void CREATE_NAME (void){
static bool has_inited = false;
if (has_inited==false) {
fill_type(&faust_type);
CLASSNAME::classInit(MIXER_get_sample_rate());
// TODO: Don't do this. Just dispatch during runtime which rt_process to call.
//
Data *data = (Data*)create_effect_plugin_data(&faust_type, NULL, NULL, MIXER_get_sample_rate(), MIXER_get_buffer_size(), false);
faust_type.name = DSP_NAME;
faust_type.num_inputs = data->voices[0].dsp_instance->getNumInputs();
faust_type.num_outputs = data->voices[0].dsp_instance->getNumOutputs();
if(data->voices[0].myUI.is_instrument()){
faust_type.is_instrument = true;
faust_type.RT_process = RT_process_instrument;
faust_type.create_plugin_data = create_instrument_plugin_data;
data->voices[0].myUI.remove_instrument_notecontrol_effects();
}else{
faust_type.is_instrument = false;
faust_type.RT_process = RT_process_effect;
faust_type.create_plugin_data = create_effect_plugin_data;
faust_type.play_note = NULL;
faust_type.set_note_volume = NULL;
faust_type.stop_note = NULL;
}
faust_type.num_effects = data->voices[0].myUI._num_effects;
has_inited = true;
}
PR_add_plugin_type(&faust_type);
}
float lineDuration(int line, int tracknum, int blocknum, int windownum){
struct Tracker_Windows *window;
struct WBlocks *wblock;
struct WTracks *wtrack = getWTrackFromNumA(windownum, &window, blocknum, &wblock, tracknum);
if (wtrack==NULL)
return 1.0;
struct Blocks *block = wblock->block;
if (line < 0){
RError("lineDuration: Line must be 0 or larger: %d",line);
return 1.0;
}
if (line >= block->num_lines){
RError("lineDuration: Line must be less than the number of lines: %d >= %d or larger: %d",line, block->num_lines);
return 1.0;
}
Place p1 = {line, 0, 1};
Place p2 = {line+1, 0, 1};
STime s1 = Place2STime(block, &p1);
STime s2 = Place2STime(block, &p2);
return (double)(s2-s1) / (double)MIXER_get_sample_rate();
}
bool SOUNDFILESAVER_write(float **outputs, int num_frames){
PaUtil_FullMemoryBarrier();
if(g_save_state==BEFORE_WRITING && is_playing()==false)
return true;
if(g_save_state==AFTER_WRITING)
return true;
bool ret=true;
int i;
float interleaved_data[num_frames*2];
int pos=0;
for(i=0;i<num_frames;i++){
interleaved_data[pos++] = outputs[0][i];
interleaved_data[pos++] = outputs[1][i];
}
//printf("Writing %d frames\n",num_frames);
if(sf_writef_float(g_sndfile, interleaved_data, num_frames) != num_frames){
g_saving_was_successful = false;
ret = false;
}
if(g_save_state==IS_WRITING){
if(is_playing()==false){
g_save_state=POST_WRITING;
}
}
if(g_save_state==POST_WRITING){
g_post_writing_left -= (float)num_frames/MIXER_get_sample_rate();
if(g_post_writing_left <= 0.0f){
stop_writing();
g_save_state=AFTER_WRITING;
}
}
return ret;
}
void CREATE_NAME (void){
fill_type(&faust_type);
Data *data = (Data*)create_effect_plugin_data(&faust_type, NULL, NULL, MIXER_get_sample_rate(), MIXER_get_buffer_size());
faust_type.data = data;
faust_type.name = DSP_NAME;
faust_type.num_inputs = data->voices[0].dsp_instance->getNumInputs();
faust_type.num_outputs = data->voices[0].dsp_instance->getNumOutputs();
if(data->voices[0].myUI.is_instrument()){
faust_type.is_instrument = true;
faust_type.RT_process = RT_process_instrument;
faust_type.create_plugin_data = create_instrument_plugin_data;
data->voices[0].myUI.remove_instrument_notecontrol_effects();
}else{
faust_type.is_instrument = false;
faust_type.RT_process = RT_process_effect;
faust_type.create_plugin_data = create_effect_plugin_data;
faust_type.play_note = NULL;
faust_type.set_note_volume = NULL;
faust_type.stop_note = NULL;
}
faust_type.num_effects = data->voices[0].myUI._num_effects;
PR_add_plugin_type(&faust_type);
}
