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
This commit is contained in:
Thom Johansen
parent
b0302f0cbb
commit
a8cc6a7454
4 changed files with 305 additions and 0 deletions
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@ -76,4 +76,8 @@ playback.c
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metadata.c
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codecs.c
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dsp.c
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eq.c
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#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
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eq_cf.S
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#endif
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#endif
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34
apps/dsp.c
34
apps/dsp.c
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@ -19,6 +19,7 @@
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#include <inttypes.h>
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#include <string.h>
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#include "dsp.h"
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#include "eq.h"
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#include "kernel.h"
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#include "playback.h"
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#include "system.h"
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@ -166,10 +167,21 @@ struct crossfeed_data
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int index;
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};
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/* Current setup is one lowshelf filters, three peaking filters and one
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highshelf filter. Varying the number of shelving filters make no sense,
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but adding peaking filters are possible. */
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struct eq_state {
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char enabled[5]; /* Flags for active filters */
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struct eqfilter ls;
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struct eqfilter pk[3];
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struct eqfilter hs;
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};
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static struct dsp_config dsp_conf[2] IBSS_ATTR;
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static struct dither_data dither_data[2] IBSS_ATTR;
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static struct resample_data resample_data[2] IBSS_ATTR;
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struct crossfeed_data crossfeed_data IBSS_ATTR;
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static struct eq_state eq_data;
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static int pitch_ratio = 1000;
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@ -608,6 +620,25 @@ static void apply_crossfeed(long* src[], int count)
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}
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#endif
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/* Apply EQ filters to those bands that have got it switched on. */
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void eq_process(long **x, unsigned num)
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{
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int i;
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unsigned int channels = dsp->stereo_mode != STEREO_MONO ? 2 : 1;
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/* filter configuration currently is 1 low shelf filter, 3 band peaking
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filters and 1 high shelf filter, in that order.
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*/
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if (eq_data.enabled[0])
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eq_filter(x, &eq_data.ls, num, channels, EQ_SHELF_SHIFT);
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for (i = 0; i < 3; i++) {
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if (eq_data.enabled[1 + i])
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eq_filter(x, &eq_data.pk[i], num, channels, EQ_PEAK_SHIFT);
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}
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if (eq_data.enabled[4])
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eq_filter(x, &eq_data.hs, num, channels, EQ_SHELF_SHIFT);
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}
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/* Apply a constant gain to the samples (e.g., for ReplayGain). May update
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* the src array if gain was applied.
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* Note that this must be called before the resampler.
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@ -713,6 +744,9 @@ long dsp_process(char* dst, char* src[], long size)
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samples = resample(tmp, samples);
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if (dsp->crossfeed_enabled && dsp->stereo_mode != STEREO_MONO)
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apply_crossfeed(tmp, samples);
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/* TODO: Might want to wrap this with a generic eq_enabled when the
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settings are in place */
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eq_process(tmp, samples);
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write_samples((short*) dst, tmp, samples);
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written += samples;
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dst += samples * sizeof(short) * 2;
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226
apps/eq.c
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226
apps/eq.c
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@ -0,0 +1,226 @@
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/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2006 Thom Johansen
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*
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* All files in this archive are subject to the GNU General Public License.
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* See the file COPYING in the source tree root for full license agreement.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "config.h"
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#include "eq.h"
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/* Coef calculation taken from Audio-EQ-Cookbook.txt by Robert Bristow-Johnson.
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Slightly faster calculation can be done by deriving forms which use tan()
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instead of cos() and sin(), but the latter are far easier to use when doing
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fixed point math, and performance is not a big point in the calculation part.
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We realise the filters as a second order direct form 1 structure. Direct
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form 1 was chosen because of better numerical properties for fixed point
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implementations.
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*/
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#define DIV64(x, y, z) (long)(((long long)(x) << (z))/(y))
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/* TODO: This macro requires the EMAC unit to be in fractional mode
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when the coef generator routines are called. If this can be guaranteeed,
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then remove the "&& 0" below for faster coef calculation on Coldfire.
