d8cb703b1e
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
397 lines
10 KiB
C
397 lines
10 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2005 Miika Pekkarinen
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*
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* All files in this archive are subject to the GNU General Public License.
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* See the file COPYING in the source tree root for full license agreement.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "kernel.h"
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#include "logf.h"
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#include "dsp.h"
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#include "playback.h"
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#include "system.h"
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/* The "dither" code to convert the 24-bit samples produced by libmad was
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taken from the coolplayer project - coolplayer.sourceforge.net */
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struct s_dither {
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int error[3];
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int random;
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};
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static struct s_dither dither[2];
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struct dsp_configuration dsp_config;
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static int channel;
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static int fracbits;
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#define SAMPLE_DEPTH 16
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/*
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* NAME: prng()
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* DESCRIPTION: 32-bit pseudo-random number generator
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*/
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static __inline
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unsigned long prng(unsigned long state)
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{
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return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
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}
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inline long dsp_noiseshape(long sample)
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{
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sample += dither[channel].error[0] - dither[channel].error[1]
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+ dither[channel].error[2];
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dither[channel].error[2] = dither[channel].error[1];
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dither[channel].error[1] = dither[channel].error[0]/2;
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return sample;
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}
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inline long dsp_bias(long sample)
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{
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sample = sample + (1L << (fracbits - SAMPLE_DEPTH));
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return sample;
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}
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inline long dsp_dither(long *mask)
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{
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long random, output;
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random = prng(dither[channel].random);
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output = (random & *mask) - (dither[channel].random & *mask);
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dither[channel].random = random;
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return output;
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}
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inline void dsp_clip(long *sample, long *output)
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{
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if (*output > dsp_config.clip_max) {
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*output = dsp_config.clip_max;
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if (*sample > dsp_config.clip_max)
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*sample = dsp_config.clip_max;
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} else if (*output < dsp_config.clip_min) {
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*output = dsp_config.clip_min;
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if (*sample < dsp_config.clip_min)
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*sample = dsp_config.clip_min;
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}
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}
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/*
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* NAME: dither()
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* DESCRIPTION: dither and scale sample
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*/
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inline int scale_dither_clip(long sample)
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{
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unsigned int scalebits;
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long output, mask;
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/* noise shape */
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sample = dsp_noiseshape(sample);
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/* bias */
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output = dsp_bias(sample);
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scalebits = fracbits + 1 - SAMPLE_DEPTH;
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mask = (1L << scalebits) - 1;
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/* dither */
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output += dsp_dither(&mask);
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/* clip */
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dsp_clip(&sample, &output);
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/* quantize */
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output &= ~mask;
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/* error feedback */
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dither->error[0] = sample - output;
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/* scale */
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return output >> scalebits;
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}
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inline int scale_clip(long sample)
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{
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unsigned int scalebits;
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long output, mask;
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output = sample;
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scalebits = fracbits + 1 - SAMPLE_DEPTH;
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mask = (1L << scalebits) - 1;
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dsp_clip(&sample, &output);
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output &= ~mask;
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return output >> scalebits;
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}
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void dsp_scale_dither_clip(short *dest, long *src, int samplecount)
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{
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dest += channel;
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while (samplecount-- > 0) {
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*dest = scale_dither_clip(*src);
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src++;
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dest += 2;
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}
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}
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void dsp_scale_clip(short *dest, long *src, int samplecount)
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{
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dest += channel;
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while (samplecount-- > 0) {
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*dest = scale_clip(*src);
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src++;
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dest += 2;
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}
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}
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struct resampler {
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long last_sample, phase, delta;
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};
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static struct resampler resample[2];
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#if CONFIG_CPU==MCF5249 && !defined(SIMULATOR)
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#define INIT() asm volatile ("move.l #0xb0, %macsr") /* frac, round, clip */
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#define FRACMUL(x, y) \
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({ \
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long t; \
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asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \
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"movclr.l %%acc0, %[t]\n\t" \
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: [t] "=r" (t) : [a] "r" (x), [b] "r" (y)); \
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t; \
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})
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#else
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#define INIT()
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#define FRACMUL(x, y) (long)(((long long)(x)*(long long)(y)) << 1)
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#endif
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/* linear resampling, introduces one sample delay, because of our inability to
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look into the future at the end of a frame */
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long downsample(long *out, long *in, int num, struct resampler *s)
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{
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long i = 1, pos;
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long last = s->last_sample;
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INIT();
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pos = s->phase >> 16;
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/* check if we need last sample of previous frame for interpolation */
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if (pos > 0)
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last = in[pos - 1];
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out[0] = last + FRACMUL((s->phase & 0xffff) << 15, in[pos] - last);
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s->phase += s->delta;
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while ((pos = s->phase >> 16) < num) {
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out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
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s->phase += s->delta;
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}
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/* wrap phase accumulator back to start of next frame */
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s->phase -= num << 16;
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s->last_sample = in[num - 1];
