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I've been wanting to do a DIY looper for quite a while. I looked into the ISD-based loopers and although I like the slightly lo-fi sound, I just can't make do with no overdubbing possibilities. I was thinking it would be cool to work together to develop a looper project for everybody here that I know is interested in this. The major stumbling block at the moment is that I have almost no experience doing any coding. However, I'm a good learner and I have the motivation to make this happen.

I do need a little help from those of you who do know what they're doing.

The major thing is this: what kind of technologies would be ideal for this? I know of DSP, AVR, PIC, but I don't know which of these would have the best architecture for a looper.

That question is also tied into this one: what kind of looper do people want? At its most basic, we need to be able to press a button to start recording, and press it again to end the loop, start playback from the beginning, and begin overdubbing. I think around 40 seconds is a good aim for maximum record time, but others can speak up about what they want.

There are lots of other options that we could consider implementing. Having owned the ridiculously full-featured Echoplex Digital Pro, as well as the very different but very fun EHX 2880, I have lots of ideas of other things it can do. I'm guessing some people want a fairly simple looper for just overdubbing a couple of parts, and some people will want very complex loop windowing, different playback speeds, de-synchronous loops, etc. so it may be difficult coming up with a feature set to satisfy everyone.

My goal is to come up with a project that is easy to build and requires no programming for the DIYer. There would be pro double-sided boards with silkscreen and soldermask, and pre-programmed chips available, as well as maybe an entire kit for those who don't want to bother with sourcing parts.

So, what kind of processor should I be looking at?

I've done extremelly basic sampling/looping on AVR. it's 8-bit and around 12khz, but it works. I had two problems:

1. RAM space. I need fast RAM to store a stream of bytes really quickly. The internal AVR RAM is plenty fast, but super limited in storage. 2KB in the largest DIP chips. enough for a tenth of a second or so. Maybe enough for some basic granular effect or delay, but not for looping. Need fast external RAM, at least a couple 100KB, and I haven't found some.

2. Operation time. You need to be able to process an entire sample in less than 45 millionths of a second to achieve 22khz. This is no problem if all you're doing is recording and playing back, even with slower languages like Basic. But in my basic looper, as soon as I added the strict minimum to be able to control the samplerate speed and a couple other basic things, I nearly halved my top possible sampling rate on my 8MHz chip.

Disclaimer: I'm still a noob at this. I've only started playign around with micros in march.

I agree it would be nice to have something solid, though. The ISD chips are cheap and easy to use, but in my opinion they sound absolutelly rubbish. To my ears they sound the quarter of the rate/depth they claim to be, even in commercial products. Plus with all the added noise. I'm not a fan of them at all, but use them because that's all we have.

I'd kill for a chip like these but with at least 12-bit CLEAN sound at reasonable sampling rates (22khz for starters), and wouldn't mind at all if it were priced at five times what the ISD chips cost. I'd gladly shell out $40-50 a chip for the upper range of lo-fi quality!

i wanna say that if i ever get my shit together as a builder, this would be my holy grail. best of luck guys.

stumbled across the following while looking for something else ... I didn't see a schematic, but the ideas seem interesting ...

http://little-scale.blogspot.com/2009/10/ram-music-audio-contortionist.html (http://little-scale.blogspot.com/2009/10/ram-music-audio-contortionist.html)

Collaborative coding sounds like a recipe for a clusterf***.

That being said, my first thought is that you might want to get a couple standard digital recording ICs and pass the signal back and forth (one chip plays back while the other records the playback mixed with the new signal).  Use an oscillator to control the switching rate.

Coding a decent sounding looper is going to take a very fast chip with a lot of external memory.  Maybe you are a savant, but I recommend starting with the basic PIC chips before diving into DSP.  There is a huge learning curve that takes a great deal of focus, dedication, and ability to deal with extended periods of frustration to overcome.

I think there are easier ways to build this than using uC's for the recording and playback, but at the end of the day you will likely want a PIC to control the other features.

Well, I've already considered using 2 ISDs and passing signal back and forth, but the consensus was that the bandwidth is too low, and distortion too high, so after about 3 repeats the signal becomes too muddy to be heard.

I wish there was a Puredata-to-hardware translator, that would make this so easy.

Regarding memory, the EHX 2880 uses compact flash cards, and they are pretty cheap for large amounts of storage. Like 4gb for $20. I don't know how quickly they're able to write, but obviously EHX was able to make it work.

Not that I'm against the idea but for the amount of work and cost in hardware you can get something like an RC-2 for less.

I've seen this with MP3 player and DIYers. People try and build a MP3 player but with the cost of shuffles and nanos its just not worth the time except purely for the learning. They don't take off as community projects and lots of builds.

Andrew

Yeah, I'm beginning to think you're right.

I'm thinking that if we do go ahead with this, the idea would be to make something very different from the simpler loopers out there. It does seem to be the case that, for simple sound on sound, it would be more practical to buy.

Quote from: scratch on November 02, 2009, 06:59:19 PM
stumbled across the following while looking for something else ... I didn't see a schematic, but the ideas seem interesting ...

http://little-scale.blogspot.com/2009/10/ram-music-audio-contortionist.html (http://little-scale.blogspot.com/2009/10/ram-music-audio-contortionist.html)

this is actually really great, thanks for the link!

I went ahead and ordered the large RAM chip he/she used for this project. it doesn't look too hard to implement. personally I'd rather control it with an AVR, though I still haven't figured out how I'm going to control a 19-bit address bus with an 8-bit chip, quickly

A dsPIC  and SD cards maybe?

How about the xmos?
http://www.xmos.com/
http://www.xmoslinkers.org/

I think a modern looper should be 24bit and 44.1kHz. It should have undo and reverse and half-speed and double speed. It should even be stereo.
It should use some regular PC DIMMs that are readily available as scrap and there should be enough sampling time to accommodate several switchable layers and
parts, like A, B and C sections of a song. MIDI tempo sync is a must. USB audio/midi I/O too. It should have an expression pedal input to control input level.

Saving all layers as WAV/AIFF to a USB stick would make it a killer for serious production.

I know I am asking for much, but why not make it great and useful from start?

double post sorry.

Stereo is very unlikely, since it would make the project much more complex for everybody and only benefit the very few people who would use it in stereo. Ditto for MIDI and USB. Those are good ideas and useful to some, but not the core audience of this forum.

Expression pedal inputs are things we could very easily add to the project, in fact input volume is something you could add to the analog part of the circuit, without need to implement that in code.

Reverse and multiple playback speeds are a must, though. In fact, I'd personally want continuously variable playback speed, not just 1/2 and double time.

Quote from: Taylor on November 03, 2009, 10:25:06 PM
Reverse and multiple playback speeds are a must, though. In fact, I'd personally want continuously variable playback speed, not just 1/2 and double time.

exactly! us droners absolutelly love continuously variable speed *grin*

24bit and 44khz would be nice but is probably pretty daunting in terms of hardware cost.

personally, my interest in those is not to create something to better what's out there (except maybe a few extra functionality features) but to learn how to work with digital

I've started going through Digi-key and Mouser looking for 16bit ADC's w/ parallel output ... not an easy task and not cheap ...

Atmelloop? (http://www.atmel.com/dyn/resources/prod_documents/doc1456.pdf)...

Interesting, but with a sampling rate of 8k, the max frequency is 4k, even less than the ISD chips. Don't think it supports simultaneous read and right either.

Update to 12bit (http://www.atmel.com/dyn/products/tools_card.asp?tool_id=4506) found @ Elektor-newsletter... (http://www.elektor.de/elektronik-news/avr-xmega-eval-board-fur-29-us-dollar.1139223.lynkx?utm_source=DE&utm_medium=email&utm_campaign=news)

from what I see, the 2 needed digichips will be <7.00 $

Quote from: scratch on November 07, 2009, 03:49:35 PM
I've started going through Digi-key and Mouser looking for 16bit ADC's w/ parallel output ... not an easy task and not cheap ...

Would any of these work?

http://focus.ti.com/apps/docs/viewdevices.tsp?blockDiagramId=6017&blockId=8833&designOptionId=10005&appId=376 (http://focus.ti.com/apps/docs/viewdevices.tsp?blockDiagramId=6017&blockId=8833&designOptionId=10005&appId=376)

Or

http://www.analog.com/en/analog-to-digital-converters/audio-ad-converters/products/index.html (http://www.analog.com/en/analog-to-digital-converters/audio-ad-converters/products/index.html)

Both of these companies offer freebies to build prototypes  ;D

Does this help???

