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|
/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2004 M. Bakker, Ahead Software AG, http://www.nero.com
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU General Public License for more details.
**
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Ahead Software through Mpeg4AAClicense@nero.com.
**
** $Id$
**/
#include "common.h"
#include "structs.h"
#ifdef SBR_DEC
#include "sbr_syntax.h"
#include "syntax.h"
#include "sbr_huff.h"
#include "sbr_fbt.h"
#include "sbr_tf_grid.h"
#include "sbr_e_nf.h"
#include "bits.h"
#ifdef PS_DEC
#include "ps_dec.h"
#endif
#ifdef DRM_PS
#include "drm_dec.h"
#endif
#include "analysis.h"
/* static function declarations */
static void sbr_header(bitfile *ld, sbr_info *sbr);
static uint8_t calc_sbr_tables(sbr_info *sbr, uint8_t start_freq, uint8_t stop_freq,
uint8_t samplerate_mode, uint8_t freq_scale,
uint8_t alter_scale, uint8_t xover_band);
static uint8_t sbr_data(bitfile *ld, sbr_info *sbr);
static uint16_t sbr_extension(bitfile *ld, sbr_info *sbr,
uint8_t bs_extension_id, uint16_t num_bits_left);
static uint8_t sbr_single_channel_element(bitfile *ld, sbr_info *sbr);
static uint8_t sbr_channel_pair_element(bitfile *ld, sbr_info *sbr);
static uint8_t sbr_grid(bitfile *ld, sbr_info *sbr, uint8_t ch);
static void sbr_dtdf(bitfile *ld, sbr_info *sbr, uint8_t ch);
static void invf_mode(bitfile *ld, sbr_info *sbr, uint8_t ch);
static void sinusoidal_coding(bitfile *ld, sbr_info *sbr, uint8_t ch);
static void sbr_reset(sbr_info *sbr)
{
#if 0
printf("%d\n", sbr->bs_start_freq_prev);
printf("%d\n", sbr->bs_stop_freq_prev);
printf("%d\n", sbr->bs_freq_scale_prev);
printf("%d\n", sbr->bs_alter_scale_prev);
printf("%d\n", sbr->bs_xover_band_prev);
printf("%d\n\n", sbr->bs_noise_bands_prev);
#endif
/* if these are different from the previous frame: Reset = 1 */
if ((sbr->bs_start_freq != sbr->bs_start_freq_prev) ||
(sbr->bs_stop_freq != sbr->bs_stop_freq_prev) ||
(sbr->bs_freq_scale != sbr->bs_freq_scale_prev) ||
(sbr->bs_alter_scale != sbr->bs_alter_scale_prev) ||
(sbr->bs_xover_band != sbr->bs_xover_band_prev) ||
(sbr->bs_noise_bands != sbr->bs_noise_bands_prev))
{
sbr->Reset = 1;
} else {
sbr->Reset = 0;
}
sbr->bs_start_freq_prev = sbr->bs_start_freq;
sbr->bs_stop_freq_prev = sbr->bs_stop_freq;
sbr->bs_freq_scale_prev = sbr->bs_freq_scale;
sbr->bs_alter_scale_prev = sbr->bs_alter_scale;
sbr->bs_xover_band_prev = sbr->bs_xover_band;
sbr->bs_noise_bands_prev = sbr->bs_noise_bands;
}
static uint8_t calc_sbr_tables(sbr_info *sbr, uint8_t start_freq, uint8_t stop_freq,
uint8_t samplerate_mode, uint8_t freq_scale,
uint8_t alter_scale, uint8_t xover_band)
{
uint8_t result = 0;
uint8_t k2;
/* calculate the Master Frequency Table */
sbr->k0 = qmf_start_channel(start_freq, samplerate_mode, sbr->sample_rate);
k2 = qmf_stop_channel(stop_freq, sbr->sample_rate, sbr->k0);
/* check k0 and k2 */
if (sbr->sample_rate >= 48000)
{
if ((k2 - sbr->k0) > 32)
result += 1;
} else if (sbr->sample_rate <= 32000) {
if ((k2 - sbr->k0) > 48)
result += 1;
} else { /* (sbr->sample_rate == 44100) */
if ((k2 - sbr->k0) > 45)
result += 1;
}
if (freq_scale == 0)
{
result += master_frequency_table_fs0(sbr, sbr->k0, k2, alter_scale);
} else {
result += master_frequency_table(sbr, sbr->k0, k2, freq_scale, alter_scale);
