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rockbox/apps/dsp.c

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/***************************************************************************
* __________ __ ___.
* 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 <inttypes.h>
#include <string.h>
#include <sound.h>
#include "dsp.h"
#include "eq.h"
#include "kernel.h"
#include "playback.h"
#include "system.h"
#include "settings.h"
#include "replaygain.h"
#include "misc.h"
#include "debug.h"
#ifndef SIMULATOR
#include <dsp_asm.h>
#endif
/* 16-bit samples are scaled based on these constants. The shift should be
* no more than 15.
*/
#define WORD_SHIFT 12
#define WORD_FRACBITS 27
#define NATIVE_DEPTH 16
#define SAMPLE_BUF_SIZE 256
#define RESAMPLE_BUF_SIZE (256 * 4) /* Enough for 11,025 Hz -> 44,100 Hz*/
#define DEFAULT_GAIN 0x01000000
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
/* Multiply two S.31 fractional integers and return the sign bit and the
* 31 most significant bits of the result.
*/
#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; \
})
/* Multiply one S.31-bit and one S8.23 fractional integer and return the
* sign bit and the 31 most significant bits of the result.
*/
#define FRACMUL_8(x, y) \
({ \
long t; \
long u; \
asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \
"move.l %%accext01, %[u]\n\t" \
"movclr.l %%acc0, %[t]\n\t" \
: [t] "=r" (t), [u] "=r" (u) : [a] "r" (x), [b] "r" (y)); \
(t << 8) | (u & 0xff); \
})
/* Multiply one S.31-bit and one S8.23 fractional integer and return the
* sign bit and the 31 most significant bits of the result. Load next value
* to multiply with into x from s (and increase s); x must contain the
* initial value.
*/
#define FRACMUL_8_LOOP_PART(x, s, d, y) \
{ \
long u; \
asm volatile ("mac.l %[a], %[b], (%[c])+, %[a], %%acc0\n\t" \
"move.l %%accext01, %[u]\n\t" \
"movclr.l %%acc0, %[t]" \
: [a] "+r" (x), [c] "+a" (s), [t] "=r" (d), [u] "=r" (u) \
: [b] "r" (y)); \
d = (d << 8) | (u & 0xff); \
}
#define FRACMUL_8_LOOP(x, y, s, d) \
{ \
long t; \
FRACMUL_8_LOOP_PART(x, s, t, y); \
asm volatile ("move.l %[t],(%[d])+" \
: [d] "+a" (d)\
: [t] "r" (t)); \
}
#define ACC(acc, x, y) \
(void)acc; \
asm volatile ("mac.l %[a], %[b], %%acc0" \
: : [a] "i,r" (x), [b] "i,r" (y));
#define GET_ACC(acc) \
({ \
long t; \
(void)acc; \
asm volatile ("movclr.l %%acc0, %[t]" \
: [t] "=r" (t)); \
t; \
})
#define ACC_INIT(acc, x, y) ACC(acc, x, y)
#elif defined(CPU_ARM) && !defined(SIMULATOR)
/* Multiply two S.31 fractional integers and return the sign bit and the
* 31 most significant bits of the result.
*/
#define FRACMUL(x, y) \
({ \
long t; \
asm volatile ("smull r0, r1, %[a], %[b]\n\t" \
"mov %[t], r1, asl #1\n\t" \
"orr %[t], %[t], r0, lsr #31\n\t" \
: [t] "=r" (t) : [a] "r" (x), [b] "r" (y) : "r0", "r1"); \
t; \
})
#define ACC_INIT(acc, x, y) acc = FRACMUL(x, y)
#define ACC(acc, x, y) acc += FRACMUL(x, y)
#define GET_ACC(acc) acc
/* Multiply one S.31-bit and one S8.23 fractional integer and store the
* sign bit and the 31 most significant bits of the result to d (and
* increase d). Load next value to multiply with into x from s (and
* increase s); x must contain the initial value.
