JailDoctor/mini
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity
Files
7ecd11a16995c52b60f89aeea8cd534f4c7131be
mini/chirp.cpp

341 lines
13 KiB
C++
Raw Blame History

#include "chirp.h"
#include <stdlib.h>
#include <string.h>
#include <SDL2/SDL.h>
//1 2 3 4 6 8 12 16 24 32
//f s c n b r
#define SILENCE 128
#define NOISE 129
#define AUDIO_NONE 0
#define AUDIO_PLAY 1
#define MAX_CHANNELS 5
typedef void (*waveform_t)(const uint16_t, const uint32_t, uint8_t*, const uint8_t, const uint8_t);
static waveform_t waveforms[6];
const static uint16_t lengths[10] = { 313, 625, 938, 1250, 1875, 2500, 3750, 5000, 7500, 10000 };
const static uint16_t tempos[10] = { 13230, 8820, 6615, 5292, 4410, 3780, 3308, 2940, 2646, 2406 };
const float periods[108] = { 1348.49207, 1272.80688, 1201.37, 1133.94214, 1070.29871, 1010.22772, 953.527893, 900.010376, 849.496887, 801.818176, 756.815613, 714.338745, 674.246033, 636.403564, 600.684875, 566.971069, 535.149475, 505.11377, 476.763947, 450.005249, 424.748352, 400.909088, 378.407806, 357.169373, 337.123016, 318.201782, 300.342438, 283.485535, 267.574738, 252.556885, 238.381973, 225.002625, 212.374176, 200.454544, 189.203888, 178.584702, 168.561508, 159.100876, 150.171234, 141.742767, 133.787354, 126.278458, 119.190987, 112.501305, 106.187096, 100.227272, 94.6019516, 89.2923508, 84.2807541, 79.5504379, 75.0856171, 70.8713837, 66.8936768, 63.139225, 59.5954933, 56.2506561, 53.0935478, 50.113636, 47.3009758, 44.6461754, 42.140377, 39.775219, 37.5428085, 35.4356918, 33.4468384, 31.5696125, 29.7977467, 28.1253281, 26.5467739, 25.056818, 23.650486, 22.3230877, 21.0701885, 19.8876095, 18.7714043, 17.7178459, 16.7234192, 15.7848072, 14.8988733, 14.0626631, 13.273387, 12.528409, 11.8252439, 11.1615429, 10.5350943, 9.94380569, 9.38570118, 8.85892296, 8.36171055, 7.89240265, 7.44943666, 7.03133202, 6.636693, 6.2642045, 5.91262197, 5.58077145, 5.26754713, 4.97190285, 4.69285059, 4.42946148, 4.18085527, 3.94620132, 3.72471833, 3.51566601, 3.3183465, 3.13210225, 2.95631051, 2.7903862 };
SDL_AudioDeviceID audio_device;
uint8_t audio_state = AUDIO_NONE;
#define RELATIVE 0
#define SEQUENTIAL 1
struct instrument_t {
uint8_t waveform = 0;
int8_t volume[8] = {0,0,0,0,0,0,0,0};
uint8_t volume_data = RELATIVE;
int8_t pitch[8] = {0,0,0,0,0,0,0,0};
uint8_t pitch_data = RELATIVE;
};
struct channel_t {
char* song = NULL;
char* song_ptr = NULL;
char* song_start = NULL;
uint32_t count = 0;
float length = 0.25f;
uint8_t volume = 32;
uint8_t octave = 4;
uint32_t tempo = 44100;
uint8_t waveform = 0;
uint8_t instrument = 0;
Uint8* play_pos;
int32_t play_len;
Uint8 play_buffer[132300];
char* stack[10] = {NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL};
uint8_t stackpos = 0;
char* labels[10] = {NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL};
instrument_t instruments[10];
};
static channel_t channels[MAX_CHANNELS];
void chirp_stop_channel(const int c) {
if (channels[c].song != NULL) {
free(channels[c].song);
channels[c].song=NULL;
}
for (int i=0;i<MAX_CHANNELS;++i) if (channels[i].song != NULL) return;
audio_state=AUDIO_NONE;
}
void audioCallback(void * userdata, uint8_t * stream, int len) {
SDL_memset(stream, 0, len);
if (audio_state == AUDIO_PLAY) {
for (int i=0;i<MAX_CHANNELS;++i) if (channels[i].song != NULL) {
int l_len=len;
uint8_t *l_stream=stream;
while( l_len > 0 ) {
while( channels[i].play_len == 0 ) {
channels[i].play_len = interpret_next_token(i);
if (channels[i].play_len == -1) {
chirp_stop_channel(i);
break;
}
channels[i].play_pos = channels[i].play_buffer;
}
if (channels[i].play_len == -1) break;
const int actual_len = ( l_len > channels[i].play_len ? channels[i].play_len : l_len );
for (int j=0;j<actual_len;++j) l_stream[j] += channels[i].play_pos[j];
l_stream += actual_len;
channels[i].play_pos += actual_len;
channels[i].play_len -= actual_len;
l_len -= actual_len;
}
}
}
}
#define COUNT (channels[c].count++ % period)
void square_waveform(const uint16_t period, const uint32_t length, uint8_t* buffer, const uint8_t volume, const uint8_t c) {
for( uint32_t i = 0; i < length; i++ ) buffer[i] = ( COUNT < (period >> 1) ? volume : -volume );
