📄 resample.c
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/* Copyright (C) 2007 Jean-Marc Valin File: resample.c Arbitrary resampling code Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: 1. Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. 2. Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. 3. The name of the author may not be used to endorse or promote products derived from this software without specific prior written permission. THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.*//* The design goals of this code are: - Very fast algorithm - SIMD-friendly algorithm - Low memory requirement - Good *perceptual* quality (and not best SNR) The code is working, but it's in a very early stage, so it may have artifacts, noise or subliminal messages from satan. Also, the API isn't stable and I can actually promise that I *will* change the API some time in the future.TODO list: - Variable calculation resolution depending on quality setting - Single vs double in float mode - 16-bit vs 32-bit (sinc only) in fixed-point mode - Make sure the filter update works even when changing params after only a few samples procesed*/#ifdef HAVE_CONFIG_H#include "config.h"#endif#ifdef OUTSIDE_SPEEX#include <stdlib.h>static void *speex_alloc (int size) {return calloc(size,1);}static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);}static void speex_free (void *ptr) {free(ptr);}#include "speex_resampler.h"#include "arch.h"#else /* OUTSIDE_SPEEX */ #include "speex/speex_resampler.h"#include "misc.h"#endif /* OUTSIDE_SPEEX */#include <math.h>#ifndef M_PI#define M_PI 3.14159263#endif#ifdef FIXED_POINT#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x))) #else#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x)))) #endif /*#define float double*/#define FILTER_SIZE 64#define OVERSAMPLE 8#define IMAX(a,b) ((a) > (b) ? (a) : (b))#ifndef NULL#define NULL 0#endiftypedef int (*resampler_basic_func)(SpeexResamplerState *, spx_uint32_t , const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);struct SpeexResamplerState_ { spx_uint32_t in_rate; spx_uint32_t out_rate; spx_uint32_t num_rate; spx_uint32_t den_rate; int quality; spx_uint32_t nb_channels; spx_uint32_t filt_len; spx_uint32_t mem_alloc_size; int int_advance; int frac_advance; float cutoff; spx_uint32_t oversample; int initialised; int started; /* These are per-channel */ spx_int32_t *last_sample; spx_uint32_t *samp_frac_num; spx_uint32_t *magic_samples; spx_word16_t *mem; spx_word16_t *sinc_table; spx_uint32_t sinc_table_length; resampler_basic_func resampler_ptr; int in_stride; int out_stride;} ;static double kaiser12_table[68] = { 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076, 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014, 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601, 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014, 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490, 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546, 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178, 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947, 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058, 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438, 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734, 0.00001000, 0.00000000};/*static double kaiser12_table[36] = { 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741, 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762, 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274, 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466, 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291, 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};*/static double kaiser10_table[36] = { 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446, 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347, 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962, 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451, 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739, 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000};static double kaiser8_table[36] = { 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200, 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126, 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272, 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758, 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490, 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000}; static double kaiser6_table[36] = { 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003, 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565, 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561, 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058, 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600, 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000};struct FuncDef { double *table; int oversample;}; static struct FuncDef _KAISER12 = {kaiser12_table, 64};#define KAISER12 (&_KAISER12)/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};#define KAISER12 (&_KAISER12)*/static struct FuncDef _KAISER10 = {kaiser10_table, 32};#define KAISER10 (&_KAISER10)static struct FuncDef _KAISER8 = {kaiser8_table, 32};#define KAISER8 (&_KAISER8)static struct FuncDef _KAISER6 = {kaiser6_table, 32};#define KAISER6 (&_KAISER6)struct QualityMapping { int base_length; int oversample; float downsample_bandwidth; float upsample_bandwidth; struct FuncDef *window_func;};/* This table maps conversion quality to internal parameters. There are two reasons that explain why the up-sampling bandwidth is larger than the down-sampling bandwidth: 1) When up-sampling, we can assume that the spectrum is already attenuated close to the Nyquist rate (from an A/D or a previous resampling filter) 2) Any aliasing that occurs very close to the Nyquist rate will be masked by the sinusoids/noise just below the Nyquist rate (guaranteed only for up-sampling).