Too bad - Allocater has a similar delay. I use my speakers to mix recordings, and having a delay on this order is a pain. It makes it hard to hear small adjustments. A delay on the order of about 10mS is considered inaudible. If UE could attain near that, it would have a major advantage over other software.
Ahhh...
The only solution I've come up so far wrt electrical noise is either separate earth ground paths between computer and power amp, or using audio isolation transformers. There seem to be good TXs out there, the are just expensive.
A very difficult problem. I never see any digital system without noise. A motherboard is very complex, low cost and it is not the priority of the designers to suppress all the noises. If the motherboard works well without bugs, it is a victory. Electronic noise don't only propagate by the wires but also by electromagnetic radiation.
I think the better way is isolate digital to analog using an external USB soundcard with a separate power supply. But the external soundcard should be well done, must separate digital and analog. It is an audiophile or a professional soundcard with a high price
I don't know what to choose
It's my experience the dominant factor in whether or not an external audio interface is better than an internal soundcard is the quality of the implementation. In principle an external interface with its own supply can be more forgiving of PC issues. But many such interfaces use two wire wall warts; those usually leave the secondary side of the transformer floating, in which case noise from the PC is still fairly free move the DACs' ground around. Particularly if the PC happens to be a laptop with a similar two wire supply. I've measured up to 240V peak to peak of ground bounce in such situations, though it's easy to "short" out the (small) free space capacitance of the laptop and its peripherals and drastically reduce the float. Connecting interconnects to a power amp is one such example.
So it's a bit of a cr*p shoot as to what you'll wind up with as noise immunity's not speced. The sense I have is most midrange (USD 100ish street price) and higher gear is generally decent, though there are still stinkers (such as Tapco's Link.USB).
So it's a bit of a cr*p shoot as to what you'll wind up with as noise immunity's not speced. The sense I have is most midrange (USD 100ish street price) and higher gear is generally decent, though there are still stinkers (such as Tapco's Link.USB).
Hi John. Yesterday was a busy day, all day.
I, too, was referring to mechanical noise. The only thing I could add to the other comments for electrical noise in the system would be to put the add-on sound card (I assume you're using a Delta 410 or other one in the SE list) into the bottom PCI slot. That might help a bit if you have a video card rather than on-board video, but the newer machine you have likely has on-board video. This assumes that it's part of the problem. I've read in the past that video cards (and now on-board video), having on-board processors can be noisy. These are in the MHz range. I wouldn't be surprised if those processors with fans might place an electrically noisy fan motor too close to other components.
If you have a video card available to replace the on-board video, it may be possible to disable the on-board through the BIOS. This might place the video farther away from a sound card in the bottom PCI slot.
I did a quick check on the topic. Here's an interesting link at a forum where one guy made a shield to determine the source of noise inside his HTPC. He points to the micro-ATX format being part of the problem. Looks like those tall tower full ATX boards might have an advantage I had not considered.
As you point out, getting it out of the PC environment can only be a good thing.
Forum thread on electromagnetic noise in HTPC
One other thing. The quality of power supply can be important as well. I don't know how you'd find a replacement that had lower EM noise problems, but since I think that these are now mostly switching power supplies, that's probably another source.
Dave
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Being more curious, I found an interesting MIT paper that documents through measurements how the visual display of a monitor, in this case a laptop PC (not an old CRT tube), can be reconstructed from RGB signals in conducted emissions in power leads.
http://ceta.mit.edu/pierc/pierc07/05.09030907.pdf
The emissions inside PCs, the video and the power supplies, is fairly significant. Sound cards don't seem to have any protection against it. Even if they had some, it likely could infiltrate through the bus connections. Anything outside the PC probably is limited to external sources such as 60Hz power conductors (unless you have a big, honking CRT monitor close by).
Dave
http://ceta.mit.edu/pierc/pierc07/05.09030907.pdf
The emissions inside PCs, the video and the power supplies, is fairly significant. Sound cards don't seem to have any protection against it. Even if they had some, it likely could infiltrate through the bus connections. Anything outside the PC probably is limited to external sources such as 60Hz power conductors (unless you have a big, honking CRT monitor close by).
Dave
SE does have two aditional features that let you linearize the phase alone or linearize phase and eq amplitude of any off the shelf speaker.
As far as processing power, the way SE emulates the filters it really doesn'tatter. SE's dsp does not cascade digital filters. It devleopes the requred transfer function for the job an emulates it directly, regardless of the complexity of the transfer function.
