Channel D Pure Vinyl

[Bessie Murrillo]

Iwanted to start a discussion on using equalisation in Pure Music to smooth out in room frequency response. My understanding is that one can add an equaliser plug in to Pure Music. Is anybody doing this, and if so, with what EQ?

I would probably use test tones and USB mic and measurement software to track the results-this is not too hard to figure out, but my concern is that the EQ software would have to be very, very good not to add problems (artifacts) to the signal.

Considering that any EQ applied in the digital domain would have to use filters, and filters, generally add artifacts of their own-I would think that the EQ would have to be very carefully designed to avoid problems. I do not think level adjustments of a handful of dB would be a problem (considering PM is running at 64 bits, I would hope any EQ would be as well).

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FWIW, I have used Channel D's Pure Vinyl to digitise my vinyl without going through a regular phono preamp, I use a flat preamp, hence, I get no RIAA EQ on these digitised files. I have then played these files with both Pure Vinyl and Pure Music (they use the same audio engine) with RIAA EQ applied in the software.

The results are stunning and would suggest to me that applying EQ with Channel D's EQ filters will cause no problems for a "handful of dB" of adjustment. After all, RIAA correction requires up to around 20 dB of adjustment.

However, I'm not sure about smoothing out room frequency response using EQ based on my experience. I have attempted on a few occasions to use test tones, mics and Audyssey software on my Denon and NAD AV receivers to compensate for room acoustics and the results were horrible. I prefer to place the speakers carefully and make use of furnishings to change the room acoustics.

Digital: iMac > Pure Vinyl/Pure Music > RME Babyface[br]Analog: NAD T763 > NAD 218 THX > Bowers & Wilkins 605 Series II[br]Also: Rega Planar 3/Dynavector 10X5/Origin Live OL1/Origin Live DC Ultra motor/SRM Tech Silent Base and Rega Enhancement kit/ > Trichord Research Dino MkII[br]

I'll be following this thread hoping to find some good info. I've tried the built-in equalizer in PM with mixed results and really haven't gotten more serious about experimenting because I'm not sure where to begin ...

The one I use corrects for amplitude and phase at one spot, or for the whole room just for amplitude. It is selectable which correction is used. I do not know if you can perform a whole room amplitude correction very easily with a plug-in. It requires multiple measurements to get an average correction. And the phase correction for a sweet spot may not be possible with tones and a mic.

I had a nasty bump around 50 Hz, and the equalizers helped smooth that out. I did it all by ear, so the results were not exact. But the end result was definitely superior. As these equalizers are free -- I believe they come with GarageBand -- it's worth giving it a shot.

First I set the desired frequency curve, for example flat. Then adjust the maximum difference I want it to allow between two adjacent filters (it is a 31-band 1/3 octave digital graphic EQ). I then run it's Auto-EQ process which does each channel independently. After I am satisfied with those results I run both channels at the same time to make sure there are no dips/peaks and save that desired curve to it's memory. It's also got a Bypass ability built-in for quick comparisons of the EQ results vs. no EQ.

It works fairly well though I'm hoping once I get some proper acoustical room treatments in and run it again that it will be the "icing on the cake" so to speak. It'd be great to have this implemented on the software side and get rid of another box in the signal chain.

is to do this manually. First get some decent analyizer software, along with a good mic and simple usb mic pre (perhaps Centrance). I suspect the measuring will be the crux of doing it manually, but the approach I would follow is to only try and adjust the gross problem areas, rather than every little wrinkle in the response. My understanding is that adjustments below about 300 Hz. will not be subject to much change as to position (ie reflections and speaker dispersion), so I would attempt to tackle the peaks here first, noting that midbass and lower midrange suckouts cannot be boosted very much without adding very powerful amplifiers and additional drivers.

I did forget to mention that the Behringer unit allows a "Max Span" where it will only adjust those frequencies that are above a certain range compared to the target curve. It also has the ability to specify frequency ranges you do not want touched.

