FV-1 Multi effect full forum project ready to go. (ShimmerVerb Also)

[potul]

looks like a good solution

you need to connect it to S0, S2, and S3 pins of the FV-1 (plus gnd and 3.3V)

[Tuff Pedals]

Hmm,

Bit of a dumb question really ! but thought I'd ask, just in case someone could think of any possible issues 

Thanks daz

[reichi]

@ice9: any chance you're gonna release your new fv-1 design's gerber files once you're satisfied with it?

[Ice-9]

Hi,

Did anyone see this as a control for program selection - http://www.hexeguitar.com/diy/utility/fv1progsel ?

What is the best option, I can see the topic has been discussed in the past but can't see a definitive solution. If I was to utilise the above solution, where would it connect into the board?

Thanks for a great project, cheers daz

Yeah this would work fine, just don't fit the pic chip to the main pcb. There are some small solder pads which sit under where the pic would normally go for this exact type of switch fitting.

On the other hand I'm sure I posted a little switch pcb which replaces the pot and connects to the 3 pot pads, this makes fitting easier than the 5 wire version above.

[Ice-9]

Here is a little update for the ShimmerVerb which is meant for the new project pedal but will work in the original as well. Give it a try, it was work done some time ago but will be the basis for the new shimmer.

Code Select Expand


;New Shimmer Reverb Program
;from 3k Room
;09/01/2013 rev 1.01 Mick Taylor
;22/11/2015 rev 2.1 Set pre Delay Reverb freq. response and gain changes
;                    pre delay removed for shimmer code space             
;07/12/2015  Shimmer code added Mick Taylor

;Pot0 = Shimmer
;Pot1 = reverb level
;Pot2 = reverb time

mem shimdel 4096 ;delay for shimmer
mem stemp 1
mem idel 4000 ;initial sound space  122mS
mem iap0 11
mem iap1 27
mem iap2 43
mem iap5 171
mem iap6 296 ;thickening all passes embedded in initial delay

mem ap1 134         ;4.1mS
mem ap2 256         ;7.8mS
mem ap3 562         ;17.1mS
mem ap4 763 ;reverb loop input all passes

mem lap1a 1421       ;43mS
mem lap1b 1945       ;59mS
mem d1 2434       ;74mS
mem lap2a 1894       ;58mS
mem lap2b 1767       ;54mS
mem d2 2645 ;80.7mS   loop constants

;write constants registers

equ kd -0.5 ;damping coefficient for shelving

;write-first registers

equ dry reg0
equ rev_in reg1
equ kirt reg2 ;coefficient to scale initial sound
equ krt reg3 ;coefficient to affect RT of loop
equ apout reg4 ;output of loop input all passes
equ temp reg5 ;temp register for filter routines
equ gain reg6 ;adjust gain with RT
equ revout reg7
equ    pitchout  reg8    ;octave up output

;read-first registers;

equ lf1 reg20 ;reverb loop filter 1
equ lf2 reg21 ;reverb loop filter 2
equ hf1 reg22 ;loop high pass 1 fixed
equ hf2 reg23 ;loop high pass 2 fixed
equ lfin1 reg24 ;LPF for imbedding in initial delay
equ lfin2 reg25 ;LPF for imbedding in initial delay
equ lf reg26 ;input low pass shelving with kd
equ lpfp reg27

equ    lpfk    0.3      ;lpf coefficent for lpfp after pitch shifting 1.85kHz
equ    lpfs    -0.5      ;Shelving coefficent for lpfp

;clear read-first registers

skp run,endclr
wrax lf1,0
wrax lf2,0
wrax hf1,0
wrax hf2,0
wrax lfin1,0
wrax lfin2,0
wrax lf,0
endclr:

;initial sound tap positions 30.5uS/location, 100=3.05mS

equ ld1 874 ;first tap, left   26.7mS
equ rd1 874 ;first tap, right 26.7mS
equ ld2 1156 ;and so on...    35.3mS
equ rd2 962                    ;29.3mS
equ ld3 1345                  ;41mS
equ rd3 1121                ;34.2mS
equ ld4 1456                ;44.4mS
equ rd4 1423            ;43.4mS
equ ld5 2121            ;64.7mS
equ rd5 2124            ;64.7mS
equ ld6 3245              ;99mS
equ rd6 3646              ;111.2mS

