You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
 
 
 
 
 
 
dexed/Source/PluginProcessor.cpp

415 lines
12 KiB

/**
*
* Copyright (c) 2013 Pascal Gauthier.
*
* 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 of
* the License, 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., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301 USA.
*/
#include "PluginProcessor.h"
#include "PluginEditor.h"
#include "msfa/synth.h"
#include "msfa/freqlut.h"
#include "msfa/sin.h"
#include "msfa/exp2.h"
#include "msfa/pitchenv.h"
#include "msfa/aligned_buf.h"
//==============================================================================
DexedAudioProcessor::DexedAudioProcessor() {
Logger *tmp = Logger::getCurrentLogger();
if ( tmp == NULL ) {
Logger::setCurrentLogger(FileLogger::createDateStampedLogger("Dexed", "DebugSession-", "log", "DexedAudioProcessor Created"));
}
TRACE("Hi");
Exp2::init();
Tanh::init();
Sin::init();
currentNote = -1;
workBlock = NULL;
initCtrl();
setCurrentProgram(0);
}
DexedAudioProcessor::~DexedAudioProcessor() {
TRACE("Bye");
}
//==============================================================================
void DexedAudioProcessor::prepareToPlay(double sampleRate, int samplesPerBlock) {
Freqlut::init(sampleRate);
Lfo::init(sampleRate);
PitchEnv::init(sampleRate);
for (int note = 0; note < MAX_ACTIVE_NOTES; ++note) {
voices[note].dx7_note = new Dx7Note;
voices[note].keydown = false;
voices[note].sustained = false;
voices[note].live = false;
}
currentNote = 0;
controllers.values_[kControllerPitch] = 0x2000;
sustain = false;
extra_buf_size = 0;
workBlockSize = samplesPerBlock;
workBlock = new SInt16[samplesPerBlock];
keyboardState.reset();
}
void DexedAudioProcessor::releaseResources() {
currentNote = -1;
for (int note = 0; note < MAX_ACTIVE_NOTES; ++note) {
delete voices[note].dx7_note;
voices[note].keydown = false;
voices[note].sustained = false;
voices[note].live = false;
}
if ( workBlock != NULL ) {
delete workBlock;
}
keyboardState.reset();
}
void DexedAudioProcessor::processBlock(AudioSampleBuffer& buffer, MidiBuffer& midiMessages) {
int numSamples = buffer.getNumSamples();
if ( refreshVoice ) {
for(int i=0;i<MAX_ACTIVE_NOTES;i++) {
if ( voices[i].live )
voices[i].dx7_note->update(data, voices[i].midi_note);
}
lfo.reset(data + 137);
refreshVoice = false;
}
// check buffer size
if ( numSamples > workBlockSize ) {
delete workBlock;
workBlockSize = numSamples;
workBlock = new SInt16[workBlockSize];
}
// Now pass any incoming midi messages to our keyboard state object, and let it
// add messages to the buffer if the user is clicking on the on-screen keys
keyboardState.processNextMidiBuffer (midiMessages, 0, numSamples, true);
// check input
MidiBuffer::Iterator it(midiMessages);
MidiMessage msg;
int pos;
while(it.getNextEvent(msg, pos)) {
processMidiMessage(&msg);
}
midiMessages.clear();
processSamples(numSamples, workBlock);
float *channelData = buffer.getSampleData(0);
for(int i = 0; i < numSamples; i++ ) {
float f = ((float) workBlock[i]) / (float) 32768;
if( f > 1 ) f = 1;
if( f < -1 ) f = -1;
channelData[i] = (double) f;
}
//fx.process(channelData, numSamples);
// DX7 is a mono synth
for (int channel = 1; channel < getNumInputChannels(); ++channel) {
buffer.copyFrom(channel, 0, channelData, numSamples, 1);
}
// In case we have more outputs than inputs, we'll clear any output
// channels that didn't contain input data, (because these aren't
// guaranteed to be empty - they may contain garbage).
for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i) {
buffer.clear (i, 0, buffer.getNumSamples());
}
if ( ! midiOut.isEmpty() ) {
midiMessages.swapWith(midiOut);
}
}
//==============================================================================
// This creates new instances of the plugin..
AudioProcessor* JUCE_CALLTYPE createPluginFilter() {
return new DexedAudioProcessor();
}
void DexedAudioProcessor::processMidiMessage(MidiMessage *msg) {
if ( msg->isSysEx() ) {
TRACE("SYSEX RECEIVED");
const uint8 *buf = msg->getSysExData();
int sz = msg->getSysExDataSize();
if ( sz < 3 )
return;
// test if it is a Yamaha Sysex
if ( buf[0] != 0x43 )
return;
// single voice dump
if ( buf[2] == 0 ) {
if ( sz < 155 ) {
TRACE("wrong single voice datasize %d", buf[2]);
return;
}
updateProgramFromSysex(buf+5);
}
return;
}
const uint8 *buf = msg->getRawData();
uint8_t cmd = buf[0];
switch(cmd & 0xf0) {
case 0x80 :
keyup(buf[1]);
return;
case 0x90 :
keydown(buf[1], buf[2]);
return;
case 0xb0 : {
int controller = buf[1];
int value = buf[2];
if (controller == 64) {
sustain = value != 0;
if (!sustain) {
