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authorThom Johansen <thomj@rockbox.org>2007-02-26 00:41:26 +0000
committerThom Johansen <thomj@rockbox.org>2007-02-26 00:41:26 +0000
commita7fabf0741c91fb0a2c28b2d8357bcc4630300af (patch)
treee1528a67f63933aa23ebb7b5d809e1ce7e60b589 /apps/eq.c
parent1915c1099431294ca9c43bc11fb1bfa41bbd83cc (diff)
downloadrockbox-a7fabf0741c91fb0a2c28b2d8357bcc4630300af.tar.gz
rockbox-a7fabf0741c91fb0a2c28b2d8357bcc4630300af.zip
Add software based bass/treble controls for targets which have no such functionality in hardware (currently only X5). They can also be used on any other SWCODEC target by adding #define HAVE_SW_TONE_CONTROLS in the relevant config-*.h file. Also remove some now unneeded zero checks when using get_replaygain_int(). Comments on sound quality are welcome as some parameters can still be fine-tuned.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12489 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps/eq.c')
-rw-r--r--apps/eq.c60
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)