+++ /dev/null
-/* Copyright (C) 2007-2008 Jean-Marc Valin
- Copyright (C) 2008 Thorvald Natvig
-
- 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)
-
- Warning: This resampler is relatively new. Although I think I got rid of
- all the major bugs and I don't expect the API to change anymore, there
- may be something I've missed. So use with caution.
-
- This algorithm is based on this original resampling algorithm:
- Smith, Julius O. Digital Audio Resampling Home Page
- Center for Computer Research in Music and Acoustics (CCRMA),
- Stanford University, 2007.
- Web published at http://www-ccrma.stanford.edu/~jos/resample/.
-
- There is one main difference, though. This resampler uses cubic
- interpolation instead of linear interpolation in the above paper. This
- makes the table much smaller and makes it possible to compute that table
- on a per-stream basis. In turn, being able to tweak the table for each
- stream makes it possible to both reduce complexity on simple ratios
- (e.g. 2/3), and get rid of the rounding operations in the inner loop.
- The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
-*/
-
-#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 "arch.h"
-#include "os_support.h"
-#endif /* OUTSIDE_SPEEX */
-
-#include "stack_alloc.h"
-#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 IMAX(a,b) ((a) > (b) ? (a) : (b))
-#define IMIN(a,b) ((a) < (b) ? (a) : (b))
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-#ifdef _USE_SSE
-#include "resample_sse.h"
-#endif
-
-/* Numer of elements to allocate on the stack */
-#ifdef VAR_ARRAYS
-#define FIXED_STACK_ALLOC 8192
-#else
-#define FIXED_STACK_ALLOC 1024
-#endif
-
-typedef 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;
- spx_uint32_t buffer_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_POINT
-static 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;
-}
-#else
-static 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];
-}
-#endif
-
-static 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)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const spx_word16_t *sinc_table = st->sinc_table;
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- spx_word32_t sum;
- int j;
-
- while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
- {
- const spx_word16_t *sinc = & sinc_table[samp_frac_num*N];
- const spx_word16_t *iptr = & in[last_sample];
-
-#ifndef OVERRIDE_INNER_PRODUCT_SINGLE
- float accum[4] = {0,0,0,0};
-
- for(j=0;j<N;j+=4) {
- accum[0] += sinc[j]*iptr[j];
- accum[1] += sinc[j+1]*iptr[j+1];
- accum[2] += sinc[j+2]*iptr[j+2];
- accum[3] += sinc[j+3]*iptr[j+3];
- }
- sum = accum[0] + accum[1] + accum[2] + accum[3];
-#else
- sum = inner_product_single(sinc, iptr, N);
-#endif
-
- out[out_stride * out_sample++] = PSHR32(sum, 15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate)
- {
- samp_frac_num -= 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
-#else
-/* This is the same as the previous function, except with a double-precision accumulator */
-static int resampler_basic_direct_double(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)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const spx_word16_t *sinc_table = st->sinc_table;
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- double sum;
- int j;
-
- while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
- {
- const spx_word16_t *sinc = & sinc_table[samp_frac_num*N];
- const spx_word16_t *iptr = & in[last_sample];
-
-#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
- double accum[4] = {0,0,0,0};
-
- for(j=0;j<N;j+=4) {
- accum[0] += sinc[j]*iptr[j];
- accum[1] += sinc[j+1]*iptr[j+1];
- accum[2] += sinc[j+2]*iptr[j+2];
- accum[3] += sinc[j+3]*iptr[j+3];
- }
- sum = accum[0] + accum[1] + accum[2] + accum[3];
-#else
- sum = inner_product_double(sinc, iptr, N);
-#endif
-
- out[out_stride * out_sample++] = PSHR32(sum, 15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate)
