[SCM] calf/master: + Added velocity sensitivity (env.mod. and amplitude) to monosynth + Added filter types (including stereo filters with separation) to monosynth
js at users.alioth.debian.org
js at users.alioth.debian.org
Tue May 7 15:36:43 UTC 2013
The following commit has been merged in the master branch:
commit 428ee885abc544564951bbfd072c541347652554
Author: kfoltman <kfoltman at 78b06b96-2940-0410-b7fc-879d825d01d8>
Date: Tue Dec 11 22:21:27 2007 +0000
+ Added velocity sensitivity (env.mod. and amplitude) to monosynth
+ Added filter types (including stereo filters with separation) to monosynth
git-svn-id: https://calf.svn.sourceforge.net/svnroot/calf/trunk@14 78b06b96-2940-0410-b7fc-879d825d01d8
diff --git a/src/calf/modules_dev.h b/src/calf/modules_dev.h
index 9687eea..54cb481 100644
--- a/src/calf/modules_dev.h
+++ b/src/calf/modules_dev.h
@@ -40,7 +40,8 @@ class monosynth_audio_module
{
public:
enum { wave_saw, wave_sqr, wave_pulse, wave_sine, wave_triangle, wave_count };
- enum { par_wave1, par_wave2, par_detune, par_osc2xpose, par_oscmode, par_oscmix, par_cutoff, par_resonance, par_envmod, par_envtores, par_decay, par_keyfollow, par_legato, param_count };
+ enum { flt_lp12, flt_lp24, flt_2lp12, flt_hp12, flt_lpbr, flt_hpbr, flt_bp6, flt_2bp6 };
+ enum { par_wave1, par_wave2, par_detune, par_osc2xpose, par_oscmode, par_oscmix, par_filtertype, par_cutoff, par_resonance, par_cutoffsep, par_envmod, par_envtores, par_decay, par_keyfollow, par_legato, par_vel2amp, par_vel2filter, param_count };
enum { in_count = 0, out_count = 2, support_midi = true, rt_capable = true };
enum { step_size = 64 };
static const char *param_names[];
@@ -53,13 +54,13 @@ public:
bool running, stopping, gate;
int last_key;
- float buffer[step_size];
+ float buffer[step_size], buffer2[step_size];
uint32_t output_pos;
biquad<float> filter;
biquad<float> filter2;
- int wave1, wave2;
- float freq, cutoff, decay_factor;
- float detune, xpose, xfade, pitchbend;
+ int wave1, wave2, filter_type;
+ float freq, cutoff, decay_factor, fgain, separation;
+ float detune, xpose, xfade, pitchbend, ampctl, fltctl, queue_vel;
int voice_age;
float odcr;
int queue_note_on;
@@ -74,6 +75,8 @@ public:
void note_on()
{
freq = 440 * pow(2.0, (queue_note_on - 69) / 12.0);
+ ampctl = 1.0 + (queue_vel - 1.0) * *params[par_vel2amp];
+ fltctl = 1.0 + (queue_vel - 1.0) * *params[par_vel2filter];
set_frequency();
osc1.waveform = waves[wave1].get_level(osc1.phasedelta);
osc2.waveform = waves[wave2].get_level(osc2.phasedelta);
@@ -123,6 +126,7 @@ public:
if (data[2]) {
queue_note_on = data[1];
last_key = data[1];
+ queue_vel = data[2] / 127.f;
}
// printf("note on %d %d\n", data[1], data[2]);
break;
@@ -144,7 +148,9 @@ public:
osc2.set_freq(freq * (detune) * pitchbend * xpose, srate);
}