}
result += derived_frequency_table(sbr, xover_band, k2);
result = (result > 0) ? 1 : 0;
return result;
}
/* table 2 */
uint8_t sbr_extension_data(bitfile *ld, sbr_info *sbr, uint16_t cnt)
{
uint8_t result = 0;
uint16_t num_align_bits = 0;
uint16_t num_sbr_bits = (uint16_t)faad_get_processed_bits(ld);
uint8_t saved_start_freq, saved_samplerate_mode;
uint8_t saved_stop_freq, saved_freq_scale;
uint8_t saved_alter_scale, saved_xover_band;
#ifdef DRM
if (!sbr->Is_DRM_SBR)
#endif
{
uint8_t bs_extension_type = (uint8_t)faad_getbits(ld, 4
DEBUGVAR(1,198,"sbr_bitstream(): bs_extension_type"));
if (bs_extension_type == EXT_SBR_DATA_CRC)
{
sbr->bs_sbr_crc_bits = (uint16_t)faad_getbits(ld, 10
DEBUGVAR(1,199,"sbr_bitstream(): bs_sbr_crc_bits"));
}
}
/* save old header values, in case the new ones are corrupted */
saved_start_freq = sbr->bs_start_freq;
saved_samplerate_mode = sbr->bs_samplerate_mode;
saved_stop_freq = sbr->bs_stop_freq;
saved_freq_scale = sbr->bs_freq_scale;
saved_alter_scale = sbr->bs_alter_scale;
saved_xover_band = sbr->bs_xover_band;
sbr->bs_header_flag = faad_get1bit(ld
DEBUGVAR(1,200,"sbr_bitstream(): bs_header_flag"));
if (sbr->bs_header_flag)
sbr_header(ld, sbr);
/* Reset? */
sbr_reset(sbr);
/* first frame should have a header */
//if (!(sbr->frame == 0 && sbr->bs_header_flag == 0))
if (sbr->header_count != 0)
{
if (sbr->Reset || (sbr->bs_header_flag && sbr->just_seeked))
{
uint8_t rt = calc_sbr_tables(sbr, sbr->bs_start_freq, sbr->bs_stop_freq,
sbr->bs_samplerate_mode, sbr->bs_freq_scale,
sbr->bs_alter_scale, sbr->bs_xover_band);
/* if an error occured with the new header values revert to the old ones */
if (rt > 0)
{
calc_sbr_tables(sbr, saved_start_freq, saved_stop_freq,
saved_samplerate_mode, saved_freq_scale, opt">, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
0.00001000, 0.00000000};
/*
static double kaiser12_table[36] = {
0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
*/
static double kaiser10_table[36] = {
0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000};
static double kaiser8_table[36] = {
0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000};
static double kaiser6_table[36] = {
0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000};
struct FuncDef {
double *table;
int oversample;
};
static struct FuncDef _KAISER12 = {kaiser12_table, 64};
#define KAISER12 (&_KAISER12)
/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
#define KAISER12 (&_KAISER12)*/
static struct FuncDef _KAISER10 = {kaiser10_table, 32};
#define KAISER10 (&_KAISER10)
static struct FuncDef _KAISER8 = {kaiser8_table, 32};
#define KAISER8 (&_KAISER8)
static struct FuncDef _KAISER6 = {kaiser6_table, 32};
#define KAISER6 (&_KAISER6)
struct QualityMapping {
int base_length;
int oversample;
float downsample_bandwidth;
float upsample_bandwidth;
struct FuncDef *window_func;
};
/* This table maps conversion quality to internal parameters. There are two
reasons that explain why the up-sampling bandwidth is larger than the
down-sampling bandwidth:
1) When up-sampling, we can assume that the spectrum is already attenuated
close to the Nyquist rate (from an A/D or a previous resampling filter)
2) Any aliasing that occurs very close to the Nyquist rate will be masked
by the sinusoids/noise just below the Nyquist rate (guaranteed only for
up-sampling).
