add reasonablesynth.lv2
A reasonably simple synth to allow new users to 'hear midi'. This is a first step. It still needs proper install and bundling.
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/* reasonable simple synth
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*
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* Copyright (C) 2013 Robin Gareus <robin@gareus.org>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2, or (at your option)
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* any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software Foundation,
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* Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#define _GNU_SOURCE
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#include <stdio.h>
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#include <stdlib.h>
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#include <stdint.h>
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/* LV2 */
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#include "lv2/lv2plug.in/ns/lv2core/lv2.h"
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#include "lv2/lv2plug.in/ns/ext/atom/util.h"
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#include "lv2/lv2plug.in/ns/ext/urid/urid.h"
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#include "lv2/lv2plug.in/ns/ext/midi/midi.h"
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#define RSY_URI "http://gareus.org/oss/lv2/reasonablesynth"
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/* the synth interface */
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static void * synth_alloc (void);
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static void synth_init (void *, double rate);
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static void synth_free (void *);
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static void synth_parse_midi (void *, uint8_t *data, size_t size);
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static uint32_t synth_sound (void *, uint32_t written, uint32_t nframes, float **out);
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#include "rsynth.c"
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typedef enum {
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RSY_MIDIIN = 0,
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RSY_OUTL,
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RSY_OUTR
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} PortIndex;
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typedef struct {
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const LV2_Atom_Sequence* midiin;
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float* outL;
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float* outR;
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LV2_URID_Map* map;
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LV2_URID midi_MidiEvent;
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double SampleRateD;
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void *synth;
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} RSynth;
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/* main LV2 */
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static LV2_Handle
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instantiate(const LV2_Descriptor* descriptor,
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double rate,
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const char* bundle_path,
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const LV2_Feature* const* features)
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{
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if (rate < 8000) {
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fprintf(stderr, "RSynth.lv2 error: unsupported sample-rate (must be > 8k)\n");
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return NULL;
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}
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RSynth* self = (RSynth*)calloc(1, sizeof(RSynth));
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if(!self) {
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return NULL;
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}
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self->SampleRateD = rate;
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int i;
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for (i=0; features[i]; ++i) {
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if (!strcmp(features[i]->URI, LV2_URID__map)) {
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self->map = (LV2_URID_Map*)features[i]->data;
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}
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}
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if (!self->map) {
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fprintf(stderr, "RSynth.lv2 error: Host does not support urid:map\n");
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free(self);
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return NULL;
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}
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self->midi_MidiEvent = self->map->map(self->map->handle, LV2_MIDI__MidiEvent);
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self->synth = synth_alloc();
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synth_init(self->synth, rate);
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return (LV2_Handle)self;
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}
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static void
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connect_port(LV2_Handle handle,
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uint32_t port,
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void* data)
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{
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RSynth* self = (RSynth*)handle;
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switch ((PortIndex)port) {
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case RSY_MIDIIN:
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self->midiin = (const LV2_Atom_Sequence*)data;
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break;
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case RSY_OUTL:
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self->outL = (float*)data;
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break;
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case RSY_OUTR:
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self->outR = (float*)data;
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break;
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}
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}
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static void
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run(LV2_Handle handle, uint32_t n_samples)
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{
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RSynth* self = (RSynth*)handle;
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float* audio[2];
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audio[0] = self->outL;
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audio[1] = self->outR;
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uint32_t written = 0;
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/* Process incoming MIDI events */
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if (self->midiin) {
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LV2_Atom_Event* ev = lv2_atom_sequence_begin(&(self->midiin)->body);
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while(!lv2_atom_sequence_is_end(&(self->midiin)->body, (self->midiin)->atom.size, ev)) {
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if (ev->body.type == self->midi_MidiEvent) {
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if (written + BUFFER_SIZE_SAMPLES < ev->time.frames
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&& ev->time.frames < n_samples) {
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/* first synthesize sound up until the message timestamp */
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written = synth_sound(self->synth, written, ev->time.frames, audio);
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}
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/* send midi message to synth */
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synth_parse_midi(self->synth, (uint8_t*)(ev+1), ev->body.size);
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}
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ev = lv2_atom_sequence_next(ev);
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}
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}
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/* synthesize [remaining] sound */
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synth_sound(self->synth, written, n_samples, audio);
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}
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static void
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cleanup(LV2_Handle handle)
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{
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RSynth* self = (RSynth*)handle;
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synth_free(self->synth);
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free(handle);
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}
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static const void*
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extension_data(const char* uri)
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{
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return NULL;
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}
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static const LV2_Descriptor descriptor = {
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RSY_URI,
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instantiate,
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connect_port,
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NULL,
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run,
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NULL,
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cleanup,
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extension_data
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};
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LV2_SYMBOL_EXPORT
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const LV2_Descriptor*
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lv2_descriptor(uint32_t index)
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{
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switch (index) {
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case 0:
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return &descriptor;
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default:
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return NULL;
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}
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}
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/* vi:set ts=8 sts=2 sw=2: */
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@ -0,0 +1,8 @@
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@prefix lv2: <http://lv2plug.in/ns/lv2core#> .