Quote from: puretube on November 07, 2009, 06:28:10 PM
Update to 12bit (http://www.atmel.com/dyn/products/tools_card.asp?tool_id=4506) found @ Elektor-newsletter... (http://www.elektor.de/elektronik-news/avr-xmega-eval-board-fur-29-us-dollar.1139223.lynkx?utm_source=DE&utm_medium=email&utm_campaign=news)

thats pretty awesome!

I need to drop Bascom and learn some C/C+ so that I can use these advanced chips

Well, something like the boomerang+ (my favorite looper) uses some sort of analog devices dsp and a stick of memory from a 486 or something it also has a foot roller for loop volume which is where an expression pedal might be handy.

I was messing with the variable speed with exp pedal on my rds-900 last night and it is awfully fun.

I don't know much about digital but I'm in for whatever I can help with on this project!

Just a quick thought:

Revisiting the idea of using the ISD recorders, what if we placed an exciter circuit after each recorder, to add back what's lost each time in the a/d/a conversion? After all, the exciter was invented to add back/resynthesize high end to tape recordings with many overdubs. That's basically what we're working with here.

This would never sound like an RC-2 or similar, but for something a little strange but without total mud, it might be cool.

I just got myself a dsPIC kit with the express intent on doing some mid-fidelity (12-bit) looping and sampling in its many forms. I intend on using an MCP4921 DAC as it's well documented, cheap and seems to be a norm among PIC users. storage is a little trickier. I don't think using RAM chips is going to be feasable since they usually employ huge address bits (19+ bit) which means either using 19+ pins on the MCU or doign some tricky binary counting stuff I'm not ready to go into. anyway I need the RAM to be random access, not sequential, putting binary counters out of the picture.

I'm looking at using memory cards for storage. SD/MMC is a good candidate since it only uses like four pins to communicate with the MCU (via SPI). need to buffer 512 bytes worth of sound and dump it all at once on the card while filling up another 512 byte buffer in the meantime.

I was using AVRs previously which would of probably been good for 8-bit looping, but having the dsPIC's 16-bit pipe instead will be a huge help for better fidelity. in fact since I'm using 12bit conversion, there's a full 4 wasted bits that have absolutelly no concequence on the speed of the processing! if I find the *right* 16-bit DAC and ADC, it would cost no more than 12-bit. I'm just using 12-bit now because the chip has in-built 12-bit ADCs and I'm just learning.

I'm using the mikroBasic compiler because I'm not smart enough for C.

Cool, interested to see what you come up with.

I was also interested in this:

http://www.diystompboxes.com/smfforum/index.php?topic=74742.msg661285#msg661285

I found this last night and just thought to pop this in here because there are probably people here who will enjoy this finding very much:

Mobius - the programmable looper
http://www.circularlabs.com/index.html

It is free and available for both Win/OSX, both standalone and VST/AU.

Quote from: Taylor on December 18, 2009, 06:29:01 PM
Cool, interested to see what you come up with.

I was also interested in this:

http://www.diystompboxes.com/smfforum/index.php?topic=74742.msg661285#msg661285

yeah that sampler looks completelly nuts! just a 40-pin AVR, a resistor ladder and some RAM. great design!

I held off buying it because not having one gives me further incentive to learn how to eventually make my own

I notice a lot of lo-fi loopers and samplers use 74lc ICs for address latching. I can't helt then agging feelign that I should learn what these are about. I've tried the datasheets, but still can't "visualize".

http://little-scale.blogspot.com/search/label/eprom%20music (http://little-scale.blogspot.com/search/label/eprom%20music)
found this the other night it sounds interesting

I'll post a report on the Where's The Party At Sampler (mentioned by Taylor a few posts ago)

I just populated it and ready to solder, I forgot to order a couple parts so it may be a few weeks...

Its not really a "pedal" without a MIDI foot controller (which I would like to DIY as well if such a project exists)
:icon_eek: more exciting is that a friend of mine, an engineer who recently built a custom midi controller for me said he's working on a small sampler :icon_eek:
I'll try to connect him to this thread although he's really busy with a Linux related day job. He uses mainly AVR for his embedded projects. He said something to the effect that he had to program the RAM functionality himself, I'm really dumb about digital so you'll have to bear with my loose description but he basically said he had to write the protocol for the way the RAM chip gets stuff written into memory because there wasn't an exactly right sort of product out there "out of the box." I do not know where they are with this project. He works with an interesting shop here in portland that sells one of a kind, really extreme pedals and touch synths:

www.buzz-r-electronics.com

- Strategy

Quote from: Strategy on December 22, 2009, 11:56:45 AM
www.buzz-r-electronics.com

Aha, I have run across this guy - I have a feeling he'll be selling about 5 Gristleizers pretty soon...

Doesn't the WTPA? have onboard switches for rec/play/etc.? If so, couldn't you just wire up some momentary SPST stomps to perform these functions without MIDI?

yes, you could totally wire up spst momentary's, no problem, but there's even more stuff you can do with MIDI than from the front panel, so that was my rationale with that (and I'd love to have a MIDI foot controller any way.) So for pedal builders that would be a good route to go. I want my WTPA? to double as a desk top module. About half the time I play shows I am using table top gear like drum machines etc. So my pedals have to be desk friendly and my desk gear has to be foot friendly when I play 'live' instruments.

Buzz-R-Electronics is awesome. I have bought a bunch of stuff from there. The guy who runs it is a long time experimental/improvisation community person here who I have played with a bunch. Glad they will have gristleizers!!

I will ask Alex from Buzz-R about their sampler project.

- Strategy

Quote from: ljzimmer on November 12, 2009, 06:33:46 AM

Quote from: scratch on November 07, 2009, 03:49:35 PM
I've started going through Digi-key and Mouser looking for 16bit ADC's w/ parallel output ... not an easy task and not cheap ...

Would any of these work?

http://focus.ti.com/apps/docs/viewdevices.tsp?blockDiagramId=6017&blockId=8833&designOptionId=10005&appId=376 (http://focus.ti.com/apps/docs/viewdevices.tsp?blockDiagramId=6017&blockId=8833&designOptionId=10005&appId=376)

Or

http://www.analog.com/en/analog-to-digital-converters/audio-ad-converters/products/index.html (http://www.analog.com/en/analog-to-digital-converters/audio-ad-converters/products/index.html)

Both of these companies offer freebies to build prototypes  ;D

Does this help???

sorry for taking so long to answer, but no these don't work in terms of the project posted at little-scale ... it uses parallel interface ADC/DAC, it makes easy to interface directly to the SRAM. The ones you posted are Sigma-Delta converters, an encoded serial 'bitstream' output which would require additional hardware to process to store in memory and recall. These I wouldn't know how to use even if they are cheaper. The parallel 16bit ADCs and DACs I'm looking at so far go for approx. $23 each.

Apologies on being a late-comer but you aren't going to be able to use a PIC or AVR and expect to get a looper that is not basically a toy (eg, low-bit sampling and low sample rate).  You can get some very cool stuff like the Where's The Party At but nothing even close to commercial quality.  For that, you'll need a DSP chip or some other much faster platform - maybe a 16 or 32 bit system, at the least.  The AVR32 and dsPIC might be able to do it but those are a totally different beast than AVR or PIC.  You would probably be better off looking into a more well-accepted platform, like TI, maybe TMS320C55x, with an external ADC and RAM (SD card would be smart).  But really embedded DSP is not at all well internet-documented, so it will certainly be an adventure!

-Colin

yeah the best I was able to do with regular AVRs was 8-bit mroe or less 20khz, and there were clicks every few fractions of a second because of the buffering I had to constantly do.

I've been doing tests with a 16-bit dsPIC (F30) since last week and this thing is a BEAST in comparison. I'm doing 12-bit sampling at 90+ khz  :icon_biggrin:
still wont be pristine though, because of limited headroom.

Quote from: ExpAnonColin on December 23, 2009, 10:26:33 PM
embedded DSP is not at all well internet-documented, so it will certainly be an adventure!

+10

Quote from: cloudscapes on December 23, 2009, 10:42:57 PM
I've been doing tests with a 16-bit dsPIC (F30) since last week and this thing is a BEAST in comparison. I'm doing 12-bit sampling at 90+ khz  :icon_biggrin:
still wont be pristine though, because of limited headroom.

What language are you writing on the dsPIC?  Why 12-bit, because that's the built-in ADC?  And what do you mean about limited headroom - you mean because you're sampling at 12 and not 16 or 24 bit, or because of 5v?  Because if it's the latter, just attenuate and reamplify!

