📄 resampv.c
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/* resampv.c -- use sinc interpolation to resample at a time-varying sample rate *//* CHANGE LOG * -------------------------------------------------------------------- * 28Apr03 dm min->MIN, max->MAX */#include "stdio.h"#ifndef mips#include "stdlib.h"#endif#include "xlisp.h"#include "sound.h"#include "falloc.h"#include "cext.h"#include "resampv.h"#include "fresample.h"#include "ffilterkit.h"#include "fsmallfilter.h"/* Algorithm: First compute a factor = ratio of new sample rate to original sample rate. We have Time, the offset into X We want Xoff = ((susp->Nmult + 1) / 2.0) * MAX(1.0, 1.0 / factor) + 10 samples on either side of Time before we interpolate. If Xoff * 2 > Xsize, then we're in trouble because X is not big enough. Assume this is a pathalogical case, raise the cutoff frequency to reduce Xoff to less than Xsize/2. If Time is too small, then we're in trouble because we've lost the beginning of the buffer. Raise the cutoff frequency until Xoff is less than Time. This should only happen if factor suddenly drops. If Time is too big, we can handle it: shift X down and load X with new samples. When X is shifted by N samples, N is subtracted from Time. To minimize the "Time is too small" case, don't shift too far: leave a cushion of Xoff * 2 samples rather than the usual Xoff. Now compute a sample at Time using SrcUD and output it. What is Time? Time is the offset into X, so Time is g_of_now - (sum of all X shifts) So, let Time = g_of_now - shift_sum Whenever shift_sum or g_of_now is updated, recompute Time To compute the next g_of_now, do a lookup of g at now + 1/sr, using linear interpolation (be sure to do computation with doubles to minimize sampling time jitter). *//* maximum ratio for downsampling (downsampling will still take place, * but the lowest prefilter cutoff frequency will be * (original_sample_rate/2) / MAX_FACTOR_INVERSE */#define MAX_FACTOR_INVERSE 64typedef struct resamplev_susp_struct { snd_susp_node susp; long terminate_cnt; boolean logically_stopped; sound_type f; long f_cnt; sample_block_values_type f_ptr; sound_type g; long g_cnt; sample_block_values_type g_ptr; double prev_g; /* data for interpolation: */ double next_g; double phase_in_g; double phase_in_g_increment; double g_of_now; float *X; long Xsize; double Time; /* location (offset) in X of next output sample */ double shift_sum; double LpScl; double factor_inverse; /* computed at every sample from g */ double factor; /* computed from factor_inverse */ sample_type *Imp; sample_type *ImpD; boolean interpFilt; int Nmult; int Nwing; int Xp; /* first free location at end of X */ int Xoff; /* number of extra samples at beginning and end of X */} resamplev_susp_node, *resamplev_susp_type;void resamplev_free();void resampv_refill(resamplev_susp_type susp);/* Sampling rate conversion subroutine * Note that this is not the same as SrcUD in resamp.c! */static float SrcUD(float X[], double dt, double Time, int Nwing, double LpScl, float Imp[], float ImpD[], boolean Interp){ mem_float *Xp; fast_float v; double dh; /* Step through filter impulse response */ long iTime = (long) Time; dh = MIN(Npc, Npc/dt); /* Filter sampling period */ Xp = &X[iTime]; /* Ptr to current input sample */ v = FilterUD(Imp, ImpD, Nwing, Interp, Xp, Time - iTime, -1, dh); /* Perform left-wing inner product */ v += FilterUD(Imp, ImpD, Nwing, Interp, Xp+1, (1 + iTime) - Time, 