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author | Adam Gashlin <agashlin@gmail.com> | 2008-02-25 21:47:56 +0000 |
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committer | Adam Gashlin <agashlin@gmail.com> | 2008-02-25 21:47:56 +0000 |
commit | 2668547a554da72093e28d69ea1294fb471e1c7e (patch) | |
tree | 3fc315a07996ba1805480f51f1620089ff10c609 /apps | |
parent | 0380bec8aff48417256d12df162fb413d43506b6 (diff) | |
download | rockbox-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.c | 185 |
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; |