Jump to content

Using REW to Determine Time Delays Between Drivers


Chris A
 Share

Recommended Posts

26 minutes ago, rplace said:

Are the "better" ones still the ones with the least total number of phase shifts/cycles/round-trips?

Yes.

 

26 minutes ago, rplace said:

BTW, what is the proper term for cycling through 360 degrees of phase over and over across the 20-20K range?

"Wraps."

 

25 minutes ago, Rudy81 said:

I am slowly getting a grip on what you are trying to teach us.

 

Here are some areas that you can look forward to exploring if you have the interest:

  1. The RT60 (reverberation time) plots to check the evenness of your in-room absorption/major acoustic reflection time periods (there's usually a rise in the RT plots around 1 kHz that can be addressed by moving around your in-room absorption and adding/subtracting absorption area).
     
  2. ETC (energy-time curve) plots looking for times (and therefore places around your loudspeakers ) of the biggest near field reflections in-room, and whether or not you've got enough absorption around the loudspeakers.
     
  3. Using step response plots (under the impulse response plot area) to see the overall end effects of your tweaking.
     
  4. Using the "Overlays" facility to plot more than one measurement at a time to see the effects on SPL and phase of your tweaking--measurement to measurement, or comparing loudspeaker to loudspeaker, etc.
     
  5. Using Stepped sweeps (i.e., very long sweeps measured in minutes) to see deeper into the noisier environment, that can reject noise and statistical disturbances
     
  6. Multi-tone measurements to see modulation distortion sidebands.  This can be used to see FM and AM distortion differences between loudspeaker types (i.e., direct radiators vs. horn loaded bass bins, the high levels of modulation distortion of turntables/needle cartridges (this is an area that will really make your hair fall out when you see the actual turntable measurements...😨
     
  7. Harmonic distortion plots to see frequency areas of resonances in your loudspeakers: horns and/or boxes.
     
  8. Using the EQ facility more, you'll become more adept at using PEQs in combination and progressively working down the peaks in response (and sometimes the dips in response, too) by using input channel PEQs in combination with output channel PEQs, and learning how to identify PEQs that you've already set that need to be either modified or cleared as you work down the SPL peaks.

________________________________________________________________________________

 

There are some tricks that can be used with the DSP crossover that flattens the loudspeaker/room phase response, but that's a subject of the "Part 2--Advanced" tutorial that I'm working on, and that tutorial will mostly apply to those people wanting to squeeze the last 5% out of their setups, and are willing to do a little acoustic treatments in-room, and to take advantage of the full-range directivity of their fully horn-loaded loudspeakers (i.e., horn-loaded bass bins).  That's where I found some real pay dirt with the sound of the system--by reducing the total phase growth to less than 360 degrees globally/45 degrees locally, and choosing better places to cross the drivers, using a more intelligent method.

 

Chris

  • Like 2
Link to comment
Share on other sites

By the way, the latest version of REW (v5.20 beta 27) has changed the color and line type (dotted) of the phase traces relative to the SPL traces in the SPL & Phase view, thus aiding the visibility of phase vs. SPL greatly--a very welcome change.

 

 

 

Chris

Link to comment
Share on other sites

Thanks to Chris I was able to make a breakthrough in my speaker setup. In all honesty, I have had a difficult time grasping some of the concepts in Chris' technique. I really struggled to understand not only the graphs I was creating, but also how those graph results could help tune the speaker.  It did not help that I was not setting REW correctly in order to have the graphs show the proper information. 

 

Until today, I had not been able to correlate Chris' instructions with my limited knowledge of acoustics and the physical characteristics of my two way Oris based system. I decided that the best thing to do was to just spend some time running some experiments that demonstrated what I did know, and then move on to the new information.  In my mind, these are scientific measurements and should follow scientific principles.  This should not be a guessing game, which is what I was doing making incremental delay modifications to my system. That was a waste to time.  I will chronicle what I did and the results.  Should the information be incorrect, please speak up. But I do know the results are conclusive.

 

First, I set the mic exactly 3' from the front of my bass bin.  Vertically, placed half way between the center of the Oris and LF drivers.  One problem with that is that the HF measurements will show a reduction in HF output due to the mic location.  Just the nature of the beast.  Since I was concerned with Xover selection and time optimization, the HF quality was irrelevant.

