* #27232: jni: added pjproject checkout as regular git content
We will remove it once the next release of pjsip (with Android support)
comes out and is merged into SFLphone.
diff --git a/jni/pjproject-android/third_party/speex/libspeex/resample.c b/jni/pjproject-android/third_party/speex/libspeex/resample.c
new file mode 100644
index 0000000..bebd1a8
--- /dev/null
+++ b/jni/pjproject-android/third_party/speex/libspeex/resample.c
@@ -0,0 +1,1131 @@
+/* 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.";
+ }
+}