;initialize sin LFO

skp run,endset
wlds sin0,25,100
wldr RMP0,16384,4096    ;load octave up
endset:
;--------------Off and Running Program Loops to Here--------------------------
;prepare decay pot Reverb Time

rdax pot2,0.97 ;get pot, limit to less than infinite
wrax krt,1 ;write loop decay time
sof 0.4,0.6 ;scale Pot to 0.6 to 1.0 range
wrax kirt,1 ;write impulse filter gains changed to 1 from 0 MT 22-11-15 gain always +0.99 before=too high
sof -0.88,0.99          ;scale to decrease gain with RT need to assess the -1 & the 0.99 range offset for  gain Vs RT now changed to -0.88
; Range allowed=-2.0 to +0.9999389 e.g. From scale above if pot2=0.6 then 0.6* -0.88+0.99=0.46;if pot2=1 then gain=0.11 23-11-2015
wrax gain,0 ;write gain factor and clear ACC

;------------------------------Octave up------------------------------------

cho rda,RMP0,REG|COMPC,shimdel
cho rda,RMP0,,shimdel+1
wra    stemp,0
cho rda,RMP0,RPTR2|COMPC,shimdel
cho rda,RMP0,RPTR2,shimdel+1
cho sof,RMP0,NA|COMPC,0
cho rda,RMP0,NA,stemp
mulx    POT0
rdfx    lpfp,   lpfk      ;Freq coef
wrhx    lpfp,   lpfs      ;Shelving coef.
wrax    pitchout,0

;-------------do inputs to predelay-------------------
;rdax pitchout,1
rdax adcl,0.5
rdax adcr,0.5 ;get inputs sum & divide by 2
wrax dry,1 ;22/11/2015 write dry input signal to dry register and keep in ACC for mulx next instruction:
mulx gain ;Acc=Acc* reg& give greater gain to short RT See code above for adjusting this 23-11-2015 Steve
wrax rev_in,1 ;22/11/2015 write gain adjusted dry input to rev_in register and clear ACC Reg1
wra shimdel,0
;--------------------read predelay and write initial all pass response delay------------------

rdax pitchout,1
rdax rev_in, 0.5              ;use 97mS delayed signal divided by 2 for reverb input
rda iap0#,0.5              ;Read from end of initial all pass memory0 divide by 2 adding to rev_in data
wrap iap0,-0.5 ;complicate input to initial delay
wrax temp,1                  ; Write ACC to register;multiply ACC x 1.         
rdfx lf,0.404              ;Low pass <2.7kHz
wrhx lf,-1                      ;Register=ACC; ACC=ACC*&#40;-1&#41;+previous contents of ACC
mulx kd                          ;ACC=ACC*Reg  kd = damping coefficient for shelving from POT0
rdax temp,1 ;low pass filter entire input
wra idel,0 ;write initial sound delay  clear ACC

;complicate initial sound

rda idel+500,1            ;read from 15.2mS position Retain ACC
rda iap1#,0.5                ;read from end of iap1 delay stream divided by 2
wrap iap1,-0.5                ;write to beginning of iap1,
wra idel+500,0            ;data at delay ram address=ACC; ACC=ACC*0 , i.e. clear ACC

rda idel+1000,1            ;read from 30.5mS position
rda iap2#,0.5                ;read from end of iap2 delay stream divided by 2
wrap iap2,-0.5                ;write to beginning of iap2,
wrax temp,1 ;save filter input
rdfx lfin1,0.2                ;0.2=~1.2kHz perhaps too high for Abbey Road reverb try 600Hz = 0.109
wrhx lfin1,-1 ;make HP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back input shelving LPF
wra idel+1000,0          ;now modify idel+1000 but clear ACC

rda idel+2500,1          ;read from 76.3mS position retain ACC
rda iap5#,0.5
wrap iap5,-0.5
wrax temp,1 ;save filter input
rdfx lfin2,0.2                ;0.2 =~1.2kHz
wrhx lfin2,-1 ;make HP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back input shelving LPF
wra idel+2500,0          ;data at delay ram address=ACC; ACC=ACC*0 , i.e. clear ACC


rda idel+3000,1          ;read from 91.5mS position retain ACC
rda iap6#,0.5
wrap iap6,-0.5
wra idel+3000,0