 

I removed all EQ, timing and xover settings from the Ashly crossover.  I then ran sweeps on the LF and HF section from 200hz-20kHz.  I started at 200hz to somewhat protect the HF driver.  Using the loopback timing feature of REW I now had a true time differential between the LF and HF sections without any crossovers, EQ or other artificial changes to what the driver was doing.  In my case, based on where the Oris horn sits on my bass bin, the difference was O.52ms.  In my mind, that correlates with the physical setup of the speaker. I then also looked at the phase plots to get an idea of what would be happening as I made other changes.

 

BTW, the red trace is the HF plot. Not sure why the trace tags were not kept in the pic.

 

Raw driver SPL.jpg

Raw Driver Phases.jpg

Link to comment
Share on other sites

Next, I added the 0.52ms delay to the fastest driver.  In the case of my mains, that is the LF bin with two 15" Eminence Kappalite 3015LF drivers.

At that point I wanted to work with a xover point of 353Hz and also wanted to use a L/R 12dB/octave setting.  After adding these changes, I then ran a full sweep and applied what Chris had taught me.  What I did not know in past plots was how to properly use REW to obtain the correct GD and Spectrogram results. So, I set the IR to zero, generated minimum phase, unwrapped the phase and then went on to work on the plots.

 

Here is what I came up with.  Chris taught me that the large peak around 417hz in the GD excess phase plot indicates a need to invert the LF.  When I did that, I got the results in the second GD plot. In my mind, the time delay I obtained of 0.52ms should be THE correct setting. I then went on to play with nearby delays either side and confirmed that any changes to that time settings made the graphs worse, not better.  I then have confidence that I had a good number for delaying the LF.

 

 

 

Full Range Normal LF Polarity GD.jpg

Full Range Inverted LF Polarity GD.jpg

Link to comment
Share on other sites

No matter what further changes I made to the timing, I could not improve the excess GD that you see above.  The corresponding spectrograms are these. No matter what I tried, I could not get the bottom graph below to have a more vertical dashed line at the lower frequencies so I am assuming this is the best I can get.

 

Full Range Normal LF Polarity Spectrogram.jpg

Full Range Inverted LF Polarity Spectrogram.jpg

Link to comment
Share on other sites

Finally, since I changed from my original 24dB/octave L/R xover to a 12dB/octave setting, I raised the xover frequency to 408hz to better protect the Fostex HF driver. Probably not necessary, but during my tests I noted that the 12dB spectrogram plots are 'fuller' in the lower frequencies than the 24dB plots. 

 

I then set up the active crossover with one set having a 24dB/octave xover and one having a 12dB/octave crossover. I spent a little bit of time just A/B playing music and trying to decide which I liked better.  Although it was not an extensive listen, I felt the 12dB/octave settings had two things that stood out.  A wider image,  which I'm not sure is better, and a much wider soundstage.  So, I have gone with the 12dB/octave settings for the xover. Below is the final plot for those settings.

 

Finally, all this makes sense to me and doing it from scratch made total sense. I'm not sure if all this is correct, but it sounds good and I think that at least I'm on the right track.

 

Fire away.....

Final Full Range Inverted LF Polarity Spectrogram.jpg

Final Full Range Inverted LF Polarity GD.jpg

  • Like 1
Link to comment
Share on other sites

Once these new settings were set, I then ran Audyssey Pro again.  The interesting thing I noted there is that the sound difference between Audyssey Off and Audyssey On is minimal now.  That has not been the case before.  Only the house curve I added to Audyssey Pro is notable.  Otherwise, the sound is very similar. Which, I take as a huge positive to what the time changes have made.

 

One other note for learning purposes. After figuring the 12dB/octave settings, I then worked on the 24dB/octave settings.  The 12dB/octave settings required the LF to have inverted polarity to obtain the optimum excess GD plot.  The 24dB/octave plot required the LF to have normal (not inverted) polarity. I suspect this has to do with the phase change induced by the higher order filter.  IIRC Chris mentioned that change in one of his exchanges while trying to tutor me.

 

I really want to thank Chris for his patience in helping me and his efforts to teach us about the benefits of DSP.

 

  • Like 2
Link to comment
Share on other sites

17 hours ago, Rudy81 said:

Finally, since I changed from my original 24dB/octave L/R xover to a 12dB/octave setting, I raised the xover frequency to 408hz to better protect the Fostex HF driver. Probably not necessary, but during my tests I noted that the 12dB spectrogram plots are 'fuller' in the lower frequencies than the 24dB plots. 

