/*****************************************************************************
*
* PROCESS_BUFFER()
*
* callback for jack to read audio data from ringbuffer
* jack uses a realtime priority thread for this
*
*****************************************************************************/
int
process_buffer(jack_nframes_t nframes, void *arg) {
jack_default_audio_sample_t *in1;
jack_default_audio_sample_t *in2;
jack_default_audio_sample_t *out1;
jack_default_audio_sample_t *out2;
int portion1;
int portion2;
int j, osc, lfo;
/* Get the input and output buffers for audio data */
in1 = jack_port_get_buffer (input_port1, nframes);
in2 = jack_port_get_buffer (input_port2, nframes);
out1 = jack_port_get_buffer (output_port1, nframes);
out2 = jack_port_get_buffer (output_port2, nframes);
/* check for jack not running or global shutdown so we don't get hung on mutex */
if (!jack_running || shutdown) {
return 0;
}
/* copy circular buffers and update index */
/* WARNING: needs to handle circular buffer properly */
if ((buffer_read_index + nframes) > buffer_size) {
portion1 = buffer_size - buffer_read_index;
portion2 = nframes - portion1;
memcpy (&(input_buffer1[buffer_read_index]), in1, sizeof (jack_default_audio_sample_t) * portion1);
memcpy (&(input_buffer2[buffer_read_index]), in2, sizeof (jack_default_audio_sample_t) * portion1);
memcpy (input_buffer1, &(in1[portion1]), sizeof (jack_default_audio_sample_t) * portion2);
memcpy (input_buffer2, &(in2[portion1]), sizeof (jack_default_audio_sample_t) * portion2);
memcpy (out1, &(output_buffer1[buffer_read_index]), sizeof (jack_default_audio_sample_t) * portion1);
memcpy (out2, &(output_buffer2[buffer_read_index]), sizeof (jack_default_audio_sample_t) * portion1);
memcpy (&(out1[portion1]), output_buffer1, sizeof (jack_default_audio_sample_t) * portion2);
memcpy (&(out2[portion1]), output_buffer2, sizeof (jack_default_audio_sample_t) * portion2);
}
else {
memcpy (&(input_buffer1[buffer_read_index]), in1, sizeof (jack_default_audio_sample_t) * nframes);
memcpy (&(input_buffer2[buffer_read_index]), in2, sizeof (jack_default_audio_sample_t) * nframes);
memcpy (out1, &(output_buffer1[buffer_read_index]), sizeof (jack_default_audio_sample_t) * nframes);
memcpy (out2, &(output_buffer2[buffer_read_index]), sizeof (jack_default_audio_sample_t) * nframes);
}
buffer_read_index += nframes;
buffer_read_index %= buffer_size;
/* Update buffer full and buffer needed counters */
buffer_full -= nframes;
if(setting_jack_transport_mode)
{
/* jack transport sync */
jack_state = jack_transport_query (client, &jack_pos);
/* reinit sync vars if transport is just started */
if((jack_prev_state == JackTransportStopped) && (jack_state == JackTransportStarting))
{
#ifdef EXTRA_DEBUG
if (debug)
{
fprintf (stderr, "Starting sync.\n");
}
#endif
need_resync = 1;
}
if(jack_state == JackTransportRolling)
{
if(need_resync)
{
prev_frame = jack_pos.frame - nframes;
frames_per_beat = sample_rate / global.bps;
frames_per_tick = sample_rate / (jack_pos.ticks_per_beat * global.bps);
current_frame = 0;
need_resync = 0;
}
/* Handle BPM change */
if(jack_pos.beats_per_minute && (global.bps != jack_pos.beats_per_minute / 60.0))
{
param[0].cc_val[program_number] = jack_pos.beats_per_minute - 64;
param[0].callback (main_window, (gpointer)(&(param[0])));
}
/* frame-based sync */
current_frame = (jack_pos.frame - prev_frame);
prev_frame = jack_pos.frame;
if(current_frame != nframes)
{
/* whoooaaaa, we're traveling through time! */
