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Fix ADX decoder, old constant coefficients were for 44.1khz only, they

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are now calculated at runtime.


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@16418 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Adam Gashlin 2008-02-25 21:47:56 +00:00
parent 0380bec8af
commit 2668547a55
1 changed files with 175 additions and 10 deletions
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185
apps/codecs/adx.c
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@ -6,7 +6,8 @@
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
*
* Copyright (C) 2006-2007 Adam Gashlin (hcs)
* Copyright (C) 2006-2008 Adam Gashlin (hcs)
* Copyright (C) 2006 Jens Arnold
*
* 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.
@ -17,23 +18,145 @@
****************************************************************************/
#include "codeclib.h"
#include "inttypes.h"
#include "math.h"
CODEC_HEADER
/* Maximum number of bytes to process in one iteration */
#define WAV_CHUNK_SIZE (1024*2)
/* Volume for ADX decoder */
#define BASE_VOL 0x2000
/* Number of times to loop looped tracks when repeat is disabled */
#define LOOP_TIMES 2
/* Length of fade-out for looped tracks (milliseconds) */
#define FADE_LENGTH 10000L
/* Default high pass filter cutoff frequency is 500 Hz.
* Others can be set, but the default is nearly always used,
* and there is no way to determine if another was used, anyway.
*/
const long cutoff = 500;
static int16_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;
/* fixed point stuff from apps/plugins/lib/fixedpoint.c */
/* Inverse gain of circular cordic rotation in s0.31 format. */
static const long cordic_circular_gain = 0xb2458939; /* 0.607252929 */
/* Table of values of atan(2^-i) in 0.32 format fractions of pi where pi = 0xffffffff / 2 */
static const unsigned long atan_table[] = {
0x1fffffff, /* +0.785398163 (or pi/4) */
0x12e4051d, /* +0.463647609 */
0x09fb385b, /* +0.244978663 */
0x051111d4, /* +0.124354995 */
0x028b0d43, /* +0.062418810 */
0x0145d7e1, /* +0.031239833 */
0x00a2f61e, /* +0.015623729 */
0x00517c55, /* +0.007812341 */
0x0028be53, /* +0.003906230 */
0x00145f2e, /* +0.001953123 */
0x000a2f98, /* +0.000976562 */
0x000517cc, /* +0.000488281 */
0x00028be6, /* +0.000244141 */
0x000145f3, /* +0.000122070 */
0x0000a2f9, /* +0.000061035 */
0x0000517c, /* +0.000030518 */
0x000028be, /* +0.000015259 */
0x0000145f, /* +0.000007629 */
0x00000a2f, /* +0.000003815 */
0x00000517, /* +0.000001907 */
0x0000028b, /* +0.000000954 */
0x00000145, /* +0.000000477 */
0x000000a2, /* +0.000000238 */
0x00000051, /* +0.000000119 */
0x00000028, /* +0.000000060 */
0x00000014, /* +0.000000030 */
0x0000000a, /* +0.000000015 */
0x00000005, /* +0.000000007 */
0x00000002, /* +0.000000004 */
0x00000001, /* +0.000000002 */
0x00000000, /* +0.000000001 */
0x00000000, /* +0.000000000 */
};
/**
* Implements sin and cos using CORDIC rotation.
*
* @param phase has range from 0 to 0xffffffff, representing 0 and
* 2*pi respectively.
* @param cos return address for cos
* @return sin of phase, value is a signed value from LONG_MIN to LONG_MAX,
* representing -1 and 1 respectively.
*/
static long fsincos(unsigned long phase, long *cos)
{
int32_t x, x1, y, y1;
unsigned long z, z1;
int i;
/* Setup initial vector */
x = cordic_circular_gain;
y = 0;
z = phase;
/* The phase has to be somewhere between 0..pi for this to work right */
if (z < 0xffffffff / 4) {
/* z in first quadrant, z += pi/2 to correct */
x = -x;
z += 0xffffffff / 4;
} else if (z < 3 * (0xffffffff / 4)) {
/* z in third quadrant, z -= pi/2 to correct */
z -= 0xffffffff / 4;
} else {
/* z in fourth quadrant, z -= 3pi/2 to correct */
x = -x;
z -= 3 * (0xffffffff / 4);
}
/* Each iteration adds roughly 1-bit of extra precision */
for (i = 0; i < 31; i++) {
x1 = x >> i;
y1 = y >> i;
z1 = atan_table[i];
/* Decided which direction to rotate vector. Pivot point is pi/2 */
if (z >= 0xffffffff / 4) {
x -= y1;
y += x1;
z -= z1;
} else {
x += y1;
y -= x1;
z += z1;
}
}
if (cos)
*cos = x;
return y;
}
/**
* Fixed point square root via Newton-Raphson.
* @param a square root argument.
* @param fracbits specifies number of fractional bits in argument.
* @return Square root of argument in same fixed point format as input.
