Mercurial > mplayer.hg
view libaf/af_resample.c @ 22341:dd297c5f9ccd
Adding an extra -I.. parameter is no longer necessary.
author | diego |
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date | Tue, 27 Feb 2007 17:04:18 +0000 |
parents | ecf562795caf |
children | fd6f824ef894 |
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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 audio filter changes the sample rate. */ #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <inttypes.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_MMX) // This machine is slow #define L8 #else #define L16 #endif #include "af_resample.h" // Filtering types #define RSMP_LIN (0<<0) // Linear interpolation #define RSMP_INT (1<<0) // 16 bit integer #define RSMP_FLOAT (2<<0) // 32 bit floating point #define RSMP_MASK (3<<0) // Defines for sloppy or exact resampling #define FREQ_SLOPPY (0<<2) #define FREQ_EXACT (1<<2) #define FREQ_MASK (1<<2) // Accuracy for linear interpolation #define STEPACCURACY 32 // local data typedef struct af_resample_s { void* w; // Current filter weights void** xq; // Circular buffers uint32_t xi; // Index for circular buffers uint32_t wi; // Index for w uint32_t i; // Number of new samples to put in x queue uint32_t dn; // Down sampling factor uint32_t up; // Up sampling factor uint64_t step; // Step size for linear interpolation uint64_t pt; // Pointer remainder for linear interpolation int setup; // Setup parameters cmdline or through postcreate } af_resample_t; // Fast linear interpolation resample with modest audio quality static int linint(af_data_t* c,af_data_t* l, af_resample_t* s) { uint32_t len = 0; // Number of input samples uint32_t nch = l->nch; // Words pre transfer uint64_t step = s->step; int16_t* in16 = ((int16_t*)c->audio); int16_t* out16 = ((int16_t*)l->audio); int32_t* in32 = ((int32_t*)c->audio); int32_t* out32 = ((int32_t*)l->audio); uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY); uint64_t pt = s->pt; uint16_t tmp; switch (nch){ case 1: while(pt < end){ out16[len++]=in16[pt>>STEPACCURACY]; pt+=step; } s->pt=pt & ((1LL<<STEPACCURACY)-1); break; case 2: end/=2; while(pt < end){ out32[len++]=in32[pt>>STEPACCURACY]; pt+=step; } len=(len<<1); s->pt=pt & ((1LL<<STEPACCURACY)-1); break; default: end /=nch; while(pt < end){ tmp=nch; do { tmp--; out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch]; } while (tmp); len+=nch; pt+=step; } s->pt=pt & ((1LL<<STEPACCURACY)-1); } return len; } /* Determine resampling type and format */ static int set_types(struct af_instance_s* af, af_data_t* data) { af_resample_t* s = af->setup; int rv = AF_OK; float rd = 0; // Make sure this filter isn't redundant if((af->data->rate == data->rate) || (af->data->rate == 0)) return AF_DETACH; /* If sloppy and small resampling difference (2%) */ rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate; if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) && (data->format != (AF_FORMAT_FLOAT_NE))) || ((s->setup & RSMP_MASK) == RSMP_LIN)){ s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN; af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; af_msg(AF_MSG_VERBOSE,"[resample] Using linear interpolation. \n"); } else{ /* If the input format is float or if float is explicitly selected use float, otherwise use int */ if((data->format == (AF_FORMAT_FLOAT_NE)) || ((s->setup & RSMP_MASK) == RSMP_FLOAT)){ s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT; af->data->format = AF_FORMAT_FLOAT_NE; af->data->bps = 4; } else{ s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT; af->data->format = AF_FORMAT_S16_NE; af->data->bps = 2; } af_msg(AF_MSG_VERBOSE,"[resample] Using %s processing and %s frequecy" " conversion.\n", ((s->setup & RSMP_MASK) == RSMP_FLOAT)?"floating point":"integer", ((s->setup & FREQ_MASK) == FREQ_SLOPPY)?"inexact":"exact"); } if(af->data->format != data->format || af->data->bps != data->bps) rv = AF_FALSE; data->format = af->data->format; data->bps = af->data->bps; af->data->nch = data->nch; return rv; } // 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; // Free space for circular bufers if(s->xq){ for(i=1;i<af->data->nch;i++) if(s->xq[i]) free(s->xq[i]); free(s->xq); s->xq = NULL; } if(AF_DETACH == (rv = set_types(af,n))) return AF_DETACH; // If linear interpolation if((s->setup & RSMP_MASK) == RSMP_LIN){ s->pt=0LL; s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL; af_msg(AF_MSG_DEBUG0,"[resample] Linear interpolation step: 0x%016"PRIX64".\n", s->step); af->mul.n = af->data->rate; af->mul.d = n->rate; af_frac_cancel(&af->mul); return rv; } // Calculate up and down sampling factors d=af_gcd(af->data->rate,n->rate); // If sloppy resampling is enabled limit the upsampling factor if(((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (af->data->rate/d > 5000)){ int up=af->data->rate/2; int dn=n->rate/2; int m=2; while(af->data->rate/(d*m) > 5000){ d=af_gcd(up,dn); up/=2; dn/=2; m*=2; } d*=m; } // Create space for circular bufers s->xq = malloc(n->nch*sizeof(void*)); for(i=0;i<n->nch;i++) s->xq[i] = malloc(2*L*af->data->bps); s->xi = 0; // 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; s->wi = 0; s->i = 0; // 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*af->data->bps); // Design prototype filter type using Kaiser window with beta = 10 if(NULL == w || NULL == s->w || -1 == af_filter_design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){ af_msg(AF_MSG_ERROR,"[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 if((s->setup & RSMP_MASK) == RSMP_INT){ float t=(float)s->up*32767.0*(*wt); ((int16_t*)s->w)[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5)); } else ((float*)s->w)[i*L+j] = (float)s->up*(*wt); wt++; } } free(w); af_msg(AF_MSG_VERBOSE,"[resample] New filter designed up: %i " "down: %i\n", s->up, s->dn); } // Set multiplier and delay af->delay = (double)(1000*L/2)/((double)n->rate); af->mul.n = s->up; af->mul.d = s->dn; return rv; } case AF_CONTROL_COMMAND_LINE:{ af_resample_t* s = (af_resample_t*)af->setup; int rate=0; int type=RSMP_INT; int sloppy=1; sscanf((char*)arg,"%i:%i:%i", &rate, &sloppy, &type); s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) | (clamp(type,RSMP_LIN,RSMP_FLOAT)); return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate); } case AF_CONTROL_POST_CREATE: if((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT) ((af_resample_t*)af->setup)->setup = RSMP_FLOAT; return AF_OK; case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET: // Reinit must be called after this function has been called // Sanity check if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){ af_msg(AF_MSG_ERROR,"[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]; af_msg(AF_MSG_VERBOSE,"[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->audio); 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 switch(s->setup & RSMP_MASK){ case(RSMP_INT): # define FORMAT_I 1 if(s->up>s->dn){ # define UP # include "af_resample.h" # undef UP } else{ # define DN # include "af_resample.h" # undef DN } break; case(RSMP_FLOAT): # undef FORMAT_I # define FORMAT_F 1 if(s->up>s->dn){ # define UP # include "af_resample.h" # undef UP } else{ # define DN # include "af_resample.h" # undef DN } break; case(RSMP_LIN): len = linint(c, l, s); break; } // Set output data c->audio = l->audio; c->len = len*l->bps; 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; ((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY; return AF_OK; } // Description of this plugin af_info_t af_info_resample = { "Sample frequency conversion", "resample", "Anders", "", AF_FLAGS_REENTRANT, open };