That is why I created a filter which does not touch anything over 300hz. Ironically, that was precisely the filter that killed the high frequencies. Results are very erratic and unpredictable, but when all the stars align you can definitely better sound, improving the entire frequency range.
Dirac Live
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That is why I created a filter which does not touch anything over 300hz. Ironically, that was precisely the filter that killed the high frequencies. Results are very erratic and unpredictable, but when all the stars align you can definitely better sound, improving the entire frequency range.
Thanks Edorr. I really would appreciate your thoughts comparing DIRAC live to Audiolense whenever you are ready to do so.
I believe audiolense involves measuring in REW, which will then define filters to be exported to Jriver. Rather eloborate, and not sure I'll have time to do this anytime soon. If Nyal reports stellar (better than Dirac) results with Audiolense, I might give it a whirl. Interesting to see what the relative improvement (if any) after I take possesion of three 96" RPG variscreens.
I read that the latest version of audiolense says:
"Implemented choice between using target as eq or not in partial correction."
I guess this is what you did with Dirac (partial correction) but didn't like the sound as much as the target eq?
When I previously used audiolense, I believe I was forced to use the target eq. I know Bob Katz had a similar experience to mine when he first started with audiolense. I am not sure whether he's overcome that problem or is just using partial eq now.
"Implemented choice between using target as eq or not in partial correction."
I guess this is what you did with Dirac (partial correction) but didn't like the sound as much as the target eq?
When I previously used audiolense, I believe I was forced to use the target eq. I know Bob Katz had a similar experience to mine when he first started with audiolense. I am not sure whether he's overcome that problem or is just using partial eq now.
I don't think I used "partial correction". I just changed the shape of the target curve and liked it better than the Dirac "house curve". My curve actually applied more correction than the Dirac curve. The "partial correction" I did was my curve 2, which applied correction just on the 20-300Hz range (like Meridian). This I did not like.
Audiolense takes its own measurements. However, if you want to see the improvements of Audiolense filters then it is easiest to load the filter into JRiver and use REW with JRiver's loopback to send the REW sweep through JRiver.
Edit: I forgot to mention that I used Audiolense's new partial correction for a recent speaker GTG in Iowa. It worked great. Mark Seaton was there with his Catalysts 12C speakers and we used the correction with his speakers and dual JTR Speakers Captivator S2 subwoofers.
I have little patience for companies that cannot get their website to work though..... Does not instill a lot of confidence, and is very frustration.
I just created some filters for Dirac Live on my Windows 8 machine, with the MSB Signature DAC, also running the new MSB Windows 8 drivers.
On native 96/24 files (or real time converted DSD at 96/24) everything works fine, but when switching to a 44/16 file, I am getting highly distorted unlistenable sound though Dirac (both with Filter On and Filter Off). If I shut down Dirac and set my MSB DAC as output device and 44/16 worked fine again.
it appears Dirac is having problem with the 16 bit files. However, I am playing Jriver in WASAPI mode (with autoselect bit depth), and the DAC is showing 44/24, when I play a 44/16 file, so JRiver does convert to 44/24. If I select any other mode than WASAPI, the 44/16 won't play at all, or Dirac does not show up as output device (in Asio mode).
I had no problems with Dirac on my Windows 7 machine and the Windows 7 version of the MSB drivers.
I am completely at a loss about what can be going wrong. Any suggestions?
On native 96/24 files (or real time converted DSD at 96/24) everything works fine, but when switching to a 44/16 file, I am getting highly distorted unlistenable sound though Dirac (both with Filter On and Filter Off). If I shut down Dirac and set my MSB DAC as output device and 44/16 worked fine again.
it appears Dirac is having problem with the 16 bit files. However, I am playing Jriver in WASAPI mode (with autoselect bit depth), and the DAC is showing 44/24, when I play a 44/16 file, so JRiver does convert to 44/24. If I select any other mode than WASAPI, the 44/16 won't play at all, or Dirac does not show up as output device (in Asio mode).
I had no problems with Dirac on my Windows 7 machine and the Windows 7 version of the MSB drivers.
I am completely at a loss about what can be going wrong. Any suggestions?
Dirac sound
Hi everybody
I have been using Dirac for some months now, I also got the Mic kit from Dirac and am using MacosX.
Everything from the installation to measurements to daily use went smoothly, and without any problems at all.
When listening to the corrected sound, i found it a bit harsh in the mid/hi frequencies. I found out that the FLAT frequency response is a no go in my listening room, so I adjusted the target curve to follow my loudspeakers (the measured) curve, and only correct the dips and peaks (see the orange/red line on the picture below).
That way I am now enjoying a fantastic 3-D sound stage, and every time I hit the "bypass" button, I can´t help to wonder why this software not is in every listening room where there is a computer streaming the music! The low end is now deep and precise, no more standing waves where one bass note is louder then the next. And the mids and the highs are just "right", and now stand in front of the speakers, and not somewhere in between or to one side or the other.