- {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-static int resampler_basic_interpolate_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)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- int j;
- spx_word32_t sum;
-
- while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
- {
- const spx_word16_t *iptr = & in[last_sample];
-
- const int offset = samp_frac_num*st->oversample/st->den_rate;
-#ifdef FIXED_POINT
- const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
-#else
- const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
-#endif
- spx_word16_t interp[4];
-
-
-#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
- spx_word32_t accum[4] = {0,0,0,0};
-
- for(j=0;j<N;j++) {
- const spx_word16_t curr_in=iptr[j];
- accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
- accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
- accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
- accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
- }
-
- cubic_coef(frac, interp);
- sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
-#else
- cubic_coef(frac, interp);
- sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
-#endif
-
- out[out_stride * out_sample++] = PSHR32(sum,15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate)
- {
- samp_frac_num -= 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
-#else
-/* This is the same as the previous function, except with a double-precision accumulator */
-static int resampler_basic_interpolate_double(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)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- int j;
- spx_word32_t sum;
-
- while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
- {
- const spx_word16_t *iptr = & in[last_sample];
-
- const int offset = samp_frac_num*st->oversample/st->den_rate;
-#ifdef FIXED_POINT
- const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
-#else
- const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
-#endif
- spx_word16_t interp[4];
-
-
-#ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
- double accum[4] = {0,0,0,0};
-
- for(j=0;j<N;j++) {
- const double curr_in=iptr[j];
- accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
- accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
- accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
- accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
- }
-
- cubic_coef(frac, interp);
- sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
-#else
- cubic_coef(frac, interp);
- sum = interpolate_product_double(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
-#endif
-
- out[out_stride * out_sample++] = PSHR32(sum,15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate)
- {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-static void update_filter(SpeexResamplerState *st)
-{
- spx_uint32_t old_length;
-
- old_length = st->filt_len;
- st->oversample = quality_map[st->quality].oversample;
- st->filt_len = quality_map[st->quality].base_length;
-
- if (st->num_rate > st->den_rate)
- {
- /* down-sampling */
- st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
- /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
- st->filt_len = st->filt_len*st->num_rate / st->den_rate;
- /* Round down to make sure we have a multiple of 4 */
- st->filt_len &= (~0x3);
- if (2*st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (4*st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (8*st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (16*st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (st->oversample < 1)
- st->oversample = 1;
- } else {
- /* up-sampling */
- st->cutoff = quality_map[st->quality].upsample_bandwidth;
- }
-
- /* Choose the resampling type that requires the least amount of memory */
- if (st->den_rate <= st->oversample)
- {
- spx_uint32_t i;
