Initial multi-band EQ support for software codec platforms. Now go start

making that user interface!


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8478 a1c6a512-1295-4272-9138-f99709370657

4 files changed, 305 insertions, 0 deletions

diff --git a/apps/SOURCES b/apps/SOURCES
index cf17bbc27c..94c081ec15 100644
--- a/apps/SOURCES
+++ b/apps/SOURCES
@@ -76,4 +76,8 @@ playback.c
 metadata.c
 codecs.c
 dsp.c
+eq.c
+#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
+eq_cf.S
+#endif
 #endif
diff --git a/apps/dsp.c b/apps/dsp.c
index 19cb669a06..b2fc0ce7a2 100644
--- a/apps/dsp.c
+++ b/apps/dsp.c
@@ -19,6 +19,7 @@
 #include <inttypes.h>
 #include <string.h>
 #include "dsp.h"
+#include "eq.h"
 #include "kernel.h"
 #include "playback.h"
 #include "system.h"
@@ -166,10 +167,21 @@ struct crossfeed_data
     int index;
 };
 
+/* Current setup is one lowshelf filters, three peaking filters and one
+   highshelf filter. Varying the number of shelving filters make no sense,
+   but adding peaking filters are possible. */
+struct eq_state {
+    char enabled[5];    /* Flags for active filters */
+    struct eqfilter ls;
+    struct eqfilter pk[3];
+    struct eqfilter hs;
+};
+
 static struct dsp_config dsp_conf[2] IBSS_ATTR;
 static struct dither_data dither_data[2] IBSS_ATTR;
 static struct resample_data resample_data[2] IBSS_ATTR;
 struct crossfeed_data crossfeed_data IBSS_ATTR;
+static struct eq_state eq_data;
 
 static int pitch_ratio = 1000;
 
@@ -608,6 +620,25 @@ static void apply_crossfeed(long* src[], int count)
 }
 #endif
 