*/
#define FRACMUL_8_LOOP(x, y, s, d) \
({ \
asm volatile ("smull r0, r1, %[a], %[b]\n\t" \
"mov %[t], r1, asl #9\n\t" \
"orr %[t], %[t], r0, lsr #23\n\t" \
: [t] "=r" (*(d)++) : [a] "r" (x), [b] "r" (y) : "r0", "r1"); \
x = *(s)++; \
})
#else
#define ACC_INIT(acc, x, y) acc = FRACMUL(x, y)
#define ACC(acc, x, y) acc += FRACMUL(x, y)
#define GET_ACC(acc) acc
#define FRACMUL(x, y) (long) (((((long long) (x)) * ((long long) (y))) >> 31))
#define FRACMUL_8(x, y) (long) (((((long long) (x)) * ((long long) (y))) >> 23))
#define FRACMUL_8_LOOP(x, y, s, d) \
({ \
long t = x; \
x = *(s)++; \
*(d)++ = (long) (((((long long) (t)) * ((long long) (y))) >> 23)); \
})
#endif
struct dsp_config
{
long codec_frequency; /* Sample rate of data coming from the codec */
long frequency; /* Effective sample rate after pitch shift (if any) */
long clip_min;
long clip_max;
long track_gain;
long album_gain;
long track_peak;
long album_peak;
long replaygain;
int sample_depth;
int sample_bytes;
int stereo_mode;
int frac_bits;
bool dither_enabled;
long dither_bias;
long dither_mask;
bool new_gain;
bool crossfeed_enabled;
bool eq_enabled;
long eq_precut;
long gain; /* Note that this is in S8.23 format. */
};
struct resample_data
{
long phase, delta;
int32_t last_sample[2];
};
struct dither_data
{
long error[3];
long random;
};
struct crossfeed_data
{
int32_t gain; /* Direct path gain */
int32_t coefs[3]; /* Coefficients for the shelving filter */
int32_t history[4]; /* Format is x[n - 1], y[n - 1] for both channels */
int32_t delay[13][2];
int index; /* Current index into the delay line */
};
/* Current setup is one lowshelf filters, three peaking filters and one
highshelf filter. Varying the number of shelving filters make no sense,
but adding peaking filters is possible. */
struct eq_state {
char enabled[5]; /* Flags for active filters */
struct eqfilter filters[5];
};
static struct dsp_config dsp_conf[2] IBSS_ATTR;
static struct dither_data dither_data[2] IBSS_ATTR;
static struct resample_data resample_data[2] IBSS_ATTR;
struct crossfeed_data crossfeed_data IBSS_ATTR;
static struct eq_state eq_data;
static int pitch_ratio = 1000;
static int channels_mode = 0;
static int32_t sw_gain, sw_cross;
extern int current_codec;
struct dsp_config *dsp;
/* The internal format is 32-bit samples, non-interleaved, stereo. This
* format is similar to the raw output from several codecs, so the amount
* of copying needed is minimized for that case.
*/
static int32_t sample_buf[SAMPLE_BUF_SIZE] IBSS_ATTR;
static int32_t resample_buf[RESAMPLE_BUF_SIZE] IBSS_ATTR;
int sound_get_pitch(void)
{
return pitch_ratio;
}
void sound_set_pitch(int permille)
{
pitch_ratio = permille;
dsp_configure(DSP_SWITCH_FREQUENCY, (int *)dsp->codec_frequency);
}
/* Convert at most count samples to the internal format, if needed. Returns
* number of samples ready for further processing. Updates src to point
* past the samples "consumed" and dst is set to point to the samples to
* consume. Note that for mono, dst[0] equals dst[1], as there is no point
* in processing the same data twice.
