Mercurial > mplayer.hg
view libaf/af_resample.c @ 7579:24860f7c716f
change i18n to autodetect
author | pontscho |
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date | Wed, 02 Oct 2002 09:24:34 +0000 |
parents | 8819fdf88b5d |
children | 255039c14525 |
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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){ \ int16_t a = (w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3]) >> 16; \ int16_t b = (w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7]) >> 16; \ (y[0]) = a+b; \ } #else /* Fast machine */ #define L 16 // Unrolled loop to speed up execution #define FIR(x,w,y){ \ int16_t a = (w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] ) >> 16; \ int16_t b = (w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] ) >> 16; \ int16_t c = (w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11]) >> 16; \ int16_t d = (w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15]) >> 16; \ y[0] = (a+b+c+d) >> 1; \ } #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 int16_t xi; // Index for circular buffers int16_t wi; // Index for w uint16_t i; // Number of new samples to put in x queue uint16_t dn; // Down sampling factor uint16_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) { uint16_t ci = l->nch; // Index for channels uint16_t len = 0; // Number of input samples uint16_t nch = l->nch; // Number of channels uint16_t inc = s->up/s->dn; uint16_t level = s->up%s->dn; uint16_t up = s->up; uint16_t dn = s->dn; register int16_t* w = s->w; register uint16_t wi = 0; register uint16_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 uint16_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) { uint16_t ci = l->nch; // Index for channels uint16_t len = 0; // Number of output samples uint16_t nch = l->nch; // Number of channels uint16_t inc = s->dn/s->up; uint16_t level = s->dn%s->up; uint16_t up = s->up; uint16_t dn = s->dn; register uint16_t i = 0; register uint16_t wi = 0; register uint16_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){ // 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 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", "", open };