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Original commit message from CVS: * gst/speexresample/README: * gst/speexresample/arch.h: * gst/speexresample/resample.c: (resampler_basic_direct_single), (resampler_basic_direct_double), (resampler_basic_interpolate_single), (resampler_basic_interpolate_double), (speex_resampler_process_native), (speex_resampler_process_float), (speex_resampler_process_int), (speex_resampler_process_interleaved_float), (speex_resampler_process_interleaved_int), (speex_resampler_get_input_latency), (speex_resampler_get_output_latency): * gst/speexresample/speex_resampler.h: Update speex resampler to latest SVN. We're now down to only the changes noted in README again. * gst/speexresample/speex_resampler_wrapper.h: * gst/speexresample/gstspeexresample.c: (gst_speex_resample_push_drain), (gst_speex_resample_query): Adjust to API changes.
1372 lines
43 KiB
C
1372 lines
43 KiB
C
/* Copyright (C) 2007 Jean-Marc Valin
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#ifdef OUTSIDE_SPEEX
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#include <stdlib.h>
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#include <glib.h>
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static inline void *
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speex_alloc (int size)
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{
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return g_malloc0 (size);
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}
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static inline void *
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speex_realloc (void *ptr, int size)
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{
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return g_realloc (ptr, size);
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}
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static inline void
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speex_free (void *ptr)
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{
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g_free (ptr);
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}
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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 <math.h>
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#ifndef M_PI
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#define M_PI 3.14159263
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#endif
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#ifdef FIXED_POINT
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#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
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#else
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#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))
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#endif
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/*#define float double*/
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#define FILTER_SIZE 64
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#define OVERSAMPLE 8
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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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typedef int (*resampler_basic_func) (SpeexResamplerState *, spx_uint32_t,
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const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
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struct SpeexResamplerState_
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{
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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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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 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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/*
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static 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 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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};
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static 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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};
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static 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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};
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struct FuncDef
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{
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double *table;
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int oversample;
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};
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static 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 struct FuncDef _KAISER10 = { kaiser10_table, 32 };
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#define KAISER10 (&_KAISER10)
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static struct FuncDef _KAISER8 = { kaiser8_table, 32 };
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#define KAISER8 (&_KAISER8)
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static struct FuncDef _KAISER6 = { kaiser6_table, 32 };
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#define KAISER6 (&_KAISER6)
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struct QualityMapping
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{
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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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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
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compute_func (float x, 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] =
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-0.3333333333 * frac + 0.5 * (frac * frac) -
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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] +
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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
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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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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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#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