-Colin

Quote from: ExpAnonColin on December 24, 2009, 02:50:30 AM
What language are you writing on the dsPIC?  Why 12-bit, because that's the built-in ADC?  And what do you mean about limited headroom - you mean because you're sampling at 12 and not 16 or 24 bit, or because of 5v?  Because if it's the latter, just attenuate and reamplify!

-Colin

in basic (mikrobasic compiler). I can't seem to ever get my head around C when I look at it.

12bit because I already had 12bit DACs lying around that are incredibly simple to use. I looked for 16bit (and over) converters but they all seemed to be much more complicated to use, with compression and SMD leads. for now since I'm just learning this new platform, I'll stick with simple.

yeah, lower headroom because of the 12bit. currently I am attenuating and reamplifying, but a bunch more noise adds itself. I've ordered some 12bit ADCs which I guess will have less noise than the internal ones I've been using.

Quote from: cloudscapes on December 24, 2009, 09:29:32 AM
in basic (mikrobasic compiler). I can't seem to ever get my head around C when I look at it.

12bit because I already had 12bit DACs lying around that are incredibly simple to use. I looked for 16bit (and over) converters but they all seemed to be much more complicated to use, with compression and SMD leads. for now since I'm just learning this new platform, I'll stick with simple.

Yeah, once you get into 16 bit ADC's normally you have to deal with a serial output, which is not so bad once you get used to it. If basic has macros for I2C then you can definitely get a 16 bit ADC which communicates over that.  Or even SPI.  You might head over to AVRfreaks and ask them if anyone is using a 16 bit adc with an avr.

-Colin

Quote from: ExpAnonColin on December 24, 2009, 03:08:31 PM
Yeah, once you get into 16 bit ADC's normally you have to deal with a serial output, which is not so bad once you get used to it. If basic has macros for I2C then you can definitely get a 16 bit ADC which communicates over that.  Or even SPI.  You might head over to AVRfreaks and ask them if anyone is using a 16 bit adc with an avr.

-Colin

well, I'm not to irked by SPI. the DAC I'm using is SPI and it's been pretty easy to use. I guess it's just that if I choose a 16-bit DAC/ADC instead, then I can no longer put all control bytes and data bytes in a single 16bit word. I'll have to do a few control bits first, then wait, then the data. to someone like me who's just learning, it doesn't fit in as neatly a tight package.
I'll eventually get to 16 bit, and I'm sure dsPICs can do 24bit even, but for now I'm gonna stick with this. 12bit is nice.

I guess it's also the surface mount stuff that puts me off. I haven't found any SPI 16bit converters in DIP

There's these, both 16 bit in DIP
http://www.newark.com/linear-technology/ltc1605cn-pbf/16bit-adc-1605-dip28/dp/38K4125
http://www.newark.com/texas-instruments/dac715p/16bit-dac-pdip28-715/dp/61M5731

Not inexpensive ...
Robert

Quote from: cardinalflyer on December 24, 2009, 06:06:02 PM
There's these, both 16 bit in DIP
http://www.newark.com/linear-technology/ltc1605cn-pbf/16bit-adc-1605-dip28/dp/38K4125
http://www.newark.com/texas-instruments/dac715p/16bit-dac-pdip28-715/dp/61M5731

Not inexpensive ...
Robert

thanks Robert
unfortunatelly they're in parallel interface. not bad for speed and simplicity, but I see that more as a last resort since it'll use so many MCU pins.

I'll still write them down, in case....  ;)

and I havent checked neward before, guess I'll browse a bit

You looking for something like this then?
http://focus.ti.com/lit/ds/symlink/dac714.pdf
http://focus.ti.com/lit/ds/symlink/ads7813.pdf

16 bit serial data in to DAC out in 16 pin DIP format.
16 bit serial data out ADC in 16 pin dip format.  40 KHz sampling, just what's needed for audio.

those are interesting ... I'll check out the price on Digikey, and see what is available in shift registers ...

Quote from: cardinalflyer on December 24, 2009, 08:02:54 PM
You looking for something like this then?
http://focus.ti.com/lit/ds/symlink/dac714.pdf
http://focus.ti.com/lit/ds/symlink/ads7813.pdf

16 bit serial data in to DAC out in 16 pin DIP format.
16 bit serial data out ADC in 16 pin dip format.  40 KHz sampling, just what's needed for audio.

at Digikey the DAC goes for almost $47, and the ADC for $35 (cdn) ... going to be easier (and cheaper) to stick to the parallel ones.

Imo designing both the hardware and software will be a large amount of effort (years probably).

Why not use:
- PC as computing platform. Its the most standardised computing platform with USB device support and user friendly OS environments.
- USB AD card. 16 bits AD with ASIO driver support go for 40 dollars or so these days, 24 bit AD for 100 dollar.
- Host software that support ASIO for low latency (a few ms is standard these days).

Then write the looper app in a nice high level language like C++ (like an ASIO plugin or VST-Instrument) or even graphically (for the non-programmers) using MAX/MSP or PD (pure data)?

But first find out that this has probably been done already by someone else, is readily available and probably freeware! ;)

My advice:
Go laptop, or perhaps even a small netbook, install MAX/MSP or something and have loads of looping fun within a day! :)

Maybe we should split this into a separate thread on good external A/D converters, but here's a very simple one, fast enough, and relatively cheap (~$9)
http://focus.ti.com/lit/ds/symlink/ads8320.pdf

And, it's also good to look at audio-specific ones, this one is stereo at about $5, and can do 48khz:
http://focus.ti.com/lit/ds/symlink/pcm1801.pdf

Unfortunately at this level of performance you have to start getting into smd.

-Colin

yeah initially I didn't want to go with parallel converters because of all the pins they requie.
I guess I could use a port expander, though like the MCP23S17.

I tend to stay away from AD's surface mount stuff, because not only is it surface mount, it's the super tiny kind. I have one of their 8-pin DDS chips and it's almost too small for tweezers (grain of rice size)! no way can I solder those. not DIY friendly at all.


flo: I dont know, as far as I'm concerned, the point of DIY is to keep away from laptops ;D
I like being limited and super-compact

I dont think it would take years to develop. I learned how to use dsPICs and how to input sound and output it again in just a few hours. add some storage and basic looping buttons and you've already surpassed most ISD-based loopers in a few more hours/days of R&D.

overdub is another story. the math in programming isn't hard, but in practice I'm eager to see how easily it'll clip.

Quote from: cloudscapes on December 27, 2009, 11:20:44 AM
flo: I dont know, as far as I'm concerned, the point of DIY is to keep away from laptops ;D

My suggestion was to use the PC/laptop as hardware+interfacing base. The actual looper software development part is still DIY, right?

Hey guys, today I coded up a looper for the FV-1. The chip only has 32k of memory, which gives one second of recording time with 20hz-16khz frequency range - definitely not a looper. But, a clever guy at the Spin forum showed me a way to internally undersample the delay, which allows us to do some things I didn't think were possible with the chip.

Right now I've got it doing 8 seconds of delay at 4khz sampling (which means 2khz is the highest frequency it can represent cleanly). It's pretty aliased, ala chiptunes/Nintendo, but with 2 4-pole filters, one before and one after the delay, you can get it sounding more like an analog delay than bit crushed. I'm going to see how far I can stretch this - at 32 seconds, it would allow up to 500hz frequency response. This is known in audio circles as "completely f-ing terrible", but for the adventurous it might be fun.

Without filtering, 500hz will probably sound ridiculously aliased, beyond 8-bit NES stuff (yes, I know there's a difference between sample rate reduction and bit reduction). but with heavy filtering (which is easy in the FV-1) it will be able to represent the fundamental frequencies of most of the notes on a guitar.

So, it'll come out either crazy, ugly, or without any harmonics, but the amazing part is just that it's possible to get something like 30 seconds of delay time out of so little memory. This is far from what I had envisioned when I started this thread, and it will never be anything like a clean commercial looper, but if you're cool with lo-fi sounds, I think you'll enjoy this. "What makes this different from a LoFi Loop Junky/Payback-type looper?" you ask. Answer: even worse sound quality, and infinite layers of overdubbing.  ;D

If anybody thinks they'd be interested in building this (perhaps after I can do some sound clips), let me know and I'll think about how to make it happen (at its simplest, you could build the Tonepad FV-1 board and I could send you an EEPROM chip with the looper code on it). Does that sound interesting to anybody?

i would try it.

