static void play_note2(Data *data, int time, note_t note){
//printf("Playing %f\n",note.pitch);
Voice *voice = data->voices_not_playing;
if(voice==NULL){
printf("no more free voices\n");
return;
}
RT_remove_voice(&data->voices_not_playing, voice);
RT_add_voice(&data->voices_playing, voice);
*(voice->myUI._gate_control) = 1.0f;
*(voice->myUI._freq_control) = midi_to_hz(note.pitch);
*(voice->myUI._gain_control) = velocity2gain(note.velocity);
voice->note_num = note.pitch;
voice->note_id = note.id;
voice->seqblock = note.seqblock;
voice->frames_since_stop = 0;
voice->delta_pos_at_start = time;
voice->delta_pos_at_end = -1;
}
static void play_note(struct SoundPlugin *plugin, int64_t time, float note_num, int64_t note_id, float volume, float pan){
Data *data = (Data*)plugin->data;
//printf("Playing %d\n",note_num);
Voice *voice = data->voices_not_playing;
if(voice==NULL){
printf("no more free voices\n");
return;
}
RT_remove_voice(&data->voices_not_playing, voice);
RT_add_voice(&data->voices_playing, voice);
//voice->dsp_instance->init((int)data->samplerate);
*(voice->myUI._gate_control) = 1.0f;
*(voice->myUI._freq_control) = midi_to_hz(note_num);
*(voice->myUI._gain_control) = velocity2gain(volume);
voice->note_num = note_num;
voice->note_id = note_id;
voice->frames_since_stop = 0;
voice->delta_pos_at_start = time;
voice->delta_pos_at_end = -1;
}
static void set_note_pitch2(Data *data, int time, note_t note){
Voice *voice = data->voices_playing;
//printf("Setting volume %f / %f\n",volume,velocity2gain(volume));
while(voice!=NULL){
if(is_note(note, voice->note_id, voice->seqblock))
*(voice->myUI._freq_control) = midi_to_hz(note.pitch);
voice=voice->next;
}
}
static void set_note_pitch(struct SoundPlugin *plugin, int64_t time, float note_num, int64_t note_id, float pitch){
Data *data = (Data*)plugin->data;
Voice *voice = data->voices_playing;
//printf("Setting volume %f / %f\n",volume,velocity2gain(volume));
while(voice!=NULL){
if(voice->note_id==note_id)
*(voice->myUI._freq_control) = midi_to_hz(pitch);
voice=voice->next;
}
}
int main( int argc, const char ** argv )
{
PaStreamParameters outputParameters;
PaError pa_err;
SF_INFO sfinfo;
int i;
for (i = 1; i < argc; i++)
{
if (!strncmp(argv[i], "-m", 2))
{
mute = 1;
}
if (!strncmp(argv[i], "-o", 2))
{
to_stdout = 1;
}
}
gen_default_drums();
pa_err = Pa_Initialize();
outputParameters.device = Pa_GetDefaultOutputDevice();
outputParameters.channelCount = CHANNELS;
outputParameters.sampleFormat = paFloat32;
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
pa_err = Pa_OpenStream( &stream, NULL, &outputParameters, SAMPLE_RATE, BUFFER_SIZE, paClipOff, audio_callback, NULL );
sfinfo.samplerate = SAMPLE_RATE;
sfinfo.channels = CHANNELS;
sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_FLOAT;
wave_output = sf_open( OUTPUT_FILE, SFM_WRITE, &sfinfo );
convert_buf = (int32_t*)malloc(SAMPLE_RATE * 4 * 2);
srand( time( 0 ) );
bass_z = 0.0f;
bass_freq = midi_to_hz( BASE_NOTE );
pa_err = Pa_StartStream( stream );
while( 1 )
Pa_Sleep( 10000 );
sf_close( wave_output );
