rockbox/apps/dsp.c
Miika Pekkarinen d8cb703b1e Initial DSP implementation. DSP supports resampling audio stream from
codecs (currently works corrently only with mp3's, somebody should fix
that).


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6877 a1c6a512-1295-4272-9138-f99709370657
2005-06-26 19:41:29 +00:00

397 lines
10 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Miika Pekkarinen
*
* 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.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "kernel.h"
#include "logf.h"
#include "dsp.h"
#include "playback.h"
#include "system.h"
/* The "dither" code to convert the 24-bit samples produced by libmad was
taken from the coolplayer project - coolplayer.sourceforge.net */
struct s_dither {
int error[3];
int random;
};
static struct s_dither dither[2];
struct dsp_configuration dsp_config;
static int channel;
static int fracbits;
#define SAMPLE_DEPTH 16
/*
* NAME: prng()
* DESCRIPTION: 32-bit pseudo-random number generator
*/
static __inline
unsigned long prng(unsigned long state)
{
return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
}
inline long dsp_noiseshape(long sample)
{
sample += dither[channel].error[0] - dither[channel].error[1]
+ dither[channel].error[2];
dither[channel].error[2] = dither[channel].error[1];
dither[channel].error[1] = dither[channel].error[0]/2;
return sample;
}
inline long dsp_bias(long sample)
{
sample = sample + (1L << (fracbits - SAMPLE_DEPTH));
return sample;
}
inline long dsp_dither(long *mask)
{
long random, output;
random = prng(dither[channel].random);
output = (random & *mask) - (dither[channel].random & *mask);
dither[channel].random = random;
return output;
}
inline void dsp_clip(long *sample, long *output)
{
if (*output > dsp_config.clip_max) {
*output = dsp_config.clip_max;
if (*sample > dsp_config.clip_max)
*sample = dsp_config.clip_max;
} else if (*output < dsp_config.clip_min) {
*output = dsp_config.clip_min;
if (*sample < dsp_config.clip_min)
*sample = dsp_config.clip_min;
}
}
/*
* NAME: dither()
* DESCRIPTION: dither and scale sample
*/
inline int scale_dither_clip(long sample)
{
unsigned int scalebits;
long output, mask;
/* noise shape */
sample = dsp_noiseshape(sample);
/* bias */
output = dsp_bias(sample);
scalebits = fracbits + 1 - SAMPLE_DEPTH;
mask = (1L << scalebits) - 1;
/* dither */
output += dsp_dither(&mask);
/* clip */
dsp_clip(&sample, &output);
/* quantize */
output &= ~mask;
/* error feedback */
dither->error[0] = sample - output;
/* scale */
return output >> scalebits;
}
inline int scale_clip(long sample)
{
unsigned int scalebits;
long output, mask;
output = sample;
scalebits = fracbits + 1 - SAMPLE_DEPTH;
mask = (1L << scalebits) - 1;
dsp_clip(&sample, &output);
output &= ~mask;
return output >> scalebits;
}
void dsp_scale_dither_clip(short *dest, long *src, int samplecount)
{
dest += channel;
while (samplecount-- > 0) {
*dest = scale_dither_clip(*src);
src++;
dest += 2;
}
}
void dsp_scale_clip(short *dest, long *src, int samplecount)
{
dest += channel;
while (samplecount-- > 0) {
*dest = scale_clip(*src);
src++;
dest += 2;
}
}
struct resampler {
long last_sample, phase, delta;
};
static struct resampler resample[2];
#if CONFIG_CPU==MCF5249 && !defined(SIMULATOR)
#define INIT() asm volatile ("move.l #0xb0, %macsr") /* frac, round, clip */
#define FRACMUL(x, y) \
({ \
long t; \
asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \
"movclr.l %%acc0, %[t]\n\t" \
: [t] "=r" (t) : [a] "r" (x), [b] "r" (y)); \
t; \
})
#else
#define INIT()
#define FRACMUL(x, y) (long)(((long long)(x)*(long long)(y)) << 1)
#endif
/* linear resampling, introduces one sample delay, because of our inability to
look into the future at the end of a frame */
long downsample(long *out, long *in, int num, struct resampler *s)
{
long i = 1, pos;
long last = s->last_sample;
INIT();
pos = s->phase >> 16;
/* check if we need last sample of previous frame for interpolation */
if (pos > 0)
last = in[pos - 1];
out[0] = last + FRACMUL((s->phase & 0xffff) << 15, in[pos] - last);
s->phase += s->delta;
while ((pos = s->phase >> 16) < num) {