+/* Apply EQ filters to those bands that have got it switched on. */
+void eq_process(long **x, unsigned num)
+{
+    int i;
+    unsigned int channels = dsp->stereo_mode != STEREO_MONO ? 2 : 1;
+        
+    /* filter configuration currently is 1 low shelf filter, 3 band peaking
+       filters and 1 high shelf filter, in that order.
+     */
+    if (eq_data.enabled[0])
+        eq_filter(x, &eq_data.ls, num, channels, EQ_SHELF_SHIFT);
+    for (i = 0; i < 3; i++) {
+        if (eq_data.enabled[1 + i])
+            eq_filter(x, &eq_data.pk[i], num, channels, EQ_PEAK_SHIFT);
+    }
+    if (eq_data.enabled[4])
+        eq_filter(x, &eq_data.hs, num, channels, EQ_SHELF_SHIFT);
+}
+
 /* Apply a constant gain to the samples (e.g., for ReplayGain). May update
  * the src array if gain was applied.
  * Note that this must be called before the resampler.
@@ -713,6 +744,9 @@ long dsp_process(char* dst, char* src[], long size)
         samples = resample(tmp, samples);
         if (dsp->crossfeed_enabled && dsp->stereo_mode != STEREO_MONO)
             apply_crossfeed(tmp, samples);
+        /* TODO: Might want to wrap this with a generic eq_enabled when the
+           settings are in place */
+        eq_process(tmp, samples);
         write_samples((short*) dst, tmp, samples);
         written += samples;
         dst += samples * sizeof(short) * 2;
diff --git a/apps/eq.c b/apps/eq.c
new file mode 100644
index 0000000000..8ad886fc0c
--- /dev/null
+++ b/apps/eq.c
@@ -0,0 +1,226 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2006 Thom Johansen 
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "config.h"
+#include "eq.h"
+
+/* Coef calculation taken from Audio-EQ-Cookbook.txt by Robert Bristow-Johnson.
+   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.
+   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.
+ */
+
+#define DIV64(x, y, z) (long)(((long long)(x) << (z))/(y))
+/* TODO: This macro requires the EMAC unit to be in fractional mode
+   when the coef generator routines are called. If this can be guaranteeed,
+   then remove the "&& 0" below for faster coef calculation on Coldfire.
+ */
+#if defined(CPU_COLDFIRE) && !defined(SIMULATOR) && 0
+#define FRACMUL(x, y) \
+({ \
+    long t; \
+    asm volatile ("mac.l    %[a], %[b], %%acc0\n\t" \
+                  "movclr.l %%acc0, %[t]\n\t" \
+                  : [t] "=r" (t) : [a] "r" (x), [b] "r" (y)); \
+    t; \
+})
+#else
+#define FRACMUL(x, y) ((long)(((((long long) (x)) * ((long long) (y))) >> 31)))
+#endif
+
+/* TODO: replaygain.c has some fixed point routines. perhaps we could reuse
+   them? */
+
+/* 128 sixteen bit sine samples + guard point */
+short sinetab[] = {
+    0, 1607, 3211, 4807, 6392, 7961, 9511, 11038, 12539, 14009, 15446, 16845,
+    18204, 19519, 20787, 22004, 23169, 24278, 25329, 26318, 27244, 28105, 28897,
+    29621, 30272, 30851, 31356, 31785, 32137, 32412, 32609,32727, 32767, 32727,
+    32609, 32412, 32137, 31785, 31356, 30851, 30272, 29621, 28897, 28105, 27244,
+    26318, 25329, 24278, 23169, 22004, 20787, 19519, 18204, 16845, 15446, 14009,
+    12539, 11038, 9511, 7961, 6392, 4807, 3211, 1607, 0, -1607, -3211, -4807,
+    -6392, -7961, -9511, -11038, -12539, -14009, -15446, -16845, -18204, -19519,
+    -20787, -22004, -23169, -24278, -25329, -26318, -27244, -28105, -28897,
+    -29621, -30272, -30851, -31356, -31785, -32137, -32412, -32609, -32727,
+    -32767, -32727, -32609, -32412, -32137, -31785, -31356, -30851, -30272,
+    -29621, -28897, -28105, -27244, -26318, -25329, -24278, -23169, -22004,
+    -20787, -19519, -18204, -16845, -15446, -14009, -12539, -11038, -9511,
+    -7961, -6392, -4807, -3211, -1607, 0
+};
+
+/* Good quality sine calculated by linearly interpolating
+ * a 128 sample sine table. First harmonic has amplitude of about -84 dB.
+ * phase has range from 0 to 0xffffffff, representing 0 and
+ * 2*pi respectively.
+ * Return value is a signed value from LONG_MIN to LONG_MAX, representing
+ * -1 and 1 respectively. 
+ */
+static long fsin(unsigned long phase)
+{
+    unsigned int pos = phase >> 25;
+    unsigned short frac = (phase & 0x01ffffff) >> 9;
+    short diff = sinetab[pos + 1] - sinetab[pos];
+    
+    return (sinetab[pos] << 16) + frac*diff;
+}
+
+static inline long fcos(unsigned long phase)
+{
+    return fsin(phase + 0xffffffff/4);
+}
+
+/* Fixed point square root via Newton-Raphson.
+ * Output is in same fixed point format as input. 
+ * fracbits specifies number of fractional bits in argument.
+ */
+static long fsqrt(long a, unsigned int fracbits)
+{
+    long b = a/2 + (1 << fracbits); /* initial approximation */
+    unsigned n;
+    const unsigned iterations = 4;
+    
+    for (n = 0; n < iterations; ++n)
+        b = (b + DIV64(a, b, fracbits))/2;
+
+    return b;
+}
+
+short dbtoatab[49] = {
+    2058, 2180, 2309, 2446, 2591, 2744, 2907, 3079, 3261, 3455, 3659, 3876,
