mpv/libaf/af_resample.c

361 lines
9.2 KiB
C

/*=============================================================================
//
// This software has been released under the terms of the GNU Public
// license. See http://www.gnu.org/copyleft/gpl.html for details.
//
// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au
//
//=============================================================================
*/
/* This audio filter changes the sample rate. */
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <inttypes.h>
#include "af.h"
#include "dsp.h"
/* Below definition selects the length of each poly phase component.
Valid definitions are L8 and L16, where the number denotes the
length of the filter. This definition affects the computational
complexity (see play()), the performance (see filter.h) and the
memory usage. The filterlenght is choosen to 8 if the machine is
slow and to 16 if the machine is fast and has MMX.
*/
#if !defined(HAVE_MMX) // This machine is slow
#define L8
#else
#define L16
#endif
#include "af_resample.h"
// Filtering types
#define TYPE_LIN 0 // Linear interpolation
#define TYPE_INT 1 // 16 bit integer
#define TYPE_FLOAT 2 // 32 bit floating point
// Accuracy for linear interpolation
#define STEPACCURACY 32
// local data
typedef struct af_resample_s
{
void* w; // Current filter weights
void** xq; // Circular buffers
uint32_t xi; // Index for circular buffers
uint32_t wi; // Index for w
uint32_t i; // Number of new samples to put in x queue
uint32_t dn; // Down sampling factor
uint32_t up; // Up sampling factor
uint64_t step; // Step size for linear interpolation
uint64_t pt; // Pointer remainder for linear interpolation
int sloppy; // Enable sloppy resampling to reduce memory usage
int type; // Filter type
} af_resample_t;
// Euclids algorithm for calculating Greatest Common Divisor GCD(a,b)
static inline int gcd(register int a, register int b)
{
register int r = min(a,b);
a=max(a,b);
b=r;
r=a%b;
while(r!=0){
a=b;
b=r;
r=a%b;
}
return b;
}
// Fast linear interpolation resample with modest audio quality
static int linint(af_data_t* c,af_data_t* l, af_resample_t* s)
{
uint32_t len = 0; // Number of input samples
uint32_t nch = l->nch; // Words pre transfer
uint64_t step = s->step;
int16_t* in16 = ((int16_t*)c->audio);
int16_t* out16 = ((int16_t*)l->audio);
int32_t* in32 = ((int32_t*)c->audio);
int32_t* out32 = ((int32_t*)l->audio);
uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY);
uint64_t pt = s->pt;
uint16_t tmp;
switch (nch){
case 1:
while(pt < end){
out16[len++]=in16[pt>>STEPACCURACY];
pt+=step;
}
s->pt=pt & ((1LL<<STEPACCURACY)-1);
break;
case 2:
end/=2;
while(pt < end){
out32[len++]=in32[pt>>STEPACCURACY];
pt+=step;
}
len=(len<<1);
s->pt=pt & ((1LL<<STEPACCURACY)-1);
break;
default:
end /=nch;
while(pt < end){
tmp=nch;
do {
tmp--;
out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch];
} while (tmp);
len+=nch;
pt+=step;
}
s->pt=pt & ((1LL<<STEPACCURACY)-1);
}
return len;
}
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
switch(cmd){
case AF_CONTROL_REINIT:{
af_resample_t* s = (af_resample_t*)af->setup;
af_data_t* n = (af_data_t*)arg; // New configureation
int i,d = 0;
int rv = AF_OK;
size_t tsz = (s->type==TYPE_INT) ? sizeof(int16_t) : sizeof(float);
// Make sure this filter isn't redundant
if(af->data->rate == n->rate)
return AF_DETACH;
// If linear interpolation
if(s->type == TYPE_LIN){
s->pt=0LL;
s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL;
af_msg(AF_MSG_VERBOSE,"[resample] Linear interpolation step: 0x%016X.\n",
s->step);
af->mul.n = af->data->rate;
af->mul.d = n->rate;
}
// Create space for circular bufers (if nesessary)
if((af->data->nch != n->nch) && (s->type != TYPE_LIN)){
// First free the old ones
if(s->xq){
for(i=1;i<af->data->nch;i++)
if(s->xq[i])
free(s->xq[i]);
free(s->xq);
}
// ... then create new
s->xq = malloc(n->nch*sizeof(void*));
for(i=0;i<n->nch;i++)
s->xq[i] = malloc(2*L*tsz);
s->xi = 0;
}
// Set parameters
af->data->nch = n->nch;
if(s->type == TYPE_INT || s->type == TYPE_LIN){
af->data->format = AF_FORMAT_NE | AF_FORMAT_SI;
af->data->bps = 2;
}
else{
af->data->format = AF_FORMAT_NE | AF_FORMAT_F;
af->data->bps = 4;
}
if(af->data->format != n->format || af->data->bps != n->bps)
rv = AF_FALSE;
n->format = af->data->format;
n->bps = af->data->bps;
// If linear interpolation is used the setup is done.
