From 644161ab16e119f1f0a946102483d4aad525cf5d Mon Sep 17 00:00:00 2001 From: Cameron Reikes Date: Sun, 19 Mar 2023 01:02:43 -0700 Subject: [PATCH] Sound system with metadesk, mono only for now --- assets.mdesk | 4 + assets/simple_text_chirp.wav | Bin 0 -> 8864 bytes buff.h | 1 - codegen.c | 13 + main.c | 529 ++- thirdparty/dr_wav.h | 8350 ++++++++++++++++++++++++++++++++++ thirdparty/sokol_audio.h | 649 ++- thirdparty/sokol_log.h | 343 ++ training.mdesk | 23 +- 9 files changed, 9362 insertions(+), 550 deletions(-) create mode 100644 assets/simple_text_chirp.wav create mode 100644 thirdparty/dr_wav.h create mode 100644 thirdparty/sokol_log.h diff --git a/assets.mdesk b/assets.mdesk index 2f248ca..7d76012 100644 --- a/assets.mdesk +++ b/assets.mdesk @@ -1,3 +1,7 @@ +@sound simple_talk: +{ + filepath: "simple_text_chirp.wav", +} @image merchant: { filepath: "copyrighted/merchant.png", diff --git a/assets/simple_text_chirp.wav b/assets/simple_text_chirp.wav new file mode 100644 index 0000000000000000000000000000000000000000..e176ecc790e1c747f7bcdcc6bb885bcedd2d2ffd GIT binary patch literal 8864 zcmZ8n2Ut@{*PdI_fRIq6w+L8JdQn6a6uor^{4BD69*j_u&n>B@VjA7eWH863~>&A9M~to!avP7&1a7H zo}Np4JokFhJ+k|vZYJGEbv5n!YnOptay?T#`+53%e)Ra}vCiW!k1&ss9(s4N#|`&v z_Za}Lv$pf5&Wk%YE8Z!>6;_I+Zeeb_U5B{ta*1$R?(E<^)TzqR-0_%0j{Rr58Fnjd zIU7@(zdD_=wzuwX^^awmg@;8~`El7vbGf-#y25OZ=}VJa5;IA&sJF;W7$uO>{!|5_ zK}mQUFX8HqBMo-U6rD+XpeDO5uO+zIr*T>RXyxkK!PTQGxiWRhuA=z`nlJZ1g?!-N z&3NPXdfkiQ=Leq*d35pq^qkjsw%p=w=3F0`F1wm~X%#x^cl7+Y`fuyMv;DaD$5T2+{fk8##GA@CZ~AloHs>7{yT0uH z@YjXCxBlLL;P#=DNAiz7KOy)>oIEOZ#@U) zXSQ>Mhl}U_bT=(?vd1UmA9J@>yzX=%dgDuiobop*MP75ukU@Xw^^uvXi{iKsI*VdK3n^w^x^v0hAj!(8CD&}hfVDJ zYu{IWYx{qgd_~;3jCb&#oJ+XRX@3>=e6>(wlOXHu!H^v7f3{BXOuqz=W;c~+LgewV`puHj? zD#1OWH2!M*gm~BZ>v6xusV82TsF?V2LZ1m=#`hclWt{K0M`JC<9v@RXdiLnFQJqIk zjeQk^VrGooF`{NTKWx;nNkiuhIUarFr#n$+2WLel4f<}Wxw3;NqxJ9oe!PV z`&G!I;Maju11|b``!4b>>tX4&y4&e4$kWYzg(Atd##!Sy)M27snoX9qnU%f8ud?;h zFQyHWNO6KNlm0;Z5V80Q_s$q(n5fTbuhB%Sm$eo(i<_1-oKQ)Wp|z>ipDH6OR+Onq zoQn?>zWp}%>-NvYr{N!7y?1<<@`ic6^Htr8&AE-w_B|Cn$$HfHq4xgHdn0nRcc0u@ zb35*qB1@K8d!zFDtBkMdZ?837ZNK7vWx(Y%m(nhNO$$z2aiRKr(D|%$oz9&*>vT5j zOvIU*)Xk~EsWmC*Qf8!tr4T7~$q$onB%e$^k$f=uO!9@~*U8_K%~HChOi$ULlAmIq zIz9Dfs_e|dGhfa`oV|6{@7#@Zz0bcmpKw8U;b2<$#oCKUE{(V>x{`Zk+tu;c?9(;r zw=+&&pL-+brXtfct32!Vt<>9p-9;!i(k!t9iA8R#_X-+UHQAR_jmK}emL^+=%-bm|M)WL>(Ac?6bvowR^(S4 zRAN5ylE*DjUi%+%A3?F2cu9ccPA1PWn^#sns+kcp``r z3Pn3bk>WSv4H6TRdnSWS8Pio}zS1kwndW8Y`(!ftJ$WCCT8kN$CRW?6x?5kgj_&lN z(`=hUn?G#ZZ1>s;?NjVKIb=AvJKlHnb;@-Lbs-@0 z!M)CXjYqA=O3y0K#4hDs=5_tnbxOB4-D0}m>mKfP&Z}FGojuHZ&hOdSGur#9w~Nm~ zpJtzFzFEFq{FeJw`uX}F^e^}CAFv~!B)}_hSzu0}AZT#V_Mk^WC^$HHLGZEQ*TL#w zMMy-*?2t_%Cqr(BybP%bsSn{p$X;f>EP6?yhHwTy`UXILxRkLo&;_V^bKqb*c%WTQ0<@O-_`$>-y}cAcZ09J??E4L zpA+7`-p6~o^-Su~ezwiULHZoS;%*|Mb>H_d7)YaG(}vLUb`yWX=tL#0q%Q+g;f z>wN2;)efpHu9;rLRd260t-e(?sH&xMV`WI?tBTDPii*PWjpgCx4P_U~rj{wn3QJQ; zrd%GEH7>^dR%n2XiZUkQSTz_B6Z=%!s~^H3YQj6 zE(|T~R!A4N7JM$aUyxRCxL{quvVutk;|c~8gctNE@G5XFa4E0@*cUiLt2nrP zmWM1k%T-ooRuiprtOGjz-AQaS%jT0!fbDTxgY9IyJ9c*VOYGm+D;(B3ym093xB!$a zb{gY!)TzSB!+DPLIp<1ecb7Pq{Vp$E7?&W|iLQUUX1Lb6(r$rnv2M%V_PJ%ay>hE_ zW8JJ2-4r2;;fgVenTq*}M8!(Q8pV3WD#Zq9FICJ_Oj1l#3{>=1cql9soLh}sp4)Y| zgKn$bCO|G|;ih)YbxmzRM1mDK4E|+MKhTH#iS+Hg$gKl;kwRN#gXv zaie3HBja$^VXlMBA#a^& z^|C6rOte&3Ub9HBXqF$4_mJnwX34~|6XpZWKS@_f&86vP(PlNKn@k-|@0yG=X_4%f zbd@|6PZsl{eWG5XT;XCNFE}UYB`Bd6(>C;ZYBW_w?k1hc=fos}B95ZLr~q%pPWTo7 zGcV^aaq*m%J^7IS^44RmQ(IkH%UhCL=CyQh(Klx{Z)qOZY|~uc zbhYXCrjbonO-+qA8}~LQG=?+^8cQ3lH|%Pd(a^8KszF=-y#7-C#`;)fUxp)h$)N zs!3&5-@QJ%eqQ}HSjqePrh3^=g~bmfBX?=Bl2dKCZ4%J86E_oYK^2 z+_ej|=e1g`cl)~b%yywJQnyFb{g&LLCDMzROBg1Sm^R1m$M zzD<*YaKR=)wty1$6RsEL2#vx%q7|a+qBfD2c%JyQxI}Ct870{!c_`6Kd`xDU95u-^ zp-lr#=b0WfePyaM^)ef4w%P2mS&i-A z+(PCli+cMXJmI|`LZe*Ba_OVkmtz@ z~XtpV1mF+&`RH;chRx52d$*; zQX8molr2?HrjskkXi`R&5EqDLLn;X?i_|2rSa%lK+8 zjoZu(;aoWtdyn18PG-GW#J)177#D+Dt&B=TmSK+}!O+`48p@e8W+yX_31TRwRDWH+ zTR%elGub-_BBuC@I|`=$2n?X%j4wY!6&tF`a67qkboOSEy?zS_=O zfwo2SR`UQ9zDu)Ilc4!Y6R5G*m}=V8haLv%hw2^9o+Ei@~<2}?})IsV#>cQ$! 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I don't care about fopen_s @@ -280,231 +284,61 @@ void make_space_and_append(Dialog *d, DialogElement elem) BUFF_APPEND(d, elem); } -void say_characters(Entity *npc, int num_characters) +typedef struct AudioSample { - Sentence *sentence_to_append_to = &npc->player_dialog.data[npc->player_dialog.cur_index-1].s; - for(int i = 0; i < num_characters; i++) - { - if(!BUFF_EMPTY(&npc->player_dialog) && !BUFF_EMPTY(&npc->sentence_to_say)) - { - char new_character = npc->sentence_to_say.data[0]; - bool found_matching_star = false; - BUFF(char, MAX_SENTENCE_LENGTH) match_buffer = {0}; - if(new_character == '*') - { - for(int ii = sentence_to_append_to->cur_index-1; ii >= 0; ii--) - { - if(sentence_to_append_to->data[ii] == '*') - { - found_matching_star = true; - break; - } - BUFF_PUSH_FRONT(&match_buffer, sentence_to_append_to->data[ii]); - } - } - if(found_matching_star) - { -#if 0 // actions - if(strcmp(match_buffer.data, "fights player") == 0 && npc->npc_kind == OLD_MAN) - { - npc->aggressive = true; - } - if(strcmp(match_buffer.data, "sells grounding boots") == 0 && npc->npc_kind == MERCHANT) - { - player->boots_modifier -= 1; + float *pcm_data; // allocated by loader, must be freed + uint64_t pcm_data_length; +} AudioSample; - } - if(strcmp(match_buffer.data, "sells swiftness boots") == 0 && npc->npc_kind == MERCHANT) - { - player->boots_modifier += 1; - } - if(strcmp(match_buffer.data, "moves") == 0 && npc->npc_kind == DEATH) - { - npc->going_to_target = true; - npc->target_goto = AddV2(npc->pos, V2(0.0, -TILE_SIZE*1.5f)); - } -#endif - } - BUFF_APPEND(sentence_to_append_to, new_character); - BUFF_REMOVE_FRONT(&npc->sentence_to_say); - } - } -} +typedef struct AudioPlayer +{ + AudioSample *sample; // if not 0, exists + double volume; // ZII, 1.0 + this again + double pitch; // zero initialized, the pitch used to play is 1.0 + this + double cursor_time; // in seconds, current audio sample is cursor_time * sample_rate +} AudioPlayer; -bool npc_is_knight_sprite(Entity *it) +AudioPlayer playing_audio[128] = {0}; + +#define SAMPLE_RATE 44100 + +AudioSample load_wav_audio(const char *path) { - return it->is_npc && ( it->npc_kind == NPC_Max || it->npc_kind == NPC_Hunter || it->npc_kind == NPC_John); + unsigned int channels; + unsigned int sampleRate; + AudioSample to_return = {0}; + to_return.pcm_data = drwav_open_file_and_read_pcm_frames_f32(path, &channels, &sampleRate, &to_return.pcm_data_length, 0); + assert(channels == 1); + assert(sampleRate == SAMPLE_RATE); + return to_return; } -void add_new_npc_sentence(Entity *npc, char *sentence) +uint64_t cursor_pcm(AudioPlayer *p) { - size_t sentence_len = strlen(sentence); - assert(sentence_len < MAX_SENTENCE_LENGTH); - Sentence new_sentence = {0}; - bool inside_star = false; - for(int i = 0; i < sentence_len; i++) - { - if(sentence[i] == '"') break; - if(sentence[i] == '\n') continue; - BUFF_APPEND(&new_sentence, sentence[i]); - } - DialogElement empty_elem = { .author = NPC }; - say_characters(npc, npc->sentence_to_say.cur_index); - make_space_and_append(&npc->player_dialog, empty_elem); - npc->sentence_to_say = new_sentence; + return (uint64_t)(p->cursor_time * SAMPLE_RATE); } - -void begin_text_input(); // called when player engages in dialog, must say something and fill text_input_buffer -// a callback, when 'text backend' has finished making text. End dialog -void end_text_input(char *what_player_said) +float float_rand( float min, float max ) { - player->state = CHARACTER_IDLE; -#ifdef WEB // hacky - _sapp_emsc_register_eventhandlers(); -#endif - - size_t player_said_len = strlen(what_player_said); - int actual_len = 0; - for(int i = 0; i < player_said_len; i++) if(what_player_said[i] != '\n') actual_len++; - if(actual_len == 0) - { - // this just means cancel the dialog - } - else + float scale = rand() / (float) RAND_MAX; /* [0, 1.0] */ + return min + scale * ( max - min ); /* [min, max] */ +} +void play_audio(AudioSample *sample) +{ + AudioPlayer *to_use = 0; + for(int i = 0; i < ARRLEN(playing_audio); i++) { - Sentence what_player_said_sentence = {0}; - assert(player_said_len < ARRLEN(what_player_said_sentence.data)); - for(int i = 0; i < player_said_len; i++) + if(playing_audio[i].sample == 0) { - char c = what_player_said[i]; - if(c == '\n') break; - BUFF_APPEND(&what_player_said_sentence, c); + to_use = &playing_audio[i]; + break; } - - // order is player message, item status message in training data. So has to be same here - Dialog *to_append = &player->talking_to->player_dialog; - Entity *talking = player->talking_to; - make_space_and_append(to_append, (DialogElement){.s = what_player_said_sentence, .author = PLAYER}); - if(talking->last_seen_holding != player->holding_item) - { - if(talking->last_seen_holding) - { - Sentence discard = from_str(item_discard_message_table[talking->last_seen_holding->item_kind]); - BUFF_APPEND(&discard, '\n'); - make_space_and_append(to_append, (DialogElement){.author = SYSTEM, .s = discard}); - assert(talking->last_seen_holding->is_item); - talking->last_seen_holding = 0; - } - if(player->holding_item) - { - assert(player->holding_item->is_item); - Sentence possess = from_str(item_possess_message_table[player->holding_item->item_kind]); - BUFF_APPEND(&possess, '\n'); - make_space_and_append(to_append, (DialogElement){.author = SYSTEM, .s = possess}); - } - talking->last_seen_holding = player->holding_item; - } - - // the npc response will be appended here, or at least be async queued to be appended here - BUFF(char, 4000) prompt_buff = {0}; - BUFF(char *, 100) to_join = {0}; - - - assert(talking->npc_kind >= 0); - assert(talking->npc_kind < ARRLEN(prompt_table)); - assert(talking->npc_kind < ARRLEN(general_prompt_table)); - assert(talking->npc_kind < ARRLEN(name_table)); - - // general prompt - BUFF_APPEND(&to_join, general_prompt_table[talking->npc_kind]); - BUFF_APPEND(&to_join, "\n"); - - // item prompt - if(player->holding_item) - { - BUFF_APPEND(&to_join, item_prompt_table[player->holding_item->item_kind]); - BUFF_APPEND(&to_join, "\n"); - } - - // characters prompt - BUFF_APPEND(&to_join, prompt_table[talking->npc_kind]); - BUFF_APPEND(&to_join, "\n"); - char *character_prompt = name_table[talking->npc_kind]; - - // all the dialog - int i = 0; - BUFF_ITER(DialogElement, &player->talking_to->player_dialog) - { - //bool is_player = - if(it->author == PLAYER) - { - BUFF_APPEND(&to_join, "Player: \""); - } - else if(it->author == NPC) - { - BUFF_APPEND(&to_join, character_prompt); - BUFF_APPEND(&to_join, ": \""); - } - else if(it->author == SYSTEM) - { - } - else - { - assert(false); - } - BUFF_APPEND(&to_join, it->s.data); - if(it->author == PLAYER || it->author == NPC) - BUFF_APPEND(&to_join, "\"\n"); - i++; - } - - BUFF_APPEND(&to_join, character_prompt); - BUFF_APPEND(&to_join, ": \""); - - // concatenate into prompt_buff - BUFF_ITER(char *, &to_join) - { - size_t cur_len = strlen(*it); - for(int i = 0; i < cur_len; i++) - { - BUFF_APPEND(&prompt_buff, (*it)[i]); - } - } - - const char * prompt = prompt_buff.data; -#ifdef DEVTOOLS - Log("Prompt: `%s`\n", prompt); -#endif -#ifdef WEB - // fire off generation request, save id - int req_id = EM_ASM_INT({ - return make_generation_request(UTF8ToString($1), UTF8ToString($0)); - }, SERVER_URL, prompt); - player->talking_to->gen_request_id = req_id; -#endif -#ifdef DESKTOP - if(player->talking_to->npc_kind == NPC_Death) - { - add_new_npc_sentence(player->talking_to, "test *moves* I am death, destroyer of games. Come join me in the afterlife, or continue onwards *moves*"); - //add_new_npc_sentence(player->talking_to, "test"); - } - if(player->talking_to->npc_kind == NPC_Hunter) - { - add_new_npc_sentence(player->talking_to, "I am hunter"); - } - if(player->talking_to->npc_kind == NPC_Max) - { - add_new_npc_sentence(player->talking_to, "I am max"); - } - if(player->talking_to->npc_kind == NPC_John) - { - add_new_npc_sentence(player->talking_to, "I am john *gives WhiteSquare*"); - } - -#endif } + assert(to_use); + *to_use = (AudioPlayer){0}; + to_use->sample = sample; + to_use->volume = -0.5f; + to_use->pitch = float_rand(0.9f, 1.1f); } - // keydown needs to be referenced when begin text input, // on web it disables event handling so the button up event isn't received bool keydown[SAPP_KEYCODE_MENU] = {0}; @@ -756,6 +590,7 @@ TileInstance get_tile(Level *l, TileCoord t) return get_tile_layer(l, 0, t); } + sg_image load_image(const char *path) { sg_image to_return = {0}; @@ -791,6 +626,234 @@ sg_image load_image(const char *path) #include "quad-sapp.glsl.h" #include "assets.gen.c" +void say_characters(Entity *npc, int num_characters) +{ + play_audio(&sound_simple_talk); + Sentence *sentence_to_append_to = &npc->player_dialog.data[npc->player_dialog.cur_index-1].s; + for(int i = 0; i < num_characters; i++) + { + if(!BUFF_EMPTY(&npc->player_dialog) && !BUFF_EMPTY(&npc->sentence_to_say)) + { + char new_character = npc->sentence_to_say.data[0]; + bool found_matching_star = false; + BUFF(char, MAX_SENTENCE_LENGTH) match_buffer = {0}; + if(new_character == '*') + { + for(int ii = sentence_to_append_to->cur_index-1; ii >= 0; ii--) + { + if(sentence_to_append_to->data[ii] == '*') + { + found_matching_star = true; + break; + } + BUFF_PUSH_FRONT(&match_buffer, sentence_to_append_to->data[ii]); + } + } + if(found_matching_star) + { +#if 0 // actions + if(strcmp(match_buffer.data, "fights player") == 0 && npc->npc_kind == OLD_MAN) + { + npc->aggressive = true; + } + if(strcmp(match_buffer.data, "sells grounding boots") == 0 && npc->npc_kind == MERCHANT) + { + player->boots_modifier -= 1; + + } + if(strcmp(match_buffer.data, "sells swiftness boots") == 0 && npc->npc_kind == MERCHANT) + { + player->boots_modifier += 1; + } + if(strcmp(match_buffer.data, "moves") == 0 && npc->npc_kind == DEATH) + { + npc->going_to_target = true; + npc->target_goto = AddV2(npc->pos, V2(0.0, -TILE_SIZE*1.5f)); + } +#endif + } + BUFF_APPEND(sentence_to_append_to, new_character); + BUFF_REMOVE_FRONT(&npc->sentence_to_say); + } + } +} + +bool npc_is_knight_sprite(Entity *it) +{ + return it->is_npc && ( it->npc_kind == NPC_Max || it->npc_kind == NPC_Hunter || it->npc_kind == NPC_John); +} + +void add_new_npc_sentence(Entity *npc, char *sentence) +{ + size_t sentence_len = strlen(sentence); + assert(sentence_len < MAX_SENTENCE_LENGTH); + Sentence new_sentence = {0}; + bool inside_star = false; + for(int i = 0; i < sentence_len; i++) + { + if(sentence[i] == '"') break; + if(sentence[i] == '\n') continue; + BUFF_APPEND(&new_sentence, sentence[i]); + } + DialogElement empty_elem = { .author = NPC }; + say_characters(npc, npc->sentence_to_say.cur_index); + make_space_and_append(&npc->player_dialog, empty_elem); + npc->sentence_to_say = new_sentence; +} + +void begin_text_input(); // called when player engages in dialog, must say something and fill text_input_buffer +// a callback, when 'text backend' has finished making text. End dialog +void end_text_input(char *what_player_said) +{ + player->state = CHARACTER_IDLE; +#ifdef WEB // hacky + _sapp_emsc_register_eventhandlers(); +#endif + + size_t player_said_len = strlen(what_player_said); + int actual_len = 0; + for(int i = 0; i < player_said_len; i++) if(what_player_said[i] != '\n') actual_len++; + if(actual_len == 0) + { + // this just means cancel the dialog + } + else + { + Sentence what_player_said_sentence = {0}; + assert(player_said_len < ARRLEN(what_player_said_sentence.data)); + for(int i = 0; i < player_said_len; i++) + { + char c = what_player_said[i]; + if(c == '\n') break; + BUFF_APPEND(&what_player_said_sentence, c); + } + + // order is player message, item status message in training data. So has to be same here + Dialog *to_append = &player->talking_to->player_dialog; + Entity *talking = player->talking_to; + make_space_and_append(to_append, (DialogElement){.s = what_player_said_sentence, .author = PLAYER}); + if(talking->last_seen_holding != player->holding_item) + { + if(talking->last_seen_holding) + { + Sentence discard = from_str(item_discard_message_table[talking->last_seen_holding->item_kind]); + BUFF_APPEND(&discard, '\n'); + make_space_and_append(to_append, (DialogElement){.author = SYSTEM, .s = discard}); + assert(talking->last_seen_holding->is_item); + talking->last_seen_holding = 0; + } + if(player->holding_item) + { + assert(player->holding_item->is_item); + Sentence possess = from_str(item_possess_message_table[player->holding_item->item_kind]); + BUFF_APPEND(&possess, '\n'); + make_space_and_append(to_append, (DialogElement){.author = SYSTEM, .s = possess}); + } + talking->last_seen_holding = player->holding_item; + } + + // the npc response will be appended here, or at least be async queued to be appended here + BUFF(char, 4000) prompt_buff = {0}; + BUFF(char *, 100) to_join = {0}; + + + assert(talking->npc_kind >= 0); + assert(talking->npc_kind < ARRLEN(prompt_table)); + assert(talking->npc_kind < ARRLEN(general_prompt_table)); + assert(talking->npc_kind < ARRLEN(name_table)); + + // general prompt + BUFF_APPEND(&to_join, general_prompt_table[talking->npc_kind]); + BUFF_APPEND(&to_join, "\n"); + + // item prompt + if(player->holding_item) + { + BUFF_APPEND(&to_join, item_prompt_table[player->holding_item->item_kind]); + BUFF_APPEND(&to_join, "\n"); + } + + // characters prompt + BUFF_APPEND(&to_join, prompt_table[talking->npc_kind]); + BUFF_APPEND(&to_join, "\n"); + char *character_prompt = name_table[talking->npc_kind]; + + // all the dialog + int i = 0; + BUFF_ITER(DialogElement, &player->talking_to->player_dialog) + { + //bool is_player = + if(it->author == PLAYER) + { + BUFF_APPEND(&to_join, "Player: \""); + } + else if(it->author == NPC) + { + BUFF_APPEND(&to_join, character_prompt); + BUFF_APPEND(&to_join, ": \""); + } + else if(it->author == SYSTEM) + { + } + else + { + assert(false); + } + BUFF_APPEND(&to_join, it->s.data); + if(it->author == PLAYER || it->author == NPC) + BUFF_APPEND(&to_join, "\"\n"); + i++; + } + + BUFF_APPEND(&to_join, character_prompt); + BUFF_APPEND(&to_join, ": \""); + + // concatenate into prompt_buff + BUFF_ITER(char *, &to_join) + { + size_t cur_len = strlen(*it); + for(int i = 0; i < cur_len; i++) + { + BUFF_APPEND(&prompt_buff, (*it)[i]); + } + } + + const char * prompt = prompt_buff.data; +#ifdef DEVTOOLS + Log("Prompt: `%s`\n", prompt); +#endif +#ifdef WEB + // fire off generation request, save id + int req_id = EM_ASM_INT({ + return make_generation_request(UTF8ToString($1), UTF8ToString($0)); + }, SERVER_URL, prompt); + player->talking_to->gen_request_id = req_id; +#endif +#ifdef DESKTOP + if(player->talking_to->npc_kind == NPC_Death) + { + add_new_npc_sentence(player->talking_to, "test *moves* I am death, destroyer of games. Come join me in the afterlife, or continue onwards *moves*"); + //add_new_npc_sentence(player->talking_to, "test"); + } + if(player->talking_to->npc_kind == NPC_Hunter) + { + add_new_npc_sentence(player->talking_to, "I am hunter"); + } + if(player->talking_to->npc_kind == NPC_Max) + { + add_new_npc_sentence(player->talking_to, "I am max"); + } + if(player->talking_to->npc_kind == NPC_John) + { + add_new_npc_sentence(player->talking_to, "I am john *gives WhiteSquare*"); + } + +#endif + } +} + + + AnimatedSprite knight_idle = { .img = &image_knight_idle, @@ -974,6 +1037,34 @@ void reset_level() item->pos = AddV2(player->pos, V2(0.0, 30.0)); } +void audio_stream_callback(float *buffer, int num_frames, int num_channels) +{ + assert(num_channels == 1); + const int num_samples = num_frames * num_channels; + double time_to_play = (double)num_frames / (double)SAMPLE_RATE; + double time_per_sample = 1.0 / (double)SAMPLE_RATE; + for(int i = 0; i < num_samples; i++) + { + float output_frame = 0.0f; + for(int audio_i = 0; audio_i < ARRLEN(playing_audio); audio_i++) + { + AudioPlayer *it = &playing_audio[audio_i]; + if(it->sample != 0) + { + if(cursor_pcm(it) >= it->sample->pcm_data_length) + { + it->sample = 0; + } + else + { + output_frame += it->sample->pcm_data[cursor_pcm(it)]*(float)(it->volume + 1.0); + it->cursor_time += time_per_sample*(it->pitch + 1.0); + } + } + } + buffer[i] = output_frame; + } +} void init(void) { @@ -981,8 +1072,12 @@ void init(void) Log("Size of %d entities: %zu kb\n", (int)ARRLEN(entities), sizeof(entities)/1024); sg_setup(&(sg_desc){ .context = sapp_sgcontext(), - }); + }); stm_setup(); + saudio_setup(&(saudio_desc){ + .stream_cb = audio_stream_callback, + .logger.func = slog_func, + }); scratch = make_arena(1024 * 10); @@ -1713,6 +1808,16 @@ AABB draw_text(TextParams t) { col = t.colors[i]; } + if(false) // drop shadow, don't really like it + if(t.world_space) + { + Quad shadow_quad = to_draw; + for(int i = 0; i < 4; i++) + { + shadow_quad.points[i] = AddV2(shadow_quad.points[i], V2(0.0, -1.0)); + } + draw_quad((DrawParams){t.world_space, shadow_quad, image_font, font_atlas_region, (Color){0.0f,0.0f,0.0f,0.4f}, t.clip_to, .y_coord_sorting = 1.0f, .queue_for_translucent = true}); + } draw_quad((DrawParams){t.world_space, to_draw, image_font, font_atlas_region, col, t.clip_to, .y_coord_sorting = 1.0f, .queue_for_translucent = true}); } } diff --git a/thirdparty/dr_wav.h b/thirdparty/dr_wav.h new file mode 100644 index 0000000..2f885a0 --- /dev/null +++ b/thirdparty/dr_wav.h @@ -0,0 +1,8350 @@ +/* +WAV audio loader and writer. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_wav - v0.13.7 - 2022-09-17 + +David Reid - mackron@gmail.com + +GitHub: https://github.com/mackron/dr_libs +*/ + +/* +Introduction +============ +This is a single file library. To use it, do something like the following in one .c file. + + ```c + #define DR_WAV_IMPLEMENTATION + #include "dr_wav.h" + ``` + +You can then #include this file in other parts of the program as you would with any other header file. Do something like the following to read audio data: + + ```c + drwav wav; + if (!drwav_init_file(&wav, "my_song.wav", NULL)) { + // Error opening WAV file. + } + + drwav_int32* pDecodedInterleavedPCMFrames = malloc(wav.totalPCMFrameCount * wav.channels * sizeof(drwav_int32)); + size_t numberOfSamplesActuallyDecoded = drwav_read_pcm_frames_s32(&wav, wav.totalPCMFrameCount, pDecodedInterleavedPCMFrames); + + ... + + drwav_uninit(&wav); + ``` + +If you just want to quickly open and read the audio data in a single operation you can do something like this: + + ```c + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalPCMFrameCount; + float* pSampleData = drwav_open_file_and_read_pcm_frames_f32("my_song.wav", &channels, &sampleRate, &totalPCMFrameCount, NULL); + if (pSampleData == NULL) { + // Error opening and reading WAV file. + } + + ... + + drwav_free(pSampleData, NULL); + ``` + +The examples above use versions of the API that convert the audio data to a consistent format (32-bit signed PCM, in this case), but you can still output the +audio data in its internal format (see notes below for supported formats): + + ```c + size_t framesRead = drwav_read_pcm_frames(&wav, wav.totalPCMFrameCount, pDecodedInterleavedPCMFrames); + ``` + +You can also read the raw bytes of audio data, which could be useful if dr_wav does not have native support for a particular data format: + + ```c + size_t bytesRead = drwav_read_raw(&wav, bytesToRead, pRawDataBuffer); + ``` + +dr_wav can also be used to output WAV files. This does not currently support compressed formats. To use this, look at `drwav_init_write()`, +`drwav_init_file_write()`, etc. Use `drwav_write_pcm_frames()` to write samples, or `drwav_write_raw()` to write raw data in the "data" chunk. + + ```c + drwav_data_format format; + format.container = drwav_container_riff; // <-- drwav_container_riff = normal WAV files, drwav_container_w64 = Sony Wave64. + format.format = DR_WAVE_FORMAT_PCM; // <-- Any of the DR_WAVE_FORMAT_* codes. + format.channels = 2; + format.sampleRate = 44100; + format.bitsPerSample = 16; + drwav_init_file_write(&wav, "data/recording.wav", &format, NULL); + + ... + + drwav_uint64 framesWritten = drwav_write_pcm_frames(pWav, frameCount, pSamples); + ``` + +dr_wav has seamless support the Sony Wave64 format. The decoder will automatically detect it and it should Just Work without any manual intervention. + + +Build Options +============= +#define these options before including this file. + +#define DR_WAV_NO_CONVERSION_API + Disables conversion APIs such as `drwav_read_pcm_frames_f32()` and `drwav_s16_to_f32()`. + +#define DR_WAV_NO_STDIO + Disables APIs that initialize a decoder from a file such as `drwav_init_file()`, `drwav_init_file_write()`, etc. + +#define DR_WAV_NO_WCHAR + Disables all functions ending with `_w`. Use this if your compiler does not provide wchar.h. Not required if DR_WAV_NO_STDIO is also defined. + + + +Notes +===== +- Samples are always interleaved. +- The default read function does not do any data conversion. Use `drwav_read_pcm_frames_f32()`, `drwav_read_pcm_frames_s32()` and `drwav_read_pcm_frames_s16()` + to