[SCM] calf/master: + monosynth: first "official" release
js at users.alioth.debian.org
js at users.alioth.debian.org
Tue May 7 15:36:45 UTC 2013
The following commit has been merged in the master branch:
commit 8a18988ab0ee5a82560d42881ad7716038faa810
Author: kfoltman <kfoltman at 78b06b96-2940-0410-b7fc-879d825d01d8>
Date: Sat Dec 15 15:19:32 2007 +0000
+ monosynth: first "official" release
git-svn-id: https://calf.svn.sourceforge.net/svnroot/calf/trunk@23 78b06b96-2940-0410-b7fc-879d825d01d8
diff --git a/src/calf/Makefile.am b/src/calf/Makefile.am
index 9247387..6c998d4 100644
--- a/src/calf/Makefile.am
+++ b/src/calf/Makefile.am
@@ -1,3 +1,3 @@
calfdir = $(includedir)/calf
-calf_HEADERS = audio_fx.h benchmark.h biquad.h buffer.h delay.h fft.h fixed_point.h giface.h gui.h inertia.h jackhost.h modules.h modules_dev.h onepole.h organ.h osc.h primitives.h synth.h wave.h
+calf_HEADERS = audio_fx.h benchmark.h biquad.h buffer.h delay.h fft.h fixed_point.h giface.h gui.h inertia.h jackhost.h modules.h modules_dev.h modules_synths.h onepole.h organ.h osc.h primitives.h synth.h wave.h
diff --git a/src/calf/modules.h b/src/calf/modules.h
index b7b5da5..ce431a7 100644
--- a/src/calf/modules.h
+++ b/src/calf/modules.h
@@ -300,4 +300,6 @@ extern std::string get_builtin_modules_rdf();
};
+#include "modules_synths.h"
+
#endif
diff --git a/src/calf/modules_dev.h b/src/calf/modules_dev.h
index 037d2a2..fda0bba 100644
--- a/src/calf/modules_dev.h
+++ b/src/calf/modules_dev.h
@@ -26,368 +26,13 @@
#include <assert.h>
#include "biquad.h"
#include "audio_fx.h"
-#include "inertia.h"
-#include "osc.h"
+#include "synth.h"
#include "organ.h"
namespace synth {
using namespace dsp;
-/// Monosynth-in-making. Parameters may change at any point, so don't make songs with it!
-/// It lacks inertia for parameters, even for those that really need it.
-class monosynth_audio_module: public null_audio_module
-{
-public:
- enum { wave_saw, wave_sqr, wave_pulse, wave_sine, wave_triangle, wave_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_attack, par_decay, par_sustain, par_release, par_keyfollow, par_legato, par_portamento, 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[];
- float *ins[in_count];
- float *outs[out_count];
- float *params[param_count];
- uint32_t srate, crate;
- waveform_family<11> waves[wave_count];
- waveform_oscillator<11> osc1, osc2;
- bool running, stopping, gate;
- int last_key;
-
- float buffer[step_size], buffer2[step_size];
- uint32_t output_pos;
- biquad<float> filter;
- biquad<float> filter2;
- int wave1, wave2, filter_type;
- float freq, start_freq, target_freq, cutoff, decay_factor, fgain, separation;
- float detune, xpose, xfade, pitchbend, ampctl, fltctl, queue_vel;
- float odcr, porta_time;
- int queue_note_on;
- int legato;
- adsr envelope;
-
- static parameter_properties param_props[];
- void set_sample_rate(uint32_t sr) {
- srate = sr;
- crate = sr / step_size;
- odcr = (float)(1.0 / crate);
- }
- void delayed_note_on()
- {
- porta_time = 0.f;
- start_freq = freq;
- target_freq = 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);