phase_correction = current_frame - nframes;
}
/* do the actual sync */
if(phase_correction && (setting_jack_transport_mode == JACK_TRANSPORT_TNP))
{
#ifdef EXTRA_DEBUG
if (debug)
{
fprintf (stderr, "Out of sync. Phase correction: %d\n", phase_correction);
}
#endif
part.delay_write_index += phase_correction;
while(part.delay_write_index < 0.0)
part.delay_write_index += part.delay_bufsize;
while(part.delay_write_index >= part.delay_bufsize)
part.delay_write_index -= part.delay_bufsize;
part.chorus_lfo_index_a += phase_correction * part.chorus_lfo_adjust;
while(part.chorus_lfo_index_a < 0.0)
part.chorus_lfo_index_a += F_WAVEFORM_SIZE;
while(part.chorus_lfo_index_a >= F_WAVEFORM_SIZE)
part.chorus_lfo_index_a -= F_WAVEFORM_SIZE;
part.chorus_lfo_index_b = part.chorus_lfo_index_a + (F_WAVEFORM_SIZE * 0.25);
while(part.chorus_lfo_index_b < 0.0)
part.chorus_lfo_index_b += F_WAVEFORM_SIZE;
while(part.chorus_lfo_index_b >= F_WAVEFORM_SIZE)
part.chorus_lfo_index_b -= F_WAVEFORM_SIZE;
part.chorus_lfo_index_c = part.chorus_lfo_index_a + (F_WAVEFORM_SIZE * 0.5);
while(part.chorus_lfo_index_c < 0.0)
part.chorus_lfo_index_c += F_WAVEFORM_SIZE;
while(part.chorus_lfo_index_c >= F_WAVEFORM_SIZE)
part.chorus_lfo_index_c -= F_WAVEFORM_SIZE;
part.chorus_lfo_index_d = part.chorus_lfo_index_a + (F_WAVEFORM_SIZE * 0.75);
while(part.chorus_lfo_index_d < 0.0)
part.chorus_lfo_index_d += F_WAVEFORM_SIZE;
while(part.chorus_lfo_index_d >= F_WAVEFORM_SIZE)
part.chorus_lfo_index_d -= F_WAVEFORM_SIZE;
for (osc = 0; osc < NUM_OSCS; osc++) {
for (j = 0; j < setting_polyphony; j++) {
if (patch->osc_freq_base[osc] >= FREQ_BASE_TEMPO)
{
switch (patch->freq_mod_type[osc]) {
case MOD_TYPE_LFO:
tmp_1 = part.lfo_out[patch->freq_lfo[osc]];
break;
case MOD_TYPE_OSC:
tmp_1 = ( voice[j].osc_out1[part.osc_freq_mod[osc]] + voice[j].osc_out2[part.osc_freq_mod[osc]] ) * 0.5;
break;
case MOD_TYPE_VELOCITY:
tmp_1 = voice[j].velocity_coef_linear;
break;
default:
tmp_1 = 0.0;
break;
}
voice[j].index[osc] += phase_correction
* halfsteps_to_freq_mult ( ( tmp_1
* patch->freq_lfo_amount[osc] )
+ part.osc_pitch_bend[osc]
+ patch->osc_transpose[osc] )
* voice[j].osc_freq[osc] * wave_period;
while (voice[j].index[osc] < 0.0)
voice[j].index[osc] += F_WAVEFORM_SIZE;
while (voice[j].index[osc] >= F_WAVEFORM_SIZE)
voice[j].index[osc] -= F_WAVEFORM_SIZE;
}
}
}
for (lfo = 0; lfo < NUM_LFOS; lfo++) {
if (patch->lfo_freq_base[lfo] >= FREQ_BASE_TEMPO)
{
part.lfo_index[lfo] += phase_correction
* part.lfo_freq[lfo]
* halfsteps_to_freq_mult (patch->lfo_transpose[lfo] + part.lfo_pitch_bend[lfo])
* wave_period;
while (part.lfo_index[lfo] < 0.0)
part.lfo_index[lfo] += F_WAVEFORM_SIZE;
while (part.lfo_index[lfo] >= F_WAVEFORM_SIZE)
part.lfo_index[lfo] -= F_WAVEFORM_SIZE;
}
}
phase_correction = 0;
}
}
else if ((jack_state == JackTransportStopped) && (jack_prev_state == JackTransportRolling))
{
/* send NOTE_OFFs on transport stop */
engine_notes_off();
}
jack_prev_state = jack_state;
} /* if(setting_jack_transport) */
/* Signal the engine that there's space again */
if (pthread_mutex_trylock (&buffer_mutex) == 0) {
pthread_cond_broadcast (&buffer_has_space);
pthread_mutex_unlock (&buffer_mutex);
}
return 0;
}
/*****************************************************************************
* process_phase_sync()
*
* Process a phase sync internal MIDI message. Currently used for syncing
* JACK Transport, this function performs phase correction for a given voice.