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sinc (float cutoff, float x, int N, 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) *
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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
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sinc (float cutoff, float x, int N, 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 cutoff * sin (M_PI * xx) / (M_PI * xx) * compute_func (fabs (2. * x /
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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
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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] =
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PSHR32 (MULT16_16 (QCONST16 (-0.16667f, 15),
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x) + MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
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interp[1] =
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EXTRACT16 (EXTEND32 (x) + SHR32 (SUB32 (EXTEND32 (x2), EXTEND32 (x3)),
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1));
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interp[3] =
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PSHR32 (MULT16_16 (QCONST16 (-0.33333f, 15),
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x) + MULT16_16 (QCONST16 (.5f, 15),
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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
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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] =
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-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] = 1. - interp[0] - interp[1] - interp[3];
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}
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#endif
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static int
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resampler_basic_direct_single (SpeexResamplerState * st,
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spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
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spx_word16_t * out, spx_uint32_t * out_len)
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{
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int N = st->filt_len;
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int out_sample = 0;
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spx_word16_t *mem;
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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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|
|
mem = st->mem + channel_index * st->mem_alloc_size;
|
|
while (!(last_sample >= (spx_int32_t) * in_len
|
|
|| out_sample >= (spx_int32_t) * out_len)) {
|
|
int j;
|
|
spx_word32_t sum = 0;
|
|
|
|
/* We already have all the filter coefficients pre-computed in the table */
|
|
const spx_word16_t *ptr;
|
|
|
|
/* Do the memory part */
|
|
for (j = 0; last_sample - N + 1 + j < 0; j++) {
|
|
sum +=
|
|
MULT16_16 (mem[last_sample + j],
|
|
st->sinc_table[samp_frac_num * st->filt_len + j]);
|
|
}
|
|
|
|
/* Do the new part */
|
|
if (in != NULL) {
|
|
ptr = in + st->in_stride * (last_sample - N + 1 + j);
|
|
for (; j < N; j++) {
|
|
sum +=
|
|
MULT16_16 (*ptr, st->sinc_table[samp_frac_num * st->filt_len + j]);
|
|
ptr += st->in_stride;
|
|
}
|
|
}
|
|
|
|
*out = PSHR32 (sum, 15);
|
|
out += st->out_stride;
|
|
out_sample++;
|
|
last_sample += st->int_advance;
|
|
samp_frac_num += st->frac_advance;
|
|
if (samp_frac_num >= st->den_rate) {
|
|
samp_frac_num -= st->den_rate;
|
|
last_sample++;
|
|
}
|
|
}
|
|
st->last_sample[channel_index] = last_sample;
|
|
st->samp_frac_num[channel_index] = samp_frac_num;
|
|
return out_sample;
|
|
}
|
|
|
|
#ifdef FIXED_POINT
|
|
#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)
|
|
{
|
|
int N = st->filt_len;
|
|
int out_sample = 0;
|
|
spx_word16_t *mem;
|
|
int last_sample = st->last_sample[channel_index];
|
|
spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
|
|
|
|
mem = st->mem + channel_index * st->mem_alloc_size;
|
|
while (!(last_sample >= (spx_int32_t) * in_len
|
|
|| out_sample >= (spx_int32_t) * out_len)) {
|
|
int j;
|
|
double sum = 0;
|
|
|
|
/* We already have all the filter coefficients pre-computed in the table */
|
|
const spx_word16_t *ptr;
|
|
|
|
/* Do the memory part */
|
|
for (j = 0; last_sample - N + 1 + j < 0; j++) {
|
|
sum +=
|
|
MULT16_16 (mem[last_sample + j],
|
|
(double) st->sinc_table[samp_frac_num * st->filt_len + j]);
|
|
}
|
|
|
|
/* Do the new part */
|
|
if (in != NULL) {
|
|
ptr = in + st->in_stride * (last_sample - N + 1 + j);
|
|
for (; j < N; j++) {
|
|
sum +=
|
|
MULT16_16 (*ptr,
|
|
(double) st->sinc_table[samp_frac_num * st->filt_len + j]);
|
|
ptr += st->in_stride;
|
|
}
|
|
}
|
|
|
|
*out = sum;
|
|
out += st->out_stride;
|
|
out_sample++;
|
|
last_sample += st->int_advance;
|
|
samp_frac_num += st->frac_advance;
|
|
if (samp_frac_num >= st->den_rate) {
|
|
samp_frac_num -= st->den_rate;
|
|
last_sample++;
|
|
}
|
|
}
|
|
st->last_sample[channel_index] = last_sample;
|
|
st->samp_frac_num[channel_index] = samp_frac_num;
|
|
return out_sample;
|
|
}
|
|
#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)
|
|
{
|
|
int N = st->filt_len;
|
|
int out_sample = 0;
|
|
spx_word16_t *mem;
|
|
int last_sample = st->last_sample[channel_index];
|
|
spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
|
|
|
|
mem = st->mem + channel_index * st->mem_alloc_size;
|
|
while (!(last_sample >= (spx_int32_t) * in_len
|
|
|| out_sample >= (spx_int32_t) * out_len)) {
|
|
int j;
|
|
spx_word32_t sum = 0;
|
|
|
|
/* We need to interpolate the sinc filter */
|
|
spx_word32_t accum[4] = { 0.f, 0.f, 0.f, 0.f };
|
|
spx_word16_t interp[4];
|
|
const spx_word16_t *ptr;
|
|
int offset;
|
|
spx_word16_t frac;
|
|
|
|
offset = samp_frac_num * st->oversample / st->den_rate;
|
|
#ifdef FIXED_POINT
|
|
frac =
|
|
PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15),
|
|
st->den_rate);
|
|
#else
|
|
frac =
|
|
((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
|
|
st->den_rate;
|
|
#endif
|
|
/* This code is written like this to make it easy to optimise with SIMD.
