mirror of
https://github.com/mpv-player/mpv
synced 2024-12-23 15:22:09 +00:00
23486f48a5
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8958 b3059339-0415-0410-9bf9-f77b7e298cf2
162 lines
4.2 KiB
C
162 lines
4.2 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 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 lenght 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 */
|