read and convert audio data to 32-bit floating point, signed 32-bit integer and signed 16-bit integer samples respectively. Tested and supported internal + formats include the following: + - Unsigned 8-bit PCM + - Signed 12-bit PCM + - Signed 16-bit PCM + - Signed 24-bit PCM + - Signed 32-bit PCM + - IEEE 32-bit floating point + - IEEE 64-bit floating point + - A-law and u-law + - Microsoft ADPCM + - IMA ADPCM (DVI, format code 0x11) +- dr_wav will try to read the WAV file as best it can, even if it's not strictly conformant to the WAV format. +*/ + +#ifndef dr_wav_h +#define dr_wav_h + +#ifdef __cplusplus +extern "C" { +#endif + +#define DRWAV_STRINGIFY(x) #x +#define DRWAV_XSTRINGIFY(x) DRWAV_STRINGIFY(x) + +#define DRWAV_VERSION_MAJOR 0 +#define DRWAV_VERSION_MINOR 13 +#define DRWAV_VERSION_REVISION 7 +#define DRWAV_VERSION_STRING DRWAV_XSTRINGIFY(DRWAV_VERSION_MAJOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_MINOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_REVISION) + +#include /* For size_t. */ + +/* Sized types. */ +typedef signed char drwav_int8; +typedef unsigned char drwav_uint8; +typedef signed short drwav_int16; +typedef unsigned short drwav_uint16; +typedef signed int drwav_int32; +typedef unsigned int drwav_uint32; +#if defined(_MSC_VER) && !defined(__clang__) + typedef signed __int64 drwav_int64; + typedef unsigned __int64 drwav_uint64; +#else + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wlong-long" + #if defined(__clang__) + #pragma GCC diagnostic ignored "-Wc++11-long-long" + #endif + #endif + typedef signed long long drwav_int64; + typedef unsigned long long drwav_uint64; + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop + #endif +#endif +#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__) + typedef drwav_uint64 drwav_uintptr; +#else + typedef drwav_uint32 drwav_uintptr; +#endif +typedef drwav_uint8 drwav_bool8; +typedef drwav_uint32 drwav_bool32; +#define DRWAV_TRUE 1 +#define DRWAV_FALSE 0 + +#if !defined(DRWAV_API) + #if defined(DRWAV_DLL) + #if defined(_WIN32) + #define DRWAV_DLL_IMPORT __declspec(dllimport) + #define DRWAV_DLL_EXPORT __declspec(dllexport) + #define DRWAV_DLL_PRIVATE static + #else + #if defined(__GNUC__) && __GNUC__ >= 4 + #define DRWAV_DLL_IMPORT __attribute__((visibility("default"))) + #define DRWAV_DLL_EXPORT __attribute__((visibility("default"))) + #define DRWAV_DLL_PRIVATE __attribute__((visibility("hidden"))) + #else + #define DRWAV_DLL_IMPORT + #define DRWAV_DLL_EXPORT + #define DRWAV_DLL_PRIVATE static + #endif + #endif + + #if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION) + #define DRWAV_API DRWAV_DLL_EXPORT + #else + #define DRWAV_API DRWAV_DLL_IMPORT + #endif + #define DRWAV_PRIVATE DRWAV_DLL_PRIVATE + #else + #define DRWAV_API extern + #define DRWAV_PRIVATE static + #endif +#endif + +typedef drwav_int32 drwav_result; +#define DRWAV_SUCCESS 0 +#define DRWAV_ERROR -1 /* A generic error. */ +#define DRWAV_INVALID_ARGS -2 +#define DRWAV_INVALID_OPERATION -3 +#define DRWAV_OUT_OF_MEMORY -4 +#define DRWAV_OUT_OF_RANGE -5 +#define DRWAV_ACCESS_DENIED -6 +#define DRWAV_DOES_NOT_EXIST -7 +#define DRWAV_ALREADY_EXISTS -8 +#define DRWAV_TOO_MANY_OPEN_FILES -9 +#define DRWAV_INVALID_FILE -10 +#define DRWAV_TOO_BIG -11 +#define DRWAV_PATH_TOO_LONG -12 +#define DRWAV_NAME_TOO_LONG -13 +#define DRWAV_NOT_DIRECTORY -14 +#define DRWAV_IS_DIRECTORY -15 +#define DRWAV_DIRECTORY_NOT_EMPTY -16 +#define DRWAV_END_OF_FILE -17 +#define DRWAV_NO_SPACE -18 +#define DRWAV_BUSY -19 +#define DRWAV_IO_ERROR -20 +#define DRWAV_INTERRUPT -21 +#define DRWAV_UNAVAILABLE -22 +#define DRWAV_ALREADY_IN_USE -23 +#define DRWAV_BAD_ADDRESS -24 +#define DRWAV_BAD_SEEK -25 +#define DRWAV_BAD_PIPE -26 +#define DRWAV_DEADLOCK -27 +#define DRWAV_TOO_MANY_LINKS -28 +#define DRWAV_NOT_IMPLEMENTED -29 +#define DRWAV_NO_MESSAGE -30 +#define DRWAV_BAD_MESSAGE -31 +#define DRWAV_NO_DATA_AVAILABLE -32 +#define DRWAV_INVALID_DATA -33 +#define DRWAV_TIMEOUT -34 +#define DRWAV_NO_NETWORK -35 +#define DRWAV_NOT_UNIQUE -36 +#define DRWAV_NOT_SOCKET -37 +#define DRWAV_NO_ADDRESS -38 +#define DRWAV_BAD_PROTOCOL -39 +#define DRWAV_PROTOCOL_UNAVAILABLE -40 +#define DRWAV_PROTOCOL_NOT_SUPPORTED -41 +#define DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED -42 +#define DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED -43 +#define DRWAV_SOCKET_NOT_SUPPORTED -44 +#define DRWAV_CONNECTION_RESET -45 +#define DRWAV_ALREADY_CONNECTED -46 +#define DRWAV_NOT_CONNECTED -47 +#define DRWAV_CONNECTION_REFUSED -48 +#define DRWAV_NO_HOST -49 +#define DRWAV_IN_PROGRESS -50 +#define DRWAV_CANCELLED -51 +#define DRWAV_MEMORY_ALREADY_MAPPED -52 +#define DRWAV_AT_END -53 + +/* Common data formats. */ +#define DR_WAVE_FORMAT_PCM 0x1 +#define DR_WAVE_FORMAT_ADPCM 0x2 +#define DR_WAVE_FORMAT_IEEE_FLOAT 0x3 +#define DR_WAVE_FORMAT_ALAW 0x6 +#define DR_WAVE_FORMAT_MULAW 0x7 +#define DR_WAVE_FORMAT_DVI_ADPCM 0x11 +#define DR_WAVE_FORMAT_EXTENSIBLE 0xFFFE + +/* Flags to pass into drwav_init_ex(), etc. */ +#define DRWAV_SEQUENTIAL 0x00000001 + +DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision); +DRWAV_API const char* drwav_version_string(void); + +typedef enum +{ + drwav_seek_origin_start, + drwav_seek_origin_current +} drwav_seek_origin; + +typedef enum +{ + drwav_container_riff, + drwav_container_w64, + drwav_container_rf64 +} drwav_container; + +typedef struct +{ + union + { + drwav_uint8 fourcc[4]; + drwav_uint8 guid[16]; + } id; + + /* The size in bytes of the chunk. */ + drwav_uint64 sizeInBytes; + + /* + RIFF = 2 byte alignment. + W64 = 8 byte alignment. + */ + unsigned int paddingSize; +} drwav_chunk_header; + +typedef struct +{ + /* + The format tag exactly as specified in the wave file's "fmt" chunk. This can be used by applications + that require support for data formats not natively supported by dr_wav. + */ + drwav_uint16 formatTag; + + /* The number of channels making up the audio data. When this is set to 1 it is mono, 2 is stereo, etc. */ + drwav_uint16 channels; + + /* The sample rate. Usually set to something like 44100. */ + drwav_uint32 sampleRate; + + /* Average bytes per second. You probably don't need this, but it's left here for informational