-Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1125 lines
40 KiB
C
1125 lines
40 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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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static void *speex_alloc (int size) {return calloc(size,1);}
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static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);}
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static void speex_free (void *ptr) {free(ptr);}
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#include "speex_resampler.h"
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#include "arch.h"
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#include "stack_alloc.h"
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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 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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/* 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 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 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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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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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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struct FuncDef {
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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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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 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] = -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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#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, 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, 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/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] = 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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int j;
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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 *sinc = & 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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float accum[4] = {0,0,0,0};
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for(j=0;j<N;j+=4) {
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accum[0] += sinc[j]*iptr[j];
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accum[1] += sinc[j+1]*iptr[j+1];
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accum[2] += sinc[j+2]*iptr[j+2];
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accum[3] += sinc[j+3]*iptr[j+3];
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}
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sum = accum[0] + accum[1] + accum[2] + accum[3];
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#else
|
|
sum = inner_product_single(sinc, iptr, N);
|
|
#endif
|
|
|
|
out[out_stride * out_sample++] = PSHR32(sum, 15);
|
|
last_sample += int_advance;
|
|
samp_frac_num += frac_advance;
|
|
if (samp_frac_num >= den_rate)
|
|
{
|
|
samp_frac_num -= den_rate;
|
|
last_sample++;
|
|
}
|
|
}
|
|
|
|
st->last_sample[channel_index] = last_sample;
|
|
st->samp_frac_num[channel_index] = samp_frac_num;
|
|
return out_sample;
|
|
}
|
|
|
|
#ifdef FIXED_POINT
|
|
#else
|
|
/* This is the same as the previous function, except with a double-precision accumulator */
|
|
static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
|
|
{
|
|
const int N = st->filt_len;
|
|
int out_sample = 0;
|
|
int last_sample = st->last_sample[channel_index];
|
|
spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
|
|
const spx_word16_t *sinc_table = st->sinc_table;
|
|
const int out_stride = st->out_stride;
|
|