*/
static int convert_to_internal(const char* src[], int count, int32_t* dst[])
{
count = MIN(SAMPLE_BUF_SIZE / 2, count);
if ((dsp->sample_depth <= NATIVE_DEPTH)
|| (dsp->stereo_mode == STEREO_INTERLEAVED))
{
dst[0] = &sample_buf[0];
dst[1] = (dsp->stereo_mode == STEREO_MONO)
? dst[0] : &sample_buf[SAMPLE_BUF_SIZE / 2];
}
else
{
dst[0] = (int32_t*) src[0];
dst[1] = (int32_t*) ((dsp->stereo_mode == STEREO_MONO) ? src[0] : src[1]);
}
if (dsp->sample_depth <= NATIVE_DEPTH)
{
short* s0 = (short*) src[0];
int32_t* d0 = dst[0];
int32_t* d1 = dst[1];
int scale = WORD_SHIFT;
int i;
if (dsp->stereo_mode == STEREO_INTERLEAVED)
{
for (i = 0; i < count; i++)
{
*d0++ = *s0++ << scale;
*d1++ = *s0++ << scale;
}
}
else if (dsp->stereo_mode == STEREO_NONINTERLEAVED)
{
short* s1 = (short*) src[1];
for (i = 0; i < count; i++)
{
*d0++ = *s0++ << scale;
*d1++ = *s1++ << scale;
}
}
else
{
for (i = 0; i < count; i++)
{
*d0++ = *s0++ << scale;
}
}
}
else if (dsp->stereo_mode == STEREO_INTERLEAVED)
{
int32_t* s0 = (int32_t*) src[0];
int32_t* d0 = dst[0];
int32_t* d1 = dst[1];
int i;
for (i = 0; i < count; i++)
{
*d0++ = *s0++;
*d1++ = *s0++;
}
}
if (dsp->stereo_mode == STEREO_NONINTERLEAVED)
{
src[0] += count * dsp->sample_bytes;
src[1] += count * dsp->sample_bytes;
}
else if (dsp->stereo_mode == STEREO_INTERLEAVED)
{
src[0] += count * dsp->sample_bytes * 2;
}
else
{
src[0] += count * dsp->sample_bytes;
}
return count;
}
static void resampler_set_delta(int frequency)
{
resample_data[current_codec].delta = (unsigned long)
frequency * 65536LL / NATIVE_FREQUENCY;
}
/* Linear interpolation resampling that introduces a one sample delay because
* of our inability to look into the future at the end of a frame.
*/
/* TODO: we really should have a separate set of resample functions for both
mono and stereo to avoid all this internal branching and looping. */
static long downsample(int32_t **dst, int32_t **src, int count,
struct resample_data *r)
{
long phase = r->phase;
long delta = r->delta;
int32_t last_sample;
int32_t *d[2] = { dst[0], dst[1] };
int pos = phase >> 16;
int i = 1, j;
int num_channels = dsp->stereo_mode == STEREO_MONO ? 1 : 2;
for (j = 0; j < num_channels; j++) {
last_sample = r->last_sample[j];
/* Do we need last sample of previous frame for interpolation? */
if (pos > 0)
{
last_sample = src[j][pos - 1];
}
*d[j]++ = last_sample + FRACMUL((phase & 0xffff) << 15,
src[j][pos] - last_sample);
}
phase += delta;
while ((pos = phase >> 16) < count)
{
for (j = 0; j < num_channels; j++)
*d[j]++ = src[j][pos - 1] + FRACMUL((phase & 0xffff) << 15,
src[j][pos] - src[j][pos - 1]);
phase += delta;
i++;
}
/* Wrap phase accumulator back to start of next frame. */
r->phase = phase - (count << 16);
r->delta = delta;
r->last_sample[0] = src[0][count - 1];
r->last_sample[1] = src[1][count - 1];
return i;
}
static long upsample(int32_t **dst, int32_t **src, int count, struct resample_data *r)
{
long phase = r->phase;
long delta = r->delta;
int32_t *d[2] = { dst[0], dst[1] };
int i = 0, j;
int pos;
int num_channels = dsp->stereo_mode == STEREO_MONO ? 1 : 2;
while ((pos = phase >> 16) == 0)
{
for (j = 0; j < num_channels; j++)
*d[j]++ = r->last_sample[j] + FRACMUL((phase & 0xffff) << 15,
src[j][pos] - r->last_sample[j]);
phase += delta;
i++;
}
while ((pos = phase >> 16) < count)
{
for (j = 0; j < num_channels; j++)
*d[j]++ = src[j][pos - 1] + FRACMUL((phase & 0xffff) << 15,
src[j][pos] - src[j][pos - 1]);
phase += delta;
i++;
}
/* Wrap phase accumulator back to start of next frame. */
r->phase = phase - (count << 16);
r->delta = delta;
r->last_sample[0] = src[0][count - 1];
r->last_sample[1] = src[1][count - 1];
return i;
}
/* Resample count stereo samples. Updates the src array, if resampling is
* done, to refer to the resampled data. Returns number of stereo samples
* for further processing.
