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add reasonablesynth.lv2

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A reasonably simple synth to allow new users to 'hear midi'.
This is a first step. It still needs proper install and bundling.
This commit is contained in:
Robin Gareus 2013-10-20 04:31:07 +02:00
parent f191bdf6a0
commit f5c386bbb4
6 changed files with 784 additions and 0 deletions
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190
libs/plugins/reasonablesynth.lv2/lv2.c Normal file
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/* reasonable simple synth
*
* Copyright (C) 2013 Robin Gareus <robin@gareus.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2, or (at your option)
* any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software Foundation,
* Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
#define _GNU_SOURCE
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
/* LV2 */
#include "lv2/lv2plug.in/ns/lv2core/lv2.h"
#include "lv2/lv2plug.in/ns/ext/atom/util.h"
#include "lv2/lv2plug.in/ns/ext/urid/urid.h"
#include "lv2/lv2plug.in/ns/ext/midi/midi.h"
#define RSY_URI "http://gareus.org/oss/lv2/reasonablesynth"
/* the synth interface */
static void * synth_alloc (void);
static void synth_init (void *, double rate);
static void synth_free (void *);
static void synth_parse_midi (void *, uint8_t *data, size_t size);
static uint32_t synth_sound (void *, uint32_t written, uint32_t nframes, float **out);
#include "rsynth.c"
typedef enum {
RSY_MIDIIN = 0,
RSY_OUTL,
RSY_OUTR
} PortIndex;
typedef struct {
const LV2_Atom_Sequence* midiin;
float* outL;
float* outR;
LV2_URID_Map* map;
LV2_URID midi_MidiEvent;
double SampleRateD;
void *synth;
} RSynth;
/* main LV2 */
static LV2_Handle
instantiate(const LV2_Descriptor* descriptor,
double rate,
const char* bundle_path,
const LV2_Feature* const* features)
{
if (rate < 8000) {
fprintf(stderr, "RSynth.lv2 error: unsupported sample-rate (must be > 8k)\n");
return NULL;
}
RSynth* self = (RSynth*)calloc(1, sizeof(RSynth));
if(!self) {
return NULL;
}
self->SampleRateD = rate;
int i;
for (i=0; features[i]; ++i) {
if (!strcmp(features[i]->URI, LV2_URID__map)) {
self->map = (LV2_URID_Map*)features[i]->data;
}
}
if (!self->map) {
fprintf(stderr, "RSynth.lv2 error: Host does not support urid:map\n");
free(self);
return NULL;
}
self->midi_MidiEvent = self->map->map(self->map->handle, LV2_MIDI__MidiEvent);
self->synth = synth_alloc();
synth_init(self->synth, rate);
return (LV2_Handle)self;
}
static void
connect_port(LV2_Handle handle,
uint32_t port,
void* data)
{
RSynth* self = (RSynth*)handle;
switch ((PortIndex)port) {
case RSY_MIDIIN:
self->midiin = (const LV2_Atom_Sequence*)data;
break;
case RSY_OUTL:
self->outL = (float*)data;
break;
case RSY_OUTR:
self->outR = (float*)data;
break;
}
}
static void
run(LV2_Handle handle, uint32_t n_samples)
{
RSynth* self = (RSynth*)handle;
float* audio[2];
audio[0] = self->outL;
audio[1] = self->outR;
uint32_t written = 0;
/* Process incoming MIDI events */
if (self->midiin) {
LV2_Atom_Event* ev = lv2_atom_sequence_begin(&(self->midiin)->body);
while(!lv2_atom_sequence_is_end(&(self->midiin)->body, (self->midiin)->atom.size, ev)) {
if (ev->body.type == self->midi_MidiEvent) {
if (written + BUFFER_SIZE_SAMPLES < ev->time.frames
&& ev->time.frames < n_samples) {
/* first synthesize sound up until the message timestamp */
written = synth_sound(self->synth, written, ev->time.frames, audio);
}
/* send midi message to synth */
synth_parse_midi(self->synth, (uint8_t*)(ev+1), ev->body.size);
}
ev = lv2_atom_sequence_next(ev);
}
}
/* synthesize [remaining] sound */
synth_sound(self->synth, written, n_samples, audio);
}
static void
cleanup(LV2_Handle handle)
{
RSynth* self = (RSynth*)handle;
synth_free(self->synth);
free(handle);
}
static const void*
extension_data(const char* uri)
{
return NULL;
}
static const LV2_Descriptor descriptor = {
RSY_URI,
instantiate,
connect_port,
NULL,
run,
NULL,
cleanup,
extension_data
};
LV2_SYMBOL_EXPORT
const LV2_Descriptor*
lv2_descriptor(uint32_t index)
{
switch (index) {
case 0:
return &descriptor;
default:
return NULL;
}
}
/* vi:set ts=8 sts=2 sw=2: */

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libs/plugins/reasonablesynth.lv2/manifest.ttl.in Normal file
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@prefix lv2: <http://lv2plug.in/ns/lv2core#> .
@prefix rdfs: <http://www.w3.org/2000/01/rdf-schema#> .
@prefix ui: <http://lv2plug.in/ns/extensions/ui#> .
<http://gareus.org/oss/lv2/reasonablesynth>
a lv2:Plugin ;
lv2:binary <reasonablesynth@LIB_EXT@> ;
rdfs:seeAlso <reasonablesynth.ttl> .

