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rockbox/apps/codecs/adx.c
Daniel Stenberg 2acc0ac542 Updated our source code header to explicitly mention that we are GPL v2 or 
later. We still need to hunt down snippets used that are not. 1324 modified
files...
http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
2008-06-28 18:10:04 +00:00

507 lines
16 KiB
C
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
*
* Copyright (C) 2006-2008 Adam Gashlin (hcs)
* Copyright (C) 2006 Jens Arnold
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#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)
/* 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)
{
int channels;
int sampleswritten, i;
uint8_t *buf;
int32_t ch1_1, ch1_2, ch2_1, ch2_2; /* ADPCM history */
size_t n;
int endofstream; /* end of stream flag */
uint32_t avgbytespersec;
int looping; /* looping flag */
int loop_count; /* number of loops done so far */
int fade_count; /* countdown for fadeout */
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 */
ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
next_track:
DEBUGF("ADX: next_track\n");
if (codec_init()) {
return CODEC_ERROR;
}
DEBUGF("ADX: after init\n");
/* init history */
ch1_1=ch1_2=ch2_1=ch2_2=0;
/* wait for track info to load */
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
codec_set_replaygain(ci->id3);
/* Get header */
DEBUGF("ADX: request initial buffer\n");
ci->seek_buffer(0);
buf = ci->request_buffer(&n, 0x38);
if (!buf || n < 0x38) {
return CODEC_ERROR;
}
bufoff = 0;
DEBUGF("ADX: read size = %lx\n",(unsigned long)n);
/* Get file header for starting offset, channel count */
chanstart = ((buf[2] << 8) | buf[3]) + 4;
channels = buf[7];
/* useful for seeking and reporting current playback position */
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=%ld ",(long)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;
if (!memcmp(buf+0x10,"\x01\xF4\x03\x00",4)) {
/* Soul Calibur 2 style (type 03) */
DEBUGF("ADX: type 03 found\n");
/* check if header is too small for loop data */
if (chanstart-6 < 0x2c) looping=0;
else {
looping = (buf[0x18]) ||
(buf[0x19]) ||
(buf[0x1a]) ||
(buf[0x1b]);
end_adr = (buf[0x28]<<24) |
(buf[0x29]<<16) |
(buf[0x2a]<<8) |
(buf[0x2b]);
start_adr = (
(buf[0x1c]<<24) |
(buf[0x1d]<<16) |
(buf[0x1e]<<8) |
(buf[0x1f])
)/32*channels*18+chanstart;
}
} else if (!memcmp(buf+0x10,"\x01\xF4\x04\x00",4)) {
/* Standard (type 04) */
DEBUGF("ADX: type 04 found\n");
/* check if header is too small for loop data */
if (chanstart-6 < 0x38) looping=0;
else {
looping = (buf[0x24]) ||
(buf[0x25]) ||
(buf[0x26]) ||
(buf[0x27]);
end_adr = (buf[0x34]<<24) |
(buf[0x35]<<16) |
(buf[0x36]<<8) |
buf[0x37];
start_adr = (
(buf[0x28]<<24) |
(buf[0x29]<<16) |
(buf[0x2a]<<8) |
(buf[0x2b])
)/32*channels*18+chanstart;
}
} else {
DEBUGF("ADX: error, couldn't determine ADX type\n");
return CODEC_ERROR;
}
if (looping) {
DEBUGF("ADX: looped, start: %lx end: %lx\n",start_adr,end_adr);
} else {
DEBUGF("ADX: not looped\n");
}
/* advance to first frame */
DEBUGF("ADX: first frame at %lx\n",chanstart);
bufoff = chanstart;
/* get in position */
ci->seek_buffer(bufoff);
/* setup pcm buffer format */
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
if (channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
} else if (channels == 1) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
} else {
DEBUGF("ADX CODEC_ERROR: more than 2 channels\n");