|
|
For most DSPs, it would be best to split the loops in two because most DSPs
|
|
have only two accumulators */
|
|
for (j = 0; last_sample - N + 1 + j < 0; j++) {
|
|
spx_word16_t curr_mem = mem[last_sample + j];
|
|
|
|
accum[0] +=
|
|
MULT16_16 (curr_mem,
|
|
st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
|
|
accum[1] +=
|
|
MULT16_16 (curr_mem,
|
|
st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
|
|
accum[2] +=
|
|
MULT16_16 (curr_mem,
|
|
st->sinc_table[4 + (j + 1) * st->oversample - offset]);
|
|
accum[3] +=
|
|
MULT16_16 (curr_mem,
|
|
st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
|
|
}
|
|
|
|
if (in != NULL) {
|
|
ptr = in + st->in_stride * (last_sample - N + 1 + j);
|
|
/* Do the new part */
|
|
for (; j < N; j++) {
|
|
spx_word16_t curr_in = *ptr;
|
|
|
|
ptr += st->in_stride;
|
|
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]);
|
|
|
|
*out = PSHR32 (sum, 15);
|
|
out += st->out_stride;
|
|
out_sample++;
|
|
last_sample += st->int_advance;
|
|
samp_frac_num += st->frac_advance;
|
|
if (samp_frac_num >= st->den_rate) {
|
|
samp_frac_num -= st->den_rate;
|
|
last_sample++;
|
|
}
|
|
}
|
|
st->last_sample[channel_index] = last_sample;
|
|
st->samp_frac_num[channel_index] = samp_frac_num;
|
|
return out_sample;
|
|
}
|
|
|
|
#ifdef FIXED_POINT
|
|
#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)
|
|
{
|
|
int N = st->filt_len;
|
|
int out_sample = 0;
|
|
spx_word16_t *mem;
|
|
int last_sample = st->last_sample[channel_index];
|
|
spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
|
|
|
|
mem = st->mem + channel_index * st->mem_alloc_size;
|
|
while (!(last_sample >= (spx_int32_t) * in_len
|
|
|| out_sample >= (spx_int32_t) * out_len)) {
|
|
int j;
|
|
spx_word32_t sum = 0;
|
|
|
|
/* We need to interpolate the sinc filter */
|
|
double accum[4] = { 0.f, 0.f, 0.f, 0.f };
|
|
float interp[4];
|
|
const spx_word16_t *ptr;
|
|
float alpha = ((float) samp_frac_num) / st->den_rate;
|
|
int offset = samp_frac_num * st->oversample / st->den_rate;
|
|
float frac = alpha * st->oversample - offset;
|
|
|
|
/* This code is written like this to make it easy to optimise with SIMD.
