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- [WIP] Migrating to jail_audio

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This commit is contained in:
Raimon Zamora
2023-01-11 18:58:15 +01:00
parent e388d046b2
commit 7ecd11a169
4 changed files with 400 additions and 53 deletions
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120
chirp.cpp
View File

@@ -13,7 +13,7 @@
#define MAX_CHANNELS 5 #define MAX_CHANNELS 5
typedef void (*waveform_t)(const uint16_t, const uint32_t, const uint8_t); //, uint8_t*, const uint8_t); typedef void (*waveform_t)(const uint16_t, const uint32_t, uint8_t*, const uint8_t, const uint8_t);
static waveform_t waveforms[6]; 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 lengths[10] = { 313, 625, 938, 1250, 1875, 2500, 3750, 5000, 7500, 10000 };
@@ -23,28 +23,35 @@ const float periods[108] = { 1348.49207, 1272.80688, 1201.37, 1133.94214, 1070.2
SDL_AudioDeviceID audio_device; SDL_AudioDeviceID audio_device;
uint8_t audio_state = AUDIO_NONE; uint8_t audio_state = AUDIO_NONE;
#define RELATIVE 0
#define SEQUENTIAL 1
struct instrument_t { struct instrument_t {
uint8_t waveform = 0; uint8_t waveform = 0;
int8_t volume[8] = {0,0,0,0,0,0,0,0}; 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}; int8_t pitch[8] = {0,0,0,0,0,0,0,0};
uint8_t pitch_data = RELATIVE;
}; };
struct channel_t { struct channel_t {
char* song = NULL; char* song = NULL;
char* song_ptr = NULL; char* song_ptr = NULL;
char* song_start = NULL;
uint32_t count = 0; uint32_t count = 0;
float length = 0.25f; float length = 0.25f;
uint8_t volume = 32; uint8_t volume = 32;
uint8_t octave = 4; uint8_t octave = 4;
uint32_t tempo = 44100; uint32_t tempo = 44100;
uint8_t waveform = 0; uint8_t waveform = 0;
uint8_t instrument = 0;
Uint8* play_pos; Uint8* play_pos;
int32_t play_len; int32_t play_len;
Uint8 play_buffer[132300]; Uint8 play_buffer[132300];
char* stack[10] = {NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL}; char* stack[10] = {NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL};
uint8_t stackpos = 0; uint8_t stackpos = 0;
char* labels[10] = {NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL}; char* labels[10] = {NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL,NULL};
instrument_t instruments[5]; instrument_t instruments[10];
}; };
static channel_t channels[MAX_CHANNELS]; static channel_t channels[MAX_CHANNELS];
@@ -90,31 +97,31 @@ void audioCallback(void * userdata, uint8_t * stream, int len) {
#define COUNT (channels[c].count++ % period) #define COUNT (channels[c].count++ % period)
void square_waveform(const uint16_t period, const uint32_t length, const uint8_t c) { 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++ ) channels[c].play_buffer[i] = ( COUNT < (period >> 1) ? channels[c].volume : -channels[c].volume ); 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, const uint8_t c) { 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++ ) channels[c].play_buffer[i] = -channels[c].volume + uint16_t( float(COUNT) / float(period) * channels[c].volume*2 ); 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, const uint8_t c) { 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++ ) { for( uint32_t i = 0; i < length; i++ ) {
uint16_t pos = COUNT; uint16_t pos = COUNT;
uint16_t half_period = period >> 1; uint16_t half_period = period >> 1;
if (pos < half_period) { if (pos < half_period) {
channels[c].play_buffer[i] = -channels[c].volume + uint16_t( (float(pos) / float(half_period)) * float(channels[c].volume*2) ); buffer[i] = -volume + uint16_t( (float(pos) / float(half_period)) * float(volume*2) );
