Dirac Live

DSPeaker is hardware which means the signal path would be high end awesome DAC with awesome output stage--->crappy chip ADC---->crappy chip DAC with junk output stage---->amp.

I think Trinnov and DEQX are the best hardware solutions if that's what you want. I personally wouldn't want that though. I think simplicity is still important.

I think the incremental benefits of simplifying the signal path and adding DRC (and trade-offs between the two) will be extremely system and application dependent. Based on extensive reviews from folks of avsforum, all else being equal, Trinnov is probably the best DRC, definitely in a complex MCH setup, not least because of its 2D and 3D remapping / channel expansion capabilities.

However, you can't run Trinnov on a PC. So for a simple 2 channel (or 4 channel) system running straight from a server into a DAC, it is very conceivable Dirac is the best sounding solution. Kal can probably shed some light shortly, because he has tried Trinnov and is now fiddling with Dirac (Dirac live on PC, going straight into exasound MCH DAC via USB, which would be a SSP giant killer setup).

I have a 4 channel Trinnov processor and am waiting for my server with Dirac to come back (it is not working properly now), so I will shortly have the hardware and software to do a comparison myself, but not the time / patience. I will simply try Dirac for 2 channel and MCH (5.0), if the MCH is as good as the Trinnov (or close), I'll stick with Dirac and sell the Trinnov. With every system change I also need to validate DRC is still beneficial for my 2 channel system. While beneficial (on 2 channel), my experience with using DRC has never been "transform my system", but always incremental improvement (icing on the cake as Kal called it in his review of the Meridian 861v6). I made a lot of system changes, so I'll have to do this in this evaluation cycle as well.
 
Oh okay. So DSPEAKER chip DAC is awesomer than my DAC. GOTCHA! Maybe it's better than your lampilater too? My point is that it's totally unnecessary to use hardware, alien technology notwithstanding.

LoL

Ahhh, no, DSPeaker can work exclusively in the digital domain, in that it accepts digital input, does its DSP/DRC thing and outputs in digital too. Only shortcoming is that Toslink is involved in the in/out stage and if you don't use that, a converter like MSB digital director is needed. 24/96 is the limit like Dirac. One key advantage is that the Anti-mode 2.0 algorithm is VERY advanced and has some kind of artificial intelligence code embedded in it.

Of course DSPeaker can be used as ADC and DAC as you describe as well, and it has a full parametric equalizer plus it has a passable analog preamp. Its very flexible and only costs $1K.
 
Oh okay. So DSPEAKER chip DAC is awesomer than my DAC. GOTCHA! Maybe it's better than your lampilater too? My point is that it's totally unnecessary to use hardware, alien technology notwithstanding.


Don't be silly, it won't be as good a digital section as T-Dac ( which I heard at the Zurich show for over an hour withe the TD Server) or Lampi nor have as good an analog section (should be decent coming from parent company VLSI). However, its the anti-mode SOFTWARE that has garnered the review acclaim. Its that aspect tat I wanted to use as a comparison to Dirac or DeQX or Trinnov or whatever. Anti-mode 2 is the key technology in play there and its ease of use and outstanding results are what I want to compare to the others.

Reviewers (GoldenEar/Stereophile/Ab Sound/ HiFimaailma/AVForums) seem to like it too: http://www.dspeaker.com

Exceprt: AVForums.Com
DSPeaker's versatile room correction unit, Anti-Mode 2.0 Dual Core was awarded a "Reference Status Award" in the review by Russell Williams at the AVForums.com. Read the full article here.
"The new kid on the block is the new king on the block"
- Russell Williams / AVForums


I also heard good things about Dirac and that is why I ask.
 
Too bad said alien technology is locked in that worthless box.
Don't be silly, it won't be as good a digital section as T-Dac ( which I heard at the Zurich show for over an hour withe the TD Server) or Lampi nor have as good an analog section (should be decent coming from parent company VLSI). However, its the anti-mode SOFTWARE that has garnered the review acclaim. Its that aspect tat I wanted to use as a comparison to Dirac or DeQX or Trinnov or whatever. Anti-mode 2 is the key technology in play there and its ease of use and outstanding results are what I want to compare to the others.