- if (!st->sinc_table)
- st->sinc_table = (spx_word16_t *)speex_alloc(st->filt_len*st->den_rate*sizeof(spx_word16_t));
- else if (st->sinc_table_length < st->filt_len*st->den_rate)
- {
- st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,st->filt_len*st->den_rate*sizeof(spx_word16_t));
- st->sinc_table_length = st->filt_len*st->den_rate;
- }
- for (i=0;i<st->den_rate;i++)
- {
- spx_int32_t j;
- for (j=0;j<st->filt_len;j++)
- {
- st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-(spx_int32_t)st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func);
- }
- }
-#ifdef FIXED_POINT
- st->resampler_ptr = resampler_basic_direct_single;
-#else
- if (st->quality>8)
- st->resampler_ptr = resampler_basic_direct_double;
- else
- st->resampler_ptr = resampler_basic_direct_single;
-#endif
- /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/
- } else {
- spx_int32_t i;
- if (!st->sinc_table)
- st->sinc_table = (spx_word16_t *)speex_alloc((st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
- else if (st->sinc_table_length < st->filt_len*st->oversample+8)
- {
- st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,(st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
- st->sinc_table_length = st->filt_len*st->oversample+8;
- }
- for (i=-4;i<(spx_int32_t)(st->oversample*st->filt_len+4);i++)
- st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func);
-#ifdef FIXED_POINT
- st->resampler_ptr = resampler_basic_interpolate_single;
-#else
- if (st->quality>8)
- st->resampler_ptr = resampler_basic_interpolate_double;
- else
- st->resampler_ptr = resampler_basic_interpolate_single;
-#endif
- /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/
- }
- st->int_advance = st->num_rate/st->den_rate;
- st->frac_advance = st->num_rate%st->den_rate;
-
-
- /* Here's the place where we update the filter memory to take into account
- the change in filter length. It's probably the messiest part of the code
- due to handling of lots of corner cases. */
- if (!st->mem)
- {
- spx_uint32_t i;
- st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
- st->mem = (spx_word16_t*)speex_alloc(st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
- for (i=0;i<st->nb_channels*st->mem_alloc_size;i++)
- st->mem[i] = 0;
- /*speex_warning("init filter");*/
- } else if (!st->started)
- {
- spx_uint32_t i;
- st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
- st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
- for (i=0;i<st->nb_channels*st->mem_alloc_size;i++)
- st->mem[i] = 0;
- /*speex_warning("reinit filter");*/
- } else if (st->filt_len > old_length)
- {
- spx_int32_t i;
- /* Increase the filter length */
- /*speex_warning("increase filter size");*/
- int old_alloc_size = st->mem_alloc_size;
- if ((st->filt_len-1 + st->buffer_size) > st->mem_alloc_size)
- {
- st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
- st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
- }
- for (i=st->nb_channels-1;i>=0;i--)
- {
- spx_int32_t j;
- spx_uint32_t olen = old_length;
- /*if (st->magic_samples[i])*/
- {
- /* Try and remove the magic samples as if nothing had happened */
-
- /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
- olen = old_length + 2*st->magic_samples[i];
- for (j=old_length-2+st->magic_samples[i];j>=0;j--)
- st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j];
- for (j=0;j<st->magic_samples[i];j++)
- st->mem[i*st->mem_alloc_size+j] = 0;
- st->magic_samples[i] = 0;
- }
- if (st->filt_len > olen)
- {
- /* If the new filter length is still bigger than the "augmented" length */
- /* Copy data going backward */
- for (j=0;j<olen-1;j++)
- st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*st->mem_alloc_size+(olen-2-j)];
- /* Then put zeros for lack of anything better */
- for (;j<st->filt_len-1;j++)
- st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = 0;
- /* Adjust last_sample */