} else { } else {
channels[c].play_buffer[i] = channels[c].volume - uint16_t( (float(pos-half_period) / float(half_period)) * float(channels[c].volume*2) ); 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, const uint8_t c) { 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++ ) channels[c].play_buffer[i] = ( COUNT < (period >> 3) ? channels[c].volume : -channels[c].volume ); 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, const uint8_t c) { 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++ ) channels[c].play_buffer[i] = ( COUNT < (period >> 2) ? channels[c].volume : -channels[c].volume ); for( uint32_t i = 0; i < length; i++ ) buffer[i] = ( COUNT < (period >> 2) ? volume : -volume );
} }
void pulse75_waveform(const uint16_t period, const uint32_t length, const uint8_t c) { 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++ ) channels[c].play_buffer[i] = ( COUNT < uint16_t(period-(period >> 2)) ? channels[c].volume : -channels[c].volume ); for( uint32_t i = 0; i < length; i++ ) buffer[i] = rand()%2==0 ? volume : -volume;
} }
void chirp_init() { void chirp_init() {
@@ -126,17 +133,34 @@ void chirp_init() {
waveforms[2] = &triangle_waveform; waveforms[2] = &triangle_waveform;
waveforms[3] = &pulse12_waveform; waveforms[3] = &pulse12_waveform;
waveforms[4] = &pulse25_waveform; waveforms[4] = &pulse25_waveform;
waveforms[5] = &pulse75_waveform; waveforms[5] = &noise_waveform;
for (uint8_t i=0;i<MAX_CHANNELS;++i) channels[i].song=NULL; 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 ) { 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; 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 == 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; } //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; }
const uint16_t period = periods[note + channels[c].octave*12]; uint32_t l = 0;
waveforms[channels[c].waveform](period, length, c); //channels[c].play_buffer, channels[c].volume); 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; return length;
} }
@@ -156,7 +180,7 @@ int chirp_play(const char* new_song) {
channels[c].stackpos=0; channels[c].stackpos=0;
channels[c].song = (char*)malloc( strlen( new_song ) + 1 ); channels[c].song = (char*)malloc( strlen( new_song ) + 1 );
strcpy( channels[c].song, new_song ); strcpy( channels[c].song, new_song );
channels[c].song_ptr = channels[c].song; channels[c].song_ptr = channels[c].song_start = channels[c].song;
audio_state = AUDIO_PLAY; audio_state = AUDIO_PLAY;
return c; return c;
@@ -209,13 +233,65 @@ int32_t interpret_next_token(const int c) { //uint8_t* buffer) {
param = *++*token; param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].tempo = tempos[param - 48] * 10; ++*token; } if( param >= 48 && param <= 57 ) { channels[c].tempo = tempos[param - 48] * 10; ++*token; }
return 0; return 0;
case 'w': case 'i':
param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].instrument = param - 48; ++*token; }
return 0;
/* case 'w':
param = *++*token; param = *++*token;
if( param >= 48 && param <= 57 ) { channels[c].waveform = param - 48; ++*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; return 0;
case '=': case '=':
channels[c].stackpos = 0; channels[c].stackpos = 0;
channels[c].song_ptr = channels[c].song; channels[c].song_ptr = channels[c].song_start;
return 0; return 0;
case '[': case '[':
param = *++*token; param = *++*token;

44
data/game.lua
View File

@@ -2,15 +2,9 @@ x=0
function _init() function _init()
