; filename: RVB2.ASM ; ; ; This code is an optimized version of the reverberation algorithm ; found in file RVB1.ASM. This program makes use of the parallel ; move functionality of the 56001. While this tends to make the ; code less readible, it does increase its speed. This version ; is intended to run on the DSP56000ADSx (Application Development ; System) board rev #2, with memory expansion. The memory should ; be configured figured for 8K p-memory, 16K x-memory, 8K y-memory. ; ; Further gains in algorithm speed could be realized on boards not ; constrained by the ADS memory limitations. For example, due to ; the need for 4 comb filters' worth of storage (each using about ; 3500 samples - which takes 4096 samples each after using the DSM ; statement) to fit into the 16K available for the x memory, we ; must start at address $0000. Thus, we are forced to use all the ; 56000's internal memory for sample storage. Without being forced ; to start sample storage at $0000, the 56000's internal memory could ; be used for coefficient storage, allowing better use of parallel ; x and y moves. ; ; Motorola DSP Group ; Austin, Texas ; ;************************************************************************** ; This program was originally available on the Motorola DSP bulletin board ; and is provided under a DISCLAIMER OF WARRANTY available from Motorola ; DSP Operation, 6501 William Cannon Dr. W Austin, Texas 78735-8598. ;************************************************************************** ;-------------------------------------------------------------------------- ; This reverberation program is a variation of the reverberation ; system and structures as described by James Moorer's article entitled ; 'About this Reverberation Business', Computer Music Journal, ; 3(2):13-28, 1979 ; ; Structure is: ; .----------. ; .------------. .-----. | All Pass | ; +->| Comb Filter|-->| SUM |-->| Reverb | ; Note: All Comb | | #1 | '-----' | | ; Filters | '------------' ^ ^ ^ '----------' ; Have a 1st | .------------. | | | | ; Order IIR |->| Comb Filter|---+ | | .---V---. ; LPF in their | | #2 | | | | align | ; feedback | '------------' | | | delay | ; loop | .------------. | | '---|---' ; |->| Comb Filter|------+ | | ; | | #3 | | ----- reverb ; | '------------' | \./ gain ; | .------------. | | ; |->| Comb Filter|---------+ | ; | | #4 | | ; | '------------' | ; | V ; .------------. | FIR gain .----------. ; | Early | | |\ | | ;input -->| Reflection |--+---------| >------------>| summer |--- output ; | | FIR | |/ | | ; | '------------' '----------' ; | ^ ; | | ; | dry gain | ; | |\ | ; +------------------------------| >----------------+ ; |/ ; ;............................................................................. ; COMB FILTER SUB STRUCTURE: ; .-------. ; comb i .-----. | long | comb i ; input ------>| sum |---->| delay |-------+---------> output ; '-----' '-------' | ; ^ | ; | V ; | /| .-----. ; +-----< ------------| sum |<--------+ ; \| '-----' | ; fdbck i | / \ lpf i gain ; gain V /_____\ ; gain .----------. | ; | 1 sample | | ; | delay |-----+ ; '----------' ;............................................................................. ; UNIT (ALL PASS) REVERBERATOR STRUCTURE: ; based on Schroeder as outlined in Griesinger: 'Practical Processors and ; Programs for Digital Reverberation', Audio in Digital Times, 7th AES ; conference, Toronto, Ontario, 1989 ; (the structure outlined in Moorer is a variation of this) ; ; -g ; |\ ; +---------->| >--------------------------+ ; | |/ | ; | | ; | V ; unit | .