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I am aware of cheap DSPs , such as the Minidsp,  and the Parks Audio Puffin, that some people have posted about.   While a bit loath to spoil a pure valve analogue audio chain, I am also interested in experimenting.  The Puffin is very interesting, but clearly a phono preamp.  I was interested in something more generic.

 

So, first the Minidsp (various models - but starting with a cheap one for a couple of hundred).  I see it being used as an active  crossover, and I think I read about using it for room correction.   Can it be used in other ways - such as a de-click / pop remover for vinyl (as I believe the Puffin can now) ?   Can users write their own plugins or program it in other custom ways?   

 

If not, are their any other devices?  I see some talk of a raspberry Pi with a Hifiberry hat that has an ADC and DAC.  I want to look into the cost and flexibility of that. Has anyone tried one?

 

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I have been down this path for some years now. I got a colleague to build a DIY minisharc. They are great units but built to a price. So I am changing tack and going down the path of the Ultimate Prea

That depends on how low in frequency you want to correct.   Filters are specified in "taps".   The higher the number of taps, the better frequency resolution is possible (and the lower in fr

No, that isn't a FIR filter.   That's a fancy way to specificy an IIR filter.   This has a go at explaining:FIR vs IIR filtering

17 minutes ago, aussievintage said:

I am aware of cheap DSPs , such as the Minidsp,  and the Parks Audio Puffin, that some people have posted about.   While a bit loath to spoil a pure valve analogue audio chain, I am also interested in experimenting.

 

I use a miniDSP 10x10HD as the heart of my system - providing 4-way active XOs plus LF room EQ.

 

I was worried that moving to a digital XO (from many years with an analogue active XO) would result in a degraded vinyl experience.  I couldn't hear it ... but, certainly, now I use an external A2D converter (rather than the ones built into the miniDSP) and use my miniDSP at 96kHz, in its "digital in ... analogue out" mode, the sound has gone up a level.  :)

 

Quote

 

  The Puffin is very interesting, but clearly a phono preamp.  I was interested in something more generic.

 

 

However, it seems like a very neat device.  :)

 

Quote

So, first the Minidsp (various models - but starting with a cheap one for a couple of hundred).  I see it being used as an active  crossover, and I think I read about using it for room correction.   Can it be used in other ways - such as a de-click / pop remover for vinyl (as I believe the Puffin can now)?

 

Yes, they all do room EQ; de-click / pop remover - not that I'm aware.

 

Quote

 

Can users write their own plugins or program it in other custom ways?

 

 

No, you use the plug-in designed for your particular model.

 

In terms of "custom programming" - yes, if you know how to design 'FIR" filters, you can enter this data into the miniDSP, instead of using the default "IIR" filters.  (I'm not that smart!  :( )

 

Good luck,

Andy

 

 

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6 minutes ago, andyr said:

I use a miniDSP 10x10HD as the heart of my system - providing 4-way active XOs plus LF room EQ.

 

I knew you were using one, yep.

 

6 minutes ago, andyr said:

However, it seems like a very neat device.  :)

It does seem to be well thought out.

8 minutes ago, andyr said:

they all do room EQ; de-click / pop remover - not that I'm aware.

7 minutes ago, andyr said:

No, you use the plug-in designed for your particular model.

 

In terms of "custom programming" - yes, if you know how to design 'FIR" filters, you can enter this data into the miniDSP, instead of using the default "IIR" filters.  (I'm not that smart!  :( )

 

That's what I thought.  I want something that can do other stuff.  Another example, other than declick, might be to add some reverb (not that I'd normally do that in a hifi situation), or intelligent muting between tracks.  Just to experiment.

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If you can find one, a WAF Najda DSP is a good thing for the money - very flexible. 

 

Companies like Danville Signal do DSP boards that are freakishly powerful though programming them isn't as plug/play as MiniDSP. You need tools and knowledge - tools (e.g. AudioWeaver) have incredible possibility but are not as trivial or as easy to use as what MiniDSP offers.

 

You could also simply do it out of a PC natively - everything from nice programs (HQ etc) to backends in Linux (Ecasound etc) that do it transparently.

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22 minutes ago, rmpfyf said:

Ecasound

Just looked that up, and reading about it.  Haven't yet found an example that processes live input and outputs it realtime.  Looks like it might be able to do it though.  I like how it can use LADSPA and LV2 plugins.  