purposes. */ + drwav_uint32 avgBytesPerSec; + + /* Block align. This is equal to the number of channels * bytes per sample. */ + drwav_uint16 blockAlign; + + /* Bits per sample. */ + drwav_uint16 bitsPerSample; + + /* The size of the extended data. Only used internally for validation, but left here for informational purposes. */ + drwav_uint16 extendedSize; + + /* + The number of valid bits per sample. When is equal to WAVE_FORMAT_EXTENSIBLE, + is always rounded up to the nearest multiple of 8. This variable contains information about exactly how + many bits are valid per sample. Mainly used for informational purposes. + */ + drwav_uint16 validBitsPerSample; + + /* The channel mask. Not used at the moment. */ + drwav_uint32 channelMask; + + /* The sub-format, exactly as specified by the wave file. */ + drwav_uint8 subFormat[16]; +} drwav_fmt; + +DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT); + + +/* +Callback for when data is read. Return value is the number of bytes actually read. + +pUserData [in] The user data that was passed to drwav_init() and family. +pBufferOut [out] The output buffer. +bytesToRead [in] The number of bytes to read. + +Returns the number of bytes actually read. + +A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until +either the entire bytesToRead is filled or you have reached the end of the stream. +*/ +typedef size_t (* drwav_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data is written. Returns value is the number of bytes actually written. + +pUserData [in] The user data that was passed to drwav_init_write() and family. +pData [out] A pointer to the data to write. +bytesToWrite [in] The number of bytes to write. + +Returns the number of bytes actually written. + +If the return value differs from bytesToWrite, it indicates an error. +*/ +typedef size_t (* drwav_write_proc)(void* pUserData, const void* pData, size_t bytesToWrite); + +/* +Callback for when data needs to be seeked. + +pUserData [in] The user data that was passed to drwav_init() and family. +offset [in] The number of bytes to move, relative to the origin. Will never be negative. +origin [in] The origin of the seek - the current position or the start of the stream. + +Returns whether or not the seek was successful. + +Whether or not it is relative to the beginning or current position is determined by the "origin" parameter which will be either drwav_seek_origin_start or +drwav_seek_origin_current. +*/ +typedef drwav_bool32 (* drwav_seek_proc)(void* pUserData, int offset, drwav_seek_origin origin); + +/* +Callback for when drwav_init_ex() finds a chunk. + +pChunkUserData [in] The user data that was passed to the pChunkUserData parameter of drwav_init_ex() and family. +onRead [in] A pointer to the function to call when reading. +onSeek [in] A pointer to the function to call when seeking. +pReadSeekUserData [in] The user data that was passed to the pReadSeekUserData parameter of drwav_init_ex() and family. +pChunkHeader [in] A pointer to an object containing basic header information about the chunk. Use this to identify the chunk. +container [in] Whether or not the WAV file is a RIFF or Wave64 container. If you're unsure of the difference, assume RIFF. +pFMT [in] A pointer to the object containing the contents of the "fmt" chunk. + +Returns the number of bytes read + seeked. + +To read data from the chunk, call onRead(), passing in pReadSeekUserData as the first parameter. Do the same for seeking with onSeek(). The return value must +be the total number of bytes you have read _plus_ seeked. + +Use the `container` argument to discriminate the fields in `pChunkHeader->id`. If the container is `drwav_container_riff` or `drwav_container_rf64` you should +use `id.fourcc`, otherwise you should use `id.guid`. + +The `pFMT` parameter can be used to determine the data format of the wave file. Use `drwav_fmt_get_format()` to get the sample format, which will be one of the +`DR_WAVE_FORMAT_*` identifiers. + +The read pointer will be sitting on the first byte after the chunk's header. You must not attempt to read beyond the boundary of the chunk. +*/ +typedef drwav_uint64 (* drwav_chunk_proc)(void* pChunkUserData, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_chunk_header* pChunkHeader, drwav_container container, const drwav_fmt* pFMT); + +typedef struct +{ + void* pUserData; + void* (* onMalloc)(size_t sz, void* pUserData); + void* (* onRealloc)(void* p, size_t sz, void* pUserData); + void (* onFree)(void* p, void* pUserData); +} drwav_allocation_callbacks; + +/* Structure for internal use. Only used for loaders opened with drwav_init_memory(). */ +typedef struct +{ + const drwav_uint8* data; + size_t dataSize; + size_t currentReadPos; +} drwav__memory_stream; + +/* Structure for internal use. Only used for writers opened with drwav_init_memory_write(). */ +typedef struct +{ + void** ppData; + size_t* pDataSize; + size_t dataSize; + size_t dataCapacity; + size_t currentWritePos; +} drwav__memory_stream_write; + +typedef struct +{ + drwav_container container; /* RIFF, W64. */ + drwav_uint32 format; /* DR_WAVE_FORMAT_* */ + drwav_uint32 channels; + drwav_uint32 sampleRate; + drwav_uint32 bitsPerSample; +} drwav_data_format; + +typedef enum +{ + drwav_metadata_type_none = 0, + + /* + Unknown simply means a chunk that drwav does not handle specifically. You can still ask to + receive these chunks as metadata objects. It is then up to you to interpret the chunk's data. + You can also write unknown metadata to a wav file. Be careful writing unknown chunks if you + have also edited the audio data. The unknown chunks could represent offsets/sizes that no + longer correctly correspond to the audio data. + */ + drwav_metadata_type_unknown = 1 << 0, + + /* Only 1 of each of these metadata items are allowed in a wav file. */ + drwav_metadata_type_smpl = 1 << 1, + drwav_metadata_type_inst = 1 << 2, + drwav_metadata_type_cue = 1 << 3, + drwav_metadata_type_acid = 1 << 4, + drwav_metadata_type_bext = 1 << 5, + + /* + Wav files often have a LIST chunk. This is a chunk that contains a set of subchunks. For this + higher-level metadata API, we don't make a distinction between a regular chunk and a LIST + subchunk. Instead, they are all just 