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Old 6th January 2013, 11:46 AM   #281
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Quote:
Originally Posted by Barleywater View Post
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:

Attachment 321808

This is completely avoidable if high pass filters are all derived from low pass counterpart via single sample inversion (subtractive filter).
Hi Andrew,

The ripples you are showing occur at -144dB. I don't think it should be considered as a problem as is.

As the HP and LP will likely have different length and window, iterative optimization will give different result, so a subtractive solution would probably not give the best result for a given impulse length...

As for the ripples you showed me starting post #9 (low level ripples at the Nyquist frequency), I have found a way to reduce them using oversampling (and double the number of taps) and subsequent downsampling (with amplitude derivation being corrected during the fist iterative optimization step).
Not sure I will implement it in the next release though, as I would like to focus on measurement import first.
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Old 6th January 2013, 11:50 AM   #282
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Originally Posted by twest820 View Post
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.)
Why not upsample to the nominal sampling freq before outputing to the DAC?
(interpolation might require an FFT though )
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Old 6th January 2013, 11:51 AM   #283
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Originally Posted by more10 View Post
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 haven't been following this thread. But what exactly do you mean by clever marketing? the wavio device has two sets of i2s outputs, 1 set is isolated by NVE IL715 isolator chip and are on pin headers (you provide 'dac side' power for the i2s output), the other set of i2s outputs is non-isolated and on u.fl connectors.

What exactly is 'marketing' about that? It's fact as far as I can tell?
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Old 6th January 2013, 12:33 PM   #284
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It is clever to put it on the board. Marketing is not all about lies :-)
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Old 6th January 2013, 02:42 PM   #285
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SHARC 21469 EZ-BOARD FOR THE ADSP-2146X SHARC FAMILY. It has the most powerful Sharc processor, ADSP-21469 (450 MHz, 5 mbit on chip sram) as well as a multichannel AD1939 DAC/ADC with balanced connectors (no i2S fiddling needed). There is also an SPDIF input. I believe this board is good enough to verify if it is possible to build a stereo 4 way filter using FIR convolution on the Sharc.
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Old 6th January 2013, 03:36 PM   #286
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Originally Posted by pos View Post
Hi Andrew,

The ripples you are showing occur at -144dB. I don't think it should be considered as a problem as is.

As the HP and LP will likely have different length and window, iterative optimization will give different result, so a subtractive solution would probably not give the best result for a given impulse length...

As for the ripples you showed me starting post #9 (low level ripples at the Nyquist frequency), I have found a way to reduce them using oversampling (and double the number of taps) and subsequent downsampling (with amplitude derivation being corrected during the fist iterative optimization step).
Not sure I will implement it in the next release though, as I would like to focus on measurement import first.
You may think it isn't problem, a simple one to fix. Calculation of high pass should be done buy calculating low pass and subtracting normalized value of one from peak sample. Additionally, linear phase version filters for IIR family is faster and more accurate to calculate IIR with 1/2 gain/slope, convolve with time reversed copy, trimming to desired tap count, and applying symmetrical fade in and fade out at ends.

Fare and away at moment is indeed import of user IR for response modeling, followed by solver for optimizing. Even REW approach is tedious compared to good technique with direct inversion.
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Old 6th January 2013, 05:00 PM   #287
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Hello there and congrats for the ideea,
But......many of are not able to figure out how exactrly to work with the app...sooo..
it there any tutorial for using this program with...foobar for example?
Thank you, and good luck with your project!
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Old 6th January 2013, 05:08 PM   #288
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Originally Posted by pos View Post
Why not upsample to the nominal sampling freq before outputing to the DAC?
Making the downsampled stop band wide enough to allow a slow rolloff antialiasing filter limits the downsampling ratio. Between that, the cost of the antialiasing, and the minor impulse blurring from the pre-DAC antialiasing round one likely ends up with a more complex implementation and little or no computational savings for three ways and a good fraction of four ways.

Downsample+upsample might let you fit a tweeter+mid+high sub+low sub four way into a DSP that's more of a three way size, though. That's the kind of thing which wants a case study. It's not something I've looked into as three ways are more than capable of delivering SPLs I find uncomfortably loud.
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Old 6th January 2013, 05:37 PM   #289
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Quote:
Originally Posted by Barleywater View Post
You may think it isn't problem, a simple one to fix. Calculation of high pass should be done buy calculating low pass and subtracting normalized value of one from peak sample. Additionally, linear phase version filters for IIR family is faster and more accurate to calculate IIR with 1/2 gain/slope, convolve with time reversed copy, trimming to desired tap count, and applying symmetrical fade in and fade out at ends.

Fare and away at moment is indeed import of user IR for response modeling, followed by solver for optimizing. Even REW approach is tedious compared to good technique with direct inversion.
Subtracting is not the way it is done in rePhase, so 100% complementarity cannot be guaranteed. But you know what you get when looking at the ripples, and frankly a -144dB complementarity error is not something that would worry me. How is the complementarity of the acoustic filters?...
You can also use more taps for the given frequency and these complementarity problems will go even deeper in level as the result curve will fit the theoretical one better (and theoretical filters generated in rephase are 100% LP/HP complementary).

The IIR=>FIR generation method using cooledit is good, but only work for textbook filters whereas here any function (for both magnitude and phase) can be generated (Linkwitz-Riley, Horbach-Keele, reject high/low, overlapping, etc...).

There are room for many tools and methods for FIR filtering (automated inversion, integrated tools, etc.), and rePhase is just one of them, with a specific approach (manual correction and wysiwyg result).
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Old 6th January 2013, 05:42 PM   #290
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Originally Posted by rvrazvan View Post
Hello there and congrats for the ideea,
But......many of are not able to figure out how exactrly to work with the app...sooo..
it there any tutorial for using this program with...foobar for example?
Thank you, and good luck with your project!
Hi rvrazvan,

Thanks for the kind words

It depends on what you want to do with rePhase: correct the phase of an existing loudspeaker? Correct both phase and magnitude? Build the whole crossover with rePhase?...

There are some tutorial online, for example Thierry's here, with a melting pot of all the possible approaches, with a focus on PC VST convolution:
EASY FIR crossover PC based+DRC

Here is Another one in French by Jimmy Thomas, explaining how to build an entier crossover with rePhase and Jriver:
http://jimmy.thomas.free.fr/Jriver/T...olmImpulse.pdf

If you read French you will find additional information along the way on this topic:
rePhase: linéarisation de phase, EQ et filtrage FIR - Enceintes

Explain you situation and I will see how I can help (and this might be the first step to another tutorial).
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