forked from osmocom/wireshark
![João Valverde](/assets/img/avatar_default.png)
For a sane plugin build environment. Include config.h as the first header in the .c file instead. Fix by moving required compiler attribute macros to a new "ws_attributes.h" API header. Change-Id: I34f58a927f68c1a0e59686c14d214825149749e1 Reviewed-on: https://code.wireshark.org/review/23400 Petri-Dish: João Valverde <j@v6e.pt> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: João Valverde <j@v6e.pt>
1212 lines
43 KiB
C
1212 lines
43 KiB
C
/* Copyright (C) 2007-2008 Jean-Marc Valin
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Copyright (C) 2008 Thorvald Natvig
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File: resample.c
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Arbitrary resampling code
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Redistribution and use in source and binary forms, with or without
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modification, are permitted provided that the following conditions are
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met:
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1. Redistributions of source code must retain the above copyright notice,
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this list of conditions and the following disclaimer.
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2. Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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3. The name of the author may not be used to endorse or promote products
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derived from this software without specific prior written permission.
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THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
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IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
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OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
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DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
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INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
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(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
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SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
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STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
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ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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POSSIBILITY OF SUCH DAMAGE.
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*/
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/*
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The design goals of this code are:
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- Very fast algorithm
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- SIMD-friendly algorithm
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- Low memory requirement
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- Good *perceptual* quality (and not best SNR)
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Warning: This resampler is relatively new. Although I think I got rid of
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all the major bugs and I don't expect the API to change anymore, there
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may be something I've missed. So use with caution.
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This algorithm is based on this original resampling algorithm:
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Smith, Julius O. Digital Audio Resampling Home Page
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Center for Computer Research in Music and Acoustics (CCRMA),
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Stanford University, 2007.
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Web published at http://www-ccrma.stanford.edu/~jos/resample/.
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There is one main difference, though. This resampler uses cubic
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interpolation instead of linear interpolation in the above paper. This
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makes the table much smaller and makes it possible to compute that table
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on a per-stream basis. In turn, being able to tweak the table for each
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stream makes it possible to both reduce complexity on simple ratios
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(e.g. 2/3), and get rid of the rounding operations in the inner loop.
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The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
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*/
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#include <config.h>
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#define OUTSIDE_SPEEX 1
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#define FLOATING_POINT 1
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#ifdef OUTSIDE_SPEEX
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#include <glib.h>
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static void *speex_alloc (size_t size) {return g_malloc0(size);}
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static void *speex_realloc (void *ptr, size_t size) {return g_realloc(ptr, size);}
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static void speex_free (void *ptr) {g_free(ptr);}
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#include "speex_resampler.h"
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#include "arch.h"
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#else /* OUTSIDE_SPEEX */
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#include "speex/speex_resampler.h"
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#include "arch.h"
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#include "os_support.h"
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#endif /* OUTSIDE_SPEEX */
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#include "stack_alloc.h"
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#include <math.h>
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#include <limits.h>
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#include "ws_attributes.h"
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#ifndef M_PI
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#define M_PI 3.14159265358979323846
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#endif
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#define IMAX(a,b) ((a) > (b) ? (a) : (b))
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#define IMIN(a,b) ((a) < (b) ? (a) : (b))
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#ifndef NULL
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#define NULL 0
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#endif
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#ifdef _USE_SSE
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#include "resample_sse.h"
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#endif
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#ifdef _USE_NEON
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#include "resample_neon.h"
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#endif
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/* Numer of elements to allocate on the stack */
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#ifdef VAR_ARRAYS
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#define FIXED_STACK_ALLOC 8192
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#else
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#define FIXED_STACK_ALLOC 1024
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#endif
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typedef int (*resampler_basic_func)(SpeexResamplerState *, spx_uint32_t , const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
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struct SpeexResamplerState_ {
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spx_uint32_t in_rate;
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spx_uint32_t out_rate;
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spx_uint32_t num_rate;
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spx_uint32_t den_rate;
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int quality;
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spx_uint32_t nb_channels;
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spx_uint32_t filt_len;
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spx_uint32_t mem_alloc_size;
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spx_uint32_t buffer_size;
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int int_advance;
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int frac_advance;
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float cutoff;
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spx_uint32_t oversample;
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int initialised;
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int started;
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/* These are per-channel */
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spx_int32_t *last_sample;
