Dirac Live Room Correction Suite Cracked 12
Austin Vermont
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 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.
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.
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.
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.
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.
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.
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.
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.
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).
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.
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.
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.
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.
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.
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.
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.
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.
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:
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. :-)
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.
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.
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.
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.
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