*/
static inline int resample(int32_t* src[], int count)
{
long new_count;
if (dsp->frequency != NATIVE_FREQUENCY)
{
int32_t* dst[2] = {&resample_buf[0], &resample_buf[RESAMPLE_BUF_SIZE / 2]};
if (dsp->frequency < NATIVE_FREQUENCY)
{
new_count = upsample(dst, src, count,
&resample_data[current_codec]);
}
else
{
new_count = downsample(dst, src, count,
&resample_data[current_codec]);
}
src[0] = dst[0];
if (dsp->stereo_mode != STEREO_MONO)
src[1] = dst[1];
else
src[1] = dst[0];
}
else
{
new_count = count;
}
return new_count;
}
static inline long clip_sample(int32_t sample, int32_t min, int32_t max)
{
if (sample > max)
{
sample = max;
}
else if (sample < min)
{
sample = min;
}
return sample;
}
/* The "dither" code to convert the 24-bit samples produced by libmad was
* taken from the coolplayer project - coolplayer.sourceforge.net
*/
void dsp_dither_enable(bool enable)
{
dsp->dither_enabled = enable;
}
static void dither_init(void)
{
memset(&dither_data[0], 0, sizeof(dither_data));
memset(&dither_data[1], 0, sizeof(dither_data));
dsp->dither_bias = (1L << (dsp->frac_bits - NATIVE_DEPTH));
dsp->dither_mask = (1L << (dsp->frac_bits + 1 - NATIVE_DEPTH)) - 1;
}
static void dither_samples(int32_t* src, int num, struct dither_data* dither)
{
int32_t output, sample;
int32_t random;
int32_t min, max;
long mask = dsp->dither_mask;
long bias = dsp->dither_bias;
int i;
for (i = 0; i < num; ++i) {
/* Noise shape and bias */
sample = src[i];
sample += dither->error[0] - dither->error[1] + dither->error[2];
dither->error[2] = dither->error[1];
dither->error[1] = dither->error[0]/2;
output = sample + bias;
/* Dither */
random = dither->random*0x0019660dL + 0x3c6ef35fL;
output += (random & mask) - (dither->random & mask);
dither->random = random;
/* Clip and quantize */
min = dsp->clip_min;
max = dsp->clip_max;
if (output > max) {
output = max;
if (sample > max)
sample = max;
} else if (output < min) {
output = min;
if (sample < min)
sample = min;
}
output &= ~mask;
/* Error feedback */
dither->error[0] = sample - output;
src[i] = output;
}
}
void dsp_set_crossfeed(bool enable)
{
dsp->crossfeed_enabled = enable;
}
void dsp_set_crossfeed_direct_gain(int gain)
{
/* Work around bug in get_replaygain_int which returns 0 for 0 dB */
if (gain == 0)
crossfeed_data.gain = 0x7fffffff;
else
crossfeed_data.gain = get_replaygain_int(gain * -10) << 7;
}
void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
{
long g1 = get_replaygain_int(lf_gain * -10) << 3;
long g2 = get_replaygain_int(hf_gain * -10) << 3;
filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*cutoff, g1, g2,
crossfeed_data.coefs);
}
/* Applies crossfeed to the stereo signal in src.
* Crossfeed is a process where listening over speakers is simulated. This
* is good for old hard panned stereo records, which might be quite fatiguing
* to listen to on headphones with no crossfeed.