 

24 minutes ago, rmpfyf said:

nice programs (HQ etc)

What is HQ ?

 

32 minutes ago, rmpfyf said:

Companies like Danville Signal do DSP boards that are freakishly powerful though programming them isn't as plug/play as MiniDSP. You need tools and knowledge - tools (e.g. AudioWeaver) have incredible possibility but are not as trivial or as easy to use as what MiniDSP offers.

The dspMusik 2/8  with Audio Weaver looks like an ideal solution.  I am finding it hard to find a price - but I suspect it is too expensive for my first experiments.

 

 

 

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Using mini dsp as a phono preamp is possible https://www.minidsp.com/forum/hometheater-av-applications/12510-2x4-hd-as-phono-preamp but it's more I can do it kind of way rather that it is good. removing click or pops not possible cos it's firmware locked and you don't have access to the mips processor. If you want to do all that you have to go FPGA way. I would skip any linux device for doing both ADC and DAC at the same time. You can do one well but not both. Theoretically you can have 2 linux device one doing the adc and one doing a dac and transfer data through network and if you have the correct buffer timing issue can be minimised. 

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3 minutes ago, mloutfie said:

Using mini dsp as a phono preamp is possible https://www.minidsp.com/forum/hometheater-av-applications/12510-2x4-hd-as-phono-preamp but it's more I can do it kind of way rather that it is good. removing click or pops not possible cos it's firmware locked and you don't have access to the mips processor. If you want to do all that you have to go FPGA way. I would skip any linux device for doing both ADC and DAC at the same time. You can do one well but not both. Theoretically you can have 2 linux device one doing the adc and one doing a dac and transfer data through network and if you have the correct buffer timing issue can be minimised. 

Appreciate the input,   but to say you can't do input and output at the same time in linux really surprises me????   Ardour is a really great DAW  that runs under linux.

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7 minutes ago, aussievintage said:

Just looked that up, and reading about it.  Haven't yet found an example that processes live input and outputs it realtime.  Looks like it might be able to do it though.  I like how it can use LADSPA and LV2 plugins.  

Pretty sure you can pipe through ALSA (been a while) - though if you're happy doing FIR's, BruteFIR will do this too and is very computationally lean.

 

9 minutes ago, aussievintage said:

What is HQ ?

HQPlayer

 

9 minutes ago, aussievintage said:

The dspMusik 2/8  with Audio Weaver looks like an ideal solution.  I am finding it hard to find a price - but I suspect it is too expensive for my first experiments.

When I last checked closer to $2k for a fully spec'd item, though they're going to release an updated, modular system (dspNexus) that can be expanded between 8 and 16 channels, and has AKM 4495 and 4495 stereo DAC modules (user selectable/configurable).

 

A bit blue for my blood tho. I'm saving for an Okto DAC8 Pro, filters on the PC, passthrough AES to a MiniDSP for the subs.

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5 minutes ago, rmpfyf said:

Pretty sure you can pipe through ALSA (been a while) - though if you're happy doing FIR's, BruteFIR will do this too and is very computationally lean.

 

Yep.  Found this

 

ecasound -c -f:16,2,44100 -a:1 -i /dev/dsp0 -o /dev/dsp3 -ev

 

Basicly this just records from one OSS input, puts the signal through an analyze (-ev) effect and outputs to an OSS output. The secret here is that you can get volume statistics with the estatus (or es) command in interactive mode. Qtecasound also offers a estatus pushbutton. This way you can adjust the mixer settings, check the statistics (after which they're reseted), adjust again, check statistics, ... and so on. Newer ecasound versions (1.8.5 and newer) come with 'ecasignalview', which is a standalone app that can monitor signal level in realtime.

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31 minutes ago, mloutfie said:

Using mini dsp as a phono preamp is possible https://www.minidsp.com/forum/hometheater-av-applications/12510-2x4-hd-as-phono-preamp but it's more I can do it kind of way rather that it is good

That was worthwhile reading. The idea occurs to me that, even if you don't build a "buffer" amplifier to go in front of it,  you could put the minidsp between any standard RIAA phono preamp and the rest of the system,  then using a similar technique, modify the RIAA response to obtain the other curves for a multi-eq phono preamp.

 

In addition, I imagine you can do low freq rumble filtering,  high freq. scratch filtering,  and all sorts of other eq tweaks like the Puffin does, to tweak your cartridge's sound to your liking ?