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@prefix rdfs: <http://www.w3.org/2000/01/rdf-schema#> .
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@prefix ui: <http://lv2plug.in/ns/extensions/ui#> .
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<http://gareus.org/oss/lv2/reasonablesynth>
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a lv2:Plugin ;
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lv2:binary <reasonablesynth@LIB_EXT@> ;
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rdfs:seeAlso <reasonablesynth.ttl> .
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@ -0,0 +1,51 @@
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@prefix atom: <http://lv2plug.in/ns/ext/atom#> .
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@prefix doap: <http://usefulinc.com/ns/doap#> .
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@prefix foaf: <http://xmlns.com/foaf/0.1/> .
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@prefix lv2: <http://lv2plug.in/ns/lv2core#> .
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@prefix rdf: <http://www.w3.org/1999/02/22-rdf-syntax-ns#> .
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@prefix rdfs: <http://www.w3.org/2000/01/rdf-schema#> .
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@prefix pg: <http://lv2plug.in/ns/ext/port-groups#> .
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@prefix units: <http://lv2plug.in/ns/extensions/units#> .
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@prefix urid: <http://lv2plug.in/ns/ext/urid#> .
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<http://gareus.org/rgareus#me>
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a foaf:Person ;
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foaf:name "Robin Gareus" ;
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foaf:mbox <mailto:robin@gareus.org> ;
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foaf:homepage <http://gareus.org/> .
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<http://gareus.org/oss/lv2/reasonablesynth>
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a lv2:Plugin, lv2:InstrumentPlugin, doap:Project;
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doap:license <http://usefulinc.com/doap/licenses/gpl> ;
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doap:maintainer <http://gareus.org/rgareus#me> ;
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doap:name "Reasonable Synth";
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lv2:optionalFeature lv2:hardRTCapable ;
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lv2:requiredFeature urid:map ;
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rdfs:comment """A simple synthesizer with no controls at all but a reasonable sound instead.""" ;
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lv2:port
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[
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a atom:AtomPort ,
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lv2:InputPort ;
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atom:bufferType atom:Sequence ;
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atom:supports <http://lv2plug.in/ns/ext/midi#MidiEvent> ;
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lv2:index 0 ;
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lv2:symbol "MidiIn" ;
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lv2:name "MIDI Input" ;
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],
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[
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a lv2:AudioPort ,
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lv2:OutputPort ;
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lv2:index 1 ;
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lv2:symbol "outL" ;
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lv2:name "Left output" ;
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lv2:designation pg:left ;
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],
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[
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a lv2:AudioPort ,
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lv2:OutputPort ;
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lv2:index 2 ;
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lv2:symbol "outR" ;
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lv2:name "Right Output" ;
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lv2:designation pg:right ;
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]
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.
