I Like AT As Well!
Particularly for MF check out,
AudioTechnology, C-Quenze Drivers
C-Quenze - our newest line of Loudspeaker Drivers.
Regards
WHG
Particularly for MF check out,
AudioTechnology, C-Quenze Drivers
C-Quenze - our newest line of Loudspeaker Drivers.
Regards
WHG
No!
Certainly not. Radiating width of the Raven is 15 mm, the ATC dome has an effective width (dia.) of 80 mm; so, in the horizontal plane the Raven wins the polar response battle. Intrinsically the Ribbon technology is far superior for reproducing the upper 1.1 decades of program signal.
| Mfg ...| Mmd ……....| Sens .| F(hp) ….| F(lp) …………...………|
| ATC ..| >> 0.04 gr | 94 dB | 500 Hz | 3,000 Hz ………..……|
ATC SM75-150S Dome Mid Measurement Data - AudioSmile Forum
|Raven| =0.04 gr …..| 98 dB | 800 Hz | None (>20,000Hz)|
http://www.orcadesign.com/images/Pdfs/RavenR3x.pdf
n.b.: Implicit in the above comparison is that the first two decades to be covered are 1-100 Hz and 100 - 1000 Hz approximately.
This strategy accommodates flexibility in the placement of subwoofer(s) as there is no perceivable directionality of signal origin in the first decade. Such placement may then be focused primarily on mitigating LF room modes.
Regards,
WHG
Certainly not. Radiating width of the Raven is 15 mm, the ATC dome has an effective width (dia.) of 80 mm; so, in the horizontal plane the Raven wins the polar response battle. Intrinsically the Ribbon technology is far superior for reproducing the upper 1.1 decades of program signal.
| Mfg ...| Mmd ……....| Sens .| F(hp) ….| F(lp) …………...………|
| ATC ..| >> 0.04 gr | 94 dB | 500 Hz | 3,000 Hz ………..……|
ATC SM75-150S Dome Mid Measurement Data - AudioSmile Forum
|Raven| =0.04 gr …..| 98 dB | 800 Hz | None (>20,000Hz)|
http://www.orcadesign.com/images/Pdfs/RavenR3x.pdf
n.b.: Implicit in the above comparison is that the first two decades to be covered are 1-100 Hz and 100 - 1000 Hz approximately.
This strategy accommodates flexibility in the placement of subwoofer(s) as there is no perceivable directionality of signal origin in the first decade. Such placement may then be focused primarily on mitigating LF room modes.
Regards,
WHG
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If I am reading description of DEQX correctly, its volume control is placed in analog domain after DA conversion? So if I want to use my own external multichannel DAC for DA conversion fed from digital output board on DEQX, I have to come up with my own volume control, correct?
Yes, you will need an analog volume control after the D/A converter. It is possible to digitally attenuate the output but that could degrade the audio quality.
I have two of these, one master and another unit slaved off of the master. simple, ultra high quality solution:
MSB DACs
It's sounds like you are looking for the ultimate solution and my advice is using your own DACs will not get you there, same with using the DEQX. It will be good, but it's processing as I stated earlier is limited. You seem computer savvy, I would go this route, and I could give you detailed, foolproof setup instructions. I've been meaning to write a PCXO webpage for about 5 years now.
And to Shinobiwan's point about taps, 4096 taps are fine for FIR, but IIR requires a lot more. In the case of Audiolense, using the high quality IIR filters it uses 65,536 taps. I still say it's botique DCX2496. I still don't see what Acourate offers that Audiolense does not, except a clunkier interface. The Acourate GUI reminds me of Sound Easy, and Audiolense is like ARTA.
I don't think it's very hard to setup a PC based XO. As far as delay is concerned there is a way around it with the right convolution plugin. Voxengo Pristine Space has a Zero Latency function. It doubles the needed processing power, in my case from ~4.5% to ~9%, and works very well.
That looks like a very reasonable price for an 8 channel DAC/Pre. Good find.
I notice that you mention the pc as the ultimate solution. I had a similar thoughts a few years ago whilst I was in the height of my PCXO exploits. But after a bunch of projects and experiences my outlook has changed somewhat. I don't really think there's an ultimate solution for anything otherwise there'd be a more definite consensus amongst the community. Instead what you have is options, lots of them and the choice of crossover is no different. The best one can hope for is something that works for you. So in the case of SashaV he asks a question and we should be offering advice to help him make an informed decision.
The PC is a valid choice for you and the DEQX works for me but will either of those suit someone else? I've no idea but I'll try hard to offer a well rounded opinion which might aid that person.