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libs/plugins/reasonablesynth.lv2/reasonablesynth.ttl.in Normal file
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@prefix atom: <http://lv2plug.in/ns/ext/atom#> .
@prefix doap: <http://usefulinc.com/ns/doap#> .
@prefix foaf: <http://xmlns.com/foaf/0.1/> .
@prefix lv2: <http://lv2plug.in/ns/lv2core#> .
@prefix rdf: <http://www.w3.org/1999/02/22-rdf-syntax-ns#> .
@prefix rdfs: <http://www.w3.org/2000/01/rdf-schema#> .
@prefix pg: <http://lv2plug.in/ns/ext/port-groups#> .
@prefix units: <http://lv2plug.in/ns/extensions/units#> .
@prefix urid: <http://lv2plug.in/ns/ext/urid#> .
<http://gareus.org/rgareus#me>
a foaf:Person ;
foaf:name "Robin Gareus" ;
foaf:mbox <mailto:robin@gareus.org> ;
foaf:homepage <http://gareus.org/> .
<http://gareus.org/oss/lv2/reasonablesynth>
a lv2:Plugin, lv2:InstrumentPlugin, doap:Project;
doap:license <http://usefulinc.com/doap/licenses/gpl> ;
doap:maintainer <http://gareus.org/rgareus#me> ;
doap:name "Reasonable Synth";
lv2:optionalFeature lv2:hardRTCapable ;
lv2:requiredFeature urid:map ;
rdfs:comment """A simple synthesizer with no controls at all but a reasonable sound instead.""" ;
lv2:port
[
a atom:AtomPort ,
lv2:InputPort ;
atom:bufferType atom:Sequence ;
atom:supports <http://lv2plug.in/ns/ext/midi#MidiEvent> ;
lv2:index 0 ;
lv2:symbol "MidiIn" ;
lv2:name "MIDI Input" ;
],
[
a lv2:AudioPort ,
lv2:OutputPort ;
lv2:index 1 ;
lv2:symbol "outL" ;
lv2:name "Left output" ;
lv2:designation pg:left ;
],
[
a lv2:AudioPort ,
lv2:OutputPort ;
lv2:index 2 ;
lv2:symbol "outR" ;
lv2:name "Right Output" ;
lv2:designation pg:right ;
]
.

