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-rw-r--r--apps/dsp.c98
-rw-r--r--apps/eq.c3
-rw-r--r--apps/eq.h8
3 files changed, 56 insertions, 53 deletions
diff --git a/apps/dsp.c b/apps/dsp.c
index 0c376a14ea..de88c97ca6 100644
--- a/apps/dsp.c
+++ b/apps/dsp.c
@@ -192,7 +192,7 @@ struct dsp_config
struct resample_data
{
long phase, delta;
- long last_sample[2];
+ int32_t last_sample[2];
};
struct dither_data
@@ -203,9 +203,9 @@ struct dither_data
struct crossfeed_data
{
- long lowpass[2];
- long highpass[2];
- long delay[2][13];
+ int32_t lowpass[2];
+ int32_t highpass[2];
+ int32_t delay[2][13];
int index;
};
@@ -233,8 +233,8 @@ struct dsp_config *dsp;
* of copying needed is minimized for that case.
*/
-static long sample_buf[SAMPLE_BUF_SIZE] IBSS_ATTR;
-static long resample_buf[RESAMPLE_BUF_SIZE] IBSS_ATTR;
+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)
{
@@ -254,7 +254,7 @@ void sound_set_pitch(int permille)
* 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, long* dst[])
+static int convert_to_internal(const char* src[], int count, int32_t* dst[])
{
count = MIN(SAMPLE_BUF_SIZE / 2, count);
@@ -267,15 +267,15 @@ static int convert_to_internal(const char* src[], int count, long* dst[])
}
else
{
- dst[0] = (long*) src[0];
- dst[1] = (long*) ((dsp->stereo_mode == STEREO_MONO) ? src[0] : src[1]);
+ 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];
- long* d0 = dst[0];
- long* d1 = dst[1];
+ int32_t* d0 = dst[0];
+ int32_t* d1 = dst[1];
int scale = WORD_SHIFT;
int i;
@@ -307,9 +307,9 @@ static int convert_to_internal(const char* src[], int count, long* dst[])
}
else if (dsp->stereo_mode == STEREO_INTERLEAVED)
{
- long* s0 = (long*) src[0];
- long* d0 = dst[0];
- long* d1 = dst[1];
+ 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++)
@@ -348,13 +348,13 @@ static void resampler_set_delta(int frequency)
/* 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(long **dst, long **src, int count,
+static long downsample(int32_t **dst, int32_t **src, int count,
struct resample_data *r)
{
long phase = r->phase;
long delta = r->delta;
- long last_sample;
- long *d[2] = { dst[0], dst[1] };
+ 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;
@@ -388,11 +388,11 @@ static long downsample(long **dst, long **src, int count,
return i;
}
-static long upsample(long **dst, long **src, int count, struct resample_data *r)
+static long upsample(int32_t **dst, int32_t **src, int count, struct resample_data *r)
{
long phase = r->phase;
long delta = r->delta;
- long *d[2] = { dst[0], dst[1] };
+ int32_t *d[2] = { dst[0], dst[1] };
int i = 0, j;
int pos;
int num_channels = dsp->stereo_mode == STEREO_MONO ? 1 : 2;
@@ -427,13 +427,13 @@ static long upsample(long **dst, long **src, int count, struct resample_data *r)
* done, to refer to the resampled data. Returns number of stereo samples
* for further processing.
*/
-static inline int resample(long* src[], int count)
+static inline int resample(int32_t* src[], int count)
{
long new_count;
if (dsp->frequency != NATIVE_FREQUENCY)
{
- long* dst[2] = {&resample_buf[0], &resample_buf[RESAMPLE_BUF_SIZE / 2]};
+ int32_t* dst[2] = {&resample_buf[0], &resample_buf[RESAMPLE_BUF_SIZE / 2]};
if (dsp->frequency < NATIVE_FREQUENCY)
{
@@ -460,7 +460,7 @@ static inline int resample(long* src[], int count)
return new_count;
}
-static inline long clip_sample(long sample, long min, long max)
+static inline long clip_sample(int32_t sample, int32_t min, int32_t max)
{
if (sample > max)
{
@@ -478,13 +478,13 @@ static inline long clip_sample(long sample, long min, long max)
* taken from the coolplayer project - coolplayer.sourceforge.net
*/
-static long dither_sample(long sample, long bias, long mask,
+static long dither_sample(int32_t sample, int32_t bias, int32_t mask,
struct dither_data* dither)
{
- long output;
- long random;
- long min;
- long max;
+ int32_t output;
+ int32_t random;
+ int32_t min;
+ int32_t max;
/* Noise shape and bias */
@@ -523,7 +523,7 @@ static const long crossfeed_coefs[6] ICONST_ATTR = {
LOW, LOW_COMP, HIGH_NEG, HIGH_COMP, ATT, ATT_COMP
};
-static void apply_crossfeed(long* src[], int count)
+static void apply_crossfeed(int32_t* src[], int count)
{