;do reverb input all passes

rda idel,0.9 ;leave some headroom
rda ap1#,0.5
wrap ap1,-0.5
rda ap2#,0.5
wrap ap2,-0.5
rda ap3#,0.5
wrap ap3,-0.5
rda ap4#,0.5
wrap ap4,-0.5
wrax apout,0                  ;Save all pass out to

;do reverb loop and sum all outputs

rda d2#,1                    ;Read from end of d2, retain ACC
mulx krt                        ;krt = Reverb Time coefficient
rdax apout,1
rda lap1a#,0.5
wrap lap1a,-0.5
rda lap1b#,0.5
wrap lap1b,-0.5
wrax temp,1 ;save filter input
rdfx lf1,0.404                ;2.7kHz
wrhx lf1,-1 ;make LP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back temporary filter input keep ACC
rdfx hf1,0.01                ;ACC=ACC+reg-ACC*0.01
wrhx hf1,-0.5 ;roll out lows in loop
wra d1,0                      ;Write sum to d1 location clear ACC

rda d1#,1                  ;Read from end of d1 memory
mulx krt
rdax apout,1
rda lap2a#,0.5
wrap lap2a,-0.5
rda lap2b#,0.5
wrap lap2b,-0.5
wrax temp,1
rdfx lf2,0.404              ;Again use 2.7kHz
wrhx lf2,-1
mulx kd
rdax temp,1
rdfx hf2,0.01
wrhx hf2,-0.5
wra d2,1.99               
rda d1,1.99               
mulx pot1                     
mulx pot1
wrax revout,0              ;Reverb output saved to register, ACC cleared

;do reverb smoothing

cho rda,sin0,sin|reg|compc,d1+100
cho rda,sin0,sin,d1+101
wra d1+200,0

cho rda,sin0,cos|reg|compc,d2+100
cho rda,sin0,cos,d2+101
wra d2+200,0

;now combine to output

rdax dry,1                      ;
rdax revout,1
wrax dacl,0


[potul]

thanks for the code, I will give it a try later this week

[Momomo391]

What does everyone think of this as a DIY shimmer?? Ive packed this with as many features as i could. Its my first prototype of the design but ill have all the build files ready soon just finalizing everything. so keep up to date on the progress with my social media
https://www.facebook.com/coloradosynthdoctor
https://www.instagram.com/coloradosynthdoctor/

[Ice-9]

Wow, I have just realised that it has been 10 years almost to this date that I first started this FV-1 forum project.  I think it's time for something new, buckle up , get your popcorn in, beers or whatever. I have a plan. New Project, New thread, different  format.

[Cybercow]

Here is a little update for the ShimmerVerb which is meant for the new project pedal but will work in the original as well. Give it a try, it was work done some time ago but will be the basis for the new shimmer.

Code Select Expand


;New Shimmer Reverb Program
;from 3k Room
;09/01/2013 rev 1.01 Mick Taylor
;22/11/2015 rev 2.1 Set pre Delay Reverb freq. response and gain changes
;                    pre delay removed for shimmer code space             
;07/12/2015  Shimmer code added Mick Taylor

;Pot0 = Shimmer
;Pot1 = reverb level
;Pot2 = reverb time

mem shimdel 4096 ;delay for shimmer
mem stemp 1
mem idel 4000 ;initial sound space  122mS
mem iap0 11
mem iap1 27
mem iap2 43
mem iap5 171
mem iap6 296 ;thickening all passes embedded in initial delay

mem ap1 134         ;4.1mS
mem ap2 256         ;7.8mS
mem ap3 562         ;17.1mS
mem ap4 763 ;reverb loop input all passes

mem lap1a 1421       ;43mS
mem lap1b 1945       ;59mS
mem d1 2434       ;74mS
mem lap2a 1894       ;58mS
mem lap2b 1767       ;54mS
mem d2 2645 ;80.7mS   loop constants

;write constants registers

equ kd -0.5 ;damping coefficient for shelving

;write-first registers

equ dry reg0
equ rev_in reg1
equ kirt reg2 ;coefficient to scale initial sound
equ krt reg3 ;coefficient to affect RT of loop
equ apout reg4 ;output of loop input all passes
equ temp reg5 ;temp register for filter routines
equ gain reg6 ;adjust gain with RT
equ revout reg7
equ    pitchout  reg8    ;octave up output