I would try this to.

Taylor, can the code be written so that the user could alter the amount of downsampling, from short, hi-fi loops to zonked, low fi, long loops?

This might just be the project to push me into checking out FV1. 

A short loop has its uses, too, like the "hold" on old digitech double pedal digital delays- very musical on the 2-second delay for instance. And those are NOT hi-fi, full bandwidth digital delays, they have their alias-hiding filtering, and its' very musical.

- Strategy

Quote from: Strategy on March 13, 2010, 12:34:42 AM
Taylor, can the code be written so that the user could alter the amount of downsampling, from short, hi-fi loops to zonked, low fi, long loops?

A short loop has its uses, too, like the "hold" on old digitech double pedal digital delays- very musical on the 2-second delay for instance. And those are NOT hi-fi, full bandwidth digital delays, they have their alias-hiding filtering, and its' very musical.

- Strategy

Yep. Coding it as a continuously variable control would be a bit hard, but what we could do is use the switch inputs to select between some preset sample rates. Depending on whether we need a switch for anything else, there could be up to 8 different sample rates.

Keep in mind I just did the first code and tested it today, so the concept isn't all that fleshed out, but I was thinking a knob for delay time, a knob for filter frequency, and a knob for feedback/repeats. Then a rotary for sample rate. Ideally I'd like to do a stomp switch to set time like a real looper if I can figure it out, but at least for now it just works like a really long delay - you set loop length with the delay time knob.

I really recommend the FV-1. It opens up a ton of possibilities outside the reach of analog DIY. There's definitely a learning curve, but it's actually pretty easy to get into.

Preset sampling times is good. If you develop the project more, I might be into trying it, but my backlog is mighty. That said, Where's The Party At and Tau Pipe Phaser are both almost done (for reals this time) so who knows, maybe I'll be able to start digging into this.

Will keep watching for updates on this

- Strategy

Quote from: Strategy on March 13, 2010, 12:34:42 AM
I would try this to.

Taylor, can the code be written so that the user could alter the amount of downsampling, from short, hi-fi loops to zonked, low fi, long loops?

This might just be the project to push me into checking out FV1. 

A short loop has its uses, too, like the "hold" on old digitech double pedal digital delays- very musical on the 2-second delay for instance. And those are NOT hi-fi, full bandwidth digital delays, they have their alias-hiding filtering, and its' very musical.

- Strategy

I still use an RDS-900 (rack version)  I've tweaked it so it will go to about 1.5 seconds, it has it's uses.  I have a boomerang+ and 16 sec in my rig and I still use the RDS for a lot of stuff.  It gets a grainy/pixilated sound at longer delays that I like, not to mention the built in LFO and the voltage control delay time, it can get real fun!

Quote from: Strategy on March 13, 2010, 02:18:44 AM
Preset sampling times is good. If you develop the project more, I might be into trying it, but my backlog is mighty. That said, Where's The Party At and Tau Pipe Phaser are both almost done (for reals this time) so who knows, maybe I'll be able to start digging into this.

Will keep watching for updates on this

- Strategy

One cool thing about the FV-1 is that the hardware doesn't need to change to build a completely different effect. So if you build this looper, you can also download tons of other programs and have other interesting effects available without having to build a new pedal. It's also way simpler to build than WTPA or any Juergen Haible stuff. Definitely interested to hear about those two builds, BTW.

Does anyone have any thoughts about other modes or features that would be fun with this? My plan has already used up the 3 available pots, however I have worked out a trick to get a 4th pot, so what else could we add? A vibrato ala the LoFi Loop Junky is possible, but it would have to have either the speed or depth preset, or else have them controlled by the same knob (giving you slow shallow vibrato, or fast deep vibrato, or the other way around). Or if I do work out a solution for a record/play stomp switch, it would probably use this 4th pot.

Any interesting suggestions welcome.

Vibrato is good- I was thinking filter cutoff freq controllable by potentiometer. On the other hand, it seems like a pedal like this could have analog stages in the same box, even a simple tone control would be useful or the Escobedo Q&D VCF...

I plan on doing comprehensive build reports for the mega projects WTPA and Tau!

- Strategy

I'm curious about this? Do you mean Digitech RDS-1900? And it has voltage control - like a CV input for delay time?!?

- Strategy

Quote from: wavley on March 13, 2010, 10:36:11 AM
I still use an RDS-900 (rack version)  I've tweaked it so it will go to about 1.5 seconds, it has it's uses.  I have a boomerang+ and 16 sec in my rig and I still use the RDS for a lot of stuff.  It gets a grainy/pixilated sound at longer delays that I like, not to mention the built in LFO and the voltage control delay time, it can get real fun!

Quote from: Strategy on March 14, 2010, 01:01:15 AM
Vibrato is good- I was thinking filter cutoff freq controllable by potentiometer. On the other hand, it seems like a pedal like this could have analog stages in the same box, even a simple tone control would be useful or the Escobedo Q&D VCF...

I plan on doing comprehensive build reports for the mega projects WTPA and Tau!

- Strategy

Yep, I've already got the filtering controlled by a pot right now. A tone control or simple analog filter really won't do here - you need a very steep filter to cut the aliasing while allowing the instrument to come through. Right now I have two 4-pole filters, one before the delay and one after.

BTW, any pot on the FV-1 is a CV input, so (with some caveats) you could have CV control over delay time, filtering, and feedback if you wanted to, without any real modding.

Oh boy, my next post will be the mythical 1337! I think that's the post # after which DIYers become men.

CV inputs? Well that is badass. FV-1 is now really really appealing. Is there a noob guide or how-to-get-going thread?

What are the caveats for CV input?

- Strategy

Quote from: Taylor on March 14, 2010, 01:42:22 AM

Yep, I've already got the filtering controlled by a pot right now. A tone control or simple analog filter really won't do here - you need a very steep filter to cut the aliasing while allowing the instrument to come through. Right now I have two 4-pole filters, one before the delay and one after.

BTW, any pot on the FV-1 is a CV input, so (with some caveats) you could have CV control over delay time, filtering, and feedback if you wanted to, without any real modding.
[/quote]

Well, one is that there is about 100ms delay on the pot inputs. This makes no difference when adjusting them with your fingers, but if you were sending a time-synced LFO or something into the pot inputs, it would be delayed by a 10th of a second, which might be a problem. The "4th pot" trick, which uses one of the audio inputs as a pot input, does not have this delay, but you can't send a regular CV into it, unfortunately.

The other thing is that the CV needs to range between 3.3v and 0v, so for most things you'd probably need to attenuate the CV if it's 0-5v.

The best noob guide is to check out the knowledge base (http://spinsemi.com/knowledge_base.html) at the Spin Semi website, and read through the forums a bit. I would be happy to help out with any questions you have about getting started with it.

my favourite "loopers" are japanese sampling delays from the 1980s.  they provide a maximum of 1 second sampling time, but there was always a good selection of different overdub and truncate features, depending on the make/model. also, because they were pre-midi, there were inputs for realtime control of sample length and speed.  it is my goal to one day reverse engineer one of these, so that I can make them myself, possibly with more sampling time.. you simply cannot beat the sound. I don't like any of these new 24bit samplers... as far as I'm concerned, nothing beats that 12-bit punch.

the best ones tend to be the most rare/esoteric, but check out the Boss RSD-10 and Arion DDS-1, you can still find them occasionally.

DOD DFX9/91/94s are pretty common too.

those old digidelays are my favorites too. I've got a DFX94, a few PDS delays, Aria DD-X10 (is amazing!)..
I've been trying to plan/build a looper for a couple months. truncating and realtime speed control will be necessities. overdub not so much.

I messed around with a couple 16bit DACs I found this morning. no luck with the TDA1543. the WS pin (left/right switch) is used to "finish" the LSB but it calls for it to be triggered before the last bit. I haven't figured out how to do that with an AVR or dsPIC. the offset options can only go so far. bitbanging it doesn't even give me random voltages. it just sits still dumbly because I guess the clock doesnt line up perfectly with WS.

a bit more luck with the PT8211 I found, but it completelly ignores the MSB no matter how I offset everything. so I just have 15bit depth on that one. it's strange, one would assume pulling the data line completelly low or completelly high would give me the min voltage and max voltage, but it instead goes to the "middle". for both of them. yet I still get that 15bit precision if I send the first 15 bits or last 15 bits.

What's truncating in a delay? Does that mean that you can move the read pointer around without changing pitch? I could see that being useful in a glitch context. I'll see if I can do a mode like that in my looper.