-
- if (!running)
- {
- if (legato >= 2)
- porta_time = -1.f;
- osc1.reset();
- osc2.reset();
- filter.reset();
- filter2.reset();
- switch((int)*params[par_oscmode])
- {
- case 1:
- osc2.phase = 0x80000000;
- break;
- case 2:
- osc2.phase = 0x40000000;
- break;
- case 3:
- osc1.phase = osc2.phase = 0x40000000;
- break;
- case 4:
- osc1.phase = 0x40000000;
- osc2.phase = 0xC0000000;
- break;
- case 5:
- // rand() is crap, but I don't have any better RNG in Calf yet
- osc1.phase = rand() << 16;
- osc2.phase = rand() << 16;
- break;
- default:
- break;
- }
- envelope.note_on();
- running = true;
- }
- if (legato >= 2 && !gate)
- porta_time = -1.f;
- gate = true;
- stopping = false;
- if (!(legato & 1) || envelope.released()) {
- envelope.note_on();
- }
- queue_note_on = -1;
- }
- void note_on(int note, int vel)
- {
- queue_note_on = note;
- last_key = note;
- queue_vel = vel / 127.f;
- }
- void note_off(int note, int vel)
- {
- if (note == last_key) {
- gate = false;
- envelope.note_off();
- }
- }
- void pitch_bend(int value)
- {
- pitchbend = pow(2.0, value / 8192.0);
- }
- void set_frequency()
- {
- osc1.set_freq(freq * (2 - detune) * pitchbend, srate);
- osc2.set_freq(freq * (detune) * pitchbend * xpose, srate);
- }
- void params_changed() {
- float sf = 0.001f;
- envelope.set(*params[par_attack] * sf, *params[par_decay] * sf, *params[par_sustain], *params[par_release] * sf, srate / step_size);
- 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);
- xpose = pow(2.0, *params[par_osc2xpose] / 12.0);
- xfade = *params[par_oscmix];
- legato = dsp::fastf2i_drm(*params[par_legato]);
- set_frequency();
- }
- void activate() {
- running = false;
- output_pos = 0;
- queue_note_on = -1;
- pitchbend = 1.f;
- filter.reset();
- filter2.reset();
- float data[2048];
- bandlimiter<11> bl;
-
- // yes these waves don't have really perfect 1/x spectrum because of aliasing
- // (so what?)
- for (int i = 0 ; i < 1024; i++)
- data[i] = (float)(i / 1024.f),
- data[i + 1024] = (float)(i / 1024.f - 1.0f);
- waves[wave_saw].make(bl, data);
-
- for (int i = 0 ; i < 2048; i++)
- data[i] = (float)(i < 1024 ? -1.f : 1.f);
- waves[wave_sqr].make(bl, data);
-
- for (int i = 0 ; i < 2048; i++)
- data[i] = (float)(i < 64 ? -1.f : 1.f);
- waves[wave_pulse].make(bl, data);
-
- // XXXKF sure this is a waste of space, this will be fixed some day by better bandlimiter
- for (int i = 0 ; i < 2048; i++)
- data[i] = (float)sin(i * PI / 1024);
- waves[wave_sine].make(bl, data);
-
- for (int i = 0 ; i < 512; i++) {
- data[i] = i / 512.0,
- data[i + 512] = 1 - i / 512.0,
- data[i + 1024] = - i / 512.0,
- data[i + 1536] = -1 + i / 512.0;
- }
- waves[wave_triangle].make(bl, data);
- }
- 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)
- delayed_note_on();
- else if (stopping)
- {
- running = false;
- dsp::zero(buffer, step_size);
- if (is_stereo_filter())
- dsp::zero(buffer2, step_size);
- return;
- }
- float porta_total_time = *params[par_portamento] * 0.001f;
-
- if (porta_total_time >= 0.00101f && porta_time >= 0) {
- // XXXKF this is criminal, optimize!