;read-first registers;

equ lf1 reg20 ;reverb loop filter 1
equ lf2 reg21 ;reverb loop filter 2
equ hf1 reg22 ;loop high pass 1 fixed
equ hf2 reg23 ;loop high pass 2 fixed
equ lfin1 reg24 ;LPF for imbedding in initial delay
equ lfin2 reg25 ;LPF for imbedding in initial delay
equ lf reg26 ;input low pass shelving with kd
equ lpfp reg27

equ    lpfk    0.3      ;lpf coefficent for lpfp after pitch shifting 1.85kHz
equ    lpfs    -0.5      ;Shelving coefficent for lpfp

;clear read-first registers

skp run,endclr
wrax lf1,0
wrax lf2,0
wrax hf1,0
wrax hf2,0
wrax lfin1,0
wrax lfin2,0
wrax lf,0
endclr:

;initial sound tap positions 30.5uS/location, 100=3.05mS

equ ld1 874 ;first tap, left   26.7mS
equ rd1 874 ;first tap, right 26.7mS
equ ld2 1156 ;and so on...    35.3mS
equ rd2 962                    ;29.3mS
equ ld3 1345                  ;41mS
equ rd3 1121                ;34.2mS
equ ld4 1456                ;44.4mS
equ rd4 1423            ;43.4mS
equ ld5 2121            ;64.7mS
equ rd5 2124            ;64.7mS
equ ld6 3245              ;99mS
equ rd6 3646              ;111.2mS

;initialize sin LFO

skp run,endset
wlds sin0,25,100
wldr RMP0,16384,4096    ;load octave up
endset:
;--------------Off and Running Program Loops to Here--------------------------
;prepare decay pot Reverb Time

rdax pot2,0.97 ;get pot, limit to less than infinite
wrax krt,1 ;write loop decay time
sof 0.4,0.6 ;scale Pot to 0.6 to 1.0 range
wrax kirt,1 ;write impulse filter gains changed to 1 from 0 MT 22-11-15 gain always +0.99 before=too high
sof -0.88,0.99          ;scale to decrease gain with RT need to assess the -1 & the 0.99 range offset for  gain Vs RT now changed to -0.88
; Range allowed=-2.0 to +0.9999389 e.g. From scale above if pot2=0.6 then 0.6* -0.88+0.99=0.46;if pot2=1 then gain=0.11 23-11-2015
wrax gain,0 ;write gain factor and clear ACC

;------------------------------Octave up------------------------------------

cho rda,RMP0,REG|COMPC,shimdel
cho rda,RMP0,,shimdel+1
wra    stemp,0
cho rda,RMP0,RPTR2|COMPC,shimdel
cho rda,RMP0,RPTR2,shimdel+1
cho sof,RMP0,NA|COMPC,0
cho rda,RMP0,NA,stemp
mulx    POT0
rdfx    lpfp,   lpfk      ;Freq coef
wrhx    lpfp,   lpfs      ;Shelving coef.
wrax    pitchout,0

;-------------do inputs to predelay-------------------
;rdax pitchout,1
rdax adcl,0.5
rdax adcr,0.5 ;get inputs sum & divide by 2
wrax dry,1 ;22/11/2015 write dry input signal to dry register and keep in ACC for mulx next instruction:
mulx gain ;Acc=Acc* reg& give greater gain to short RT See code above for adjusting this 23-11-2015 Steve
wrax rev_in,1 ;22/11/2015 write gain adjusted dry input to rev_in register and clear ACC Reg1
wra shimdel,0
;--------------------read predelay and write initial all pass response delay------------------

rdax pitchout,1
rdax rev_in, 0.5              ;use 97mS delayed signal divided by 2 for reverb input
rda iap0#,0.5              ;Read from end of initial all pass memory0 divide by 2 adding to rev_in data
wrap iap0,-0.5 ;complicate input to initial delay
wrax temp,1                  ; Write ACC to register;multiply ACC x 1.         
rdfx lf,0.404              ;Low pass <2.7kHz
wrhx lf,-1                      ;Register=ACC; ACC=ACC*&#40;-1&#41;+previous contents of ACC
mulx kd                          ;ACC=ACC*Reg  kd = damping coefficient for shelving from POT0
rdax temp,1 ;low pass filter entire input
wra idel,0 ;write initial sound delay  clear ACC