This is by far the biggest improvement in my system, the change of amps, source or cablings not even comes close to this. It´s a revelation (revolution?), and I will not live without Dirac!
Best Regards
KnockKnock
Hi everybody
I have been using Dirac for some months now, I also got the Mic kit from Dirac and am using MacosX.
Everything from the installation to measurements to daily use went smoothly, and without any problems at all.
When listening to the corrected sound, i found it a bit harsh in the mid/hi frequencies. I found out that the FLAT frequency response is a no go in my listening room, so I adjusted the target curve to follow my loudspeakers (the measured) curve, and only correct the dips and peaks (see the orange/red line on the picture below).
That way I am now enjoying a fantastic 3-D sound stage, and every time I hit the "bypass" button, I can´t help to wonder why this software not is in every listening room where there is a computer streaming the music! The low end is now deep and precise, no more standing waves where one bass note is louder then the next. And the mids and the highs are just "right", and now stand in front of the speakers, and not somewhere in between or to one side or the other.
This is by far the biggest improvement in my system, the change of amps, source or cablings not even comes close to this. It´s a revelation (revolution?), and I will not live without Dirac!
Best Regards
KnockKnock
Is the huge dip below 50Hz a result of limitations of your speakers bandwidth or are these room related cancellations?
The dip around 50 Hz is my homemade target curve, the orange curve. The speakers go down to 40 Hz ( the blue / white curve).
The dip around 50 Hz is my homemade target curve, the orange curve. The speakers go down to 40 Hz ( the blue / white curve).
Why are you EQ'ing down all those low frequencies?
I am actually turning UP the bass a little by adding a little more in the 50-80 Hz region, and cutting everything not needed below 45 Hz. Why I do this? Because it sounds best that way to my ears ;=)
(I made one more correction where I add 2 dB more around 50-60 Hz, which suits some music (mostly acoustic) and some NOT (rock and pop with loads of bass). So I can switch between these two settings with a flick of a button)
Best Regards
KK
I see. It appears your approach is to cut frequencies your speakers are not really well suited to reproduce to begin with (taking some load off your speakers), rather than trying to jack them up. Given your constraints (speakers) this may indeed get you the best results. However, this does not mean the frequencies below 45Hz are "no needed". I suspect you would benefit greatly from adding a sub to your system - dependent on what kind of music you listen to.
I've demoed Acourate. I used to own Audiolense. Recently, Nyal and I eventually came up with some excellent LF filters using REW.
Now, I am in day 2 of my DIRAC LIVE 2 channel demo. The thing that seperates DIRAC from the others is the HF transients. This has been the common theme in other full range target based computer DSP solutions. The HF details sound untouched to my ears with DIRAC. Just perfect bass and midbass. The spatial details are superb!
At this point, I feel sort of dumb that I hadn't yet given DIRAC LIVE a demo. I guess I stupidly thought it being limited to 96k meant that it wasn't worth it. I was wrong.
DIRAC LIVE is also super easy to use. It takes a lot of measuring to get it right. In the "sofa" mode, you need 9 measurements. No calls to customer service needed!
Now, I am in day 2 of my DIRAC LIVE 2 channel demo. The thing that seperates DIRAC from the others is the HF transients. This has been the common theme in other full range target based computer DSP solutions. The HF details sound untouched to my ears with DIRAC. Just perfect bass and midbass. The spatial details are superb!
At this point, I feel sort of dumb that I hadn't yet given DIRAC LIVE a demo. I guess I stupidly thought it being limited to 96k meant that it wasn't worth it. I was wrong.
DIRAC LIVE is also super easy to use. It takes a lot of measuring to get it right. In the "sofa" mode, you need 9 measurements. No calls to customer service needed!
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Hello
I'm using my first post to inform those who are demoing Dirac Live of a fractional update (1.0.4) while waiting for the free update that will lift the limit of 96 KHz.
1.0.4 allows to fine tune the DSP gain (headroom) applied to the signal before it is passed from the DAP to the selected output device.
The purpose of the DSP Gain is to adjust the level of the processed signal in order to avoid clipping at or before the DAC or audio card.
I'll be pleased to send a link to the 1.0.4 beta and to the revised manual with relative instructions to those trial users who will send me a PM with their email saying if they are using a Mac or Pc.
Ciaooo, Flavio
I'm using my first post to inform those who are demoing Dirac Live of a fractional update (1.0.4) while waiting for the free update that will lift the limit of 96 KHz.
1.0.4 allows to fine tune the DSP gain (headroom) applied to the signal before it is passed from the DAP to the selected output device.
The purpose of the DSP Gain is to adjust the level of the processed signal in order to avoid clipping at or before the DAC or audio card.
I'll be pleased to send a link to the 1.0.4 beta and to the revised manual with relative instructions to those trial users who will send me a PM with their email saying if they are using a Mac or Pc.
Ciaooo, Flavio
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 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.
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.
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