}
void saw_waveform(const uint16_t period, const uint32_t length, uint8_t* buffer, const uint8_t volume, const uint8_t c) {
for( uint32_t i = 0; i < length; i++ ) buffer[i] = -volume + uint16_t( float(COUNT) / float(period) * volume*2 );
}
void triangle_waveform(const uint16_t period, const uint32_t length, uint8_t* buffer, const uint8_t volume, const uint8_t c) {
for( uint32_t i = 0; i < length; i++ ) {
uint16_t pos = COUNT;
uint16_t half_period = period >> 1;
if (pos < half_period) {
buffer[i] = -volume + uint16_t( (float(pos) / float(half_period)) * float(volume*2) );
} else {
buffer[i] = volume - uint16_t( (float(pos-half_period) / float(half_period)) * float(volume*2) );
}
}
}
void pulse12_waveform(const uint16_t period, const uint32_t length, uint8_t* buffer, const uint8_t volume, const uint8_t c) {
for( uint32_t i = 0; i < length; i++ ) buffer[i] = ( COUNT < (period >> 3) ? volume : -volume );
}
void pulse25_waveform(const uint16_t period, const uint32_t length, uint8_t* buffer, const uint8_t volume, const uint8_t c) {
for( uint32_t i = 0; i < length; i++ ) buffer[i] = ( COUNT < (period >> 2) ? volume : -volume );
}
void noise_waveform(const uint16_t period, const uint32_t length, uint8_t* buffer, const uint8_t volume, const uint8_t c) {
for( uint32_t i = 0; i < length; i++ ) buffer[i] = rand()%2==0 ? volume : -volume;
}
void chirp_init() {
SDL_AudioSpec audioSpec = {22050, AUDIO_S8, 1, 0, 512, 0, 0, audioCallback, NULL};
audio_device = SDL_OpenAudioDevice(NULL, 0, &audioSpec, NULL, 0);
SDL_PauseAudioDevice(audio_device, 0);
waveforms[0] = &square_waveform;
waveforms[1] = &saw_waveform;
waveforms[2] = &triangle_waveform;
waveforms[3] = &pulse12_waveform;
waveforms[4] = &pulse25_waveform;
waveforms[5] = &noise_waveform;
for (uint8_t i=0;i<MAX_CHANNELS;++i) channels[i].song=NULL;
}
uint32_t interpret_note(const int c, const uint8_t note, const char param ) {
const uint32_t length = ( param == -1 ? channels[c].length : ((float)lengths[uint8_t(param)])/10000.0f ) * channels[c].tempo;
if( note == SILENCE ) { memset( channels[c].play_buffer, 0, length ); return length; }
//if( note == NOISE ) { for( uint32_t i = 0; i < length; i++ ) channels[c].play_buffer[i] = rand()%2==0 ? channels[c].volume : -channels[c].volume; return length; }
uint32_t l = 0;
uint8_t envelope_pos=0;
uint8_t* buffer = channels[c].play_buffer;
uint32_t min_len = uint32_t(0.0313f * channels[c].tempo);
uint8_t fullnote = note + channels[c].octave*12;
uint8_t fullvolume = channels[c].volume;
while (l<length) {
uint8_t actualnote = fullnote + channels[c].instruments[channels[c].instrument].pitch[envelope_pos];
if (channels[c].instruments[channels[c].instrument].pitch_data==SEQUENTIAL) fullnote = actualnote;
const uint16_t period = periods[actualnote];
uint8_t actualvolume = fullvolume + (channels[c].instruments[channels[c].instrument].volume[envelope_pos] << 3);
if (channels[c].instruments[channels[c].instrument].volume_data==SEQUENTIAL) fullvolume = actualvolume;
waveforms[channels[c].instruments[channels[c].instrument].waveform](period, min_len, buffer, actualvolume, c); //channels[c].play_buffer, channels[c].volume);
l+=min_len;
envelope_pos=(envelope_pos+1)&7;
buffer += min_len;
}
return length;
}
int chirp_play(const char* new_song) {
int c = 0;
while (c<MAX_CHANNELS && channels[c].song != NULL) c++;
if (c==MAX_CHANNELS) return -1;
channels[c].play_pos = channels[c].play_buffer;
channels[c].play_len = 0;
channels[c].length=0.25f;
channels[c].volume=32;
channels[c].octave=4;
channels[c].tempo=44100;
channels[c].waveform=0;
channels[c].stackpos=0;
channels[c].song = (char*)malloc( strlen( new_song ) + 1 );
strcpy( channels[c].song, new_song );
channels[c].song_ptr = channels[c].song_start = channels[c].song;
audio_state = AUDIO_PLAY;
return c;
}
int32_t interpret_next_token(const int c) { //uint8_t* buffer) {
char** token = &channels[c].song_ptr;
char note = 0;
char param = -1;
switch( **token ) {
case 'b': note += 2;
case 'a': note += 2;
case 'g': note += 2;
case 'f': note += 1;
case 'e': note += 2;
case 'd': note += 2;
case 'c':