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spx_uint32_t *samp_frac_num;
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spx_uint32_t *magic_samples;
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spx_word16_t *mem;
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spx_word16_t *sinc_table;
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spx_uint32_t sinc_table_length;
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resampler_basic_func resampler_ptr;
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int in_stride;
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int out_stride;
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} ;
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static const double kaiser12_table[68] = {
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0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
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0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
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0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
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0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
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0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
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0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
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0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
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0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
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0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
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0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
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0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
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0.00001000, 0.00000000};
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/*
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static const double kaiser12_table[36] = {
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0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
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0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
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0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
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0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
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0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
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0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
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*/
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static const double kaiser10_table[36] = {
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0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
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0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
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0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
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0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
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0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
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0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000};
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static const double kaiser8_table[36] = {
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0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
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0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
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0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
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0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
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0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
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0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000};
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static const double kaiser6_table[36] = {
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0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
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0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
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0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
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0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
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0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
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0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000};
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struct FuncDef {
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const double *table;
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int oversample;
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};
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static const struct FuncDef _KAISER12 = {kaiser12_table, 64};
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#define KAISER12 (&_KAISER12)
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/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
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#define KAISER12 (&_KAISER12)*/
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static const struct FuncDef _KAISER10 = {kaiser10_table, 32};
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#define KAISER10 (&_KAISER10)
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static const struct FuncDef _KAISER8 = {kaiser8_table, 32};
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#define KAISER8 (&_KAISER8)
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static const struct FuncDef _KAISER6 = {kaiser6_table, 32};
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#define KAISER6 (&_KAISER6)
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struct QualityMapping {
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int base_length;
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int oversample;
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float downsample_bandwidth;
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float upsample_bandwidth;
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const struct FuncDef *window_func;
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};
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/* This table maps conversion quality to internal parameters. There are two
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reasons that explain why the up-sampling bandwidth is larger than the
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down-sampling bandwidth:
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1) When up-sampling, we can assume that the spectrum is already attenuated
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close to the Nyquist rate (from an A/D or a previous resampling filter)
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2) Any aliasing that occurs very close to the Nyquist rate will be masked
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by the sinusoids/noise just below the Nyquist rate (guaranteed only for
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up-sampling).
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*/
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static const struct QualityMapping quality_map[11] = {
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{ 8, 4, 0.830f, 0.860f, KAISER6 }, /* Q0 */
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{ 16, 4, 0.850f, 0.880f, KAISER6 }, /* Q1 */
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{ 32, 4, 0.882f, 0.910f, KAISER6 }, /* Q2 */ /* 82.3% cutoff ( ~60 dB stop) 6 */
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{ 48, 8, 0.895f, 0.917f, KAISER8 }, /* Q3 */ /* 84.9% cutoff ( ~80 dB stop) 8 */
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{ 64, 8, 0.921f, 0.940f, KAISER8 }, /* Q4 */ /* 88.7% cutoff ( ~80 dB stop) 8 */
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{ 80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 */ /* 89.1% cutoff (~100 dB stop) 10 */
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{ 96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 */ /* 91.5% cutoff (~100 dB stop) 10 */
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{128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */ /* 93.1% cutoff (~100 dB stop) 10 */
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{160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */ /* 94.5% cutoff (~100 dB stop) 10 */
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{192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 */ /* 95.5% cutoff (~100 dB stop) 10 */
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{256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
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};
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/*8,24,40,56,80,104,128,160,200,256,320*/
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static double compute_func(float x, const struct FuncDef *func)