-----. .--------. |\ .-----. unit ; input ------+--->| sum |-->| delay |--+--->| >-->| sum |-----> output ; '-----' '--------' | |/ '-----' ; ^ | 1-g**2 ; | g | ; | /| | ; +-----< |---------+ ; \| ; ;........................................................................... ; ; one multi-tap fir structure - to handle early reflections ; ; followed by 4 parallel comb (iir) filters (each comb having ; a first order LPF in its feedback loop ; ; followed by an 'allpass' reverberator whose output is then ; delayed so that its first output follows after the last "early ; reflection" output ; ;__________________________________________________________________________ ; opt cex page 132 ;--------------------------------------------------constant declarations adc equ $ffef ; ADC address dac equ $ffef ; DAC address ntap equ 7 ; number of taps ntapp10 equ ntap+10 ; # taps + 10 for other variables tapmod equ ntapp10-1 ; number of taps minus 1 dlymx equ 4000 ; length of delay line in samples cmbdly1 equ 2205 ; COMB FILTER CONSTANTS cmbdly2 equ 2690 ; delay values (in samples) cmbdly3 equ 3175 cmbdly4 equ 3440 cmbmod1 equ 3490 ; modulo for comb #1 delay line THESE ARE cmbmod2 equ 3490 ; " " #2 " " ALSO THE cmbmod3 equ 3490 ; " " #3 " " MAX DELAYS cmbmod4 equ 3490 ; " " #4 " " ALLOWED FOR ; CHOSEN DSM ; VALUE untrvbdly equ 265 ; delay UNIT (or ALLPASS) unt_g equ 0.7 ; gains (g) REVERBERATOR neg_g equ -unt_g ; (-g) one_m_g2 equ (1-unt_g*unt_g) ; " (1-g**2) cmb_g equ 0.86 ; (OVERALL) COMB FILTER FEEDBACK GAIN ; controls reverb decay time: smaller ; values give quicker decay, larger ; values yield slower decay lpf1 equ 0.408 ; lpf filter coefficient in the lpf2 equ 0.448 ; feedback loop of the combs, lpf3 equ 0.476 ; these LPF's simulate the high freq lpf4 equ 0.496 ; attentuation in real acoustic reverb fdbck1 equ cmb_g*(1-lpf1) ; actual comb feedback gain fdbck2 equ cmb_g*(1-lpf2) ; is determined by the lpf filter fdbck3 equ cmb_g*(1-lpf3) ; coefficient and the overall feed- fdbck4 equ cmb_g*(1-lpf4) ; back gain to insure stability aligndly equ 1305 ; alignment delay (see diagram above) alignmod equ 1390 ; " mod reverbg equ 0.35 ; reverberation output gain value firg equ 0.15 ; early rflctn FIR output gain value dryg equ 0.9999-(reverbg+firg) ; dry signal gain is rest ; CHOOSE THE MIX YOU LIKE org x:0 chead1 dsm 3500 ; allocates 3500 data spaces for use as chead2 dsm 3500 ; comb filter tap delay lines, CHEAD1 refers chead3 dsm 3500 ; to the Comb filter HEAD (beginning) for chead4 dsm 3500 ; filter #1 org x:$0f00 ofst_bf dsm ntapp10 ; this reserves "ntap" contiguous addresses org x:ofst_bf ; starting at the closest appropriate modulo ofst0 dc 1 ; x address space. Then starts filling the ofst1 dc 877 ; offset buffer at this location with ofst2 dc 1561 ; the right delay values for the ofst3 dc 1716 ; EARLY REFLECTION FIR ofst4 dc 1826 ofst5 dc 3083 ; IMPORTANT - to get no delay for the first ofst6 dc 3510 ; early reflection, use a "1". This is due the use ; of an instruction code pre-decrement, a "0" value ; will cause a maximum delay equal to the tap delay ; line length untdly dc