const int int_advance = st->int_advance;
|
|
const int frac_advance = st->frac_advance;
|
|
const spx_uint32_t den_rate = st->den_rate;
|
|
double sum;
|
|
int j;
|
|
|
|
while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
|
|
{
|
|
const spx_word16_t *sinc = & sinc_table[samp_frac_num*N];
|
|
const spx_word16_t *iptr = & in[last_sample];
|
|
|
|
#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
|
|
double accum[4] = {0,0,0,0};
|
|
|
|
for(j=0;j<N;j+=4) {
|
|
accum[0] += sinc[j]*iptr[j];
|
|
accum[1] += sinc[j+1]*iptr[j+1];
|
|
accum[2] += sinc[j+2]*iptr[j+2];
|
|
accum[3] += sinc[j+3]*iptr[j+3];
|
|
}
|
|
sum = accum[0] + accum[1] + accum[2] + accum[3];
|
|
#else
|
|
sum = inner_product_double(sinc, iptr, N);
|
|
#endif
|
|
|
|
out[out_stride * out_sample++] = PSHR32(sum, 15);
|
|
last_sample += int_advance;
|
|
samp_frac_num += frac_advance;
|
|
if (samp_frac_num >= den_rate)
|
|
{
|
|
samp_frac_num -= den_rate;
|
|
last_sample++;
|
|
}
|
|
}
|
|
|
|
st->last_sample[channel_index] = last_sample;
|
|
st->samp_frac_num[channel_index] = samp_frac_num;
|
|
return out_sample;
|
|
}
|
|
#endif
|
|
|
|
static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
|
|
{
|
|
const int N = st->filt_len;
|
|
int out_sample = 0;
|
|
int last_sample = st->last_sample[channel_index];
|
|
spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
|
|
const int out_stride = st->out_stride;
|
|
const int int_advance = st->int_advance;
|
|
const int frac_advance = st->frac_advance;
|
|
const spx_uint32_t den_rate = st->den_rate;
|
|
int j;
|
|
spx_word32_t sum;
|
|
|
|
while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
|
|
{
|
|
const spx_word16_t *iptr = & in[last_sample];
|
|
|
|
const int offset = samp_frac_num*st->oversample/st->den_rate;
|
|
#ifdef FIXED_POINT
|
|
const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
|
|
#else
|
|
const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
|
|
#endif
|
|
spx_word16_t interp[4];
|
|
|
|
|
|
#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
|
|
spx_word32_t accum[4] = {0,0,0,0};
|
|
|
|
for(j=0;j<N;j++) {
|
|
const spx_word16_t curr_in=iptr[j];
|
|
accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
|
|
accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
|
|
accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
|
|
accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
|
|
}
|
|
|
|
cubic_coef(frac, interp);
|
|
sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
|
|
#else
|
|
cubic_coef(frac, interp);
|
|
sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
|
|
#endif
|
|
|
|
out[out_stride * out_sample++] = PSHR32(sum,15);
|
|
last_sample += int_advance;
|
|
samp_frac_num += frac_advance;
|
|
if (samp_frac_num >= den_rate)
|
|
{
|
|
samp_frac_num -= den_rate;
|
|
last_sample++;
|
|
}
|
|
}
|
|
|
|
st->last_sample[channel_index] = last_sample;
|
|
st->samp_frac_num[channel_index] = samp_frac_num;
|
|
return out_sample;
|
|
}
|
|
|
|
#ifdef FIXED_POINT
|
|
#else
|
|
/* This is the same as the previous function, except with a double-precision accumulator */
|
|
static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
|
|
{
|
|
const int N = st->filt_len;
|
|
int out_sample = 0;
|
|
int last_sample = st->last_sample[channel_index];
|
|
spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
|
|
const int out_stride = st->out_stride;
|
|
const int int_advance = st->int_advance;
|
|
const int frac_advance = st->frac_advance;
|
|
const spx_uint32_t den_rate = st->den_rate;
|
|
int j;
|
|
spx_word32_t sum;
|
|
|
|
while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len))
|
|
{
|
|
const spx_word16_t *iptr = & in[last_sample];
|
|
|
|
const int offset = samp_frac_num*st->oversample/st->den_rate;
|
|
#ifdef FIXED_POINT
|
|
const spx_word16_t frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate);
|
|
#else
|
|
const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate;
