jackaudio
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jack1
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finish what i started with earlier transformations - several functions were missed; fix up merge functions to avoid overflow; follow fons' advice and make min/max values symmetrical 

git-svn-id: svn+ssh://jackaudio.org/trunk/jack@3048 0c269be4-1314-0410-8aa9-9f06e86f4224
tags/0.115.6
paul 16 years ago
parent
d66301b86c
commit
2e10e088d0
1 changed files with 81 additions and 67 deletions
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  1. +81
    -67
      drivers/alsa/memops.c

+ 81
- 67
drivers/alsa/memops.c View File

@@ -65,13 +65,15 @@
#define SAMPLE_16BIT_SCALING 32767.0f

/* these are just values to use if the floating point value was out of range
advice from Fons Adriaensen: make the limits symmetrical
*/

#define SAMPLE_24BIT_MAX 8388607 // 2^23 - 1
#define SAMPLE_24BIT_MIN -8388608 // - (2^23)
#define SAMPLE_24BIT_MAX 8388607
#define SAMPLE_24BIT_MIN -8388607

#define SAMPLE_16BIT_MAX 32767 // 2^15-1
#define SAMPLE_16BIT_MIN -32768 // - (2^15)
#define SAMPLE_16BIT_MAX 32767
#define SAMPLE_16BIT_MIN -32767

/* these mark the outer edges of the range considered "within" range
for a floating point sample value. values outside (and on the boundaries)
@@ -229,19 +231,17 @@ void sample_move_dS_s32u24 (jack_default_audio_sample_t *dst, char *src, unsigne

void sample_move_d24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state)
{
int64_t y;
int z;
int32_t z;

while (nsamples--) {
y = (int64_t)(*src * SAMPLE_24BIT_SCALING);

if (y > (INT_MAX >> 8 )) {
z = (INT_MAX >> 8);
} else if (y < (INT_MIN >> 8 )) {
z = (INT_MIN >> 8 );
if (*src <= NORMALIZED_FLOAT_MIN) {
z = SAMPLE_24BIT_MIN;
} else if (*src >= NORMALIZED_FLOAT_MAX) {
z = SAMPLE_24BIT_MAX;
} else {
z = (int)y;
z = (int32_t)f_round (*src * SAMPLE_24BIT_SCALING);
}

#if __BYTE_ORDER == __LITTLE_ENDIAN
dst[0]=(char)(z>>16);
dst[1]=(char)(z>>8);
@@ -258,20 +258,21 @@ void sample_move_d24_sSs (char *dst, jack_default_audio_sample_t *src, unsigned

void sample_move_d24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state)
{
int64_t y;
int32_t z;
while (nsamples--) {
y = (int64_t)(*src * SAMPLE_24BIT_SCALING);
if (y > (INT_MAX >> 8 )) {
y = (INT_MAX >> 8);
} else if (y < (INT_MIN >> 8 )) {
y = (INT_MIN >> 8 );
if (*src <= NORMALIZED_FLOAT_MIN) {
z = SAMPLE_24BIT_MIN;
} else if (*src >= NORMALIZED_FLOAT_MAX) {
z = SAMPLE_24BIT_MAX;
} else {
z = (int32_t) f_round (*src * SAMPLE_24BIT_SCALING);
}

#if __BYTE_ORDER == __LITTLE_ENDIAN
memcpy (dst, &y, 3);
memcpy (dst, &z, 3);
#elif __BYTE_ORDER == __BIG_ENDIAN
memcpy (dst, (char *)&y + 5, 3);
memcpy (dst, (char *)&z + 1, 3);
#endif
dst += dst_skip;
src++;
@@ -402,7 +403,6 @@ void sample_move_dither_rect_d16_sSs (char *dst, jack_default_audio_sample_t *s
void sample_move_dither_rect_d16_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state)
{
jack_default_audio_sample_t val;
int16_t tmp;

while (nsamples--) {
val = *src - (fast_rand()/((float)INT_MAX - 1.0f));
@@ -433,7 +433,7 @@ void sample_move_dither_tri_d16_sSs (char *dst, jack_default_audio_sample_t *sr
if (x <= NORMALIZED_FLOAT_MIN) {
y = SAMPLE_16BIT_MIN;
} else if (y >= NORMALIZED_FLOAT_MAX) {
} else if (x >= NORMALIZED_FLOAT_MAX) {
y = SAMPLE_16BIT_MAX;
} else {
y = f_round(x);
@@ -457,21 +457,19 @@ void sample_move_dither_tri_d16_sS (char *dst, jack_default_audio_sample_t *src
jack_default_audio_sample_t x;
float r;
float rm1 = state->rm1;
int y;

while (nsamples--) {
x = *src * SAMPLE_16BIT_SCALING;
r = 2.0f * (float)fast_rand() / (float)INT_MAX - 1.0f;
x += r - rm1;
rm1 = r;
y = f_round(x);

if (y > SHRT_MAX) {
*((short *)dst) = SHRT_MAX;
} else if (y < SHRT_MIN) {
*((short *)dst) = SHRT_MIN;
if (x <= NORMALIZED_FLOAT_MIN) {
*((int16_t*)dst) = SAMPLE_16BIT_MIN;
} else if (x >= NORMALIZED_FLOAT_MAX) {
*((int16_t*)dst) = SAMPLE_16BIT_MAX;
} else {
*((short *) dst) = (short)y;
*((int16_t*)dst) = (int16_t) f_round (x * SAMPLE_16BIT_SCALING);
}

dst += dst_skip;
@@ -488,7 +486,7 @@ void sample_move_dither_shaped_d16_sSs (char *dst, jack_default_audio_sample_t
float r;
float rm1 = state->rm1;
unsigned int idx = state->idx;
int y;
int16_t tmp;

while (nsamples--) {
x = *src * SAMPLE_16BIT_SCALING;
@@ -504,24 +502,24 @@ void sample_move_dither_shaped_d16_sSs (char *dst, jack_default_audio_sample_t
xp = xe + r - rm1;
rm1 = r;

