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Multi-Way Conventional loudspeakers with crossovers

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Old 5th November 2013, 06:34 PM   #551
KSTR is offline KSTR  Germany
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Quote:
Originally Posted by TOINO View Post
Just Divide 48KHz by 4 ??
Si.
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Old 5th November 2013, 07:35 PM   #552
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danke :-)
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Old 5th November 2013, 09:58 PM   #553
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Quote:
Originally Posted by Pano View Post
At the moment I'm using my M-Audio Fast Track Pro as the output, it has 4 channels out. The player is JRiver. I just tell JRiver that I want 4 output channels and the convolver script text file does the routing, crossovers and such. I currently have a file for HP and a file for LP, each 24 bit mono wav. JRiver hits the 4 channels with Left Low, Right Low, Left High, Right High, then it's straight out of the Fast Track to the amps. So JRiver is doing the splitting into 4 channels, I tell the convolver which impulse to use and where to route it. Simple, but took some time to figure out.
What I'm suggesting is to refine the "script", for natively supporting IIR BiQuad filters and FIR filters.
The FIR filter gets described just as usual, using a monophonic .wav file.
Each IIR BiQuad filter would get described using a short stereo .wav containing 3 stereo samples. The non-recursive part of the IIR BiQuad would be described using the left channel. The recursive part of the IIR BiQuad would be described using the right channel.

The script would be a text file just as usual, naming a succession of .wav files. Say 8 x IIR BiQuad filter, plus 1 x FIR filter.
A typical two-way Xover crossing a 2kHz would specify 8 x IIR BiQuad filters in series, followed by a single 128-tap FIR.
A typical two-way Xover crossing at 200 Hz would specify 8 x IIR BiQuad filters in series, followed by a single 1024-tap FIR.

Basically, the aim of the 8 x IIR BiQuad filters is to shorten the speaker impulse response, prior to applying the FIR filter. Most bass and midbass drivers exhibit a sharp cone resonance at approx 5 kHz. Some exhibit a +5 dB resonance, other exhibit a +15 dB resonance. Their impulse responses exhibit thus ringing. If you intend to suppress such ringing using a FIR filter, the FIR filter needs to be long, covering tens of milliseconds. A more suitable approach is to suppress such ringing using an IIR BiQuad filter configured as -5 dB or -15 dB notch at 5 kHz with the correct Q factor. Consequently, the FIR filter that's following doesn't need to cover tens of milliseconds. The FIR filter that's following can be 128-tap (less than 3 ms long at 44,100 Hz), fully dedicated to the fine gain and phase linearizations.

Oh, and I'm forgetting an advantage. Tuning an IIR BiQuad filter is easy and intuitive. It's like operating the pots of a PEQ. After having suppressed the main ringing, you'll get motivated for suppressing second-order ringings, for filling bumps in the frequency response, and for pre-filtering the signal using a lowpass and/or a highpass. Doing so you will be amazed how a succession of eight IIR BiQuad filters can massively improve (shorten) the impulse response.

This way, the FIR filter that's following appears as a nice refinement, not only for linearizing the amplitude response into a 2 dB corridor, but also for reaching a smooth controlled phase.

Such approach got applied in the Philips DSS930 and DSS940 digital speakers, back in 1993. Two-way speakers. Crossover around 3.5 kHz. DSP at 44.1 kHz. Only 41 taps in the FIR filter, if I remember. Outstanding overall impulse response, as result.
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Old 5th November 2013, 10:37 PM   #554
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FIR filters are very good devices for audio, at the condition that they get used in a proper way.