*/
#ifndef DSP_HAVE_ASM_CROSSFEED
void apply_crossfeed(int32_t* src[], int count)
{
int32_t *hist_l = &crossfeed_data.history[0];
int32_t *hist_r = &crossfeed_data.history[2];
int32_t *delay = &crossfeed_data.delay[0][0];
int32_t *coefs = &crossfeed_data.coefs[0];
int32_t gain = crossfeed_data.gain;
int di = crossfeed_data.index;
int32_t acc;
int32_t left, right;
int i;
for (i = 0; i < count; i++) {
left = src[0][i];
right = src[1][i];
/* Filter delayed sample from left speaker */
ACC_INIT(acc, delay[di*2], coefs[0]);
ACC(acc, hist_l[0], coefs[1]);
ACC(acc, hist_l[1], coefs[2]);
/* Save filter history for left speaker */
hist_l[1] = GET_ACC(acc);
hist_l[0] = delay[di*2];
/* Filter delayed sample from right speaker */
ACC_INIT(acc, delay[di*2 + 1], coefs[0]);
ACC(acc, hist_r[0], coefs[1]);
ACC(acc, hist_r[1], coefs[2]);
/* Save filter history for right speaker */
hist_r[1] = GET_ACC(acc);
hist_r[0] = delay[di*2 + 1];
delay[di*2] = left;
delay[di*2 + 1] = right;
/* Now add the attenuated direct sound and write to outputs */
src[0][i] = FRACMUL(left, gain) + hist_r[1];
src[1][i] = FRACMUL(right, gain) + hist_l[1];
/* Wrap delay line index if bigger than delay line size */
if (++di > 12)
di = 0;
}
/* Write back local copies of data we've modified */
crossfeed_data.index = di;
}
#endif
/* Combine all gains to a global gain. */
static void set_gain(void)
{
dsp->gain = DEFAULT_GAIN;
if (dsp->replaygain)
{
dsp->gain = dsp->replaygain;
}
if (dsp->eq_enabled && dsp->eq_precut)
{
dsp->gain = (long) (((int64_t) dsp->gain * dsp->eq_precut) >> 24);
}
if (dsp->gain == DEFAULT_GAIN)
{
dsp->gain = 0;
}
else
{
dsp->gain >>= 1;
}
}
/**
* Use to enable the equalizer.
*
* @param enable true to enable the equalizer
*/
void dsp_set_eq(bool enable)
{
dsp->eq_enabled = enable;
}
/**
* Update the amount to cut the audio before applying the equalizer.
*
* @param precut to apply in decibels (multiplied by 10)
*/
void dsp_set_eq_precut(int precut)
{
dsp->eq_precut = get_replaygain_int(precut * -10);
set_gain();
}
/**
* Synchronize the equalizer filter coefficients with the global settings.
*
* @param band the equalizer band to synchronize
*/
void dsp_set_eq_coefs(int band)
{
const int *setting;
long gain;
unsigned long cutoff, q;
/* Adjust setting pointer to the band we actually want to change */
setting = &global_settings.eq_band0_cutoff + (band * 3);
/* Convert user settings to format required by coef generator functions */
cutoff = 0xffffffff / NATIVE_FREQUENCY * (*setting++);
q = ((*setting++) << 16) / 10; /* 16.16 */
gain = ((*setting++) << 16) / 10; /* s15.16 */
if (q == 0)
q = 1;
/* NOTE: The coef functions assume the EMAC unit is in fractional mode,
which it should be, since we're executed from the main thread. */
/* Assume a band is disabled if the gain is zero */
if (gain == 0) {
eq_data.enabled[band] = 0;
} else {
if (band == 0)
eq_ls_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
else if (band == 4)
eq_hs_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
else
eq_pk_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
eq_data.enabled[band] = 1;
}
}
/* Apply EQ filters to those bands that have got it switched on. */
static void eq_process(int32_t **x, unsigned num)
{
int i;
unsigned int channels = dsp->stereo_mode != STEREO_MONO ? 2 : 1;
unsigned shift;
/* filter configuration currently is 1 low shelf filter, 3 band peaking
filters and 1 high shelf filter, in that order. we need to know this
so we can choose the correct shift factor.
