rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 58 - diyAudio
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Old 29th January 2014, 03:34 PM   #571
bobkatz is offline bobkatz  United States
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Nice to see you, Tom! Bonjour(soir). As a purist I'm concerned about the decimation steps you describe. It remains to be heard. What I do to avoid the question with Acourate is run a long filter when I don't care about latency, 65536 sample IR length. I think that means the same as "taps" (crazy long, huh, but that's what Uli believes in and it sure sounds pure and accurate!

When I care about latency, only for editing, PQ coding and cleanup tasks in mastering, I push one button on a remote that Uli and I have devised, and it switches to a short minimum phase filter and minimum phase crossover. So I can have the best of both worlds and not question anything.

So, until I can do a shootout of a tricked-out decimated sharc versus my Acourate system I prefer to stick with what I know to sound absolutely transparent, and stunningly musical. I also run the same Acourate-generated filters in Jriver so I can have perfect audio-video sync with these enormous FIR filters.

So for me, conquering the stability issues of Windows would be nice. An embedded system that just boots to the software of interest in less than a minute would be my goal. My Shuttle Computer that runs Acourate Convolver has a main and a backup SSD, but it's always asking for Windows updates (even for Windows 8) and it runs antivirus and other stuff that bother me. And it takes about 3-5 minutes to get everything up and running from a cold or warm boot. That's not an appliance, that's a home computer. I want an appliance :-). An appliance would solve the day-to-day issues of occasional reboots, etc. But it would not solve the issue of total breakdown.

So instead I'm building two Open-DRC boxes as soon as they arrive from Mini-DSP and I'll do something like my Acourate filters dumbed down to 6144 taps or a combination of IIR and Rephase and that will be my backup for the stereo portion of my system which I need daily for my mastering work. The surround portion, bass-managed and DRC'ed by Acourate, can live with being down if the Shuttle computer dies as I don't do surround mastering that often.

To answer the other poster's question, why do I need an i7 to run convolution? I'm not sure, I just wanted to have the most memory and speed possible. For 5.1, bass management, DRC and LFE, I have 12 or 13 individual 65536-long FIR filters that can run 192 kHz material direct to the DAC at 192 kHz with no decimation or ASRC in the path or necessary. I don't know if something less than my six-core i7 with 16 Gigs of RAM is suitable but I didn't want to take a chance so I did some overkill. The system is stable and reliable and that's what I need.

Latency? When in the high resolution mode... Oh, about a second :-).

Best wishes,


Bob

Quote:
Originally Posted by pos View Post
Hello Bob

Nice to see you here
I don't know if that would be easy to operate an Intel-based CPU without a full-blown (ie bloated) OS, but the hardware sure looks sexy: the Intel nuke plateform can run passive, especially with the new ultra low Wattage CPU coming this year, and SSD disks are now a commodity as well.

That said, even the current crop of SHARC DSP could run much longer FIRs if downsampling (decimation) and/or FFT convolution was to be used (the later requires more memory, and also implies more delay, which is a huge drawback in my opinion).

I know more10 (a member here as well) has been playing with downsampling with a SHARC DSP board with great results : JBL Master Reference Monitor - Page 36
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Old 14th February 2014, 09:11 AM   #572
wesayso is offline wesayso  Netherlands
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Why would you run or need updates on a dedicated PC for media/audio?
Cut off the internet and other file transfers and there's no need for updates and virus scans.
If you only run audio trough the dedicated machine it should be stable for a long time.
Look at it as an industrial PC. You can enhance performance by shutting down all unneeded services. Run two SSD's in raid mirror for safety.
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Old 3rd July 2014, 02:35 AM   #573
ra7 is offline ra7  United States
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Hi Pos, I have a question about delay generated by the convolution file. I'm using the USBStreamer from miniDSP for a three-way crossover implemented through JRiver. Details here:

Setting up a PC-based multichannel DSP system

The speaker is a corner-loaded expanding array:
Corner Expanding Line Array with KEF Q100

I'm getting a 20ms delay between the center element and the mid and outer drivers. Now, I know this cannot be due to the physical separation. Other likely explanations:
1. The measurement in Holm using convolution through JRiver is fairly complicated involving three USB soundcards. There could be latency but you would expect it to be the same for all three drivers.

2. There is a delay generated due to the FIR filters that needs to be accounted for.

See the attached picture from RePhase. Is this the delay I need to account for due to the filter for each driver? Is there a way to precisely measure it? Time lock in Holm is giving wonky results.