*/
static long fsqrt(long a, unsigned int fracbits)
{
long b = a/2 + (1 << fracbits); /* initial approximation */
unsigned n;
const unsigned iterations = 8; /* bumped up from 4 as it wasn't
nearly enough for 28 fractional bits */
for (n = 0; n < iterations; ++n)
b = (b + (long)(((long long)(a) << fracbits)/b))/2;
return b;
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
@ -50,6 +173,8 @@ enum codec_status codec_main(void)
int fade_frames; /* length of fade in frames */
off_t start_adr, end_adr; /* loop points */
off_t chanstart, bufoff;
/*long coef1=0x7298L,coef2=-0x3350L;*/
long coef1, coef2;
/* Generic codec initialisation */
/* we only render 16 bits */
@ -90,6 +215,46 @@ next_track:
avgbytespersec = ci->id3->frequency * 18 * channels / 32;
DEBUGF("avgbytespersec=%ld\n",(unsigned long)avgbytespersec);
/* calculate filter coefficients */
/**
* A simple table of these coefficients would be nice, but
* some very odd frequencies are used and if I'm going to
* interpolate I might as well just go all the way and
* calclate them precisely.
* Speed is not an issue as this only needs to be done once per file.
*/
{
const int64_t big28 = 0x10000000LL;
const int64_t big32 = 0x100000000LL;
int64_t frequency = ci->id3->frequency;
int64_t phasemultiple = cutoff*big32/frequency;
long z;
int64_t a;
const int64_t b = (M_SQRT2*big28)-big28;
int64_t c;
int64_t d;
fsincos((unsigned long)phasemultiple,&z);
a = (M_SQRT2*big28)-(z*big28/LONG_MAX);
/**
* In the long passed to fsqrt there are only 4 nonfractional bits,
* which is sufficient here, but this is the only reason why I don't
* use 32 fractional bits everywhere.
*/
d = fsqrt((a+b)*(a-b)/big28,28);
c = (a-d)*big28/b;
coef1 = (c*8192) >> 28;
coef2 = (c*c/big28*-4096) >> 28;
DEBUGF("ADX: samprate=%lld ",frequency);
DEBUGF("coef1 %04x ",(unsigned int)(coef1*4));
DEBUGF("coef2 %04x\n",(unsigned int)(coef2*-4));
}
/* Get loop data */
looping = 0; start_adr = 0; end_adr = 0;
@ -248,13 +413,13 @@ next_track:
return CODEC_ERROR;
}
scale = (((buf[0] << 8) | (buf[1])) +1) * BASE_VOL;
scale = ((buf[0] << 8) | (buf[1])) +1;
for (i = 2; i < 18; i++)
{
d = (buf[i] >> 4) & 15;
if (d & 8) d-= 16;
ch1_0 = (d*scale + 0x7298L*ch1_1 - 0x3350L*ch1_2) >> 14;
ch1_0 = d*scale + ((coef1*ch1_1 + coef2*ch1_2) >> 12);
if (ch1_0 > 32767) ch1_0 = 32767;
else if (ch1_0 < -32768) ch1_0 = -32768;
samples[sampleswritten] = ch1_0;
@ -263,7 +428,7 @@ next_track:
d = buf[i] & 15;
if (d & 8) d -= 16;
ch1_0 = (d*scale + 0x7298L*ch1_1 - 0x3350L*ch1_2) >> 14;
ch1_0 = d*scale + ((coef1*ch1_1 + coef2*ch1_2) >> 12);
if (ch1_0 > 32767) ch1_0 = 32767;
else if (ch1_0 < -32768) ch1_0 = -32768;
samples[sampleswritten] = ch1_0;
@ -286,7 +451,7 @@ next_track:
return CODEC_ERROR;
}
scale = (((buf[0] << 8)|(buf[1]))+1)*BASE_VOL;
scale = ((buf[0] << 8)|(buf[1]))+1;
sampleswritten-=63;
@ -294,7 +459,7 @@ next_track:
{
d = (buf[i] >> 4) & 15;
if (d & 8) d-= 16;
ch2_0 = (d*scale + 0x7298L*ch2_1 - 0x3350L*ch2_2) >> 14;
ch2_0 = d*scale + ((coef1*ch2_1 + coef2*ch2_2) >> 12);
if (ch2_0 > 32767) ch2_0 = 32767;
else if (ch2_0 < -32768) ch2_0 = -32768;
samples[sampleswritten] = ch2_0;
@ -303,7 +468,7 @@ next_track:
d = buf[i] & 15;
if (d & 8) d -= 16;
ch2_0 = (d*scale + 0x7298L*ch2_1 - 0x3350L*ch2_2) >> 14;
ch2_0 = d*scale + ((coef1*ch2_1 + coef2*ch2_2) >> 12);
if (ch2_0 > 32767) ch2_0 = 32767;
else if (ch2_0 < -32768) ch2_0 = -32768;
samples[sampleswritten] = ch2_0;