ref: 27f834ac163241a5ebb09a7952d8b704b30e1a60
dir: /src/rate.c/
/* Effect: change sample rate Copyright (c) 2008 robs@users.sourceforge.net
*
* This library is free software; you can redistribute it and/or modify it
* under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or (at
* your option) any later version.
*
* This library 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 Lesser
* General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with this library; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/* Based upon the techniques described in `The Quest For The Perfect Resampler'
* by Laurent De Soras; http://ldesoras.free.fr/doc/articles/resampler-en.pdf */
#ifdef NDEBUG /* Enable assert always. */
#undef NDEBUG /* Must undef above assert.h or other that might include it. */
#endif
#include "sox_i.h"
#include "fft4g.h"
#include "getopt.h"
#define FIFO_SIZE_T int
#include "fifo.h"
#include <assert.h>
#include <string.h>
#define calloc lsx_calloc
#define malloc lsx_malloc
#define raw_coef_t double
#define sample_t double
#define TO_SOX SOX_FLOAT_64BIT_TO_SAMPLE
#define FROM_SOX SOX_SAMPLE_TO_FLOAT_64BIT
#define coef(coef_p, interp_order, fir_len, phase_num, coef_interp_num, fir_coef_num) coef_p[(fir_len) * ((interp_order) + 1) * (phase_num) + ((interp_order) + 1) * (fir_coef_num) + (interp_order - coef_interp_num)]
static sample_t * prepare_coefs(raw_coef_t const * coefs, int num_coefs,
int num_phases, int interp_order, int multiplier)
{
int i, j, length = num_coefs * num_phases;
sample_t * result = malloc(length * (interp_order + 1) * sizeof(*result));
double fm1 = coefs[0], f1 = 0, f2 = 0;
for (i = num_coefs - 1; i >= 0; --i)
for (j = num_phases - 1; j >= 0; --j) {
double f0 = fm1, b = 0, c = 0, d = 0; /* = 0 to kill compiler warning */
int pos = i * num_phases + j - 1;
fm1 = (pos > 0 ? coefs[pos - 1] : 0) * multiplier;
switch (interp_order) {
case 1: b = f1 - f0; break;
case 2: b = f1 - (.5 * (f2+f0) - f1) - f0; c = .5 * (f2+f0) - f1; break;
case 3: c=.5*(f1+fm1)-f0;d=(1/6.)*(f2-f1+fm1-f0-4*c);b=f1-f0-d-c; break;
default: if (interp_order) assert(0);
}
#define coef_coef(x) \
coef(result, interp_order, num_coefs, j, x, num_coefs - 1 - i)
coef_coef(0) = f0;
if (interp_order > 0) coef_coef(1) = b;
if (interp_order > 1) coef_coef(2) = c;
if (interp_order > 2) coef_coef(3) = d;
#undef coef_coef
f2 = f1, f1 = f0;
}
return result;
}
typedef struct {
int dft_length, num_taps, post_peak;
sample_t * coefs;
} half_band_t; /* Note: not half-band as in symmetric about Fn/2 (Fs/4) */
typedef struct { /* Data that are shared between channels and filters */
sample_t * poly_fir_coefs;
half_band_t half_band[2]; /* [0]: halve; [1]: down/up: halve/double */
} rate_shared_t;
struct stage;
typedef void (* stage_fn_t)(struct stage * input, fifo_t * output);
typedef struct stage {
rate_shared_t * shared;
fifo_t fifo;
int pre; /* Number of past samples to store */
int pre_post; /* pre + number of future samples to store */
int preload; /* Number of zero samples to pre-load the fifo */
int which; /* Which of the 2 half-band filters to use */
stage_fn_t fn;
/* For poly_fir & spline: */
union { /* 32bit.32bit fixed point arithmetic */
#if defined(WORDS_BIGENDIAN)
struct {int32_t integer; uint32_t fraction;} parts;
#else
struct {uint32_t fraction; int32_t integer;} parts;
#endif
int64_t all;
#define MULT32 (65536. * 65536.)