1, dh); /* Perform right-wing inner product */ v *= LpScl; /* Normalize for unity filter gain */ return (float) v;}void resamplev__fetch(register resamplev_susp_type susp, snd_list_type snd_list){ int cnt = 0; /* how many samples computed */ sample_block_type out; /* note that in this fetch routine, out_ptr is used to remember where * to put the "real" output, while X_ptr_reg is used in the inner * loop that copies input samples into X, a buffer */ register sample_block_values_type out_ptr; falloc_sample_block(out, "resamplev__fetch"); out_ptr = out->samples; snd_list->block = out; while (cnt < max_sample_block_len) { /* outer loop */ /* fetch g until we have points to interpolate */ while (susp->phase_in_g >= 1.0) { susp->prev_g = susp->next_g; if (susp->g_cnt == 0) { susp_get_samples(g, g_ptr, g_cnt); if (susp->g->logical_stop_cnt == susp->g->current - susp->g_cnt) { if (susp->susp.log_stop_cnt == UNKNOWN) { susp->susp.log_stop_cnt = susp->susp.current + cnt; nyquist_printf("g logically stops at %ld\n", susp->susp.log_stop_cnt); } } if (susp->g_ptr == zero_block->samples && susp->terminate_cnt == UNKNOWN) { susp->terminate_cnt = susp->susp.current + cnt; nyquist_printf("g terminates at %ld\n", susp->terminate_cnt); } } susp->next_g = susp_fetch_sample(g, g_ptr, g_cnt); susp->phase_in_g -= 1.0; if (susp->next_g < susp->prev_g) { susp->next_g = susp->prev_g; } /* factor_inverse = 1/factor = how many samples of f per * output sample = change-in-g / output-samples-per-g-sample * = change-in-g * phase_in_g_increment */ susp->factor_inverse = susp->phase_in_g_increment * (susp->next_g - susp->prev_g); if (susp->factor_inverse > MAX_FACTOR_INVERSE) susp->factor_inverse = MAX_FACTOR_INVERSE; /* update Xoff, which depends upon factor_inverse: */ susp->Xoff = (int) (((susp->Nmult + 1) / 2.0) * MAX(1.0, susp->factor_inverse)) + 10; if (susp->Xoff * 2 > susp->Xsize) { /* bad because X is not big enough for filter, so we'll * raise the cutoff frequency as necessary */ susp->factor_inverse = ((susp->Xsize / 2) - 10 ) / ((susp->Nmult + 1) / 2.0); susp->Xoff = (susp->Xsize / 2) - 2 /* fudge factor */; } } susp->g_of_now = susp->prev_g + susp->phase_in_g * (susp->next_g - susp->prev_g); susp->Time = susp->g_of_now - susp->shift_sum; susp->phase_in_g += susp->phase_in_g_increment; /* now we have a position (g_of_now) and a factor */ /* See if enough of f is in X */ if (susp->Xoff > susp->Time) { /* there are not enough samples before Time in X, so * modify factor_inverse to fix it */ susp->factor_inverse = (susp->Time - 10.0) / ((susp->Nmult + 1) / 2.0); } else if ((susp->Xsize - susp->Xoff) < susp->Time) { /* if there's room, leave 2*Xoff samples at beginning of * buffer. Otherwise leave only Xoff:*/ int i, dist; susp->Xp = susp->Xoff * 2; dist = ((int) susp->Time) - susp->Xp; if (dist < 1 && (susp->Xp * 2 < susp->Xsize)) { susp->Xp = susp->Xoff; dist = ((int) susp->Time) - susp->Xoff; } for (i = 0; i < susp->Xsize - dist; i++) { susp->X[i] = susp->X[i + dist]; } resampv_refill(susp); susp->shift_sum += dist; susp->Time = susp->g_of_now - susp->shift_sum; } /* second, compute a sample to output */ /* don't run past terminate time */ if (susp->terminate_cnt == susp->susp.current + cnt) { snd_list->block_len = cnt; susp->susp.current += cnt; snd_list = snd_list->u.next; snd_list->u.next = snd_list_create(&susp->susp); snd_list->block = internal_zero_block; nyquist_printf("resamp calling