- st->last_sample[i] += (st->filt_len - olen)/2;
- } else {
- /* Put back some of the magic! */
- st->magic_samples[i] = (olen - st->filt_len)/2;
- for (j=0;j<st->filt_len-1+st->magic_samples[i];j++)
- st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
- }
- }
- } else if (st->filt_len < old_length)
- {
- spx_uint32_t i;
- /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
- samples so they can be used directly as input the next time(s) */
- for (i=0;i<st->nb_channels;i++)
- {
- spx_uint32_t j;
- spx_uint32_t old_magic = st->magic_samples[i];
- st->magic_samples[i] = (old_length - st->filt_len)/2;
- /* We must copy some of the memory that's no longer used */
- /* Copy data going backward */
- for (j=0;j<st->filt_len-1+st->magic_samples[i]+old_magic;j++)
- st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
- st->magic_samples[i] += old_magic;
- }
- }
-
-}
-
-EXPORT SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
-{
- return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality, err);
-}
-
-EXPORT SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate, int quality, int *err)
-{
- spx_uint32_t i;
- SpeexResamplerState *st;
- if (quality > 10 || quality < 0)
- {
- if (err)
- *err = RESAMPLER_ERR_INVALID_ARG;
- return NULL;
- }
- st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState));
- st->initialised = 0;
- st->started = 0;
- st->in_rate = 0;
- st->out_rate = 0;
- st->num_rate = 0;
- st->den_rate = 0;
- st->quality = -1;
- st->sinc_table_length = 0;
- st->mem_alloc_size = 0;
- st->filt_len = 0;
- st->mem = 0;
- st->resampler_ptr = 0;
-
- st->cutoff = 1.f;
- st->nb_channels = nb_channels;
- st->in_stride = 1;
- st->out_stride = 1;
-
-#ifdef FIXED_POINT
- st->buffer_size = 160;
-#else
- st->buffer_size = 160;
-#endif
-
- /* Per channel data */
- st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(int));
- st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
- st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
- for (i=0;i<nb_channels;i++)
- {
- st->last_sample[i] = 0;
- st->magic_samples[i] = 0;
- st->samp_frac_num[i] = 0;
- }
-
- speex_resampler_set_quality(st, quality);
- speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);
-
-
- update_filter(st);
-
- st->initialised = 1;
- if (err)
- *err = RESAMPLER_ERR_SUCCESS;
-
- return st;
-}
-
-EXPORT void speex_resampler_destroy(SpeexResamplerState *st)
-{
- speex_free(st->mem);
- speex_free(st->sinc_table);
- speex_free(st->last_sample);
- speex_free(st->magic_samples);
- speex_free(st->samp_frac_num);
- speex_free(st);
-}
-
-static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t channel_index, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
-{
- int j=0;
- const int N = st->filt_len;
- int out_sample = 0;
- spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
- spx_uint32_t ilen;
-
- st->started = 1;
-
- /* Call the right resampler through the function ptr */
- out_sample = st->resampler_ptr(st, channel_index, mem, in_len, out, out_len);
-
- if (st->last_sample[channel_index] < (spx_int32_t)*in_len)
- *in_len = st->last_sample[channel_index];
- *out_len = out_sample;
- st->last_sample[channel_index] -= *in_len;
-
- ilen = *in_len;
-
- for(j=0;j<N-1;++j)
- mem[j] = mem[j+ilen];
-
- return RESAMPLER_ERR_SUCCESS;
-}
-
-static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_index, spx_word16_t **out, spx_uint32_t out_len) {
- spx_uint32_t tmp_in_len = st->magic_samples[channel_index];
- spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
- const int N = st->filt_len;
-
- speex_resampler_process_native(st, channel_index, &tmp_in_len, *out, &out_len);
-
- st->magic_samples[channel_index] -= tmp_in_len;
-
- /* If we couldn't process all "magic" input samples, save the rest for next time */
- if (st->magic_samples[channel_index])
- {
- spx_uint32_t i;
- for (i=0;i<st->magic_samples[channel_index];i++)
