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Actualizado a la última versión de jail_audio

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This commit is contained in:
Sergio
2022-08-28 22:41:24 +02:00
parent fcdd2ffd21
commit 462a64bb3e
6 changed files with 106 additions and 60 deletions
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2
data/map/05.map
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@@ -2,7 +2,7 @@ tileset_img=surface.png
bgColor1=234,171,159
bgColor2=144,225,231
room_up=0.map
room_up=0
room_down=06.map
room_left=02.map
room_right=07.map

2
source/game.cpp
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@@ -27,7 +27,7 @@ Game::Game(SDL_Renderer *renderer, Screen *screen, Asset *asset, Input *input)
section.subsection = SUBSECTION_GAME_PLAY;
musicEnabled = true;
debug = false;
debug = true;
musicEnabled = !debug;
}

39
source/jail_audio.cpp
View File

@@ -1,6 +1,7 @@
#ifndef __MIPSEL__
#include "jail_audio.h"
#include "stb_vorbis.c"
#include <SDL2/SDL.h>
#include <stdio.h>
#define JA_MAX_SIMULTANEOUS_CHANNELS 5
@@ -30,16 +31,17 @@ 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;
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_memcpy(stream, current_music->output+current_music->pos, size);
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_memcpy(stream+size, current_music->output, len-size);
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 {
@@ -52,11 +54,11 @@ void audioCallback(void * userdata, uint8_t * stream, int len) {
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, 64);
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, 64);
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 {
@@ -79,7 +81,19 @@ void JA_Init(const int freq, const SDL_AudioFormat format, const int channels) {
JA_Music JA_LoadMusic(const char* filename) {
int chan, samplerate;
JA_Music music = new JA_Music_t();
music->samples = stb_vorbis_decode_filename(filename, &chan, &samplerate, &music->output);
// [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);
@@ -134,6 +148,13 @@ void JA_DeleteMusic(JA_Music music) {
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;
@@ -210,4 +231,8 @@ JA_Channel_state JA_GetChannelState(const int channel) {
if (channel < 0 || channel >= JA_MAX_SIMULTANEOUS_CHANNELS) return JA_CHANNEL_INVALID;
return channels[channel].state;
}
#endif
int JA_SetVolume(int volume) {
JA_volume = volume > 128 ? 128 : volume < 0 ? 0 : volume;
return JA_volume;
}

4
source/jail_audio.h
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@@ -1,5 +1,4 @@
#pragma once
#include <SDL2/SDL.h>
enum JA_Channel_state { JA_CHANNEL_INVALID, JA_CHANNEL_FREE, JA_CHANNEL_PLAYING, JA_CHANNEL_PAUSED };
@@ -18,6 +17,7 @@ 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);
@@ -25,3 +25,5 @@ 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);

2
source/prog.cpp
View File

@@ -26,7 +26,7 @@ Prog::Prog(std::string executablePath)
}
else
{
section.name = SECTION_PROG_LOGO;
section.name = SECTION_PROG_GAME;
}
input = new Input(asset->get("gamecontrollerdb.txt"));
screen = new Screen(window, renderer, options);

