1 files changed, 56 insertions, 4 deletions

diff --git a/apps/eq.c b/apps/eq.c
index 588c23f89f..1d74db790e 100644
--- a/apps/eq.c
+++ b/apps/eq.c
@@ -7,7 +7,7 @@
  *                     \/            \/     \/    \/            \/
  * $Id$
  *
- * Copyright (C) 2006 Thom Johansen 
+ * Copyright (C) 2006-2007 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.
@@ -127,7 +127,7 @@ static long fsincos(unsigned long phase, long *cos) {
  * @param an gain at Nyquist frequency. s3.27 fixed point.
  * @param c pointer to coefficient storage. The coefs are s0.31 format.
  */
-void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
+void filter_shelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
 {
     const long one = 1 << 27;
     long a0, a1;
@@ -137,7 +137,7 @@ void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
     cs = one + (cs >> 4);
 
     /* For max A = 4 (24 dB) */
-    b0 = FRACMUL_SHL(an, cs, 4) + FRACMUL_SHL(ad, s, 4);
+    b0 = FRACMUL_SHL(ad, s, 4) + FRACMUL_SHL(an, cs, 4);
     b1 = FRACMUL_SHL(ad, s, 4) - FRACMUL_SHL(an, cs, 4);
     a0 = s + cs;
     a1 = s - cs;
@@ -147,6 +147,58 @@ void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
     c[2] = -DIV64(a1, a0, 31);
 }
 
+/** 
+ * Calculate second order section filter consisting of one low-shelf and one
+ * high-shelf section.
+ * @param cutoff_low low-shelf midpoint frequency. See eq_pk_coefs for format.
+ * @param cutoff_high high-shelf midpoint frequency.
+ * @param A_low decibel value multiplied by ten, describing gain/attenuation of
+ * low-shelf part. Max value is 24 dB.
+ * @param A_high decibel value multiplied by ten, describing gain/attenuation of
+ * high-shelf part. Max value is 24 dB.
+ * @param A decibel value multiplied by ten, describing additional overall gain.
+ * @param c pointer to coefficient storage. Coefficients are s4.27 format.
+ */
+void filter_bishelf_coefs(unsigned long cutoff_low, unsigned long cutoff_high,
+                          long A_low, long A_high, long A, int32_t *c)
+{
+    long sin1, cos2;        /* s0.31 */
+    long cos1, sin2;        /* s3.28 */
+    int32_t b0, b1, b2, b3; /* s3.28 */
+    int32_t a0, a1, a2, a3;
+    const long gd = get_replaygain_int(A_low*5) << 4; /* 10^(db/40), s3.28 */
+    const long gn = get_replaygain_int(A_high*5) << 4; /* 10^(db/40), s3.28 */
+    const long g = get_replaygain_int(A*10) << 7; /* 10^(db/20), s0.31 */
+
+    sin1 = fsincos(cutoff_low/2, &cos1);
+    sin2 = fsincos(cutoff_high/2, &cos2) >> 3;
+    cos1 >>= 3;
+
+    /* lowshelf filter, ranges listed are for all possible cutoffs */
+    b0 = FRACMUL(sin1, gd) + cos1;   /* 0.25 .. 4.10 */
+    b1 = FRACMUL(sin1, gd) - cos1;   /* -1 .. 3.98 */
+    a0 = DIV64(sin1, gd, 25) + cos1; /* 0.25 .. 4.10 */
+    a1 = DIV64(sin1, gd, 25) - cos1; /* -1 .. 3.98 */
+
+    /* highshelf filter */
+    b2 = sin2 + FRACMUL(cos2, gn);   /* 0.25 .. 4.10 */
+    b3 = sin2 - FRACMUL(cos2, gn);   /* -3.98 .. 1 */
+    a2 = sin2 + DIV64(cos2, gn, 25); /* 0.25 .. 4.10 */
+    a3 = sin2 - DIV64(cos2, gn, 25); /* -3.98 .. 1 */
+
+    /* now we cascade the two first order filters to one second order filter
+     * which can be used by eq_filter(). these resulting coefficients have a
+     * really wide numerical range, so we use a fixed point format which will
+     * work for the selected cutoff frequencies (in dsp.c) only.
+     */
+    const int32_t rcp_a0 = DIV64(1, FRACMUL(a0, a2), 53); /* s3.28 */
+    *c++ = FRACMUL(g, FRACMUL_SHL(FRACMUL(b0, b2), rcp_a0, 5));
+    *c++ = FRACMUL(g, FRACMUL_SHL(FRACMUL(b0, b3) + FRACMUL(b1, b2), rcp_a0, 5));
+    *c++ = FRACMUL(g, FRACMUL_SHL(FRACMUL(b1, b3), rcp_a0, 5));
+    *c++ = -FRACMUL_SHL(FRACMUL(a0, a3) + FRACMUL(a1, a2), rcp_a0, 5);
+    *c++ = -FRACMUL_SHL(FRACMUL(a1, a3), rcp_a0, 5);
+}
+
 /* 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
@@ -162,7 +214,7 @@ void filter_bishelf_coefs(unsigned long cutoff, long ad, long an, int32_t *c)
  * @param Q Q factor value multiplied by ten. Lower bound is artificially set
  * at 0.5.
  * @param db decibel value multiplied by ten, describing gain/attenuation at
- * peak freq.
+ * peak freq. Max value is 24 dB.
  * @param c pointer to coefficient storage. Coefficients are s3.28 format.
  */
 void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, int32_t *c)