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mpv/libaf/af_resample.c

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/*=============================================================================
//
// 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. */
#define PLUGIN
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <inttypes.h>
#include "../config.h"
#include "../mp_msg.h"
#include "../libao2/afmt.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_SSE) && !defined(HAVE_3DNOW) // This machine is slow
#define L 8 // Filter length
// Unrolled loop to speed up execution
#define FIR(x,w,y) \
(y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \
+ w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) >> 16
#else /* Fast machine */
#define L 16
// Unrolled loop to speed up execution
#define FIR(x,w,y) \
y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
+ w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
+ w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
+ w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) >> 16
#endif /* Fast machine */
// Macro to add data to circular que
#define ADDQUE(xi,xq,in)\
xq[xi]=xq[xi+L]=(*in);\
xi=(--xi)&(L-1);
// local data
typedef struct af_resample_s
{
int16_t* w; // Current filter weights
int16_t** 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
} af_resample_t;
// Euclids algorithm for calculating Greatest Common Divisor GCD(a,b)
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;
}
static int upsample(af_data_t* c,af_data_t* l, af_resample_t* s)
{
uint32_t ci = l->nch; // Index for channels
uint32_t len = 0; // Number of input samples
uint32_t nch = l->nch; // Number of channels
uint32_t inc = s->up/s->dn;
uint32_t level = s->up%s->dn;
uint32_t up = s->up;
uint32_t dn = s->dn;
register int16_t* w = s->w;
register uint32_t wi = 0;
register uint32_t xi = 0;
// Index current channel
while(ci--){
// Temporary pointers
register int16_t* x = s->xq[ci];
register int16_t* in = ((int16_t*)c->audio)+ci;
register int16_t* out = ((int16_t*)l->audio)+ci;
int16_t* end = in+c->len/2; // Block loop end
wi = s->wi; xi = s->xi;
while(in < end){
register uint32_t i = inc;
if(wi<level) i++;
ADDQUE(xi,x,in);
in+=nch;
while(i--){
// Run the FIR filter
FIR((&x[xi]),(&w[wi*L]),out);
len++; out+=nch;
// Update wi to point at the correct polyphase component
wi=(wi+dn)%up;
}
}
}
// Save values that needs to be kept for next time
s->wi = wi;
s->xi = xi;
return len;
}
static int downsample(af_data_t* c,af_data_t* l, af_resample_t* s)
{
uint32_t ci = l->nch; // Index for channels
uint32_t len = 0; // Number of output samples
uint32_t nch = l->nch; // Number of channels
uint32_t inc = s->dn/s->up;
uint32_t level = s->dn%s->up;
uint32_t up = s->up;
uint32_t dn = s->dn;
register int32_t i = 0;
register uint32_t wi = 0;
register uint32_t xi = 0;
// Index current channel
while(ci--){
// Temporary pointers
register int16_t* x = s->xq[ci];
register int16_t* in = ((int16_t*)c->audio)+ci;
register int16_t* out = ((int16_t*)l->audio)+ci;
register int16_t* end = in+c->len/2; // Block loop end
i = s->i; wi = s->wi; xi = s->xi;
while(in < end){
ADDQUE(xi,x,in);
in+=nch;
if((--i)<=0){
// Run the FIR filter
FIR((&x[xi]),(&s->w[wi*L]),out);
len++; out+=nch;
// Update wi to point at the correct polyphase component
wi=(wi+dn)%up;
// Insert i number of new samples in queue
i = inc;
if(wi<level) i++;
}
}
}
// Save values that needs to be kept for next time
s->wi = wi;
s->xi = xi;
s->i = i;
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;
// Make sure this filter isn't redundant
if(af->data->rate == n->rate)
return AF_DETACH;
// Create space for circular bufers (if nesessary)
if(af->data->nch != n->nch){
// 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(int16_t*));
for(i=0;i<n->nch;i++)
s->xq[i] = malloc(2*L*sizeof(int16_t));
s->xi = 0;
}
// Set parameters
af->data->nch = n->nch;
af->data->format = AFMT_S16_LE;
af->data->bps = 2;
if(af->data->format != n->format || af->data->bps != n->bps)
rv = AF_FALSE;
n->format = AFMT_S16_LE;
n->bps = 2;
// Calculate up and down sampling factors
d=gcd(af->data->rate,n->rate);
// 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*sizeof(int16_t));
// 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)){
mp_msg(MSGT_AFILTER,MSGL_ERR,"[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
float t=(float)s->up*32767.0*(*wt);
s->w[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
wt++;
}
}
free(w);
mp_msg(MSGT_AFILTER,MSGL_V,"[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_RESAMPLE:
// Reinit must be called after this function has been called
// Sanity check
if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
mp_msg(MSGT_AFILTER,MSGL_ERR,"[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];
mp_msg(MSGT_AFILTER,MSGL_V,"[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
if(s->up>s->dn)
len = upsample(c,l,s);
else
len = downsample(c,l,s);
// Set output data
c->audio = l->audio;
c->len = len*2;
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
};
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