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{
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float y, frac;
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double interp[4];
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int ind;
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y = x*func->oversample;
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ind = (int)floor(y);
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frac = (y-ind);
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/* CSE with handle the repeated powers */
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interp[3] = -0.1666666667*frac + 0.1666666667*(frac*frac*frac);
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interp[2] = frac + 0.5*(frac*frac) - 0.5*(frac*frac*frac);
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/*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
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interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac);
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/* Just to make sure we don't have rounding problems */
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interp[1] = 1.f-interp[3]-interp[2]-interp[0];
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/*sum = frac*accum[1] + (1-frac)*accum[2];*/
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return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3];
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}
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#if 0
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#include <stdio.h>
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int main(int argc, char **argv)
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{
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int i;
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for (i=0;i<256;i++)
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{
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printf ("%f\n", compute_func(i/256., KAISER12));
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}
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return 0;
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}
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#endif
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DIAG_OFF_CLANG(self-assign) /* SATURATE32PSHR */
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#ifdef FIXED_POINT
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/* The slow way of computing a sinc for the table. Should improve that some day */
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static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func)
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{
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/*fprintf (stderr, "%f ", x);*/
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float xx = x * cutoff;
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if (fabs(x)<1e-6f)
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return WORD2INT(32768.*cutoff);
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else if (fabs(x) > .5f*N)
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return 0;
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/*FIXME: Can it really be any slower than this? */
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return WORD2INT(32768.*cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func));
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}
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#else
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/* The slow way of computing a sinc for the table. Should improve that some day */
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static spx_word16_t sinc(float cutoff, float x, int N, const struct FuncDef *window_func)
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{
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/*fprintf (stderr, "%f ", x);*/
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float xx = x * cutoff;
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if (fabs(x)<1e-6)
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return cutoff;
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else if (fabs(x) > .5*N)
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return 0;
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/*FIXME: Can it really be any slower than this? */
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return (spx_word16_t)(cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func((float)fabs(2.*x/N), window_func));
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}
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#endif
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#ifdef FIXED_POINT
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static void cubic_coef(spx_word16_t x, spx_word16_t interp[4])
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{
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/* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
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but I know it's MMSE-optimal on a sinc */
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spx_word16_t x2, x3;
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x2 = MULT16_16_P15(x, x);
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x3 = MULT16_16_P15(x, x2);
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interp[0] = PSHR32(MULT16_16(QCONST16(-0.16667f, 15),x) + MULT16_16(QCONST16(0.16667f, 15),x3),15);
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interp[1] = EXTRACT16(EXTEND32(x) + SHR32(SUB32(EXTEND32(x2),EXTEND32(x3)),1));
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interp[3] = PSHR32(MULT16_16(QCONST16(-0.33333f, 15),x) + MULT16_16(QCONST16(.5f,15),x2) - MULT16_16(QCONST16(0.16667f, 15),x3),15);
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/* Just to make sure we don't have rounding problems */
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interp[2] = Q15_ONE-interp[0]-interp[1]-interp[3];
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if (interp[2]<32767)
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interp[2]+=1;
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}
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#else
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static void cubic_coef(spx_word16_t frac, spx_word16_t interp[4])
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{
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/* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
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but I know it's MMSE-optimal on a sinc */
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interp[0] = -0.16667f*frac + 0.16667f*frac*frac*frac;
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interp[1] = frac + 0.5f*frac*frac - 0.5f*frac*frac*frac;
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/*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
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interp[3] = -0.33333f*frac + 0.5f*frac*frac - 0.16667f*frac*frac*frac;
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/* Just to make sure we don't have rounding problems */
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interp[2] = (spx_word16_t)(1.-interp[0]-interp[1]-interp[3]);
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}
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#endif
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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)
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{
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const int N = st->filt_len;
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int out_sample = 0;
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int last_sample = st->last_sample[channel_index];
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spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
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const spx_word16_t *sinc_table = st->sinc_table;
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const int out_stride = st->out_stride;
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const int int_advance = st->int_advance;
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const int frac_advance = st->frac_advance;
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const spx_uint32_t den_rate = st->den_rate;
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spx_word32_t sum;
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while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
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{
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const spx_word16_t *sinct = & sinc_table[samp_frac_num*N];
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const spx_word16_t *iptr = & in[last_sample];
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#ifndef OVERRIDE_INNER_PRODUCT_SINGLE
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int j;
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sum = 0;
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for(j=0;j<N;j++) sum += MULT16_16(sinct[j], iptr[j]);
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/* This code is slower on most DSPs which have only 2 accumulators.
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Plus this this forces truncation to 32 bits and you lose the HW guard bits.
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I think we can trust the compiler and let it vectorize and/or unroll itself.