;complicate initial sound

rda idel+500,1            ;read from 15.2mS position Retain ACC
rda iap1#,0.5                ;read from end of iap1 delay stream divided by 2
wrap iap1,-0.5                ;write to beginning of iap1,
wra idel+500,0            ;data at delay ram address=ACC; ACC=ACC*0 , i.e. clear ACC

rda idel+1000,1            ;read from 30.5mS position
rda iap2#,0.5                ;read from end of iap2 delay stream divided by 2
wrap iap2,-0.5                ;write to beginning of iap2,
wrax temp,1 ;save filter input
rdfx lfin1,0.2                ;0.2=~1.2kHz perhaps too high for Abbey Road reverb try 600Hz = 0.109
wrhx lfin1,-1 ;make HP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back input shelving LPF
wra idel+1000,0          ;now modify idel+1000 but clear ACC

rda idel+2500,1          ;read from 76.3mS position retain ACC
rda iap5#,0.5
wrap iap5,-0.5
wrax temp,1 ;save filter input
rdfx lfin2,0.2                ;0.2 =~1.2kHz
wrhx lfin2,-1 ;make HP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back input shelving LPF
wra idel+2500,0          ;data at delay ram address=ACC; ACC=ACC*0 , i.e. clear ACC


rda idel+3000,1          ;read from 91.5mS position retain ACC
rda iap6#,0.5
wrap iap6,-0.5
wra idel+3000,0

;do reverb input all passes

rda idel,0.9 ;leave some headroom
rda ap1#,0.5
wrap ap1,-0.5
rda ap2#,0.5
wrap ap2,-0.5
rda ap3#,0.5
wrap ap3,-0.5
rda ap4#,0.5
wrap ap4,-0.5
wrax apout,0                  ;Save all pass out to

;do reverb loop and sum all outputs

rda d2#,1                    ;Read from end of d2, retain ACC
mulx krt                        ;krt = Reverb Time coefficient
rdax apout,1
rda lap1a#,0.5
wrap lap1a,-0.5
rda lap1b#,0.5
wrap lap1b,-0.5
wrax temp,1 ;save filter input
rdfx lf1,0.404                ;2.7kHz
wrhx lf1,-1 ;make LP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back temporary filter input keep ACC
rdfx hf1,0.01                ;ACC=ACC+reg-ACC*0.01
wrhx hf1,-0.5 ;roll out lows in loop
wra d1,0                      ;Write sum to d1 location clear ACC

rda d1#,1                  ;Read from end of d1 memory
mulx krt
rdax apout,1
rda lap2a#,0.5
wrap lap2a,-0.5
rda lap2b#,0.5
wrap lap2b,-0.5
wrax temp,1
rdfx lf2,0.404              ;Again use 2.7kHz
wrhx lf2,-1
mulx kd
rdax temp,1
rdfx hf2,0.01
wrhx hf2,-0.5
wra d2,1.99               
rda d1,1.99               
mulx pot1                     
mulx pot1
wrax revout,0              ;Reverb output saved to register, ACC cleared

;do reverb smoothing

cho rda,sin0,sin|reg|compc,d1+100
cho rda,sin0,sin,d1+101
wra d1+200,0

cho rda,sin0,cos|reg|compc,d2+100
cho rda,sin0,cos,d2+101
wra d2+200,0

;now combine to output

rdax dry,1                      ;
rdax revout,1
wrax dacl,0


I'm wondering how difficult it would be to modify your New Shimmer Reverb for a stereo output. The input could be mono, but a stereo input as well would be awesome.

I'm still shite at ASM, but I can work with the SpinCAD Designer with ease. If you have a SpinCAD Designer file you could share, the would be cool. Then I could futz with it myself and sort things out.

I'm working with a full stereo implementation of an FV-1 circuit from PedalPCB; the "Binaura". And I've "approximated" as stereo shimmer, but it's not quite right. When I try to enhance it, I run over the 128 line limit for the FV-1.

Thanks for your time and consideration.

Moo

[Ice-9]

Here is a little update for the ShimmerVerb which is meant for the new project pedal but will work in the original as well. Give it a try, it was work done some time ago but will be the basis for the new shimmer.