By the way, it's quite easy to bit reduce on the FV-1, so there's no reason we can't have a 12-bit version of this looper - it already sounds rather crunchy due to the low sampling rate.

Whoa.  :o

Today I added a delay time control to the looper. Instead of acting nice and polite like I expected, it turned this thing into a completely schizo glitch machine. With the time pot stationary, it works just as before, and you can dial in different times just as I expected and wanted. But when you play something into it and let it loop, and then change the time control, it glitches out in a fantastic way.

Instead of just repeating a snippet of sound, the snippet that you hear changes from moment to moment. So let's just say I recording myself saying abcdefghijklmnop into the loop. When I adjust the time parameter to a shorter time, I expected to hear

abcd abcd abcd abcd

(this I think is the truncation you guys were talking about above) but instead I get

abc nop def abdef abdeg fghijk pppp mno

all without playing with the knobs. So you can record a longer loop, then let it spit out random snippets of your loop in a jumbled and ever-changing manner.

By setting the time super-short, I get a very nice "playing inside of a rusty steel cube" kind of pseudo-reverb. I will try to record some clips today.

That sounds fantastic! any chance for clips?

great job Taylor. The sample-crumpling "truncation"  - i.e., sample time does not affect pitch but crumples up the data as you modify the delay time- there ARE some precedents for this, but they seem to always get programmed out of the line. I once borrowed an early Boss pitch shifting delay (possibly the sought after PS-2 - can't recall now, its 10 years ago) and one of the delay settings did that, I used it extensively. The later PS series did not feature this sample crumpling. IIRC, the EH "#1" delay has a nice crumple to it, and it can be really nice to be able to modify delay time without causing the huge pitch bends.

I really look forward to this project, once you have coded it up, it will get added to the queue.

Re: minor CV delay times, it's hard for me to conceptualize the lag in audio terms, without hearing it. If its glitch/delayed signal anyways, it might not be very noticeable, but on a filter setting, that may result in a kind of subtle stair stepping. It's worth a shot in any case, adding CV input jacks, esp being able to modify delay time sample crumple-glitching with my modular synth or analog sequencer.

The CV attenuation could be a pot after the input jack. Although I'm not sure what the ideal variable resistance would be for attenuating incoming CV's. Typical synth CV would be 5V, but they go higher. During the trouble shooting phase on the Tau, we actually cooked the CA3086 (ouch) giving it LFOs that were too hot from a CGS psycho LFO. But I did not measure the voltages. I will have to research what the hottest theoretical CV outputs are.

The recently issued Tiptop Z-DSP modular synth module (Eurorack format) uses FV-1, I think. A couple emails to Tip Top might clear up how to implement voltage control effectively with FV1.

- Strategy

-
Stratey

RE: the lag, it's not stair stepping, it's just that the parameter changes 1/10th of a second after you move the knob. It's not something I hear moving knobs with fingers, but if sweeping with an expression pedal, or if you had a tempo-synced LFO going into it, it would be out of sync with whatever else is going on.

@Taylor

regarding your glitch "time control" ; you've essentially created a simple granulator. that's a whole other project I've wanted to do for a long time.

I forget exactly how the truncation was implemented in the older samplers. I recall a diode array and a switch matrix, but obviously you're trying to keep things simple and in software.

what makes the samplers from the early 1980s so special is that there wasn't really a precedent for the ideal sampler back then. so you got a lot of engineers experimenting with various options they thought the musicians might like. strange circuit designs, usage of local or proprietary japanese ICs, and constant partnerships between various (bubble-rich) Japanese manufacturers also helped cultivate this.  it wasn't until pop music and akai took over samplers in the late 80s early 90s, when you got a solidified set of expected parameters and controls for every sampler. they're all pretty faceless these days. 

sorry i am late to this thread but i am sort of working on something similar. liek a looper/grain sequencer. i will try to make some clips tomorrow.

i made a test on breadboard that is just a sampler with a sort of "zero crossing" playback rate pot so say once your left of center it starts playing the loop backwards. you can screw with it in real time so it sounds like:

helllloooooo mmmmmmmmyyyyyyyyyyyyyyyyyyyy  nnnnnnnnnnnnnnnnnnnnnnnnaaaaaaammmaaaaaaannnnnnnnnnnnnnnnnnnnnn yyyyyyyyyym ooooolleh. and also a pot to truncate the beginning and end loop points of the sample once its recorded.

What hardware are you using?

dsPIC 33

but for what im using in it you could just use a pic24. Im not actually using any dsp functions. its more that i bought something high end to prototype with.

"ooolleh"  :icon_lol:

which dacs/adcs are you using scott?

those truncate/chop/start/end knobs sound like my Akai S-612- one slider for start, one for end, and when you reverse them you are reversing the sample. Awesome hardware sampler,  I read somewhere that the engineer on it had gone from Electro-Harmonix to Akai, hence the pedal like nature and hands on functionality...

- Strategy

oh man so weird i had a very similar idea to originally just have two sliders and a record button, maybe i saw this akai thing once and dont remember. but instead i decided to make the rate and direction in one knob. hmm.

i was thinking also somehting along the lines of having 4 sliders that act more as markers. then having different play modes for example one could have a pattern like, play from one marker to the next, but then skip from that marker to the next, then play from that marker to the next. etc... to rearrange the sample by chopping and reordering it.

or even having 1 slider and a button to "add a marker at the position of the slider" and a button to clear all markers or something like that. something simple.

also for hardware i am not using anything fancy right now. it was mostly about developing the algorithm before dealing with finer details. adc is built in and dac is some microchip spi dac.

http://www.youtube.com/watch?v=FhuDJ2jMUr0

Quote from: Strategy on March 17, 2010, 11:11:59 PM
those truncate/chop/start/end knobs sound like my Akai S-612- one slider for start, one for end, and when you reverse them you are reversing the sample. Awesome hardware sampler,  I read somewhere that the engineer on it had gone from Electro-Harmonix to Akai, hence the pedal like nature and hands on functionality...

- Strategy

you're thinking of David %^&*rell. he worked for EMS in england, then went to design for EH, and finally Akai, where he worked on their first MPCs. absolutely wonderful engineer, I love all of his work.

Quote from: Rectangular on March 18, 2010, 07:27:55 PM

you're thinking of David %^&*rell. he worked for EMS in england, then went to design for EH, and finally Akai, where he worked on their first MPCs. absolutely wonderful engineer, I love all of his work.

Ha, that would be Co.ckrell. Censoring software...

seeing how precise and flexible filters are needed for those of us doing ada conversions, this tool is super useful:

http://www.microchip.com/stellent/idcplg?IdcService=SS_GET_PAGE&nodeId=1406&dDocName=en010007

introductory video
http://techtrain.microchip.com/media/filterlab/filterlab.html

yes that software is nice. There are a lot of excellent TI application notes too that have similar information.

also maybe you could use somethign like the switched capacitor filters maxim makes because then you can have up to 8th order in an 8 pin chip which saves some board space/soldering time ;). maybe TI makes somethign similar too since thay love signal processing so much.

just go to the maxim website and go products->analogfilters, neat stuff!

sort of side track:

in terms of 16 bit sampling dont you think it is excessive for soemthing like this? i think about 50 uV for the LSB (wiht a 3.3 volt chip). not to be pesimestic but your not likely to beat that noise floor without some nice filtering/pcb layout. and plus you are taking up 25 percent more memory (than 12 bit, boo). Maybe you could use an ADC with nonlinear quantization with ~10 bits and still get a nice dynamic range.

Quote from: Taylor on March 18, 2010, 07:58:07 PM

Quote from: Rectangular on March 18, 2010, 07:27:55 PM

you're thinking of David %^&*rell. he worked for EMS in england, then went to design for EH, and finally Akai, where he worked on their first MPCs. absolutely wonderful engineer, I love all of his work.

Ha, that would be Co.ckrell. Censoring software...

haha, da.mn, I forgot this forum auto-censored things. god co.cking da.mnit.  anyway, Co.ckrell is a great engineer.

any new developments on this project?

Do you mean the original idea, which was to make a hi-fi powerful looper, or the tangent I went down, which was a glitchy lo fi looper?

The former is not happening at the moment. I have begun learning DSP and will be investigating hardware for it. Since it's such a large project and beyond my personal abilities at the moment, it's a long-term project for sure. It would be great if some of the heavier-hitters w.r.t. DSP would volunteer, but as long as it's just me working on it, it might be a multi-year project.

