherence, but I have less to correct there since I chose speakers that are designed to be good on that area too.

 

Stereophile is nice since they publish in-room measurements too.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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...anyway I'm happy to measure flat tweeter response to 24k+ at listening position from my system, without EQ. Bass side needs some EQ, so I have some amount of room EQ at <500 Hz, but nothing else. And moving to a new place helped on the bass area too.

 

Another important topic is time coherence, but I have less to correct there since I chose speakers that are designed to be good on that area too.

 

Stereophile is nice since they publish in-room measurements too.

 

Got a frequency response to share :-)

 

I have measured a few of the drivers you listed, some make it past 20 kHz, but again, not many. For example, the AMT's I have measured don't make it too well past 20 kHz. But, RAAL ribbons have real measureable ultrasonic response.

 

Speaking of ribbons, you have one of the same speakers that I measured here It does not make it very well to 20 kHz though.

 

Agreed on time coherence and there are a number of ways to achieve it.

 

Cheers!

 

Mitch

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Got a frequency response to share :-)

 

Yeah, I'll have to make a graph for the measurements I have or a new measurement, I have not stored the plot, only measurement data (and that only for RoomEq Wizard). My listening room has Dynaudio speakers, optionally supported by super-tweeters.

 

For example, the AMT's I have measured don't make it too well past 20 kHz.

 

Adams I've measured manage to 30 kHz*and then drop like a stone. And IIRC, Elac did a bit better. Although I don't trust my measurement data for 20k+ because I don't have calibration data for the measurement mic past that point.

 

Speaking of ribbons, you have one of the same speakers that I measured here It does not make it very well to 20 kHz though.

 

My speakers have the tweeter mounted behind the front baffle and have different guide, because the original one behaves badly. I have those speakers now in the living room attached to a tube amp (on 70cm high sand-filled Target speaker stands). It just has to be measured exactly at vertical tweeter level due to DI and d'Appolito configuration. Measurements don't differ too much from the originals:

13020025s.png

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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  • 3 weeks later...

Hi!

 

I also downloaded the free trial version of Dirac and tested it over the weekend.

First I have a question to anyone who also tried. Dirac has an audio processor that acts as a soundcard. It also shows up in Jriver but only works under certain circumstances.

1. Asio out of Jriver does not work. Dirac does not show up as soundcard, only my RME does.

2. Asio out of Dirac into my RME soundcard does not work either.

3. Selecting Kernel Streaming (or WASAPI) in Jriver works only if the output is set to 24bit. Dirac out needs to be set to Kernel Streaming.

 

This actually means that there are two DSPs in place, one within Jriver that converts 16/44.1 to 24/44.1 and the DSP in Dirac that applied the filters. Also, Dirac crashes Jriver quite often, especially when changing the output mode in Jriver.

 

There is another caveat; when you run the music through the Dirac processor it significantly decreases the volume of the output signal. Also, but this is expected, the dynamics are somewhat decreased.

 

So much to the negative side. On the positive: The software is very easy to use, extremely intuitive but still gives the user enough flexibility. I did not do a correction on the entire spectrum but only in the 20-400 Hz range. The result was promising, some of the boom at 50-60 Hz and 100-200 Hz was gone and the sound did get much cleaner. Really nice!

 

With the shortcomings of the software today, I am not sure it is worth the 480 EUR FOR ME. If someone is interested I can post before/after response curves measured with REW. At least I know now that my room does have some trouble and that I need to do something about it before investing in any further hardware. :-)

 

Suggestions are very much appreciated! Maybe some others have tried Dirac and could post their experiences? Maybe there is room for improvement or I made some user mistakes?

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Hi!

 

 

 

There is another caveat; when you run the music through the Dirac processor it significantly decreases the volume of the output signal. Also, but this is expected, the dynamics are somewhat decreased.

 

Hi,

this is correct and shows a good understanding of the mechanisms of equalization/room correction. In fact, a well-made filter should be aware that there are situations in which it makes, for certain frequencies, amplifications also the order of 10 db.

So in order to avoid clipping of the amplification, the filter must reduce the overall volume.

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Good point! I always thought amplification in that order of magnitude should be avoided. But if applied, you are right, the overall volume needs to be decreased.

 

Nevertheless, it needs to be considered when using Dirac for room correction and comparing it to without room correction. While Dirac does have an "on" and "off" button it is still in the signal path. The fair comparison is: Jriver Asio -> soundard vs Jriver Kernel Streaming -> Dirac with filters on -> soundcard.

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  • 2 weeks later...

I'm finally ready to give Dirac live a whirl. I got Dirac live up and running on my PC. I am using Scarlett 2i2 mic USB preamp. Everything appears to work fine and the preamp is recognized as input device.