So for example, say you wanted to iron out any peaks from 300Hz through 16kHz that are more than 3dB higher than your reference. So if you wanted everything flat @ 80dB and at 2khz it measured 85dB, the EQ will bring it down to 83dB or below if you allow it to adjust it down more than 3dB.

Barrows I am doing exactly this- addressing some low frequency room issues with an EQ plugin within Puremusic. I measured my room (ECM8000 mic and REW software) and found quite severe modes in the region of 30Hz-250Hz. I used REW to auto-generate a set of 12 filters in that regoin that flatten the response to around +/-3dB. I have these filters set up in a very good EQ plugin called FabFilter Pro-Q. I address everything above 250Hz with room treatment as 2" thick panels will absorb these frequencies effectively. My avatar is a screen grab of some of the filters I use around the 100Hz range.

The difference in the bass region is night and day, I simply cannot live without it (am therefore stuck with PureMusic or Fidelia which are the only 2 programs that support AU/VST currently). I hear no ill-effects from the Pro-Q in the entire frequency spectrum, to my ears it is 100% transparent. When I set the ProQ filter completely flat my weiss dac still passes the bit transparency test, this tells me it is keeping the non-EQed section in tact.

Put Pure Music into playthrough mode by selecting Audio Playthrough from the Music Server menu. I always turn off upsampling and memory play first. A note of caution here: until you are used to how your hardware behaves, I would recommend always turning off all amplification whenever doing this with a microphone connected, as at least some audio interfaces (Presonus Firebox, for example) will route inputs straight through to the outputs at this time, resulting in a horrendous and potentially speaker-damaging feedback loop.

Ideally, these measurements are done outdoors, with the speaker elevated off the ground and as far away from other reflecting surfaces as possible. If measurements must be done indoors, place the speaker as far as possible from walls and other reflecting surfaces and angled relative to the walls. Raise the speaker so the tweeter is halfway between the ceiling and floor, and place the microphone level with the tweeter and a meter away.

The lower curves in Figure 4 show the responses of the woofer and tweeter in the example system with the crossover frequency set back to the target 3 kHz. The upper curve shows the result when both drivers are measured together.

With the lower crossover point set to 1 kHz, I ran a sweep of the subwoofer. Equalizing the subwoofer meant using three notches to mitigate the effect of room modes, and a low-frequency shelf to boost the lows a little for this sealed sub. Figure 5 shows the before and after plots of the sub.

With the mains in the intended listening position, it is now a matter of integrating the subwoofer and the mains to obtain the smoothest response at and around the crossover frequency. At these frequencies, a simple crossover often does not work and some experimentation will be needed, as well as the use of the delay adjustment. Since the woofer on this speaker rolls off naturally at around 90-100 Hz, I used a 12 dB/octave highpass on the woofer (to result in a total 24 dB/octave slope) and a 24 dB/octave lowpass on the subwoofer set to 90 Hz. To get the best response, I used an overlap of 1.15. Figure 6 shows the resulting in-room response.

Running your active crossover in-computer has, as noted in the introduction, some key advantages. For one thing, since you already have the computer there is plenty of processing power on tap without the need to purchase units with dedicated digital signal processors in them. For another, the wordlength and therefore resolution is likely higher than dedicated DSPs, and potentially, so is the flexibility from being able to distribute and update software for a general-purpose computer.

There are some limitations, though. Pure Music is designed for playing from computer files, and playing from other sources is more difficult. If you have a vinyl source, then digitizing your collection with Pure Vinyl would certainly be one approach. It is possible to route other digital sources on the computer through Pure Music using the PAD (as described above for measurement), and I use this for things like streaming from GrooveShark.

The crossover is also limited in some features compared to dedicated digital crossovers. For example, the number of different types of crossover filter is limited to four, whereas off-the-shelf digital crossovers offer a dozen or more. Another limitation is that there is no way to save multiple configurations, to allow rapid switching for auditioning changes, or to have different settings for e.g. normal listening and late-night listening.

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