Optimizing Acourate Room+Speaker+Sub Correction

[The Computer Audiophile]

24 minutes ago, ecwl said:

First, I really have to thank @The Computer Audiophile for inspiring this project.

My subs are set to crossover to compensate my main bookshelf speakers (which I don’t use crossovers for transparency reasons). I have been using parametric EQ to address bass issues from 20-100Hz for years (and lately 20-300Hz). But in the midst of the pandemic, and because of our great leader’s wonderful experience with Audiolense’s convolution filters that @mitchco created for him, I decided to get Acourate and try to optimize my setup further. And I have to admit I learnt a few things. Obviously, YMMV. And you’ll probably disagree with me on a lot of this.

Before, I start, the main reason why I use Chord DACs and Chord M-Scaler is because I find the long tap length filters from Chord to really create realistic transients. I have found many people who drop by my local dealer can’t hear the differences, e.g. hand clapping, drum strikes, guitar plucks, cymbals, triangles. In fact, when I first got my Chord DAVE and compared it to Mojo, I can hear many improvements but just not the transients. The other thing I found Chord DAVE to shine at is soundstage depth if it’s in the recording, whereas I’ve heard a lot of DACs create and homogenized artificial soundstage width while shortening soundstage depth even if it’s in the recording.

Now despite most people feeling that parametric EQs are transparent, I have to disagree slightly as I often do find that there is always a very, very subtle loss in transparency and soundstage depth. However, because my bass issue is sufficiently significant (as with most real room systems), there is no question that parametric EQ of the bass peaks improves the sound dramatically compared to the minuscule loss in soundstage depth.

But for Acourate, it was quite easy to decide the target frequency response to mimick what I previously used for parametric EQ and the natural roll-off of the high frequency in my room (which happens to co-incident with EBU 3276. The real challenge I found was choosing the appropriate Phase Correction and Pre-ringing Correction. Since I sometimes do turn off my subwoofers at night and just listen to the bookshelves on their own, I also created separate filters for them. And the bottom line in order to understand how the filters affect the sound, I ended up creating more than 30 filters before reaching what to me, was optimal.

My major experience/findings were:

1. Convolution filters (at least via Roon) are even “less transparent” than parametric EQ. Not in the sense that I really noticed any loss of transparency to the music but I do notice a much more significant loss in soundstage depth. However, the phase correction when done optimally (or close to optimal) offers so many other benefits that once again this mild loss of soundstage depth to me was acceptable. That said, I suspect the soundstage depth change may or may not be audible in your system. 

2. While it is theoretically possible to use a long excess phase Frequency Dependent Window to alter the step response to mimick the the ideal, it is fairly easy to go overboard and introduce excessive ringing or group delays above 100Hz. Given my room, the speaker design, and ultimately, I already use “suboptimal” integration of subwoofer to speakers, there is only so much phase correction I can do before unpleasant artifacts get introduced to the system

3. So within the limits not introducing group delay or ringing artifacts, there is still lots of options to set for the length/width of low-frequency vs high frequency excess phase FDW. And since the group delay artifacts from excessive correction are often generated in my system between 100-1000Hz, I essentially have a choice to set the width of low-frequency and high frequency excess phase FDW the same, or I can set the low-frequency higher or the high-frequency higher before group delay artifacts come in. And what I found was that while setting low-frequency excess phase FDW higher does create more realistic and wholesome drum sounds and resonances, it is important to set high-frequency excess phase FDW higher in general because for me, that determines the accuracy of the transients which are highly audible in my system. Basically, if the high-frequency excess phase FDW is set too low for me, the whole system loses PRaT. Moreover, even though graphically I can correct pre-ringing in the step response by engaging in more pre-ringing correction, I also found that the correction it self also worsens the transient accuracy as I can hear them so engaging pre-ringing correction also always seems to lose PRaT for me. For reference, I would cross check by plugging my headphones directly into Chord DAVE to ensure the transient attacks are not an artifact of phase error.

4. Because the system is asymmetrically setup, I can usually do more correction on the right side than the left. But I find that sometimes with excessively long/wide excessive phase FDW correction on the right, I can actually hear that the same transients on the left don’t sound the same as the ones no the right. In fact, my favorite test track for this is Unsquare Dance from Further Time Out. So I find that I’m always writing down the Interaural Coherence Coefficient to see what the maximal value I can get between the two channels and then while I can push the right channel correction more, I don’t let the IACC drift off by more than 0.1%-0.2% optimal.