} at, step;
int divisor; /* For step: > 1 for rational; 1 otherwise */
double out_in_ratio;
} stage_t;
#define stage_occupancy(s) max(0, fifo_occupancy(&(s)->fifo) - (s)->pre_post)
#define stage_read_p(s) ((sample_t *)fifo_read_ptr(&(s)->fifo) + (s)->pre)
static void cubic_spline(stage_t * p, fifo_t * output_fifo)
{
int i, num_in = stage_occupancy(p), max_num_out = 1 + num_in*p->out_in_ratio;
sample_t const * input = stage_read_p(p);
sample_t * output = fifo_reserve(output_fifo, max_num_out);
for (i = 0; p->at.parts.integer < num_in; ++i, p->at.all += p->step.all) {
sample_t const * s = input + p->at.parts.integer;
sample_t x = p->at.parts.fraction * (1 / MULT32);
sample_t b = .5*(s[1]+s[-1])-*s, a = (1/6.)*(s[2]-s[1]+s[-1]-*s-4*b);
sample_t c = s[1]-*s-a-b;
output[i] = ((a*x + b)*x + c)*x + *s;
}
assert(max_num_out - i >= 0);
fifo_trim_by(output_fifo, max_num_out - i);
fifo_read(&p->fifo, p->at.parts.integer, NULL);
p->at.parts.integer = 0;
}
static void half_sample(stage_t * p, fifo_t * output_fifo)
{
sample_t * output;
int i, j, num_in = max(0, fifo_occupancy(&p->fifo));
rate_shared_t const * s = p->shared;
half_band_t const * f = &s->half_band[p->which];
int const overlap = f->num_taps - 1;
while (num_in >= f->dft_length) {
sample_t const * input = fifo_read_ptr(&p->fifo);
fifo_read(&p->fifo, f->dft_length - overlap, NULL);
num_in -= f->dft_length - overlap;
output = fifo_reserve(output_fifo, f->dft_length);
fifo_trim_by(output_fifo, (f->dft_length + overlap) >> 1);
memcpy(output, input, f->dft_length * sizeof(*output));
lsx_rdft(f->dft_length, 1, output, lsx_fft_br, lsx_fft_sc);
output[0] *= f->coefs[0];
output[1] *= f->coefs[1];
for (i = 2; i < f->dft_length; i += 2) {
sample_t tmp = output[i];
output[i ] = f->coefs[i ] * tmp - f->coefs[i+1] * output[i+1];
output[i+1] = f->coefs[i+1] * tmp + f->coefs[i ] * output[i+1];
}
lsx_rdft(f->dft_length, -1, output, lsx_fft_br, lsx_fft_sc);
for (j = 1, i = 2; i < f->dft_length - overlap; ++j, i += 2)
output[j] = output[i];
}
}
static void double_sample(stage_t * p, fifo_t * output_fifo)
{
sample_t * output;
int i, j, num_in = max(0, fifo_occupancy(&p->fifo));
rate_shared_t const * s = p->shared;
half_band_t const * f = &s->half_band[1];
int const overlap = f->num_taps - 1;
while (num_in > f->dft_length >> 1) {
sample_t const * input = fifo_read_ptr(&p->fifo);
fifo_read(&p->fifo, (f->dft_length - overlap) >> 1, NULL);
num_in -= (f->dft_length - overlap) >> 1;
output = fifo_reserve(output_fifo, f->dft_length);
fifo_trim_by(output_fifo, overlap);
for (j = i = 0; i < f->dft_length; ++j, i += 2)
output[i] = input[j], output[i+1] = 0;
lsx_rdft(f->dft_length, 1, output, lsx_fft_br, lsx_fft_sc);
output[0] *= f->coefs[0];
output[1] *= f->coefs[1];
for (i = 2; i < f->dft_length; i += 2) {
sample_t tmp = output[i];
output[i ] = f->coefs[i ] * tmp - f->coefs[i+1] * output[i+1];
output[i+1] = f->coefs[i+1] * tmp + f->coefs[i ] * output[i+1];
}
lsx_rdft(f->dft_length, -1, output, lsx_fft_br, lsx_fft_sc);
}
}
static double * make_lpf(int num_taps, double Fc, double beta, double scale)
{
double * h = malloc(num_taps * sizeof(*h)), sum = 0;
int i, m = num_taps - 1;
assert(Fc >= 0 && Fc <= 1);
for (i = 0; i <= m / 2; ++i) {
double x = M_PI * (i - .5 * m), y = 2. * i / m - 1;
h[i] = x? sin(Fc * x) / x : Fc;
sum += h[i] *= lsx_bessel_I_0(beta * sqrt(1 - y * y));
if (m - i != i)
sum += h[m - i] = h[i];
}
for (i = 0; i < num_taps; ++i) h[i] *= scale / sum;
return h;
}
#define TO_6dB .5869
#define TO_3dB ((2/3.) * (.5 + TO_6dB))
#define MAX_TBW0 36.