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@ -0,0 +1,490 @@
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/* reasonable simple synth
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*
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* Copyright (C) 2013 Robin Gareus <robin@gareus.org>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2, or (at your option)
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* any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software Foundation,
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* Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#ifndef _GNU_SOURCE
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#define _GNU_SOURCE // needed for M_PI
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#endif
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#include <math.h>
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#include <stdlib.h>
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#include <string.h>
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#include <stdint.h>
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#include <assert.h>
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#ifndef BUFFER_SIZE_SAMPLES
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#define BUFFER_SIZE_SAMPLES 64
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#endif
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#ifndef MIN
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#define MIN(A, B) ( (A) < (B) ? (A) : (B) )
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#endif
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/* internal MIDI event abstraction */
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enum RMIDI_EV_TYPE {
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INVALID=0,
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NOTE_ON,
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NOTE_OFF,
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PROGRAM_CHANGE,
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CONTROL_CHANGE,
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};
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struct rmidi_event_t {
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enum RMIDI_EV_TYPE type;
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uint8_t channel; /**< the MIDI channel number 0-15 */
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union {
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struct {
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uint8_t note;
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uint8_t velocity;
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} tone;
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struct {
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uint8_t param;
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uint8_t value;
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} control;
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} d;
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};
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typedef struct {
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uint32_t tme[3]; // attack, decay, release times [settings:ms || internal:samples]
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float vol[2]; // attack, sustain volume [0..1]
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uint32_t off[3]; // internal use (added attack,decay,release times)
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} ADSRcfg;
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typedef struct _RSSynthChannel {
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uint32_t keycomp;
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uint32_t adsr_cnt[128];
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float adsr_amp[128];
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float phase[128]; // various use, zero'ed on note-on
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int8_t miditable[128]; // internal, note-on/off velocity
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ADSRcfg adsr;
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void (*synthesize) (struct _RSSynthChannel* sc,
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const uint8_t note, const float vol, const float pc,
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const size_t n_samples, float* left, float* right);
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} RSSynthChannel;
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typedef void (*SynthFunction) (RSSynthChannel* sc,
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const uint8_t note, const float vol, const float pc,
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const size_t n_samples, float* left, float* right);
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typedef struct {
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uint32_t boffset;
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float buf [2][BUFFER_SIZE_SAMPLES];
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RSSynthChannel sc[16];
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float freqs[128];
