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rockbox/apps/dsp.c
Michael Sevakis 36175ac945 SWCODEC: DSP optimizations for conversion to internal format and resampling. Assembly resampling for Coldfire. Word has it ARM will get that soon. 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@12399 a1c6a512-1295-4272-9138-f99709370657
2007-02-19 02:49:26 +00:00

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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 "config.h"
#include <stdbool.h>
#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_COUNT 256
#define RESAMPLE_BUF_COUNT (256 * 4) /* Enough for 11,025 Hz -> 44,100 Hz*/
#define DEFAULT_GAIN 0x01000000
enum
{
CONVERT_LE_NATIVE_I_STEREO = STEREO_INTERLEAVED,
CONVERT_LE_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED,
CONVERT_LE_NATIVE_MONO = STEREO_MONO,
CONVERT_GT_NATIVE_I_STEREO = STEREO_INTERLEAVED + STEREO_NUM_MODES,
CONVERT_GT_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED + STEREO_NUM_MODES,
CONVERT_GT_NATIVE_MONO = STEREO_MONO + STEREO_NUM_MODES,
CONVERT_GT_NATIVE_1ST_INDEX = STEREO_NUM_MODES
};
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 num_channels;
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. */
int (*convert_to_internal)(const char* src[], int32_t* dst[], int count);
};
struct resample_data
{
long phase;
long 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;
static 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_COUNT] IBSS_ATTR;
static int32_t resample_buf[RESAMPLE_BUF_COUNT] IBSS_ATTR;
int sound_get_pitch(void)
{
return pitch_ratio;
}
void sound_set_pitch(int permille)
{
pitch_ratio = permille;
dsp_configure(DSP_SWITCH_FREQUENCY, 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.
*/
/* convert count 16-bit mono to 32-bit mono */
static int convert_lte_native_mono(
const char *src[], int32_t *dst[], int count)
{
count = MIN(SAMPLE_BUF_COUNT/2, count);
const short *s = (short*) src[0];
const short * const send = s + count;
int32_t *d = dst[0] = dst[1] = sample_buf;
const int scale = WORD_SHIFT;
do
{
*d++ = *s++ << scale;
}
while (s < send);
src[0] = (char *)s;
return count;
}
/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
static int convert_lte_native_interleaved_stereo(
const char *src[], int32_t *dst[], int count)
{
count = MIN(SAMPLE_BUF_COUNT/2, count);
const int32_t *s = (int32_t *) src[0];
const int32_t * const send = s + count;
int32_t *dl = dst[0] = sample_buf;
int32_t *dr = dst[1] = sample_buf + SAMPLE_BUF_COUNT/2;
const int scale = WORD_SHIFT;
do
{
short slr = *s++;
#ifdef ROCKBOX_LITTLE_ENDIAN
*dl++ = (slr >> 16) << scale;
*dr++ = (int32_t)(short)slr << scale;
#else /* ROCKBOX_BIG_ENDIAN */
*dl++ = (int32_t)(short)slr << scale;
*dr++ = (slr >> 16) << scale;
#endif
}
while (s < send);
src[0] = (char *)s;
return count;
}
/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
static int convert_lte_native_noninterleaved_stereo(
const char *src[], int32_t *dst[], int count)
{
const short *sl = (short *) src[0];
const short *sr = (short *) src[1];
const short * const slend = sl + count;
int32_t *dl = dst[0] = sample_buf;
int32_t *dr = dst[1] = sample_buf + SAMPLE_BUF_COUNT/2;
const int scale = WORD_SHIFT;
do
{
*dl++ = *sl++ << scale;
*dr++ = *sr++ << scale;
}
while (sl < slend);
src[0] = (char *)sl;
src[1] = (char *)sr;
return count;
}
/* convert count 32-bit mono to 32-bit mono */
static int convert_gt_native_mono(
const char *src[], int32_t *dst[], int count)
{
count = MIN(SAMPLE_BUF_COUNT/2, count);
dst[0] = dst[1] = (int32_t *)src[0];
src[0] = (char *)(dst[0] + count);
return count;
}
/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
static int convert_gt_native_interleaved_stereo(
const char *src[], int32_t *dst[], int count)
{
count = MIN(SAMPLE_BUF_COUNT/2, count);
const int32_t *s = (int32_t *)src[0];
const int32_t * const send = s + 2*count;
int32_t *dl = sample_buf;
int32_t *dr = sample_buf + SAMPLE_BUF_COUNT/2;
dst[0] = dl;
dst[1] = dr;
do
{
*dl++ = *s++;
*dr++ = *s++;
}
while (s < send);
src[0] = (char *)send;
return count;
}
/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
static int convert_gt_native_noninterleaved_stereo(
