sampleCount EffectReverb::ProcessBlock(float **inBlock, float **outBlock, sampleCount blockLen)
{
float *ichans[2] = {NULL, NULL};
float *ochans[2] = {NULL, NULL};
for (int c = 0; c < mNumChans; c++)
{
ichans[c] = inBlock[c];
ochans[c] = outBlock[c];
}
float const dryMult = mParams.mWetOnly ? 0 : dB_to_linear(mParams.mDryGain);
sampleCount remaining = blockLen;
while (remaining)
{
sampleCount len = wxMin(remaining, BLOCK);
for (int c = 0; c < mNumChans; c++)
{
// Write the input samples to the reverb fifo. Returned value is the address of the
// fifo buffer which contains a copy of the input samples.
mP[c].dry = (float *) fifo_write(&mP[c].reverb.input_fifo, len, ichans[c]);
reverb_process(&mP[c].reverb, len);
}
if (mNumChans == 2)
{
for (sampleCount i = 0; i < len; i++)
{
for (int w = 0; w < 2; w++)
{
ochans[w][i] = dryMult *
mP[w].dry[i] +
0.5 *
(mP[0].wet[w][i] + mP[1].wet[w][i]);
}
}
}
else
{
for (sampleCount i = 0; i < len; i++)
{
ochans[0][i] = dryMult *
mP[0].dry[i] +
mP[0].wet[0][i];
}
}
remaining -= len;
for (int c = 0; c < mNumChans; c++)
{
ichans[c] += len;
ochans[c] += len;
}
}
return blockLen;
}
static void *_init(void *arg) {
_t *data = (_t *)arg;
void *p = noise_init(&data->noise); if (p != &data->noise) return p;
noise_configure(&data->noise, &data->opts.n);
data->gain = dB_to_linear(data->opts.dBgain);
return arg;
}
static int _command(ClientData clientData, Tcl_Interp *interp, int argc, Tcl_Obj* const *objv) {
_t *data = (_t *)clientData;
options_t save = data->opts;
if (framework_command(clientData, interp, argc, objv) != TCL_OK) {
data->opts = save;
return TCL_ERROR;
}
if (data->opts.dBgain != save.dBgain) {
data->gain = dB_to_linear(data->opts.dBgain);
}
return TCL_OK;
}
pa_volume_t pa_sw_volume_from_dB(double dB) {
if (isinf(dB) < 0 || dB <= PA_DECIBEL_MININFTY)
return PA_VOLUME_MUTED;
return pa_sw_volume_from_linear(dB_to_linear(dB));
}
static int start(sox_effect_t * effp)
{
priv_t * p = (priv_t *)effp->priv;
double w0, A, alpha, mult;
if (p->filter_type == filter_deemph) { /* See deemph.plt for documentation */
if (effp->in_signal.rate == 44100) {
p->fc = 5283;
p->width = 0.4845;
p->gain = -9.477;
}
else if (effp->in_signal.rate == 48000) {
p->fc = 5356;
p->width = 0.479;
p->gain = -9.62;
}
else {
lsx_fail("sample rate must be 44100 (audio-CD) or 48000 (DAT)");
return SOX_EOF;
}
}
w0 = 2 * M_PI * p->fc / effp->in_signal.rate;
A = exp(p->gain / 40 * log(10.));
alpha = 0, mult = dB_to_linear(max(p->gain, 0));
if (w0 > M_PI) {
lsx_fail("frequency must be less than half the sample-rate (Nyquist rate)");
return SOX_EOF;
}
/* Set defaults: */
p->b0 = p->b1 = p->b2 = p->a1 = p->a2 = 0;
p->a0 = 1;
if (p->width) switch (p->width_type) {
case width_slope:
alpha = sin(w0)/2 * sqrt((A + 1/A)*(1/p->width - 1) + 2);
break;
case width_Q:
alpha = sin(w0)/(2*p->width);
break;
case width_bw_oct:
alpha = sin(w0)*sinh(log(2.)/2 * p->width * w0/sin(w0));
break;
case width_bw_Hz:
alpha = sin(w0)/(2*p->fc/p->width);
break;
case width_bw_kHz: assert(0); /* Shouldn't get here */
case width_bw_old:
alpha = tan(M_PI * p->width / effp->in_signal.rate);
break;
}
switch (p->filter_type) {
case filter_LPF: /* H(s) = 1 / (s^2 + s/Q + 1) */
p->b0 = (1 - cos(w0))/2;
p->b1 = 1 - cos(w0);
p->b2 = (1 - cos(w0))/2;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_HPF: /* H(s) = s^2 / (s^2 + s/Q + 1) */
p->b0 = (1 + cos(w0))/2;
p->b1 = -(1 + cos(w0));
p->b2 = (1 + cos(w0))/2;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_BPF_CSG: /* H(s) = s / (s^2 + s/Q + 1) (constant skirt gain, peak gain = Q) */
p->b0 = sin(w0)/2;
p->b1 = 0;
p->b2 = -sin(w0)/2;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_BPF: /* H(s) = (s/Q) / (s^2 + s/Q + 1) (constant 0 dB peak gain) */
p->b0 = alpha;
p->b1 = 0;
p->b2 = -alpha;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_notch: /* H(s) = (s^2 + 1) / (s^2 + s/Q + 1) */
p->b0 = 1;
p->b1 = -2*cos(w0);
