rePhase, loudspeaker phase linearization, EQ and FIR filtering tool

[pos]

That will be the first plugin.
Last news they wanted to stick to 96khz, which with some power kept for IIR biquads will leave only 512 taps per channel (which is really not much at that sampling freq!).
After some discussions on their forum the plugin seems to be delayed (was supposed to be before christmas), so I hope they will go down to 48khz as the main sampling freq (1024 taps at 48khz is already quite a bit better) and allow for taps distribution among channels for the first version of the plugin (not need for 1024 taps for a non-brickwall crossover above a few hundreds of Hz...).

Further versions could implement downsampling of some channels and/or FFT convolution, time will tell...

[john k...]

Originally Posted by more10

Coding is fun?

For a 4 way stereo filter that means 1500 taps. Is 1500 taps enough?

Simple answer, No. No way if you are doing anything below 500 Hz (guesstimate on the lower frequency limit) .

[john k...]

Originally Posted by pos

That will be the first plugin.
Last news they wanted to stick to 96khz, which with some power kept for IIR biquads will leave only 512 taps per channel (which is really not much at that sampling freq!).
After some discussions on their forum the plugin seems to be delayed (was supposed to be before christmas), so I hope they will go down to 48khz as the main sampling freq (1024 taps at 48khz is already quite a bit better) and allow for taps distribution among channels for the first version of the plugin (not need for 1024 taps for a non-brickwall crossover above a few hundreds of Hz...).

Further versions could implement downsampling of some channels and/or FFT convolution, time will tell...

512 tap at 96k. Are they kidding? That basicly useless. 1024 at 48k isn't going to do much at low frequency for any kind of eq either. You can't down sample if yuo are trying eq a full range signal, say for room eq.

[twest820]

Let's just say the software sitting atop the miniSHARC seems primitive and the pace of development slow---I would tend to bet on Hypex porting Grimm's computationally efficient FIR phase correction to the DLCP first though the pace there is slow as well (but the coding's better). One reason why DIY with an M4 or SHARC eval board could be attractive if one doesn't want a PC in the loop (though doing one time EQ and phase correction offline and just playing back the equalized track with IIR XO can avoid quite a lot of fuss about real time processing power---perhaps I should do a Cross Time DSP DCR to support rePhase impulse files ).

[Barleywater]

Originally Posted by john k...

512 tap at 96k. Are they kidding? That basicly useless. 1024 at 48k isn't going to do much at low frequency for any kind of eq either. You can't down sample if yuo are trying eq a full range signal, say for room eq.

Brute force with PC CPU is quite doable, but don't sell DSP chips short. DCX2496 may not do lots of taps, but good sample rate conversion management covers a lot of ground.

Example:

Loopback of DCX channel with soundcard is used to capture IR of PEQ set to 47Hz, gain 9.5dB and Q=10.0. Capture done at fs 48kHz results in IR with >24k effective sample length, thus >48k samples active in DCX 96kHz sample rate output.


Future is bright with continuing development.

[more10]

Chapter 28: Digital Signal Processors How Fast are DSPs?:

FFT convolution can be carried out in about 60 clock cycles per sample.

Chapter 18: FFT Convolution:

With FFT convolution, the filter kernel can be made as long as you like, with very little penalty in execution time. For instance, a 16,000 point filter kernel only requires about twice as long to execute as one with only 64 points.

120 cycles per sample for a 16 k point filter.

A 300 MHz Sharc performs 1500 cycles per 192 kHz sample. It should be possible to filter 12 channels (ignoring memory bottlenecks).

[CopperTop]

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.

[pos]

Originally Posted by john k...

512 tap at 96k. Are they kidding? That basicly useless. 1024 at 48k isn't going to do much at low frequency for any kind of eq either.

I posted some simulations on the miniDSP forum, in the hope they change their mind: Forum | miniDSP
(pic uploaded below for full resolution)

Originally Posted by john k...

You can't down sample if yuo are trying eq a full range signal, say for room eq.

So you want 8 channels of fullrange correction
That is a lot to ask to a $175 kit!

Another solution to spare some of these taps (in the active crossover scenario) is to do the filtering with the biquads and use the FIR only for phase linearization.
In that case you know you will always get the precision you want for the magnitude (and no ripples), and as the impulse is not symmetrical you will make a better use of the available taps with energy centering...

[more10]

Why use a specialised DSP board when a near-silent PC can be had for a couple of quid?

I am a programmer. I don´t trust PCs :-).

It will also be fun learning some DSP programming.

[CopperTop]

Originally Posted by more10

It will also be fun learning some DSP programming.

What's the difference between writing DSP on a PC and a dedicated DSP? I ask in all sincerity, because I have been busily writing my crossover software in 'C' for the last few months, and I thought I was doing DSP! Will you be writing in another language?