I have been getting into PICs, and am envisioning a looper based on the ISD chipcorders, which would have something like 4-6 of the chips all linked up by an MCU which would step through them, recording on one while playing back the others. The Tone God said they don't sync up perfectly, but that's not a huge concern of mine. This unit would be more for e experimental/ambient/drone types.

I've decided to take a look at the ISD17000 series and take a stab at programming a controller for it, much in the same way your tremolo design was done.  I think I do enough programming in research that it shouldn't be to hard to figure it out the logic and the language.  The hard part will finding the time.

Ok, sounds cool. So this'll be in the tradition of the Payback, etc. more than a modern digital looper? I'll be interested to check it out. Did you get an email from some musicians asking you to build such a thing by chance?  :icon_wink: I got the mail as well.

Quote from: Taylor on August 06, 2010, 03:34:24 PM
Ok, sounds cool. So this'll be in the tradition of the Payback, etc. more than a modern digital looper? I'll be interested to check it out. Did you get an email from some musicians asking you to build such a thing by chance?  :icon_wink: I got the mail as well.

yes, I've received a few emails and eventually found that thread.  I've made myself a ISD2500 looper in the past, but I won't be making any loopers for bassists anytime soon.  They all want tiny, lovepedal-sized single loopers and I had more of a vision to create a Echoplex-voiced Payback style thing.  Tone God may still beat me to it with v2, but I'm more interested in learning a way to program chips with a fun project.

Both the ideas proposed in this thread were very interesting, I hope they eventually pan out.

Cool, that sounds interesting.

You forgot to mention that they want those hand-made teeny tiny loopers for less than a Boss pedal!  :icon_biggrin:

Have you looked into the "Where's the party at?"? It's a DIY project that does overdubbing, multiple separate loops, and all kinds of crazy granular loop slicing. It basically does all the usual pedal looper stuff but adds a bunch of weird stuff as well. The only real drawback is that it is 8-bit quality, but you can tack a lowpass filter on the end and it should sound similar to the ISD chips but with a lot more functionality. I bought 2 of the PCBs and chips and hope to start building soon (haven't gotten the boards yet). It looks like a really cool thing and I think this in combination with a Payback 1.0 I'm building will probably end up replacing my EHX 2880. The EHX is of course more clean, but I'm not sure that matters too much for me.

Sorry if this is a rediculous question, but what is wrong with the example circuit on the ISD1740A/50A/60A or whatever those numbers were?

Quote from: Taylor on August 06, 2010, 05:23:00 PM
Cool, that sounds interesting.

You forgot to mention that they want those hand-made teeny tiny loopers for less than a Boss pedal!  :icon_biggrin:

Have you looked into the "Where's the party at?"? It's a DIY project that does overdubbing, multiple separate loops, and all kinds of crazy granular loop slicing. It basically does all the usual pedal looper stuff but adds a bunch of weird stuff as well. The only real drawback is that it is 8-bit quality, but you can tack a lowpass filter on the end and it should sound similar to the ISD chips but with a lot more functionality. I bought 2 of the PCBs and chips and hope to start building soon (haven't gotten the boards yet). It looks like a really cool thing and I think this in combination with a Payback 1.0 I'm building will probably end up replacing my EHX 2880. The EHX is of course more clean, but I'm not sure that matters too much for me.

is there a thread on the "where's the party at?"  I've been search to no avail.

EDIT: found it

Quote from: mth5044 on August 06, 2010, 05:24:45 PM
Sorry if this is a rediculous question, but what is wrong with the example circuit on the ISD1740A/50A/60A or whatever those numbers were?

Eh, well it doesn't have buffers or mixers, and the switching is a little awkward, and the input expects a microphone signal rather than an instrument signal. I think it does have an analog output rather than the speaker out only of the earlier chips, though. Oh, and it doesn't loop AFAIK, so you'd have to retrigger playback each time.

Quote from: aziltz on August 06, 2010, 05:42:56 PM
"where's the party at?"  I've been search to no avail.
EDIT: found it

:icon_lol:

You guys are being silly!  USE A PIC.  They have 1G of memory they are dirt cheap and easy to work with.  I have been thinking about programming a looper for a while now.  This was just the motivation I needed.  I'll make one and post my finding somewhere on here.  A few questions first how do you guys think it should function?  What would be the best way for all of you?

Which PIC has that much memory? What kind of bit depth and sample rate does the ADC have? Sounds cool to me, hope that comes to fruition.

You won't get any real consensus, since this is a really open-ended kind of unit and people use them in different ways.

I really like multiple layers that can be independently controlled for continuous pitch/speed control, feedback level, and volume. Something like 4 simultaneous loops would be great. Reverse is important.

Quote from: lil PopTart kid on August 10, 2010, 12:45:16 AM
You guys are being silly!  USE A PIC.  They have 1G of memory they are dirt cheap and easy to work with.  I have been thinking about programming a looper for a while now.  This was just the motivation I needed.  I'll make one and post my finding somewhere on here.  A few questions first how do you guys think it should function?  What would be the best way for all of you?

There isn't any PIC with 1GB of memory!

I've been working on an idea to get a second ISD 1700 chipcorder to quantize to the loop point of a first one. Basically, it's the same way you get the first one to loop: route the signal from the -sp out lead to the play input lead. When play stops, the voltage drops to ground, and the chip (almost) instantly starts playing again. My idea is to just route the -sp out lead to BOTH the play switch input of the 1st AND the second chip. You could have the second chip be switchable to quantize either with the loop point of the first chip, or to itself. Adding a third chip, you could then have much more permutations of quantization for the second and the third chip (both to the first, second to the first and third to second, second to itself and third to second, second to itself and third to first, second and third to themselves). Loop length doesn't matter. Say first loop is two bars, and second is 1 bar. Second starts with first, and then stops. When first restarts, -sp goes to ground, and triggers both. Say loop one is 2 bars and loop 2 is 4 bars. Same idea, first loop will go twice, but will trigger the second loop when it ends. The trick will be that you will need to end the second loop a little shorter than an equal increment of the first (ie, if first is EXACTLY 2 bars, you'll need to end the second at 1.99 bars). Otherwise I'm not sure if it will trigger at the right time (ie, if it is too long, it will not get triggered until the NEXT time around).

All the audio is taken from the +sp lead, and will need to mixed down together. I suggest "volume control" style passive mixer circuits for each loop.

I've successfully built a single looper with this concept. It self-quantizes just fine. Not totally seamless, but perfectly adequate for my needs. I need to buy a second ISD 1700 board (the fully populated test boards from digikey are AWESOME time savers!) and some more momentary switches in order to test this idea (slaving the second loop to the quantize point of the first), but I am 99% sure it will work.

I don`t understand that quantize thing? Is it like synchronizing?

If you are using two chips in parallel, to record separate parts, you need these two chips to remain in sync. So let's say you record an octaved bassline, then a lead on top on the other chip. If the chips don't remain in sync with each other, then after a little while it sounds like your lead guitar is dragging behind the bass player (perhaps more realistic this way, but most don't want the "guitarist-has-had-3-Jamesons" effect). So you need to work out a scheme where they are clocked together.

That is if sync is a big deal to you. My looping interests tend towards droney stuff where sync is not a real issue.

Hey Taylor, what was the programmed chip technology used in the Tap Tremolo?  I want to start reading up.

Quote from: aziltz on August 10, 2010, 07:06:14 AM
Hey Taylor, what was the programmed chip technology used in the Tap Tremolo?  I want to start reading up.

PIC16F684. Code, datasheet for the TapLFO: http://www.electricdruid.net/index.php?page=projects.taplfo

Mike

Quote from: JKowalski on August 10, 2010, 01:50:34 AM
There isn't any PIC with 1GB of memory!

Excuse me ment 1MB.  However you can easily hook up extra memory. 
Also If I can come up with a smart way of going about making the circuit
and the program all the memory isn't needed.

And oooo a reverse! I think I saw that one time on an electro harmonix
looper and it was AWESOME.

Quote from: Lurco on August 10, 2010, 03:01:02 AM
I don`t understand that quantize thing? Is it like synchronizing?