void params_changed() {
+ filter_type = fastf2i_drm(*params[par_filtertype]);
decay_factor = odcr * 1000.0 / *params[par_decay];
+ separation = pow(2.0, *params[par_cutoffsep] / 1200.0);
wave1 = dsp::clip(dsp::fastf2i_drm(*params[par_wave1]), 0, (int)wave_count - 1);
wave2 = dsp::clip(dsp::fastf2i_drm(*params[par_wave2]), 0, (int)wave_count - 1);
detune = pow(2.0, *params[par_detune] / 1200.0);
@@ -193,6 +199,57 @@ public:
}
void deactivate() {
}
+ inline float softclip(float wave) const
+ {
+ float abswave = fabs(wave);
+ if (abswave > 0.75)
+ {
+ abswave = abswave - 0.5 * (abswave - 0.75);
+ if (abswave > 1.0)
+ abswave = 1.0;
+ wave = (wave > 0.0) ? abswave : - abswave;
+ }
+ return wave;
+ }
+ void calculate_buffer_ser()
+ {
+ for (uint32_t i = 0; i < step_size; i++)
+ {
+ float osc1val = osc1.get();
+ float osc2val = osc2.get();
+ float wave = fgain * (osc1val + (osc2val - osc1val) * xfade);
+ wave = filter.process_d1(wave);
+ wave = filter2.process_d1(wave);
+ buffer[i] = softclip(wave);
+ }
+ }
+ void calculate_buffer_single()
+ {
+ for (uint32_t i = 0; i < step_size; i++)
+ {
+ float osc1val = osc1.get();
+ float osc2val = osc2.get();
+ float wave = fgain * (osc1val + (osc2val - osc1val) * xfade);
+ wave = filter.process_d1(wave);
+ buffer[i] = softclip(wave);
+ }
+ }
+ void calculate_buffer_stereo()
+ {
+ for (uint32_t i = 0; i < step_size; i++)
+ {
+ float osc1val = osc1.get();
+ float osc2val = osc2.get();
+ float wave1 = osc1val + (osc2val - osc1val) * xfade;
+ float wave2 = osc1val + ((-osc2val) - osc1val) * xfade;
+ buffer[i] = softclip(fgain * filter.process_d1(wave1));
+ buffer2[i] = softclip(fgain * filter2.process_d1(wave2));
+ }
+ }
+ bool is_stereo_filter() const
+ {
+ return filter_type == flt_2lp12 || filter_type == flt_2bp6;
+ }
void calculate_step() {
if (queue_note_on != -1)
note_on();
@@ -200,46 +257,84 @@ public:
{
running = false;
dsp::zero(buffer, step_size);
+ if (is_stereo_filter())
+ dsp::zero(buffer2, step_size);
return;
}
set_frequency();
float env = max(0.f, 1.f - voice_age * decay_factor);
- cutoff = *params[par_cutoff] * pow(2.0f, env * *params[par_envmod] * (1.f / 1200.f));
+ cutoff = *params[par_cutoff] * pow(2.0f, env * fltctl * *params[par_envmod] * (1.f / 1200.f));
if (*params[par_keyfollow] >= 0.5f)
cutoff *= freq / 264.0f;
- if (cutoff < 10.f)
- cutoff = 10.f;
- if (cutoff > 16000.f)
- cutoff = 16000.f;
+ cutoff = dsp::clip(cutoff , 10.f, 18000.f);
float resonance = *params[par_resonance];
float e2r = *params[par_envtores];
resonance = resonance * (1 - e2r) + (0.7 + (resonance - 0.7) * env) * e2r;
- filter.set_lp_rbj(cutoff, resonance, srate);
- float fgain = min(0.5, 0.5 / resonance);
- filter2.copy_coeffs(filter);
- for (uint32_t i = 0; i < step_size; i++)
+ float cutoff2 = dsp::clip(cutoff * separation, 10.f, 18000.f);