Your using IIR filters? The same as the DCX2496? Hmm... 😉
I was talking about FIR filters. I brought the DEQX and the PC for this functionality. For me the selling point of digital crossovers is FIR functionality.
I agree with you regarding Acourate being overly drawn out in its approach but its not something you really use often after the initial setup and once you do get used to the filter generation process you tend to fly through it quickly so I could forgive it for those sins. The important thing for me was that the filters sound good and in that it had no problems. However anyone is concerned about UI and ease of use issues then the DEQX is impressive because its a tailored solution with no hardware or software variables unlike the PC which is a set of general purpose hardware and software that's pressed into service as a crossover.
I tried Voxengo PS and the zero latency function doesn't remove the latency of an FIR, it only reduces the latency of the plugin. Convolver was more successful in this regard because you could use the 'partition' function to sub divide the processing task. You can never remove the filter latency completely with this but a partition of 2 will have half the latency of a single partition and a partition of 4 will have a quarter of the latency of a single partition and so on. However it quickly eats up CPU and I needed 8 channels of processing for a stereo pair so it if your using high partition numbers you need a suitably powerful CPU.
MVC looks very interesting, I wish there was a balanced version, but still it is probably much better than digital volume control.
Out of curiously, what DACs are you using in your solution?
I use both IIR (not 100% IIR but a type of minimum phase filter) and FIR filters. And no it's not like a DCX, LOL, I have a one laying around if someone wants it! Audio quality is terrible on that thing... I'm not sure what issues you ran into with latency but an FIR filter with 131,072 taps, 8 channels (which is what I use), 32bit, 96K sampling rate and zero latency set on Pristine Space will result in about 1.4 ms of delay.
Adding the ASIO delay the system is still sub 10ms, otherwise, if zero latency is not checked, the system will result in 350+ ms of delay. Pristine Space must be reloaded for zero latency to work correctly. A VST hosts might not report the delay correctly internally and that can cause an issue.
As far as the community consensus goes, I don't really subscribe to that. I hardly ever post on forums because the audio community can be pretty annoying, full range speakers, SETs, and EnABLE???
The MVC does not have any DAC capabilities, well unless you buy that option. Although it does use two buffer and a resistor ladder from a DAC to attenuate the signal. So it does have a DAC inside but does not use it conventionally. Everything is kept in the analog domain, hence the 120dB (or 130dB, the site has two specs) SNR. I would contact MSB, I think they can modify it for balanced inputs and outputs.
Sasha I use the Echo Audiofire8 for my DAC, it's my favorite audio adapter to date. Just for fun, if I had money to burn I would look at buying this beast:
Prism ADA-8XR FW-MDSD | VintageKing.com
Rob
I'm just a beginner on FIR but how is it possible to have such a low latency with 131072 taps at 96 kHz ? it's beyond me. Could you develop ?
Best from France
Jean Claude
What's the resolution in the low frequency (say under 100 Hz) with only 4096 taps at 96 kHz sample rate ?
I'm afraid not enough to correct any strong room modes
Best from France
Jean Claude
What's the resolution in the low frequency (say under 100 Hz) with only 4096 taps at 96 kHz sample rate ?
I'm afraid not enough to correct any strong room modes
Best from France
Jean Claude
There's a separate PEQ used for dealing with room related bass issues. The filters are only used for crossover and driver correction NOT room correction.
My DCX question was rhetorical and meant as a light humoured nod towards your 'boutique DCX' inference 🙂
If your delay is that low then your using IIR and not FIR. There's a very definite delay that's defined by sample rate and the number of taps contained within the filter.
There's still no avoiding that what you find to be the ultimate is going to be someone else's idea of hell. Just like you don't really find those full ranges or tubes all that endearing.
Understood, so IIR filters are in use for room correction ?
and sorry for the previous double post
JC
I've wondered about that myself and I'd say yes the PEQ section is more than likely IIR because otherwise there'd be a significant delay involved due to the large number of taps needed to achieve the resolution the room correction filters exhibit. I'm not 100% sure on this but its the most logical conclusion for me.
So my understanding of the DEQX is that the crossover and driver filtering is FIR based and the room correction component, if used, is IIR.
SashaV,
I would like to suggest some thinking in a different direction. Using high quality DSP allows you to use much cheaper drivers than the very expensive drivers proposed so far.
Why would you use a very expensive 75 mm dome, that can hardly be used below 600Hz even if filtered very steeply at the low end and stronly beams above 2.5 kHz.
Used a a midrange, I seriously doubt the DT Q15 will, after proper filtering, sound any better than your average Vifa.
I would like to take Shins statement about different cone materials even further than he does: once properly equalized and filtered with DSP, physically similar drivers will measure and sound identical by and large no matter how exoticly designed. It is the physics that take over then.