'metadata' items. + + There can be multiple of these metadata items in a wav file. + */ + drwav_metadata_type_list_label = 1 << 6, + drwav_metadata_type_list_note = 1 << 7, + drwav_metadata_type_list_labelled_cue_region = 1 << 8, + + drwav_metadata_type_list_info_software = 1 << 9, + drwav_metadata_type_list_info_copyright = 1 << 10, + drwav_metadata_type_list_info_title = 1 << 11, + drwav_metadata_type_list_info_artist = 1 << 12, + drwav_metadata_type_list_info_comment = 1 << 13, + drwav_metadata_type_list_info_date = 1 << 14, + drwav_metadata_type_list_info_genre = 1 << 15, + drwav_metadata_type_list_info_album = 1 << 16, + drwav_metadata_type_list_info_tracknumber = 1 << 17, + + /* Other type constants for convenience. */ + drwav_metadata_type_list_all_info_strings = drwav_metadata_type_list_info_software + | drwav_metadata_type_list_info_copyright + | drwav_metadata_type_list_info_title + | drwav_metadata_type_list_info_artist + | drwav_metadata_type_list_info_comment + | drwav_metadata_type_list_info_date + | drwav_metadata_type_list_info_genre + | drwav_metadata_type_list_info_album + | drwav_metadata_type_list_info_tracknumber, + + drwav_metadata_type_list_all_adtl = drwav_metadata_type_list_label + | drwav_metadata_type_list_note + | drwav_metadata_type_list_labelled_cue_region, + + drwav_metadata_type_all = -2, /*0xFFFFFFFF & ~drwav_metadata_type_unknown,*/ + drwav_metadata_type_all_including_unknown = -1 /*0xFFFFFFFF,*/ +} drwav_metadata_type; + +/* +Sampler Metadata + +The sampler chunk contains information about how a sound should be played in the context of a whole +audio production, and when used in a sampler. See https://en.wikipedia.org/wiki/Sample-based_synthesis. +*/ +typedef enum +{ + drwav_smpl_loop_type_forward = 0, + drwav_smpl_loop_type_pingpong = 1, + drwav_smpl_loop_type_backward = 2 +} drwav_smpl_loop_type; + +typedef struct +{ + /* The ID of the associated cue point, see drwav_cue and drwav_cue_point. As with all cue point IDs, this can correspond to a label chunk to give this loop a name, see drwav_list_label_or_note. */ + drwav_uint32 cuePointId; + + /* See drwav_smpl_loop_type. */ + drwav_uint32 type; + + /* The byte offset of the first sample to be played in the loop. */ + drwav_uint32 firstSampleByteOffset; + + /* The byte offset into the audio data of the last sample to be played in the loop. */ + drwav_uint32 lastSampleByteOffset; + + /* A value to represent that playback should occur at a point between samples. This value ranges from 0 to UINT32_MAX. Where a value of 0 means no fraction, and a value of (UINT32_MAX / 2) would mean half a sample. */ + drwav_uint32 sampleFraction; + + /* Number of times to play the loop. 0 means loop infinitely. */ + drwav_uint32 playCount; +} drwav_smpl_loop; + +typedef struct +{ + /* IDs for a particular MIDI manufacturer. 0 if not used. */ + drwav_uint32 manufacturerId; + drwav_uint32 productId; + + /* The period of 1 sample in nanoseconds. */ + drwav_uint32 samplePeriodNanoseconds; + + /* The MIDI root note of this file. 0 to 127. */ + drwav_uint32 midiUnityNote; + + /* The fraction of a semitone up from the given MIDI note. This is a value from 0 to UINT32_MAX, where 0 means no change and (UINT32_MAX / 2) is half a semitone (AKA 50 cents). */ + drwav_uint32 midiPitchFraction; + + /* Data relating to SMPTE standards which are used for syncing audio and video. 0 if not used. */ + drwav_uint32 smpteFormat; + drwav_uint32 smpteOffset; + + /* drwav_smpl_loop loops. */ + drwav_uint32 sampleLoopCount; + + /* Optional sampler-specific data. */ + drwav_uint32 samplerSpecificDataSizeInBytes; + + drwav_smpl_loop* pLoops; + drwav_uint8* pSamplerSpecificData; +} drwav_smpl; + +/* +Instrument Metadata + +The inst metadata contains data about how a sound should be played as part of an instrument. This +commonly read by samplers. See https://en.wikipedia.org/wiki/Sample-based_synthesis. +*/ +typedef struct +{ + drwav_int8 midiUnityNote; /* The root note of the audio as a MIDI note number. 0 to 127. */ + drwav_int8 fineTuneCents; /* -50 to +50 */ + drwav_int8 gainDecibels; /* -64 to +64 */ + drwav_int8 lowNote; /* 0 to 127 */ + drwav_int8 highNote; /* 0 to 127 */ + drwav_int8 lowVelocity; /* 1 to 127 */ + drwav_int8 highVelocity; /* 1 to 127 */ +} drwav_inst; + +/* +Cue Metadata + +Cue points are markers at specific points in the audio. They often come with an associated piece of +drwav_list_label_or_note metadata which contains the text for the marker. +*/ +typedef struct +{ + /* Unique identification value. */ + drwav_uint32 id; + + /* Set to 0. This is only relevant if there is a 'playlist' chunk - which is not supported by dr_wav. */ + drwav_uint32 playOrderPosition; + + /* Should always be "data". This represents the fourcc value of the chunk that this cue point corresponds to. dr_wav only supports a single data chunk so this should always be "data". */ + drwav_uint8 dataChunkId[4]; + + /* Set to 0. This is only relevant if there is a wave list chunk. dr_wav, like lots of readers/writers, do not support this. */ + drwav_uint32 chunkStart; + + /* Set to 0 for uncompressed formats. Else the last byte in compressed wave data where decompression can begin to find the value of the corresponding sample value. */ + drwav_uint32 blockStart; + + /* For uncompressed formats this is the byte offset of the cue point into the audio data. For compressed formats this is relative to the block specified with blockStart. */ + drwav_uint32 sampleByteOffset; +} drwav_cue_point; + +typedef struct +{ + drwav_uint32 cuePointCount; + drwav_cue_point *pCuePoints; +} drwav_cue; + +/* +Acid Metadata + +This chunk contains some information about the time signature and the tempo of the audio. +*/ +typedef enum +{ + drwav_acid_flag_one_shot = 1, /* If this is not set, then it is a loop instead of a one-shot. */ + drwav_acid_flag_root_note_set = 2, + drwav_acid_flag_stretch = 4, + drwav_acid_flag_disk_based = 8, + drwav_acid_flag_acidizer = 16 /* Not sure what this means. */ +} drwav_acid_flag; + +typedef struct +{ + /* A bit-field, see drwav_acid_flag. */ + drwav_uint32 flags; + + /* Valid if flags contains drwav_acid_flag_root_note_set. It represents the MIDI root note the file - a value