for (int note = 0; note < MAX_ACTIVE_NOTES; note++) {
if (voices[note].sustained && !voices[note].keydown) {
voices[note].dx7_note->keyup();
voices[note].sustained = false;
}
}
}
}
}
return;
case 0xc0 :
setCurrentProgram(buf[1]);
return;
}
switch (cmd) {
case 0xe0 :
controllers.values_[kControllerPitch] = (char) buf[1] | (buf[2] << 7);
break;
}
}
void DexedAudioProcessor::keydown(uint8_t pitch, uint8_t velo) {
if ( velo == 0 ) {
keyup(pitch);
return;
}
int note = currentNote;
for (int i = 0; i < MAX_ACTIVE_NOTES; i++) {
if (!voices[note].keydown) {
currentNote = (note + 1) % MAX_ACTIVE_NOTES;
lfo.keydown(); // TODO: should only do this if # keys down was 0
voices[note].midi_note = pitch;
voices[note].keydown = true;
voices[note].sustained = sustain;
voices[note].live = true;
voices[note].dx7_note->init(data, pitch, velo);
return;
}
note = (note + 1) % MAX_ACTIVE_NOTES;
}
}
void DexedAudioProcessor::keyup(uint8_t pitch) {
for (int note = 0; note < MAX_ACTIVE_NOTES; ++note) {
if (voices[note].midi_note == pitch && voices[note].keydown) {
if (sustain) {
voices[note].sustained = true;
} else {
voices[note].dx7_note->keyup();
}
voices[note].keydown = false;
}
}
}
void DexedAudioProcessor::processSamples(int n_samples, int16_t *buffer) {
int i;
for (i = 0; i < n_samples && i < extra_buf_size; i++) {
buffer[i] = extra_buf[i];
}
if (extra_buf_size > n_samples) {
for (int j = 0; j < extra_buf_size - n_samples; j++) {
extra_buf[j] = extra_buf[j + n_samples];
}
extra_buf_size -= n_samples;
return;
}
for (; i < n_samples; i += N) {
AlignedBuf<int32_t, N> audiobuf;
for (int j = 0; j < N; ++j) {
audiobuf.get()[j] = 0;
}
int32_t lfovalue = lfo.getsample();
int32_t lfodelay = lfo.getdelay();
for (int note = 0; note < MAX_ACTIVE_NOTES; ++note) {
if (voices[note].live) {
voices[note].dx7_note->compute(audiobuf.get(), lfovalue, lfodelay, &controllers);
}
}
int jmax = n_samples - i;
for (int j = 0; j < N; ++j) {
int32_t val = audiobuf.get()[j] >> 4;
int clip_val = val < -(1 << 24) ? 0x8000 : val >= (1 << 24) ? 0x7fff :
val >> 9;
// TODO: maybe some dithering?
if (j < jmax) {
buffer[i + j] = clip_val;
} else {
extra_buf[j - jmax] = clip_val;
}
}
}
extra_buf_size = i - n_samples;
}
// ====================================================================
bool DexedAudioProcessor::peekEnvStatus(int32_t *env) {
if ( currentNote == -1 )
return false;
// we are trying to find the last "keydown" note
int note = currentNote;
for (int i = 0; i < MAX_ACTIVE_NOTES; i++) {
if (voices[note].keydown) {
voices[note].dx7_note->peekEnvStatus(env);
return true;
}
if ( --note < 0 )
note = MAX_ACTIVE_NOTES-1;
}
// not found; try a live note
note = currentNote;
for (int i = 0; i < MAX_ACTIVE_NOTES; i++) {
if (voices[note].live) {
voices[note].dx7_note->peekEnvStatus(env);
return true;
}
if ( --note < 0 )
note = MAX_ACTIVE_NOTES-1;
}
// all the notes are stopped, return 0;
memset(env, 0, sizeof(int32_t) * 6);
return true;
}
const String DexedAudioProcessor::getInputChannelName (int channelIndex) const {
return String (channelIndex + 1);
}
const String DexedAudioProcessor::getOutputChannelName (int channelIndex) const {
return String (channelIndex + 1);
}
bool DexedAudioProcessor::isInputChannelStereoPair (int index) const {
return true;
}
bool DexedAudioProcessor::isOutputChannelStereoPair (int index) const {
return true;
}
bool DexedAudioProcessor::acceptsMidi() const {
#if JucePlugin_WantsMidiInput
return true;
#else
return false;
#endif
}
bool DexedAudioProcessor::producesMidi() const {
#if JucePlugin_ProducesMidiOutput
return true;
#else
return false;
#endif
}
bool DexedAudioProcessor::silenceInProducesSilenceOut() const {
return false;
}
double DexedAudioProcessor::getTailLengthSeconds() const {
return 0.0;
}
const String DexedAudioProcessor::getName() const {
return JucePlugin_Name;
}
//==============================================================================
bool DexedAudioProcessor::hasEditor() const {
return true; // (change this to false if you choose to not supply an editor)
}
void DexedAudioProcessor::updateUI() {
AudioProcessorEditor *editor = getActiveEditor();
if ( editor == NULL ) {
TRACE("no editor found!?");
return;
}
DexedAudioProcessorEditor *dexedEditor = (DexedAudioProcessorEditor *) editor;
dexedEditor->updateUI();
}
AudioProcessorEditor* DexedAudioProcessor::createEditor() {
return new DexedAudioProcessorEditor (this);
}
void DexedAudioProcessor::log(const char *source, const char *fmt, ...) {
char output[4096];
va_list argptr;
va_start(argptr, fmt);
vsnprintf(output, 4095, fmt, argptr);
va_end(argptr);
String dest;
dest << source << " " << output;
Logger::writeToLog(dest);
}