|
|
For most DSPs, it would be best to split the loops in two because most DSPs
|
|
have only two accumulators */
|
|
for (j = 0; last_sample - N + 1 + j < 0; j++) {
|
|
double curr_mem = mem[last_sample + j];
|
|
|
|
accum[0] +=
|
|
MULT16_16 (curr_mem,
|
|
st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
|
|
accum[1] +=
|
|
MULT16_16 (curr_mem,
|
|
st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
|
|
accum[2] +=
|
|
MULT16_16 (curr_mem,
|
|
st->sinc_table[4 + (j + 1) * st->oversample - offset]);
|
|
accum[3] +=
|
|
MULT16_16 (curr_mem,
|
|
st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
|
|
}
|
|
if (in != NULL) {
|
|
ptr = in + st->in_stride * (last_sample - N + 1 + j);
|
|
/* Do the new part */
|
|
for (; j < N; j++) {
|
|
double curr_in = *ptr;
|
|
|
|
ptr += st->in_stride;
|
|
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 =
|
|
interp[0] * accum[0] + interp[1] * accum[1] + interp[2] * accum[2] +
|
|
interp[3] * accum[3];
|
|
|
|
*out = PSHR32 (sum, 15);
|
|
out += st->out_stride;
|
|
out_sample++;
|
|
last_sample += st->int_advance;
|
|
samp_frac_num += st->frac_advance;
|
|
if (samp_frac_num >= st->den_rate) {
|
|
samp_frac_num -= st->den_rate;
|
|
last_sample++;
|
|
}
|
|
}
|
|
st->last_sample[channel_index] = last_sample;
|
|
st->samp_frac_num[channel_index] = samp_frac_num;
|
|
return out_sample;
|
|
}
|
|
#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 =
|
|
(spx_word16_t *) speex_alloc (st->nb_channels * (st->filt_len -
|
|
1) * sizeof (spx_word16_t));
|
|
for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
|
|
st->mem[i] = 0;
|
|
st->mem_alloc_size = st->filt_len - 1;
|
|
/*speex_warning("init filter"); */
|
|
} else if (!st->started) {
|
|
spx_uint32_t i;
|
|
|
|
st->mem =
|
|
(spx_word16_t *) speex_realloc (st->mem,
|
|
st->nb_channels * (st->filt_len - 1) * sizeof (spx_word16_t));
|
|
for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
|
|
st->mem[i] = 0;
|
|
st->mem_alloc_size = st->filt_len - 1;
|
|
/*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->mem_alloc_size) {
|
|
st->mem =
|
|
(spx_word16_t *) speex_realloc (st->mem,
|
|
st->nb_channels * (st->filt_len - 1) * sizeof (spx_word16_t));
|
|
st->mem_alloc_size = st->filt_len - 1;
|
|
}
|
|
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;
|
|
}
|
|
}
|
|
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
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;
|
|
|
|
/* 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;
|
|
}
|
|
|
|
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, const spx_word16_t * in, spx_uint32_t * in_len,
|
|
spx_word16_t * out, spx_uint32_t * out_len)
|
|
{
|
|
int j = 0;
|
|
int N = st->filt_len;
|
|
int out_sample = 0;
|
|
spx_word16_t *mem;
|
|
spx_uint32_t tmp_out_len = 0;
|
|
|
|
mem = st->mem + channel_index * st->mem_alloc_size;
|
|
st->started = 1;
|
|
|
|
/* Handle the case where we have samples left from a reduction in filter length */
|
|
if (st->magic_samples[channel_index]) {
|
|
int istride_save;
|
|
spx_uint32_t tmp_in_len;
|
|
spx_uint32_t tmp_magic;
|
|
|
|
istride_save = st->in_stride;
|
|
tmp_in_len = st->magic_samples[channel_index];
|
|
tmp_out_len = *out_len;
|
|
/* magic_samples needs to be set to zero to avoid infinite recursion */
|
|
tmp_magic = st->magic_samples[channel_index];
|
|
st->magic_samples[channel_index] = 0;
|
|
st->in_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, mem + N - 1, &tmp_in_len,