text="HOLA MINI" text="HOLA MINI"
--playchirp("l0v4o4fg#o5cv2o4fg#o5cv1o4fg#o5cv4o4fg#o5cv2o4fg#o5cv1o4fg#o5c") playchirp("{0w3vs-10000000ps0-5-6-90000}{1w5v0-1-2-30000p00000000}!t9v4o2 l3i0f l6r i1l3f l5r l3f i0f r f r i1f l6r=") -- drums
playchirp("t9v4{0w0v0-1-2-30000}o2 ! l9 rrrrrrrr l3 [0 frrfg#rgr][1 frrfa#>c<a#g#]@0@1@0@1@0[2 f>fr<f>frfr<] =") -- bass
--playchirp("w3t4l0o2fv2fv4l1fv2l0fv1fv4l1fv2l0fv1fv4l1fv2l0fv1fv4fv2fv4l1fv2l0fv1fv4l1g#v2l0g#v1g#v4l1a#v2l0a#v1a#v4o3l1cv2l0cv1c") playchirp("t9v1 {1w0v44422211p037037037} l9 rrrrrrrrrrrrrrrr i1 [0 l5frfl3rl5dl3rl5<a#raa#ra#l3rl5al3rl8r>]") -- arp
--playchirp("t9 v4l0 o2fco1f#o0g# l6r o2l0f v3f v2f v1f v4l4r l0rv1rv4fv3fv2fv1fv4fco1f#o0g#l3ro2l0fco1f#o0g#l3ro2l0fv3fv2fv1fl6r=")
playchirp("w4t9l0v4o3fg#o4cv2o3fg#o4cv1o3fg#o4cl4rl0rv4o3fg#o4cv2o3fg#o4cv1o3fg#o4cl4rl0r")
playchirp("w0t9 [0v4o2l0fv3fv2fv1fv4fv3fv2fv1fl3rv4l0fv3fv2fv1frv3rv2rv1rv4fv3fv2fv1fl3rv4l0fv3fv2fv1f][1v4fv3fv2fv1fv4l3rl0dv3dv2dv1dv4l3rl0d#v3d#v2d#v1d#v4l3rl0ev3ev2ev1el3r]@0[2v4fv3fv2fv1fv4l3rl0gv3gv2gv1gv4l3rl0g#v3g#v2g#v1g#v4l3rl0av3av2av1al3r]=")
playchirp("w3t9 v3l0o2fco1f#o0g# l6r o2l0nv3nv2nv1n l5r v4l0nv3nv2nv1n v3l0o2fco1f#o0g# l3r v3l0o2fco1f#o0g# l3r o2l0nv3nv2nv1n l6r=")
end end
function _update() function _update()
@@ -19,26 +13,14 @@ function _update()
x=x+1 x=x+1
if x>160 then x=-strlen(text)*4 end if x>160 then x=-strlen(text)*4 end
end end
0 x
1 xx
2 xxx
3 xxxx
4 xxxxxx
5 xxxxxxxx
6 xxxxxxxxxxxx
7 xxxxxxxxxxxxxxxx
8 xxxxxxxxxxxxxxxxxxxxxxxx
9 xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
t9 {1w4v44422211a037037037} i1 l5 f l4 r l0 r l5 f l4 r l0 r -- 0 *
-- 1 **
w4 t9 -- 2 ***
l0 -- 3 ****
v4 o3f g# o4c -- 4 ******
v2 o3f g# o4c -- 5 ********
v1 o3f g# o4c -- 6 ************
l4r l0r -- 7 ****************
v4 o3f g# o4c -- 8 ************************
v2 o3f g# o4c -- 9 ********************************
v1 o3f g# o4c
l4r l0r

259
jail_audio.cpp Normal file
View File

@@ -0,0 +1,259 @@
#include "jail_audio.h"
#include "stb_vorbis.c"
#include <SDL2/SDL.h>
#include <stdio.h>
#include "jfile.h"
#define JA_MAX_SIMULTANEOUS_CHANNELS 5
struct JA_Sound_t {
Uint32 length {0};
Uint8* buffer {NULL};
};
struct JA_Channel_t {
JA_Sound sound;
int pos {0};
int times {0};
JA_Channel_state state { JA_CHANNEL_FREE };
};
struct JA_Music_t {
int samples {0};
int pos {0};
int times {0};
short* output {NULL};
JA_Music_state state {JA_MUSIC_INVALID};
};
JA_Music current_music{NULL};
JA_Channel_t channels[JA_MAX_SIMULTANEOUS_CHANNELS];
int JA_freq {48000};
SDL_AudioFormat JA_format {AUDIO_S16};
Uint8 JA_channels {2};
int JA_volume = 128;
SDL_AudioDeviceID sdlAudioDevice = 0;
void audioCallback(void * userdata, uint8_t * stream, int len) {
SDL_memset(stream, 0, len);
if (current_music != NULL && current_music->state == JA_MUSIC_PLAYING) {
const int size = SDL_min(len, current_music->samples*2-current_music->pos);
SDL_MixAudioFormat(stream, (Uint8*)(current_music->output+current_music->pos), AUDIO_S16, size, JA_volume);
current_music->pos += size/2;
if (size < len) {
if (current_music->times != 0) {
SDL_MixAudioFormat(stream+size, (Uint8*)current_music->output, AUDIO_S16, len-size, JA_volume);
current_music->pos = (len-size)/2;
if (current_music->times > 0) current_music->times--;
} else {
current_music->pos = 0;
current_music->state = JA_MUSIC_STOPPED;
}
}
}
// Mixar els channels mi amol
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
if (channels[i].state == JA_CHANNEL_PLAYING) {
const int size = SDL_min(len, channels[i].sound->length - channels[i].pos);