if(s->type == TYPE_LIN)
return rv;
// Calculate up and down sampling factors
d=gcd(af->data->rate,n->rate);
// If sloppy resampling is enabled limit the upsampling factor
if(s->sloppy && (af->data->rate/d > 5000)){
int up=af->data->rate/2;
int dn=n->rate/2;
int m=2;
while(af->data->rate/(d*m) > 5000){
d=gcd(up,dn);
up/=2; dn/=2; m*=2;
}
d*=m;
}
// Check if the the design needs to be redone
if(s->up != af->data->rate/d || s->dn != n->rate/d){
float* w;
float* wt;
float fc;
int j;
s->up = af->data->rate/d;
s->dn = n->rate/d;
// Calculate cuttof frequency for filter
fc = 1/(float)(max(s->up,s->dn));
// Allocate space for polyphase filter bank and protptype filter
w = malloc(sizeof(float) * s->up *L);
if(NULL != s->w)
free(s->w);
s->w = malloc(L*s->up*tsz);
// Design prototype filter type using Kaiser window with beta = 10
if(NULL == w || NULL == s->w ||
-1 == design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
af_msg(AF_MSG_ERROR,"[resample] Unable to design prototype filter.\n");
return AF_ERROR;
}
// Copy data from prototype to polyphase filter
wt=w;
for(j=0;j<L;j++){//Columns
for(i=0;i<s->up;i++){//Rows
if(s->type == TYPE_INT){
float t=(float)s->up*32767.0*(*wt);
((int16_t*)s->w)[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
}
else
((float*)s->w)[i*L+j] = (float)s->up*(*wt);
wt++;
}
}
free(w);
af_msg(AF_MSG_VERBOSE,"[resample] New filter designed up: %i "
"down: %i\n", s->up, s->dn);
}
// Set multiplier and delay
af->delay = (double)(1000*L/2)/((double)n->rate);
af->mul.n = s->up;
af->mul.d = s->dn;
return rv;
}
case AF_CONTROL_COMMAND_LINE:{
af_resample_t* s = (af_resample_t*)af->setup;
int rate=0;
int lin=0;
sscanf((char*)arg,"%i:%i:%i", &rate, &(s->sloppy), &lin);
if(lin)
s->type = TYPE_LIN;
return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate);
}
case AF_CONTROL_POST_CREATE:
((af_resample_t*)af->setup)->type =
((af_cfg_t*)arg)->force == AF_INIT_SLOW ? TYPE_INT : TYPE_FLOAT;
return AF_OK;
case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
// Reinit must be called after this function has been called
// Sanity check
if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
af_msg(AF_MSG_ERROR,"[resample] The output sample frequency "
"must be between 8kHz and 192kHz. Current value is %i \n",
((int*)arg)[0]);
return AF_ERROR;
}
af->data->rate=((int*)arg)[0];
af_msg(AF_MSG_VERBOSE,"[resample] Changing sample rate "
"to %iHz\n",af->data->rate);
return AF_OK;
}
return AF_UNKNOWN;
}
// Deallocate memory
static void uninit(struct af_instance_s* af)
{
if(af->data)
free(af->data);
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
int len = 0; // Length of output data
af_data_t* c = data; // Current working data
af_data_t* l = af->data; // Local data
af_resample_t* s = (af_resample_t*)af->setup;
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
return NULL;
// Run resampling
switch(s->type){
case(TYPE_INT):
# define FORMAT_I 1
if(s->up>s->dn){
# define UP
# include "af_resample.h"
# undef UP
}
else{
# define DN
# include "af_resample.h"
# undef DN
}
break;
case(TYPE_FLOAT):
# undef FORMAT_I
# define FORMAT_F 1
if(s->up>s->dn){
# define UP
# include "af_resample.h"
# undef UP
}
else{
# define DN
# include "af_resample.h"
# undef DN
}
break;
case(TYPE_LIN):
len = linint(c, l, s);
break;
}
// Set output data
c->audio = l->audio;
c->len = len*l->bps;
c->rate = l->rate;
return c;
}
// Allocate memory and set function pointers
static int open(af_instance_t* af){
af->control=control;
af->uninit=uninit;
af->play=play;
af->mul.n=1;
af->mul.d=1;
af->data=calloc(1,sizeof(af_data_t));
af->setup=calloc(1,sizeof(af_resample_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
return AF_OK;
}
// Description of this plugin
af_info_t af_info_resample = {
"Sample frequency conversion",
"resample",
"Anders",
"",
AF_FLAGS_REENTRANT,
open
};