115
source/stb_vorbis.c
View File

@@ -1,4 +1,4 @@
// Ogg Vorbis audio decoder - v1.20 - public domain
// Ogg Vorbis audio decoder - v1.22 - public domain
// http://nothings.org/stb_vorbis/
//
// Original version written by Sean Barrett in 2007.
@@ -29,12 +29,15 @@
// Bernhard Wodo Evan Balster github:alxprd
// Tom Beaumont Ingo Leitgeb Nicolas Guillemot
// Phillip Bennefall Rohit Thiago Goulart
// github:manxorist saga musix github:infatum
// github:manxorist Saga Musix github:infatum
// Timur Gagiev Maxwell Koo Peter Waller
// github:audinowho Dougall Johnson David Reid
// github:Clownacy Pedro J. Estebanez Remi Verschelde
// AnthoFoxo github:morlat Gabriel Ravier
//
// Partial history:
// 1.22 - 2021-07-11 - various small fixes
// 1.21 - 2021-07-02 - fix bug for files with no comments
// 1.20 - 2020-07-11 - several small fixes
// 1.19 - 2020-02-05 - warnings
// 1.18 - 2020-02-02 - fix seek bugs; parse header comments; misc warnings etc.
@@ -220,6 +223,12 @@ extern int stb_vorbis_decode_frame_pushdata(
// channel. In other words, (*output)[0][0] contains the first sample from
// the first channel, and (*output)[1][0] contains the first sample from
// the second channel.
//
// *output points into stb_vorbis's internal output buffer storage; these
// buffers are owned by stb_vorbis and application code should not free
// them or modify their contents. They are transient and will be overwritten
// once you ask for more data to get decoded, so be sure to grab any data
// you need before then.
extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
// inform stb_vorbis that your next datablock will not be contiguous with
@@ -579,7 +588,7 @@ enum STBVorbisError
#if defined(_MSC_VER) || defined(__MINGW32__)
#include <malloc.h>
#endif
#if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__)
#if defined(__linux__) || defined(__linux) || defined(__sun__) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__)
#include <alloca.h>
#endif
#else // STB_VORBIS_NO_CRT
@@ -646,6 +655,12 @@ typedef signed int int32;
typedef float codetype;
#ifdef _MSC_VER
#define STBV_NOTUSED(v) (void)(v)
#else
#define STBV_NOTUSED(v) (void)sizeof(v)
#endif
// @NOTE
//
// Some arrays below are tagged "//varies", which means it's actually
@@ -1046,7 +1061,7 @@ static float float32_unpack(uint32 x)
uint32 sign = x & 0x80000000;
uint32 exp = (x & 0x7fe00000) >> 21;
double res = sign ? -(double)mantissa : (double)mantissa;
return (float) ldexp((float)res, exp-788);
return (float) ldexp((float)res, (int)exp-788);
}
@@ -1077,6 +1092,7 @@ static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
// find the first entry
for (k=0; k < n; ++k) if (len[k] < NO_CODE) break;
if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
assert(len[k] < 32); // no error return required, code reading lens checks this
// add to the list
add_entry(c, 0, k, m++, len[k], values);
// add all available leaves
@@ -1090,6 +1106,7 @@ static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
uint32 res;
int z = len[i], y;
if (z == NO_CODE) continue;
assert(z < 32); // no error return required, code reading lens checks this
// find lowest available leaf (should always be earliest,
// which is what the specification calls for)
// note that this property, and the fact we can never have
@@ -1099,12 +1116,10 @@ static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
while (z > 0 && !available[z]) --z;
if (z == 0) { return FALSE; }
res = available[z];
assert(z >= 0 && z < 32);
available[z] = 0;
add_entry(c, bit_reverse(res), i, m++, len[i], values);
// propagate availability up the tree
if (z != len[i]) {
assert(len[i] >= 0 && len[i] < 32);
for (y=len[i]; y > z; --y) {
assert(available[y] == 0);
available[y] = res + (1 << (32-y));
@@ -2577,34 +2592,33 @@ static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A,
while (z > base) {
float k00,k11;
float l00,l11;
k00 = z[-0] - z[-8];
k11 = z[-1] - z[-9];
z[-0] = z[-0] + z[-8];
z[-1] = z[-1] + z[-9];
z[-8] = k00;
z[-9] = k11 ;
k00 = z[-0] - z[ -8];
k11 = z[-1] - z[ -9];
l00 = z[-2] - z[-10];
l11 = z[-3] - z[-11];
z[ -0] = z[-0] + z[ -8];
z[ -1] = z[-1] + z[ -9];
z[ -2] = z[-2] + z[-10];
z[ -3] = z[-3] + z[-11];
z[ -8] = k00;
z[ -9] = k11;
z[-10] = (l00+l11) * A2;
z[-11] = (l11-l00) * A2;
k00 = z[ -2] - z[-10];
k11 = z[ -3] - z[-11];
z[ -2] = z[ -2] + z[-10];
z[ -3] = z[ -3] + z[-11];
z[-10] = (k00+k11) * A2;
z[-11] = (k11-k00) * A2;
k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation
k00 = z[ -4] - z[-12];
k11 = z[ -5] - z[-13];
l00 = z[ -6] - z[-14];
l11 = z[ -7] - z[-15];
z[ -4] = z[ -4] + z[-12];
z[ -5] = z[ -5] + z[-13];
z[-12] = k11;
z[-13] = k00;
k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation
k11 = z[ -7] - z[-15];
z[ -6] = z[ -6] + z[-14];
z[ -7] = z[ -7] + z[-15];
z[-14] = (k00+k11) * A2;
z[-15] = (k00-k11) * A2;
z[-12] = k11;
z[-13] = -k00;
z[-14] = (l11-l00) * A2;
z[-15] = (l00+l11) * -A2;
iter_54(z);
iter_54(z-8);
@@ -3069,6 +3083,7 @@ static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *f
for (q=1; q < g->values; ++q) {
j = g->sorted_order[q];
#ifndef STB_VORBIS_NO_DEFER_FLOOR