pa_err = Pa_StopStream( stream );
pa_err = Pa_CloseStream( stream );
Pa_Terminate();
return 0;
}
int main( void )
{
PaStreamParameters outputParameters;
PaError pa_err;
SF_INFO sfinfo;
signal( SIGINT, interrupt );
gen_default_drums();
pa_err = Pa_Initialize();
outputParameters.device = Pa_GetDefaultOutputDevice();
outputParameters.channelCount = CHANNELS;
outputParameters.sampleFormat = paFloat32;
outputParameters.suggestedLatency =
Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
pa_err = Pa_OpenStream( &stream, NULL, &outputParameters, SAMPLE_RATE,
BUFFER_SIZE, paClipOff, audio_callback, NULL );
sfinfo.samplerate = SAMPLE_RATE;
sfinfo.channels = CHANNELS;
sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_FLOAT;
wave_output = sf_open( OUTPUT_FILE, SFM_WRITE, &sfinfo );
srand( time( 0 ) );
bass_z = 0.0f;
bass_freq = midi_to_hz( BASE_NOTE );
pa_err = Pa_StartStream( stream );
while( 1 )
Pa_Sleep( 1000 );
sf_close( wave_output );
pa_err = Pa_StopStream( stream );
pa_err = Pa_CloseStream( stream );
Pa_Terminate();
return 0;
}
static void play_note(struct SoundPlugin *plugin, int64_t time, int note_num, float volume){
Data *data = (Data*)plugin->data;
//printf("Playing %d\n",note_num);
Voice *voice = data->voices_not_playing;
if(voice==NULL){
printf("no more free voices\n");
return;
}
RT_remove_voice(&data->voices_not_playing, voice);
RT_add_voice(&data->voices_playing, voice);
*(voice->myUI._gate_control) = 1.0f;
*(voice->myUI._freq_control) = midi_to_hz(note_num);
*(voice->myUI._gain_control) = velocity2gain(volume);
voice->note_num = note_num;
}
void runSO_404( LV2_Handle arg, uint32_t nframes ) {
so_404* so=(so_404*)arg;
lv2_event_begin(&so->in_iterator,so->MidiIn);
float* outbuffer=so->output;
if(*so->controlmode_p >0) {
so->cutoff=*so->cutoff_p;
so->portamento=*so->portamento_p;
so->release=*so->release_p;
so->volume=*so->volume_p;
so->envmod=*so->envmod_p;
so->resonance=*so->resonance_p;
}
int i;
for( i=0; i<nframes; i++ ) {
while(lv2_event_is_valid(&so->in_iterator)) {
uint8_t* data;
LV2_Event* event= lv2_event_get(&so->in_iterator,&data);
if (event->type == 0) {
so->event_ref->lv2_event_unref(so->event_ref->callback_data, event);
} else if(event->type==so->midi_event_id) {
if(event->frames > i) {
break;
} else {
const uint8_t* evt=(uint8_t*)data;
if((evt[0]&MIDI_CHANNELMASK)==(int) (*so->channel_p)) {
if((evt[0]&MIDI_COMMANDMASK)==MIDI_NOTEON) {
unsigned int note = evt[1];
so->tfreq=midi_to_hz(note);
if( so->noteson == 0 )
{
so->freq = so->tfreq;
so->amp=1.0;
so->vel = ((float)evt[2]);
so->env=so->vel/127.0;
so->cdelay = 0;
}
so->noteson += 1;
}
else if((evt[0]&MIDI_COMMANDMASK)==MIDI_NOTEOFF ) {
so->noteson -= 1;
if(so->noteson<0) {
so->noteson=0;
}
}
else if((*so->controlmode_p<=0) && (evt[0]&MIDI_COMMANDMASK)==MIDI_CONTROL ) {
unsigned int command_val=evt[2];
switch(evt[1]) {
case 74:
so->cutoff =command_val;
break;
case 65:
so->portamento = command_val;
break;
case 72:
so->release = command_val;
break;
case 7:
so->volume = command_val;
break;
case 79:
so->envmod = command_val;
break;
case 71:
so->resonance = command_val;