#define MAX_TBW0A (MAX_TBW0 / (1 + TO_3dB))
#define MAX_TBW3 floor(MAX_TBW0 * TO_3dB)
#define MAX_TBW3A floor(MAX_TBW0A * TO_3dB)
static double * design_lpf(
double Fp, /* End of pass-band; ~= 0.01dB point */
double Fc, /* Start of stop-band */
double Fn, /* Nyquist freq; e.g. 0.5, 1, PI */
sox_bool allow_aliasing,
double att, /* Stop-band attenuation in dB */
int * num_taps, /* (Single phase.) 0: value will be estimated */
int k) /* Number of phases; 0 for single-phase */
{
double tr_bw, beta;
if (allow_aliasing)
Fc += (Fc - Fp) * TO_3dB;
Fp /= Fn, Fc /= Fn; /* Normalise to Fn = 1 */
tr_bw = TO_6dB * (Fc - Fp); /* Transition band-width: 6dB to stop points */
if (*num_taps == 0) { /* TODO this could be cleaner, esp. for k != 0 */
double n160 = (.0425* att - 1.4) / tr_bw; /* Half order for att = 160 */
int n = n160 * (16.556 / (att - 39.6) + .8625) + .5; /* For att [80,160) */
*num_taps = k? 2 * n : 2 * (n + (n & 1)) + 1; /* =1 %4 (0 phase 1/2 band) */
}
assert(att >= 80);
beta = att < 100 ? .1102 * (att - 8.7) : .1117 * att - 1.11;
if (k)
*num_taps = *num_taps * k - 1;
else k = 1;
return make_lpf(*num_taps, (Fc - tr_bw) / k, beta, (double)k);
}
static void fir_to_phase(double * * h, int * len,
int * post_len, double phase0)
{
double * work, phase = (phase0 > 50 ? 100 - phase0 : phase0) / 50;
int work_len, begin, end, peak = 0, i = *len;
for (work_len = 32; i > 1; work_len <<= 1, i >>= 1);
work = calloc(work_len, sizeof(*work));
for (i = 0; i < *len; ++i) work[i] = (*h)[i];
lsx_safe_rdft(work_len, 1, work); /* Cepstral: */
work[0] = log(fabs(work[0])), work[1] = log(fabs(work[1]));
for (i = 2; i < work_len; i += 2) {
work[i] = log(sqrt(sqr(work[i]) + sqr(work[i + 1])));
work[i + 1] = 0;
}
lsx_safe_rdft(work_len, -1, work);
for (i = 0; i < work_len; ++i) work[i] *= 2. / work_len;
for (i = 1; i < work_len / 2; ++i) { /* Window to reject acausal components */
work[i] *= 2;
work[i + work_len / 2] = 0;
}
lsx_safe_rdft(work_len, 1, work);
/* Some filters require phase unwrapping at this point. Ours give dis-
* continuities only in the stop band, so no need to unwrap in this case. */
for (i = 2; i < work_len; i += 2) /* Interpolate between linear & min phase */
work[i + 1] = phase * M_PI * .5 * i + (1 - phase) * work[i + 1];
work[0] = exp(work[0]), work[1] = exp(work[1]);
for (i = 2; i < work_len; i += 2) {
double x = exp(work[i]);
work[i ] = x * cos(work[i + 1]);
work[i + 1] = x * sin(work[i + 1]);
}
lsx_safe_rdft(work_len, -1, work);
for (i = 0; i < work_len; ++i) work[i] *= 2. / work_len;
for (i = 1; i < work_len; ++i) if (work[i] > work[peak]) /* Find peak pos. */
peak = i; /* N.B. peak val. > 0 */
if (phase == 0)
begin = 0;
else if (phase == 1)
begin = 1 + (work_len - *len) / 2;
else {
if (peak < work_len / 4) { /* Low phases can wrap impulse, so unwrap: */
memmove(work + work_len / 4, work, work_len / 2 * sizeof(*work));
memmove(work, work + work_len * 3 / 4, work_len / 4 * sizeof(*work));
peak += work_len / 4;
}
begin = (.997 - (2 - phase) * .22) * *len + .5;
end = (.997 + (0 - phase) * .22) * *len + .5;
begin = peak - begin - (begin & 1);
end = peak + 1 + end + (end & 1);