- float point = porta_time / porta_total_time;
- if (point >= 1.0f) {
- freq = target_freq;
- porta_time = -1;
- } else {
- freq = start_freq * pow(target_freq / start_freq, point);
- porta_time += odcr;
- }
- }
- set_frequency();
- envelope.advance();
- float env = envelope.value;
- 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;
- 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 * env) * e2r;
- float cutoff2 = dsp::clip(cutoff * separation, 10.f, 18000.f);
- switch(filter_type)
- {
- 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 (envelope.state == adsr::STOP)
- {
- 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;
- }
- }
- uint32_t process(uint32_t offset, uint32_t nsamples, uint32_t inputs_mask, uint32_t outputs_mask) {
- if (!running && queue_note_on == -1)
- return 0;
- uint32_t op = offset;
- uint32_t op_end = offset + nsamples;
- while(op < op_end) {
- if (output_pos == 0) {
- if (running || queue_note_on != -1)
- calculate_step();
- else
- dsp::zero(buffer, step_size);
- }
- if(op < op_end) {
- uint32_t ip = output_pos;
- uint32_t len = std::min(step_size - output_pos, op_end - op);
- 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)
- output_pos = 0;
- }
- }
-
- return 3;
- }
-};
-
struct organ_audio_module: public null_audio_module, public drawbar_organ
{
public:
diff --git a/src/calf/modules_dev.h b/src/calf/modules_synths.h
similarity index 90%
copy from src/calf/modules_dev.h
copy to src/calf/modules_synths.h
index 037d2a2..bb5e0d6 100644
--- a/src/calf/modules_dev.h
+++ b/src/calf/modules_synths.h
@@ -1,5 +1,5 @@
/* Calf DSP Library
- * Prototype audio modules
+ * Audio modules - synthesizers
*
* Copyright (C) 2001-2007 Krzysztof Foltman
*
@@ -18,22 +18,18 @@
* Free Software Foundation, Inc., 59 Temple Place, Suite 330,
* Boston, MA 02111-1307, USA.
*/
-#ifndef __CALF_MODULES_DEV_H
-#define __CALF_MODULES_DEV_H
-
-#if ENABLE_EXPERIMENTAL
+#ifndef __CALF_MODULES_SYNTHS_H
+#define __CALF_MODULES_SYNTHS_H
#include <assert.h>
#include "biquad.h"
#include "audio_fx.h"
#include "inertia.h"
#include "osc.h"
-#include "organ.h"
+#include "synth.h"
namespace synth {
-using namespace dsp;
-
/// Monosynth-in-making. Parameters may change at any point, so don't make songs with it!
/// It lacks inertia for parameters, even for those that really need it.
class monosynth_audio_module: public null_audio_module
@@ -64,7 +60,7 @@ public:
float odcr, porta_time;
int queue_note_on;
int legato;
- adsr envelope;
+ synth::adsr envelope;
static parameter_properties param_props[];
void set_sample_rate(uint32_t sr) {
@@ -388,50 +384,6 @@ public:
}
};
-struct organ_audio_module: public null_audio_module, public drawbar_organ
-{
-public:
- using drawbar_organ::note_on;
- using drawbar_organ::note_off;
- using drawbar_organ::control_change;
- enum { par_drawbar1, par_drawbar2, par_drawbar3, par_drawbar4, par_drawbar5, par_drawbar6, par_drawbar7, par_drawbar8, par_drawbar9, par_foldover,
- par_percmode, par_percharm, par_vibrato, par_master, param_count };
- enum { in_count = 0, out_count = 2, support_midi = true, rt_capable = true };
- static const char *param_names[];
- float *ins[in_count];
- float *outs[out_count];
- float *params[param_count];
- organ_parameters par_values;
- uint32_t srate;
-
- organ_audio_module()
- : drawbar_organ(&par_values)
- {
- }
- static parameter_properties param_props[];
- void set_sample_rate(uint32_t sr) {
- srate = sr;
- }
- void params_changed() {
- for (int i = 0; i < param_count; i++)
- ((float *)&par_values)[i] = *params[i];
- set_vibrato();
- }
- void activate() {
- setup(srate);
- }
- void deactivate() {
- }
- uint32_t process(uint32_t offset, uint32_t nsamples, uint32_t inputs_mask, uint32_t outputs_mask) {
