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authorAdam Gashlin <agashlin@gmail.com>2008-02-25 21:47:56 +0000
committerAdam Gashlin <agashlin@gmail.com>2008-02-25 21:47:56 +0000
commit2668547a554da72093e28d69ea1294fb471e1c7e (patch)
tree3fc315a07996ba1805480f51f1620089ff10c609 /apps
parent0380bec8aff48417256d12df162fb413d43506b6 (diff)
downloadrockbox-2668547a554da72093e28d69ea1294fb471e1c7e.tar.gz
rockbox-2668547a554da72093e28d69ea1294fb471e1c7e.zip
Fix ADX decoder, old constant coefficients were for 44.1khz only, they
are now calculated at runtime. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@16418 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps')
-rw-r--r--apps/codecs/adx.c185
1 files changed, 175 insertions, 10 deletions
diff --git a/apps/codecs/adx.c b/apps/codecs/adx.c
index f558bae135..c3a64b1efe 100644
--- a/apps/codecs/adx.c
+++ b/apps/codecs/adx.c
@@ -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;