*****************************************************************************/
void
process_phase_sync(MIDI_EVENT *event, unsigned int part_num)
{
PART *part = get_part(part_num);
PATCH_STATE *state = get_active_state(part_num);
DELAY *delay = get_delay(part_num);
CHORUS *chorus = get_chorus(part_num);
VOICE *voice;
int voice_num;
int osc;
int lfo;
int phase_correction = event->value;
sample_t f_phase_correction = (sample_t) phase_correction;
sample_t tmp_1;
delay->write_index += phase_correction;
while (delay->write_index < 0.0) {
delay->write_index += delay->bufsize;
}
while (delay->write_index >= delay->bufsize) {
delay->write_index -= delay->bufsize;
}
chorus->lfo_index_a += f_phase_correction * chorus->lfo_adjust;
while (chorus->lfo_index_a < 0.0) {
chorus->lfo_index_a += F_WAVEFORM_SIZE;
}
while (chorus->lfo_index_a >= F_WAVEFORM_SIZE) {
chorus->lfo_index_a -= F_WAVEFORM_SIZE;
}
chorus->lfo_index_b = chorus->lfo_index_a + (F_WAVEFORM_SIZE * 0.25);
while (chorus->lfo_index_b < 0.0) {
chorus->lfo_index_b += F_WAVEFORM_SIZE;
}
while (chorus->lfo_index_b >= F_WAVEFORM_SIZE) {
chorus->lfo_index_b -= F_WAVEFORM_SIZE;
}
chorus->lfo_index_c = chorus->lfo_index_a + (F_WAVEFORM_SIZE * 0.5);
while (chorus->lfo_index_c < 0.0) {
chorus->lfo_index_c += F_WAVEFORM_SIZE;
}
while (chorus->lfo_index_c >= F_WAVEFORM_SIZE) {
chorus->lfo_index_c -= F_WAVEFORM_SIZE;
}
chorus->lfo_index_d = chorus->lfo_index_a + (F_WAVEFORM_SIZE * 0.75);
while (chorus->lfo_index_d < 0.0) {
chorus->lfo_index_d += F_WAVEFORM_SIZE;
}
while (chorus->lfo_index_d >= F_WAVEFORM_SIZE) {
chorus->lfo_index_d -= F_WAVEFORM_SIZE;
}
for (voice_num = 0; voice_num < setting_polyphony; voice_num++) {
voice = get_voice(part_num, voice_num);
for (osc = 0; osc < NUM_OSCS; osc++) {
if (state->osc_freq_base[osc] >= FREQ_BASE_TEMPO) {
switch (state->freq_mod_type[osc]) {
case MOD_TYPE_LFO:
tmp_1 = part->lfo_out[state->freq_lfo[osc]];
break;
case MOD_TYPE_OSC:
tmp_1 = (voice->osc_out1[part->osc_freq_mod[osc]] +
voice->osc_out2[part->osc_freq_mod[osc]]) * 0.5;
break;
case MOD_TYPE_VELOCITY:
tmp_1 = voice->velocity_coef_linear;
break;
default:
tmp_1 = 0.0;
break;
}
voice->index[osc] +=
f_phase_correction *
halfsteps_to_freq_mult((tmp_1 *
state->freq_lfo_amount[osc]) +
part->osc_pitch_bend[osc] +
state->osc_transpose[osc] +
state->voice_osc_tune[voice->id] ) *
voice->osc_freq[osc] * wave_period;
while (voice->index[osc] < 0.0) {
voice->index[osc] += F_WAVEFORM_SIZE;
}
while (voice->index[osc] >= F_WAVEFORM_SIZE) {
voice->index[osc] -= F_WAVEFORM_SIZE;
}
}
}
}
for (lfo = 0; lfo < NUM_LFOS; lfo++) {
if (state->lfo_freq_base[lfo] >= FREQ_BASE_TEMPO) {
part->lfo_index[lfo] +=
f_phase_correction *
part->lfo_freq[lfo] *
halfsteps_to_freq_mult(state->lfo_transpose[lfo] + part->lfo_pitch_bend[lfo]) *
wave_period;
while (part->lfo_index[lfo] < 0.0) {
part->lfo_index[lfo] += F_WAVEFORM_SIZE;
}
while (part->lfo_index[lfo] >= F_WAVEFORM_SIZE) {
part->lfo_index[lfo] -= F_WAVEFORM_SIZE;
}
}
}
}