STBV_NOTUSED(step2_flag);
if (finalY[j] >= 0)
#else
if (step2_flag[j])
@@ -3171,6 +3186,7 @@ static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start,
// WINDOWING
STBV_NOTUSED(left_end);
n = f->blocksize[m->blockflag];
map = &f->mapping[m->mapping];
@@ -3368,7 +3384,7 @@ static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start,
// this isn't to spec, but spec would require us to read ahead
// and decode the size of all current frames--could be done,
// but presumably it's not a commonly used feature
f->current_loc = -n2; // start of first frame is positioned for discard
f->current_loc = 0u - n2; // start of first frame is positioned for discard (NB this is an intentional unsigned overflow/wrap-around)
// we might have to discard samples "from" the next frame too,
// if we're lapping a large block then a small at the start?
f->discard_samples_deferred = n - right_end;
@@ -3642,8 +3658,10 @@ static int start_decoder(vorb *f)
f->vendor[len] = (char)'\0';
//user comments
f->comment_list_length = get32_packet(f);
if (f->comment_list_length > 0) {
f->comment_list = (char**)setup_malloc(f, sizeof(char*) * (f->comment_list_length));
f->comment_list = NULL;
if (f->comment_list_length > 0)
{
f->comment_list = (char**) setup_malloc(f, sizeof(char*) * (f->comment_list_length));
if (f->comment_list == NULL) return error(f, VORBIS_outofmem);
}
@@ -3867,8 +3885,7 @@ static int start_decoder(vorb *f)
unsigned int div=1;
for (k=0; k < c->dimensions; ++k) {
int off = (z / div) % c->lookup_values;
float val = mults[off];
val = mults[off]*c->delta_value + c->minimum_value + last;
float val = mults[off]*c->delta_value + c->minimum_value + last;
c->multiplicands[j*c->dimensions + k] = val;
if (c->sequence_p)
last = val;
@@ -3951,7 +3968,7 @@ static int start_decoder(vorb *f)
if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
}
for (k=0; k < 1 << g->class_subclasses[j]; ++k) {
g->subclass_books[j][k] = get_bits(f,8)-1;
g->subclass_books[j][k] = (int16)get_bits(f,8)-1;
if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
}
}
@@ -4509,6 +4526,7 @@ stb_vorbis *stb_vorbis_open_pushdata(
*error = VORBIS_need_more_data;
else
*error = p.error;
vorbis_deinit(&p);
return NULL;
}
f = vorbis_alloc(&p);
@@ -4566,7 +4584,7 @@ static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
header[i] = get8(f);
if (f->eof) return 0;
if (header[4] != 0) goto invalid;
goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24);
goal = header[22] + (header[23] << 8) + (header[24]<<16) + ((uint32)header[25]<<24);
for (i=22; i < 26; ++i)
header[i] = 0;
crc = 0;
@@ -4970,7 +4988,7 @@ unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f)
// set. whoops!
break;
}
previous_safe = last_page_loc+1;
//previous_safe = last_page_loc+1; // NOTE: not used after this point, but note for debugging
last_page_loc = stb_vorbis_get_file_offset(f);
}
@@ -5081,7 +5099,10 @@ stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const st
stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc)
{
stb_vorbis *f, p;
if (data == NULL) return NULL;
if (!data) {
if (error) *error = VORBIS_unexpected_eof;
return NULL;
}
vorbis_init(&p, alloc);
p.stream = (uint8 *) data;
p.stream_end = (uint8 *) data + len;
@@ -5156,11 +5177,11 @@ static void copy_samples(short *dest, float *src, int len)
static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
{
#define BUFFER_SIZE 32
float buffer[BUFFER_SIZE];
int i,j,o,n = BUFFER_SIZE;
#define STB_BUFFER_SIZE 32
float buffer[STB_BUFFER_SIZE];
int i,j,o,n = STB_BUFFER_SIZE;
check_endianness();
for (o = 0; o < len; o += BUFFER_SIZE) {
for (o = 0; o < len; o += STB_BUFFER_SIZE) {
memset(buffer, 0, sizeof(buffer));
if (o + n > len) n = len - o;
for (j=0; j < num_c; ++j) {
@@ -5177,16 +5198,17 @@ static void compute_samples(int mask, short *output, int num_c, float **data, in
output[o+i] = v;
}
}
#undef STB_BUFFER_SIZE
}
static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
{
#define BUFFER_SIZE 32
float buffer[BUFFER_SIZE];
int i,j,o,n = BUFFER_SIZE >> 1;
#define STB_BUFFER_SIZE 32
float buffer[STB_BUFFER_SIZE];
int i,j,o,n = STB_BUFFER_SIZE >> 1;
// o is the offset in the source data
check_endianness();
for (o = 0; o < len; o += BUFFER_SIZE >> 1) {
for (o = 0; o < len; o += STB_BUFFER_SIZE >> 1) {
// o2 is the offset in the output data
int o2 = o << 1;
memset(buffer, 0, sizeof(buffer));
@@ -5216,6 +5238,7 @@ static void compute_stereo_samples(short *output, int num_c, float **data, int d
output[o2+i] = v;
}
}
#undef STB_BUFFER_SIZE
}
static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
@@ -5288,8 +5311,6 @@ int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short
float **outputs;
int len = num_shorts / channels;
int n=0;
int z = f->channels;
if (z > channels) z = channels;
while (n < len) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;
@@ -5308,8 +5329,6 @@ int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, in
{
float **outputs;
int n=0;
int z = f->channels;
if (z > channels) z = channels;
while (n < len) {
int k = f->channel_buffer_end - f->channel_buffer_start;
if (n+k >= len) k = len - n;