1- FIR filters can be long, and describe low orders, say an order of 3
2- avoid high orders tending to brickwall filtering, as they generate ringing
3- when crossing at 2 kHz, very good results can be obtained using a 128-tap FIR filter
4- when crossing at 200 Hz, very good results can be obtained using a 1024-tap FIR filter
5- an FIR-based crossover can be designed for implementing a symmetric phase-linear 3rd-order (highpass 3rd-order and lowpass 3rd-order)
6- AND designed for zero relative phase shifts
7- AND designed for zero overall phase shift

The IIR BiQuad filters I'm referring in the post above, are only there for maximizing the FIR-based crossover efficiency. The FIR filter efficiency gets maximized when the FIR filter doesn't need to battle against the speaker cone breakup and ringing at high frequency. A single carefully tuned IIR BiQuad filter will eradicate the nasty effect of the main (first) speaker breakup mode. Adding a few more tuned IIR BiQuad filters will eradicate the effects of higher-order speaker breakup modes. This way the tweeter emission won't get garbled by spurious emissions coming from the midbass speaker.
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Old 5th November 2013, 11:04 PM   #555
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look post #16 here:
iDFT-based XOs (FIR)
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File Type: png 200 Hz 3rd-order.png (63.3 KB, 485 views)
File Type: png 2000 Hz 3rd-order.png (70.6 KB, 477 views)
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Old 5th November 2013, 11:28 PM   #556
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Regarding IIR BiQuad filters, it's here :
IIR_Lab mini : a design help for digital audio filters

A0, A1, A1 are the coefficients in the recursive path.
B0, B1, B2 are the coefficients in the non-recursive path.

Those six values can sit into a stereo .wav containing 3 stereo audio samples.
For avoiding numbers outside of the (+1,-1) range, we would divide them all by two inside the .wav.
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File Type: png 4000 Hz notch -6dB Q3.0.png (65.5 KB, 475 views)

Last edited by steph_tsf; 5th November 2013 at 11:33 PM.
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Old 31st December 2013, 10:28 AM   #557
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There is a nice series of articles about FIR filtering on Pro Sound Web, written by Pat Brown.
As of now there are two articles, but probably more to come:

An introduction to FIR: AV: FIR-ward Thinking: Examining Finite Impulse Response Filtering In Sound Reinforcement Systems - Pro Sound Web

Brickwall filters using rePhase: AV: A Useful Tool: Creating & Applying FIR Filters - Pro Sound Web
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Old 6th January 2014, 05:35 PM   #558
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Originally Posted by pos View Post
Art, what are the crossovers and (acoustical) filters used in your system?
Missed your question, the crossover used for the FR posted in #527 is a DBX Driverack PA, filters are 24LR and BW.

Thanks for all the explanations and examples in post #544.
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Old 7th January 2014, 02:22 PM   #559
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You're welcome.

If you post a measurement of your system (in wav, txt, or freq/amplitude/phase columns) I can show you different scenarios of what can be corrected for what delay.
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Old 28th January 2014, 02:16 AM   #560
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Excuse me for answering an old post in this thread. I'm just treading water here...

Quote:
Originally Posted by CopperTop View Post
Why use a specialised DSP board when a near-silent PC can be had for a couple of quid?
And then you can develop on the target machine using free software. 32 bit floating point is very quick and it's powerful enough to use 64 bit if you so desire (even on a 32 bit machine). Plus you lock everything to the same sample clock (sound card is the destination for the audio stream, source for your crossover app, and destination for your crossover's output).

All this talk of SHARCs etc. makes me think I'm missing something, and also makes me head spin at the learning curve involved.
There's only one reason to justify the SHARC and Mini DSP at this point in the game: Making a reliable backup to your Windowz computer. For example, I'm using Acourate Convolver for my main audio system, 64-bit, dithered volume control, to 24 bit on the way to the DACs, superb convolver, 65k impulse response length, etc. etc. etc. What's not to like? Uh... yeah, Windowz. Can you say, "excuse me, I have to reboot my loudspeakers." So, basically, I'm learning how to get the best performance possible from the Mini-Sharc as a backup system. It boots up instantly, it's stable, and with the best optimization it should get me through a few days of main computer being down.

I do wish someone would conquer developing an Intel-based convolver in an embedded system that doesn't require windowz and does what the Mini-DSP system does. We're just not going to see a dedicated DSP chip with the speed and power of an Intel i7.
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