Skip to content
This repository has been archived by the owner on Feb 11, 2021. It is now read-only.

Commit

Permalink
Began implementing IIR and FIR filters, and got zero stuffing and sam…
Browse files Browse the repository at this point in the history 
…ple discarding working.
  • Loading branch information
@schnarf
schnarf committed Jun 18, 2008
1 parent 0a35ce5 commit fa3dee0
Show file tree
Hide file tree
Showing 2 changed files with 109 additions and 27 deletions.
4 changes: 2 additions & 2 deletions include/SDL_audio.h
View file
Expand Up @@ -142,8 +142,8 @@ typedef struct SDL_AudioCVT
SDL_AudioFormat dst_format; /* Target audio format */
double rate_incr; /* Rate conversion increment */
Uint8 *buf; /* Buffer to hold entire audio data */
Uint8 *sinc; /* Windowed sinc filter */
Uint8 *state_buf; /* Sample history for either the FIR or IIR filter */
Uint8 *coeff; /* Filter coefficients: either big windowed sinc filter, or 6 IIR lowpass coefficients*/
Uint8 *state_buf; /* Sample history for either the FIR or IIR filter. For IIR filter, first two elements are X, second two are Y, and state_pos toggles the order */
int state_pos; /* Position in the state */
int len_sinc; /* Length of windowed sinc filter, in appropriate units (not necessarily bytes) */
int len; /* Length of original audio buffer */
Expand Down
132 changes: 107 additions & 25 deletions src/audio/SDL_audiocvt.c
View file
Expand Up @@ -27,6 +27,8 @@
#include "SDL_audio.h"
#include "SDL_audio_c.h"

#define DEBUG_CONVERT

/* Effectively mix right and left channels into a single channel */
static void SDLCALL
SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
Expand Down Expand Up @@ -1237,7 +1239,7 @@ SDL_Resample(SDL_AudioCVT * cvt, SDL_AudioFormat format)
int i, j;

#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate via proper resampling (mono)\n");
printf("Converting audio rate via proper resampling (mono)\n");
#endif

#define zerostuff_mono(type) { \
Expand All @@ -1256,14 +1258,17 @@ SDL_Resample(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#define discard_mono(type) { \
const type *src = (const type *) (cvt->buf); \
type *dst = (type *) (cvt->buf); \
for (i = 0; i < cvt->len_cvt / sizeof (type); ++i) { \
for (i = 0; i < cvt->len_cvt / cvt->len_div / sizeof (type); ++i) { \
dst[0] = src[0]; \
src += cvt->len_div; \
++dst; \
} \
}

// Step 1: Zero stuff the conversion buffer
#ifdef DEBUG_CONVERT
printf("Zero-stuffing by a factor of %u\n", cvt->len_mult);
#endif
switch (SDL_AUDIO_BITSIZE(format)) {
case 8:
zerostuff_mono(Uint8);
Expand All @@ -1281,6 +1286,9 @@ SDL_Resample(SDL_AudioCVT * cvt, SDL_AudioFormat format)
// Step 2: Use either a windowed sinc FIR filter or IIR lowpass filter to remove all alias frequencies

// Step 3: Discard unnecessary samples
#ifdef DEBUG_CONVERT
printf("Discarding samples by a factor of %u\n", cvt->len_div);
#endif
switch (SDL_AUDIO_BITSIZE(format)) {
case 8:
discard_mono(Uint8);
Expand Down Expand Up @@ -1418,13 +1426,53 @@ int SDL_BuildIIRLowpass(SDL_AudioCVT * cvt, SDL_AudioFormat format) {
coeff[4] = -2.0f * cosw0 * scale;
coeff[5] = (1.0f - alpha) * scale;

/* Convert coefficients to fixed point, using the range (-2.0, 2.0) */
/* Copy the coefficients to the struct. If necessary, convert coefficients to fixed point, using the range (-2.0, 2.0) */
if(SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
float *cvt_coeff = (float *)cvt->coeff;
int i;
for(i = 0; i < 6; ++i) {
cvt_coeff[i] = coeff[i];
}
} else {
}

/* Initialize the state buffer to all zeroes, and set initial position */
memset(cvt->state_buf, 0, 4 * SDL_AUDIO_BITSIZE(format) / 4);
cvt->state_pos = 0;
}

/* Apply the lowpass IIR filter to the given SDL_AudioCVT struct */
int SDL_FilterIIR(SDL_AudioCVT * cvt, SDL_AudioFormat format) {
int i, n;

n = cvt->len_cvt / (SDL_AUDIO_BITSIZE(format) / 4);

if(SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
float *coeff = (float *)cvt->coeff;
float *state = (float *)cvt->state_buf;
float *buf = (float *)cvt->buf;
float temp;


for(i = 0; i < n; ++i) {
/* y[n] = b0 * x[n] + b1 * x[n-1] + b2 * x[n-2] - a1 * y[n-1] - a[2] * y[n-2] */
temp = buf[n];
if( cvt->state_pos ) {
buf[n] = coeff[0] * buf[n] + coeff[1] * state[0] + coeff[2] * state[1] - coeff[4] * state[2] - coeff[5] * state[3];
state[1] = temp;
state[3] = buf[n];
cvt->state_pos = 0;
} else {
buf[n] = coeff[0] * buf[n] + coeff[1] * state[1] + coeff[2] * state[0] - coeff[4] * state[3] - coeff[5] * state[2];
state[0] = temp;
state[2] = buf[n];
cvt->state_pos = 1;
}
}
} else {
}
}