break;
}
}
}
}
}
lv2_event_increment(&so->in_iterator);
}
if( so->cdelay <= 0 )
{
so->freq = ((so->portamento/127.0)*0.9)*so->freq + (1.0-((so->portamento/127.0)*0.9))*so->tfreq;
if( so->noteson > 0 ) {
so->amp *= 0.99;
} else {
so->amp *= 0.5;
}
so->env*=0.8+powf(so->release/127.0,0.25)/5.1;
so->fcutoff = powf(so->cutoff/127.0,2.0)+powf(so->env,2.0)*powf(so->envmod/127.0,2.0);
so->fcutoff = tanh(so->fcutoff);
so->freso = powf(so->resonance/130.0,0.25);
so->cdelay = so->samplerate/100;
}
so->cdelay--;
float max = so->samplerate / so->freq;
float sample = (so->phase/max)*(so->phase/max)-0.25;
so->phase++;
if( so->phase >= max ) {
so->phase -= max;
}
if(so->vel>100) {
sample*=so->env;
} else {
sample*=so->amp;
}
so->fpos += so->fspeed;
so->fspeed *= so->freso;
so->fspeed += (sample-so->fpos)*so->fcutoff;
sample = so->fpos;
sample = sample*0.5+so->lastsample*0.5;
so->lastsample = sample;
outbuffer[i] = sample * (so->volume/127.0);
}
}
static bool load_sample_with_libsndfile(Data *data, const wchar_t *filename){
SF_INFO sf_info; memset(&sf_info,0,sizeof(sf_info));
data->num_different_samples = 1;
float *samples = load_interleaved_samples(filename, &sf_info);
if(samples==NULL){
fprintf(stderr,"could not open file\n");
return false;
}
{
int num_channels = sf_info.channels;
printf("Num channels: %d\n",num_channels);
if(num_channels > 2) // TODO
num_channels = 2;
int ch;
for(ch=0;ch<num_channels;ch++){
Sample *sample = (Sample*)&data->samples[ch];
sample->num_frames = sf_info.frames;
sample->sound = malloc(sizeof(float)*sample->num_frames);
}
int interleaved_pos=0;
int i;
for(i=0;i<sf_info.frames;i++){
for(ch=0;ch<sf_info.channels;ch++){
if(ch<2){
Sample *sample=(Sample*)&data->samples[ch];
sample->sound[i] = samples[interleaved_pos];
}
interleaved_pos++;
}
}
for(ch=0;ch<num_channels;ch++){
Sample *sample=(Sample*)&data->samples[ch];
set_legal_loop_points(sample,-1,-1); // By default, don't loop, but if set, loop all.
if((sf_info.format&0xffff0000) == SF_FORMAT_WAV){
printf("format: 0x%x. sections: %d, num_frames: %d. SF_FORMAT_WAV: 0x%x. og: 0x%x\n",sf_info.format,sf_info.sections,(int)sf_info.frames,SF_FORMAT_WAV,sf_info.format&SF_FORMAT_WAV);
set_wav_loop_points(sample,filename);
}
if(num_channels==1)
sample->ch = -1; // i.e play the sample in both channels.
else
sample->ch = ch;
int i;
for(i=1;i<128;i++){
Note *note=(Note*)&data->notes[i];
note->num_samples = num_channels;
note->samples[ch] = sample;
sample->frequency_table[i] = sf_info.samplerate * midi_to_hz(i)/midi_to_hz(48);
//printf("%d: %f, data: %f, sample: %f, midifreq: %f\n",i,sample->samplerate,(float)data->samplerate,(float)sf_info.samplerate,midi_to_hz(i));
}
}
}
return true;
}
static double get_ratio(int sample_note_num, int play_note_num){
return midi_to_hz(sample_note_num) / midi_to_hz(play_note_num);
}
static int audio_callback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
int i;
float *out = (float*)outputBuffer;
float b, d, f, mod;
int beat;
float v;
for( i = 0; i < framesPerBuffer; i++ )
{
d = 0;
beat = global_frame % ( SAMPLE_RATE / ( DEFAULT_BPM / 60 ) / 4 );