*len = end - begin;
*h = realloc(*h, *len * sizeof(**h));
}
for (i = 0; i < *len; ++i)
(*h)[i] = work[begin + (phase0 > 50 ? *len - 1 - i : i)];
*post_len = begin + *len - (peak + 1);
free(work);
}
static void half_band_filter_init(rate_shared_t * p, unsigned which,
int num_taps, sample_t const h[], double Fp, double atten, int multiplier,
double phase, sox_bool allow_aliasing)
{
half_band_t * f = &p->half_band[which];
int dft_length, i;
if (f->num_taps)
return;
if (h) {
dft_length = lsx_set_dft_length(num_taps);
f->coefs = calloc(dft_length, sizeof(*f->coefs));
for (i = 0; i < num_taps; ++i)
f->coefs[(i + dft_length - num_taps + 1) & (dft_length - 1)]
= h[abs(num_taps / 2 - i)] / dft_length * 2 * multiplier;
f->post_peak = num_taps / 2;
}
else {
/* Adjustment to negate att degradation with intermediate phase */
double att = phase && phase != 50 && phase != 100? atten * (34./33) : atten;
double * h = design_lpf(Fp, 1., 2., allow_aliasing, att, &num_taps, 0);
if (phase != 50)
fir_to_phase(&h, &num_taps, &f->post_peak, phase);
else f->post_peak = num_taps / 2;
dft_length = lsx_set_dft_length(num_taps);
f->coefs = calloc(dft_length, sizeof(*f->coefs));
for (i = 0; i < num_taps; ++i)
f->coefs[(i + dft_length - num_taps + 1) & (dft_length - 1)]
= h[i] / dft_length * 2 * multiplier;
free(h);
}
assert(num_taps & 1);
f->num_taps = num_taps;
f->dft_length = dft_length;
lsx_debug("fir_len=%i dft_length=%i Fp=%g atten=%g mult=%i",
num_taps, dft_length, Fp, atten, multiplier);
lsx_safe_rdft(dft_length, 1, f->coefs);
}
#include "rate_filters.h"
typedef struct {
double factor;
size_t samples_in, samples_out;
int level, input_stage_num, output_stage_num;
sox_bool upsample;
stage_t * stages;
} rate_t;
#define pre_stage p->stages[-1]
#define last_stage p->stages[p->level]
#define post_stage p->stages[p->level + 1]
typedef enum {Default = -1, Quick, Low, Medium, High, Very} quality_t;
static void rate_init(rate_t * p, rate_shared_t * shared, double factor,
quality_t quality, int interp_order, double phase, double bandwidth,
sox_bool allow_aliasing)
{
int i, mult, divisor = 1;
assert(factor > 0);
p->factor = factor;
if (quality < Quick || quality > Very)
quality = High;
if (quality != Quick) {
const int max_divisor = 2048; /* Keep coef table size ~< 500kb */
const double epsilon = 4 / MULT32; /* Scaled to half this at max_divisor */
p->upsample = p->factor < 1;
for (i = factor, p->level = 0; i >>= 1; ++p->level); /* log base 2 */
factor /= 1 << (p->level + !p->upsample);
for (i = 2; i <= max_divisor && divisor == 1; ++i) {
double try_d = factor * i;
int try = try_d + .5;
if (fabs(try - try_d) < try * epsilon * (1 - (.5 / max_divisor) * i)) {
if (try == i) /* Rounded to 1:1? */
factor = 1, divisor = 2, p->upsample = sox_false;
else factor = try, divisor = i;
}
}
}
p->stages = (stage_t *)calloc((size_t)p->level + 4, sizeof(*p->stages)) + 1;
for (i = -1; i <= p->level + 1; ++i) p->stages[i].shared = shared;
last_stage.step.all = factor * MULT32 + .5;
last_stage.out_in_ratio = MULT32 * divisor / last_stage.step.all;
if (divisor != 1)
assert(!last_stage.step.parts.fraction);
else if (quality != Quick)