|
|
out, &tmp_out_len);
|
|
st->in_stride = istride_save;
|
|
/*speex_warning_int("extra samples:", tmp_out_len); */
|
|
/* If we couldn't process all "magic" input samples, save the rest for next time */
|
|
if (tmp_in_len < tmp_magic) {
|
|
spx_uint32_t i;
|
|
|
|
st->magic_samples[channel_index] = tmp_magic - tmp_in_len;
|
|
for (i = 0; i < st->magic_samples[channel_index]; i++)
|
|
mem[N - 1 + i] = mem[N - 1 + i + tmp_in_len];
|
|
}
|
|
out += tmp_out_len * st->out_stride;
|
|
*out_len -= tmp_out_len;
|
|
}
|
|
|
|
/* Call the right resampler through the function ptr */
|
|
out_sample = st->resampler_ptr (st, channel_index, in, 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 + tmp_out_len;
|
|
st->last_sample[channel_index] -= *in_len;
|
|
|
|
for (j = 0; j < N - 1 - (spx_int32_t) * in_len; j++)
|
|
mem[j] = mem[j + *in_len];
|
|
if (in != NULL) {
|
|
for (; j < N - 1; j++)
|
|
mem[j] = in[st->in_stride * (j + *in_len - N + 1)];
|
|
} else {
|
|
for (; j < N - 1; j++)
|
|
mem[j] = 0;
|
|
}
|
|
return RESAMPLER_ERR_SUCCESS;
|
|
}
|
|
|
|
#define FIXED_STACK_ALLOC 1024
|
|
|
|
#ifdef FIXED_POINT
|
|
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)
|
|
{
|
|
spx_uint32_t i;
|
|
int istride_save, ostride_save;
|
|
|
|
#ifdef VAR_ARRAYS
|
|
spx_word16_t x[*in_len];
|
|
spx_word16_t y[*out_len];
|
|
|
|
/*VARDECL(spx_word16_t *x);
|
|
VARDECL(spx_word16_t *y);
|
|
ALLOC(x, *in_len, spx_word16_t);
|
|
ALLOC(y, *out_len, spx_word16_t); */
|
|
istride_save = st->in_stride;
|
|
ostride_save = st->out_stride;
|
|
if (in != NULL) {
|
|
for (i = 0; i < *in_len; i++)
|
|
x[i] = WORD2INT (in[i * st->in_stride]);
|
|
st->in_stride = st->out_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, x, in_len, y, out_len);
|
|
} else {
|
|
st->in_stride = st->out_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, NULL, in_len, y,
|
|
out_len);
|
|
}
|
|
st->in_stride = istride_save;
|
|
st->out_stride = ostride_save;
|
|
for (i = 0; i < *out_len; i++)
|
|
out[i * st->out_stride] = y[i];
|
|
#else
|
|
spx_word16_t x[FIXED_STACK_ALLOC];
|
|
spx_word16_t y[FIXED_STACK_ALLOC];
|
|
spx_uint32_t ilen = *in_len, olen = *out_len;
|
|
|
|
istride_save = st->in_stride;
|
|
ostride_save = st->out_stride;
|
|
while (ilen && olen) {
|
|
spx_uint32_t ichunk, ochunk;
|
|
|
|
ichunk = ilen;
|
|
ochunk = olen;
|
|
if (ichunk > FIXED_STACK_ALLOC)
|
|
ichunk = FIXED_STACK_ALLOC;
|
|
if (ochunk > FIXED_STACK_ALLOC)
|
|
ochunk = FIXED_STACK_ALLOC;
|
|
if (in != NULL) {
|
|
for (i = 0; i < ichunk; i++)
|
|
x[i] = WORD2INT (in[i * st->in_stride]);
|
|
st->in_stride = st->out_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, x, &ichunk, y,
|
|
&ochunk);
|
|
} else {
|
|
st->in_stride = st->out_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, NULL, &ichunk, y,
|
|
&ochunk);
|
|
}
|
|
st->in_stride = istride_save;
|
|
st->out_stride = ostride_save;
|
|
for (i = 0; i < ochunk; i++)
|
|
out[i * st->out_stride] = y[i];
|
|
out += ochunk;
|
|
in += ichunk;
|
|
ilen -= ichunk;
|
|
olen -= ochunk;
|
|
}
|
|
*in_len -= ilen;
|
|
*out_len -= olen;
|
|
#endif
|
|
return RESAMPLER_ERR_SUCCESS;
|
|
}
|
|
|
|
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)
|
|
{
|
|
return speex_resampler_process_native (st, channel_index, in, in_len, out,
|
|
out_len);
|
|
}
|
|
#else
|
|
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)
|
|
{
|
|