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spx_word32_t accum[4] = {0,0,0,0};
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for(j=0;j<N;j+=4) {
|
|
accum[0] += MULT16_16(sinct[j], iptr[j]);
|
|
accum[1] += MULT16_16(sinct[j+1], iptr[j+1]);
|
|
accum[2] += MULT16_16(sinct[j+2], iptr[j+2]);
|
|
accum[3] += MULT16_16(sinct[j+3], iptr[j+3]);
|
|
}
|
|
sum = accum[0] + accum[1] + accum[2] + accum[3];
|
|
*/
|
|
sum = SATURATE32PSHR(sum, 15, 32767);
|
|
#else
|
|
sum = inner_product_single(sinct, iptr, N);
|
|
#endif
|
|
|
|
out[out_stride * out_sample++] = sum;
|
|
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;
|
|
|
|
while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
|
|
{
|
|
const spx_word16_t *sinct = & sinc_table[samp_frac_num*N];
|
|
const spx_word16_t *iptr = & in[last_sample];
|
|
|
|
#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
|
|
int j;
|
|
double accum[4] = {0,0,0,0};
|
|
|
|
for(j=0;j<N;j+=4) {
|
|
accum[0] += sinct[j]*iptr[j];
|
|
accum[1] += sinct[j+1]*iptr[j+1];
|
|
accum[2] += sinct[j+2]*iptr[j+2];
|
|
accum[3] += sinct[j+3]*iptr[j+3];
|
|
}
|
|
sum = accum[0] + accum[1] + accum[2] + accum[3];
|
|
#else
|
|
sum = inner_product_double(sinct, 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;
|
|
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
|
|
int j;
|
|
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],SHR32(accum[0], 1)) + MULT16_32_Q15(interp[1],SHR32(accum[1], 1)) + MULT16_32_Q15(interp[2],SHR32(accum[2], 1)) + MULT16_32_Q15(interp[3],SHR32(accum[3], 1));
|
|
sum = SATURATE32PSHR(sum, 15, 32767);
|
|
#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++] = sum;
|
|
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;
|
|
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
|
|
int j;
|
|
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
|
|
|
|
/* This resampler is used to produce zero output in situations where memory
|
|
for the filter could not be allocated. The expected numbers of input and
|
|
output samples are still processed so that callers failing to check error
|
|
codes are not surprised, possibly getting into infinite loops. */
|
|
static int resampler_basic_zero(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in _U_, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_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;
|
|
|
|
while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
|
|
{
|
|
out[out_stride * out_sample++] = 0;
|
|
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;
|
|
}
|
|
|
|
static int update_filter(SpeexResamplerState *st)
|
|
{
|
|
spx_uint32_t old_length = st->filt_len;
|
|
spx_uint32_t old_alloc_size = st->mem_alloc_size;
|
|
int use_direct;
|
|
spx_uint32_t min_sinc_table_length;
|
|
spx_uint32_t min_alloc_size;
|
|
|
|
st->int_advance = st->num_rate/st->den_rate;
|
|
st->frac_advance = st->num_rate%st->den_rate;
|
|
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 up to make sure we have a multiple of 8 for SSE */
|
|
st->filt_len = ((st->filt_len-1)&(~0x7))+8;
|
|
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 */
|
|
#ifdef RESAMPLE_FULL_SINC_TABLE
|
|
use_direct = 1;
|
|
if (INT_MAX/sizeof(spx_word16_t)/st->den_rate < st->filt_len)
|
|
goto fail;
|
|
#else
|
|
use_direct = st->filt_len*st->den_rate <= st->filt_len*st->oversample+8
|
|
&& INT_MAX/(spx_uint32_t)sizeof(spx_word16_t)/st->den_rate >= st->filt_len;
|
|
#endif
|
|
if (use_direct)
|
|
{
|
|
min_sinc_table_length = st->filt_len*st->den_rate;
|
|
} else {
|
|
if ((INT_MAX/(spx_uint32_t)sizeof(spx_word16_t)-8)/st->oversample < st->filt_len)
|
|
goto fail;
|
|
|
|
min_sinc_table_length = st->filt_len*st->oversample+8;
|
|
}
|
|
if (st->sinc_table_length < min_sinc_table_length)
|
|
{
|
|
spx_word16_t *sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,min_sinc_table_length*sizeof(spx_word16_t));
|
|
if (!sinc_table)
|
|
goto fail;
|
|
|
|
st->sinc_table = sinc_table;
|
|
st->sinc_table_length = min_sinc_table_length;
|
|
}
|
|
if (use_direct)
|
|
{
|
|
spx_uint32_t i;
|
|
for (i=0;i<st->den_rate;i++)
|
|
{
|
|
spx_uint32_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;
|
|
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);*/
|
|
}
|
|
|
|
/* 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. */
|
|
|
|
/* Adding buffer_size to filt_len won't overflow here because filt_len
|
|
could be multiplied by sizeof(spx_word16_t) above. */
|
|
min_alloc_size = st->filt_len-1 + st->buffer_size;
|
|
if (min_alloc_size > st->mem_alloc_size)
|
|
{
|
|
spx_word16_t *mem;
|
|
if (INT_MAX/(spx_uint32_t)sizeof(spx_word16_t)/st->nb_channels < min_alloc_size)
|
|
goto fail;
|
|
else if (!(mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*min_alloc_size * sizeof(*mem))))
|
|
goto fail;
|
|
|
|
st->mem = mem;
|
|
st->mem_alloc_size = min_alloc_size;
|
|
}
|
|
if (!st->started)
|
|
{
|
|
spx_uint32_t i;
|
|
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_uint32_t i;
|
|
/* Increase the filter length */
|
|
