mpv/libaf/af_resample.h

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/*=============================================================================
//
// This software has been released under the terms of the GNU General Public
// license. See http://www.gnu.org/copyleft/gpl.html for details.
//
// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au
//
//=============================================================================
*/
/* This file contains the resampling engine, the sample format is
controlled by the FORMAT parameter, the filter length by the L
parameter and the resampling type by UP and DN. This file should
only be included by af_resample.c
*/
#undef L
#undef SHIFT
#undef FORMAT
#undef FIR
#undef ADDQUE
/* The length Lxx definition selects the length of each poly phase
component. Valid definitions are L8 and L16 where the number
defines the nuber of taps. This definition affects the
computational complexity, the performance and the memory usage.
*/
/* The FORMAT_x parameter selects the sample format type currently
float and int16 are supported. Thes two formats are selected by
defining eiter FORMAT_F or FORMAT_I. The advantage of using float
is that the amplitude and therefore the SNR isn't affected by the
filtering, the disadvantage is that it is a lot slower.
*/
#if defined(FORMAT_I)
#define SHIFT >>16
#define FORMAT int16_t
#else
#define SHIFT
#define FORMAT float
#endif
// Short filter
#if defined(L8)
#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] ) SHIFT
#else /* L8/L16 */
#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] ) SHIFT
#endif /* L8/L16 */
// Macro to add data to circular que
#define ADDQUE(xi,xq,in)\
xq[xi]=xq[(xi)+L]=*(in);\
xi=((xi)-1)&(L-1);
#if defined(UP)
uint32_t ci = l->nch; // Index for channels
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;
uint32_t ns = c->len/l->bps;
register FORMAT* w = s->w;
register uint32_t wi = 0;
register uint32_t xi = 0;
// Index current channel
while(ci--){
// Temporary pointers
register FORMAT* x = s->xq[ci];
register FORMAT* in = ((FORMAT*)c->audio)+ci;
register FORMAT* out = ((FORMAT*)l->audio)+ci;
FORMAT* end = in+ns; // 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;
#endif /* UP */
#if defined(DN) /* DN */
uint32_t ci = l->nch; // Index for channels
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;
uint32_t ns = c->len/l->bps;
FORMAT* w = s->w;
register int32_t i = 0;
register uint32_t wi = 0;
register uint32_t xi = 0;
// Index current channel
while(ci--){
// Temporary pointers
register FORMAT* x = s->xq[ci];
register FORMAT* in = ((FORMAT*)c->audio)+ci;
register FORMAT* out = ((FORMAT*)l->audio)+ci;
register FORMAT* end = in+ns; // 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]),(&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;
#endif /* DN */