return speex_resampler_process_native (st, channel_index, in, in_len, out,
|
|
out_len);
|
|
}
|
|
|
|
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)
|
|
{
|
|
spx_uint32_t i;
|
|
int istride_save, ostride_save;
|
|
|
|
#ifdef VAR_ARRAYS
|
|
spx_word16_t x[*in_len];
|
|
spx_word16_t y[*out_len];
|
|
|
|
/*VARDECL(spx_word16_t *x);
|
|
VARDECL(spx_word16_t *y);
|
|
ALLOC(x, *in_len, spx_word16_t);
|
|
ALLOC(y, *out_len, spx_word16_t); */
|
|
istride_save = st->in_stride;
|
|
ostride_save = st->out_stride;
|
|
if (in != NULL) {
|
|
for (i = 0; i < *in_len; i++)
|
|
x[i] = in[i * st->in_stride];
|
|
st->in_stride = st->out_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, x, in_len, y, out_len);
|
|
} else {
|
|
st->in_stride = st->out_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, NULL, in_len, y,
|
|
out_len);
|
|
}
|
|
st->in_stride = istride_save;
|
|
st->out_stride = ostride_save;
|
|
for (i = 0; i < *out_len; i++)
|
|
out[i * st->out_stride] = WORD2INT (y[i]);
|
|
#else
|
|
spx_word16_t x[FIXED_STACK_ALLOC];
|
|
spx_word16_t y[FIXED_STACK_ALLOC];
|
|
spx_uint32_t ilen = *in_len, olen = *out_len;
|
|
|
|
istride_save = st->in_stride;
|
|
ostride_save = st->out_stride;
|
|
while (ilen && olen) {
|
|
spx_uint32_t ichunk, ochunk;
|
|
|
|
ichunk = ilen;
|
|
ochunk = olen;
|
|
if (ichunk > FIXED_STACK_ALLOC)
|
|
ichunk = FIXED_STACK_ALLOC;
|
|
if (ochunk > FIXED_STACK_ALLOC)
|
|
ochunk = FIXED_STACK_ALLOC;
|
|
if (in != NULL) {
|
|
for (i = 0; i < ichunk; i++)
|
|
x[i] = in[i * st->in_stride];
|
|
st->in_stride = st->out_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, x, &ichunk, y,
|
|
&ochunk);
|
|
} else {
|
|
st->in_stride = st->out_stride = 1;
|
|
speex_resampler_process_native (st, channel_index, NULL, &ichunk, y,
|
|
&ochunk);
|
|
}
|
|
st->in_stride = istride_save;
|
|
st->out_stride = ostride_save;
|
|
for (i = 0; i < ochunk; i++)
|
|
out[i * st->out_stride] = WORD2INT (y[i]);
|
|
out += ochunk;
|
|
in += ichunk;
|
|
ilen -= ichunk;
|
|
olen -= ochunk;
|
|
}
|
|
*in_len -= ilen;
|
|
*out_len -= olen;
|
|
#endif
|
|
return RESAMPLER_ERR_SUCCESS;
|
|
}
|
|
#endif
|
|
|
|
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;
|
|
}
|
|
|
|
|
|
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;
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
void
|
|
speex_resampler_get_quality (SpeexResamplerState * st, int *quality)
|
|
{
|
|
*quality = st->quality;
|
|
}
|
|
|
|
void
|
|
speex_resampler_set_input_stride (SpeexResamplerState * st, spx_uint32_t stride)
|
|
{
|
|
st->in_stride = stride;
|
|
}
|
|
|
|
void
|
|
speex_resampler_get_input_stride (SpeexResamplerState * st,
|
|
spx_uint32_t * stride)
|
|
{
|
|
*stride = st->in_stride;
|
|
}
|
|
|
|
void
|
|
speex_resampler_set_output_stride (SpeexResamplerState * st,
|
|
spx_uint32_t stride)
|
|
{
|
|
st->out_stride = stride;
|
|
}
|
|
|
|
void
|
|
speex_resampler_get_output_stride (SpeexResamplerState * st,
|
|
spx_uint32_t * stride)
|
|
{
|
|
*stride = st->out_stride;
|
|
}
|
|
|
|
int
|
|
speex_resampler_get_input_latency (SpeexResamplerState * st)
|
|
{
|
|
return st->filt_len / 2;
|
|
}
|
|
|
|
int
|
|
speex_resampler_get_output_latency (SpeexResamplerState * st)
|
|
{
|
|
return ((st->filt_len / 2) * st->den_rate +
|
|
(st->num_rate >> 1)) / st->num_rate;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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.";
|
|
}
|
|
}
|