untrvbdly ; define the UNIT REVERBERATOR delay untmod dc untrvbdly-1 ; modulo for " " algndly dc aligndly algnmod dc alignmod org y:$0f00 gain_bf dsm ntapp10 org y:gain_bf ; starting at y address zero this reserves ; "ntap" contiguous addresses then starts ; filling these addresses with the desired gain0 dc 0.213 ; tap "gain" values for EARLY REFLECTION FIR gain1 dc 0.217 ; recommended by moorer gain2 dc 0.174 ; gain3 dc 0.135 ; it is recommended to keep many early gain4 dc 0.153 ; reflection returns, as these are fed into gain5 dc 0.157 ; comb filters, than can contribute greatly gain6 dc 0.051 ; to the later impulse response density c1g2 dc fdbck1 c2g2 dc fdbck2 c3g2 dc fdbck3 c4g2 dc fdbck4 c1out ds 1 untg2 dc neg_g untg3 dc one_m_g2 untg1 dc unt_g firgain dc firg drygain dc dryg rvbgain dc reverbg org y:0 c1g1 dc lpf1 ; comb #1 g1 (LPF gain) c2g1 dc lpf2 ; #2 c3g1 dc lpf3 ; #3 c4g1 dc lpf4 ; #4 sample ds 1 ; input sample storage address firout ds 1 ; early ref FIR OUTput storage address ; COMB filter output storage address c2out ds 1 ; " " #2 " " " lpfst1 ds 1 ; " " #1 Low Pass Filter state lpfst2 ds 1 ; " " #2 " " " " lpfst3 ds 1 ; " " #3 " " " " lpfst4 ds 1 ; " " #4 " " " " ; allocates modulo memory for the unit udlyln dsm 300 ; (allpass) reverberator delay line algndlyln dsm 1400 ; allocates modulo memory for alignment delay ; line dlyline dsm dlymx ; allocates mod memory for the FIR delay line ;-------------------------------------------------------------------------- org p:$40 ; program start address ; Set up ADS board in case of force break instead of force reset movep #0,x:$FFFE ;set bcr to zero movec #0,sp ;init stack pointer movec #0,sr ;clear loop flag ; Set up the SSI for operation with the DSP56ADC16EVB ; The following code sets port C to function as SCI/SSI move #$0,a0 ;zero PCC to cycle it movep a0,x:$FFE1 move #$0001ff,a0 movep a0,x:$FFE1 ;write PCC ; The following code sets the SSI CRA and CRB control registers for external ; continuous clock, synchronous, normal mode. move #$004000,a0 ;CRA pattern for word length=16 bits movep a0,x:$FFEC move #$003200,a0 ;CRB pattern for continous ck,sych,normal mode movep a0,x:$FFED ;word long frame sync: FSL=0;ext ck/fs ; ------------------------------------------------------------------------- ; initialize registers MO,RO, etc move #dlymx-1,m0 ; tap line modulus move #dlyline,r0 ; start of delay line move #0,n0 ; move #tapmod,m1 ; tap gain line modulo move #gain_bf,r1 ; tap gain & tap delay pointer move #cmbmod1,m2 ; initialize the modulo registers move #cmbmod2,m3 ; for comb filter buffers 1 thru 5 move #cmbmod3,m4 move #cmbmod4,m5 move #cmbdly1,n2 ; initialize the offset register move #cmbdly2,n3 ; for Comb filters 1 thru 5 move #cmbdly3,n4 move #cmbdly4,n5 move #chead1,r2 ; initialize the pointer values for move #chead2,r3 ; Comb filters 1 thru 5 move #chead3,r4 move #chead4,r5 move x:algnmod,m6 ; initialize alignment delay line move x:algndly,n6 ; offset, modulo, and pointer registers move #algndlyln,r6 move x:untmod,m7 ; initialize unit reverberator (allpass) move x:untdly,n7 ; offset, modulo, and pointer registers move #udlyln,r7 ;--------------------------------------------------------------------------- ; The following code polls the RDF flag in the SSI-SR and waits for RDF=1 ; and then reads the RX register to