assert(!last_stage.step.parts.integer);
lsx_debug("i/o=%g; %.9g:%i @ level %i", p->factor, factor, divisor, p->level);
mult = 1 + p->upsample; /* Compensate for zero-stuffing in double_sample */
p->input_stage_num = -p->upsample;
p->output_stage_num = p->level;
if (quality == Quick) {
++p->output_stage_num;
last_stage.fn = cubic_spline;
last_stage.pre_post = max(3, last_stage.step.parts.integer);
last_stage.preload = last_stage.pre = 1;
}
else if (last_stage.out_in_ratio != 2 || (p->upsample && quality == Low)) {
poly_fir_t const * f;
poly_fir1_t const * f1;
int n = 4 * p->upsample + range_limit(quality, Medium, Very) - Medium;
if (interp_order < 0)
interp_order = quality > High;
interp_order = divisor == 1? 1 + interp_order : 0;
last_stage.divisor = divisor;
p->output_stage_num += 2;
if (p->upsample && quality == Low)
mult = 1, ++p->input_stage_num, --p->output_stage_num, --n;
f = &poly_firs[n];
f1 = &f->interp[interp_order];
if (!last_stage.shared->poly_fir_coefs) {
int num_taps = 0, phases = divisor == 1? (1 << f1->phase_bits) : divisor;
raw_coef_t * coefs =
design_lpf(f->pass, f->stop, 1., sox_false, f->att, &num_taps, phases);
assert(num_taps == f->num_coefs * phases - 1);
last_stage.shared->poly_fir_coefs =
prepare_coefs(coefs, f->num_coefs, phases, interp_order, mult);
lsx_debug("fir_len=%i phases=%i coef_interp=%i mult=%i size=%s",
f->num_coefs, phases, interp_order, mult,
lsx_sigfigs3((num_taps + 1) * (interp_order + 1) * sizeof(sample_t)));
free(coefs);
}
last_stage.fn = f1->fn;
last_stage.pre_post = f->num_coefs - 1;
last_stage.pre = 0;
last_stage.preload = last_stage.pre_post >> 1;
mult = 1;
}
if (quality > Low) {
typedef struct {int len; sample_t const * h; double bw, a;} filter_t;
static filter_t const filters[] = {
{2 * array_length(half_fir_coefs_low) - 1, half_fir_coefs_low, 0,0},
{0, NULL, .931, 110}, {0, NULL, .931, 125}, {0, NULL, .931, 170}};
filter_t const * f = &filters[quality - Low];
double att = allow_aliasing? (34./33)* f->a : f->a; /* negate att degrade */
double bw = bandwidth? 1 - (1 - bandwidth / 100) / TO_3dB : f->bw;
double min = 1 - (allow_aliasing? MAX_TBW0A : MAX_TBW0) / 100;
assert((size_t)(quality - Low) < array_length(filters));
half_band_filter_init(shared, p->upsample, f->len, f->h, bw, att, mult, phase, allow_aliasing);
if (p->upsample) {
pre_stage.fn = double_sample; /* Finish off setting up pre-stage */
pre_stage.preload = shared->half_band[1].post_peak >> 1;
/* Start setting up post-stage; TODO don't use dft for short filters */
if ((1 - p->factor) / (1 - bw) > 2)
half_band_filter_init(shared, 0, 0, NULL, max(p->factor, min), att, 1, phase, allow_aliasing);
else shared->half_band[0] = shared->half_band[1];
}
else if (p->level > 0 && p->output_stage_num > p->level) {
double pass = bw * divisor / factor / 2;
if ((1 - pass) / (1 - bw) > 2)
half_band_filter_init(shared, 1, 0, NULL, max(pass, min), att, 1, phase, allow_aliasing);
}
post_stage.fn = half_sample;
post_stage.preload = shared->half_band[0].post_peak;
}
else if (quality == Low && !p->upsample) { /* dft is slower here, so */
post_stage.fn = half_sample_low; /* use normal convolution */