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30 minutes ago, rmpfyf said:

When I last checked closer to $2k for a fully spec'd item,

That's the sort of price I expected, just looking at it.

 

I might start small, and connect an rPi to a small Yamaha recording interface I have, and have a play with ecasound, and various front-ends it has.  There's also something called Puredata which I saw someone using on a Pi.

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First experiment successful

 

- loaded up a new image of raspbian in a Pi 3B 

- plugged the Yamaha USB audio interface

- installed and ran Audacity and recorded and played back with this device.

- installed ecasound

 

I was then able to chain the audio using ecasound in realtime from input straight through to output.  I didn't try to use any processing on the audio, just sent it straight back out. 

 

 Only problem I foresee so far is to keep the sound free of computer noise.  With the Pi running bare on the desk, using power from my laptop USB ports,  and the sound source being my phone headphone socket plugged into the Yamaha with cheap RCA leads,  there was some background computer noise.   Using a separate USB charger for power removed a lot of that.  I can see a metal case, clean power, and the usual good practice with shielding will be very necessary doing this stuff.

 

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I have been down this path for some years now. I got a colleague to build a DIY minisharc. They are great units but built to a price. So I am changing tack and going down the path of the Ultimate Preamp. 

 

The guy is extremely knowledgeable and has built a very very good product which is modular in design. Furthermore he is representative of the quality home grown products that this country is renown for. 

 

See - 

 

https://analog-precision.com/home/upp/

 

 

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6 hours ago, aussievintage said:

First experiment successful

 

- loaded up a new image of raspbian in a Pi 3B 

- plugged the Yamaha USB audio interface

- installed and ran Audacity and recorded and played back with this device.

- installed ecasound

 

I was then able to chain the audio using ecasound in realtime from input straight through to output.  I didn't try to use any processing on the audio, just sent it straight back out. 

 

 Only problem I foresee so far is to keep the sound free of computer noise.  With the Pi running bare on the desk, using power from my laptop USB ports,  and the sound source being my phone headphone socket plugged into the Yamaha with cheap RCA leads,  there was some background computer noise.   Using a separate USB charger for power removed a lot of that.  I can see a metal case, clean power, and the usual good practice with shielding will be very necessary doing this stuff.

 

Boom! Awesome.

 

1 hour ago, ghost4man said:

I have been down this path for some years now. I got a colleague to build a DIY minisharc. They are great units but built to a price. So I am changing tack and going down the path of the Ultimate Preamp. 

 

The guy is extremely knowledgeable and has built a very very good product which is modular in design. Furthermore he is representative of the quality home grown products that this country is renown for. 

 

See - 

 

https://analog-precision.com/home/upp/

 

 

I love the spec on that box - pricey against the Danville Box which does the same thing in an AKM flavour, though local support counts for much.

 

The Dev's previous servobass DSP solution looked pretty special too. 

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Moving along nicely.

 

I have been able to use LADSPA plugins, filters etc.  Even added some more plugins I downloaded.  Discovered the download was a sourcefile, but compiled easily enough.

 

Added a second USB device, a FIIO headphone amp/DAC, and the sound is very nice (also quiet).  No problem routing sound in from one device and out to the other.   I have connected a simple vinyl rig now so I can play with eq, scratch filters etc.

 

Not happy with any of the GUI frontends for ecasound (that I have found so far anyway), so all command line at present.  That's a good thing anyway.

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15 minutes ago, aussievintage said:

Moving along nicely.

 

I have been able to use LADSPA plugins, filters etc.  Even added some more plugins I downloaded.  Discovered the download was a sourcefile, but compiled easily enough.

 

Added a second USB device, a FIIO headphone amp/DAC, and the sound is very nice (also quiet).  No problem routing sound in from one device and out to the other.   I have connected a simple vinyl rig now so I can play with eq, scratch filters etc.

 

Not happy with any of the GUI frontends for ecasound (that I have found so far anyway), so all command line at present.  That's a good thing anyway.

 

Wow. Your progress is making me look like what I am - a master procrastinator.

 

When you say 2nd USB device - are you outputting to 2 USB devices simultaneously? I'm about to start on this much with 2x stereo DACs (yes, timing issues etc) for some DSP filter testing.

 

The GUIs were rubbish years ago, interesting to hear they're still rubbish now :D 

 

What kind of sample rates are you pushing through the Pi? Any idea what kind of utilisation you're getting?