retrieve the data from the A/D converter. ; Sample rate is controlled by DSP56ADC16 board. poll jclr #7,x:$FFEE,poll ;loop until RDF bit = 1 movep x:adc,a ;get A/D converter data ;--------------------------------------------------------------------------- asr a asr a ; obtain data sample ; and shift right for headroom move a,y:sample ; keep dry sample ; - - - - - - - - - - - - - - - - - - - - - - - - ; FIR for early reflections move a,y:(r0)- ; push sample into delay line clr a x:(r1),n0 ; clr A and load offset into N0 do #ntap,fir ; loop over the (ntaps-1),exclude fb tap now move y:(r1)+,y1 ; " " " gain " " Y move y:(r0+n0),x1 ; get delayed sample mac x1,y1,a x:(r1),n0 ; MAC gain and sample & update offset fir move a,y:firout ; save FIR OUTput to y memory ; - - - - - - - - - - - - - - - - - - - - - - - - ; COMB 1 move x:(r2+n2),b ; get delayed sample & put in b move b,y:c1out ; move output to y memory space move y:c1g1,x1 ; get LPF filter coeff move y:lpfst1,y1 ; get LPF filter state mac x1,y1,b y:(r1)+,x1 ; compute LPF output move b,y:lpfst1 ; save LPF output to LPF state move b,y1 ; put LPF output in Y1 mac x1,y1,a x:(r3+n3),b ; compute feedback term in A move a,x:(r2)- ; store output into delay queue ; - - - - - - - - - - - - - - - - - - - - - - - - ; COMB 2 move y:firout,a ; comb filter #1 with LPF in feed back move b,y:c2out ; move output to y memory space move y:c2g1,x1 ; get LPF filter coeff move y:lpfst2,y1 ; get LPF filter state mac x1,y1,b y:(r1)+,x1 ; compute LPF output move b,y:lpfst2 ; save LPF output to LPF state move b,y1 ; put LPF output in Y1 mac x1,y1,a x:(r4+n4),b ; compute feedback term in A move a,x:(r3)- ; store output into delay queue ; - - - - - - - - - - - - - - - - - - - - - - - - ; COMB 3 move y:firout,a ; comb filter #1 with LPF in feed back move b,x0 move y:c3g1,x1 ; get LPF filter coeff move y:lpfst3,y1 ; get LPF filter state mac x1,y1,b y:(r1)+,x1 ; compute LPF output move b,y:lpfst3 ; save LPF output to LPF state move b,y1 ; put LPF output in Y1 mac x1,y1,a x:(r5+n5),b ; compute feedback term in A move a,x:(r4)- ; store output into delay queue ; - - - - - - - - - - - - - - - - - ; COMB 4 move y:firout,a ; move b,y0 move y:c4g1,x1 ; get LPF filter coeff move y:lpfst4,y1 ; get LPF filter state mac x1,y1,b y:(r1)+,x1 ; compute LPF output move b,y:lpfst4 ; save LPF output to LPF state move b,y1 ; put LPF output in Y1 mac x1,y1,a y:(r1)+,x1 ; compute feedback term in A,c1out to x1 move a,x:(r5)- ; store output into delay queue ; - - - - - - - - - - - - - - - - - - - - - - - - move y:c2out,a ; add output of add four combs add x1,a ; c2out + c1out add x0,a y:(r7+n7),x0 ; +c3out add y0,a y:(r1)+,y1 ; +c4out ; asr a ; - - - - - - - - - - - - - - - - - - ; all pass unit reverberator move a,x1 mpy x1,y1,b y:(r1)+,y1 mac x0,y1,b y:(r1)+,y1 ; MAC into b for output (B CONTAINS OUTPUT) move b,y:(r6)- mac x1,y1,a y:(r6+n6),y1 move a,y:(r7)- ; put new+g*delayed into delay line/inc r7 ; - - - - - - - - - - - - - move y:rvbgain,y0 mpy y1,y0,a y:(r1)+,x1 move y:firout,y1 mac x1,y1,a y:(r1)+,x1 move y:sample,y1 ; mix in the dry sample mac x1,y1,a ; A now contains the output sample asr a ; - - - - - - - - - - - - - - - - - - - - - - - - ; Write DSP56ADC16 A/D converter data to the PCM-56 move a,x:dac ;write the PCM-56 D/A via SSI xmt reg. jmp poll ;loop indefinitely end