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/* reasonable simple synth
*
* Copyright (C) 2013 Robin Gareus <robin@gareus.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2, or (at your option)
* any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software Foundation,
* Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
#ifndef _GNU_SOURCE
#define _GNU_SOURCE // needed for M_PI
#endif
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <stdint.h>
#include <assert.h>
#ifndef BUFFER_SIZE_SAMPLES
#define BUFFER_SIZE_SAMPLES 64
#endif
#ifndef MIN
#define MIN(A, B) ( (A) < (B) ? (A) : (B) )
#endif
/* internal MIDI event abstraction */
enum RMIDI_EV_TYPE {
INVALID=0,
NOTE_ON,
NOTE_OFF,
PROGRAM_CHANGE,
CONTROL_CHANGE,
};
struct rmidi_event_t {
enum RMIDI_EV_TYPE type;
uint8_t channel; /**< the MIDI channel number 0-15 */
union {
struct {
uint8_t note;
uint8_t velocity;
} tone;
struct {
uint8_t param;
uint8_t value;
} control;
} d;
};
typedef struct {
uint32_t tme[3]; // attack, decay, release times [settings:ms || internal:samples]
float vol[2]; // attack, sustain volume [0..1]
uint32_t off[3]; // internal use (added attack,decay,release times)
} ADSRcfg;
typedef struct _RSSynthChannel {
uint32_t keycomp;
uint32_t adsr_cnt[128];
float adsr_amp[128];
float phase[128]; // various use, zero'ed on note-on
int8_t miditable[128]; // internal, note-on/off velocity
ADSRcfg adsr;
void (*synthesize) (struct _RSSynthChannel* sc,
const uint8_t note, const float vol, const float pc,
const size_t n_samples, float* left, float* right);
} RSSynthChannel;
typedef void (*SynthFunction) (RSSynthChannel* sc,
const uint8_t note, const float vol, const float pc,
const size_t n_samples, float* left, float* right);
typedef struct {
uint32_t boffset;
float buf [2][BUFFER_SIZE_SAMPLES];
RSSynthChannel sc[16];
float freqs[128];
float kcgain;
float kcfilt;
double rate;
} RSSynthesizer;
/* initialize ADSR values
*
* @param rate sample-rate
* @param a attack time in seconds
* @param d decay time in seconds
* @param r release time in seconds
* @param avol attack gain [0..1]
* @param svol sustain volume level [0..1]
*/
static void init_adsr(ADSRcfg *adsr, const double rate,
const uint32_t a, const uint32_t d, const uint32_t r,
const float avol, const float svol) {
adsr->vol[0] = avol;
adsr->vol[1] = svol;
adsr->tme[0] = a * rate / 1000.0;
adsr->tme[1] = d * rate / 1000.0;
adsr->tme[2] = r * rate / 1000.0;
assert(adsr->tme[0] > 32);
assert(adsr->tme[1] > 32);
assert(adsr->tme[2] > 32);
assert(adsr->vol[0] >=0 && adsr->vol[1] <= 1.0);
assert(adsr->vol[1] >=0 && adsr->vol[1] <= 1.0);
adsr->off[0] = adsr->tme[0];
adsr->off[1] = adsr->tme[1] + adsr->off[0];
adsr->off[2] = adsr->tme[2] + adsr->off[1];
}
/* calculate per-sample, per-key envelope */
static inline float adsr_env(RSSynthChannel *sc, const uint8_t note) {
if (sc->adsr_cnt[note] < sc->adsr.off[0]) {
// attack
const uint32_t p = ++sc->adsr_cnt[note];
if (p == sc->adsr.tme[0]) {
sc->adsr_amp[note] = sc->adsr.vol[0];
return sc->adsr.vol[0];
} else {
const float d = sc->adsr.vol[0] - sc->adsr_amp[note];
return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[0]) * d;
}
}
else if (sc->adsr_cnt[note] < sc->adsr.off[1]) {
// decay
const uint32_t p = ++sc->adsr_cnt[note] - sc->adsr.off[0];
if (p == sc->adsr.tme[1]) {
sc->adsr_amp[note] = sc->adsr.vol[1];
return sc->adsr.vol[1];
} else {
const float d = sc->adsr.vol[1] - sc->adsr_amp[note];
return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[1]) * d;
}
}
else if (sc->adsr_cnt[note] == sc->adsr.off[1]) {
// sustain
return sc->adsr.vol[1];
}
else if (sc->adsr_cnt[note] < sc->adsr.off[2]) {
// release
const uint32_t p = ++sc->adsr_cnt[note] - sc->adsr.off[1];
if (p == sc->adsr.tme[2]) {
sc->adsr_amp[note] = 0;
return 0;
} else {
const float d = 0 - sc->adsr_amp[note];
return sc->adsr_amp[note] + (p / (float) sc->adsr.tme[2]) * d;
}
}
else {
sc->adsr_cnt[note] = 0;
return 0;
}
}
/*****************************************************************************/
/* piano like sound w/slight stereo phase */
static void synthesize_sineP (RSSynthChannel* sc,
const uint8_t note, const float vol, const float fq,
const size_t n_samples, float* left, float* right) {
float phase = sc->phase[note];
for (size_t i=0; i < n_samples; ++i) {
float env = adsr_env(sc, note);
if (sc->adsr_cnt[note] == 0) break;
const float amp = vol * env;
left[i] += amp * sinf(2.0 * M_PI * phase);
left[i] += .300 * amp * sinf(2.0 * M_PI * phase * 2.0);
left[i] += .150 * amp * sinf(2.0 * M_PI * phase * 3.0);
left[i] += .080 * amp * sinf(2.0 * M_PI * phase * 4.0);
//left[i] -= .007 * amp * sinf(2.0 * M_PI * phase * 5.0);
//left[i] += .010 * amp * sinf(2.0 * M_PI * phase * 6.0);
//left[i] += .020 * amp * sinf(2.0 * M_PI * phase * 7.0);
phase += fq;
right[i] += amp * sinf(2.0 * M_PI * phase);
right[i] += .300 * amp * sinf(2.0 * M_PI * phase * 2.0);
right[i] += .150 * amp * sinf(2.0 * M_PI * phase * 3.0);
right[i] -= .080 * amp * sinf(2.0 * M_PI * phase * 4.0);
//right[i] += .007 * amp * sinf(2.0 * M_PI * phase * 5.0);
//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: */

44
libs/plugins/reasonablesynth.lv2/wscript Normal file
View File

@ -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:

1
wscript
View File

@ -35,6 +35,7 @@ children = [
'libs/gtkmm2ext',
'libs/clearlooks-newer',
'libs/audiographer',
'libs/plugins/reasonablesynth.lv2',
'gtk2_ardour',
'export',
'midi_maps',