Code Select Expand


;New Shimmer Reverb Program
;from 3k Room
;09/01/2013 rev 1.01 Mick Taylor
;22/11/2015 rev 2.1 Set pre Delay Reverb freq. response and gain changes
;                    pre delay removed for shimmer code space             
;07/12/2015  Shimmer code added Mick Taylor

;Pot0 = Shimmer
;Pot1 = reverb level
;Pot2 = reverb time

mem shimdel 4096 ;delay for shimmer
mem stemp 1
mem idel 4000 ;initial sound space  122mS
mem iap0 11
mem iap1 27
mem iap2 43
mem iap5 171
mem iap6 296 ;thickening all passes embedded in initial delay

mem ap1 134         ;4.1mS
mem ap2 256         ;7.8mS
mem ap3 562         ;17.1mS
mem ap4 763 ;reverb loop input all passes

mem lap1a 1421       ;43mS
mem lap1b 1945       ;59mS
mem d1 2434       ;74mS
mem lap2a 1894       ;58mS
mem lap2b 1767       ;54mS
mem d2 2645 ;80.7mS   loop constants

;write constants registers

equ kd -0.5 ;damping coefficient for shelving

;write-first registers

equ dry reg0
equ rev_in reg1
equ kirt reg2 ;coefficient to scale initial sound
equ krt reg3 ;coefficient to affect RT of loop
equ apout reg4 ;output of loop input all passes
equ temp reg5 ;temp register for filter routines
equ gain reg6 ;adjust gain with RT
equ revout reg7
equ    pitchout  reg8    ;octave up output

;read-first registers;

equ lf1 reg20 ;reverb loop filter 1
equ lf2 reg21 ;reverb loop filter 2
equ hf1 reg22 ;loop high pass 1 fixed
equ hf2 reg23 ;loop high pass 2 fixed
equ lfin1 reg24 ;LPF for imbedding in initial delay
equ lfin2 reg25 ;LPF for imbedding in initial delay
equ lf reg26 ;input low pass shelving with kd
equ lpfp reg27

equ    lpfk    0.3      ;lpf coefficent for lpfp after pitch shifting 1.85kHz
equ    lpfs    -0.5      ;Shelving coefficent for lpfp

;clear read-first registers

skp run,endclr
wrax lf1,0
wrax lf2,0
wrax hf1,0
wrax hf2,0
wrax lfin1,0
wrax lfin2,0
wrax lf,0
endclr:

;initial sound tap positions 30.5uS/location, 100=3.05mS

equ ld1 874 ;first tap, left   26.7mS
equ rd1 874 ;first tap, right 26.7mS
equ ld2 1156 ;and so on...    35.3mS
equ rd2 962                    ;29.3mS
equ ld3 1345                  ;41mS
equ rd3 1121                ;34.2mS
equ ld4 1456                ;44.4mS
equ rd4 1423            ;43.4mS
equ ld5 2121            ;64.7mS
equ rd5 2124            ;64.7mS
equ ld6 3245              ;99mS
equ rd6 3646              ;111.2mS

;initialize sin LFO

skp run,endset
wlds sin0,25,100
wldr RMP0,16384,4096    ;load octave up
endset:
;--------------Off and Running Program Loops to Here--------------------------
;prepare decay pot Reverb Time

rdax pot2,0.97 ;get pot, limit to less than infinite
wrax krt,1 ;write loop decay time
sof 0.4,0.6 ;scale Pot to 0.6 to 1.0 range
wrax kirt,1 ;write impulse filter gains changed to 1 from 0 MT 22-11-15 gain always +0.99 before=too high
sof -0.88,0.99          ;scale to decrease gain with RT need to assess the -1 & the 0.99 range offset for  gain Vs RT now changed to -0.88
; Range allowed=-2.0 to +0.9999389 e.g. From scale above if pot2=0.6 then 0.6* -0.88+0.99=0.46;if pot2=1 then gain=0.11 23-11-2015
wrax gain,0 ;write gain factor and clear ACC

;------------------------------Octave up------------------------------------

cho rda,RMP0,REG|COMPC,shimdel
cho rda,RMP0,,shimdel+1
wra    stemp,0
cho rda,RMP0,RPTR2|COMPC,shimdel
cho rda,RMP0,RPTR2,shimdel+1
cho sof,RMP0,NA|COMPC,0
cho rda,RMP0,NA,stemp
mulx    POT0
rdfx    lpfp,   lpfk      ;Freq coef
wrhx    lpfp,   lpfs      ;Shelving coef.
wrax    pitchout,0