Anyway, if you go analog active with dsp phase correction you have to design the analog filters for the specific application. With a dsp approach a general dsp board is required which can then be flashed for what ever application is required. Once the board is designed to handle the dsp algorithms used it's a done deal.
Would you have links to this "allpass, phase correction" discussion? I have active crossovers and I would like to learn more.
Im googling it and there is some links but more links is always better.
Thanks!
I don't have a particular link, I just read in some thread here (that it sounds better). The Soundeasy homepage seems to have links to such information.
There is a PC crossover thread too, which mentions BruteFIR. It's a free Linux tool which can doo all that I believe but I didn't try it.
There is a PC crossover thread too, which mentions BruteFIR. It's a free Linux tool which can doo all that I believe but I didn't try it.
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Greetings Cuibono, earlier in the thread you talked of the possibility of developing an analogue active xo for the violet. Any news on that? Also I have a pair of h frames with alpha 15's in them ala MJK and wondered about the suitability of crossing to the top half of Violet, my concern being the Tang Band is crossed higher than the MJK h frame should be possibly leaving a whole in the response or alternatively running the TB's lower and consequently running out of puff ealier with spl. Anny thoughts would be greatly appreciated
cheers fergs
cheers fergs
Hi Fergs - I considered developing an analog version, but have basically abandoned the idea. The reason being that I started playing with phase linearization, and really liked it. Analog active XOs can't do phase linear, and they are a ton of work to put together - so that is probably the end of that.
I think there should be no problem using the MJK H frame. I would just try and run the H-frame up to the TB - its about 275Hz, IIRC. I don't think you will have any problem with the H-frame resonance. I probably used to do the same thing. Good luck!
I think there should be no problem using the MJK H frame. I would just try and run the H-frame up to the TB - its about 275Hz, IIRC. I don't think you will have any problem with the H-frame resonance. I probably used to do the same thing. Good luck!
Yes, but unless you're running a StigErik style crossover the savings aren't meaningful. So it makes more sense to leave the crossover digital; most of the properties John mentions are general to DSP (or at least a few different types of DSP) and not specific to SoundEasy.
Well, sort of. There's not exactly a shortage of turnkey crossovers like the Ashly XR-x001 series where "design" consists of adjusting some knobs.
See the Arbitrator forum at Thuneau, specifically the OB thread.
I would just point out that there is a big difference between theory and implementation.
Twisting knobs isn't exactly "designing" a filter. There is a rigorous mathematical development behind what is done in SoundEasy.
I glanced at your comments at the thread above. I would pint out that the SE System Phase Linearized is basically similar to the Phase Arbitrator. The SE Advanced System linearizer is some where between the SE UE and the System Linearizer. However there is one thing that the UE can do with really makes the Advanced System Linearizer some what superfluous.
For example, if you were to measure a speaker system, any system, at a design point and in SE save that response as a diver file the response would include both the system amplitude and phase, including the effects of the crossovers and delays (as, for example, used in the Orion which you discuss in the OB thread). Then you can load this "driver" as one of the drivers (the woofer) of a project to be used in the UE. The UE can then be used to linearize the phase of this driver (system). The advantage here is that you do not have to fool around setting up filters to try to emulate the system phase response. The UE will invert the measured phase directly.
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Sure. Got anything specific in mind or are you just sayin'?
Which gets back to our previous remarks about garbage in, garbage out. Since a one sample window alignment error causes a 180 degree phase error at the Nyquist frequency finding the minimum phase window position in SE v16 was quite a hassle even at high sample rates. I, at least, often ended up settling for good enough as I knew enough about where the window ideally needed to be. Does v17 improve this any?
The concepts are general but implimentaion can vary greatly.
First, setting a window for minimum phase is irrelevant. You don't need it. Being off a fraction of an sample period just adds a linear phase component to the phase. What is important about phase is that it be measured relative to some well defined reference point and done so consistently.
But if you want to obtain minimum phase then all you need to is compensate with by specifying a delay, equal to some fraction of a sample interval, to be removed or added to the starting position of the time window. Of course, the capability to do this has always been present in SoundEasy, and has been discussed at length on the SE user's group, so there is no reason to address it in V17.
And once again, I fail to see why you direct this comment toward SoundEasy. It is not a SoundEasy related issue but a sample rate issue, common to all digital processing/acquisition systems.
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