ardour/livetrax
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update amp for negative (polarity-invert) gain

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fixes monitor-section “inv”
This commit is contained in:
Robin Gareus 2015-04-26 02:30:06 +02:00
parent a43901ead9
commit 1b54b85da0
1 changed files with 19 additions and 16 deletions
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35
libs/ardour/amp.cc
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@ -35,6 +35,9 @@
using namespace ARDOUR;
using namespace PBD;
// used for low-pass filter denormal protection
#define GAIN_COEFF_TINY (1e-10) // -200dB
Amp::Amp (Session& s, std::string type)
: Processor(s, "Amp")
, _apply_gain(true)
@ -104,8 +107,8 @@ Amp::run (BufferSet& bufs, framepos_t /*start_frame*/, framepos_t /*end_frame*/,
}
const float a = 62.78 / _session.nominal_frame_rate(); // 10 Hz LPF
float lpf = _current_gain;
const double a = 62.78 / _session.nominal_frame_rate(); // 10 Hz LPF; see Amp::apply_gain for details
double lpf = _current_gain;
for (BufferSet::audio_iterator i = bufs.audio_begin(); i != bufs.audio_end(); ++i) {
Sample* const sp = i->data();
@ -116,8 +119,8 @@ Amp::run (BufferSet& bufs, framepos_t /*start_frame*/, framepos_t /*end_frame*/,
}
}
if (lpf < 1e-10) {
_current_gain = 0;
if (fabs (lpf) < GAIN_COEFF_TINY) {
_current_gain = GAIN_COEFF_ZERO;
} else {
_current_gain = lpf;
}
@ -207,11 +210,11 @@ Amp::apply_gain (BufferSet& bufs, framecnt_t sample_rate, framecnt_t nframes, ga
/* Low pass filter coefficient: 1.0 - e^(-2.0 * π * f / 48000) f in Hz.
* for f << SR, approx a ~= 6.2 * f / SR;
*/
const float a = 62.78 / sample_rate; // 10 Hz LPF
const double a = 62.78 / sample_rate; // 10 Hz LPF
for (BufferSet::audio_iterator i = bufs.audio_begin(); i != bufs.audio_end(); ++i) {
Sample* const buffer = i->data();
float lpf = initial;
double lpf = initial;
for (pframes_t nx = 0; nx < nframes; ++nx) {
buffer[nx] *= lpf;
@ -221,9 +224,8 @@ Amp::apply_gain (BufferSet& bufs, framecnt_t sample_rate, framecnt_t nframes, ga
rv = lpf;
}
}
// 1e-10 ~ 200dB, prevent denormals.
if (rv < 1e-10) return 0;
if (fabsf(rv - 1.0) < 1e-10) return 1.0;
if (fabsf (rv - target) < GAIN_COEFF_TINY) return target;
if (fabsf (rv) < GAIN_COEFF_TINY) return GAIN_COEFF_ZERO;
return rv;
}
@ -287,23 +289,23 @@ Amp::apply_gain (AudioBuffer& buf, framecnt_t sample_rate, framecnt_t nframes, g
}
Sample* const buffer = buf.data();
const float a = 62.78 / sample_rate; // 10 Hz LPF, see [other] Amp::apply_gain() above,
const double a = 62.78 / sample_rate; // 10 Hz LPF, see [other] Amp::apply_gain() above,
float lpf = initial;
double lpf = initial;
for (pframes_t nx = 0; nx < nframes; ++nx) {
buffer[nx] *= lpf;
lpf += a * (target - lpf);
}
if (lpf < 1e-10) return 0; // TODO use GAIN_COEFF_TINY or _DENORMAL
if (fabsf(lpf - GAIN_COEFF_UNITY) < 1e-10) return GAIN_COEFF_UNITY;
if (fabs (lpf - target) < GAIN_COEFF_TINY) return target;
if (fabs (lpf) < GAIN_COEFF_TINY) return GAIN_COEFF_ZERO;
return lpf;
}
void
Amp::apply_simple_gain (BufferSet& bufs, framecnt_t nframes, gain_t target, bool midi_amp)
{
if (target < GAIN_COEFF_SMALL) {
if (fabsf (target) < GAIN_COEFF_SMALL) {
if (midi_amp) {
/* don't Trim midi velocity -- only relevant for Midi on Audio tracks */
@ -348,7 +350,7 @@ Amp::apply_simple_gain (BufferSet& bufs, framecnt_t nframes, gain_t target, bool
void
Amp::apply_simple_gain (AudioBuffer& buf, framecnt_t nframes, gain_t target)
{
if (target < GAIN_COEFF_SMALL) {
if (fabsf (target) < GAIN_COEFF_SMALL) {
memset (buf.data(), 0, sizeof (Sample) * nframes);
} else if (target != GAIN_COEFF_UNITY) {
apply_gain_to_buffer (buf.data(), nframes, target);
@ -360,7 +362,8 @@ Amp::inc_gain (gain_t factor, void *src)
{
float desired_gain = _gain_control->user_double();
if (desired_gain < GAIN_COEFF_SMALL) {
if (fabsf (desired_gain) < GAIN_COEFF_SMALL) {
// really?! what's the idea here?
set_gain (0.000001f + (0.000001f * factor), src);
} else {
set_gain (desired_gain + (desired_gain * factor), src);