Well, yes, sort of. The Quantize point is the loop initiation trigger point. In more sophisticated loopers, you can set the quantize point to be any even increment or divisor of the loop. So for example, you record a loop that is 4 seconds long. Normally, the quantize point is 4 seconds, so the loop plays for 4 seconds, then starts at the beginning. With a sophisticated software looper like sooperlooper, you can set the loop to quantize at, for example, 8 seconds. So you get 4 seconds of playing, 4 seconds of silence, then 4 seconds of playing again. If you have two loops, you can make the second loop quantize to the start of the first loop (which is the way it "normally" works), so that regardless of how long or short the second loop is, it will always start at the "beginning" of the first loop. With very sophisticated loopers (again, like sooperlooper) you can set the quantize point to be, for example, the nearest 8th note, and thus trigger the second loop BEFORE the first loop ends. If you use such complicated quantization schemes with multiple loops, you get very cool polyrhythms between the multiple loops.

  However, with the way I'm suggesting to trigger the loops with the ISD chips, there is only 1 quantize point per loop. That point is the start of the loop. If you have more than one chip, each chip has it's own quantize point (starting point) that has the potential to trigger the loop to restart. In the case of only two ISD chips, you can either opt to have the second loop trigger based on the starting point of the first loop, or you can have the second loop trigger based on it's own starting point. If it is important that the loops be "aligned", then one ought to take the first option. If it doesn't matter, then you can do the second one. Doing the second one also gives you the option to build looped polyrythms, depending upon the difference in loop intervals and the timing of your loop triggering. Doing this all with the ISD chips and such "analog" triggering methods means that if you want to have multiple loops and have them all aligned on the same beats, you have to be much more precise while setting the loop point than you do with other loopers, who "do it for you".  The advantages of doing it this way are that it's actually cheaper than buying a commercial hardware looper with multiple loops, the whole thing can be powered off of 3AA batteries (ie. it's ultraportable for street performance), you can set up the control layout to be exactly the way you want it (ie. you can put the switches in the box so they fit your needs), and it's totally cool to build your own! The disadvantages are that you can't actually overdub with sound-on-sound, but instead can only just have two independent single loops that are syncing their initial loop points together, you have to be more precise when setting the loops because there is less "fudge factor" than with commercial units, and finally, you have to build the thing yourself which means sourcing all the components, testing everything, assembling, retesting, debugging, retesting, etc, etc., before everything works.

Thanks for those explanations! Much appreciated.

There's nothing even remotely solid yet, but this project (that of a powerful, modern, high fidelity looper) may be on again.

With that in mind, let's start discussing the feature set. Here's the basis that I'd plan for:

High sampling quality - somewhere around 32khz sample rate, 12 bit or higher.
At least a minute of sampling time
Overdubs
Reverse
Loop playback speed change - I'd want continuously variable speed/pitch, but a half speed button might be possible as well
Feedback control


Here's some other stuff I'd want personally, but let me know whether you like these ideas or not:
Multiple parallel loops with individual feedback, volume, and playback speed controls - maybe 2 to 4 loops.
Stuff like punch-in, replace, insert for putting in snippets of sound into a loop - this is borrowed from the Echoplex Digital Pro.


Feel free to add your own thoughts on features. But keep in mind that the overarching concept for this unit is to be a pedal looper in the style of the Boomerang, DL4, Echoplex Digital, eHX 2880, etc., and aimed at live use by instrument players. To that extent, studio-oriented stuff like USB sample backup and stereo operation will probably not be included. The key is to not add features which would vastly increase the cost and build complexity, but which would only be useful to a small number of people.

Once some ideas have been posted, I'll make a little poll so we can gauge interest in various features and concepts to get a consensus on the core feature set.

Figured I'd work a little on my looper project today, here's a 3-second aphex twin loop getting mangled:

http://www.youtube.com/watch?v=eka7dL_mH2M

The two pots define both the start and end loop points, thoguh right now they're buggy. the pushbutton reverses. Slowing down and speeding up is planned, as well as other stuff. It'll have delay functions too.

Right now I'm mostly working in code. I'm using an AVR clocked at 12MHz. I tried clocking it faster but I had problems. Sound is 8-bit and I'm using the AVR's own ADC for conversion. It's then stored in an external RAM (23K256, only 32KB, around 3 seconds worth at 11khz lo-fi) and then outputted on an external DAC (MCP4921).

I'm kinda pushing the chip's limits right now, and the additional functions I want to add will bog it down even further (meaning less khz), so I ordered an xmega kit which will be MUCH beefier! Only problem is that they're surface-mount only. But the advantages are nice! 32mhz (some people push it to 40 with no stability problems), it's supported by my compiler so porting will be easy, direct memory access, more faster memory, longer loop times, etc! I'll aim for 12bit 22khz top speed.

I figured an easy way to change the speed is to simply use an analog clock to drive the MCU instead of a crystal! like an opamp and squarewave, or some CDxxxx.

Awesome! ucephalus sulahpecuB ceph cephbbbbbbbbbbbbbb

Did you see that:

http://www.diystompboxes.com/smfforum/index.php?topic=86729.0

? Might provide some ideas for you, as it's similar. The guy who designed is cool and helpful, and his code and hardware are all open. narrat1ve.com.

yep, I knew about it! that stuff is an inspiration right there!

This is from the other thread but I think it makes more sense to reply here:

Quote from: cloudscapes on September 12, 2010, 08:13:19 PM
nice job! I've been keeping a close eye on this, studyign the schematic, etc.
it's sort of what I want to achieve with my own looper, but done "my way"
the window-shifting is great! the timefactor has that too.

Here are a few things that I'd like to see that the WTPA doesn't do:

- Separate playback speeds for each loop (this can probably be done with the WTPA using MIDI, but not with the built-in interface)
- Feedback control. The WTPA essentially has 100% feedback, so if you add overdubs they just stack up, and there's no way to fade out older layers.
- Copy one loop to the other/another loop location, so you can phase them against each other, mangle one up and then go back to the original, etc.

Of course, this is your thing and I know you'll do it your way, so these are just some ideas I've had since I've been using the WTPA.

Even though we disagree on the subject of Roman Polanski, I always enjoy seeing the stuff you're doing.  ;)

I'll probably start small. like a version without separate loops and loop shifting, or midi. mostly just a compact digital looper/delay with loop window/grain mangling stuff. there will definitely be feedback control. not that difficult. if I can, I'll do this with these small AVRs. if I have to bring the max samplerate down to 6 or 8khz, then so be it.

then several months down the line I can make a WTPA-like looper. ;) we'll see. plans can always change.

experimenting with time stretching

http://nearworlds.org/stuff/glitcher_test_2.mp3

what I'm doing is moving a very sample short window forward in the loop (slowly!), and inside the window the "playback head" is pingponging back and forth and a normal speed.

Ah cool. Are you able to move through the loop automatically or are you scrubbing through with a pot? i can do this with the WTPA manually, but if it could be automated to do jungle-style time stretch that would be really useful.

Something I just thought of that might be cool is to randomize playback speed per grain, so if you jumble up a bunch of segments of a loop, they not only play in the wrong order but each at a different speed.

right now the scrubbing is manual-ish... hard to explain. I found the time-stretching by accident. I have two pots (start point and end point), keep them close together and I push the reverse button very rapidly all throughout the manual scrubbing of my two pots. no random playback speed planned I'm afraid, and it'll be hard to do that with how I want to set it up. playback speed will be analog. it'll be an opamp generating a MHz pulse pretending to be the AVRs crystal clock. this way I can have smooth sample/playback speed bending and no need for code (saving precious cycles). down side is I cant do any fancy tricks to the speed.

adding an LFO to the grain window shifting along with feedback and continuous recording (turning it into a delay basically) should give pretty interesting results! if the grain window is only 300ms long (typical short delay length), we'll hear older samples (2-3 seconds old) bleed into the delay as it sweeps back and forth!  :icon_mrgreen:

Hmm, that does sound interesting.

The WTPA actually does it's clock the way you say: the sample rate knobs are controlling an analog oscillator's frequency. And he has implemented a random grain playback thing, but I do think he said it was difficult, so I can see why you wouldn't want to mess with it.

the WTPA uses an analog clock source?  ??? thats weird, because I'm looking at it's schematic and it's very much a crystal (along with the mandatory 20pf caps to ground). and I'm pretty sure I see a crystal to the left of the big chip in the picture on the WTPA site

If you look on the last page of the PDF, you can see the analog oscillator which goes to the analog clock input of the AVR. But, there's also a crystal. So I guess it uses both - I think maybe it uses the analog one when working with the built-in controls, and uses the xtal when using MIDI. I don't have the midi hooked up yet so I can't check if this is true.