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2 hours ago, rmpfyf said:

 

Wow. Your progress is making me look like what I am - a master procrastinator.

 

When you say 2nd USB device - are you outputting to 2 USB devices simultaneously? I'm about to start on this much with 2x stereo DACs (yes, timing issues etc) for some DSP filter testing.

 

No, inputting from one USB device (Yamaha Audiogram6 mixer/recording interface), and outputting to another different USB Fiio E10K headphone amp/DAC.

 

2 hours ago, rmpfyf said:

The GUIs were rubbish years ago, interesting to hear they're still rubbish now :D 

Partly because they are aimed more at using ecasound as a DAW than as a realtime processor, and partly because they have been updated and it gets tricky with old software versions.  Most run on top of Ruby or Tcl or Python which adds more confusion.

 

2 hours ago, rmpfyf said:

What kind of sample rates are you pushing through the Pi? Any idea what kind of utilisation you're getting?

Running at 16-bit 44.1/48 kHz input due to limitation of the interface.

 

The snapshot below was with ecasound running a simple low pass filter.  Just a couple of percent.

ecasound -f s16_le,2,48000  -i alsahw,3,0  -o alsahw,2,0 -el:lpf,5000 -c --daemon

 

image.png.f5f450002b6cf69569f51b387996a2e3.png

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On 03/06/2020 at 11:10 AM, aussievintage said:

So, first the Minidsp (various models - but starting with a cheap one for a couple of hundred).  I see it being used as an active  crossover, and I think I read about using it for room correction. 

Some of their cheap ones can make FIR filters, which is what you typically want for "room correction".

 

The more expensive ones can run long FIR filters (what you need for <100Hz correction), or can run Dirac Live.

 

On 03/06/2020 at 11:10 AM, aussievintage said:

Can it be used in other ways - such as a de-click / pop remover for vinyl (as I believe the Puffin can now) ?   Can users write their own plugins or program it in other custom ways?   

No.

On 03/06/2020 at 11:10 AM, aussievintage said:

If not, are their any other devices?  I see some talk of a raspberry Pi with a Hifiberry hat that has an ADC and DAC.  I want to look into the cost and flexibility of that. Has anyone tried one?

Sure.  If you use a computer with ADC/DAC, you can do anything you can find software for.

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On 03/06/2020 at 11:26 AM, andyr said:

Yes, they all do room EQ

This depends on your definition.

 

They can all do "PEQ".

 

They do no all do FIR filters (which is what "room EQ" systems typically use).   They do not all do long FIR filters (needed for LF "room EQ").

On 03/06/2020 at 11:26 AM, andyr said:

 

In terms of "custom programming" - yes, if you know how to design 'FIR" filters, you can enter this data into the miniDSP, instead of using the default "IIR" filters.  (I'm not that smart!  :( )

... but not into all of them.   For example.  Not into your 10x10HD model.

 

Th 2x4HD model, yes.... but only short filters (not suitable for "room EQ", at least not to any significantly low frequency).

 

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23 hours ago, aussievintage said:

That's the sort of price I expected, just looking at it.

 

I might start small, and connect an rPi to a small Yamaha recording interface I have, and have a play with ecasound, and various front-ends it has.  There's also something called Puredata which I saw someone using on a Pi.

What I'd recommend really depends on what you want to achieve.

 

If you want to do something which needs a computer.... then, do that.

 

Otherwise, if you're wanting to do active crossovers and other filters..... then something like the miniDSP (pay attention to gain structure) is much more accessible.

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4 minutes ago, davewantsmoore said:

What I'd recommend really depends on what you want to achieve.

 

If you want to do something which needs a computer.... then, do that.

 

Otherwise, if you're wanting to do active crossovers and other filters..... then something like the miniDSP (pay attention to gain structure) is much more accessible.

The former.  As you have probably read by now, I am playing with a simple rPi setup at the moment.

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Wow,  Found this thing called Puredata.   It's a graphical programming language for audio

 

 

One quick simple "program" and I have the sound looping through, just like with ecasound

 

eimage.png.59ba477b07573604041dc71eff2b823b.png

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39 minutes ago, davewantsmoore said:

They do not all do long FIR filters (needed for LF "room EQ").

Can you define how long a "long filter" is and how is it measured?