if( beat == 0 )
{
if( beat_count % 4 == 0 )
bass_lfofreq = ((float)DEFAULT_BPM/60.0f) * flt_freq[ rand()%4 ];
if( rand()%RANDOM_NOTE_CHANGE_CHANCE == 0 )
{
bass_freq = midi_to_hz( BASE_NOTE + minor_scale[ rand()%7 ] + ((rand()%RANDOM_OCTAVE_JUMP_CHANCE==0)?12:0) );
}
if( ( beat_count % 4 == 0 ) && ( rand()%RANDOM_GLITCH_CHANCE == 0 ) )
{
bass_fmindex = (float)( rand()%999 + 1 );
}
if( ( beat_count % 4 == 0 ) && ( rand()%RANDOM_GLITCH_RETURN_CHANCE == 0 ) )
{
bass_fmindex = 0.0f;
}
if( rand()%RANDOM_MODULATION == 0 )
{
bass_fmmod = (float)( rand()%3 + 1 );
}
if( beat_count % 16 == 0 )
bd_time = 0;
if( beat_count % 16 == 8 )
{
if( rand()%RANDOM_SNARE_SILENCE_CHANCE == 0 )
bass_vol = 0.0f;
sd_time = 0;
}
if( ( beat_count % 16 == 10 ) && ( rand()%RANDOM_SNARE_CHANCE == 0 ) )
sd_time = 0;
if( beat_count % 16 == 12 )
bass_vol = 1.0f;
if( beat_count % 16 == 6 )
bd_time = 0;
if( beat_count % 2 == 0 )
hh_time = 0;
beat_count++;
}
if( bd_time >= 0 )
{
d = bd[bd_time++];
if( bd_time > SAMPLE_RATE )
bd_time = -1;
}
if( sd_time >= 0 )
{
d = d * 0.8f + sd[sd_time++];
if( sd_time > SAMPLE_RATE )
sd_time = -1;
}
if( hh_time >= 0 )
{
d = d * 0.8f + hh[hh_time++] * 0.05f;
if( hh_time > SAMPLE_RATE )
hh_time = -1;
}
bass_lfoval = sin( 2.0 * M_PI * bass_lfofreq * (float)global_frame / (float)SAMPLE_RATE );
bass_lfoval = bass_lfoval/100.0f + 0.99f;
mod = sin( 2.0 * M_PI * bass_fmmod * bass_freq * (float)global_frame / (float)SAMPLE_RATE );
b = sgn( sin( 2.0 * M_PI * ( bass_freq * (float)global_frame / (float)SAMPLE_RATE ) + mod * bass_fmindex ) );
bass_z = bass_lfoval * bass_z + ( 1.0f - bass_lfoval ) * b;
v = 0.3f;
out[i] = bass_vol * bass_z * v + d * ( 1.0f-v );
if( out[i] > 1.0f ) out[i] = 1.0f;
if( out[i] < -1.0f ) out[i] = -1.0f;
global_frame++;
}
sf_write_float( wave_output, out, framesPerBuffer );
if (to_stdout)
{
for (i = 0; i < framesPerBuffer; i++)
{
double f = (double)out[i];
convert_buf[i] = (int32_t)(f * 0x7fffffffL);
}
write(STDOUT_FILENO, convert_buf, framesPerBuffer * 4);
}
if (mute)
memset(out, 0, sizeof(float) * framesPerBuffer);
return paContinue;
}
/* main function that makes the dubstep sound */
static int audio_callback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
int i;
float *out = (float*)outputBuffer;
float b, d, f, mod;
int beat;
float v;
for( i = 0; i < framesPerBuffer; i++ )
{
d = 0;
beat = global_frame % ( SAMPLE_RATE / ( DEFAULT_BPM / 60 ) / 4 );
/* if the stream has hit a beat */
if( beat == 0 )
{
/* 4 beats is half a measure; alter the filter's LFO */
if( beat_count % 4 == 0 )
bass_lfofreq = ((float)DEFAULT_BPM/60.0f) *
flt_freq[ rand()%4 ];
/* change the note by choosing another from the provided scale */
if( rand()%RANDOM_NOTE_CHANGE_CHANCE == 0 )
{
bass_freq = midi_to_hz( BASE_NOTE + minor_scale[ rand()%7 ] +
((rand()%RANDOM_OCTAVE_JUMP_CHANCE==0)?12:0) );
}
/* alter the FM index (modulator amplitude), to add overtones to
the bassline, thus creating the 'glitched' sound */
if( ( beat_count % 4 == 0 ) && ( rand()%RANDOM_GLITCH_CHANCE == 0 ) )
{