 

However, when I play a test tone, the preamp respond (i.e. mic pics up signal), but the meters in Dirac show nothing. So the signal does not appear to make it from the preamp into the PC over USB.

 

Do anyone have any idea what I could be going wrong? The Dirac manual troubleshooting guide does not offer a solution that works.

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  • 2 months later...

I finally got around to listening to Dirac Live (I did the calibration a few weeks ago) and here are my observations. I created four target curves:

 

1. +5db at 20hz sloping down to flat at 300Hz flat through 2000hz then sloping down to -5dB at 18,000 hz - This is very similar to my target curve in Trinnov

2. +5db at 20hz sloping down to flat at 300Hz - no correction beyond 300 Hz

3. The Dirac house curve (standard generated)

4. Flat curve

 

First of all, the difference betwen the four curves and no DRC were quite dramatic. Curve (2) and (4) were immediately eliminated. Somehow (2) muffled the sound, and (4) just did not sound right. Betweeen (1) and (3) was a close call but I preferd (1).

 

The interesting comparison was between no DRC and curve (1). Preferences appeared to vary depending on reportaire, but most importantly I believe recording quality. There was a very distinct difference in imaging, imaging of the vocals (for example, the voice of JJ Cale on his "Live" album changes completely!), and mid bass. Surprisingly, not so much on low bass, which is where most problems happen according to the measurement. In the end I ended up preffering curve (1) over no DRC. it appears this setting was more unforgiving but I believe it got me closer to what is actually on the recording.

 

So with that settled I kicked back and did some listening with Dirac engaged. It then occured to me that the Dirac in bypass mode does not sound as good as not loading Dirac at all (which was what I previously did). So now I need to go back to the drawing board to compare Dirac with filter (1) with taking Dirac completely out (instead of just disengaging the filter in Dirac).

 

Interesting stuff. Working from memory, I believe the difference with engaging and bypassing Dirac were more pronounced than doing the same in Trinnov, which appeared to be more subtle.

 

When I have my lynx card in, I will repeat the whole experiment in multi channel.

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  • 2 months later...

I have been using Dirac for some months now, and it really is amazing. The sound now just stands EXACTLY in between the speakers, and several meters in front of them, compared to Dirac in standby/off.

 

Best Regards KK ;)

 

I tried the "Dirac Live Room Correction Suite".

 

This is commercial software, which sells for €487.50, which is about US$650, which is a bit more than I am willing to spend. I suppose if you think of it as replacing the need to purchase a box with an analogue equalizer, that isn't so unreasonable, but it is far more than I have ever paid for any software. Hence my observations are limited to the free trial, which gives you 2 weeks of uncrippled use.

 

There are essentially two pieces of software in this suite. The first one does room measurements, and the second is a virtual audio interface that applies the corrections.

 

I used a decent but sub-audiophile-grade Samson USB mic that I had purchased for recording podcasts, based on its flat profile between 20 Hz and 20kHz, and a favorable review in the NYT. I figured it was good enough to get an idea for how this worked.

 

The first thing to do is the room measurements. The software leads you through it, and it is pretty much idiot-proof, unlike the REW, which I still have yet to grok.

 

Here is what my room + speakers look like. Try not to laugh:

 

[ATTACH=CONFIG]4335[/ATTACH]

 

The light blue lines are the L and R averages of several measurement positions (individual measurements are the darker blue lines). These measurements are all from slightly different positions on the listening chair. The software guides you through it. Roughly speaking, this gives you some idea of the error of measurement with respect to positioning the mic.

 

Next, the software allows you to provide a correction. You can correct the entire spectrum, or just a part of it (say below 150 Hz). I did both. I made a bunch of filters, including a perfectly flat one, and one that slightly descends as the frequency gets higher.

 

The software corrects both amplitude and phase, which makes it a bit different from a parametric equalizer correction (which some people prefer).

 

Here is an example of one such correction:

 

[ATTACH=CONFIG]4336[/ATTACH]

 

You then can save the filter, and apply it. The filters you design are specific to your favorite audio interface (DAC, SPDIF converter or whatever) and you choose any and all sample frequencies available. It makes one filter for each of these.

 

The second piece of software functions as a virtual audio interface, soft of like Soundflower, except it isn't (thankfully) a kernel extension. So to use the Dirac filter, you select the Dirac audio interface in your playback software, and the filter has the real audio interface programmed into it. So all you have to do is start the Dirac software at login (or whenever you want it) and start playback.

 

One thing I immediately noticed is Audirvana only works with this when in stand-alone playback mode. It also is incompatible with exclusive access and direct mode playback, not surprisingly. I tried it with both Audirvana and iTunes.

 

Here is what the Dirac playback software looks like when running:

 

[ATTACH=CONFIG]4337[/ATTACH]

 

This gives you the ability to switch between up to four separate filters, and to turn the whole thing on and off.