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float kcgain;
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float kcfilt;
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double rate;
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} RSSynthesizer;
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/* initialize ADSR values
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*
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* @param rate sample-rate
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* @param a attack time in seconds
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* @param d decay time in seconds
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* @param r release time in seconds
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* @param avol attack gain [0..1]
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* @param svol sustain volume level [0..1]
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*/
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static void init_adsr(ADSRcfg *adsr, const double rate,
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const uint32_t a, const uint32_t d, const uint32_t r,
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const float avol, const float svol) {
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adsr->vol[0] = avol;
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adsr->vol[1] = svol;
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adsr->tme[0] = a * rate / 1000.0;
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adsr->tme[1] = d * rate / 1000.0;
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adsr->tme[2] = r * rate / 1000.0;
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assert(adsr->tme[0] > 32);
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assert(adsr->tme[1] > 32);
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assert(adsr->tme[2] > 32);
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assert(adsr->vol[0] >=0 && adsr->vol[1] <= 1.0);
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assert(adsr->vol[1] >=0 && adsr->vol[1] <= 1.0);
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adsr->off[0] = adsr->tme[0];
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adsr->off[1] = adsr->tme[1] + adsr->off[0];
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adsr->off[2] = adsr->tme[2] + adsr->off[1];
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}
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/* calculate per-sample, per-key envelope */
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static inline float adsr_env(RSSynthChannel *sc, const uint8_t note) {
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if (sc->adsr_cnt[note] < sc->adsr.off[0]) {
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// attack
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const uint32_t p = ++sc->adsr_cnt[note];
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if (p == sc->adsr.tme[0]) {
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sc->adsr_amp[note] = sc->adsr.vol[0];
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return sc->adsr.vol[0];
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} else {
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const float d = sc->adsr.vol[0] - sc->adsr_amp[note];
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return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[0]) * d;
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}
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}
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else if (sc->adsr_cnt[note] < sc->adsr.off[1]) {
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// decay
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const uint32_t p = ++sc->adsr_cnt[note] - sc->adsr.off[0];
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if (p == sc->adsr.tme[1]) {
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sc->adsr_amp[note] = sc->adsr.vol[1];
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return sc->adsr.vol[1];
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} else {
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const float d = sc->adsr.vol[1] - sc->adsr_amp[note];
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return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[1]) * d;
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}
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}
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else if (sc->adsr_cnt[note] == sc->adsr.off[1]) {
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// sustain
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return sc->adsr.vol[1];
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}
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else if (sc->adsr_cnt[note] < sc->adsr.off[2]) {
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// release
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const uint32_t p = ++sc->adsr_cnt[note] - sc->adsr.off[1];
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if (p == sc->adsr.tme[2]) {