return CODEC_ERROR;
}
endofstream = 0;
loop_count = 0;
fade_count = -1; /* disable fade */
fade_frames = 1;
/* The main decoder loop */
while (!endofstream) {
ci->yield();
if (ci->stop_codec || ci->new_track) {
break;
}
/* do we need to loop? */
if (bufoff > end_adr-18*channels && looping) {
DEBUGF("ADX: loop!\n");
/* check for endless looping */
if (ci->global_settings->repeat_mode==REPEAT_ONE) {
loop_count=0;
fade_count = -1; /* disable fade */
} else {
/* otherwise start fade after LOOP_TIMES loops */
loop_count++;
if (loop_count >= LOOP_TIMES && fade_count < 0) {
/* frames to fade over */
fade_frames = FADE_LENGTH*ci->id3->frequency/32/1000;
/* volume relative to fade_frames */
fade_count = fade_frames;
DEBUGF("ADX: fade_frames = %d\n",fade_frames);
}
}
bufoff = start_adr;
ci->seek_buffer(bufoff);
}
/* do we need to seek? */
if (ci->seek_time) {
uint32_t newpos;
DEBUGF("ADX: seek to %ldms\n",ci->seek_time);
endofstream = 0;
loop_count = 0;
fade_count = -1; /* disable fade */
fade_frames = 1;
newpos = (((uint64_t)avgbytespersec*(ci->seek_time - 1))
/ (1000LL*18*channels))*(18*channels);
bufoff = chanstart + newpos;
while (bufoff > end_adr-18*channels) {
bufoff-=end_adr-start_adr;
loop_count++;
}
ci->seek_buffer(bufoff);
ci->seek_complete();
}
if (bufoff>ci->filesize-channels*18) break; /* End of stream */
sampleswritten=0;
while (
/* Is there data left in the file? */
(bufoff <= ci->filesize-(18*channels)) &&
/* Is there space in the output buffer? */
(sampleswritten <= WAV_CHUNK_SIZE-(32*channels)) &&
/* Should we be looping? */
((!looping) || bufoff <= end_adr-18*channels))
{
/* decode first/only channel */
int32_t scale;
int32_t ch1_0, d;
/* fetch a frame */
buf = ci->request_buffer(&n, 18);
if (!buf || n!=18) {
DEBUGF("ADX: couldn't get buffer at %lx\n",
bufoff);
return CODEC_ERROR;
}
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 + ((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;
sampleswritten+=channels;
ch1_2 = ch1_1; ch1_1 = ch1_0;
d = buf[i] & 15;
if (d & 8) d -= 16;
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;
sampleswritten+=channels;
ch1_2 = ch1_1; ch1_1 = ch1_0;
}
bufoff+=18;
ci->advance_buffer(18);
if (channels == 2) {
/* decode second channel */
int32_t scale;
int32_t ch2_0, d;
buf = ci->request_buffer(&n, 18);
if (!buf || n!=18) {
DEBUGF("ADX: couldn't get buffer at %lx\n",
bufoff);
return CODEC_ERROR;
}
scale = ((buf[0] << 8)|(buf[1]))+1;
sampleswritten-=63;
for (i = 2; i < 18; i++)
{
d = (buf[i] >> 4) & 15;
if (d & 8) d-= 16;
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;
sampleswritten+=2;
ch2_2 = ch2_1; ch2_1 = ch2_0;
d = buf[i] & 15;
if (d & 8) d -= 16;
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;
sampleswritten+=2;
ch2_2 = ch2_1; ch2_1 = ch2_0;
}
bufoff+=18;
ci->advance_buffer(18);
sampleswritten--; /* go back to first channel's next sample */
}
if (fade_count>0) {
fade_count--;
for (i=0;i<(channels==1?32:64);i++) samples[sampleswritten-i-1]=
((int32_t)samples[sampleswritten-i-1])*fade_count/fade_frames;
if (fade_count==0) {endofstream=1; break;}
}
}
if (channels == 2)
sampleswritten >>= 1; /* make samples/channel */
ci->pcmbuf_insert(samples, NULL, sampleswritten);
ci->set_elapsed(
((end_adr-start_adr)*loop_count + bufoff-chanstart)*
1000LL/avgbytespersec);
}
if (ci->request_next_track())
goto next_track;
return CODEC_OK;
}
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