out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
s->phase += s->delta;
}
/* wrap phase accumulator back to start of next frame */
s->phase -= num << 16;
s->last_sample = in[num - 1];
return i;
}
long upsample(long *out, long *in, int num, struct resampler *s)
{
long i = 0, pos;
INIT();
while ((pos = s->phase >> 16) == 0) {
out[i++] = s->last_sample + FRACMUL((s->phase & 0xffff) << 15, in[pos] - s->last_sample);
s->phase += s->delta;
}
while ((pos = s->phase >> 16) < num) {
out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
s->phase += s->delta;
}
/* wrap phase accumulator back to start of next frame */
s->phase -= num << 16;
s->last_sample = in[num - 1];
return i;
}
#define MAX_CHUNK_SIZE 1024
static char samplebuf[MAX_CHUNK_SIZE*4];
/* enough to cope with 11khz upsampling */
long resampled[MAX_CHUNK_SIZE * 4];
int process(short *dest, long *src, int samplecount)
{
long *p;
int length = samplecount;
p = resampled;
/* Resample as necessary */
if (dsp_config.frequency > NATIVE_FREQUENCY)
length = upsample(resampled, src, samplecount, &resample[channel]);
else if (dsp_config.frequency < NATIVE_FREQUENCY)
length = downsample(resampled, src, samplecount, &resample[channel]);
else
p = src;
/* Scale & dither */
if (dsp_config.dither_enabled) {
dsp_scale_dither_clip(dest, p, length);
} else {
dsp_scale_clip(dest, p, length);
}
return length;
}
void convert_stereo_mode(long *dest, long *src, int samplecount)
{
int i;
samplecount /= 2;
for (i = 0; i < samplecount; i++) {
dest[i] = src[i*2 + 0];
dest[i+samplecount] = src[i*2 + 1];
}
}
/* Not yet functional. */
void scale_up(long *dest, short *src, int samplecount)
{
int i;
for (i = 0; i < samplecount; i++)
dest[i] = (long)(src[i] << 8);
}
void scale_up_convert_stereo_mode(long *dest, short *src, int samplecount)
{
int i;
samplecount /= 2;
for (i = 0; i < samplecount; i++) {
dest[i] = (long)(src[i*2+0] << SAMPLE_DEPTH);
dest[i+samplecount] = (long)(src[i*2+1] << SAMPLE_DEPTH);
//dest[i] = (long)(((src[i*2 + 0] << 8)&0x7fff) | ((1L << 31) & src[i*2+0]<<15));
//dest[i+samplecount] = (long)(((src[i*2 + 1] << 8)&0x7fff) | ((1L << 31) & src[i*2+1]<<15));
}
}
int dsp_process(char *dest, char *src, int samplecount)
{
int copy_n, rc;
char *p;
int processed_bytes = 0;
fracbits = dsp_config.sample_depth;
while (samplecount > 0) {
yield();
copy_n = MIN(MAX_CHUNK_SIZE / 4, samplecount);
p = src;
/* Scale up to 32-bit samples. */
if (dsp_config.sample_depth <= SAMPLE_DEPTH) {
if (dsp_config.stereo_mode == STEREO_INTERLEAVED)
scale_up_convert_stereo_mode((long *)samplebuf,
(short *)p, copy_n);
else
scale_up((long *)samplebuf, (short *)p, copy_n);
p = samplebuf;
fracbits = 31;
}
/* Convert to non-interleaved stereo. */
else if (dsp_config.stereo_mode == STEREO_INTERLEAVED) {
convert_stereo_mode((long *)samplebuf, (long *)p, copy_n);
p = samplebuf;
}
/* Apply DSP functions. */
if (dsp_config.stereo_mode == STEREO_INTERLEAVED) {
channel = 0;
rc = process((short *)dest, (long *)p, copy_n / 2) * 4;
p += copy_n * 2;
channel = 1;
process((short *)dest, (long *)p, copy_n / 2);
dest += rc;
} else {
rc = process((short *)dest, (long *)p, copy_n) * 2;
dest += rc * 2;
}
samplecount -= copy_n;
if (dsp_config.sample_depth <= SAMPLE_DEPTH)
src += copy_n * 2;
else
src += copy_n * 4;
processed_bytes += rc;
}
/* Set stereo channel */
channel = channel ? 0 : 1;
return processed_bytes;
}
bool dsp_configure(int setting, void *value)
{
switch (setting) {
case DSP_SET_FREQUENCY:
dsp_config.frequency = (int)value;
resample[0].delta = resample[1].delta =
(unsigned long)value*65536/NATIVE_FREQUENCY;
break ;
case DSP_SET_CLIP_MIN:
dsp_config.clip_min = (long)value;
break ;
case DSP_SET_CLIP_MAX:
dsp_config.clip_max = (long)value;
break ;
case DSP_SET_SAMPLE_DEPTH:
dsp_config.sample_depth = (long)value;
break ;
case DSP_SET_STEREO_MODE:
dsp_config.stereo_mode = (long)value;
channel = 0;
break ;
case DSP_RESET:
dsp_config.dither_enabled = false;
dsp_config.clip_max = 0x7fffffff;
dsp_config.clip_min = 0x80000000;
dsp_config.frequency = NATIVE_FREQUENCY;
channel = 0;
break ;
case DSP_DITHER:
dsp_config.dither_enabled = (bool)value;
break ;
default:
return 0;
}
return 1;
}