- mem[N-1+i]=mem[N-1+i+tmp_in_len];
- }
- *out += out_len*st->out_stride;
- return out_len;
-}
-
-#ifdef FIXED_POINT
-EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
-#else
-EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
-#endif
-{
- int j;
- spx_uint32_t ilen = *in_len;
- spx_uint32_t olen = *out_len;
- spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
- const int filt_offs = st->filt_len - 1;
- const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
- const int istride = st->in_stride;
-
- if (st->magic_samples[channel_index])
- olen -= speex_resampler_magic(st, channel_index, &out, olen);
- if (! st->magic_samples[channel_index]) {
- while (ilen && olen) {
- spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
- spx_uint32_t ochunk = olen;
-
- if (in) {
- for(j=0;j<ichunk;++j)
- x[j+filt_offs]=in[j*istride];
- } else {
- for(j=0;j<ichunk;++j)
- x[j+filt_offs]=0;
- }
- speex_resampler_process_native(st, channel_index, &ichunk, out, &ochunk);
- ilen -= ichunk;
- olen -= ochunk;
- out += ochunk * st->out_stride;
- if (in)
- in += ichunk * istride;
- }
- }
- *in_len -= ilen;
- *out_len -= olen;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-#ifdef FIXED_POINT
-EXPORT int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
-#else
-EXPORT int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
-#endif
-{
- int j;
- const int istride_save = st->in_stride;
- const int ostride_save = st->out_stride;
- spx_uint32_t ilen = *in_len;
- spx_uint32_t olen = *out_len;
- spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
- const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1);
-#ifdef VAR_ARRAYS
- const unsigned int ylen = (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC;
- VARDECL(spx_word16_t *ystack);
- ALLOC(ystack, ylen, spx_word16_t);
-#else
- const unsigned int ylen = FIXED_STACK_ALLOC;
- spx_word16_t ystack[FIXED_STACK_ALLOC];
-#endif
-
- st->out_stride = 1;
-
- while (ilen && olen) {
- spx_word16_t *y = ystack;
- spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
- spx_uint32_t ochunk = (olen > ylen) ? ylen : olen;
- spx_uint32_t omagic = 0;
-
- if (st->magic_samples[channel_index]) {
- omagic = speex_resampler_magic(st, channel_index, &y, ochunk);
- ochunk -= omagic;
- olen -= omagic;
- }
- if (! st->magic_samples[channel_index]) {
- if (in) {
- for(j=0;j<ichunk;++j)
-#ifdef FIXED_POINT
- x[j+st->filt_len-1]=WORD2INT(in[j*istride_save]);
-#else
- x[j+st->filt_len-1]=in[j*istride_save];
-#endif
- } else {
- for(j=0;j<ichunk;++j)
- x[j+st->filt_len-1]=0;
- }
-
- speex_resampler_process_native(st, channel_index, &ichunk, y, &ochunk);
- } else {
- ichunk = 0;
- ochunk = 0;
- }
-
- for (j=0;j<ochunk+omagic;++j)
-#ifdef FIXED_POINT
- out[j*ostride_save] = ystack[j];
-#else
- out[j*ostride_save] = WORD2INT(ystack[j]);
-#endif
-
- ilen -= ichunk;
- olen -= ochunk;
- out += (ochunk+omagic) * ostride_save;
- if (in)
- in += ichunk * istride_save;
- }
- st->out_stride = ostride_save;
- *in_len -= ilen;
- *out_len -= olen;
-
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
- spx_uint32_t bak_len = *out_len;
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- st->in_stride = st->out_stride = st->nb_channels;
- for (i=0;i<st->nb_channels;i++)
- {
- *out_len = bak_len;
- if (in != NULL)
- speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len);
- else
- speex_resampler_process_float(st, i, NULL, in_len, out+i, out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
- spx_uint32_t bak_len = *out_len;
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- st->in_stride = st->out_stride = st->nb_channels;
- for (i=0;i<st->nb_channels;i++)
- {
- *out_len = bak_len;
- if (in != NULL)
- speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len);
- else
- speex_resampler_process_int(st, i, NULL, in_len, out+i, out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int speex_resampler_set_rate(SpeexResamplerState *st, spx_uint32_t in_rate, spx_uint32_t out_rate)