/*speex_warning("increase filter size");*/
|
|
for (i=st->nb_channels;i--;)
|
|
{
|
|
spx_uint32_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-1+st->magic_samples[i];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;
|
|
}
|
|
}
|
|
return RESAMPLER_ERR_SUCCESS;
|
|
|
|
fail:
|
|
st->resampler_ptr = resampler_basic_zero;
|
|
/* st->mem may still contain consumed input samples for the filter.
|
|
Restore filt_len so that filt_len - 1 still points to the position after
|
|
the last of these samples. */
|
|
st->filt_len = old_length;
|
|
return RESAMPLER_ERR_ALLOC_FAILED;
|
|
}
|
|
|
|
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;
|
|
int filter_err;
|
|
|
|
if (nb_channels == 0 || ratio_num == 0 || ratio_den == 0 || 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;
|
|
|
|
st->buffer_size = 160;
|
|
|
|
/* Per channel data */
|
|
st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t));
|
|
st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t));
|
|
st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t));
|
|
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);
|
|
|
|
filter_err = update_filter(st);
|
|
if (filter_err == RESAMPLER_ERR_SUCCESS)
|
|
{
|
|
st->initialised = 1;
|
|
} else {
|
|
speex_resampler_destroy(st);
|
|
st = NULL;
|
|
}
|
|
if (err)
|
|
*err = filter_err;
|
|
|
|
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
|
|
{
|
|
spx_uint32_t 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 st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : 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
|
|
{
|
|
spx_uint32_t 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 st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : 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_out_len = *out_len;
|
|
spx_uint32_t bak_in_len = *in_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_out_len;
|
|
*in_len = bak_in_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 st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : 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_out_len = *out_len;
|
|
spx_uint32_t bak_in_len = *in_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_out_len;
|
|
*in_len = bak_in_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 st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : 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 (ratio_num == 0 || ratio_den == 0)
|
|
return RESAMPLER_ERR_INVALID_ARG;
|
|
|
|
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)
|
|
return 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)
|
|
return 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)
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{
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spx_uint32_t i;
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for (i=0;i<st->nb_channels;i++)
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{
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st->last_sample[i] = 0;
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st->magic_samples[i] = 0;
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st->samp_frac_num[i] = 0;
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}
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for (i=0;i<st->nb_channels*(st->filt_len-1);i++)
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st->mem[i] = 0;
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return RESAMPLER_ERR_SUCCESS;
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}
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|
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EXPORT const char *speex_resampler_strerror(int err)
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|
{
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|
switch (err)
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|
{
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|
case RESAMPLER_ERR_SUCCESS:
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|
return "Success.";
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|
case RESAMPLER_ERR_ALLOC_FAILED:
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|
return "Memory allocation failed.";
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|
case RESAMPLER_ERR_BAD_STATE:
|
|
return "Bad resampler state.";
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|
case RESAMPLER_ERR_INVALID_ARG:
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|
return "Invalid argument.";
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|
case RESAMPLER_ERR_PTR_OVERLAP:
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|
return "Input and output buffers overlap.";
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|
default:
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|
return "Unknown error. Bad error code or strange version mismatch.";
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|
}
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}
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