From 462a64bb3e40cc930df7fce6497f92ab3ecde37e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sergio=20Valor=20Mart=C3=ADnez?= Date: Sun, 28 Aug 2022 22:41:24 +0200 Subject: [PATCH] =?UTF-8?q?Actualizado=20a=20la=20=C3=BAltima=20versi?= =?UTF-8?q?=C3=B3n=20de=20jail=5Faudio?= MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit --- data/map/05.map | 2 +- source/game.cpp | 2 +- source/jail_audio.cpp | 39 +++++++++++--- source/jail_audio.h | 4 +- source/prog.cpp | 2 +- source/stb_vorbis.c | 117 ++++++++++++++++++++++++------------------ 6 files changed, 106 insertions(+), 60 deletions(-) diff --git a/data/map/05.map b/data/map/05.map index c75cc93..63098ff 100644 --- a/data/map/05.map +++ b/data/map/05.map @@ -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 diff --git a/source/game.cpp b/source/game.cpp index 5114b88..94b1d29 100644 --- a/source/game.cpp +++ b/source/game.cpp @@ -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; } diff --git a/source/jail_audio.cpp b/source/jail_audio.cpp index e22b355..fc23241 100644 --- a/source/jail_audio.cpp +++ b/source/jail_audio.cpp @@ -1,6 +1,7 @@ -#ifndef __MIPSEL__ #include "jail_audio.h" #include "stb_vorbis.c" +#include +#include #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; +} \ No newline at end of file diff --git a/source/jail_audio.h b/source/jail_audio.h index 17164b7..2a2fc9f 100644 --- a/source/jail_audio.h +++ b/source/jail_audio.h @@ -1,5 +1,4 @@ #pragma once - #include 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); \ No newline at end of file diff --git a/source/prog.cpp b/source/prog.cpp index 61bbf3d..0b0d60e 100644 --- a/source/prog.cpp +++ b/source/prog.cpp @@ -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); diff --git a/source/stb_vorbis.c b/source/stb_vorbis.c index 49df433..7e5daa3 100644 --- a/source/stb_vorbis.c +++ b/source/stb_vorbis.c @@ -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 #endif - #if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__) + #if defined(__linux__) || defined(__linux) || defined(__sun__) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__) #include #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; @@ -3642,9 +3658,11 @@ 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)); - if (f->comment_list == NULL) return error(f, VORBIS_outofmem); + 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); } for(i=0; i < f->comment_list_length; ++i) { @@ -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;