|
|
#endif
|
|
spx_word16_t interp[4];
|
|
|
|
|
|
#ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
|
|
double accum[4] = {0,0,0,0};
|
|
|
|
for(j=0;j<N;j++) {
|
|
const double curr_in=iptr[j];
|
|
accum[0] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-2]);
|
|
accum[1] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset-1]);
|
|
accum[2] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset]);
|
|
accum[3] += MULT16_16(curr_in,st->sinc_table[4+(j+1)*st->oversample-offset+1]);
|
|
}
|
|
|
|
cubic_coef(frac, interp);
|
|
sum = MULT16_32_Q15(interp[0],accum[0]) + MULT16_32_Q15(interp[1],accum[1]) + MULT16_32_Q15(interp[2],accum[2]) + MULT16_32_Q15(interp[3],accum[3]);
|
|
#else
|
|
cubic_coef(frac, interp);
|
|
sum = interpolate_product_double(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp);
|
|
#endif
|
|
|
|
out[out_stride * out_sample++] = PSHR32(sum,15);
|
|
last_sample += int_advance;
|
|
samp_frac_num += frac_advance;
|
|
if (samp_frac_num >= den_rate)
|
|
{
|
|
samp_frac_num -= den_rate;
|
|
last_sample++;
|
|
}
|
|
}
|
|
|
|
st->last_sample[channel_index] = last_sample;
|
|
st->samp_frac_num[channel_index] = samp_frac_num;
|
|
return out_sample;
|
|
}
|
|
#endif
|
|
|
|
static void update_filter(SpeexResamplerState *st)
|
|
{
|
|
spx_uint32_t old_length;
|
|
|
|
old_length = st->filt_len;
|
|
st->oversample = quality_map[st->quality].oversample;
|
|
st->filt_len = quality_map[st->quality].base_length;
|
|
|
|
if (st->num_rate > st->den_rate)
|
|
{
|
|
/* down-sampling */
|
|
st->cutoff = quality_map[st->quality].downsample_bandwidth * st->den_rate / st->num_rate;
|
|
/* FIXME: divide the numerator and denominator by a certain amount if they're too large */
|
|
st->filt_len = st->filt_len*st->num_rate / st->den_rate;
|
|
/* Round down to make sure we have a multiple of 4 */
|
|
st->filt_len &= (~0x3);
|
|
if (2*st->den_rate < st->num_rate)
|
|
st->oversample >>= 1;
|
|
if (4*st->den_rate < st->num_rate)
|
|
st->oversample >>= 1;
|
|
if (8*st->den_rate < st->num_rate)
|
|
st->oversample >>= 1;
|
|
if (16*st->den_rate < st->num_rate)
|
|
st->oversample >>= 1;
|
|
if (st->oversample < 1)
|
|
st->oversample = 1;
|
|
} else {
|
|
/* up-sampling */
|
|
st->cutoff = quality_map[st->quality].upsample_bandwidth;
|
|
}
|
|
|
|
/* Choose the resampling type that requires the least amount of memory */
|
|
if (st->den_rate <= st->oversample)
|
|
{
|
|
spx_uint32_t i;
|
|
if (!st->sinc_table)
|
|
st->sinc_table = (spx_word16_t *)speex_alloc(st->filt_len*st->den_rate*sizeof(spx_word16_t));
|
|
else if (st->sinc_table_length < st->filt_len*st->den_rate)
|
|
{
|
|
st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,st->filt_len*st->den_rate*sizeof(spx_word16_t));
|
|
st->sinc_table_length = st->filt_len*st->den_rate;
|
|
}
|
|
for (i=0;i<st->den_rate;i++)
|
|
{
|
|
spx_int32_t j;
|
|
for (j=0;j<st->filt_len;j++)
|
|
{
|
|
st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-(spx_int32_t)st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func);
|
|
}
|
|
}
|
|
#ifdef FIXED_POINT
|
|
st->resampler_ptr = resampler_basic_direct_single;
|
|
#else
|
|
if (st->quality>8)
|
|
st->resampler_ptr = resampler_basic_direct_double;
|
|
else
|
|
st->resampler_ptr = resampler_basic_direct_single;
|
|
#endif
|
|
/*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/
|
|
} else {
|
|
spx_int32_t i;
|
|
if (!st->sinc_table)
|
|
st->sinc_table = (spx_word16_t *)speex_alloc((st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
|
|
else if (st->sinc_table_length < st->filt_len*st->oversample+8)
|
|
{
|
|
st->sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,(st->filt_len*st->oversample+8)*sizeof(spx_word16_t));
|
|
st->sinc_table_length = st->filt_len*st->oversample+8;
|
|
}
|
|
for (i=-4;i<(spx_int32_t)(st->oversample*st->filt_len+4);i++)
|
|
st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func);