I know this type of statement usually causes a lot of stir up with seasoned audiophies, but few of them have experimented with DSP and steep filtering and A/B comparisons between equalized drivers.
I would rather go for something like The Ardor, recently shown in DIYaudio.
DSP, no exotic drivers, but intelligently designed. An almost omni character to approx. 6 kHz is maintained.
Have a look at the Ardor thread and think how you want the desgn to be and what you are aiming at before making heavy investments in boutique drivers.
Good luck,
Eelco
I would like to suggest some thinking in a different direction. Using high quality DSP allows you to use much cheaper drivers than the very expensive drivers proposed so far.
Why would you use a very expensive 75 mm dome, that can hardly be used below 600Hz even if filtered very steeply at the low end and stronly beams above 2.5 kHz.
Used a a midrange, I seriously doubt the DT Q15 will, after proper filtering, sound any better than your average Vifa.
I would like to take Shins statement about different cone materials even further than he does: once properly equalized and filtered with DSP, physically similar drivers will measure and sound identical by and large no matter how exoticly designed. It is the physics that take over then.
I know this type of statement usually causes a lot of stir up with seasoned audiophies, but few of them have experimented with DSP and steep filtering and A/B comparisons between equalized drivers.
I would rather go for something like The Ardor, recently shown in DIYaudio.
DSP, no exotic drivers, but intelligently designed. An almost omni character to approx. 6 kHz is maintained.
Have a look at the Ardor thread and think how you want the desgn to be and what you are aiming at before making heavy investments in boutique drivers.
Good luck,
Eelco
Not true, I'm using FIR and here's a couple of pictures to demonstrate the delay using and impulse test signal. The first test is conventional FIR processing and the second is using the zero latency function. The file used for testing is 32bit/96kHz, 131,072 taps. If you look at the scale at the top of the scope you can see the delay between the impulses, the first pulse (red) is from the impulse generator, the second (white) is through Pristine Space. This test is on my new computer, and as you can see the results are even better than 1.4ms:
Zero Latency Off
An externally hosted image should be here but it was not working when we last tested it.
Zero Latency On
An externally hosted image should be here but it was not working when we last tested it.
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My apologies, if that's the total latency from the playback software to the processed outputs of Voxengo PS and not the just the delay caused by the sample buffer then it appears Voxengo PS works similarly to Convolver with its partition function.
Besides the test figures have tried a tasks that requires near realtime playback such as gaming or a movie for example and if so is there no appreciable lag of audio behind the video when using FIR filters(not IIR)? A good test for this is checking for lip sync.
Besides the test figures have tried a tasks that requires near realtime playback such as gaming or a movie for example and if so is there no appreciable lag of audio behind the video when using FIR filters(not IIR)? A good test for this is checking for lip sync.
I don't have any issues with audio and video sync, but if I need to free up resources I can do that by switching filters, which I never need to do on my new computer. Just for you Shin!!!.... A convolverVST analysis using the same filter:
No Partitions:
An externally hosted image should be here but it was not working when we last tested it.
16 Partitions:
An externally hosted image should be here but it was not working when we last tested it.
And as you can see convolverVST is still not up to what Voxengo Pristine Space can do. The best convolverVST can achieve is almost 100ms of delay.
Rob
Why not just increase the number of partitions? 16 is very low. I used to use 256 but that was a few years ago with a 3.2Ghz P4. I now use an i7 4.2Ghz which would likely handle around double that. Based on the 150ms your system is showing a 256 partition would be 16 times quicker or around 10ms and a 512 partition would be under 5ms.
Also I read this on Voxengo's website regarding pristine space:
The "Zero Latency" setting enables a true zero-latency processing mode. Please note that this mode has its own limitations. It will work only with audiocard block sizes (latencies) which are a power of 2, in between 32 and 16384 samples. For example, if the current audiocard block size is 5512 samples, Pristine Space will be silent given that the "Zero Latency" mode is enabled. Another limitation this mode imposes is the stability of the CPU load: you may experience CPU spikes and overloads, especially if you are using more than two instances of Pristine Space. This mode can be useful for tracking sessions, when you don't need many plug-in instances, but where zero latency operation is useful.
Its pretty clear to me that Voxengo is using the same partition approach as Convolver because of the need to keep the audio card latency as a power of 2.
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Even though its been around 4-5years since I used Voxengo I still have most of my old PCXO software. I checked pristine space within Console and the version I have doesn't include the zero-latency functionality... Guess that's why I never used it back then. It does have a 'set latency' within the option menu which explains the confusion.
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