So I had to take my RDS-900 out of my rig because it is just to danged noisey, while I was looking for a schematic for it so I could think about making some changes (a fet input op amp, cap values, and maybe some 5532s instead of 4558s should do it) I found this site about the RDS-7.6 that has some info that might help you considering that RDS stuff fits some of your parameters.  The schem on that site doesn't apply to much to my 900 because the later models use 4164 RAM and mine uses 4116, sucks because I have some 4164.  There's a guy that shows how to make an adapter for them to fit, but I have two Apple ][+'s and they have 4116 as motherboard ram.  Unfortunately, while re-seating all my chips trying to find the cracking noise, my hand slipped and I mangled my 570 compandor, I have a 571 I can pull out of something this weekend that may do, but I haven't measured my voltages yet, if that 570 was running over 18 volts then I'm just gonna have to wait and order one before I can mess with upgrades.  I was thinking about putting a pot on the front panel to replace the clock trimmer on the board so that I can chose how much bit crushing is going to happen on the fly.  And I'm going to work on shoving a little more memory in the thing.

effectivly doubled ram space, but had to rewrite half the code.  :icon_cry:

Quote from: Taylor on September 13, 2010, 11:59:33 PM
If you look on the last page of the PDF, you can see the analog oscillator which goes to the analog clock input of the AVR. But, there's also a crystal. So I guess it uses both - I think maybe it uses the analog one when working with the built-in controls, and uses the xtal when using MIDI. I don't have the midi hooked up yet so I can't check if this is true.

thanks. I think the PDF I have is missing some graphics (where the analog CLCK can be switched int othe oscillator), but I'll go ahead anyway.

here's a neat little feature/knob I added today. pushing time-stretching and mangling to the limit

http://nearworlds.org/stuff/wtf_looper.mp3

what happens when you "virtually" push the reverse button up and down really fast? how about hundreds or thousands of times a second? this.

this "looper" is gonna be fun to play  ;D

 :icon_eek:

I like it.

I'm having a schmitt trigger feed a square wave into the reverse/don't reverse input pin. anywhere between a few hz to 1 khz.  ;D the duty cycle isn't a perfect 50% which is why it drifts a bit.

I did away with the 'reverse' pushbutton. it's now a pot, where one extreme is forward and the other is reverse. in the middle is varing degrees of what you can hear in my last sample, who's speed is set by another pot.

now trying to implement overdub. I'm trying to feed the output back into the input (to hell with sound quality), but unfortunatelly am not getting anything. even when I buffer all my inputs/outputs. that's my biggest weakness in this I think. the stuff that should be easy but actually has a lot of theory behind it.

For something nasty, I think analog overdubbing is cool, because it allows to effect each repeat progressively with an effects loop. As you note, resampling is bad practice from a hi fi perspective, but that's obviously not the point of 8 bit sampling to begin with.

looks like I'm going to have to forget about overdubbing though. I've spent 7 (yes, 7) hours today trying to get only that working. something with the micro just doesn't seem to want to accept feedback. tried digital overdubbing as well. nonsensical results.

Quote from: cloudscapes on September 12, 2010, 07:46:22 PM

I'm kinda pushing the chip's limits right now, and the additional functions I want to add will bog it down even further (meaning less khz), so I ordered an xmega kit which will be MUCH beefier! Only problem is that they're surface-mount only. But the advantages are nice! 32mhz (some people push it to 40 with no stability problems), it's supported by my compiler so porting will be easy, direct memory access, more faster memory, longer loop times, etc! I'll aim for 12bit 22khz top speed.

Did you ever start working with the XMEGA kit? I was looking for inexpensive controllers with larger memory sizes and the XMEGA chips look really good. They also have cheap eval boards, although I'm not sure how you'd use an external audio codec with one.

Thanks,

--Will

Quote from: hoopshot on October 26, 2010, 04:29:45 PM
Did you ever start working with the XMEGA kit? I was looking for inexpensive controllers with larger memory sizes and the XMEGA chips look really good. They also have cheap eval boards, although I'm not sure how you'd use an external audio codec with one.

Thanks,

--Will

hi Will

I haven't really done anything with the xmega kit aside from making sure I could compile code for it and get a blinking light going. in any case, the compiler and language/environment I use doesn't yet support every feature so it's not yet worthwhile for me. stuff like DMA (though preliminary support was added a couple days ago), and the internal DACs.

so for now, I'm going to finish my looper/sampler with the "lesser" chip.

I haven't looked into how well the xmega will handle 24bit codec chips.

Quote from: cloudscapes on October 26, 2010, 10:53:14 PM
hi Will

I haven't really done anything with the xmega kit aside from making sure I could compile code for it and get a blinking light going. in any case, the compiler and language/environment I use doesn't yet support every feature so it's not yet worthwhile for me. stuff like DMA (though preliminary support was added a couple days ago), and the internal DACs.

so for now, I'm going to finish my looper/sampler with the "lesser" chip.

I haven't looked into how well the xmega will handle 24bit codec chips.

Thanks for the info. I was looking for an Xplain evaluation kit and found that they have them at DigiKey for $30.16. These boards have an XMEGA chip with multiple 12-bit ADCs and DACs and 8MB of SDRAM on the board.

--Will

waiting for a PCB is haaaard!   :icon_mrgreen:
I'd do it myself but it's double sided, and I want this proto to be reliable. not patched up with jumpers and lifting copper.

(http://nearworlds.org/stuff/scwt.gif)

basic interface. no graphics yet. no stompswitches as it's meant more as a tabletop looper.
(http://nearworlds.org/stuff/crushed.gif)

Awesome. Will you have a second board for the LEDs and pots, or will you wire all that offboard?

...and will this be a DIY-able project?

How do I get one?  :D

well done!~

Yeah, I'm totally in to this, my boomerang is giving me trouble with digital noise and ground loops, I'm tired of repairing it (for the longest time it would start recording whenever a loud noise happened, turned out to be a bad solder on a fet)  The only thing I can reasonably replace it with is another boomerang and a can't afford the new one and I feel that nothing else is quite as performance oriented.  I don't care about saving two hours of crap, I might as well be playing with a computer or sequencer and there is nothing rock and roll about that.  This looks really promising it looks like all the features I need to be foot switchable can happen.  I can't wait for the new WTPA to come out and try converting that to a pedal.  You guys rock because my digital skills are marginal at best, I'm an old school tube amp guy.

Quote from: Taylor on November 18, 2010, 10:55:49 PM
Awesome. Will you have a second board for the LEDs and pots, or will you wire all that offboard?

...and will this be a DIY-able project?

the 16 leds in the middle will have their own little board and own co-processor.

for this prototype, everything will be hand-wired (I've done worse)
in the future, I hope to have EVERYTHING on a single board. pots and all. save for maybe the jacks.

PCB is done. enclosure is drilled and primed. need to start on paintjob/decal/finish. then, assembly!

(http://nearworlds.org/stuff/crushed_glass_populated_pcb.jpg)

AWESOME! Assuming the project goes forward the way you hope, will you be doing runs of boards etc?
well done
strategy

there is a second (smaller) pcb i naddition to this one. it has the 16 chaser leds in a row and a microcontroller that acts as a 4-to-16 multiplexer.

Quote from: Strategy on December 04, 2010, 07:01:17 PM
AWESOME! Assuming the project goes forward the way you hope, will you be doing runs of boards etc?
well done
strategy

I... might....

Not before I redo the board so that everything is PCB mounted. pots and leds. except maybe the two toggle switches. pre-programmed chips would also need to be included.

http://soniccrayon.blogspot.com/2010/12/crushed-glass-granular.html

:)

Very cool as always. Rick/FrequencyCentral had a tube noise generator called "Crushed Glass" awhile back, funny coincidence.

The noise-beat guys will eat this up when you start making them for sale.  :)

Quote from: Taylor on December 12, 2010, 07:57:57 PM
Very cool as always. Rick/FrequencyCentral had a tube noise generator called "Crushed Glass" awhile back, funny coincidence.

The noise-beat guys will eat this up when you start making them for sale.  :)

now that you mention it, it does kinda sorta ring a bell..

my brain sort of reads a cool term or word and years later I use it for a song or project, totally forgetting where it's from and thinking it's my own idea.  :o

sorry rick!

I'll have to restart the code from scratch if I'm to sell a few of these. it has some quirks that I've gotten used to but I'm sure others will dislike. the code is so patched up and hacked that it proved impossible fixing these.

I also think next time I make this sort of looper, it will be a good opportunity to finally learn C and start using dsPICs in a real project.

Your marketing skills need to catch up with your design skills! In weirdo processor land, quirks are selling points, not problems.  ;) But I do know what you mean.