param = *++*token;
if( param == '#' || param == '+' ) { note++; param = *++*token; } else if( param == '-' ) { note--; param = *++*token; }
if( param >= 48 && param <= 57 ) { param -= 48; ++*token; } else { param = -1; }
return interpret_note( c, note, param );
case 'r':
param = *++*token;
if( param >= 48 && param <= 57 ) { param -= 48; ++*token; } else { param = -1; }
return interpret_note( c, SILENCE, param );
case 'n':
param = *++*token;
if( param >= 48 && param <= 57 ) { param -= 48; ++*token; } else { param = -1; }
return interpret_note( c, NOISE, param );
case 'o':
param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].octave = (param - 48) % 8; ++*token; }
return 0;
case '>':
channels[c].octave = (channels[c].octave+1) % 8; ++*token;
return 0;
case '<':
channels[c].octave = (channels[c].octave-1) % 8; ++*token;
return 0;
case 'l':
param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].length = ((float)lengths[param - 48])/10000.0f; ++*token; }
return 0;
case 'v':
param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].volume = (param - 48) << 3; ++*token; }
return 0;
case 't':
param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].tempo = tempos[param - 48] * 10; ++*token; }
return 0;
case 'i':
param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].instrument = param - 48; ++*token; }
return 0;
/* case 'w':
param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].waveform = param - 48; ++*token; }
return 0;*/
case '{':
{
uint8_t instrument = 0;
param = *++*token;
if( param >= 48 && param <= 57 ) { instrument = param - 48; param = *++*token; }
while (param != '}') {
switch (param) {
case 'w':
param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].instruments[instrument].waveform = param - 48; param = *++*token; }
break;
case 'v':
channels[c].instruments[instrument].volume_data = RELATIVE;
param = *++*token;
if (param=='r') { channels[c].instruments[instrument].volume_data = RELATIVE; param = *++*token; }
if (param=='s') { channels[c].instruments[instrument].volume_data = SEQUENTIAL; param = *++*token; }
for (int i=0;i<8;++i) {
int8_t sign = 1;
if (param=='-') { sign=-1; param = *++*token; }
if( param >= 48 && param <= 57 ) channels[c].instruments[instrument].volume[i] = (param-48)*sign;
param = *++*token;
}
break;
case 'p':
channels[c].instruments[instrument].pitch_data = RELATIVE;
param = *++*token;
if (param=='r') { channels[c].instruments[instrument].pitch_data = RELATIVE; param = *++*token; }
if (param=='s') { channels[c].instruments[instrument].pitch_data = SEQUENTIAL; param = *++*token; }
for (int i=0;i<8;++i) {
int8_t sign = 1;
if (param=='-') { sign=-1; param = *++*token; }
if( param >= 48 && param <= 57 ) channels[c].instruments[instrument].pitch[i] = (param-48)*sign;
param = *++*token;
}
break;
case '}':
default:
param = *++*token;
break;
}
//param = *++*token;
}
return 0;
}
case '!':
channels[c].stackpos = 0;
channels[c].song_start = ++*token;
return 0;
case '=':
channels[c].stackpos = 0;
channels[c].song_ptr = channels[c].song_start;
return 0;
case '[':
param = *++*token;
if( param >= 48 && param <= 57 ) { param -= 48; ++*token; } else { param = 0; }
channels[c].labels[uint8_t(param)] = *token; //channels[c].song_ptr;
{
char* nextpos = *token;
uint8_t innerblocks=0;
while(innerblocks>0 || *nextpos!=']') {
if (*nextpos=='[') innerblocks++;
if (*nextpos==']') innerblocks--;
nextpos++;
}
channels[c].stack[channels[c].stackpos]=nextpos+1;
}
channels[c].stackpos++;
return 0;
case ']':
++*token;
channels[c].stackpos--;
channels[c].song_ptr=channels[c].stack[channels[c].stackpos];
return 0;
case '@':
param = *++*token;
if( param >= 48 && param <= 57 ) { param -= 48; ++*token; } else { param = 0; }
channels[c].stack[channels[c].stackpos]=*token;
channels[c].stackpos++;
channels[c].song_ptr=channels[c].labels[uint8_t(param)];
return 0;
case '\0':
return -1;
default:
++*token;
return 0;
};
}
void chirp_stop() {
for (int i=0;i<MAX_CHANNELS;++i) {
if (channels[i].song != NULL) {
free(channels[i].song);
channels[i].song=NULL;
}
}
audio_state=AUDIO_NONE;
}
Reference in New Issue View Git Blame Copy Permalink