|
|
#ifdef FIXED_POINT
|
|
st->resampler_ptr = resampler_basic_interpolate_single;
|
|
#else
|
|
if (st->quality>8)
|
|
st->resampler_ptr = resampler_basic_interpolate_double;
|
|
else
|
|
st->resampler_ptr = resampler_basic_interpolate_single;
|
|
#endif
|
|
/*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/
|
|
}
|
|
st->int_advance = st->num_rate/st->den_rate;
|
|
st->frac_advance = st->num_rate%st->den_rate;
|
|
|
|
|
|
/* Here's the place where we update the filter memory to take into account
|
|
the change in filter length. It's probably the messiest part of the code
|
|
due to handling of lots of corner cases. */
|
|
if (!st->mem)
|
|
{
|
|
spx_uint32_t i;
|
|
st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
|
|
st->mem = (spx_word16_t*)speex_alloc(st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
|
|
for (i=0;i<st->nb_channels*st->mem_alloc_size;i++)
|
|
st->mem[i] = 0;
|
|
/*speex_warning("init filter");*/
|
|
} else if (!st->started)
|
|
{
|
|
spx_uint32_t i;
|
|
st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
|
|
st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
|
|
for (i=0;i<st->nb_channels*st->mem_alloc_size;i++)
|
|
st->mem[i] = 0;
|
|
/*speex_warning("reinit filter");*/
|
|
} else if (st->filt_len > old_length)
|
|
{
|
|
spx_int32_t i;
|
|
/* Increase the filter length */
|
|
/*speex_warning("increase filter size");*/
|
|
int old_alloc_size = st->mem_alloc_size;
|
|
if ((st->filt_len-1 + st->buffer_size) > st->mem_alloc_size)
|
|
{
|
|
st->mem_alloc_size = st->filt_len-1 + st->buffer_size;
|
|
st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*st->mem_alloc_size * sizeof(spx_word16_t));
|
|
}
|
|
for (i=st->nb_channels-1;i>=0;i--)
|
|
{
|
|
spx_int32_t j;
|
|
spx_uint32_t olen = old_length;
|
|
/*if (st->magic_samples[i])*/
|
|
{
|
|
/* Try and remove the magic samples as if nothing had happened */
|
|
|
|
/* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
|
|
olen = old_length + 2*st->magic_samples[i];
|
|
for (j=old_length-2+st->magic_samples[i];j>=0;j--)
|
|
st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]] = st->mem[i*old_alloc_size+j];
|
|
for (j=0;j<st->magic_samples[i];j++)
|
|
st->mem[i*st->mem_alloc_size+j] = 0;
|
|
st->magic_samples[i] = 0;
|
|
}
|
|
if (st->filt_len > olen)
|
|
{
|
|
/* If the new filter length is still bigger than the "augmented" length */
|
|
/* Copy data going backward */
|
|
for (j=0;j<olen-1;j++)
|
|
st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*st->mem_alloc_size+(olen-2-j)];
|
|
/* Then put zeros for lack of anything better */
|
|
for (;j<st->filt_len-1;j++)
|
|
st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = 0;
|
|
/* Adjust last_sample */
|
|
st->last_sample[i] += (st->filt_len - olen)/2;
|
|
} else {
|
|
/* Put back some of the magic! */
|
|
st->magic_samples[i] = (olen - st->filt_len)/2;
|
|
for (j=0;j<st->filt_len-1+st->magic_samples[i];j++)
|
|
st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
|
|
}
|
|
}
|
|
} else if (st->filt_len < old_length)
|
|
{
|
|
spx_uint32_t i;
|
|
/* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
|
|
samples so they can be used directly as input the next time(s) */
|
|
for (i=0;i<st->nb_channels;i++)
|
|
{
|
|
spx_uint32_t j;
|
|
spx_uint32_t old_magic = st->magic_samples[i];
|
|
st->magic_samples[i] = (old_length - st->filt_len)/2;
|
|
/* We must copy some of the memory that's no longer used */
|
|
/* Copy data going backward */
|
|
for (j=0;j<st->filt_len-1+st->magic_samples[i]+old_magic;j++)
|
|
st->mem[i*st->mem_alloc_size+j] = st->mem[i*st->mem_alloc_size+j+st->magic_samples[i]];
|
|
st->magic_samples[i] += old_magic;
|
|
}
|
|
}
|
|
|
|
}
|
|
|
|
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;
|
|
|
|
#ifdef FIXED_POINT
|
|
st->buffer_size = 160;
|
|
#else
|
|
st->buffer_size = 160;
|
|
#endif
|
|
|
|
/* Per channel data */
|
|