*/
for (i = 0; i < 5; i++) {
if (eq_data.enabled[i]) {
if (i == 0 || i == 4) /* shelving filters */
shift = EQ_SHELF_SHIFT;
else
shift = EQ_PEAK_SHIFT;
eq_filter(x, &eq_data.filters[i], num, channels, shift);
}
}
}
/* Apply a constant gain to the samples (e.g., for ReplayGain). May update
* the src array if gain was applied.
* Note that this must be called before the resampler.
*/
static void apply_gain(int32_t* _src[], int _count)
{
if (dsp->gain)
{
int32_t** src = _src;
int count = _count;
int32_t* s0 = src[0];
int32_t* s1 = src[1];
long gain = dsp->gain;
int32_t s;
int i;
int32_t *d;
if (s0 != s1)
{
d = &sample_buf[SAMPLE_BUF_SIZE / 2];
src[1] = d;
s = *s1++;
for (i = 0; i < count; i++)
FRACMUL_8_LOOP(s, gain, s1, d);
}
else
{
src[1] = &sample_buf[0];
}
d = &sample_buf[0];
src[0] = d;
s = *s0++;
for (i = 0; i < count; i++)
FRACMUL_8_LOOP(s, gain, s0, d);
}
}
void channels_set(int value)
{
channels_mode = value;
}
void stereo_width_set(int value)
{
long width, straight, cross;
width = value * 0x7fffff / 100;
if (value <= 100) {
straight = (0x7fffff + width) / 2;
cross = straight - width;
} else {
/* straight = (1 + width) / (2 * width) */
straight = ((int64_t)(0x7fffff + width) << 22) / width;
cross = straight - 0x7fffff;
}
sw_gain = straight << 8;
sw_cross = cross << 8;
}
/* Implements the different channel configurations and stereo width.
* We might want to combine this with the write_samples stage for efficiency,
* but for now we'll just let it stay as a stage of its own.
*/
static void channels_process(int32_t **src, int num)
{
int i;
int32_t *sl = src[0], *sr = src[1];
if (channels_mode == SOUND_CHAN_STEREO)
return;
switch (channels_mode) {
case SOUND_CHAN_MONO:
for (i = 0; i < num; i++)
sl[i] = sr[i] = sl[i]/2 + sr[i]/2;
break;
case SOUND_CHAN_CUSTOM:
for (i = 0; i < num; i++) {
int32_t left_sample = sl[i];
sl[i] = FRACMUL(sl[i], sw_gain) + FRACMUL(sr[i], sw_cross);
sr[i] = FRACMUL(sr[i], sw_gain) + FRACMUL(left_sample, sw_cross);
}
break;
case SOUND_CHAN_MONO_LEFT:
for (i = 0; i < num; i++)
sr[i] = sl[i];
break;
case SOUND_CHAN_MONO_RIGHT:
for (i = 0; i < num; i++)
sl[i] = sr[i];
break;
case SOUND_CHAN_KARAOKE:
for (i = 0; i < num; i++) {
int32_t left_sample = sl[i]/2;
sl[i] = left_sample - sr[i]/2;
sr[i] = sr[i]/2 - left_sample;
}
break;
}
}
static void write_samples(short* dst, int32_t* src[], int count)
{
int32_t* s0 = src[0];
int32_t* s1 = src[1];
int scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
if (dsp->dither_enabled)
{
dither_samples(src[0], count, &dither_data[0]);
dither_samples(src[1], count, &dither_data[1]);
while (count-- > 0)
{
*dst++ = (short) (*s0++ >> scale);
*dst++ = (short) (*s1++ >> scale);
}
}
else
{
long min = dsp->clip_min;
long max = dsp->clip_max;
while (count-- > 0)
{
*dst++ = (short) (clip_sample(*s0++, min, max) >> scale);
*dst++ = (short) (clip_sample(*s1++, min, max) >> scale);
}
}
}
/* Process and convert src audio to dst based on the DSP configuration,
* reading size bytes of audio data. dst is assumed to be large enough; use
* dst_get_dest_size() to get the required size. src is an array of
* pointers; for mono and interleaved stereo, it contains one pointer to the
* start of the audio data; for non-interleaved stereo, it contains two
* pointers, one for each audio channel. Returns number of bytes written to
* dest.