5. So in the end, in Acourate, the way I found the optimal settings for my system was to set Pre-ringing correction to 0/0, start with excess phase FDW at 1/3 on each channel and just slowly increase it by 0.1, e.g. 1.1/3.1, 1.2/3.2, etc. Until I get Group Delay >100Hz or pre-ringing. And then let’s say the optimal I can get on that channel is 1.4/3.4 for low vs high-frequency excess phase FDW, I would then try increasing the low-frequency setting by .1 to see if it’ll introduce Group delay >100Hz or pre-ringing. And then I would do the same for the other channel while looking at the IACC to see what is optimal and how far I can push the other channel. At least to my ears, this seems to be the optimal balance between trying to obtaining optimal transient attacks (which is why I got my Chord DACs in the first place) while getting as much low-frequency correction so that I can get improved bass resonance sound, e.g. drums that I hear in Copland’s Fanfare for the Common Man.

I know this approach seems a bit different from what’s been written here on forums. I definitely re-read over and over again what @mitchco wrote here and in his outstanding book and of course most of what I’m doing is recommended by him. It was just that the end optimization for me was to preserve the transient attack while trying to improve the phase response of the system. It’s been a long month of trial and error but I’m very happy with the sound I got. It definitely pushed my system beyond what I was able to achieve with parametric EQ alone. However, my experience also makes me wonder if it is truly possible to automate this type of process as I have previously found that Dirac has been a hit or miss in various systems at my dealer’s, almost always resulting in a loss of transient attack, and I wonder if it’s because the phase correction was excessive at times and not at other times. I know Uli the designer at Acourate has mentioned that his experience with phase correction is that it is very system dependent so it is likely that my method might not even work for somebody else’s system. However, I thought i would share my experience.

What a great post! Thanks for sharing all that info and your honest conclusions. So neat. 

[The Computer Audiophile]

Getting the absolute best out of my system and these apps required me to engage @mitchco at https://accuratesound.ca/. He’s the professional who eats, sleeps, and breathes this stuff. I attempted to read everything but I just couldn’t convince myself that wouldn’t cause more harm than good. 
 

It seems like you guys understand this quite a bit more than I did, so you likely have it under control. If others reading this are like me, I suggest bringing in the best because it can get ugly quick on your own. 

[The Computer Audiophile]

1 minute ago, ecwl said:

While I do think @mitchco is the master at this, I think my experience has made me question: Is there an optimal convolution filter (assuming you've picked your favorite frequency response, e.g. EBU3276)? Mathematically and graphically, there may be an "optimal" convolution filter that generates the best looking step response possible for specific speakers and room. However, I'm wondering for those generating convolution filters at home, whether some like me would prioritize high-frequency phase response for better transient attacks while others would prioritize a more even low- vs high-frequency phase response correction. 

Unfortunately because of the pandemic, I have not invited my friends over to listen to the various filters I generated. I would be curious whether they would prefer the pure parametric EQ correction, my favorite convolution filter or the more even handed convolution filter I created. To my ears, each filter seems to involve a set of trade-offs amongst soundstage depth/transparency vs bass phase response accuracy (bass coherence/drum sounds) vs transient attack accuracy. But all of them, when set correctly are a significant improvement over uncorrected, uneven bass response.

Yes +100.

 

All of these are the important questions to ask when using DSP. In my experience, there is no optimal anything, let alone convolution filter, in HiFi. When designing a component it's all about tradeoffs and DSP is no different. 

 

When I worked with Mitch, he sent me filters and asked for my opinion. through this easy process he delivered the filter that sounded best and most correct to my ears. Given that we all hear and process sound in different ways, there just can't be an optimal filter. 

 

This is all really fun stuff because it has such a dramatic effect on the sound and the cost is peanuts compared to the rest of this hobby. 

[The Computer Audiophile]

Just now, asdf1000 said:

 

Wait till software can play test tones through your speakers to test what frequencies you can and can't hear and then the EQ for music playback is not just room correction, but also 'hearing correction' at the same time !

 

 

 

This would be a really cool development. Digital Room Correction and Digital Ear Correction.