+ switch(filter_type)
{
- float osc1val = osc1.get();
- float osc2val = osc2.get();
- float wave = fgain * (osc1val + (osc2val - osc1val) * xfade);
- wave = filter.process_d1(wave);
- wave = filter2.process_d1(wave);
- // primitive waveshaping (hard-knee)
- float abswave = fabs(wave);
- if (abswave > 0.75)
- {
- abswave = abswave - 0.5 * (abswave - 0.75);
- if (abswave > 1.0)
- abswave = 1.0;
- wave = (wave > 0.0) ? abswave : - abswave;
-
- }
- buffer[i] = dsp::clip(wave, -1.f, +1.f);
+ case flt_lp12:
+ filter.set_lp_rbj(cutoff, resonance, srate);
+ fgain = min(0.7f, 0.7f / resonance) * ampctl;
+ break;
+ case flt_hp12:
+ filter.set_hp_rbj(cutoff, resonance, srate);
+ fgain = min(0.7f, 0.7f / resonance) * ampctl;
+ break;
+ case flt_lp24:
+ filter.set_lp_rbj(cutoff, resonance, srate);
+ filter2.set_lp_rbj(cutoff2, resonance, srate);
+ fgain = min(0.5f, 0.5f / resonance) * ampctl;
+ break;
+ case flt_lpbr:
+ filter.set_lp_rbj(cutoff, resonance, srate);
+ filter2.set_br_rbj(cutoff2, resonance, srate);
+ fgain = min(0.5f, 0.5f / resonance) * ampctl;
+ break;
+ case flt_hpbr:
+ filter.set_hp_rbj(cutoff, resonance, srate);
+ filter2.set_br_rbj(cutoff2, resonance, srate);
+ fgain = min(0.5f, 0.5f / resonance) * ampctl;
+ break;
+ case flt_2lp12:
+ filter.set_lp_rbj(cutoff, resonance, srate);
+ filter2.set_lp_rbj(cutoff2, resonance, srate);
+ fgain = min(0.7f, 0.7f / resonance) * ampctl;
+ break;
+ case flt_bp6:
+ filter.set_bp_rbj(cutoff, resonance, srate);
+ fgain = ampctl;
+ break;
+ case flt_2bp6:
+ filter.set_bp_rbj(cutoff, resonance, srate);
+ filter2.set_bp_rbj(cutoff2, resonance, srate);
+ fgain = ampctl;
+ break;
+ }
+ switch(filter_type)
+ {
+ case flt_lp24:
+ case flt_lpbr:
+ case flt_hpbr: // Oomek's wish
+ calculate_buffer_ser();
+ break;
+ case flt_lp12:
+ case flt_hp12:
+ case flt_bp6:
+ calculate_buffer_single();
+ break;
+ case flt_2lp12:
+ case flt_2bp6:
+ calculate_buffer_stereo();
+ break;
}
if (!gate)
{
for (int i = 0; i < step_size; i++)
buffer[i] *= (step_size - i) * (1.0f / step_size);
+ if (is_stereo_filter())
+ for (int i = 0; i < step_size; i++)
+ buffer2[i] *= (step_size - i) * (1.0f / step_size);
stopping = true;
}
@@ -260,8 +355,13 @@ public:
if(op < op_end) {
uint32_t ip = output_pos;
uint32_t len = std::min(step_size - output_pos, op_end - op);
- for(uint32_t i = 0 ; i < len; i++)
- outs[0][op + i] = outs[1][op + i] = buffer[ip + i];
+ if (is_stereo_filter())
+ for(uint32_t i = 0 ; i < len; i++)
+ outs[0][op + i] = buffer[ip + i],
+ outs[1][op + i] = buffer2[ip + i];
+ else
+ for(uint32_t i = 0 ; i < len; i++)
+ outs[0][op + i] = outs[1][op + i] = buffer[ip + i];
op += len;
output_pos += len;
if (output_pos == step_size)
diff --git a/src/modules.cpp b/src/modules.cpp