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sc->adsr_amp[note] = 0;
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return 0;
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} else {
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const float d = 0 - sc->adsr_amp[note];
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return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[2]) * d;
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}
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}
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else {
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sc->adsr_cnt[note] = 0;
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return 0;
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}
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}
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/*****************************************************************************/
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/* piano like sound w/slight stereo phase */
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static void synthesize_sineP (RSSynthChannel* sc,
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const uint8_t note, const float vol, const float fq,
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const size_t n_samples, float* left, float* right) {
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float phase = sc->phase[note];
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for (size_t i=0; i < n_samples; ++i) {
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float env = adsr_env(sc, note);
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if (sc->adsr_cnt[note] == 0) break;
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const float amp = vol * env;
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left[i] += amp * sinf(2.0 * M_PI * phase);
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left[i] += .300 * amp * sinf(2.0 * M_PI * phase * 2.0);
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left[i] += .150 * amp * sinf(2.0 * M_PI * phase * 3.0);
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left[i] += .080 * amp * sinf(2.0 * M_PI * phase * 4.0);
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//left[i] -= .007 * amp * sinf(2.0 * M_PI * phase * 5.0);
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//left[i] += .010 * amp * sinf(2.0 * M_PI * phase * 6.0);
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//left[i] += .020 * amp * sinf(2.0 * M_PI * phase * 7.0);
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phase += fq;
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right[i] += amp * sinf(2.0 * M_PI * phase);
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right[i] += .300 * amp * sinf(2.0 * M_PI * phase * 2.0);
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right[i] += .150 * amp * sinf(2.0 * M_PI * phase * 3.0);
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right[i] -= .080 * amp * sinf(2.0 * M_PI * phase * 4.0);
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//right[i] += .007 * amp * sinf(2.0 * M_PI * phase * 5.0);
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||||
//right[i] += .010 * amp * sinf(2.0 * M_PI * phase * 6.0);
|
||||
//right[i] -= .020 * amp * sinf(2.0 * M_PI * phase * 7.0);
|
||||
if (phase > 1.0) phase -= 2.0;
|
||||
}
|
||||
sc->phase[note] = phase;
|
||||
}
|
||||
|
||||
static const ADSRcfg piano_adsr = {{ 5, 1300, 100}, { 1.0, 0.0}, {0,0,0}};
|
||||
|
||||
/*****************************************************************************/
|
||||
|
||||
|
||||
/* process note - move through ADSR states, count active keys,.. */
|
||||
static void process_key (void *synth,
|
||||
const uint8_t chn, const uint8_t note,
|
||||
const size_t n_samples, float *left, float *right)
|
||||
{
|
||||
RSSynthesizer* rs = (RSSynthesizer*)synth;
|
||||
RSSynthChannel* sc = &rs->sc[chn];
|
||||
const int8_t vel = sc->miditable[note];
|
||||
const float vol = /* master_volume */ 0.25 * fabsf(vel) / 127.0;
|
||||
const float phase = sc->phase[note];
|
||||
|
||||
if (phase == -10 && vel > 0) {
|
||||
// new note on
|
||||
assert(sc->adsr_cnt[note] == 0);
|
||||
sc->adsr_amp[note] = 0;
|
||||
sc->adsr_cnt[note] = 0;
|
||||
sc->phase[note] = 0;
|
||||
sc->keycomp++;
|
||||
//printf("[On] Now %d keys active on chn %d\n", sc->keycomp, chn);
|
||||
}
|
||||
else if (phase >= -1.0 && phase <= 1.0 && vel > 0) {
|
||||
// sustain note or re-start note while adsr in progress:
|
||||
if (sc->adsr_cnt[note] > sc->adsr.off[1]) {
|
||||
// x-fade to attack
|
||||
sc->adsr_amp[note] = adsr_env(sc, note);
|
||||
sc->adsr_cnt[note] = 0;
|
||||
}
|
||||
}
|
||||
else if (phase >= -1.0 && phase <= 1.0 && vel < 0) {
|
||||
// note off
|
||||
if (sc->adsr_cnt[note] <= sc->adsr.off[1]) {
|
||||
if (sc->adsr_cnt[note] != sc->adsr.off[1]) {
|
||||
// x-fade to release
|
||||
sc->adsr_amp[note] = adsr_env(sc, note);
|
||||
}
|
||||
sc->adsr_cnt[note] = sc->adsr.off[1] + 1;
|
||||
}
|
||||
}
|
||||
else {
|
||||
/* note-on + off in same cycle */
|
||||
sc->miditable[note] = 0;
|
||||
sc->adsr_cnt[note] = 0;
|
||||
sc->phase[note] = -10;
|
||||
return;
|
||||
}
|
||||
|
||||
// synthesize actual sound