const char *src[], int32_t *dst[], int count)
{
count = MIN(SAMPLE_BUF_COUNT/2, count);
dst[0] = (int32_t *)src[0];
dst[1] = (int32_t *)src[1];
src[0] = (char *)(dst[0] + count);
src[1] = (char *)(dst[1] + count);
return count;
}
/* set the to-native sample conversion function based on dsp sample parameters */
static void new_sample_conversion(void)
{
static int (*convert_to_internal_functions[])(
const char* src[], int32_t *dst[], int count) =
{
[CONVERT_LE_NATIVE_MONO] = convert_lte_native_mono,
[CONVERT_LE_NATIVE_I_STEREO] = convert_lte_native_interleaved_stereo,
[CONVERT_LE_NATIVE_NI_STEREO] = convert_lte_native_noninterleaved_stereo,
[CONVERT_GT_NATIVE_MONO] = convert_gt_native_mono,
[CONVERT_GT_NATIVE_I_STEREO] = convert_gt_native_interleaved_stereo,
[CONVERT_GT_NATIVE_NI_STEREO] = convert_gt_native_noninterleaved_stereo,
};
int convert = dsp->stereo_mode;
if (dsp->sample_depth > NATIVE_DEPTH)
convert += CONVERT_GT_NATIVE_1ST_INDEX;
dsp->convert_to_internal = convert_to_internal_functions[convert];
}
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.
*/
#ifndef DSP_HAVE_ASM_RESAMPLING
static int dsp_downsample(int channels, int count, struct resample_data *r,
int32_t **src, int32_t **dst)
{
long delta = r->delta;
long phase, pos;
int32_t *d;
/* Rolled channel loop actually showed slightly faster. */
do
{
/* Just initialize things and not worry too much about the relatively
* uncommon case of not being able to spit out a sample for the frame.
*/
int32_t *s = src[--channels];
int32_t last = r->last_sample[channels];
r->last_sample[channels] = s[count - 1];
d = dst[channels];
phase = r->phase;
pos = phase >> 16;
/* Do we need last sample of previous frame for interpolation? */
if (pos > 0)
last = s[pos - 1];
while (pos < count)
{
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
phase += delta;
pos = phase >> 16;
last = s[pos - 1];
}
}
while (channels > 0);
/* Wrap phase accumulator back to start of next frame. */
r->phase = phase - (count << 16);
return d - dst[0];
}
static int dsp_upsample(int channels, int count, struct resample_data *r,
int32_t **src, int32_t **dst)
{
long delta = r->delta;
long phase, pos;
int32_t *d;
/* Rolled channel loop actually showed slightly faster. */
do
{
/* Should always be able to output a sample for a ratio up to
RESAMPLE_BUF_COUNT / SAMPLE_BUF_COUNT. */
int32_t *s = src[--channels];
int32_t last = r->last_sample[channels];
r->last_sample[channels] = s[count - 1];
d = dst[channels];
phase = r->phase;
pos = phase >> 16;
while (pos == 0)
{
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[0] - last);
phase += delta;
pos = phase >> 16;
}
while (pos < count)
{
last = s[pos - 1];
*d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
phase += delta;
pos = phase >> 16;
}
}
while (channels > 0);
/* Wrap phase accumulator back to start of next frame. */
r->phase = phase & 0xffff;
return d - dst[0];
}
#endif /* DSP_HAVE_ASM_RESAMPLING */
/* 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 = count;
if (dsp->frequency != NATIVE_FREQUENCY)
{
int32_t *dst[2] =
{
resample_buf,
resample_buf + RESAMPLE_BUF_COUNT/2,
};
int channels = dsp->num_channels;
if (dsp->frequency < NATIVE_FREQUENCY)
new_count = dsp_upsample(channels, count,
&resample_data[current_codec],
src, dst);
else
new_count = dsp_downsample(channels, count,
&resample_data[current_codec],
src, dst);
src[0] = dst[0];
src[1] = dst[channels - 1];
}
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, 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
static 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++;
gain = *setting++;
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->num_channels;
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_COUNT / 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 count number of audio samples. dst is assumed to be large
* enough; use dsp_output_count() to get the required number. src is an
* array of pointers; for mono and interleaved stereo, it contains one
* pointer to the start of the audio data and the other is ignored; for
* non-interleaved stereo, it contains two pointers, one for each audio
* channel. Returns number of bytes written to dst.