*/
static const struct QualityMapping quality_map[11] = {
{ 8, 4, 0.830f, 0.860f, KAISER6 }, /* Q0 */
{ 16, 4, 0.850f, 0.880f, KAISER6 }, /* Q1 */
{ 32, 4, 0.882f, 0.910f, KAISER6 }, /* Q2 */ /* 82.3% cutoff ( ~60 dB stop) 6 */
{ 48, 8, 0.895f, 0.917f, KAISER8 }, /* Q3 */ /* 84.9% cutoff ( ~80 dB stop) 8 */
{ 64, 8, 0.921f, 0.940f, KAISER8 }, /* Q4 */ /* 88.7% cutoff ( ~80 dB stop) 8 */
{ 80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 */ /* 89.1% cutoff (~100 dB stop) 10 */
{ 96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 */ /* 91.5% cutoff (~100 dB stop) 10 */
{128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */ /* 93.1% cutoff (~100 dB stop) 10 */
{160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */ /* 94.5% cutoff (~100 dB stop) 10 */
{192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 */ /* 95.5% cutoff (~100 dB stop) 10 */
{256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
};
/*8,24,40,56,80,104,128,160,200,256,320*/
static double compute_func(float x, struct FuncDef *func)
{
float y, frac;
double interp[4];
int ind;
y = x*func->oversample;
ind = (int)floor(y);
frac = (y-ind);
/* CSE with handle the repeated powers */
interp[3] = -0.1666666667*frac + 0.1666666667*(frac*frac*frac);
interp[2] = frac + 0.5*(frac*frac) - 0.5*(frac*frac*frac);
/*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac);
/* Just to make sure we don't have rounding problems */
interp[1] = 1.f-interp[3]-interp[2]-interp[0];
/*sum = frac*accum[1] + (1-frac)*accum[2];*/
return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3];
}
#if 0
#include <stdio.h>
int main(int argc, char **argv)
{
int i;
for (i=0;i<256;i++)
{
printf ("%f\n", compute_func(i/256., KAISER12));
}
return 0;
}
#endif
#ifdef FIXED_POINT
/* The slow way of computing a sinc for the table. Should improve that some day */
static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
{
/*fprintf (stderr, "%f ", x);*/
float xx = x * cutoff;
if (fabs(x)<1e-6f)
return WORD2INT(32768.*cutoff);
else if (fabs(x) > .5f*N)
return 0;
/*FIXME: Can it really be any slower than this? */
return WORD2INT(32768.*cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func));
}
#else
/* The slow way of computing a sinc for the table. Should improve that some day */
static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
{
/*fprintf (stderr, "%f ", x);*/
float xx = x * cutoff;
if (fabs(x)<1e-6)
return cutoff;
else if (fabs(x) > .5*N)
return 0;
/*FIXME: Can it really be any slower than this? */
return cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func(fabs(2.*x/N), window_func);
}
#endif
#ifdef FIXED_POINT
static void cubic_coef(spx_word16_t x, spx_word16_t interp[4])
{
/* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
but I know it's MMSE-optimal on a sinc */
spx_word16_t x2, x3;
x2 = MULT16_16_P15(x, x);
x3 = MULT16_16_P15(x, x2);
interp[0] = PSHR32(MULT16_16(QCONST16(-0.16667f, 15),x) + MULT16_16(QCONST16(0.16667f, 15),x3),15);
interp[1] = EXTRACT16(EXTEND32(x) + SHR32(SUB32(EXTEND32(x2),EXTEND32(x3)),1));
interp[3] = PSHR32(MULT16_16(QCONST16(-0.33333f, 15),x) + MULT16_16(QCONST16(.5f,15),x2) - MULT16_16(QCONST16(0.16667f, 15),x3),15);
/* Just to make sure we don't have rounding problems */
interp[2] = Q15_ONE-interp[0]-interp[1]-interp[3];
if (interp[2]<32767)
interp[2]+=1;
}
#else
static void cubic_coef(spx_word16_t frac, spx_word16_t interp[4])
{
/* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
but I know it's MMSE-optimal on a sinc */
interp[0] = -0.16667f*frac + 0.16667f*frac*frac*frac;
interp[1] = frac + 0.5f*frac*frac - 0.5f*frac*frac*frac;
/*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac;*/
interp[3] = -0.33333f*frac + 0.5f*frac*frac - 0.16667f*frac*frac*frac;
/* Just to make sure we don't have rounding problems */
interp[2] = 1.-interp[0]-interp[1]-interp[3];
}
#endif
static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t channel_index, const spx_word16_t *in, spx_uint32_t *in_len, spx_word16_t *out, spx_uint32_t *out_len)
{
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 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];
for (;j<N-1;j++)
mem[j] = in[st->in_stride*(j+*in_len-N+1)];
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;
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);
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;
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);
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;
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);
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;
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);
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_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.";
}
}
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