*/
long dsp_process(char* dst, const char* src[], long size)
{
int32_t* tmp[2];
long written = 0;
long factor;
int samples;
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
/* set emac unit for dsp processing, and save old macsr, we're running in
codec thread context at this point, so can't clobber it */
unsigned long old_macsr = coldfire_get_macsr();
coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
#endif
dsp = &dsp_conf[current_codec];
factor = (dsp->stereo_mode != STEREO_MONO) ? 2 : 1;
size /= dsp->sample_bytes * factor;
dsp_set_replaygain(false);
while (size > 0)
{
samples = convert_to_internal(src, size, tmp);
size -= samples;
apply_gain(tmp, samples);
samples = resample(tmp, samples);
if (dsp->crossfeed_enabled && dsp->stereo_mode != STEREO_MONO)
apply_crossfeed(tmp, samples);
if (dsp->eq_enabled)
eq_process(tmp, samples);
if (dsp->stereo_mode != STEREO_MONO)
channels_process(tmp, samples);
write_samples((short*) dst, tmp, samples);
written += samples;
dst += samples * sizeof(short) * 2;
yield();
}
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
/* set old macsr again */
coldfire_set_macsr(old_macsr);
#endif
return written * sizeof(short) * 2;
}
/* Given size bytes of input data, calculate the maximum number of bytes of
* output data that would be generated (the calculation is not entirely
* exact and rounds upwards to be on the safe side; during resampling,
* the number of samples generated depends on the current state of the
* resampler).
*/
/* dsp_input_size MUST be called afterwards */
long dsp_output_size(long size)
{
dsp = &dsp_conf[current_codec];
if (dsp->sample_depth > NATIVE_DEPTH)
{
size /= 2;
}
if (dsp->frequency != NATIVE_FREQUENCY)
{
size = (long) ((((unsigned long) size * NATIVE_FREQUENCY)
+ (dsp->frequency - 1)) / dsp->frequency);
}
/* round to the next multiple of 2 (these are shorts) */
size = (size + 1) & ~1;
if (dsp->stereo_mode == STEREO_MONO)
{
size *= 2;
}
/* now we have the size in bytes for two resampled channels,
* and the size in (short) must not exceed RESAMPLE_BUF_SIZE to
* avoid resample buffer overflow. One must call dsp_input_size()
* to get the correct input buffer size. */
if (size > RESAMPLE_BUF_SIZE*2)
size = RESAMPLE_BUF_SIZE*2;
return size;
}
/* Given size bytes of output buffer, calculate number of bytes of input
* data that would be consumed in order to fill the output buffer.
*/
long dsp_input_size(long size)
{
dsp = &dsp_conf[current_codec];
/* convert to number of output stereo samples. */
size /= 2;
/* Mono means we need half input samples to fill the output buffer */
if (dsp->stereo_mode == STEREO_MONO)
size /= 2;
/* size is now the number of resampled input samples. Convert to
original input samples. */
if (dsp->frequency != NATIVE_FREQUENCY)
{
/* Use the real resampling delta =
* (unsigned long) dsp->frequency * 65536 / NATIVE_FREQUENCY, and
* round towards zero to avoid buffer overflows. */
size = ((unsigned long)size *
resample_data[current_codec].delta) >> 16;
}
/* Convert back to bytes. */
if (dsp->sample_depth > NATIVE_DEPTH)
size *= 4;
else
size *= 2;
return size;
}
int dsp_stereo_mode(void)
{
dsp = &dsp_conf[current_codec];
return dsp->stereo_mode;
}
bool dsp_configure(int setting, void *value)
{
dsp = &dsp_conf[current_codec];
switch (setting)
{
case DSP_SET_FREQUENCY:
memset(&resample_data[current_codec], 0,
sizeof(struct resample_data));
/* Fall through!!! */
case DSP_SWITCH_FREQUENCY:
dsp->codec_frequency = ((long) value == 0) ? NATIVE_FREQUENCY : (long) value;
/* Account for playback speed adjustment when setting dsp->frequency
if we're called from the main audio thread. Voice UI thread should
not need this feature.