Reviewers (GoldenEar/Stereophile/Ab Sound/ HiFimaailma/AVForums) seem to like it too: http://www.dspeaker.com

Exceprt: AVForums.Com
DSPeaker's versatile room correction unit, Anti-Mode 2.0 Dual Core was awarded a "Reference Status Award" in the review by Russell Williams at the AVForums.com. Read the full article here.
"The new kid on the block is the new king on the block"
- Russell Williams / AVForums


I also heard good things about Dirac and that is why I ask.
 
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Let me explain our admittedly biased point of view about differences between Dirac Live and DSpeaker...

DSspeaker is a standalone unit so it can be easily used with a number of different sources while Dirac Live is an application that consequently can be used with a PC or Mac only (Dirac Live may eventualy process one external source only, and only if an audio card is used instead of a DAC)
Needless to say no additional conversions are necessary with Dirac Live.

I understand that DSpeaker is a minimum-phase solution, so that its designers quite correctly concieved and limited it to low frequencies correction.
The reason is that minimum-phase filters can properly correct room behaviour in minimum-phase regions only, which tend to be at the lower frequencies (even at those frequencies that is not always true).

Let me cite an excerpt from the REW's measurement software developer site:
"Minimum phase systems can be inverted, which means that a filter can be designed that, if applied to the system, would produce a flat response and correct the phase response at the same time. That is clearly a nice property to find if we want to apply EQ. If we apply EQ to a system that is not minimum phase, or more particularly in a region where it is not minimum phase, the EQ will not produce the results we would like. It may still be possible to achieve a flat response, but correcting the phase response would elude us. It is simply not possible"
This quotation comes from this well informed document:
http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/minimumphase.html

On the contrary Dirac Live is a mixed-phase solution that can successfully apply a full bandwidth correction to speakers/room behaviour which is mixed-phase.
We explain that as follows:
"Infinitely many different filters can be designed to have the exact same magnitude response. They differ only in their impulse response. Therefore, it is useful to classify filters according to how their impulse responses behave.
Two commonly used filter classes in audio applications are minimum-phase filters and linear-phase filters. They are two special cases that are relatively easy to design, but that come with tightly constrained impulse response characteristics. A minimum-phase filter, by definition, is constrained to apply only the smallest possible delay to the signal given a desired magnitude response. A linear-phase filter, by definition, applies a delay which is constant across the whole frequency range. Therefore, neither of these two filter designs can make a desired change to the phase or impulse response, unless the desired change is exactly the particular change they make by definition. Minimum-phase and linear-phase filters may even worsen both the impulse response and the magnitude response of a system, simply by applying their magnitude response corrections at the wrong time.
A more difficult design task is to make a mixed-phase filter that matches a desired magnitude response while also having a customized impulse response. A properly designed mixed-phase filter can make significant improvements to the impulse response of a sound system at the listening position"

In other words we are talking about two different products of different nature with different design goals :)

Ciao, Flavio
 
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Thanks Flak for the detailed explanation. Very useful perspective. It is now much clearer what the differences are.

DSPeaker can be used over the entire spectrum but they do advise never to go over 500hz and preferably not over 250hz.
 
Norman,
If you are looking for good LF filters and good measurement solution, you should use REW to measure and develop filters. You can then export said filters to whatever program you want. I did this and exported my 14 LF filters to Fabfilter linear phase filters. This works very well and can be very effective. However, it's not even close to what DIRAC LIVE will do; Apples and Oranges, IMO.

Thanks Flak for the detailed explanation. Very useful perspective. It is now much clearer what the differences are.

DSPeaker can be used over the entire spectrum but they do advise never to go over 500hz and preferably not over 250hz.
 
Thanks Mike,

But DSPeaker does that automatically with their algorithm PLUS they do temporal alignment and you can use EQ curves on top…

Dirac seems to be similar, but is more full frequency appropriate. Seems like Dirac is a more economical version of Spatial Computing by Clayton Shaw, without the remote customer service setup feature.
 
Thanks Mike,

Seems like Dirac is a more economical version of Spatial Computing by Clayton Shaw, without the remote customer service setup feature.