SDL_MixAudioFormat(stream, channels[i].sound->buffer + channels[i].pos, AUDIO_S16, size, JA_volume/2);
channels[i].pos += size;
if (size < len) {
if (channels[i].times != 0) {
SDL_MixAudioFormat(stream + size, channels[i].sound->buffer, AUDIO_S16, len-size, JA_volume/2);
channels[i].pos = len-size;
if (channels[i].times > 0) channels[i].times--;
} else {
JA_StopChannel(i);
}
}
}
}
}
void JA_Init(const int freq, const SDL_AudioFormat format, const int channels) {
JA_freq = freq;
JA_format = format;
JA_channels = channels;
SDL_AudioSpec audioSpec{JA_freq, JA_format, JA_channels, 0, 1024, 0, 0, audioCallback, NULL};
if (sdlAudioDevice != 0) SDL_CloseAudioDevice(sdlAudioDevice);
sdlAudioDevice = SDL_OpenAudioDevice(NULL, 0, &audioSpec, NULL, 0);
SDL_PauseAudioDevice(sdlAudioDevice, 0);
}
void JA_Quit() {
SDL_PauseAudioDevice(sdlAudioDevice, 1);
if (sdlAudioDevice != 0) SDL_CloseAudioDevice(sdlAudioDevice);
sdlAudioDevice = 0;
}
JA_Music JA_LoadMusic(const char* filename) {
int chan, samplerate;
JA_Music music = new JA_Music_t();
int fsize;
Uint8 *buffer = (Uint8*)file_getfilebuffer(filename, fsize);
// [RZC 28/08/22] Carreguem primer el arxiu en memòria i després el descomprimim. Es algo més rapid.
/*
FILE *f = fopen(filename, "rb");
fseek(f, 0, SEEK_END);
long fsize = ftell(f);
fseek(f, 0, SEEK_SET);
Uint8 *buffer = (Uint8*)malloc(fsize + 1);
fread(buffer, fsize, 1, f);
fclose(f);
*/
music->samples = stb_vorbis_decode_memory(buffer, fsize, &chan, &samplerate, &music->output);
free(buffer);
// [RZC 28/08/22] Abans el descomprimiem mentre el teniem obert
// music->samples = stb_vorbis_decode_filename(filename, &chan, &samplerate, &music->output);
SDL_AudioCVT cvt;
SDL_BuildAudioCVT(&cvt, AUDIO_S16, chan, samplerate, JA_format, JA_channels, JA_freq);
if (cvt.needed) {
cvt.len = music->samples * chan * 2;
cvt.buf = (Uint8 *) SDL_malloc(cvt.len * cvt.len_mult);
SDL_memcpy(cvt.buf, music->output, cvt.len);
SDL_ConvertAudio(&cvt);
free(music->output);
music->output = (short*)cvt.buf;
}
music->pos = 0;
music->state = JA_MUSIC_STOPPED;
return music;
}
void JA_PlayMusic(JA_Music music, const int loop) {
if (current_music != NULL) {
current_music->pos = 0;
current_music->state = JA_MUSIC_STOPPED;
}
current_music = music;
current_music->pos = 0;
current_music->state = JA_MUSIC_PLAYING;
current_music->times = loop;
}
void JA_PauseMusic() {
if (current_music == NULL || current_music->state == JA_MUSIC_INVALID) return;
current_music->state = JA_MUSIC_PAUSED;
}
void JA_ResumeMusic() {
if (current_music == NULL || current_music->state == JA_MUSIC_INVALID) return;
current_music->state = JA_MUSIC_PLAYING;
}
void JA_StopMusic() {
if (current_music == NULL || current_music->state == JA_MUSIC_INVALID) return;
current_music->pos = 0;
current_music->state = JA_MUSIC_STOPPED;
}
JA_Music_state JA_GetMusicState() {
if (current_music == NULL) return JA_MUSIC_INVALID;
return current_music->state;
}
void JA_DeleteMusic(JA_Music music) {
if (current_music == music) current_music = NULL;
free(music->output);
delete music;
}
JA_Sound JA_NewSound(Uint8* buffer, Uint32 length) {
JA_Sound sound = new JA_Sound_t();
sound->buffer = buffer;
sound->length = length;
return sound;
}
JA_Sound JA_LoadSound(const char* filename) {
JA_Sound sound = new JA_Sound_t();
SDL_AudioSpec wavSpec;
//SDL_LoadWAV(filename, &wavSpec, &sound->buffer, &sound->length);
int size;
char *buffer = file_getfilebuffer(filename, size);
SDL_LoadWAV_RW(SDL_RWFromMem(buffer, size),1, &wavSpec, &sound->buffer, &sound->length);