*/
if (current_codec == CODEC_IDX_AUDIO)
dsp->frequency = pitch_ratio * dsp->codec_frequency / 1000;
else
dsp->frequency = dsp->codec_frequency;
resampler_set_delta(dsp->frequency);
break;
case DSP_SET_CLIP_MIN:
dsp->clip_min = (long) value;
break;
case DSP_SET_CLIP_MAX:
dsp->clip_max = (long) value;
break;
case DSP_SET_SAMPLE_DEPTH:
dsp->sample_depth = (long) value;
if (dsp->sample_depth <= NATIVE_DEPTH)
{
dsp->frac_bits = WORD_FRACBITS;
dsp->sample_bytes = sizeof(short);
dsp->clip_max = ((1 << WORD_FRACBITS) - 1);
dsp->clip_min = -((1 << WORD_FRACBITS));
}
else
{
dsp->frac_bits = (long) value;
dsp->sample_bytes = 4; /* samples are 32 bits */
dsp->clip_max = (1 << (long)value) - 1;
dsp->clip_min = -(1 << (long)value);
}
dither_init();
break;
case DSP_SET_STEREO_MODE:
dsp->stereo_mode = (long) value;
break;
case DSP_RESET:
dsp->stereo_mode = STEREO_NONINTERLEAVED;
dsp->clip_max = ((1 << WORD_FRACBITS) - 1);
dsp->clip_min = -((1 << WORD_FRACBITS));
dsp->track_gain = 0;
dsp->album_gain = 0;
dsp->track_peak = 0;
dsp->album_peak = 0;
dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY;
dsp->sample_depth = NATIVE_DEPTH;
dsp->frac_bits = WORD_FRACBITS;
dsp->new_gain = true;
break;
case DSP_SET_TRACK_GAIN:
dsp->track_gain = (long) value;
dsp->new_gain = true;
break;
case DSP_SET_ALBUM_GAIN:
dsp->album_gain = (long) value;
dsp->new_gain = true;
break;
case DSP_SET_TRACK_PEAK:
dsp->track_peak = (long) value;
dsp->new_gain = true;
break;
case DSP_SET_ALBUM_PEAK:
dsp->album_peak = (long) value;
dsp->new_gain = true;
break;
default:
return 0;
}
return 1;
}
void dsp_set_replaygain(bool always)
{
dsp = &dsp_conf[current_codec];
if (always || dsp->new_gain)
{
long gain = 0;
dsp->new_gain = false;
if (global_settings.replaygain || global_settings.replaygain_noclip)
{
bool track_mode = get_replaygain_mode(dsp->track_gain != 0,
dsp->album_gain != 0) == REPLAYGAIN_TRACK;
long peak = (track_mode || !dsp->album_peak)
? dsp->track_peak : dsp->album_peak;
if (global_settings.replaygain)
{
gain = (track_mode || !dsp->album_gain)
? dsp->track_gain : dsp->album_gain;
if (global_settings.replaygain_preamp)
{
long preamp = get_replaygain_int(
global_settings.replaygain_preamp * 10);
gain = (long) (((int64_t) gain * preamp) >> 24);
}
}
if (gain == 0)
{
/* So that noclip can work even with no gain information. */
gain = DEFAULT_GAIN;
}
if (global_settings.replaygain_noclip && (peak != 0)
&& ((((int64_t) gain * peak) >> 24) >= DEFAULT_GAIN))
{
gain = (((int64_t) DEFAULT_GAIN << 24) / peak);
}
if (gain == DEFAULT_GAIN)
{
/* Nothing to do, disable processing. */
gain = 0;
}
}
/* Store in S8.23 format to simplify calculations. */
dsp->replaygain = gain;
set_gain();
}
}
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