*/static const struct QualityMapping quality_map[11] = { { 8, 4, 0.830f, 0.860f, KAISER6 }, /* Q0 */ { 16, 4, 0.850f, 0.880f, KAISER6 }, /* Q1 */ { 32, 4, 0.882f, 0.910f, KAISER6 }, /* Q2 */ /* 82.3% cutoff ( ~60 dB stop) 6 */ { 48, 8, 0.895f, 0.917f, KAISER8 }, /* Q3 */ /* 84.9% cutoff ( ~80 dB stop) 8 */ { 64, 8, 0.921f, 0.940f, KAISER8 }, /* Q4 */ /* 88.7% cutoff ( ~80 dB stop) 8 */ { 80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 */ /* 89.1% cutoff (~100 dB stop) 10 */ { 96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 */ /* 91.5% cutoff (~100 dB stop) 10 */ {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */ /* 93.1% cutoff (~100 dB stop) 10 */ {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */ /* 94.5% cutoff (~100 dB stop) 10 */ {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 */ /* 95.5% cutoff (~100 dB stop) 10 */ {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */};/*8,24,40,56,80,104,128,160,200,256,320*/static double compute_func(float x, struct FuncDef *func){ float y, frac; double interp[4]; int ind; y = x*func->oversample; ind = (int)floor(y); frac = (y-ind); /* CSE with handle the repeated powers */ interp[3] = -0.1666666667*frac + 0.1666666667*(frac*frac*frac); interp[2] = frac + 0.5*(frac*frac) - 0.5*(frac*frac*frac); /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/ interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac); /* Just to make sure we don't have rounding problems */ interp[1] = 1.f-interp[3]-interp[2]-interp[0]; /*sum = frac*accum[1] + (1-frac)*accum[2];*/ return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3];}#if 0#include <stdio.h>int main(int argc, char **argv){ int i; for (i=0;i<256;i++) { printf ("%f\n", compute_func(i/256., KAISER12)); } return 0;}#endif#ifdef FIXED_POINT/* The slow way of computing a sinc for the table. Should improve that some day */static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func){ /*fprintf (stderr, "%f ", x);*/ float xx = x * cutoff; if (fabs(x)<1e-6f) return WORD2INT(32768.*cutoff); else if (fabs(x) > .5f*N) return 0; /*FIXME: Can it really be any slower than this? */ return WORD2INT(32768.*cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func));}#else/* The slow way of computing a sinc for the table. Should improve that some day */static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func){ /*fprintf (stderr, "%f ", x);*/ float xx = x * cutoff; if (fabs(x)<1e-6) return cutoff; else if (fabs(x) > .5*N) return 0; /*FIXME: Can it really be any slower than this? */ return cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func);}#endif#ifdef FIXED_POINTstatic void cubic_coef(spx_word16_t x, spx_word16_t interp[4]){ /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation but I know it's MMSE-optimal on a sinc */ spx_word16_t x2, x3; x2 = MULT16_16_P15(x, x); x3 = MULT16_16_P15(x, x2); interp[0] = PSHR32(MULT16_16(QCONST16(-0.16667f, 15),x) + MULT16_16(QCONST16(0.16667f, 15),x3),15); interp[1] = EXTRACT16(EXTEND32(x) + SHR32(SUB32(EXTEND32(x2),EXTEND32(x3)),1)); interp[3] = PSHR32(MULT16_16(QCONST16(-0.33333f, 15),x) + MULT16_16(QCONST16(.5f,15),x2) - MULT16_16(QCONST16(0.16667f, 15),x3),15); /* Just to make sure we don't have rounding problems */ interp[2] = Q15_ONE-interp[0]-interp[1]-interp[3]; if (interp[2]<32767) interp[2]+=1;}#elsestatic void cubic_coef(spx_word16_t frac, spx_word16_t interp[4]){ /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation but I know it's MMSE-optimal on a sinc */ interp[0] = -0.16667f*frac + 0.16667f*frac*frac*frac; interp[1] = frac + 0.5f*frac*frac - 0.5f*frac*frac*frac; /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/ interp[3] = -0.33333f*frac + 0.5f*frac*frac - 0.16667f*frac*frac*frac; /* Just to make sure we don't have rounding problems */ interp[2] = 1.-interp[0]-interp[1]-interp[3];}#endifstatic int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len){ int N = st->filt_len; int out_sample = 0; spx_word16_t *mem; int last_sample = st->last_sample[channel_index]; spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index]; mem = st->mem + channel_index * st->mem_alloc_size; while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) { int j; spx_word32_t sum=0; /* We already have all the filter coefficients pre-computed in the table */ const spx_word16_t *ptr; /* Do the memory part */ for (j=0;last_sample-N+1+j < 0;j++) { sum += MULT16_16(mem[last_sample+j],st->sinc_table[samp_frac_num*st->filt_len+j]); } /* Do the new part */ ptr = in+st->in_stride*(last_sample-N+1+j); for (;j<N;j++) { sum += MULT16_16(*ptr,st->sinc_table[samp_frac_num*st->filt_len+j]); ptr += st->in_stride; } *out = PSHR32(sum,15); out += st->out_stride; out_sample++; last_sample += st->int_advance; samp_frac_num += st->frac_advance; if (samp_frac_num >= st->den_rate) { samp_frac_num -= st->den_rate; last_sample++; } } st->last_sample[channel_index] = last_sample; st->samp_frac_num[channel_index] = samp_frac_num; return out_sample;}#ifdef FIXED_POINT
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