bass_fmindex = (float)( rand()%999 + 1 );
}
/* reset the FM index to return to the normal square bass sound */
if( ( beat_count % 4 == 0 ) && ( rand()%RANDOM_GLITCH_RETURN_CHANCE == 0 ) )
{
bass_fmindex = 0.0f;
}
/* alter the FM mod (modulator frequency multiplier) */
if( rand()%RANDOM_MODULATION == 0 )
{
bass_fmmod = (float)( rand()%3 + 1 );
}
/* a bassdrum hits every two measures */
if( beat_count % 16 == 0 )
bd_time = 0;
/* sometimes, the snare won't trigger */
if( beat_count % 16 == 8 )
{
if( rand()%RANDOM_SNARE_SILENCE_CHANCE == 0 )
bass_vol = 0.0f;
sd_time = 0;
}
/* trigger a snare drum on the 10th beat of two measures combined */
if( ( beat_count % 16 == 10 ) && ( rand()%RANDOM_SNARE_CHANCE == 0 ) )
sd_time = 0;
/* reset the volume of the bassline, if it has been previously
diminished, to create the impression of a compressor */
if( beat_count % 16 == 12 )
bass_vol = 1.0f;
/* trigger a bass drum */
if( beat_count % 16 == 6 )
bd_time = 0;
/* trigger a hihat */
if( beat_count % 2 == 0 )
hh_time = 0;
beat_count++;
}
/* advance sample counters and mix the drum channel */
if( bd_time >= 0 )
{
d = bd[bd_time++];
if( bd_time > SAMPLE_RATE )
bd_time = -1;
}
if( sd_time >= 0 )
{
d = d * 0.8f + sd[sd_time++];
if( sd_time > SAMPLE_RATE )
sd_time = -1;
}
if( hh_time >= 0 )
{
d = d * 0.8f + hh[hh_time++] * 0.05f;
if( hh_time > SAMPLE_RATE )
hh_time = -1;
}
/* compute the LFO value */
bass_lfoval = sin( 2.0 * M_PI * bass_lfofreq * (float)global_frame /
(float)SAMPLE_RATE );
/* the LFO value is in range 0.00 .. 0.01 */
bass_lfoval = bass_lfoval/100.0f + 0.99f;
/* compute the modulator */
mod = sin( 2.0 * M_PI * bass_fmmod * bass_freq * (float)global_frame /
(float)SAMPLE_RATE );
/* compute the bassline */
b = sgn( sin( 2.0 * M_PI * ( bass_freq * (float)global_frame /
(float)SAMPLE_RATE ) + mod * bass_fmindex ) );
/* process the bassline through a lowpass filter */
bass_z = bass_lfoval * bass_z + ( 1.0f - bass_lfoval ) * b;
v = 0.3f;
/* final mix of the two tracks */
out[i] = bass_vol * bass_z * v + d * ( 1.0f-v );
if( out[i] > 1.0f ) out[i] = 1.0f;
if( out[i] < -1.0f ) out[i] = -1.0f;
global_frame++;
}
sf_write_float( wave_output, out, framesPerBuffer );
return paContinue;
}
static bool load_sf2_instrument(Data *data, const wchar_t *filename, int preset_bag_number, bool set_loop_on_off){
EVENTLOG_add_event(talloc_format("load_sf2_instrument -%S-", filename));
hash_t *info = SF2_get_info(filename);
if(info==NULL){
GFX_Message(NULL, "Unable to open soundfont file \"%S\"\n", filename);
return false;
}
//HASH_save(info,stdout);
hash_t *sample_infos = HASH_get_hash(info,"samples");
hash_t *presets = HASH_get_hash(info,"presets");
hash_t *preset = HASH_get_hash_at(presets, "", preset_bag_number);
if(preset==NULL){
GFX_Message(NULL, "No such preset number %d in instrument \"%S\"\n", preset_bag_number, filename);
return false;
}
int bank_num = HASH_get_int32(preset,"bank");
hash_t *instrument = NULL;
// Try to find an instrument from the region. A preset may use several instruments, but that's not supported yet. We just use the first and best/worst.