asm volatile (
"lea.l crossfeed_data, %%a1 \n"
@@ -598,20 +598,20 @@ static void apply_crossfeed(long* src[], int count)
);
}
#else
-static void apply_crossfeed(long* src[], int count)
+static void apply_crossfeed(int32_t* src[], int count)
{
- long a; /* accumulator */
+ int32_t a; /* accumulator */
- long low_left = crossfeed_data.lowpass[0];
- long low_right = crossfeed_data.lowpass[1];
- long high_left = crossfeed_data.highpass[0];
- long high_right = crossfeed_data.highpass[1];
+ int32_t low_left = crossfeed_data.lowpass[0];
+ int32_t low_right = crossfeed_data.lowpass[1];
+ int32_t high_left = crossfeed_data.highpass[0];
+ int32_t high_right = crossfeed_data.highpass[1];
unsigned int index = crossfeed_data.index;
- long left, right;
+ int32_t left, right;
- long * delay_l = crossfeed_data.delay[0];
- long * delay_r = crossfeed_data.delay[1];
+ int32_t* delay_l = crossfeed_data.delay[0];
+ int32_t* delay_r = crossfeed_data.delay[1];
int i;
@@ -711,7 +711,7 @@ void dsp_eq_update_data(bool enabled, int band)
}
/* Apply EQ filters to those bands that have got it switched on. */
-static void eq_process(long **x, unsigned num)
+static void eq_process(int32_t **x, unsigned num)
{
int i;
unsigned int channels = dsp->stereo_mode != STEREO_MONO ? 2 : 1;
@@ -736,19 +736,19 @@ static void eq_process(long **x, unsigned num)
* the src array if gain was applied.
* Note that this must be called before the resampler.
*/
-static void apply_gain(long* _src[], int _count)
+static void apply_gain(int32_t* _src[], int _count)
{
struct dsp_config *my_dsp = dsp;
if (my_dsp->replaygain)
{
- long** src = _src;
+ int32_t** src = _src;
int count = _count;
- long* s0 = src[0];
- long* s1 = src[1];
+ int32_t* s0 = src[0];
+ int32_t* s1 = src[1];
long gain = my_dsp->replaygain;
- long s;
+ int32_t s;
int i;
- long *d;
+ int32_t *d;
if (s0 != s1)
{
@@ -773,10 +773,10 @@ static void apply_gain(long* _src[], int _count)
}
}
-static void write_samples(short* dst, long* src[], int count)
+static void write_samples(short* dst, int32_t* src[], int count)
{
- long* s0 = src[0];
- long* s1 = src[1];
+ int32_t* s0 = src[0];
+ int32_t* s1 = src[1];
int scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
if (dsp->dither_enabled)
@@ -815,7 +815,7 @@ static void write_samples(short* dst, long* src[], int count)
*/
long dsp_process(char* dst, const char* src[], long size)
{
- long* tmp[2];
+ int32_t* tmp[2];
long written = 0;
long factor;
int samples;
@@ -980,7 +980,7 @@ bool dsp_configure(int setting, void *value)
else
{
dsp->frac_bits = (long) value;
- dsp->sample_bytes = sizeof(long);
+ dsp->sample_bytes = 4; /* samples are 32 bits */
dsp->clip_max = (1 << (long)value) - 1;
dsp->clip_min = -(1 << (long)value);
}
diff --git a/apps/eq.c b/apps/eq.c
index 0a835d5484..cb7086a7e3 100644
--- a/apps/eq.c
+++ b/apps/eq.c
@@ -17,6 +17,7 @@
*
****************************************************************************/
+#include <inttypes.h>
#include "config.h"
#include "eq.h"
@@ -214,7 +215,7 @@ void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c)
}
#if (!defined(CPU_COLDFIRE) && !defined(CPU_ARM)) || defined(SIMULATOR)
-void eq_filter(long **x, struct eqfilter *f, unsigned num,
+void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
unsigned channels, unsigned shift)
{
unsigned c, i;
diff --git a/apps/eq.h b/apps/eq.h
index f29d765325..9228658eca 100644
--- a/apps/eq.h
+++ b/apps/eq.h
@@ -20,6 +20,8 @@
#ifndef _EQ_H
#define _EQ_H
+#include <inttypes.h>
+
/* These depend on the fixed point formats used by the different filter types
and need to be changed when they change.
*/
@@ -27,14 +29,14 @@
#define EQ_SHELF_SHIFT 8
struct eqfilter {
- long coefs[5]; /* Order is b0, b1, b2, a1, a2 */
- long history[2][4];
+ int32_t coefs[5]; /* Order is b0, b1, b2, a1, a2 */
+ int32_t history[2][4];
};
void eq_pk_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
void eq_ls_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
void eq_hs_coefs(unsigned long cutoff, unsigned long Q, long db, long *c);
-void eq_filter(long **x, struct eqfilter *f, unsigned num,
+void eq_filter(int32_t **x, struct eqfilter *f, unsigned num,
unsigned channels, unsigned shift);
#endif