index d85d84d..f752085 100644
--- a/src/modules.cpp
+++ b/src/modules.cpp
@@ -140,9 +140,22 @@ synth::ladspa_wrapper<organ_audio_module> organ(organ_info);
////////////////////////////////////////////////////////////////////////////
#ifdef ENABLE_EXPERIMENTAL
+const char *monosynth_audio_module::param_names[] = {"Out L", "Out R", "Osc1 Wave", "Osc2 Wave", "Osc 1/2 Detune", "Osc 2 Transpose", "Phase Mode", "Osc Mix", "Filter", "Cutoff", "Resonance", "Separation", "Env->Cutoff", "Env->Res", "Decay", "Key Follow", "Legato", "Vel->Amp", "Vel->Filter"};
+
const char *monosynth_waveform_names[] = { "Sawtooth", "Square", "Pulse", "Sine", "Triangle" };
const char *monosynth_mode_names[] = { "0 : 0", "0 : 180", "0 : 90", "90 : 90", "90 : 270", "Random" };
+const char *monosynth_filter_choices[] = {
+ "12dB/oct Lowpass",
+ "24dB/oct Lowpass",
+ "2x12dB/oct Lowpass",
+ "12dB/oct Highpass",
+ "Lowpass+Notch",
+ "Highpass+Notch",
+ "6dB/oct Bandpass",
+ "2x6dB/oct Bandpass",
+};
+
parameter_properties monosynth_audio_module::param_props[] = {
{ wave_saw, 0,wave_count - 1, 1, PF_ENUM | PF_CTL_COMBO, monosynth_waveform_names },
{ wave_pulse, 0,wave_count - 1, 1, PF_ENUM | PF_CTL_COMBO, monosynth_waveform_names },
@@ -150,17 +163,19 @@ parameter_properties monosynth_audio_module::param_props[] = {
{ 12, -24, 24, 1.01, PF_INT | PF_SCALE_LINEAR | PF_CTL_KNOB | PF_UNIT_SEMITONES, NULL },
{ 0, 0, 5, 1.01, PF_ENUM | PF_CTL_COMBO, monosynth_mode_names },
{ 0.5, 0, 1, 1.01, PF_FLOAT | PF_SCALE_PERC, NULL },
+ { 0, 0, 7, 1.01, PF_ENUM | PF_CTL_COMBO, monosynth_filter_choices },
{ 33, 10,16000, 1.01, PF_FLOAT | PF_SCALE_LOG | PF_CTL_KNOB | PF_UNIT_HZ, NULL },
{ 2, 0.7, 8, 1.01, PF_FLOAT | PF_SCALE_LOG | PF_CTL_KNOB, NULL },
+ { 0, -2400, 2400, 1.01, PF_FLOAT | PF_SCALE_LINEAR | PF_CTL_KNOB | PF_UNIT_CENTS, NULL },
{ 8000, -10800,10800, 1.01, PF_FLOAT | PF_SCALE_LINEAR | PF_CTL_KNOB | PF_UNIT_CENTS, NULL },
{ 0.5, 0, 1, 1.01, PF_FLOAT | PF_SCALE_LINEAR | PF_CTL_KNOB, NULL },
{ 350, 10,20000, 1.01, PF_FLOAT | PF_SCALE_LOG | PF_CTL_KNOB | PF_UNIT_MSEC, NULL },
{ 0, 0, 1, 1.01, PF_BOOL | PF_CTL_TOGGLE, NULL },
{ 0, 0, 1, 1.01, PF_BOOL | PF_CTL_TOGGLE, NULL },
+ { 0, 0, 1, 0.1, PF_FLOAT | PF_SCALE_PERC | PF_CTL_KNOB, NULL },
+ { 0, 0, 1, 0.1, PF_FLOAT | PF_SCALE_PERC | PF_CTL_KNOB, NULL },
};
-const char *monosynth_audio_module::param_names[] = {"Out L", "Out R", "Osc1 Wave", "Osc2 Wave", "Osc 1/2 Detune", "Osc 2 Transpose", "Phase Mode", "Osc Mix", "Cutoff", "Resonance", "Env->Cutoff", "Env->Res", "Decay", "Key Follow", "Legato"};
-
static synth::ladspa_info monosynth_info = { 0x8480, "Monosynth", "Calf Monosynth", "Krzysztof Foltman", copyright, "SynthesizerPlugin" };
#if USE_LADSPA
--
calf audio plugins packaging
More information about the pkg-multimedia-commits
mailing list