-{
- return speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate);
-}
-
-EXPORT void speex_resampler_get_rate(SpeexResamplerState *st, spx_uint32_t *in_rate, spx_uint32_t *out_rate)
-{
- *in_rate = st->in_rate;
- *out_rate = st->out_rate;
-}
-
-EXPORT int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_num, spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
-{
- spx_uint32_t fact;
- spx_uint32_t old_den;
- spx_uint32_t i;
- if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den)
- return RESAMPLER_ERR_SUCCESS;
-
- old_den = st->den_rate;
- st->in_rate = in_rate;
- st->out_rate = out_rate;
- st->num_rate = ratio_num;
- st->den_rate = ratio_den;
- /* FIXME: This is terribly inefficient, but who cares (at least for now)? */
- for (fact=2;fact<=IMIN(st->num_rate, st->den_rate);fact++)
- {
- while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0))
- {
- st->num_rate /= fact;
- st->den_rate /= fact;
- }
- }
-
- if (old_den > 0)
- {
- for (i=0;i<st->nb_channels;i++)
- {
- st->samp_frac_num[i]=st->samp_frac_num[i]*st->den_rate/old_den;
- /* Safety net */
- if (st->samp_frac_num[i] >= st->den_rate)
- st->samp_frac_num[i] = st->den_rate-1;
- }
- }
-
- if (st->initialised)
- update_filter(st);
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT void speex_resampler_get_ratio(SpeexResamplerState *st, spx_uint32_t *ratio_num, spx_uint32_t *ratio_den)
-{
- *ratio_num = st->num_rate;
- *ratio_den = st->den_rate;
-}
-
-EXPORT int speex_resampler_set_quality(SpeexResamplerState *st, int quality)
-{
- if (quality > 10 || quality < 0)
- return RESAMPLER_ERR_INVALID_ARG;
- if (st->quality == quality)
- return RESAMPLER_ERR_SUCCESS;
- st->quality = quality;
- if (st->initialised)
- update_filter(st);
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT void speex_resampler_get_quality(SpeexResamplerState *st, int *quality)
-{
- *quality = st->quality;
-}
-
-EXPORT void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride)
-{
- st->in_stride = stride;
-}
-
-EXPORT void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride)
-{
- *stride = st->in_stride;
-}
-
-EXPORT void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride)
-{
- st->out_stride = stride;
-}
-
-EXPORT void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride)
-{
- *stride = st->out_stride;
-}
-
-EXPORT int speex_resampler_get_input_latency(SpeexResamplerState *st)
-{
- return st->filt_len / 2;
-}
-
-EXPORT int speex_resampler_get_output_latency(SpeexResamplerState *st)
-{
- return ((st->filt_len / 2) * st->den_rate + (st->num_rate >> 1)) / st->num_rate;
-}
-
-EXPORT int speex_resampler_skip_zeros(SpeexResamplerState *st)
-{
- spx_uint32_t i;
- for (i=0;i<st->nb_channels;i++)
- st->last_sample[i] = st->filt_len/2;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int speex_resampler_reset_mem(SpeexResamplerState *st)
-{
- spx_uint32_t i;
- for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
- st->mem[i] = 0;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT const char *speex_resampler_strerror(int err)
-{
- switch (err)
- {
- case RESAMPLER_ERR_SUCCESS:
- return "Success.";
- case RESAMPLER_ERR_ALLOC_FAILED:
- return "Memory allocation failed.";
- case RESAMPLER_ERR_BAD_STATE:
- return "Bad resampler state.";
- case RESAMPLER_ERR_INVALID_ARG:
- return "Invalid argument.";
- case RESAMPLER_ERR_PTR_OVERLAP:
- return "Input and output buffers overlap.";
- default:
- return "Unknown error. Bad error code or strange version mismatch.";
- }
-}
+++ /dev/null
-/* Copyright (C) 2007 Jean-Marc Valin
-
- File: speex_resampler.h
- Resampling code
-
- The design goals of this code are:
- - Very fast algorithm
- - Low memory requirement
- - Good *perceptual* quality (and not best SNR)
-
- 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.