Well, it could also be seen the other way.... Spatial Computing digital room correction is Dirac Live with customer setup service, at least this is what I get by reading at the bottom of this page: http://www.spatialcomputer.com/page3/page3.html
So I can only say that it is an excellent choice :)

Good evening,
Flavio
 
Norman,
Another thing to consider when evaluating these DSP solutions is the inherent limitations associated with the omnimic; inaccurate beyond transition frequency at single location. To my knowledge, only DIRAC LIVE and Trinnov deal with this limitation. In the case of DSPEAKER, I wouldn't be getting accurate measurements in my room beyond about 130hz.

Thanks Mike,

But DSPeaker does that automatically with their algorithm PLUS they do temporal alignment and you can use EQ curves on top…

Dirac seems to be similar, but is more full frequency appropriate. Seems like Dirac is a more economical version of Spatial Computing by Clayton Shaw, without the remote customer service setup feature.
 
I think you have to measure from 5 separate locations with Dspeaker and the algorithm figures it out….They recommend correction to 150hz ideally anyway.
 
Thank you guys for the eye opening condo. Very informative and straight to the point.
 
Thanks Art! Art has also been a DIRAC user, albeit sub rosa. :)

I don't think I mentioned this before but I've tried different microphones with their respective calibration files. I started with a cross spectrum emm-6 and now I use an Earthworks m23. I can certainly tell the difference. The earthworks creates better, more refined filters using DIRAC. I can't wait to bring my Tascam USB ADC/DAC and m23 mic over to your place, Art. I think the TASCAM is a little better than using the Dayton USB MIC going in and your DAC going out. The TASCAM will be totally synchronous in and out, using the same clock. You can't really do that with a USB mic. Maybe DIRAC has a way of compensating for that type of setup but I don't think it's ideal, IMO.
 
Thanks Art! Art has also been a DIRAC user, albeit sub rosa. :)

I don't think I mentioned this before but I've tried different microphones with their respective calibration files. I started with a cross spectrum emm-6 and now I use an Earthworks m23. I can certainly tell the difference. The earthworks creates better, more refined filters using DIRAC. I can't wait to bring my Tascam USB ADC/DAC and m23 mic over to your place, Art. I think the TASCAM is a little better than using the Dayton USB MIC going in and your DAC going out. The TASCAM will be totally synchronous in and out, using the same clock. You can't really do that with a USB mic. Maybe DIRAC has a way of compensating for that type of setup but I don't think it's ideal, IMO.

We'll be able to do a good comparison soon enough! Looking froward to it....
 
I think the TASCAM is a little better than using the Dayton USB MIC going in and your DAC going out. The TASCAM will be totally synchronous in and out, using the same clock. You can't really do that with a USB mic. Maybe DIRAC has a way of compensating for that type of setup but I don't think it's ideal, IMO.

That's a keen observation... as far as I know other DRCs require to connect the mic to the same usb port as the output device to make sure they share the same clock source (so a USB mic on one port and a DAC on another USB port is a problem)

But you may have noticed that when you do each stereo measurement with Dirac Live three sweeps are played instead of two as expected...
this way you can have completely different clock sources on the playback and recording device because we compensate for the timing problems.

This is why we play three sweeps when we do a stereo measurement, first left, then right, then left again.
When we do like this we can calculate the clock drift between the playback and recording device and compensate for it.

Ciao, Flavio
 
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Thanks Flavio! I always wondered why DIRAC double measures one side. That answer that question. :) That's a very clever design feature. And, as you said, it's totally unique in comparison to Audiolense or Acourate.

The reason I don't use my TotalDac to measure is because it has a 1024 sample asynchronous buffer in it. I just assumed that DIRAC wouldn't be able to put humpty-dumpty back together again on the other side of my TotalDac. Was I wrong? Should I use my DAC to measure?

Thanks,
Michael.
That's a keen observation... as far as I know other DRCs require to connect the mic to the same usb port as the output device to make sure they share the same clock source (so a USB mic on one port and a DAC on another USB port is a problem)

But you may have noticed that when you do each stereo measurement with Dirac Live three sweeps are played instead of two as expected...
this way you can have completely different clock sources on the playback and recording device because we compensate for the timing problems.

This is why we play three sweeps when we do a stereo measurement, first left, then right, then left again.
When we do like this we can calculate the clock drift between the playback and recording device and compensate for it.

Ciao, Flavio
 

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