/* This could be some inline asm on x86 */
y = f_round(xp);
if (xp <= NORMALIZED_FLOAT_MIN) {
tmp = SAMPLE_16BIT_MIN;
} else if (xp >= NORMALIZED_FLOAT_MAX) {
tmp = SAMPLE_16BIT_MAX;
} else {
tmp = (int16_t) f_round (xp * SAMPLE_16BIT_SCALING);
}

/* Intrinsic z^-1 delay */
idx = (idx + 1) & DITHER_BUF_MASK;
state->e[idx] = y - xe;
state->e[idx] = xp - xe;

if (y > SHRT_MAX) {
y = SHRT_MAX;
} else if (y < SHRT_MIN) {
y = SHRT_MIN;
}
#if __BYTE_ORDER == __LITTLE_ENDIAN
dst[0]=(char)(y>>8);
dst[1]=(char)(y);
dst[0]=(char)(tmp>>8);
dst[1]=(char)(tmp);
#elif __BYTE_ORDER == __BIG_ENDIAN
dst[0]=(char)(y);
dst[1]=(char)(y>>8);
dst[0]=(char)(tmp);
dst[1]=(char)(tmp>>8);
#endif
dst += dst_skip;
src++;
@@ -538,7 +536,6 @@ void sample_move_dither_shaped_d16_sS (char *dst, jack_default_audio_sample_t *
float r;
float rm1 = state->rm1;
unsigned int idx = state->idx;
int y;

while (nsamples--) {
x = *src * SAMPLE_16BIT_SCALING;
@@ -554,20 +551,18 @@ void sample_move_dither_shaped_d16_sS (char *dst, jack_default_audio_sample_t *
xp = xe + r - rm1;
rm1 = r;

/* This could be some inline asm on x86 */
y = f_round(xp);
if (xp <= NORMALIZED_FLOAT_MIN) {
*((int16_t*) dst) = SAMPLE_16BIT_MIN;
} else if (xp >= NORMALIZED_FLOAT_MIN) {
*((int16_t*) dst) = SAMPLE_16BIT_MAX;
} else {
*((int16_t*) dst) = (int16_t) f_round (*src * SAMPLE_16BIT_SCALING);
}

/* Intrinsic z^-1 delay */
idx = (idx + 1) & DITHER_BUF_MASK;
state->e[idx] = y - xe;
state->e[idx] = *((int16_t*) dst) - xe;

if (y > SHRT_MAX) {
*((short *)dst) = SHRT_MAX;
} else if (y < SHRT_MIN) {
*((short *)dst) = SHRT_MIN;
} else {
*((short *) dst) = (short)y;
}
dst += dst_skip;
src++;
}
@@ -609,20 +604,25 @@ void sample_move_dS_s16 (jack_default_audio_sample_t *dst, char *src, unsigned l

void sample_merge_d16_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state)
{
short val;
int16_t val;
/* ALERT: signed sign-extension portability !!! */
while (nsamples--) {
val = (short) (*src * SAMPLE_16BIT_SCALING);
if (val > SHRT_MAX - *((short *) dst)) {
*((short *)dst) = SHRT_MAX;
} else if (val < SHRT_MIN - *((short *) dst)) {
*((short *)dst) = SHRT_MIN;
if (*src <= NORMALIZED_FLOAT_MIN) {
val = SAMPLE_16BIT_MIN;
} else if (*src >= NORMALIZED_FLOAT_MAX) {
val = SAMPLE_16BIT_MAX;
} else {
*((short *) dst) += val;
val = (int16_t) f_round (*src * SAMPLE_16BIT_SCALING);
}

if (val > SAMPLE_16BIT_MAX - *((int16_t*)dst)) {
*((int16_t*) dst) = SAMPLE_16BIT_MAX;
} else {
*((int16_t*) dst) += val;
}

dst += dst_skip;
src++;
}
@@ -630,10 +630,24 @@ void sample_merge_d16_sS (char *dst, jack_default_audio_sample_t *src, unsigned

void sample_merge_d32u24_sS (char *dst, jack_default_audio_sample_t *src, unsigned long nsamples, unsigned long dst_skip, dither_state_t *state)
{
/* ALERT: signed sign-extension portability !!! */
int32_t val;

while (nsamples--) {
*((int *) dst) += (((int) (*src * SAMPLE_24BIT_SCALING)) << 8);

if (*src <= NORMALIZED_FLOAT_MIN) {
val = SAMPLE_16BIT_MIN;
} else if (*src >= NORMALIZED_FLOAT_MAX) {
val = SAMPLE_16BIT_MAX;
} else {
val = f_round (*src * SAMPLE_24BIT_SCALING);
}

if (val > SAMPLE_24BIT_MAX - *((int32_t*) dst)) {
*((int32_t*) dst) = SAMPLE_24BIT_MAX;
} else {
*((int32_t*) dst) += val;
}
dst += dst_skip;
src++;
}


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