p->b2 = 1;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_APF: /* H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1) */
p->b0 = 1 - alpha;
p->b1 = -2*cos(w0);
p->b2 = 1 + alpha;
p->a0 = 1 + alpha;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha;
break;
case filter_peakingEQ: /* H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1) */
if (A == 1)
return SOX_EFF_NULL;
p->b0 = 1 + alpha*A;
p->b1 = -2*cos(w0);
p->b2 = 1 - alpha*A;
p->a0 = 1 + alpha/A;
p->a1 = -2*cos(w0);
p->a2 = 1 - alpha/A;
break;
case filter_lowShelf: /* H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1) */
if (A == 1)
return SOX_EFF_NULL;
p->b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha );
p->b1 = 2*A*( (A-1) - (A+1)*cos(w0) );
p->b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha );
p->a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha;
p->a1 = -2*( (A-1) + (A+1)*cos(w0) );
p->a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha;
break;
case filter_deemph: /* Falls through to high-shelf... */
case filter_highShelf: /* H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A) */
if (!A)
return SOX_EFF_NULL;
p->b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha );
p->b1 = -2*A*( (A-1) + (A+1)*cos(w0) );
p->b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha );
p->a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha;
p->a1 = 2*( (A-1) - (A+1)*cos(w0) );
p->a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha;
break;
case filter_LPF_1: /* single-pole */
p->a1 = -exp(-w0);
p->b0 = 1 + p->a1;
break;
case filter_HPF_1: /* single-pole */
p->a1 = -exp(-w0);
p->b0 = (1 - p->a1)/2;
p->b1 = -p->b0;
break;
case filter_BPF_SPK: case filter_BPF_SPK_N: {
double bw_Hz;
if (!p->width)
p->width = p->fc / 2;
bw_Hz = p->width_type == width_Q? p->fc / p->width :
p->width_type == width_bw_Hz? p->width :
p->fc * (pow(2., p->width) - 1) * pow(2., -0.5 * p->width); /* bw_oct */
#include "band.h" /* Has different licence */
break;
}
case filter_AP1: /* Experimental 1-pole all-pass from Tom Erbe @ UCSD */
p->b0 = exp(-w0);
p->b1 = -1;
p->a1 = -exp(-w0);
break;
case filter_AP2: /* Experimental 2-pole all-pass from Tom Erbe @ UCSD */
p->b0 = 1 - sin(w0);
p->b1 = -2 * cos(w0);
p->b2 = 1 + sin(w0);
p->a0 = 1 + sin(w0);
p->a1 = -2 * cos(w0);
p->a2 = 1 - sin(w0);
break;
case filter_riaa: /* http://www.dsprelated.com/showmessage/73300/3.php */
if (effp->in_signal.rate == 44100) {
static const double zeros[] = {-0.2014898, 0.9233820};
static const double poles[] = {0.7083149, 0.9924091};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else if (effp->in_signal.rate == 48000) {
static const double zeros[] = {-0.1766069, 0.9321590};
static const double poles[] = {0.7396325, 0.9931330};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else if (effp->in_signal.rate == 88200) {
static const double zeros[] = {-0.1168735, 0.9648312};
static const double poles[] = {0.8590646, 0.9964002};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else if (effp->in_signal.rate == 96000) {
static const double zeros[] = {-0.1141486, 0.9676817};
static const double poles[] = {0.8699137, 0.9966946};
make_poly_from_roots(zeros, (size_t)2, &p->b0);
make_poly_from_roots(poles, (size_t)2, &p->a0);
}
else {
lsx_fail("Sample rate must be 44.1k, 48k, 88.2k, or 96k");
return SOX_EOF;
}
{ /* Normalise to 0dB at 1kHz (Thanks to Glenn Davis) */
double y = 2 * M_PI * 1000 / effp->in_signal.rate;
double b_re = p->b0 + p->b1 * cos(-y) + p->b2 * cos(-2 * y);
double a_re = p->a0 + p->a1 * cos(-y) + p->a2 * cos(-2 * y);
double b_im = p->b1 * sin(-y) + p->b2 * sin(-2 * y);
double a_im = p->a1 * sin(-y) + p->a2 * sin(-2 * y);
double g = 1 / sqrt((sqr(b_re) + sqr(b_im)) / (sqr(a_re) + sqr(a_im)));
p->b0 *= g; p->b1 *= g; p->b2 *= g;
}
mult = (p->b0 + p->b1 + p->b2) / (p->a0 + p->a1 + p->a2);
lsx_debug("gain=%f", linear_to_dB(mult));
break;
}
if (effp->in_signal.mult)
*effp->in_signal.mult /= mult;
return lsx_biquad_start(effp);
}
