diff options
Diffstat (limited to 'audio/filter/af_resample_template.c')
| -rw-r--r-- | audio/filter/af_resample_template.c | 171 |
1 files changed, 0 insertions, 171 deletions
diff --git a/audio/filter/af_resample_template.c b/audio/filter/af_resample_template.c deleted file mode 100644 index 4d4c5922ca..0000000000 --- a/audio/filter/af_resample_template.c +++ /dev/null @@ -1,171 +0,0 @@ -/* - * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au - * - * This file is part of MPlayer. - * - * MPlayer is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * MPlayer is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with MPlayer; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -/* 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 */ |