From 65721f0b3573460d306528ff34aa395c45e94ea3 Mon Sep 17 00:00:00 2001 From: Thom Johansen Date: Sun, 29 Jan 2006 17:52:13 +0000 Subject: Slight change of coef format. Removed erronous hard code of channel number in EQ filtering routine and added some minor changes. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8486 a1c6a512-1295-4272-9138-f99709370657 --- apps/eq.c | 14 ++++++++------ apps/eq_cf.S | 17 ++++++++++------- 2 files changed, 18 insertions(+), 13 deletions(-) diff --git a/apps/eq.c b/apps/eq.c index 8ad886fc0c..3d2f8d133d 100644 --- a/apps/eq.c +++ b/apps/eq.c @@ -24,6 +24,8 @@ Slightly faster calculation can be done by deriving forms which use tan() instead of cos() and sin(), but the latter are far easier to use when doing fixed point math, and performance is not a big point in the calculation part. + All the 'a' filter coefficients are negated so we can use only additions + in the filtering equation. We realise the filters as a second order direct form 1 structure. Direct form 1 was chosen because of better numerical properties for fixed point implementations. @@ -153,8 +155,8 @@ void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, long *c) c[0] = DIV64(b0, a0, 28); c[1] = DIV64(b1, a0, 28); c[2] = DIV64(b2, a0, 28); - c[3] = DIV64(a1, a0, 28); - c[4] = DIV64(a2, a0, 28); + c[3] = DIV64(-a1, a0, 28); + c[4] = DIV64(-a2, a0, 28); } /* Calculate coefficients for lowshelf filter */ @@ -180,8 +182,8 @@ void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, long *c) c[0] = DIV64(b0, a0, 24); c[1] = DIV64(b1, a0, 24); c[2] = DIV64(b2, a0, 24); - c[3] = DIV64(a1, a0, 24); - c[4] = DIV64(a2, a0, 24); + c[3] = DIV64(-a1, a0, 24); + c[4] = DIV64(-a2, a0, 24); } /* Calculate coefficients for highshelf filter */ @@ -207,8 +209,8 @@ void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c) c[0] = DIV64(b0, a0, 24); c[1] = DIV64(b1, a0, 24); c[2] = DIV64(b2, a0, 24); - c[3] = DIV64(a1, a0, 24); - c[4] = DIV64(a2, a0, 24); + c[3] = DIV64(-a1, a0, 24); + c[4] = DIV64(-a2, a0, 24); } #if !defined(CPU_COLDFIRE) || defined(SIMULATOR) diff --git a/apps/eq_cf.S b/apps/eq_cf.S index 3876ca72d6..0a34d7707e 100644 --- a/apps/eq_cf.S +++ b/apps/eq_cf.S @@ -26,21 +26,24 @@ eq_filter: movem.l (11*4+16, %sp), %d6-%d7 | load num. channels and shift count movem.l (%a5), %a0-%a4 | load coefs lea.l (5*4, %a5), %a5 | point to filter history - moveq.l #2, %d6 | number of channels (hardcode to stereo) .filterloop: move.l (11*4+4, %sp), %a6 | load input channel pointer + addq.l #4, (11*4+4, %sp) | point x to next channel move.l (%a6), %a6 move.l (11*4+12, %sp), %d5 | number of samples - addq.l #4, (11*4+4, %sp) | point x to next channel movem.l (%a5), %d0-%d3 | load filter history .loop: - move.l (%a6), %d4 - mac.l %a0, %d4, %acc0 | acc = b0*x[i] - mac.l %a1, %d0, %acc0 | acc += b1*x[i - 1] + /* Direct form 1 filtering code. We assume DSP has put EMAC in frac mode. + y[n] = b0*x[i] + b1*x[i - 1] + b2*x[i - 2] + a1*y[i - 1] + a2*y[i - 2], + where y[] is output and x[] is input. This is performed out of order + to do parallel load of input value. + */ + mac.l %a1, %d0, (%a6), %d4, %acc0 | acc = b1*x[i - 1], x[i] -> d4 mac.l %a2, %d1, %acc0 | acc += b2*x[i - 2] - msac.l %a3, %d2, %acc0 | acc -= a1*y[i - 1] - msac.l %a4, %d3, %acc0 | acc -= a2*y[i - 2] + mac.l %a0, %d4, %acc0 | acc += b0*x[i] + mac.l %a3, %d2, %acc0 | acc += a1*y[i - 1] + mac.l %a4, %d3, %acc0 | acc += a2*y[i - 2] move.l %d0, %d1 | fix history move.l %d4, %d0 move.l %d2, %d3 -- cgit v1.2.3