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1 hour ago, davewantsmoore said:

They do not all do FIR filters (which is what "room EQ" systems typically use).   They do not all do long FIR filters (needed for LF "room EQ").

... but not into all of them.   For example.  Not into your 10x10HD model.

 

 

Dave, the software for my 10x10HD allows me to select either 'basic' or 'advanced' filters.  If you select:

  • 'basic' - you then get the options of BW, L_R or Bessel ... at various slopes.
  • 'advanced' - you have to enter BiQuads.  I thought this was what a 'FIR' filter required?

 

Andy

 

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4 hours ago, frednork said:

Can you define how long a "long filter" is and how is it measured?

That depends on how low in frequency you want to correct.

 

Filters are specified in "taps".   The higher the number of taps, the better frequency resolution is possible (and the lower in frequency you can design filters for).

 

 

For example.   1000 taps, is going to get you around 50Hz frequency resolution .... meaning there is one sample point ever 50Hz in your filter.    At 1000Hz, this isn't a big problem ..... but at low frequencies this is terrible.

 

I think the minidsp 2x4HD has about 1000 taps per channel .... but this resolution is fine, as it's just intended for linear phase crossovers and typically crossover operate a relatively high frequency.....   The OpenDRC MiniDSP has more than 6000 per channel.

 

 

All that being said.... any device marketed for "room correction" using FIR filters, will have all of this sorted out for the user   (which is why the miniDSP 2x4HD doesn't market itself for "room correction").

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4 hours ago, andyr said:

'advanced' - you have to enter BiQuads.  I thought this was what a 'FIR' filter required?

No, that isn't a FIR filter.

 

That's a fancy way to specificy an IIR filter.

 

This has a go at explaining:FIR vs IIR filtering

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15 hours ago, davewantsmoore said:

That depends on how low in frequency you want to correct.

 

Filters are specified in "taps".   The higher the number of taps, the better frequency resolution is possible (and the lower in frequency you can design filters for).

 

 

For example.   1000 taps, is going to get you around 50Hz frequency resolution .... meaning there is one sample point ever 50Hz in your filter.    At 1000Hz, this isn't a big problem ..... but at low frequencies this is terrible.

 

I think the minidsp 2x4HD has about 1000 taps per channel .... but this resolution is fine, as it's just intended for linear phase crossovers and typically crossover operate a relatively high frequency.....   The OpenDRC MiniDSP has more than 6000 per channel.

 

 

All that being said.... any device marketed for "room correction" using FIR filters, will have all of this sorted out for the user   (which is why the miniDSP 2x4HD doesn't market itself for "room correction").

 

Dave any idea why they don't do the MiniSHARC anymore? I just managed to get a piece for this very reason (was going to do a 2x8 kit but wanted FIRs), though it doesn't seem to be current production anymore... but it's a nice bit of kit?

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3 minutes ago, rmpfyf said:

 

Dave any idea why they don't do the MiniSHARC anymore? I just managed to get a piece for this very reason (was going to do a 2x8 kit but wanted FIRs), though it doesn't seem to be current production anymore... but it's a nice bit of kit?

I wasn't aware of this.... They let you put it in your shopping cart.... but it is listed under the EOL products page too.

Confusing.

 

 

Perhaps some of the reason is the taps per channel = 2048.

 

This isn't enough to run a very long FIR filter.... so you can't really correct the bass, ie. run any sort of "room correction" type filter.

 

Using FIR to correct higher frequencies is a very big trap IMVHO.   Speakers are minimum phase.   If we think a speaker needs a FIR filter (ie. you want to manipulate the phase independantly of the amplitude), then we have probably made a "mistake" somewhere in the design or measurement.

 

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Actually.... perhaps I take that back, somewhat (after reading the manual again).

 

MiniSHARC will do two channels at 48khz with 6000+ taps per channel.... which is something like enough to correct the full frequency range.

 

I think I had it stuck in my head somewhere that "the taps we not reassignable" ... and so this wouldn't be possible.

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5 minutes ago, davewantsmoore said:

Actually.... perhaps I take that back, somewhat (after reading the manual again).

 

MiniSHARC will do two channels at 48khz with 6000+ taps per channel.... which is something like enough to correct the full frequency range.

 

I think I had it stuck in my head somewhere that "the taps we not reassignable" ... and so this wouldn't be possible.

 

OpenDRC plugin - seems a reasonable way out for bass work. 