 

I could not hear any difference between on and off until I added in a second filter. Then I could readily hear differences between my low-frequency and full-range filters, and hear differences between on and off for all the filters. I am not sure whether this is user error or a bug, but I suspect it is a bug. At first I dismissed it completely as inaudible, so it is kind of a bad bug for them to have in their software for those demoing it.

 

The difference with the low-frequency filter I made on/off is quite subtle. I guess I don't have an overwhelming bass problem to begin with (although the measurements sure aren't inspiring).

 

With the full-range filter, it definitely sounds different, and arguably better, with it on vs. off. However, I could easily replicate most of the audible difference, using the AU parametric equalizer available for free in Vox, just guessing the frequency and Q-value from inspection of the measurement plot.

 

There is also a discussion of this and other software here: The Well-Tempered Computer

 

It was also interesting to read the comments about the limitations of this approach by the guy who invented it: The Well-Tempered Computer

 

 

 

So, I'm not sure what to do now. I might play around a bit more with REW and try to replicate (or not) my results, and perhaps make some filters for a future version of Audirvana, if that becomes a possibility.

 

This is a nice neat self-contained solution if you can live within the constraints imposed by the software.

 

I think the idea of using the computer to do room corrections for computer audio is an ideal goal, especially for those of us who use the computer as our only audio source component. I'm just not persuaded this is the realization of that promise.

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I have heard of folks borrowing a Windows laptop and using Audiolense to measure and design/generate the correction filters. Then host the generated filters in a convolution engine that runs on the Mac. I can't vouch for the approach as I have not tried it, but I do have a Mac and if I find some time, I will give it a try.

 

According to the Audiolense author, it is very important to use the same audio interface for recording and for playback - especially if you are doing correction in the time domain. So seperate USB mic and DAC is out unless you have them connected via a master clock. At least if you want the best result.

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  • 2 months later...

A new Stereo Version of Dirac Live is now available with the main difference that it supports two-channel playback only. It features full mixed-phase impulse- and frequency response optimization for a listening area of arbitrary size chosen by the user.

 

With the Dirac Live Room Correction suite and a microphone available from Dirac’s online store the user performs a simple multipoint measurement of the listening room (nine measurements instead of one are necessary for optimal room correction)

 

The stereo suite has been tailored for those persons who use their computer as an audio source, and leverages the computer’s built-in processing power. It has been designed with simplicity in mind without compromise in terms of sound quality and/or flexibility for the advanced user. Dirac Live acts as a "virtual soundcard" so that all sound from the computer regardless of whether is a media player, YouTube, video game, Skype conversation, etc., is optimized.

 

A free Trial Version with full functionality is available for download here: Dirac RCS | Dirac Research

 

Ciao, Flavio

Warning: My posts may be biased even if in good faith, I work for Dirac Research :-)

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I have been using Dirac on a daily basis now for quite some time. It´s stunning, its fabtastique! ;=)

 

It´s worth every penny, and its better to spend the money on Dirac, then on a new amp or new speakers

 

Best Regards

KK

 

A new Stereo Version of Dirac Live is now available with the main difference that it supports two-channel playback only. It features full mixed-phase impulse- and frequency response optimization for a listening area of arbitrary size chosen by the user.

 

With the Dirac Live Room Correction suite and a microphone available from Dirac’s online store the user performs a simple multipoint measurement of the listening room (nine measurements instead of one are necessary for optimal room correction)

 

The stereo suite has been tailored for those persons who use their computer as an audio source, and leverages the computer’s built-in processing power. It has been designed with simplicity in mind without compromise in terms of sound quality and/or flexibility for the advanced user. Dirac Live acts as a "virtual soundcard" so that all sound from the computer regardless of whether is a media player, YouTube, video game, Skype conversation, etc., is optimized.

 

A free Trial Version with full functionality is available for download here: Dirac RCS | Dirac Research

 

Ciao, Flavio

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  • 1 year later...

I am interested in using Dirac and visualizing the sound in my room, and then improving it if I can. I am a complete noob with this, however. Does anyone have any advice and/or link to an appropriate microphone to purchase?

Third Floor: AE>Pioneer solid state integrated>Sony PS-x70 turntable>KEF 103.2 speakers

Second Floor: Intel NUC>LampizatOr GA TRP/LampizatOr Integrated Solid State amp>triode wire labs speaker cables & power cord and wywires power cords>vapor über auroras speakers

Old school: VPI Prime Signature turntable w/ Ortofon Bronze Cadenza cartridge and Technics SP-10 mk2

First Floor: AE>lifatec silflex glass toslink>schiit bifrost über>Kimber kable hero RCA>audioengine 5

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