*/
int dsp_process(char *dst, const char *src[], int count)
{
int32_t* tmp[2];
int written = 0;
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];
dsp_set_replaygain(false);
while (count > 0)
{
samples = dsp->convert_to_internal(src, tmp, count);
count -= samples;
apply_gain(tmp, samples);
samples = resample(tmp, samples);
if (samples <= 0)
break; /* I'm pretty sure we're downsampling here */
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;
}
/* Given count number of input samples, calculate the maximum number of
* samples 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 */
int dsp_output_count(int count)
{
dsp = &dsp_conf[current_codec];
if (dsp->frequency != NATIVE_FREQUENCY)
{
count = (int)(((unsigned long)count * NATIVE_FREQUENCY
+ (dsp->frequency - 1)) / dsp->frequency);
}
/* Now we have the resampled sample count which must not exceed
* RESAMPLE_BUF_COUNT/2 to avoid resample buffer overflow. One
* must call dsp_input_count() to get the correct input sample
* count.
*/
if (count > RESAMPLE_BUF_COUNT/2)
count = RESAMPLE_BUF_COUNT/2;
return count;
}
/* Given count output samples, calculate number of input samples
* that would be consumed in order to fill the output buffer.
*/
int dsp_input_count(int count)
{
dsp = &dsp_conf[current_codec];
/* count is now the number of resampled input samples. Convert to
original input samples. */
if (dsp->frequency != NATIVE_FREQUENCY)
{
/* Use the real resampling delta =
* dsp->frequency * 65536 / NATIVE_FREQUENCY, and
* round towards zero to avoid buffer overflows. */
count = (int)(((unsigned long)count *
resample_data[current_codec].delta) >> 16);
}
return count;
}
int dsp_stereo_mode(void)
{
dsp = &dsp_conf[current_codec];
return dsp->stereo_mode;
}
bool dsp_configure(int setting, intptr_t 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 = (value == 0) ? NATIVE_FREQUENCY : 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 = value;
break;
case DSP_SET_CLIP_MAX:
dsp->clip_max = value;
break;
case DSP_SET_SAMPLE_DEPTH:
dsp->sample_depth = 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 = value;
dsp->sample_bytes = 4; /* samples are 32 bits */
dsp->clip_max = (1 << value) - 1;
dsp->clip_min = -(1 << value);
}
new_sample_conversion();
dither_init();
break;
case DSP_SET_STEREO_MODE:
dsp->stereo_mode = value;
dsp->num_channels = value == STEREO_MONO ? 1 : 2;
new_sample_conversion();
break;
case DSP_RESET:
dsp->stereo_mode = STEREO_NONINTERLEAVED;
dsp->num_channels = 2;
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;
new_sample_conversion();
break;
case DSP_FLUSH:
memset(&resample_data[current_codec], 0,
sizeof (struct resample_data));
resampler_set_delta(dsp->frequency);
dither_init();
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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