rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 28 - diyAudio
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Multi-Way Conventional loudspeakers with crossovers

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Old 5th January 2013, 09:14 PM   #271
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the just midlle,

using a basic laptop with usb sound card.(7.1+spdif in+line in).
you have:
-the power
-low consumption
-remote
-multisource
-and a keyboard+screen to type your message on DIY audio while listenning.

-(and lot a flexibility to load/create impulse response/measure)

total consumption with a 3 stereo (8000+4000+1000 taps) IRs
about 15 Watts (estimated).

what a basic laptop is able to convolve (91% cpu load on 1 core)
18 stereo convolution of 8192 taps (FFT of course Click the image to open in full size.)

Click the image to open in full size.

Last edited by thierry38efd; 5th January 2013 at 09:18 PM.
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Old 5th January 2013, 09:25 PM   #272
more10 is offline more10  Sweden
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I understand the fear of PC based crossovers (I've had the problem), but the Behringer? Never had one output what it was not supposed to, ever. You can set it wrong, but you can do that with a DSP board.

What's your experience?
I had a Behringer CX 3400 Pro. It broke down the first nigth I used it (MTBF 6 hours). Possibly a heatsink broke loose because of the vibrations and square or triangular waves was output. It was in a rack next to my 4530. Party sound levels.
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Old 5th January 2013, 09:38 PM   #273
more10 is offline more10  Sweden
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What sort of source is that? As I understood it, I2S is for linking together ICs on the one PCB, or in the same box at least.
It could be a USB interface like miniStreamer or an ADC.

The important thing here is that I2S is a proper protocol with a clock, not a Mickey Mouse protocol like spdif where you need to extract the clock from the signal.

I will put all the cards in the same box.
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Old 5th January 2013, 10:37 PM   #274
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Quote:
Originally Posted by twest820 View Post
Which problems, specifically? I'm familiar with the motivation for apodizing filters but my experience is subjective audio quality tends to be dominated by the impulse ringing introduced by the filter. Since the Fourier relation mandates an abrupt transition in the frequency domain is a slow one in the time domain brickwall is often a worst case choice in this regard---it's my experience a more even tradeoff of frequency and time domain behavior sounds better. Upsampling from 44.1 mitigates the issue but requires caution as most sample rate conversion defaults to brickwall antialiasing (miniDSP is unlikely to document their implementation on the miniSHARC---someone'll have to get one and measure it---but Analog's ASRC implementations are all brickwall so I'd expect miniDSP's to also be brickwal), as do the majority of DACs. So adding a third brickwall in one's own filtering is an interesting choice.
You would agree that to do the downsampling, for example from 48khz to 16khz, you first have to remove any frequency above the Nyquist frequency of the targeted sampling freq.
So you first have to apply a LP filter that reject anything above 8khz low enough (lets say -100dB), and then take 1 sample out of 3.

I said "brickwall", but of course with the shortest possible impulse it will end up much more like a gentle filter (but this will reduce the usable range).

Here are two examples of such filters, generated with rephase for a 48khz initial sampling freq, targeting a -100dB rejection at 8khz.

They were generated by trial and error

First example is with a 128 taps LP filter, and you get a usable range up to 6khz.
Click the image to open in full size.

Good enough if you are planning to use a 48dB/oct LP at 2000Hz for example, as it end up "brickwall" (and possibly ringing) or at least steeper than expected -80dB down the target slope (albeit this could also be taken into account with EQ...).

Now with a 64 taps LP filter, the slope is more gentle and you end up with a usable band up to 5khz.

Click the image to open in full size.

With the same -80dB target you can do a 48dB/oct LP at ~1500Hz.

Now with the first example (128 taps), lets calculate the available power that would get us. Lets pretend we have 1024 taps available at 48khz, and just count naively for the sake of an example.
After that first 128taps convolution we might have 896 taps "left".

After the filter you get 1 sample out of 3 and do the convolution on that.
So for each sample you have 3 times the processing time: 896 => 2688.

And then we are applying the convolution on a 16khz signal, so we have basically the same filtering capabilities than if we had 3 times the taps at 48khz (8064 taps, that is!).
That means that be can for example do a 48dB/oct HP filter at 80Hz without a stretch:

Click the image to open in full size.

The final implied delay with direct convolution will be 1.33ms for the initial 128taps filter + 84ms for the 2688taps@16khz linear-phase filter.
So the overhead is only 1.33ms (and half that with th 64 taps scenario), nothing compared to an FFT based convolution...