st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(int));
|
|
st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
|
|
st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int));
|
|
for (i=0;i<nb_channels;i++)
|
|
{
|
|
st->last_sample[i] = 0;
|
|
st->magic_samples[i] = 0;
|
|
st->samp_frac_num[i] = 0;
|
|
}
|
|
|
|
speex_resampler_set_quality(st, quality);
|
|
speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate);
|
|
|
|
|
|
update_filter(st);
|
|
|
|
st->initialised = 1;
|
|
if (err)
|
|
*err = RESAMPLER_ERR_SUCCESS;
|
|
|
|
return st;
|
|
}
|
|
|
|
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
|
|
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
|
|
int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_index, const float *in, spx_uint32_t *in_len, float *out, spx_uint32_t *out_len)
|
|
#endif
|
|
{
|
|
int j;
|
|
spx_uint32_t ilen = *in_len;
|
|
spx_uint32_t olen = *out_len;
|
|
spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
|
|
const int filt_offs = st->filt_len - 1;
|
|
const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
|
|
const int istride = st->in_stride;
|
|
|
|
if (st->magic_samples[channel_index])
|
|
olen -= speex_resampler_magic(st, channel_index, &out, olen);
|
|
if (! st->magic_samples[channel_index]) {
|
|
while (ilen && olen) {
|
|
spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
|
|
spx_uint32_t ochunk = olen;
|
|
|
|
if (in) {
|
|
for(j=0;j<ichunk;++j)
|
|
x[j+filt_offs]=in[j*istride];
|
|
} else {
|
|
for(j=0;j<ichunk;++j)
|
|
x[j+filt_offs]=0;
|
|
}
|
|
speex_resampler_process_native(st, channel_index, &ichunk, out, &ochunk);
|
|
ilen -= ichunk;
|
|
olen -= ochunk;
|
|
out += ochunk * st->out_stride;
|
|
if (in)
|
|
in += ichunk * istride;
|
|
}
|
|
}
|
|
*in_len -= ilen;
|
|
*out_len -= olen;
|
|
return RESAMPLER_ERR_SUCCESS;
|
|
}
|
|
|
|
#ifdef FIXED_POINT
|
|
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
|
|
int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len)
|
|
#endif
|
|
{
|
|
int j;
|
|
const int istride_save = st->in_stride;
|
|
const int ostride_save = st->out_stride;
|
|
spx_uint32_t ilen = *in_len;
|
|
spx_uint32_t olen = *out_len;
|
|
spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
|
|
const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1);
|
|
#ifdef VAR_ARRAYS
|
|
const unsigned int ylen = (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC;
|
|
VARDECL(spx_word16_t *ystack);
|
|
ALLOC(ystack, ylen, spx_word16_t);
|
|
#else
|
|
const unsigned int ylen = FIXED_STACK_ALLOC;
|
|
spx_word16_t ystack[FIXED_STACK_ALLOC];
|
|
#endif
|
|
|
|
st->out_stride = 1;
|
|
|
|
while (ilen && olen) {
|
|
spx_word16_t *y = ystack;
|
|
spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
|
|
spx_uint32_t ochunk = (olen > ylen) ? ylen : olen;
|
|
spx_uint32_t omagic = 0;
|
|
|
|
if (st->magic_samples[channel_index]) {
|
|
omagic = speex_resampler_magic(st, channel_index, &y, ochunk);
|
|
ochunk -= omagic;
|
|
olen -= omagic;
|
|
}
|
|
if (! st->magic_samples[channel_index]) {
|
|
if (in) {
|
|
for(j=0;j<ichunk;++j)
|
|
#ifdef FIXED_POINT
|
|
x[j+st->filt_len-1]=WORD2INT(in[j*istride_save]);
|
|
#else
|
|
x[j+st->filt_len-1]=in[j*istride_save];
|
|
#endif
|
|
} else {
|
|
for(j=0;j<ichunk;++j)
|
|
x[j+st->filt_len-1]=0;
|
|
}
|
|
|
|
speex_resampler_process_native(st, channel_index, &ichunk, y, &ochunk);
|
|
} else {
|
|
ichunk = 0;
|
|
ochunk = 0;
|
|
}
|
|
|
|
for (j=0;j<ochunk+omagic;++j)
|
|
#ifdef FIXED_POINT
|
|
out[j*ostride_save] = ystack[j];
|
|
#else
|
|
out[j*ostride_save] = WORD2INT(ystack[j]);
|
|
#endif
|
|
|
|
ilen -= ichunk;
|
|
olen -= ochunk;
|
|
out += (ochunk+omagic) * ostride_save;
|
|
if (in)
|
|
in += ichunk * istride_save;
|
|
}
|
|
st->out_stride = ostride_save;
|
|
*in_len -= ilen;
|
|
*out_len -= olen;
|
|
|
|
return RESAMPLER_ERR_SUCCESS;
|
|
}
|
|
|
|
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.";
|
|
}
|
|
}
|