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return i;
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}
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long upsample(long *out, long *in, int num, struct resampler *s)
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{
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long i = 0, pos;
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INIT();
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while ((pos = s->phase >> 16) == 0) {
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out[i++] = s->last_sample + FRACMUL((s->phase & 0xffff) << 15, in[pos] - s->last_sample);
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s->phase += s->delta;
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}
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while ((pos = s->phase >> 16) < num) {
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out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
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s->phase += s->delta;
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}
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/* wrap phase accumulator back to start of next frame */
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s->phase -= num << 16;
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s->last_sample = in[num - 1];
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return i;
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}
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#define MAX_CHUNK_SIZE 1024
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static char samplebuf[MAX_CHUNK_SIZE*4];
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/* enough to cope with 11khz upsampling */
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long resampled[MAX_CHUNK_SIZE * 4];
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int process(short *dest, long *src, int samplecount)
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{
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long *p;
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int length = samplecount;
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p = resampled;
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/* Resample as necessary */
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if (dsp_config.frequency > NATIVE_FREQUENCY)
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length = upsample(resampled, src, samplecount, &resample[channel]);
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else if (dsp_config.frequency < NATIVE_FREQUENCY)
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length = downsample(resampled, src, samplecount, &resample[channel]);
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else
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p = src;
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/* Scale & dither */
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if (dsp_config.dither_enabled) {
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dsp_scale_dither_clip(dest, p, length);
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} else {
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dsp_scale_clip(dest, p, length);
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}
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return length;
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}
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void convert_stereo_mode(long *dest, long *src, int samplecount)
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{
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int i;
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samplecount /= 2;
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for (i = 0; i < samplecount; i++) {
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dest[i] = src[i*2 + 0];
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dest[i+samplecount] = src[i*2 + 1];
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}
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}
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/* Not yet functional. */
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void scale_up(long *dest, short *src, int samplecount)
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{
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int i;
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for (i = 0; i < samplecount; i++)
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dest[i] = (long)(src[i] << 8);
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}
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void scale_up_convert_stereo_mode(long *dest, short *src, int samplecount)
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{
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int i;
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samplecount /= 2;
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for (i = 0; i < samplecount; i++) {
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dest[i] = (long)(src[i*2+0] << SAMPLE_DEPTH);
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dest[i+samplecount] = (long)(src[i*2+1] << SAMPLE_DEPTH);
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//dest[i] = (long)(((src[i*2 + 0] << 8)&0x7fff) | ((1L << 31) & src[i*2+0]<<15));
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//dest[i+samplecount] = (long)(((src[i*2 + 1] << 8)&0x7fff) | ((1L << 31) & src[i*2+1]<<15));
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}
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}
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int dsp_process(char *dest, char *src, int samplecount)
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{
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int copy_n, rc;
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char *p;
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int processed_bytes = 0;
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fracbits = dsp_config.sample_depth;
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while (samplecount > 0) {
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yield();
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copy_n = MIN(MAX_CHUNK_SIZE / 4, samplecount);
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p = src;
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/* Scale up to 32-bit samples. */
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if (dsp_config.sample_depth <= SAMPLE_DEPTH) {
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if (dsp_config.stereo_mode == STEREO_INTERLEAVED)
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scale_up_convert_stereo_mode((long *)samplebuf,
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(short *)p, copy_n);
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else
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scale_up((long *)samplebuf, (short *)p, copy_n);
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p = samplebuf;
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fracbits = 31;
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}
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/* Convert to non-interleaved stereo. */
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else if (dsp_config.stereo_mode == STEREO_INTERLEAVED) {
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convert_stereo_mode((long *)samplebuf, (long *)p, copy_n);
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p = samplebuf;
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}
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/* Apply DSP functions. */
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if (dsp_config.stereo_mode == STEREO_INTERLEAVED) {
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channel = 0;
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rc = process((short *)dest, (long *)p, copy_n / 2) * 4;
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p += copy_n * 2;
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channel = 1;
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process((short *)dest, (long *)p, copy_n / 2);
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dest += rc;
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} else {
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rc = process((short *)dest, (long *)p, copy_n) * 2;
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dest += rc * 2;
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}
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samplecount -= copy_n;
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if (dsp_config.sample_depth <= SAMPLE_DEPTH)
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src += copy_n * 2;
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else
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src += copy_n * 4;
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processed_bytes += rc;
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}
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/* Set stereo channel */
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channel = channel ? 0 : 1;
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return processed_bytes;
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}
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bool dsp_configure(int setting, void *value)
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{
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switch (setting) {
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case DSP_SET_FREQUENCY:
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dsp_config.frequency = (int)value;
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resample[0].delta = resample[1].delta =
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(unsigned long)value*65536/NATIVE_FREQUENCY;
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break ;
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case DSP_SET_CLIP_MIN:
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dsp_config.clip_min = (long)value;
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break ;
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case DSP_SET_CLIP_MAX:
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dsp_config.clip_max = (long)value;
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break ;
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case DSP_SET_SAMPLE_DEPTH:
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dsp_config.sample_depth = (long)value;
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break ;
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case DSP_SET_STEREO_MODE:
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dsp_config.stereo_mode = (long)value;
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channel = 0;
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break ;
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case DSP_RESET:
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dsp_config.dither_enabled = false;
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dsp_config.clip_max = 0x7fffffff;
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dsp_config.clip_min = 0x80000000;
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dsp_config.frequency = NATIVE_FREQUENCY;
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channel = 0;
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break ;
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case DSP_DITHER:
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dsp_config.dither_enabled = (bool)value;
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break ;
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default:
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return 0;
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}
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return 1;
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}
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