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Diffstat (limited to 'audio/filter/af_resample_template.c')
| -rw-r--r-- | audio/filter/af_resample_template.c | 171 |
1 files changed, 171 insertions, 0 deletions
diff --git a/audio/filter/af_resample_template.c b/audio/filter/af_resample_template.c new file mode 100644 index 0000000000..4d4c5922ca --- /dev/null +++ b/audio/filter/af_resample_template.c @@ -0,0 +1,171 @@ +/* + * 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 */ |