// Think I found this pan calculation method in the ardour source many years ago.
// TODO: Optimize panning when smoothing is necessary.
void SMOOTH_apply_pan(Smooth *smooth, float **sound, int num_channels, int num_frames){
R_ASSERT(smooth->target_audio_will_be_modified==true);
int i;
if(num_channels>=2){
float *sound0 = sound[0];
float *sound1 = sound[1];
if(is_smoothing_necessary(smooth)==true){
//fprintf(stderr,"s1\n");
Panvals pan_start = das_get_pan_vals_vector(scale(smooth->values[0],0,1,-1,1),2);
//fprintf(stderr,"s2\n");
Panvals pan_end = das_get_pan_vals_vector(scale(smooth->values[num_frames-1],0,1,-1,1),2);
//fprintf(stderr,"s3\n");
for(i=0;i<num_frames;i++) {
float sound0i = sound0[i];
float sound1i = sound1[i];
sound0[i] = sound0i*scale(i,0,num_frames,pan_start.vals[0][0],pan_end.vals[0][0]) + sound1i*scale(i,0,num_frames,pan_start.vals[1][0],pan_end.vals[1][0]);
sound1[i] = sound0i*scale(i,0,num_frames,pan_start.vals[0][1],pan_end.vals[0][1]) + sound1i*scale(i,0,num_frames,pan_start.vals[1][1],pan_end.vals[1][1]);
}
} else {
if(smooth->value!=0.5f){
Panvals pan = get_pan_vals_vector(scale(smooth->value,0,1,-1,1), 2);
#if 1
if(smooth->value < 0.5f){
for(i=0;i<num_frames;i++)
sound0[i] += sound1[i]*pan.vals[1][0];
for(i=0;i<num_frames;i++)
sound1[i] *= pan.vals[1][1];
} else {
for(i=0;i<num_frames;i++)
sound1[i] += sound0[i]*pan.vals[0][1];
for(i=0;i<num_frames;i++)
sound0[i] *= pan.vals[0][0];
}
#else
// Same, but perhaps a bit slower.
for(i=0;i<num_frames;i++) {
float sound0i = sound0[i];
float sound1i = sound1[i];
sound0[i] = sound0i*pan.vals[0][0] + sound1i*pan.vals[1][0];
sound1[i] = sound0i*pan.vals[0][1] + sound1i*pan.vals[1][1];
}
#endif
}
}
}
}
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;
//fprintf(stderr,"playing note %d. Pitch: %d, time: %d\n",(int)note_id,(int)note_num,(int)time);
const Note *note = &data->notes[(int)note_num];
int i;
for(i=0;i<note->num_samples;i++){
if(data->voices_not_playing==NULL){
printf("No more free voices\n");
return;
}
Voice *voice = data->voices_not_playing;
RT_remove_voice(&data->voices_not_playing, voice);
RT_add_voice(&data->voices_playing, voice);
voice->last_finetune_value = data->finetune;
voice->note_num = note_num;
voice->note_id = note_id;
voice->start_volume = velocity2gain(volume);
voice->end_volume = voice->start_volume;
voice->start_pitch = note_num;
voice->end_pitch = note_num;
voice->sample = note->samples[i];
if(data->loop_onoff==true && voice->sample->loop_end > voice->sample->loop_start)
voice->pos=scale(data->startpos, // set startpos between 0 and loop_end
0,1,
0,voice->sample->loop_end);
else
voice->pos=scale(data->startpos, // set startpos between 0 and sound length
0,1,
0,voice->sample->num_frames);
voice->pan = get_pan_vals_vector(pan,voice->sample->ch==-1?1:2);
RESAMPLER_reset(voice->resampler);
ADSR_reset(voice->adsr);
ADSR_set_adsr(voice->adsr, data->a, data->h, data->d, data->s, data->r);
voice->delta_pos_at_start=time;
voice->delta_pos_at_end=-1;
voice->is_fading_out=false;
}
}
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;
if(data->voices_not_playing==NULL){
printf("No more free voices\n");
return;
}
Voice *voice = data->voices_not_playing;
RT_remove_voice(&data->voices_not_playing, voice);
RT_add_voice(&data->voices_playing, voice);
voicd->note_id = note_id;
voice->velocity = volume;
voice->pan = get_pan_vals_vector(pan,2);
voice->delta_pos_at_start=time;
voice->delta_pos_at_end=-1;
voice->audio = data->audio[0]
}
static int get_peaks(struct SoundPlugin *plugin,
float note_num,
int ch,
float das_pan,
int64_t start_time,
int64_t end_time,
float *min_value, float *max_value
)
{
Data *data = (Data*)plugin->data;
if(ch==-1){
int i;
for(i=0;i<MAX_NUM_SAMPLES;i++){
Sample *sample=(Sample*)&data->samples[i];
if(sample->sound!=NULL){
if(sample->ch==1)
return 2;
}
}
return 1;
}
int start_frame = time_to_frame(data, start_time, note_num);
int end_frame = time_to_frame(data, end_time, note_num);
{
const Note *note=&data->notes[(int)note_num];
float min=0.0f;
float max=0.0f;
int samplenum;
for(samplenum=0;samplenum<note->num_samples;samplenum++){
const Sample *sample=note->samples[samplenum];
Panvals pan = get_pan_vals_vector(das_pan, sample->ch==-1 ? 1 : 2);
int input_channel = sample->ch==-1 ? 0 : sample->ch;
float panval = pan.vals[input_channel][ch];
if(panval>0.0f){
float min2;
float max2;
get_peaks_from_sample(sample, start_frame, end_frame, &min2, &max2);
min2 *= panval;
max2 *= panval;
if(min2<min)
min=min2;
if(max2>max)
max=max2;
}
}
*min_value = min;
*max_value = max;
}
apply_adsr_to_peak(data, (start_time+end_time)/2, min_value, max_value);
return 2;
}