 

Can slave a second one under a common controller on the higher channel count plugins and make quite a nice multiway system with display/remote/etc for a not-unreasonable sum... 96kHz isn't leading in the space but then again neither's the budget.

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@Ittaku    I have this all setup running ecasound with the ladspa-dsp plugin, configured for a simple low pass filter to test.  I feel I am only a step away from doing full room correction on a rPi.  This wasn't a goal when I set out, but I do find it interesting, and I like pushing an rPi to it's limits.

 

Could you supply a effects chain config for me to try and see the load it causes etc?  What does it use, just a FIR filter?

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1 minute ago, aussievintage said:

@Ittaku    I have this all setup running ecasound with the ladspa-dsp plugin, configured for a simple low pass filter to test.  I feel I am only a step away from doing full room correction on a rPi.  This wasn't a goal when I set out, but I do find it interesting, and I like pushing an rPi to it's limits.

 

Could you supply a effects chain config for me to try and see the load it causes etc?  What does it use, just a FIR filter?

Just a fir filter, and you can make the low pass filter or any other filter all part of the room correction filter too, so it's all just loading one FIR filter.

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16 hours ago, davewantsmoore said:

For example.   1000 taps, is going to get you around 50Hz frequency resolution .... meaning there is one sample point ever 50Hz in your filter.    At 1000Hz, this isn't a big problem ..... but at low frequencies this is terrible.

 

I think the minidsp 2x4HD has about 1000 taps per channel .... but this resolution is fine, as it's just intended for linear phase crossovers and typically crossover operate a relatively high frequency.....   The OpenDRC MiniDSP has more than 6000 per channel.

Ok thanks, so higher taps means better resolution?Is there a notional point at which a higher number of taps has no benefit? I have generated filters which have close to a million taps when using rephase are there other benefits to longer filters?

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Yes.

 

It means a longer filter ... and a longr filter means more points on the filter curve (vs frequency).

 

As mentioned, if you have a data point for your filter curve spaced every 50Hz .... than at 1khz, this is totally fine.....  but at bass frequencies this is as little as one data point per octave.

 

12 minutes ago, frednork said:

Is there a notional point at which a higher number of taps has no benefit? I have generated filters which have close to a million taps when using rephase are there other benefits to longer filters?

Higher frequency resolution.

 

In RePhase, once you have designed a filter, you change the settings to see the result with a given number of taps.   Try it, you will see what I am talking about.    For a good example, make some amplitude changes around 30Hz, and/or a highpass filter at low frequencies.    They will look perfect with a very high number of taps.... and as you reduce down to 5000, 1000, 100, etc.  You will see what happens.

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1 hour ago, Ittaku said:

Just a fir filter, and you can make the low pass filter or any other filter all part of the room correction filter too, so it's all just loading one FIR filter.

 

OK, I am new to this.  So  I have this doco

 

Quote

fir [~/]impulse_path
Non-partitioned 64-bit FFT convolution. Latency is equal to the length of the impulse.

 

What does it expect to find in the impulse_path file ?

 

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1 hour ago, frednork said:

Ok thanks, so higher taps means better resolution?Is there a notional point at which a higher number of taps has no benefit? I have generated filters which have close to a million taps when using rephase are there other benefits to longer filters?

This is good reading

https://www.prosoundtraining.com/2016/05/20/fir-ward-thinking-part-5/

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2 hours ago, aussievintage said:

What does it expect to find in the impulse_path file ?

A wav file. I use 32 bit mono wav files for mine at the playback resolution (176 or 192 or whatever). You'll need to use something like rephase to generate one.

https://rephase.org/

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20 minutes ago, Ittaku said:

A wav file. I use 32 bit mono wav files for mine at the playback resolution (176 or 192 or whatever). You'll need to use something like rephase to generate one.

https://rephase.org/

 

Thanks,  I tried it - nice.     I must be having some weird permissions problem.  ladspa_dsp is reporting "failed to open" on anything I point it at in it's config file

 

# This is a comment
input_channels=1
output_channels=1
LC_NUMERIC=C
# effects_chain=gain -3.0 lowpass 2000 .7
effects_chain=gain -3.0 fir impulse.txt

 

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Just says

 

ladspa_dsp: fir: error: failed to open impulse file: /home/pi/.config/ladspa_dsp/impulse.txt
ladspa_dsp: error: failed to initialize effect: fir
 

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