|
||||
sc->synthesize(sc, note, vol, rs->freqs[note], n_samples, left, right);
|
||||
|
||||
if (sc->adsr_cnt[note] == 0) {
|
||||
//printf("Note %d,%d released\n", chn, note);
|
||||
sc->miditable[note] = 0;
|
||||
sc->adsr_amp[note] = 0;
|
||||
sc->phase[note] = -10;
|
||||
sc->keycomp--;
|
||||
//printf("[off] Now %d keys active on chn %d\n", sc->keycomp, chn);
|
||||
}
|
||||
}
|
||||
|
||||
/* synthesize a BUFFER_SIZE_SAMPLES's of audio-data */
|
||||
static void synth_fragment (void *synth, const size_t n_samples, float *left, float *right) {
|
||||
RSSynthesizer* rs = (RSSynthesizer*)synth;
|
||||
memset (left, 0, n_samples * sizeof(float));
|
||||
memset (right, 0, n_samples * sizeof(float));
|
||||
uint8_t keycomp = 0;
|
||||
|
||||
for (int c=0; c < 16; ++c) {
|
||||
for (int k=0; k < 128; ++k) {
|
||||
if (rs->sc[c].miditable[k] == 0) continue;
|
||||
process_key(synth, c, k, n_samples, left, right);
|
||||
}
|
||||
keycomp += rs->sc[c].keycomp;
|
||||
}
|
||||
|
||||
#if 1 // key-compression
|
||||
float kctgt = 8.0 / (float)(keycomp + 7.0);
|
||||
if (kctgt < .5) kctgt = .5;
|
||||
if (kctgt > 1.0) kctgt = 1.0;
|
||||
const float _w = rs->kcfilt;
|
||||
for (unsigned int i=0; i < n_samples; ++i) {
|
||||
rs->kcgain += _w * (kctgt - rs->kcgain);
|
||||
left[i] *= rs->kcgain;
|
||||
right[i] *= rs->kcgain;
|
||||
}
|
||||
rs->kcgain += 1e-12;
|
||||
#endif
|
||||
}
|
||||
|
||||
static void synth_reset_channel(RSSynthChannel* sc) {
|
||||
for (int k=0; k < 128; ++k) {
|
||||
sc->adsr_cnt[k] = 0;
|
||||
sc->adsr_amp[k] = 0;
|
||||
sc->phase[k] = -10;
|
||||
sc->miditable[k] = 0;
|
||||
}
|
||||
sc->keycomp = 0;
|
||||
}
|
||||
|
||||
static void synth_reset(void *synth) {
|
||||
RSSynthesizer* rs = (RSSynthesizer*)synth;
|
||||
for (int c=0; c < 16; ++c) {
|
||||
synth_reset_channel(&(rs->sc[c]));
|
||||
}
|
||||
rs->kcgain = 0;
|
||||
}
|
||||
|
||||
static void synth_load(RSSynthChannel *sc, const double rate,
|
||||
SynthFunction synthesize,
|
||||
ADSRcfg const * const adsr) {
|
||||
synth_reset_channel(sc);
|
||||
init_adsr(&sc->adsr, rate,
|
||||
adsr->tme[0], adsr->tme[1], adsr->tme[2],
|
||||
adsr->vol[0], adsr->vol[1]);
|
||||
sc->synthesize = synthesize;
|
||||
}
|
||||
|
||||
|
||||
/**
|
||||
* internal abstraction of MIDI data handling
|
||||
*/
|
||||
static void synth_process_midi_event(void *synth, struct rmidi_event_t *ev) {
|
||||
RSSynthesizer* rs = (RSSynthesizer*)synth;
|
||||
switch(ev->type) {
|
||||
case NOTE_ON:
|
||||
if (rs->sc[ev->channel].miditable[ev->d.tone.note] <= 0)
|
||||
rs->sc[ev->channel].miditable[ev->d.tone.note] = ev->d.tone.velocity;
|
||||
break;
|
||||
case NOTE_OFF:
|
||||
if (rs->sc[ev->channel].miditable[ev->d.tone.note] > 0)
|
||||
rs->sc[ev->channel].miditable[ev->d.tone.note] *= -1.0;
|
||||
break;
|
||||
case PROGRAM_CHANGE:
|
||||
break;
|
||||
case CONTROL_CHANGE:
|
||||
if (ev->d.control.param == 0x00 || ev->d.control.param == 0x20) {
|
||||
/* 0x00 and 0x20 are used for BANK select */
|
||||
break;
|
||||
} else
|
||||
if (ev->d.control.param == 121) {
|
||||
/* reset all controllers */
|
||||
break;
|
||||
} else
|
||||
if (ev->d.control.param == 120 || ev->d.control.param == 123) {
|
||||
/* Midi panic: 120: all sound off, 123: all notes off*/
|
||||
synth_reset_channel(&(rs->sc[ev->channel]));
|
||||
break;
|
||||
} else
|
||||
if (ev->d.control.param >= 120) {
|
||||
/* params 122-127 are reserved - skip them. */
|
||||
break;
|
||||
}
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
/******************************************************************************
|
||||
* PUBLIC API (used by lv2.c)
|
||||
*/
|
||||
|
||||
/**
|
||||
* align LV2 and internal synth buffers
|
||||
* call synth_fragment as often as needed for the given LV2 buffer size
|
||||
*
|
||||
* @param synth synth-handle
|
||||
* @param written samples written so far (offset in \ref out)
|
||||
* @param nframes total samples to synthesize and write to the \out buffer
|
||||
* @param out pointer to stereo output buffers
|
||||
* @return end of buffer (written + nframes)
|
||||
*/
|
||||
static uint32_t synth_sound (void *synth, uint32_t written, const uint32_t nframes, float **out) {
|
||||
RSSynthesizer* rs = (RSSynthesizer*)synth;
|
||||
|
||||
while (written < nframes) {
|
||||
uint32_t nremain = nframes - written;
|
||||
|
||||
if (rs->boffset >= BUFFER_SIZE_SAMPLES) {
|
||||
rs->boffset = 0;
|
||||
synth_fragment(rs, BUFFER_SIZE_SAMPLES, rs->buf[0], rs->buf[1]);
|
||||
}
|
||||
|
||||
uint32_t nread = MIN(nremain, (BUFFER_SIZE_SAMPLES - rs->boffset));
|
||||
|
||||
memcpy(&out[0][written], &rs->buf[0][rs->boffset], nread*sizeof(float));
|
||||
memcpy(&out[1][written], &rs->buf[1][rs->boffset], nread*sizeof(float));
|
||||
|
||||
written += nread;
|
||||
rs->boffset += nread;
|
||||
}
|
||||
return written;
|
||||
}
|
||||
|
||||
/**
|
||||
* parse raw midi-data.
|
||||
*
|
||||
* @param synth synth-handle
|
||||
* @param data 8bit midi message
|
||||
* @param size number of bytes in the midi-message
|
||||
*/
|
||||
static void synth_parse_midi(void *synth, uint8_t *data, size_t size) {
|
||||
if (size < 2 || size > 3) return;
|
||||
// All messages need to be 3 bytes; except program-changes: 2bytes.