{
hash_t *instruments = HASH_get_hash(info,"instruments");
hash_t *regions = HASH_get_hash(preset,"regions");
int i;
for(i=0;i<HASH_get_array_size(regions, "");i++){
hash_t *region = HASH_get_hash_at(regions,"",i);
if(HASH_has_key(region,"instrument")==true){
instrument = HASH_get_hash(instruments,HASH_get_chars(region,"instrument"));
break;
}
}
}
if(instrument==NULL){
GFX_Message(NULL, "load_sf2_instrument: Preset \"%S\" (bank %d / preset %d) in \"%S\" doesn't point to an instrument\n",
HASH_get_string(preset,"name"), bank_num, HASH_get_int32(preset,"num"), filename
);
return false;
}
hash_t *regions = HASH_get_hash(instrument, "regions");
for(int i=0;i<128;i++){
Note *note = new Note;
data->notes[i] = note;
data->note_storage.push_back(note);
}
int num_samples=0;
int i;
for(i=0;i<HASH_get_array_size(regions, "");i++){
hash_t *region = HASH_get_hash_at(regions,"",i);
const char *sample_name = HASH_get_chars(region, "sample_name");
if(strcmp(sample_name,"<no sample!>")){
hash_t *sample_info = HASH_get_hash(sample_infos, sample_name);
int sample_num = HASH_get_int32(sample_info,"num");
Sample &sample = data->samples[num_samples++];
new (&sample) Sample();
sample.data = data;
sample.volume = 1.0f;
sample.num_frames = HASH_get_int(sample_info,"num_frames");
set_legal_loop_points(sample,-1,-1, set_loop_on_off); // By default, loop all.
set_legal_loop_points(sample,
HASH_get_int(sample_info,"loop start"),
HASH_get_int(sample_info,"loop end"),
set_loop_on_off
);
printf("Loop start / end: %d %d\n",(int)sample.loop_start,(int)sample.loop_end);
{
sample.ch = -1;
const char *type = HASH_get_chars(sample_info,"type");
if(!strcmp(type,"Left Sample") || !strcmp(type,"ROM Left Sample"))
sample.ch = 0;
if(!strcmp(type,"Right Sample") || !strcmp(type,"ROM Right Sample"))
sample.ch = 1;
}
sample.sound = SF2_load_sample(filename, sample_num);
int root_key = HASH_get_int32(region, "root key");
int coarsetune = HASH_get_int32(region, "coarse tune");
int finetune = HASH_get_int32(region, "fine tune");
printf("root: %d, coarse: %d, fine: %d, sample pitch: %d\n",root_key,coarsetune,finetune,(int)HASH_get_int(sample_info,"pitch"));
int note;
for(note=0;note<128;note++)
if(HASH_get_int(sample_info,"pitch")==255)
sample.frequency_table[note] = HASH_get_int(sample_info, "samplerate");
else
sample.frequency_table[note] = HASH_get_int(sample_info, "samplerate") * midi_to_hz(note+coarsetune+(float)finetune/100.0) / midi_to_hz(root_key);
int note_num;
for(note_num=HASH_get_int32(region,"key start");note_num<=HASH_get_int32(region,"key end");note_num++){
Note *note = const_cast<Note*>(data->notes[note_num]);
note->samples.push_back(&sample);
//printf("%d: %f. middle_note: %d, finetune: %f. Sample: %p. Frequency: %f\n",i,dest->ratio,dest->middle_note,dest->finetune,dest->interleaved_samples,get_frequency(i,dest->finetune));
}
}
}
// Optimize data->notes so that as few Note objects as possible are used (better for cache)
for(int i=0;i<128;i++){
Note *old_note = const_cast<Note*>(data->notes[i]);
old_note->sort_samples();
for(int i2=0;i2<i;i2++){
if(data->notes[i]->is_equal(data->notes[i2])){
data->notes[i] = data->notes[i2];
data->note_storage.remove(old_note);
delete old_note;
printf(" Load SF2: Move notes %d to %d. Size of note_storage: %d\n", i, i2, data->note_storage.size());
break;
}
}
}
return true;
}