post_stage.pre_post = 2 * (array_length(half_fir_coefs_low) - 1);
post_stage.preload = post_stage.pre = post_stage.pre_post >> 1;
}
if (p->level > 0) {
stage_t * s = & p->stages[p->level - 1];
if (shared->half_band[1].num_taps) {
s->fn = half_sample;
s->preload = shared->half_band[1].post_peak;
s->which = 1;
}
else *s = post_stage;
}
for (i = p->input_stage_num; i <= p->output_stage_num; ++i) {
stage_t * s = &p->stages[i];
if (i >= 0 && i < p->level - 1) {
s->fn = half_sample_25;
s->pre_post = 2 * (array_length(half_fir_coefs_25) - 1);
s->preload = s->pre = s->pre_post >> 1;
}
fifo_create(&s->fifo, (int)sizeof(sample_t));
memset(fifo_reserve(&s->fifo, s->preload), 0, sizeof(sample_t)*s->preload);
if (i < p->output_stage_num)
lsx_debug("stage=%-3ipre_post=%-3ipre=%-3ipreload=%i",
i, s->pre_post, s->pre, s->preload);
}
}
static void rate_process(rate_t * p)
{
stage_t * stage = p->stages + p->input_stage_num;
int i;
for (i = p->input_stage_num; i < p->output_stage_num; ++i, ++stage)
stage->fn(stage, &(stage+1)->fifo);
}
static sample_t * rate_input(rate_t * p, sample_t const * samples, size_t n)
{
p->samples_in += n;
return fifo_write(&p->stages[p->input_stage_num].fifo, (int)n, samples);
}
static sample_t const * rate_output(rate_t * p, sample_t * samples, size_t * n)
{
fifo_t * fifo = &p->stages[p->output_stage_num].fifo;
p->samples_out += *n = min(*n, (size_t)fifo_occupancy(fifo));
return fifo_read(fifo, (int)*n, samples);
}
static void rate_flush(rate_t * p)
{
fifo_t * fifo = &p->stages[p->output_stage_num].fifo;
size_t samples_out = p->samples_in / p->factor + .5;
size_t remaining = samples_out - p->samples_out;
sample_t * buff = calloc(1024, sizeof(*buff));
if ((int)remaining > 0) {
while ((size_t)fifo_occupancy(fifo) < remaining) {
rate_input(p, buff, (size_t) 1024);
rate_process(p);
}
fifo_trim_to(fifo, (int)remaining);
p->samples_in = 0;
}
free(buff);
}
static void rate_close(rate_t * p)
{
rate_shared_t * shared = p->stages[0].shared;
int i;
for (i = p->input_stage_num; i <= p->output_stage_num; ++i)
fifo_delete(&p->stages[i].fifo);
free(shared->half_band[0].coefs);
if (shared->half_band[1].coefs != shared->half_band[0].coefs)
free(shared->half_band[1].coefs);
free(shared->poly_fir_coefs);
memset(shared, 0, sizeof(*shared));
free(p->stages - 1);
}
/*------------------------------- SoX Wrapper --------------------------------*/
typedef struct {
sox_rate_t out_rate;
int quality;
double coef_interp, phase, bandwidth;
sox_bool allow_aliasing;
rate_t rate;
rate_shared_t shared, * shared_ptr;
} priv_t;
static int create(sox_effect_t * effp, int argc, char **argv)
{
priv_t * p = (priv_t *) effp->priv;
int c;
char * dummy_p, * found_at, * opts = "+i:b:p:MILasqlmhv", * qopts = opts +12;
p->quality = -1;
p->phase = 25;
p->shared_ptr = &p->shared;
while ((c = getopt(argc, argv, opts)) != -1) switch (c) {
GETOPT_NUMERIC('i', coef_interp, 1 , 3)
GETOPT_NUMERIC('p', phase, 0 , 100)
GETOPT_NUMERIC('b', bandwidth, 100 - MAX_TBW3, 99.7)
case 'M': p->phase = 0; break;
case 'I': p->phase = 25; break;
case 'L': p->phase = 50; break;
case 's': p->bandwidth = 99; break;