+    4106, 4349, 4607, 4880, 5169, 5475, 5799, 6143, 6507, 6893, 7301, 7734,
+    8192, 8677, 9192, 9736, 10313, 10924, 11572, 12257, 12983, 13753, 14568,
+    15431, 16345, 17314, 18340, 19426, 20577, 21797, 23088, 24456, 25905, 27440,
+    29066, 30789, 32613
+};
+
+/* Function for converting dB to squared amplitude factor (A = 10^(dB/40)).
+   Range is -24 to 24 dB. If gain values outside of this is needed, the above
+   table needs to be extended.
+   Parameter format is s15.16 fixed point. Return format is s2.29 fixed point.
+ */
+static long dbtoA(long db)
+{
+    const unsigned long bias = 24 << 16;
+    unsigned short frac = (db + bias) & 0x0000ffff;
+    unsigned short pos = (db + bias) >> 16;
+    short diff = dbtoatab[pos + 1] - dbtoatab[pos];
+    
+    return (dbtoatab[pos] << 16) + frac*diff;
+}
+
+/* Calculate second order section peaking filter coefficients.
+   cutoff is a value from 0 to 0xffffffff, where 0 represents 0 hz and
+   0xffffffff represents nyquist (samplerate/2).
+   Q is an unsigned 6.26 fixed point number, lower bound is artificially set
+   at 0.5.
+   db is s15.16 fixed point and describes gain/attenuation at peak freq.
+   c is a pointer where the coefs will be stored.
+ */
+void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, long *c)
+{
+    const long one = 1 << 28; /* s3.28 */
+    const long A = dbtoA(db);
+    const long alpha = DIV64(fsin(cutoff), 2*Q, 25); /* s1.30 */
+    long a0, a1, a2; /* these are all s3.28 format */
+    long b0, b1, b2;
+
+    /* possible numerical ranges listed after each coef */
+    b0 = one + FRACMUL(alpha, A);     /* [1.25..5] */
+    b1 = a1 = -2*(fcos(cutoff) >> 3); /* [-2..2] */
+    b2 = one - FRACMUL(alpha, A);     /* [-3..0.75] */
+    a0 = one + DIV64(alpha, A, 27);   /* [1.25..5] */
+    a2 = one - DIV64(alpha, A, 27);   /* [-3..0.75] */
+
+    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);
+}
+
+/* Calculate coefficients for lowshelf filter */
+void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, long *c)
+{
+    const long one = 1 << 24; /* s7.24 */
+    const long A = dbtoA(db);
+    const long alpha = DIV64(fsin(cutoff), 2*Q, 25); /* s1.30 */
+    const long ap1 = (A >> 5) + one;
+    const long am1 = (A >> 5) - one;
+    const long twosqrtalpha = 2*(FRACMUL(fsqrt(A >> 5, 24), alpha) << 1);
+    long a0, a1, a2; /* these are all s7.24 format */
+    long b0, b1, b2;
+    long cs = fcos(cutoff);
+
+    b0 = FRACMUL(A, ap1 - FRACMUL(am1, cs) + twosqrtalpha) << 2;
+    b1 = FRACMUL(A, am1 - FRACMUL(ap1, cs)) << 3; 
+    b2 = FRACMUL(A, ap1 - FRACMUL(am1, cs) - twosqrtalpha) << 2; 
+    a0 = ap1 + FRACMUL(am1, cs) + twosqrtalpha; 
+    a1 = -2*((am1 + FRACMUL(ap1, cs))); 
+    a2 = ap1 + FRACMUL(am1, cs) - twosqrtalpha;
+
+    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);
+}
+
+/* Calculate coefficients for highshelf filter */
+void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c)
+{
+    const long one = 1 << 24; /* s7.24 */
+    const long A = dbtoA(db);
+    const long alpha = DIV64(fsin(cutoff), 2*Q, 25); /* s1.30 */
+    const long ap1 = (A >> 5) + one;
+    const long am1 = (A >> 5) - one;
+    const long twosqrtalpha = 2*(FRACMUL(fsqrt(A >> 5, 24), alpha) << 1);
+    long a0, a1, a2; /* these are all s7.24 format */
+    long b0, b1, b2;
+    long cs = fcos(cutoff);
+
+    b0 = FRACMUL(A, ap1 + FRACMUL(am1, cs) + twosqrtalpha) << 2;
+    b1 = -FRACMUL(A, am1 + FRACMUL(ap1, cs)) << 3;
+    b2 = FRACMUL(A, ap1 + FRACMUL(am1, cs) - twosqrtalpha) << 2;
+    a0 = ap1 - FRACMUL(am1, cs) + twosqrtalpha;
+    a1 = 2*((am1 - FRACMUL(ap1, cs)));
+    a2 = ap1 - FRACMUL(am1, cs) - twosqrtalpha;
+
+    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);
+}
+
+#if !defined(CPU_COLDFIRE) || defined(SIMULATOR)
+void eq_filter(long **x, struct eqfilter *f, unsigned num,
+               unsigned channels, unsigned shift)
+{
+    /* TODO: Implement generic filtering routine. */
+    (void)x;
+    (void)f;
+    (void)num;
+    (void)channels;
+    (void)shift;
+}
+#endif
+
diff --git a/apps/eq.h b/apps/eq.h
new file mode 100644
index 0000000000..f7616387b4
--- /dev/null
+++ b/apps/eq.h
@@ -0,0 +1,41 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2006 Thom Johansen
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#ifndef _EQ_H
+#define _EQ_H
+
+/* These depend on the fixed point formats used by the different filter types
+   and need to be changed when they change.
+ */
+#define EQ_PEAK_SHIFT 3
+#define EQ_SHELF_SHIFT 7
+
+struct eqfilter {
+    long coefs[5];        /* Order is b0, b1, b2, a1, a2 */
+    long history[2][4];
+};
+
+void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
+void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
+void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
+void eq_filter(long **x, struct eqfilter *f, unsigned num,
+               unsigned samples, unsigned shift);
+
+#endif
+