-*/
-
-
-#ifndef SPEEX_RESAMPLER_H
-#define SPEEX_RESAMPLER_H
-
-#ifdef OUTSIDE_SPEEX
-
-/********* WARNING: MENTAL SANITY ENDS HERE *************/
-
-/* If the resampler is defined outside of Speex, we change the symbol names so that
- there won't be any clash if linking with Speex later on. */
-
-/* #define RANDOM_PREFIX your software name here */
-#ifndef RANDOM_PREFIX
-#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes"
-#endif
-
-#define CAT_PREFIX2(a,b) a ## b
-#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b)
-
-#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init)
-#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac)
-#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy)
-#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float)
-#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int)
-#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float)
-#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int)
-#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate)
-#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate)
-#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac)
-#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio)
-#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality)
-#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality)
-#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride)
-#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride)
-#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride)
-#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride)
-#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency)
-#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency)
-#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros)
-#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem)
-#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror)
-
-#define spx_int16_t short
-#define spx_int32_t int
-#define spx_uint16_t unsigned short
-#define spx_uint32_t unsigned int
-
-#else /* OUTSIDE_SPEEX */
-
-#include "speex/speex_types.h"
-
-#endif /* OUTSIDE_SPEEX */
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#define SPEEX_RESAMPLER_QUALITY_MAX 10
-#define SPEEX_RESAMPLER_QUALITY_MIN 0
-#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
-#define SPEEX_RESAMPLER_QUALITY_VOIP 3
-#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
-
-enum {
- RESAMPLER_ERR_SUCCESS = 0,
- RESAMPLER_ERR_ALLOC_FAILED = 1,
- RESAMPLER_ERR_BAD_STATE = 2,
- RESAMPLER_ERR_INVALID_ARG = 3,
- RESAMPLER_ERR_PTR_OVERLAP = 4,
-
- RESAMPLER_ERR_MAX_ERROR
-};
-
-struct SpeexResamplerState_;
-typedef struct SpeexResamplerState_ SpeexResamplerState;
-
-/** Create a new resampler with integer input and output rates.
- * @param nb_channels Number of channels to be processed
- * @param in_rate Input sampling rate (integer number of Hz).
- * @param out_rate Output sampling rate (integer number of Hz).
- * @param quality Resampling quality between 0 and 10, where 0 has poor quality
- * and 10 has very high quality.
- * @return Newly created resampler state
- * @retval NULL Error: not enough memory
- */
-SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate,
- int quality,
- int *err);
-
-/** Create a new resampler with fractional input/output rates. The sampling
- * rate ratio is an arbitrary rational number with both the numerator and
- * denominator being 32-bit integers.
- * @param nb_channels Number of channels to be processed
- * @param ratio_num Numerator of the sampling rate ratio
- * @param ratio_den Denominator of the sampling rate ratio
- * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
- * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
- * @param quality Resampling quality between 0 and 10, where 0 has poor quality
- * and 10 has very high quality.
- * @return Newly created resampler state
- * @retval NULL Error: not enough memory
- */
-SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
- spx_uint32_t ratio_num,
- spx_uint32_t ratio_den,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate,
- int quality,
- int *err);
-
-/** Destroy a resampler state.
- * @param st Resampler state
- */
-void speex_resampler_destroy(SpeexResamplerState *st);
-
-/** Resample a float array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param channel_index Index of the channel to process for the multi-channel
- * base (0 otherwise)
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the
- * number of samples processed
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written
- */
-int speex_resampler_process_float(SpeexResamplerState *st,
- spx_uint32_t channel_index,
- const float *in,
- spx_uint32_t *in_len,
- float *out,
- spx_uint32_t *out_len);
-
-/** Resample an int array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param channel_index Index of the channel to process for the multi-channel
- * base (0 otherwise)
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written
- */
-int speex_resampler_process_int(SpeexResamplerState *st,
- spx_uint32_t channel_index,
- const spx_int16_t *in,
- spx_uint32_t *in_len,
- spx_int16_t *out,
- spx_uint32_t *out_len);
-
-/** Resample an interleaved float array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed. This is all per-channel.
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written.
- * This is all per-channel.
- */
-int speex_resampler_process_interleaved_float(SpeexResamplerState *st,
- const float *in,
- spx_uint32_t *in_len,
- float *out,
- spx_uint32_t *out_len);
-
-/** Resample an interleaved int array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed. This is all per-channel.
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written.
- * This is all per-channel.
- */
-int speex_resampler_process_interleaved_int(SpeexResamplerState *st,
- const spx_int16_t *in,
- spx_uint32_t *in_len,
- spx_int16_t *out,
- spx_uint32_t *out_len);
-
-/** Set (change) the input/output sampling rates (integer value).
- * @param st Resampler state
- * @param in_rate Input sampling rate (integer number of Hz).