If you add the possibility of freely assigning the taps accross channels I think any real world 8-way crossover could be generated with a single miniSHARC and direct convolution...
Attached Images
File Type: png rePhase 16kHz downsampling 128taps.PNG (63.6 KB, 230 views)
File Type: png rePhase 16kHz downsampling 64taps.PNG (63.7 KB, 225 views)
File Type: png rePhase 16kHz downsampling HP example.PNG (65.1 KB, 224 views)

Last edited by pos; 5th January 2013 at 10:39 PM.
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Old 6th January 2013, 12:40 AM   #275
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Quote:
Originally Posted by pos View Post
You would agree that to do the downsampling, for example from 48khz to 16khz, you first have to remove any frequency above the Nyquist frequency of the targeted sampling freq.
Thanks for clarifying. It's not actually necessary to antialias before downsampling. In particular, polyphase filters filter after decimation and are commonly used in multirate filtering like the decimated mid/woofer/subwoofer/whatever structure we're discussing here. I've not coded polyphase myself but in the examples I'm aware of it saves on multiplies but not adds compared to a conventional FIR. Useful for minimizing ASIC die area or fitting more filter into an FPGA but, I suspect, probably not so useful for getting more out of a processor whose instruction set includes single cycle MAC. Could be worth a look, though.

Another multirate implementation that's long looked interesting to me is IIR XO followed by decimation. This is mainly an LR6 or LR8 kind of thing (or B7 or higher order) to get good image rejection but in a typical three way crossed around 200Hz and 2kHz it allows downsample by 2 on the mid and by 20 or so on the sub. Like FIR this could benefit from polyphase IIR to move the filtering post decimation but IIR is pretty cheap already.

Unfortunately multirate doesn't play nicely with multichannel DACs and most audio DACs that are interesting for use with hi-fi aren't specified for use with sampling rates below 32kHz. It's probably fine but I would chacterize the DACs one intends to use at low sampling rates before investing heavily in multirate. (There are other, more minor, issues too; clock generation becomes more involved as can cable routing.)

Last edited by twest820; 6th January 2013 at 12:42 AM.
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Old 6th January 2013, 01:37 AM   #276
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Originally Posted by more10 View Post
It could be a USB interface like miniStreamer or an ADC.

The important thing here is that I2S is a proper protocol with a clock, not a Mickey Mouse protocol like spdif where you need to extract the clock from the signal.

I will put all the cards in the same box.
Am I correct in thinking that Ministreamer cannot do asynchronous, however? (there may be products that can do it..?)

Forum | miniDSP

And is there a significant difference between I2S and SPDIF anyway, except for the clock being on a separate wire from the data? Once you introduce a less-than-perfect interconnect you have the same jitter problem regardless of which you use. Short PCB traces between chips introduce one level of jitter, and cables and connectors presumably are a little worse. The problem may not be significant, but it's another imaginary worry.

Can't you get your DSP board to do the asynchronous transfer from the source itself via USB or some such?

(but it's these sorts of headaches that I had in mind when being smug about the PC earlier :-) )
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Old 6th January 2013, 03:26 AM   #277
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I use rePhase for creating linear phase Linkwitz-Riley 48dB/oct 80Hz high/low pass filters for crossover, sample rate 16kHz and 2688 taps. These I import into Cool Edit for inspection. Summation of filters produces ripple in tails:

ripple time.png

This is completely avoidable if high pass filters are all derived from low pass counterpart via single sample inversion (subtractive filter).
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Old 6th January 2013, 10:47 AM   #278
more10 is offline more10  Sweden
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Am I correct in thinking that Ministreamer cannot do asynchronous, however? (there may be products that can do it..?)
You are correct. Thanks for finding that info.

WaveIO Asynchronous USB-to-I2S interface is a better choice.

Of course there is a risk interconnecting equipment not using differential signalling. Grounding will be a major headace. The picture of the Wave device says "isolated I2S outputs", very clever marketing. I will be looking for DACs marked "isolated I2S inputs".

I like your persistence CopperTop! I could do an Arch Linux based filter using BruteFIR for example. Using an ASUS Xonar D2X for input/output. It will be fun doing it. The Intel® Xeon® Processor E3-1220LV2 has only 17W TDP, it can be passively cooled. The E3-1265LV2 is 45W. It will need a slow fan.
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Old 6th January 2013, 10:51 AM   #279
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Their USBStreamer is async and multi-channel.
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Old 6th January 2013, 11:31 AM   #280
more10 is offline more10  Sweden
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Originally Posted by TheShaman View Post
Their USBStreamer is async and multi-channel.

http://www.minidsp.com/images/docume...SBstreamer.pdf

Quote:
I2S format for input/outputs: 24 bits I2S master
Supported sample rate: 44.1/48/88.2/96/176.4/ 192kHz
You cannot run it as slave though.
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