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*/
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#if defined(CPU_COLDFIRE) && !defined(SIMULATOR) && 0
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#define FRACMUL(x, y) \
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({ \
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long t; \
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asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \
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"movclr.l %%acc0, %[t]\n\t" \
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: [t] "=r" (t) : [a] "r" (x), [b] "r" (y)); \
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t; \
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})
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#else
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#define FRACMUL(x, y) ((long)(((((long long) (x)) * ((long long) (y))) >> 31)))
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#endif
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/* TODO: replaygain.c has some fixed point routines. perhaps we could reuse
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them? */
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/* 128 sixteen bit sine samples + guard point */
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short sinetab[] = {
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0, 1607, 3211, 4807, 6392, 7961, 9511, 11038, 12539, 14009, 15446, 16845,
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18204, 19519, 20787, 22004, 23169, 24278, 25329, 26318, 27244, 28105, 28897,
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29621, 30272, 30851, 31356, 31785, 32137, 32412, 32609,32727, 32767, 32727,
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32609, 32412, 32137, 31785, 31356, 30851, 30272, 29621, 28897, 28105, 27244,
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26318, 25329, 24278, 23169, 22004, 20787, 19519, 18204, 16845, 15446, 14009,
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12539, 11038, 9511, 7961, 6392, 4807, 3211, 1607, 0, -1607, -3211, -4807,
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-6392, -7961, -9511, -11038, -12539, -14009, -15446, -16845, -18204, -19519,
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-20787, -22004, -23169, -24278, -25329, -26318, -27244, -28105, -28897,
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-29621, -30272, -30851, -31356, -31785, -32137, -32412, -32609, -32727,
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-32767, -32727, -32609, -32412, -32137, -31785, -31356, -30851, -30272,
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-29621, -28897, -28105, -27244, -26318, -25329, -24278, -23169, -22004,
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-20787, -19519, -18204, -16845, -15446, -14009, -12539, -11038, -9511,
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-7961, -6392, -4807, -3211, -1607, 0
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};
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/* Good quality sine calculated by linearly interpolating
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* a 128 sample sine table. First harmonic has amplitude of about -84 dB.
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* phase has range from 0 to 0xffffffff, representing 0 and
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* 2*pi respectively.
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* Return value is a signed value from LONG_MIN to LONG_MAX, representing
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* -1 and 1 respectively.
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*/
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static long fsin(unsigned long phase)
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{
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unsigned int pos = phase >> 25;
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unsigned short frac = (phase & 0x01ffffff) >> 9;
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short diff = sinetab[pos + 1] - sinetab[pos];
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return (sinetab[pos] << 16) + frac*diff;
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}
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static inline long fcos(unsigned long phase)
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{
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return fsin(phase + 0xffffffff/4);
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}
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/* Fixed point square root via Newton-Raphson.
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* Output is in same fixed point format as input.
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* fracbits specifies number of fractional bits in argument.
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*/
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static long fsqrt(long a, unsigned int fracbits)
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{
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long b = a/2 + (1 << fracbits); /* initial approximation */
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unsigned n;
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const unsigned iterations = 4;
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for (n = 0; n < iterations; ++n)
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b = (b + DIV64(a, b, fracbits))/2;
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return b;
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}
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short dbtoatab[49] = {
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2058, 2180, 2309, 2446, 2591, 2744, 2907, 3079, 3261, 3455, 3659, 3876,
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4106, 4349, 4607, 4880, 5169, 5475, 5799, 6143, 6507, 6893, 7301, 7734,
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8192, 8677, 9192, 9736, 10313, 10924, 11572, 12257, 12983, 13753, 14568,
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15431, 16345, 17314, 18340, 19426, 20577, 21797, 23088, 24456, 25905, 27440,
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29066, 30789, 32613
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};
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/* Function for converting dB to squared amplitude factor (A = 10^(dB/40)).
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Range is -24 to 24 dB. If gain values outside of this is needed, the above
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table needs to be extended.
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Parameter format is s15.16 fixed point. Return format is s2.29 fixed point.
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*/
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static long dbtoA(long db)
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{
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const unsigned long bias = 24 << 16;
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unsigned short frac = (db + bias) & 0x0000ffff;
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unsigned short pos = (db + bias) >> 16;
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short diff = dbtoatab[pos + 1] - dbtoatab[pos];
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return (dbtoatab[pos] << 16) + frac*diff;
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}
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/* Calculate second order section peaking filter coefficients.
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cutoff is a value from 0 to 0xffffffff, where 0 represents 0 hz and
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0xffffffff represents nyquist (samplerate/2).
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Q is an unsigned 6.26 fixed point number, lower bound is artificially set
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at 0.5.
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db is s15.16 fixed point and describes gain/attenuation at peak freq.
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c is a pointer where the coefs will be stored.