- * @param out_rate Output sampling rate (integer number of Hz).
- */
-int speex_resampler_set_rate(SpeexResamplerState *st,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate);
-
-/** Get the current input/output sampling rates (integer value).
- * @param st Resampler state
- * @param in_rate Input sampling rate (integer number of Hz) copied.
- * @param out_rate Output sampling rate (integer number of Hz) copied.
- */
-void speex_resampler_get_rate(SpeexResamplerState *st,
- spx_uint32_t *in_rate,
- spx_uint32_t *out_rate);
-
-/** Set (change) the input/output sampling rates and resampling ratio
- * (fractional values in Hz supported).
- * @param st Resampler state
- * @param ratio_num Numerator of the sampling rate ratio
- * @param ratio_den Denominator of the sampling rate ratio
- * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
- * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
- */
-int speex_resampler_set_rate_frac(SpeexResamplerState *st,
- spx_uint32_t ratio_num,
- spx_uint32_t ratio_den,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate);
-
-/** Get the current resampling ratio. This will be reduced to the least
- * common denominator.
- * @param st Resampler state
- * @param ratio_num Numerator of the sampling rate ratio copied
- * @param ratio_den Denominator of the sampling rate ratio copied
- */
-void speex_resampler_get_ratio(SpeexResamplerState *st,
- spx_uint32_t *ratio_num,
- spx_uint32_t *ratio_den);
-
-/** Set (change) the conversion quality.
- * @param st Resampler state
- * @param quality Resampling quality between 0 and 10, where 0 has poor
- * quality and 10 has very high quality.
- */
-int speex_resampler_set_quality(SpeexResamplerState *st,
- int quality);
-
-/** Get the conversion quality.
- * @param st Resampler state
- * @param quality Resampling quality between 0 and 10, where 0 has poor
- * quality and 10 has very high quality.
- */
-void speex_resampler_get_quality(SpeexResamplerState *st,
- int *quality);
-
-/** Set (change) the input stride.
- * @param st Resampler state
- * @param stride Input stride
- */
-void speex_resampler_set_input_stride(SpeexResamplerState *st,
- spx_uint32_t stride);
-
-/** Get the input stride.
- * @param st Resampler state
- * @param stride Input stride copied
- */
-void speex_resampler_get_input_stride(SpeexResamplerState *st,
- spx_uint32_t *stride);
-
-/** Set (change) the output stride.
- * @param st Resampler state
- * @param stride Output stride
- */
-void speex_resampler_set_output_stride(SpeexResamplerState *st,
- spx_uint32_t stride);
-
-/** Get the output stride.
- * @param st Resampler state copied
- * @param stride Output stride
- */
-void speex_resampler_get_output_stride(SpeexResamplerState *st,
- spx_uint32_t *stride);
-
-/** Get the latency in input samples introduced by the resampler.
- * @param st Resampler state
- */
-int speex_resampler_get_input_latency(SpeexResamplerState *st);
-
-/** Get the latency in output samples introduced by the resampler.
- * @param st Resampler state
- */
-int speex_resampler_get_output_latency(SpeexResamplerState *st);
-
-/** Make sure that the first samples to go out of the resamplers don't have
- * leading zeros. This is only useful before starting to use a newly created
- * resampler. It is recommended to use that when resampling an audio file, as
- * it will generate a file with the same length. For real-time processing,
- * it is probably easier not to use this call (so that the output duration
- * is the same for the first frame).
- * @param st Resampler state
- */
-int speex_resampler_skip_zeros(SpeexResamplerState *st);
-
-/** Reset a resampler so a new (unrelated) stream can be processed.
- * @param st Resampler state
- */
-int speex_resampler_reset_mem(SpeexResamplerState *st);
-
-/** Returns the English meaning for an error code
- * @param err Error code
- * @return English string
- */
-const char *speex_resampler_strerror(int err);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
projects / profile / ivi / pulseaudio.git / commitdiff