 

I then set up the active crossover with one set having a 24dB/octave xover and one having a 12dB/octave crossover. I spent a little bit of time just A/B playing music and trying to decide which I liked better.  Although it was not an extensive listen, I felt the 12dB/octave settings had two things that stood out.  A wider image,  which I'm not sure is better, and a much wider soundstage.  So, I have gone with the 12dB/octave settings for the xover. Below is the final plot for those settings.

 

Finally, all this makes sense to me and doing it from scratch made total sense. I'm not sure if all this is correct, but it sounds good and I think that at least I'm on the right track.

 

Fire away.....

Final Full Range Inverted LF Polarity Spectrogram.jpg

Final Full Range Inverted LF Polarity GD.jpg

So you've got the total excess group delay growth down to 1ms at 100 Hz: that's pretty good. 

 

If you now listen to recordings that have good phase coherence, such as an excellently recorded orchestral selection or stereo-pair microphone recording of a solo artist (such as acoustic guitar), you should hear a much smoother and realistic sound than prior DSP settings.

 

Chris

Link to comment
Share on other sites

Chris, as soon as I get a chance to sit and listen for a while I will do that.  Thank you.  I also plan to spend some time switching back and forth between 12dB and 24dB L/R filters.  Again, thank you for all your help. It is greatly appreciated!

Link to comment
Share on other sites

33 minutes ago, Rudy81 said:

I will take a look at my Ashly xover and see if it has first 6dB filters.

 

I just looked through the smallish manual and it looks like 12dB is the smallest you can do. Hope you find out otherwise.

 

Bruce

Link to comment
Share on other sites

9 hours ago, Rudy81 said:

I had thought that but wasn't sure.  Thank you.  I a contemplating going to the Xilica XP4080. Chris indicates the Xilica flexibility  may be a better fit for what we are trying to do.

Get the 8080 it's not much more $. I'm using 6 inputs and 7 outputs an have yet to touch the surrounds

Link to comment
Share on other sites

9 hours ago, rplace said:

Get the 8080 it's not much more $. I'm using 6 inputs and 7 outputs an have yet to touch the surrounds

 

I saw the 8080 has non XLR connectors on the back.  Do they require new cabling?  I have not looked into that at all.

 

.....never mind. Youtube is my friend

Link to comment
Share on other sites

4 hours ago, Rudy81 said:

 

I saw the 8080 has non XLR connectors on the back.  Do they require new cabling?  I have not looked into that at all.

 

.....never mind. Youtube is my friend

 

Ha, this way you can buy 50% less cables, cut them in half and you have everything you need in and out of the Xilica

Link to comment
Share on other sites

Xilica XP4080 is on its way from Paducah Home Theater. I am looking forward to having more EQ flexibility.

 

Did some more listening with the current settings and the center image is most definitely wider than in the past.  Not sure if it is due to the change in the Xover/slope selection or the adjustments in GD.

 

I do like the changes though.

  • Like 1
Link to comment
Share on other sites

Long week since I got involved in really applying what I learned in this thread.  Got the new XP4080 installed.  What a nice unit, the noise floor on this thing is ridiculous....totally quiet as best I can tell.

I found out through much trial and error that each crossover point, type, and slope you pick affects the outcome of the GD and spectrogram. I then had to re-work the timing and at times polarity.  I did set one of the presets to the Danley style no-crossover crossover.  Sounds really good upon a cursory listen.  I have spent so much time working on this I had to take a break.  For now, the system sounds really good.  I have to admit that I didn't even run Audyssey!  Frankly, don't plan on it.  I got my bass back by doing the tuning manually. I was always aware that Audyssey Pro messed with my Bass too much and this is proof. 

 

Once again, thank you Chris for all your help! I really like having full control of the entire system and how it sounds. 

  • Like 2
Link to comment
Share on other sites

15 hours ago, Rudy81 said:

I have spent so much time working on this I had to take a break.  For now, the system sounds really good. 

I find that the learning coupled with the resulting better sound makes it even more satisfying-- knowing what the root causes are and how to deal with them. 

 

The re-listening to recordings with improved sound is the icing--which is enjoyed over a much longer period of time. 

 

Chris

  • Like 1
Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

 Share

×
×
  • Create New...