WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions.
Click here to have your audio equipment measured for free!
Yup, I already own it . It was obviously written a few years ago and I wondered if Acourate had evolved to become more user friendly, but it seems not. I think AL is going to be a better choice for me.
Mitch has indicated that he would like to write a similar guide for AL, but I suspect that may be some way off, if ever.
I've been too busy working with my builder to select drivers for achieving all of Gnarly's performance objectives. Thus, I've yet to begin serious study of room acoustics and how to correct problems-first acousticallyt, and only then with software filters, which as all of my music sources are pc based, I will be able to can do-assuming I can get past the learning curves.
I was hoping that by now apps like Henrik's Camilla DSP for Windows would be a lot friendlier for dummies, rather than just for code writing geeks.
After a little looksee around StereoNET, it became obvious there wasn't much information about CamillaDSP or its fantastic DSP pipeline capability. So I thought, why not share how I'm currently using CamillaDSP in my setup, and maybe more users will share their experience, implementations, and tw...
Room correction is just a bonus of using something like AL from my point of view. Its the crossover and time / phase correction capabilities that I’m after.
Room correction is just a bonus of using something like AL from my point of view. Its the crossover and time / phase correction capabilities that I’m after.
Just for your possible interest and reference based on your rather easy/simple DSP requirement of XO/EQ/Gain/Time/Phase, a simple and straightforward DSP software "EKIO" may well fit for yourself. You would please refer to my post #112 on this thread.
After a little looksee around StereoNET, it became obvious there wasn't much information about CamillaDSP or its fantastic DSP pipeline capability. So I thought, why not share how I'm currently using CamillaDSP in my setup, and maybe more users will share their experience, implementations, and tw...
There is a reason why that convolver is free Not to say that free stuff isn't good, because you may be surprised how powerful it is (another example is REW/rePhase/MSO). I only use REW casually, but I have been told that it can do anything that Acourate/Audiolense can do. The downside is that it is extremely manual and you really have to know what you are doing. Same with Camilla, if it was as polished as (say) Ekio or Hang Loose ... Henrik would (and should!) charge money for it.
Trouble is, unlike most here who use multichannel DACs, I will be limited to a stereo DAC like this one. https://www.kitsunehifi.com/product/holo-audio-may-dac/ So, even though my music sources are all via my PC, I don't see how I could include both my main speakers and my pair of powered subs into any room correcting DSP filter loop with only two DAC channels.
Even with one stereo DAC unit and one stereo amplifier driving passive LCR-network SP system plus subwoofers, you may test/evaluate a kind of naive compromise which I personally call Quasi-Time Alignment (or Sham-Time Alignment) DSP setup like I suggested in my post here and here.
The paid EKIO supports non-limited numbers of output channels, and IN and OUT routings are very much flexible through all the ASIO drivers available in the PC; simple straightforward and well designed GUI for easy and stable operation. Almost all of the parameters can be flexibly changed/adjusted "on the fly".
EKIO uses IIR filters. The processing is done using a cascade of second order transposed direct form II biquad sections. Every calculation is done using 64 bit floating point numbers. EKIO works very fast and stable, small usage of CPU power of my completely silent audio dedicated PCs.
I confirmed that EKIO gives no audible post-ringing nor pre-ringing at all in my rather sensitive test environment.
I recently established 0.1 msec precision time alignment (proved by objective measurement of room air sound, summary here) over all the SP drivers using EKIO's reliable (and fully validated) group delay capabilities.
In pure physical/theoretical perspectives, I assume you are right on this point, but some of the audio authority people might possibly say "in ultimate microscopic observation and sense, phase and time-alignment is just identical, same things!". I would like to leave this important "theoretical discussion" for other authority/professional people looking this thread.
In general practices in our home audio setup, the phase/time-alignment tuning would be most important between subwoofers and woofers, and less critical between woofers and midranges/tweeters, and especially so, I mean less critical, in your specific setup where you are planning to continue using the passive SP system, even though 0.1 msec precision time-alignment can be set in DSP "EKIO" using IIR filters.
If you use rather mild IIR filters of Linkwitz-Riley -12 dB/Oct at both side of XO Fq, usually the phase/polarity matching is not so difficult/complicated; you just need to inverse the subwoofers against woofers to give flat theoretical phase response. Furthermore, your excellent active subwoofers should have phase inversion remote switch can be controlled (further inverse or not) from your listening position depending on your room acoustics. Some of the advanced active subwoofers even has more sophisticated "remote phase controller" to be flexibly adjusted/optimized depending on your room acoustic and your subjective sensations. The active subwoofer also usually has built-in low-pass (high-cut) filters of -24 dB/Oct or steeper at any where in 30 Hz - 120 Hz; which can act as "additional" filter to DSP EKIO's mild -12 dB/Oct filter.
Then, you may give more or less group delay for woofers (plus midranges and tweeters) to time-align with subwoofers; usually such time-alignment would be 1 msec to 20 msec range in 1 msec precision which is precise enough for the low XO Fq range of say 30 Hz to 100 Hz zone in consideration of actual wavelength of acoustic sound thereof.