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*/
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void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, long *c)
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{
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const long one = 1 << 28; /* s3.28 */
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const long A = dbtoA(db);
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const long alpha = DIV64(fsin(cutoff), 2*Q, 25); /* s1.30 */
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long a0, a1, a2; /* these are all s3.28 format */
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long b0, b1, b2;
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/* possible numerical ranges listed after each coef */
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b0 = one + FRACMUL(alpha, A); /* [1.25..5] */
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b1 = a1 = -2*(fcos(cutoff) >> 3); /* [-2..2] */
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b2 = one - FRACMUL(alpha, A); /* [-3..0.75] */
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a0 = one + DIV64(alpha, A, 27); /* [1.25..5] */
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a2 = one - DIV64(alpha, A, 27); /* [-3..0.75] */
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c[0] = DIV64(b0, a0, 28);
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c[1] = DIV64(b1, a0, 28);
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c[2] = DIV64(b2, a0, 28);
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c[3] = DIV64(a1, a0, 28);
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c[4] = DIV64(a2, a0, 28);
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}
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/* Calculate coefficients for lowshelf filter */
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void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, long *c)
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{
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const long one = 1 << 24; /* s7.24 */
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const long A = dbtoA(db);
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const long alpha = DIV64(fsin(cutoff), 2*Q, 25); /* s1.30 */
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const long ap1 = (A >> 5) + one;
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const long am1 = (A >> 5) - one;
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const long twosqrtalpha = 2*(FRACMUL(fsqrt(A >> 5, 24), alpha) << 1);
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long a0, a1, a2; /* these are all s7.24 format */
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long b0, b1, b2;
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long cs = fcos(cutoff);
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b0 = FRACMUL(A, ap1 - FRACMUL(am1, cs) + twosqrtalpha) << 2;
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b1 = FRACMUL(A, am1 - FRACMUL(ap1, cs)) << 3;
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b2 = FRACMUL(A, ap1 - FRACMUL(am1, cs) - twosqrtalpha) << 2;
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a0 = ap1 + FRACMUL(am1, cs) + twosqrtalpha;
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a1 = -2*((am1 + FRACMUL(ap1, cs)));
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a2 = ap1 + FRACMUL(am1, cs) - twosqrtalpha;
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c[0] = DIV64(b0, a0, 24);
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c[1] = DIV64(b1, a0, 24);
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c[2] = DIV64(b2, a0, 24);
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c[3] = DIV64(a1, a0, 24);
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c[4] = DIV64(a2, a0, 24);
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}
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/* Calculate coefficients for highshelf filter */
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void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c)
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{
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const long one = 1 << 24; /* s7.24 */
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const long A = dbtoA(db);
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const long alpha = DIV64(fsin(cutoff), 2*Q, 25); /* s1.30 */
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const long ap1 = (A >> 5) + one;
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const long am1 = (A >> 5) - one;
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const long twosqrtalpha = 2*(FRACMUL(fsqrt(A >> 5, 24), alpha) << 1);
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long a0, a1, a2; /* these are all s7.24 format */
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long b0, b1, b2;
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long cs = fcos(cutoff);
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b0 = FRACMUL(A, ap1 + FRACMUL(am1, cs) + twosqrtalpha) << 2;
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b1 = -FRACMUL(A, am1 + FRACMUL(ap1, cs)) << 3;
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b2 = FRACMUL(A, ap1 + FRACMUL(am1, cs) - twosqrtalpha) << 2;
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a0 = ap1 - FRACMUL(am1, cs) + twosqrtalpha;
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a1 = 2*((am1 - FRACMUL(ap1, cs)));
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a2 = ap1 - FRACMUL(am1, cs) - twosqrtalpha;
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c[0] = DIV64(b0, a0, 24);
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c[1] = DIV64(b1, a0, 24);
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c[2] = DIV64(b2, a0, 24);
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c[3] = DIV64(a1, a0, 24);
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c[4] = DIV64(a2, a0, 24);
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}
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#if !defined(CPU_COLDFIRE) || defined(SIMULATOR)
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void eq_filter(long **x, struct eqfilter *f, unsigned num,
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unsigned channels, unsigned shift)
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{
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/* TODO: Implement generic filtering routine. */
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(void)x;
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(void)f;
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(void)num;
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(void)channels;
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(void)shift;
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}
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#endif
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41
apps/eq.h
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41
apps/eq.h
Normal file
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@ -0,0 +1,41 @@
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/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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||||
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2006 Thom Johansen
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*
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* All files in this archive are subject to the GNU General Public License.
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* See the file COPYING in the source tree root for full license agreement.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#ifndef _EQ_H
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#define _EQ_H
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/* These depend on the fixed point formats used by the different filter types
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and need to be changed when they change.
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*/
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#define EQ_PEAK_SHIFT 3
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#define EQ_SHELF_SHIFT 7
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struct eqfilter {
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long coefs[5]; /* Order is b0, b1, b2, a1, a2 */
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long history[2][4];
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};
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void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
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void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
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void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
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void eq_filter(long **x, struct eqfilter *f, unsigned num,
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unsigned samples, unsigned shift);
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#endif
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