Thanks for your wonderful software, btw.
Attached Images
File Type: jpg rephase_delay.jpg (7.6 KB, 271 views)
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Old 3rd July 2014, 08:06 AM   #574
pos is offline pos  Europe
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Hello ra7

Nice system!

The delay you need to account for is the one from the impulse itself (the one shown by rePhase, 90.2ms in your case) + the delay of the convolution engine (none if direct convolution is used, and the size of the buffer for FFT convolution, potentially reduced via portioning) + the delay of your soundcard I/O buffers (the same for all your channels I suppose).
Then if you measure with through software that might add some additional delays...

As a side note, Holm will try to adjust the impulse with t=0 on the biggest positive peak, which is not always correct (especially when polarity is reversed of course). You must alter the time offset of your measurement in Holm until you get the phase response you are looking for.
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Old 3rd July 2014, 03:27 PM   #575
ra7 is offline ra7  United States
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Thanks for the reply, Pos!

I'm only concerned about the delay between drivers. As long as the relative phase between drivers is correct, I'm good. I assume that the delay due to the convolution engine and the delay of soundcard I/O buffers is the same.

So, after the impulse is generated, the info box I pasted in the earlier post is the one to pay attention to, right? Is the delay between two drivers due to the filters simply the difference between the values indicated in that box in RePhase?

I understand the time offset in Holm. I usually set it to largest peak (neg or pos). I'll play with it. The problem I'm having is that two consecutive measurements result in different phase sometimes. Other times they are ok. This may be due to the long circuitous path of the signal output during measurement. As a result, there is no way to tell whether the time lock in Holm has worked or not.
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Old 3rd July 2014, 03:41 PM   #576
pos is offline pos  Europe
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Yes you are correct about the relative delay between drivers.
If the hardware is the same, as well as the impulse length (FFT convolution will imply a different delay for different impulse lengths, even if the peak is at the same distance from the start of the impulse), then the only thing that counts is the delay given by rePhase.

But then, the measurements you have used to generate the rePhase correction also come with their own hidden delay (auto centering in HOLM)...

Normally if all your drivers are phase coherent the delay between drivers should be exactly the distance difference of the emitting surfaces at the listening position.

If you get that right then the sum of the drivers should follow the phase of each driver (ie linear phase if that was your target), and a reversed polarity of one driver should get you a deep null.

I have never used the time lock functionality in HOLM. Dual channel measurement is a better option for this I think (more solid), but even that is not necessary with a little bit of trial and error and the "reverse=null" method.
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Old 23rd July 2014, 01:10 AM   #577
beanbag is offline beanbag  United States
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Hello, I hope this is a simple question that can be answered easily.

Is there a convolution engine that can run as some kind of driver or virtual device or whatever, so that every sound that windows plays (e.g. Firefox, windows media player, foobar, REW, etc) can connect to it? I think Jriver can do something like that, but it seems fairly convoluted (lol pun).

I dunno if that VST or Virtual audio cable is the way to go, since I couldn't tell from their description what they did.

If possible, a (link to a) setup guide would be helpful.

Thanks
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Old 23rd July 2014, 02:02 AM   #578
ra7 is offline ra7  United States
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I think you need a VST host, a program that hosts a VST plugin. Then use the plugin called virtual audio cable. Select input as windows or whatever soundcard you will use to play sounds and output as the DAC you will use. I haven't tried it, but that's the general idea.
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Old 23rd July 2014, 02:52 AM   #579
Pano is offline Pano  United States
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Agreed. I just use the convolution engine in JRiver, which is very easy to do, but of course it only affects the JRiver playback.

If you find the virtual cable patch, please let us know.
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Old 23rd July 2014, 03:48 AM   #580
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Hello,

a more and efficient software,freeware.(virtual audio cable is a little capricious and not freeware)

VBcable,just choose the correct sampling frequency.(for me 48KHz)
need to reboot the computer.

VB-Audio Virtual Apps

once installed,select by default the output device.
then,you can play every shoutcast radio or you tube video.,or any sound played in windows.(via VSThost,and VSTconvolver,this is the same Jriver config file )
http://convolver.sourceforge.net/vst.html(for any crossover).
or SIR V1.011 ( for a single stereo impulse)
http://www.knufinke.de/sir/sir1.php


BR.

Last edited by thierry38efd; 23rd July 2014 at 04:03 AM.
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