- float *o[2] = { outs[0] + offset, outs[1] + offset };
- render_to(o, nsamples);
- return 3;
- }
-
-};
-
};
#endif
-
-#endif
diff --git a/src/jackhost.cpp b/src/jackhost.cpp
index 68f4d0a..7a50611 100644
--- a/src/jackhost.cpp
+++ b/src/jackhost.cpp
@@ -58,6 +58,7 @@ static struct option long_options[] = {
{"version", 0, 0, 'v'},
{"client", 1, 0, 'c'},
{"effect", 1, 0, 'e'},
+ {"plugin", 1, 0, 'p'},
{"input", 1, 0, 'i'},
{"output", 1, 0, 'o'},
{0,0,0,0},
@@ -74,14 +75,14 @@ int main(int argc, char *argv[])
gtk_init(&argc, &argv);
while(1) {
int option_index;
- int c = getopt_long(argc, argv, "c:e:i:o:m:hv", long_options, &option_index);
+ int c = getopt_long(argc, argv, "c:e:i:o:m:p:hv", long_options, &option_index);
if (c == -1)
break;
switch(c) {
case 'h':
case '?':
printf("JACK host for Calf effects\n"
- "Syntax: %s [--effect reverb|flanger|filter] [--client <name>] [--input <name>]"
+ "Syntax: %s [--plugin reverb|flanger|filter|monosynth] [--client <name>] [--input <name>]"
" [--output <name>] [--midi <name>] [--help] [--version]\n",
argv[0]);
return 0;
@@ -89,6 +90,7 @@ int main(int argc, char *argv[])
printf("%s\n", PACKAGE_STRING);
return 0;
case 'e':
+ case 'p':
effect_name = optarg;
break;
case 'c':
@@ -113,17 +115,17 @@ int main(int argc, char *argv[])
jh = new jack_host<flanger_audio_module>();
else if (!strcmp(effect_name, "filter"))
jh = new jack_host<filter_audio_module>();
-#ifdef ENABLE_EXPERIMENTAL
else if (!strcmp(effect_name, "monosynth"))
jh = new jack_host<monosynth_audio_module>();
+#ifdef ENABLE_EXPERIMENTAL
else if (!strcmp(effect_name, "organ"))
jh = new jack_host<organ_audio_module>();
#endif
else {
#ifdef ENABLE_EXPERIMENTAL
- fprintf(stderr, "Unknown filter name; allowed are: reverb, flanger, filter, organ, monosynth\n");
+ fprintf(stderr, "Unknown plugin name; allowed are: reverb, flanger, filter, monosynth, organ\n");
#else
- fprintf(stderr, "Unknown filter name; allowed are: reverb, flanger, filter\n");
+ fprintf(stderr, "Unknown plugin name; allowed are: reverb, flanger, filter, monosynth\n");
#endif
return 1;
}
diff --git a/src/modules.cpp b/src/modules.cpp
index 13a0fe5..42bd88b 100644
--- a/src/modules.cpp
+++ b/src/modules.cpp
@@ -138,7 +138,6 @@ synth::ladspa_wrapper<organ_audio_module> organ(organ_info);
#endif
////////////////////////////////////////////////////////////////////////////
-#ifdef ENABLE_EXPERIMENTAL
const char *monosynth_audio_module::param_names[] = {
"Out L", "Out R",
@@ -196,8 +195,6 @@ static synth::ladspa_info monosynth_info = { 0x8480, "Monosynth", "Calf Monosynt
synth::ladspa_wrapper<monosynth_audio_module> monosynth(monosynth_info);
#endif
-#endif
-
////////////////////////////////////////////////////////////////////////////
#if USE_LADSPA
@@ -215,17 +212,19 @@ const LADSPA_Descriptor *ladspa_descriptor(unsigned long Index)
};
-#if USE_DSSI && ENABLE_EXPERIMENTAL
+#if USE_DSSI
extern "C" {
const DSSI_Descriptor *dssi_descriptor(unsigned long Index)
{
switch (Index) {
- case 0: return &::monosynth.dssi_descriptor;
- case 1: return &::organ.dssi_descriptor;
- case 2: return &::filter.dssi_descriptor;
- case 3: return &::flanger.dssi_descriptor;
- case 4: return &::reverb.dssi_descriptor;
+ case 0: return &::filter.dssi_descriptor;
+ case 1: return &::flanger.dssi_descriptor;
+ case 2: return &::reverb.dssi_descriptor;
+ case 3: return &::monosynth.dssi_descriptor;
+#ifdef ENABLE_EXPERIMENTAL
+ case 4: return &::organ.dssi_descriptor;
+#endif
default: return NULL;
}
}
@@ -240,6 +239,7 @@ std::string synth::get_builtin_modules_rdf()
rdf += ::flanger.generate_rdf();
rdf += ::reverb.generate_rdf();
rdf += ::filter.generate_rdf();
+ rdf += ::monosynth.generate_rdf();
return rdf;
}
--
calf audio plugins packaging
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