;-------------do inputs to predelay-------------------
;rdax pitchout,1
rdax adcl,0.5
rdax adcr,0.5 ;get inputs sum & divide by 2
wrax dry,1 ;22/11/2015 write dry input signal to dry register and keep in ACC for mulx next instruction:
mulx gain ;Acc=Acc* reg& give greater gain to short RT See code above for adjusting this 23-11-2015 Steve
wrax rev_in,1 ;22/11/2015 write gain adjusted dry input to rev_in register and clear ACC Reg1
wra shimdel,0
;--------------------read predelay and write initial all pass response delay------------------

rdax pitchout,1
rdax rev_in, 0.5              ;use 97mS delayed signal divided by 2 for reverb input
rda iap0#,0.5              ;Read from end of initial all pass memory0 divide by 2 adding to rev_in data
wrap iap0,-0.5 ;complicate input to initial delay
wrax temp,1                  ; Write ACC to register;multiply ACC x 1.         
rdfx lf,0.404              ;Low pass <2.7kHz
wrhx lf,-1                      ;Register=ACC; ACC=ACC*&#40;-1&#41;+previous contents of ACC
mulx kd                          ;ACC=ACC*Reg  kd = damping coefficient for shelving from POT0
rdax temp,1 ;low pass filter entire input
wra idel,0 ;write initial sound delay  clear ACC

;complicate initial sound

rda idel+500,1            ;read from 15.2mS position Retain ACC
rda iap1#,0.5                ;read from end of iap1 delay stream divided by 2
wrap iap1,-0.5                ;write to beginning of iap1,
wra idel+500,0            ;data at delay ram address=ACC; ACC=ACC*0 , i.e. clear ACC

rda idel+1000,1            ;read from 30.5mS position
rda iap2#,0.5                ;read from end of iap2 delay stream divided by 2
wrap iap2,-0.5                ;write to beginning of iap2,
wrax temp,1 ;save filter input
rdfx lfin1,0.2                ;0.2=~1.2kHz perhaps too high for Abbey Road reverb try 600Hz = 0.109
wrhx lfin1,-1 ;make HP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back input shelving LPF
wra idel+1000,0          ;now modify idel+1000 but clear ACC

rda idel+2500,1          ;read from 76.3mS position retain ACC
rda iap5#,0.5
wrap iap5,-0.5
wrax temp,1 ;save filter input
rdfx lfin2,0.2                ;0.2 =~1.2kHz
wrhx lfin2,-1 ;make HP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back input shelving LPF
wra idel+2500,0          ;data at delay ram address=ACC; ACC=ACC*0 , i.e. clear ACC


rda idel+3000,1          ;read from 91.5mS position retain ACC
rda iap6#,0.5
wrap iap6,-0.5
wra idel+3000,0

;do reverb input all passes

rda idel,0.9 ;leave some headroom
rda ap1#,0.5
wrap ap1,-0.5
rda ap2#,0.5
wrap ap2,-0.5
rda ap3#,0.5
wrap ap3,-0.5
rda ap4#,0.5
wrap ap4,-0.5
wrax apout,0                  ;Save all pass out to

;do reverb loop and sum all outputs

rda d2#,1                    ;Read from end of d2, retain ACC
mulx krt                        ;krt = Reverb Time coefficient
rdax apout,1
rda lap1a#,0.5
wrap lap1a,-0.5
rda lap1b#,0.5
wrap lap1b,-0.5
wrax temp,1 ;save filter input
rdfx lf1,0.404                ;2.7kHz
wrhx lf1,-1 ;make LP filter
mulx kd ;multiply by negative shelving coef
rdax temp,1 ;add back temporary filter input keep ACC
rdfx hf1,0.01                ;ACC=ACC+reg-ACC*0.01
wrhx hf1,-0.5 ;roll out lows in loop
wra d1,0                      ;Write sum to d1 location clear ACC

rda d1#,1                  ;Read from end of d1 memory
mulx krt
rdax apout,1
rda lap2a#,0.5
wrap lap2a,-0.5
rda lap2b#,0.5
wrap lap2b,-0.5
wrax temp,1
rdfx lf2,0.404              ;Again use 2.7kHz
wrhx lf2,-1
mulx kd
rdax temp,1
rdfx hf2,0.01
wrhx hf2,-0.5
wra d2,1.99               
rda d1,1.99               
mulx pot1                     
mulx pot1
wrax revout,0              ;Reverb output saved to register, ACC cleared

;do reverb smoothing

cho rda,sin0,sin|reg|compc,d1+100
cho rda,sin0,sin,d1+101
wra d1+200,0

cho rda,sin0,cos|reg|compc,d2+100
cho rda,sin0,cos,d2+101
wra d2+200,0

;now combine to output

rdax dry,1                      ;
rdax revout,1
wrax dacl,0


I'm wondering how difficult it would be to modify your New Shimmer Reverb for a stereo output. The input could be mono, but a stereo input as well would be awesome.