snd_list_terminate\n"); snd_list_terminate(snd_list); return; } else { double scale = susp->LpScl; float tmp; if (susp->factor_inverse > 1) scale /= susp->factor_inverse; tmp = SrcUD(susp->X, susp->factor_inverse, susp->Time, susp->Nwing, scale, susp->Imp, susp->ImpD, susp->interpFilt); *out_ptr++ = tmp; } cnt++; } snd_list->block_len = cnt; susp->susp.current += cnt;} /* resamplev__fetch */void resampv_refill(resamplev_susp_type susp) { int togo, n; register sample_type *f_ptr_reg; register sample_type *X_ptr_reg; while (susp->Xp < susp->Xsize) { /* outer loop */ /* read samples from susp->f into X */ togo = susp->Xsize - susp->Xp; /* don't run past the f input sample block: */ susp_check_samples(f, f_ptr, f_cnt); togo = MIN(togo, susp->f_cnt); n = togo; f_ptr_reg = susp->f_ptr; X_ptr_reg = &(susp->X[susp->Xp]); if (n) do { /* the inner sample computation loop */ *X_ptr_reg++ = *f_ptr_reg++; } while (--n); /* inner loop */ /* using f_ptr_reg is a bad idea on RS/6000: */ susp->f_ptr += togo; susp_took(f_cnt, togo); susp->Xp += togo; } /* outer loop */}void resamplev_mark(resamplev_susp_type susp){ sound_xlmark(susp->f); sound_xlmark(susp->g);}void resamplev_free(resamplev_susp_type susp){ nyquist_printf("resamplev_free called\n"); sound_unref(susp->f); sound_unref(susp->g); free(susp->X); ffree_generic(susp, sizeof(resamplev_susp_node), "resamplev_free");}void resamplev_print_tree(resamplev_susp_type susp, int n){ indent(n); nyquist_printf("f:"); sound_print_tree_1(susp->f, n); indent(n); nyquist_printf("g:"); sound_print_tree_1(susp->g, n);}sound_type snd_make_resamplev(sound_type f, rate_type sr, sound_type g){ register resamplev_susp_type susp; int i; falloc_generic(susp, resamplev_susp_node, "snd_make_resamplev"); susp->susp.fetch = resamplev__fetch; susp->Nmult = SMALL_FILTER_NMULT; susp->Imp = SMALL_FILTER_IMP; susp->ImpD = SMALL_FILTER_IMPD; susp->LpScl = SMALL_FILTER_SCALE / 32768.0; susp->LpScl /= 16384.0; /* this is just a fudge factor, is SMALL_FILTER_SCALE is wrong? */ susp->LpScl /= 1.0011; susp->Nwing = SMALL_FILTER_NWING; susp->terminate_cnt = UNKNOWN; /* initialize susp state */ susp->susp.free = resamplev_free; susp->susp.sr = sr; susp->susp.t0 = f->t0; susp->susp.mark = resamplev_mark; susp->susp.print_tree = resamplev_print_tree; susp->susp.name = "resamplev"; susp->logically_stopped = false; susp->susp.log_stop_cnt = logical_stop_cnt_cvt(f); susp->susp.current = 0; susp->f = f; susp->f_cnt = 0; susp->g = g; susp->g_cnt = 0; susp->next_g = 0; susp->phase_in_g_increment = g->sr / sr; susp->phase_in_g = 2.0; susp->Xoff = (int) (((susp->Nmult + 1) / 2.0) * 2.0) /* MAX(1.0, 1.0 / susp->factor) */ + 10; /* this size is not critical unless it is too small */ susp->Xsize = max_sample_block_len + 2 * susp->Xoff; susp->X = calloc(susp->Xsize, sizeof(sample_type)); susp->Xp = susp->Xsize; susp->shift_sum = -susp->Xsize; susp->interpFilt = true; for (i = 0; i < susp->Xoff; i++) susp->X[i] = 0.0F; susp->LpScl *= 0.95; /* reduce probability of clipping */ nyquist_printf("resampv: scaling by 0.95\n"); return sound_create((snd_susp_type)susp, susp->susp.t0, susp->susp.sr, 1.0 /* scale factor */);}sound_type snd_resamplev(sound_type f, rate_type sr, sound_type g){ sound_type f_copy = sound_copy(f); sound_type g_copy = sound_copy(g); g_copy->scale *= (float) sr; /* put g_copy in units of samples */ return snd_make_resamplev(f_copy, sr, g_copy);}
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