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#include "dssi-render.h"
#include "midi/midi_loader.h"
#define DEBUG 0
/* Instead of creating an alsa midi input, we fill in two events
* note-on and note-off */
inline int
min(int x, int y) {
return (x < y) ? x : y;
}
// where is nframes defined?
void ladspa_run_sample_callback(event_table_t *event_table, void *userdata){
nframes = event_table->nframes_since_last;
nframes = nframes > 0 ? nframes : 1;
//is it nframes that has the negative value? or event_table length? well they are both size_t, can't be negative. i guess need to add prints to see what's going on
if(DEBUG){
printf("nframes: 0x%x\n", nframes);
printf("outs: 0x%x\n", outs);
}
for (int i = 0; i < outs; i++) {
pluginOutputBuffers[i] = (float *)realloc(pluginOutputBuffers[i], nframes * sizeof(float));
if (DEBUG){
printf("i: 0x%x buff: 0x%x\n", i, pluginOutputBuffers[i]);
}
memset(pluginOutputBuffers[i], 0, nframes * sizeof(float));
}
connect_ports();
if (descriptor->run_synth) {
descriptor->run_synth(instanceHandle,
nframes,
event_table->events,
event_table->length
);
} else if (descriptor->run_multiple_synths) {
descriptor->run_multiple_synths(1,
&instanceHandle,
nframes,
&event_table->events,
&event_table->length);
}
/* Interleaving for libsndfile. */
float sf_output[nchannels * nframes];
for (int i = 0; i < nframes; i++) {
/* First, write all the obvious channels */
for (int j = 0; j < min(outs, nchannels); j++) {
/* If outs > nchannels, we *could* do mixing - but don't. */
sf_output[i * nchannels + j] = pluginOutputBuffers[j][i];
}
/* Then, if user wants *more* output channels than there are
* audio output ports (ie outs < nchannels), copy the last audio
* out to all the remaining channels. If outs >= nchannels, this
* loop is never entered. */
for (int j = outs; j < nchannels; j++) {
sf_output[i * nchannels + j] = pluginOutputBuffers[outs - 1][i];
}
}
if (clip) {
for (int i = 0; i < nframes * nchannels; i++) {
if (!finite(sf_output[i])) {
if (!have_warned) {
have_warned = 1;
fprintf(stderr,
"%s: Warning: clipping NaN or Inf in synthesized data\n",
my_name);
}
if (sf_output[i] < 0.0f) {
sf_output[i] = -1.0f;
} else {
sf_output[i] = 1.0f;
}
} else {
if (sf_output[i] < -1.0f) {
if (!have_warned) {
have_warned = 1;
fprintf(stderr,
"%s: Warning: clipping out-of-bounds value in synthesized data\n",
my_name);
}
sf_output[i] = -1.0f;
} else if (sf_output[i] > 1.0f) {
if (!have_warned) {
have_warned = 1;
fprintf(stderr,
"%s: Warning: clipping out-of-bounds value in synthesized data\n",
my_name);
}
sf_output[i] = 1.0f;
}
}
}
} else {
for (int i = 0; i < nframes * nchannels; i++) {
if (!finite(sf_output[i])) {
fprintf(stderr, "%s: Error: NaN or Inf in synthesized data\n",
my_name);
exit(1);
}
if (sf_output[i] > 1.0f
|| sf_output[i] < -1.0f) {
fprintf(stderr, "%s: Error: sample data out of bounds\n",
my_name);
exit(1);
}
}
}
/* Write the audio */
//here it writes to file, and it has a buffer sf_output, most likely filled from plugin. anyway, tempo problem, if it doesn't pick correct note duration it must be something
//with either reading tempo/setting tempo settings to player, or calulcations of time then it calulate each note. I don't think so...it's because it doesn't do the
//calculations gradually, seems to do them all at the end (like a final loop) and whatever the last tempo was, it uses that for the whole piece, you know what I mean?
//well it's not how send_events works, it send events one by one, so it should work fine. how did you test tempo? ok I'm going to make some perfect test-cases, sound good? yep
//
if ((items_written = sf_writef_float(outfile,
sf_output,
nframes)) != nframes) {
fprintf(stderr, "%s: Error: can't write data to output file %s\n",
my_name, output_file);
fprintf(stderr, "%s: %s\n", my_name, sf_strerror(outfile));
return;
}
if(DEBUG){
fprintf(stdout, "%s: Wrote %d frames to %s\n",
my_name, items_written, output_file);
}
}
void ladspa_run_synth(void){
load_midi_file(midi_filename, sample_rate, ladspa_run_sample_callback, NULL);
}
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