I'm still shite at ASM, but I can work with the SpinCAD Designer with ease. If you have a SpinCAD Designer file you could share, the would be cool. Then I could futz with it myself and sort things out.

I'm working with a full stereo implementation of an FV-1 circuit from PedalPCB; the "Binaura". And I've "approximated" as stereo shimmer, but it's not quite right. When I try to enhance it, I run over the 128 line limit for the FV-1.

Thanks for your time and consideration.

Moo

Sorry I am not familiar enough with Spincad to be able to convert this to that format. Maybe build a mono in stereo out reverb block in Spincad and add an octave up block before the reverb with feedback and delay to see how it works out.

[Digital Larry]

I can't remember which file it is, but it seems the de facto FV-1 shimmer code that I've seen reproduced several places struck me as weird.  Like the code just "ran off the end" rather than being conclusive.  Sorry if it doesn't make any sense. It worked OK by itself but was not setup up properly to incorporate into SpinCAD.

I did a number of shimmery things as Ice-9 suggests.  Just put a 2-1 mixer before the reverb block and put the pitch shift in different places. 

- before the reverb, outside the feedback loop (before the mixer)
- before the reverb, inside the feedback loop (after the mixer)
- after the reverb, outside the feedback loop
- after the reverb, inside the feedback loop (before the loop input)

Try shifts of both +12 and +7 (a fifth) the +7 is actually my favorite shimmer sound.

You can also experiment with delays, multi-tap, MN3011, etc. in the configuration.  OMG years of fun!  Shreeeeikkk!!   

[Cybercow]

I can't remember which file it is, but it seems the de facto FV-1 shimmer code that I've seen reproduced several places struck me as weird.  Like the code just "ran off the end" rather than being conclusive.  Sorry if it doesn't make any sense. It worked OK by itself but was not setup up properly to incorporate into SpinCAD.

I did a number of shimmery things as Ice-9 suggests.  Just put a 2-1 mixer before the reverb block and put the pitch shift in different places. 

- before the reverb, outside the feedback loop (before the mixer)
- before the reverb, inside the feedback loop (after the mixer)
- after the reverb, outside the feedback loop
- after the reverb, inside the feedback loop (before the loop input)

Try shifts of both +12 and +7 (a fifth) the +7 is actually my favorite shimmer sound.

You can also experiment with delays, multi-tap, MN3011, etc. in the configuration.  OMG years of fun!  Shreeeeikkk!!   

Thanks for that Larry. I'll giver a go.

On another note . . .  have you ever done up a Barber-Pole Flanger in SpinCAD Designer?

[Digital Larry]

I can't remember which file it is, but it seems the de facto FV-1 shimmer code that I've seen reproduced several places struck me as weird.  Like the code just "ran off the end" rather than being conclusive.  Sorry if it doesn't make any sense. It worked OK by itself but was not setup up properly to incorporate into SpinCAD.

I did a number of shimmery things as Ice-9 suggests.  Just put a 2-1 mixer before the reverb block and put the pitch shift in different places. 

- before the reverb, outside the feedback loop (before the mixer)
- before the reverb, inside the feedback loop (after the mixer)
- after the reverb, outside the feedback loop
- after the reverb, inside the feedback loop (before the loop input)

Try shifts of both +12 and +7 (a fifth) the +7 is actually my favorite shimmer sound.

You can also experiment with delays, multi-tap, MN3011, etc. in the configuration.  OMG years of fun!  Shreeeeikkk!!   

Thanks for that Larry. I'll giver a go.

On another note . . .  have you ever done up a Barber-Pole Flanger in SpinCAD Designer?

Heya Mr. Cow, the answer to that is "two moos" or "no".  That was left as an exercise for the reader.  If you can put one together in Spin ASM I'll add a block and name it after you!

DL