|
||||
if (size == 2 && (data[0] & 0xf0) != 0xC0) return;
|
||||
|
||||
struct rmidi_event_t ev;
|
||||
|
||||
ev.channel = data[0]&0x0f;
|
||||
switch (data[0] & 0xf0) {
|
||||
case 0x80:
|
||||
ev.type=NOTE_OFF;
|
||||
ev.d.tone.note=data[1]&0x7f;
|
||||
ev.d.tone.velocity=data[2]&0x7f;
|
||||
break;
|
||||
case 0x90:
|
||||
ev.type=NOTE_ON;
|
||||
ev.d.tone.note=data[1]&0x7f;
|
||||
ev.d.tone.velocity=data[2]&0x7f;
|
||||
break;
|
||||
case 0xB0:
|
||||
ev.type=CONTROL_CHANGE;
|
||||
ev.d.control.param=data[1]&0x7f;
|
||||
ev.d.control.value=data[2]&0x7f;
|
||||
break;
|
||||
case 0xC0:
|
||||
ev.type=PROGRAM_CHANGE;
|
||||
ev.d.control.value=data[1]&0x7f;
|
||||
break;
|
||||
default:
|
||||
return;
|
||||
}
|
||||
synth_process_midi_event(synth, &ev);
|
||||
}
|
||||
|
||||
/**
|
||||
* initialize the synth
|
||||
* This should be called after synth_alloc()
|
||||
* as soon as the sample-rate is known
|
||||
*
|
||||
* @param synth synth-handle
|
||||
* @param rate sample-rate
|
||||
*/
|
||||
static void synth_init(void *synth, double rate) {
|
||||
RSSynthesizer* rs = (RSSynthesizer*)synth;
|
||||
rs->rate = rate;
|
||||
rs->boffset = BUFFER_SIZE_SAMPLES;
|
||||
const float tuning = 440;
|
||||
for (int k=0; k < 128; k++) {
|
||||
rs->freqs[k] = (2.0 * tuning / 32.0f) * powf(2, (k - 9.0) / 12.0) / rate;
|
||||
assert(rs->freqs[k] < M_PI/2); // otherwise spatialization may phase out..
|
||||
}
|
||||
rs->kcfilt = 12.0 / rate;
|
||||
synth_reset(synth);
|
||||
|
||||
for (int c=0; c < 16; c++) {
|
||||
synth_load(&rs->sc[c], rate, &synthesize_sineP, &piano_adsr);
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Allocate data-structure, create a handle for all other synth_* functions.
|
||||
*
|
||||
* This data should be freeded with \ref synth_free when the synth is no
|
||||
* longer needed.
|
||||
*
|
||||
* The synth can only be used after calling \rev synth_init as well.
|
||||
*
|
||||
* @return synth-handle
|
||||
*/
|
||||
static void * synth_alloc(void) {
|
||||
return calloc(1, sizeof(RSSynthesizer));
|
||||
}
|
||||
|
||||
/**
|
||||
* release synth data structure
|
||||
* @param synth synth-handle
|
||||
*/
|
||||
static void synth_free(void *synth) {
|
||||
free(synth);
|
||||
}
|
||||
/* vi:set ts=8 sts=2 sw=2: */
|
||||
|
|
@ -0,0 +1,44 @@
|
|||
#!/usr/bin/env python
|
||||
import os
|
||||
import re
|
||||
import shutil
|
||||
import waflib.extras.autowaf as autowaf
|
||||
|
||||
# Mandatory variables
|
||||
top = '.'
|
||||
out = 'build'
|
||||
|
||||
def options(opt):
|
||||
autowaf.set_options(opt)
|
||||
|
||||
def configure(opt):
|
||||
conf.load('compiler_c')
|
||||
autowaf.configure(conf)
|
||||
autowaf.set_c99_mode(conf)
|
||||
autowaf.check_pkg(conf, 'lv2', atleast_version='1.4.1',
|
||||
uselib_store='LV2_1_4_1')
|
||||
|
||||
def build(bld):
|
||||
bundle = 'reasonablesynth.lv2'
|
||||
module_pat = re.sub('^lib', '', bld.env.cshlib_PATTERN)
|
||||
module_ext = module_pat[module_pat.rfind('.'):]
|
||||
|
||||
# Build RDF files
|
||||
for i in ['manifest.ttl', 'reasonablesynth.ttl']:
|
||||
bld(features = 'subst',
|
||||
source = i + '.in',
|
||||
target = '%s/%s' % (bundle, i),
|
||||
install_path = '${LV2DIR}/%s' % bundle,
|
||||
LIB_EXT = module_ext)
|
||||
|
||||
# Build plugin library
|
||||
obj = bld(features = 'c cshlib',
|
||||
source = 'lv2.c',
|
||||
dep_files = 'rsynth.c',
|
||||
name = 'reasonablesynth',
|
||||
target = '%s/reasonablesynth' % bundle,
|
||||
install_path = '${LV2DIR}/%s' % bundle,
|
||||
use = 'LV2_1_4_1')
|
||||
obj.env.cshlib_PATTERN = module_pat
|
||||
|
||||
# vi:set ts=4 sw=4 et:
|
||||
Loading…
Reference in New Issue