free(buffer);
SDL_AudioCVT cvt;
SDL_BuildAudioCVT(&cvt, wavSpec.format, wavSpec.channels, wavSpec.freq, JA_format, JA_channels, JA_freq);
cvt.len = sound->length;
cvt.buf = (Uint8 *) SDL_malloc(cvt.len * cvt.len_mult);
SDL_memcpy(cvt.buf, sound->buffer, sound->length);
SDL_ConvertAudio(&cvt);
SDL_FreeWAV(sound->buffer);
sound->buffer = cvt.buf;
sound->length = cvt.len_cvt;
return sound;
}
int JA_PlaySound(JA_Sound sound, const int loop) {
int channel = 0;
while (channel < JA_MAX_SIMULTANEOUS_CHANNELS && channels[channel].state != JA_CHANNEL_FREE) { channel++; }
if (channel == JA_MAX_SIMULTANEOUS_CHANNELS) channel = 0;
channels[channel].sound = sound;
channels[channel].times = loop;
channels[channel].pos = 0;
channels[channel].state = JA_CHANNEL_PLAYING;
return channel;
}
void JA_DeleteSound(JA_Sound sound) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
if (channels[i].sound == sound) JA_StopChannel(i);
}
SDL_free(sound->buffer);
delete sound;
}
void JA_PauseChannel(const int channel) {
if (channel == -1) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
if (channels[i].state == JA_CHANNEL_PLAYING) channels[i].state = JA_CHANNEL_PAUSED;
}
} else if (channel >= 0 && channel < JA_MAX_SIMULTANEOUS_CHANNELS) {
if (channels[channel].state == JA_CHANNEL_PLAYING) channels[channel].state = JA_CHANNEL_PAUSED;
}
}
void JA_ResumeChannel(const int channel) {
if (channel == -1) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
if (channels[i].state == JA_CHANNEL_PAUSED) channels[i].state = JA_CHANNEL_PLAYING;
}
} else if (channel >= 0 && channel < JA_MAX_SIMULTANEOUS_CHANNELS) {
if (channels[channel].state == JA_CHANNEL_PAUSED) channels[channel].state = JA_CHANNEL_PLAYING;
}
}
void JA_StopChannel(const int channel) {
if (channel == -1) {
for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
channels[i].state = JA_CHANNEL_FREE;
channels[i].pos = 0;
channels[i].sound = NULL;
}
} else if (channel >= 0 && channel < JA_MAX_SIMULTANEOUS_CHANNELS) {
channels[channel].state = JA_CHANNEL_FREE;
channels[channel].pos = 0;
channels[channel].sound = NULL;
}
}
JA_Channel_state JA_GetChannelState(const int channel) {
if (channel < 0 || channel >= JA_MAX_SIMULTANEOUS_CHANNELS) return JA_CHANNEL_INVALID;
return channels[channel].state;
}
int JA_SetVolume(int volume) {
JA_volume = volume > 128 ? 128 : volume < 0 ? 0 : volume;
return JA_volume;
}

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jail_audio.h Normal file
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#pragma once
#include <SDL2/SDL.h>
enum JA_Channel_state { JA_CHANNEL_INVALID, JA_CHANNEL_FREE, JA_CHANNEL_PLAYING, JA_CHANNEL_PAUSED };
enum JA_Music_state { JA_MUSIC_INVALID, JA_MUSIC_PLAYING, JA_MUSIC_PAUSED, JA_MUSIC_STOPPED };
typedef struct JA_Sound_t *JA_Sound;
typedef struct JA_Music_t *JA_Music;
void JA_Init(const int freq, const SDL_AudioFormat format, const int channels);
void JA_Quit();
JA_Music JA_LoadMusic(const char* filename);
void JA_PlayMusic(JA_Music music, const int loop = -1);
void JA_PauseMusic();
void JA_ResumeMusic();
void JA_StopMusic();
JA_Music_state JA_GetMusicState();
void JA_DeleteMusic(JA_Music music);
JA_Sound JA_NewSound(Uint8* buffer, Uint32 length);
JA_Sound JA_LoadSound(const char* filename);
int JA_PlaySound(JA_Sound sound, const int loop = 0);
void JA_PauseChannel(const int channel);
void JA_ResumeChannel(const int channel);
void JA_StopChannel(const int channel);
JA_Channel_state JA_GetChannelState(const int channel);
void JA_DeleteSound(JA_Sound sound);
int JA_SetVolume(int volume);