case 'a': p->allow_aliasing = sox_true; break;
default: if ((found_at = strchr(qopts, c))) p->quality = found_at - qopts;
else {lsx_fail("unknown option `-%c'", optopt); return lsx_usage(effp);}
}
argc -= optind, argv += optind;
if ((unsigned)p->quality < 2 && (p->bandwidth || p->phase != 25 || p->allow_aliasing)) {
lsx_fail("override options not allowed with this quality level");
return SOX_EOF;
}
if (p->bandwidth && p->bandwidth < 100 - MAX_TBW3A && p->allow_aliasing) {
lsx_fail("minimum allowed bandwidth with aliasing is %g%%", 100 - MAX_TBW3A);
return SOX_EOF;
}
if (argc) {
if ((p->out_rate = lsx_parse_frequency(*argv, &dummy_p)) <= 0 || *dummy_p)
return lsx_usage(effp);
argc--; argv++;
effp->out_signal.rate = p->out_rate;
}
return argc? lsx_usage(effp) : SOX_SUCCESS;
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *) effp->priv;
double out_rate = p->out_rate != 0 ? p->out_rate : effp->out_signal.rate;
if (effp->in_signal.rate == out_rate)
return SOX_EFF_NULL;
effp->out_signal.channels = effp->in_signal.channels;
effp->out_signal.rate = out_rate;
rate_init(&p->rate, p->shared_ptr, effp->in_signal.rate / out_rate,
p->quality, (int)p->coef_interp - 1, p->phase, p->bandwidth, p->allow_aliasing);
return SOX_SUCCESS;
}
static int flow(sox_effect_t * effp, const sox_sample_t * ibuf,
sox_sample_t * obuf, size_t * isamp, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
size_t i, odone = *osamp;
sample_t const * s = rate_output(&p->rate, NULL, &odone);
for (i = 0; i < odone; ++i) *obuf++ = TO_SOX(*s++, effp->clips);
if (*isamp && odone < *osamp) {
sample_t * t = rate_input(&p->rate, NULL, *isamp);
for (i = *isamp; i; --i) *t++ = FROM_SOX(*ibuf++, effp->clips);
rate_process(&p->rate);
}
else *isamp = 0;
*osamp = odone;
return SOX_SUCCESS;
}
static int drain(sox_effect_t * effp, sox_sample_t * obuf, size_t * osamp)
{
priv_t * p = (priv_t *)effp->priv;
static size_t isamp = 0;
rate_flush(&p->rate);
return flow(effp, 0, obuf, &isamp, osamp);
}
static int stop(sox_effect_t * effp)
{
priv_t * p = (priv_t *) effp->priv;
rate_close(&p->rate);
return SOX_SUCCESS;
}
sox_effect_handler_t const * sox_rate_effect_fn(void)
{
static sox_effect_handler_t handler = {
"rate", 0, SOX_EFF_RATE | SOX_EFF_GETOPT,
create, start, flow, drain, stop, 0, sizeof(priv_t)
};
static char const * lines[] = {
"[-q|-l|-m|-h|-v] [override-options] RATE[k]",
" PHASE BAND-",
" QUALITY RESPONSE WIDTH REJ dB TYPICAL USE",
" -q quick linear n/a ~30 @ Fs/4 playback on ancient hardware",
" -l low linear 80% 100 playback on old hardware",
" -m medium interm. 95% 100 audio playback",
" -h high (default) interm. 95% 125 16-bit mastering (use with dither)",
" -v very high interm. 95% 175 24-bit mastering",
" OVERRIDE OPTIONS (only with -m, -h, -v)",
" -M/-I/-L Phase response = minimum/intermediate/linear",
" -s Steep filter (band-width = 99%)",
" -a Allow aliasing above the pass-band",
" -b 74-99.7 Any band-width %",
" -p 0-100 Any phase response (0 = minimum, 25 = intermediate,",
" 50 = linear, 100 = maximum)",
};
static char * usage;
handler.usage = lsx_usage_lines(&usage, lines, array_length(lines));
return &handler;
}