/* Apply the windowed sinc FIR filter to the given SDL_AudioCVT struct */
int SDL_FilterFIR(SDL_AudioCVT * cvt, SDL_AudioFormat format) {
int n = cvt->len_cvt / (SDL_AUDIO_BITSIZE(format) / 4);
Expand All @@ -1437,7 +1485,7 @@ int SDL_FilterFIR(SDL_AudioCVT * cvt, SDL_AudioFormat format) {
significantly fewer multiplications and additions.
*/
#define filter_sinc(type, shift_bits) { \
type *sinc = (type *)cvt->sinc; \
type *sinc = (type *)cvt->coeff; \
type *state = (type *)cvt->state_buf; \
type *buf = (type *)cvt->buf; \
for(i = 0; i < n; ++i) { \
Expand All @@ -1449,17 +1497,33 @@ int SDL_FilterFIR(SDL_AudioCVT * cvt, SDL_AudioFormat format) {
} \
} \
}

switch (SDL_AUDIO_BITSIZE(format)) {
case 8:
filter_sinc(Uint8, 4);
break;
case 16:
filter_sinc(Uint16, 8);
break;
case 32:
filter_sinc(Uint32, 16);
break;

/* If it's floating point, we don't need to do any shifting */
if(SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
float *sinc = (float *)cvt->coeff;
float *state = (float *)cvt->state_buf;
float *buf = (float *)cvt->buf;

for(i = 0; i < n; ++i) {
state[cvt->state_pos++] = buf[i];
if(cvt->state_pos == m) cvt->state_pos = 0;
buf[i] = 0.0f;
for(j = 0; j < m; ++j) {
buf[i] += state[j] * sinc[j];
}
}
} else {
switch (SDL_AUDIO_BITSIZE(format)) {
case 8:
filter_sinc(Uint8, 4);
break;
case 16:
filter_sinc(Uint16, 8);
break;
case 32:
filter_sinc(Uint32, 16);
break;
}
}

#undef filter_sinc
Expand All @@ -1482,7 +1546,7 @@ SDL_BuildWindowedSinc(SDL_AudioCVT * cvt, SDL_AudioFormat format, unsigned int m
unsigned int i;

/* Check that the buffer is allocated */
if( cvt->sinc == NULL ) {
if( cvt->coeff == NULL ) {
return -1;
}

Expand Down Expand Up @@ -1519,15 +1583,15 @@ SDL_BuildWindowedSinc(SDL_AudioCVT * cvt, SDL_AudioFormat format, unsigned int m

#define convert_fixed(type, size) { \
norm_fact = size / norm_sum; \
type *dst = (type *)cvt->sinc; \
type *dst = (type *)cvt->coeff; \
for( i = 0; i <= m; ++i ) { \
dst[i] = (type)(fSinc[i] * norm_fact); \
} \
}

/* If we're using floating point, we only need to normalize */
if(SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
float *fDest = (float *)cvt->sinc;
float *fDest = (float *)cvt->coeff;
norm_fact = 1.0f / norm_sum;
for(i = 0; i <= m; ++i) {
fDest[i] = fSinc[i] * norm_fact;
Expand Down Expand Up @@ -1555,6 +1619,17 @@ SDL_BuildWindowedSinc(SDL_AudioCVT * cvt, SDL_AudioFormat format, unsigned int m
free(fSinc);
}

/* This is used to reduce the resampling ratio */
inline int SDL_GCD(int a, int b) {
int temp;
while(b != 0) {
temp = a % b;
a = b;
b = temp;
}
return a;
}


/* Creates a set of audio filters to convert from one format to another.
Returns -1 if the format conversion is not supported, 0 if there's
Expand Down Expand Up @@ -1644,7 +1719,14 @@ SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
}

/* Do rate conversion */
cvt->rate_incr = 0.0;
int rate_gcd;
rate_gcd = SDL_GCD(src_rate, dst_rate);
cvt->len_mult = dst_rate / rate_gcd;
cvt->len_div = src_rate / rate_gcd;
cvt->len_ratio = (double)cvt->len_mult / (double)cvt->len_div;
cvt->filters[cvt->filter_index++] = SDL_Resample;

/*cvt->rate_incr = 0.0;
if ((src_rate / 100) != (dst_rate / 100)) {
Uint32 hi_rate, lo_rate;
int len_mult;
Expand Down Expand Up @@ -1693,16 +1775,16 @@ SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
}
len_mult = 2;
len_ratio = 2.0;
}
}*/
/* If hi_rate = lo_rate*2^x then conversion is easy */
while (((lo_rate * 2) / 100) <= (hi_rate / 100)) {
/*while (((lo_rate * 2) / 100) <= (hi_rate / 100)) {
cvt->filters[cvt->filter_index++] = rate_cvt;
cvt->len_mult *= len_mult;
lo_rate *= 2;
cvt->len_ratio *= len_ratio;
}
}*/
/* We may need a slow conversion here to finish up */
if ((lo_rate / 100) != (hi_rate / 100)) {
/*if ((lo_rate / 100) != (hi_rate / 100)) {*/
#if 1
/* The problem with this is that if the input buffer is
say 1K, and the conversion rate is say 1.1, then the
Expand All @@ -1722,8 +1804,8 @@ SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
}
cvt->filters[cvt->filter_index++] = SDL_RateSLOW;
#endif
}
}
/* }
}*/

/* Set up the filter information */
if (cvt->filter_index != 0) {
Expand Down

0 comments on commit fa3dee0

Please sign in to comment.