The given group-delay of woofers against subwoofers may give more-or-less complexities to the theoretical digital domain phase and gain curves of EKIO's phase/gain representation, but at first you should just ignore such "digital domain theoretical up-down", and you should concentrate yourself for subjective listening evaluation (trust your ears and brain for your preference) at your listening position by listening to your own consistent and familiar "audio reference/sampler music playlist" (in my case, ref. here and here), and, only if possible, objective microphone measurements.
(If you would be really serious in objectively observing the delay and room acoustics at your listening position, you can utilize REW's wavelet analysis, of course using measurement microphone, as I did at the very early stage of my project; e.g. ref. here, that was still one-DAC one-amp passive SP setup.)
As I recently wrote here, all of our DSP and room treatment efforts towards better and optimal sound quality at listening position should be achieved only by various combination of "compromises" in DSP and room treatments in our own individual room environment, and they would differ our room to room, gears to gears, DSP to DSP.
Even though I myself did rather intensive objective measurements for optimization of subwoofer-woofer and woofer-midrange crossover using DSP "EKIO",
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507
- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495, #497, #503, #507
I highly recommend you to start "playing" with paid-up "EKIO" (only US$149) for your subjective evaluations which enables unlimited numbers of input and output channels in the Quasi-Time Alignment (or Sham-Time Alignment) setup (ref. here) with your single stereo DAC and single stereo amplifier; I believe you may clearly hear the difference in actual room sound at your listening position before and after the EKIO's XO/Gain/Group-Delay configurations, and this would be your entrance gateway to the wonderful and amazing DSP world.
Edit:
And as I recently wrote here, now I believe that "In DSP, the simpler the better, the simplest the best".
Don't hold your breath on Dirac doing that. Pretty easy to do the XO without Dirac. I create my XO in REW & rePhase, flatten and level match the initial response of all the drivers and then some form of room correction. Currently I'm playing with DLBC so I let Dirac control the sub/bass XO and I only need XO points between bass/mid/tweeter.
mvMeter2 are used as VU and Peak meters to monitor the level of the stereo input. Dirac Live Bass Control is being used to apply digital room correction for two subs. Hang Loose Convolver is hosting the digital crossovers. And MEqualizer is being used to apply a high pass and low pass filter on one sub.
It's his cluttered diagram that makes it look complex. If he removed all the stuff that doesn't need to be there (e.g. show one channel instead of 2, remove the VU meters, the distro panel, etc) his system is as simple as any other multichannel DSP system.
I fully understand your "LOL" if you just look at my latest system setup reached after my almost 5-year system evolve.
On the other hand, I assume that @Dichotome is now standing in front of the entrance gateway to DSP world where I too was standing 5 years ago; I can well understand, therefore, his possible hesitation and/or frustration to have one-step forward into the gateway.
As for my wording of "simple" "simpler" "simplest" in first touch on DSP, I hope you may understand my naive primitive subjective impressions written in my very early posts around April of 2020, e.g. posts #37 and #39; at that time I was just initially testing DSP EKIO in single-stereo-DAC single-stereo-amp passive-3-way-SP-system plus L&R active subwoofers similar to what @Dichotome is now planning to go forward with; that was just before I received my OKTO DAC8PRO.
As far as my latest DSP EKIO configuration;
I believe it is still "simple" enough in consistent with my policy of "In DSP, the simpler the better, the simplest the best", even though I have implemented flexible on-the-fly relative gain controls in analog domain by using HiFi integrated amplifiers and active L&R subwoofers; my DSP configuration includes no "black-box-type" signal manipulation like automatic room-mode corrections.
Consequently, I should say that my latest system setup would be optimal compromised combinations of "simple" DSP and analog level flexible tunings as well as physical room acoustic treatments.
It's his cluttered diagram that makes it look complex. If he removed all the stuff that doesn't need to be there (e.g. show one channel instead of 2, remove the VU meters, the distro panel, etc) his system is as simple as any other multichannel DSP system.
I do have a pretty basic question. I've been trying to understand when, how and why you would use Digital Crossovers in Audiolense, but i seem to be too stupid to understand the whole concept of it i'm afraid. Now i have 2 active 3-way speakers, Adam A77h, no subwoofer. The internal crossover is handled by a dsp. Should i still use the digital crossover in Audiolense and if so, should i just adapt the Crossover Frequencies supplied by Adam which the internal crossover does?
If you can bypass the crossover in your speakers, then you can use an external crossover of any type.
To use Audiolense's crossover function, you would need one DAC channel and one amplifier channel for every driver. The Adam A77H is a 3 way monitor. This means you need 6 DAC channels and 6 amplifier channels, as well as performing surgery to bypass the internal DSP crossover and amplifiers. This obviously is not trivial, it involves a lot of expense purchasing additional electronics, as well as bypassing good electronics and DSP that you have already paid money for. You would do this if you thought that you can do a better job with your DSP software than the engineers at Adam.
The other use case for Audiolense's crossover function would be to integrate a subwoofer properly. However, you aren't using a subwoofer.
I would suggest it is wiser to keep your Adam speakers the way they are, and use Audiolense for room correction only.
www.stereonet.com
Not to say that free stuff isn't good, because you may be surprised how powerful it is (another example is REW/rePhase/MSO). I only use REW casually, but I have been told that it can do anything that Acourate/Audiolense can do. The downside is that it is extremely manual and you really have to know what you are doing. Same with Camilla, if it was as polished as (say) Ekio or Hang Loose ... Henrik would (and should!) charge money for it.