from 0 to 127. */ + drwav_uint16 midiUnityNote; + + /* Reserved values that should probably be ignored. reserved1 seems to often be 128 and reserved2 is 0. */ + drwav_uint16 reserved1; + float reserved2; + + /* Number of beats. */ + drwav_uint32 numBeats; + + /* The time signature of the audio. */ + drwav_uint16 meterDenominator; + drwav_uint16 meterNumerator; + + /* Beats per minute of the track. Setting a value of 0 suggests that there is no tempo. */ + float tempo; +} drwav_acid; + +/* +Cue Label or Note metadata + +These are 2 different types of metadata, but they have the exact same format. Labels tend to be the +more common and represent a short name for a cue point. Notes might be used to represent a longer +comment. +*/ +typedef struct +{ + /* The ID of a cue point that this label or note corresponds to. */ + drwav_uint32 cuePointId; + + /* Size of the string not including any null terminator. */ + drwav_uint32 stringLength; + + /* The string. The *init_with_metadata functions null terminate this for convenience. */ + char* pString; +} drwav_list_label_or_note; + +/* +BEXT metadata, also known as Broadcast Wave Format (BWF) + +This metadata adds some extra description to an audio file. You must check the version field to +determine if the UMID or the loudness fields are valid. +*/ +typedef struct +{ + /* + These top 3 fields, and the umid field are actually defined in the standard as a statically + sized buffers. In order to reduce the size of this struct (and therefore the union in the + metadata struct), we instead store these as pointers. + */ + char* pDescription; /* Can be NULL or a null-terminated string, must be <= 256 characters. */ + char* pOriginatorName; /* Can be NULL or a null-terminated string, must be <= 32 characters. */ + char* pOriginatorReference; /* Can be NULL or a null-terminated string, must be <= 32 characters. */ + char pOriginationDate[10]; /* ASCII "yyyy:mm:dd". */ + char pOriginationTime[8]; /* ASCII "hh:mm:ss". */ + drwav_uint64 timeReference; /* First sample count since midnight. */ + drwav_uint16 version; /* Version of the BWF, check this to see if the fields below are valid. */ + + /* + Unrestricted ASCII characters containing a collection of strings terminated by CR/LF. Each + string shall contain a description of a coding process applied to the audio data. + */ + char* pCodingHistory; + drwav_uint32 codingHistorySize; + + /* Fields below this point are only valid if the version is 1 or above. */ + drwav_uint8* pUMID; /* Exactly 64 bytes of SMPTE UMID */ + + /* Fields below this point are only valid if the version is 2 or above. */ + drwav_uint16 loudnessValue; /* Integrated Loudness Value of the file in LUFS (multiplied by 100). */ + drwav_uint16 loudnessRange; /* Loudness Range of the file in LU (multiplied by 100). */ + drwav_uint16 maxTruePeakLevel; /* Maximum True Peak Level of the file expressed as dBTP (multiplied by 100). */ + drwav_uint16 maxMomentaryLoudness; /* Highest value of the Momentary Loudness Level of the file in LUFS (multiplied by 100). */ + drwav_uint16 maxShortTermLoudness; /* Highest value of the Short-Term Loudness Level of the file in LUFS (multiplied by 100). */ +} drwav_bext; + +/* +Info Text Metadata + +There a many different types of information text that can be saved in this format. This is where +things like the album name, the artists, the year it was produced, etc are saved. See +drwav_metadata_type for the full list of types that dr_wav supports. +*/ +typedef struct +{ + /* Size of the string not including any null terminator. */ + drwav_uint32 stringLength; + + /* The string. The *init_with_metadata functions null terminate this for convenience. */ + char* pString; +} drwav_list_info_text; + +/* +Labelled Cue Region Metadata + +The labelled cue region metadata is used to associate some region of audio with text. The region +starts at a cue point, and extends for the given number of samples. +*/ +typedef struct +{ + /* The ID of a cue point that this object corresponds to. */ + drwav_uint32 cuePointId; + + /* The number of samples from the cue point forwards that should be considered this region */ + drwav_uint32 sampleLength; + + /* Four characters used to say what the purpose of this region is. */ + drwav_uint8 purposeId[4]; + + /* Unsure of the exact meanings of these. It appears to be acceptable to set them all to 0. */ + drwav_uint16 country; + drwav_uint16 language; + drwav_uint16 dialect; + drwav_uint16 codePage; + + /* Size of the string not including any null terminator. */ + drwav_uint32 stringLength; + + /* The string. The *init_with_metadata functions null terminate this for convenience. */ + char* pString; +} drwav_list_labelled_cue_region; + +/* +Unknown Metadata + +This chunk just represents a type of chunk that dr_wav does not understand. + +Unknown metadata has a location attached to it. This is because wav files can have a LIST chunk +that contains subchunks. These LIST chunks can be one of two types. An adtl list, or an INFO +list. This enum is used to specify the location of a chunk that dr_wav currently doesn't support. +*/ +typedef enum +{ + drwav_metadata_location_invalid, + drwav_metadata_location_top_level, + drwav_metadata_location_inside_info_list, + drwav_metadata_location_inside_adtl_list +} drwav_metadata_location; + +typedef struct +{ + drwav_uint8 id[4]; + drwav_metadata_location chunkLocation; + drwav_uint32 dataSizeInBytes; + drwav_uint8* pData; +} drwav_unknown_metadata; + +/* +Metadata is saved as a union of all the supported types. +*/ +typedef struct +{ + /* Determines which item in the union is valid. */ + drwav_metadata_type type; + + union + { + drwav_cue cue; + drwav_smpl smpl; + drwav_acid acid; + drwav_inst inst; + drwav_bext bext; + drwav_list_label_or_note labelOrNote; /* List label or list note. */ + drwav_list_labelled_cue_region labelledCueRegion; + drwav_list_info_text infoText; /* Any of the list info types. */ + drwav_unknown_metadata unknown; + } data; +} drwav_metadata; + +typedef struct +{ + /* A pointer to the function to call when more data is needed. */ + drwav_read_proc onRead; + + /* A pointer to the function to call when data needs to be written. Only used when the drwav object is opened in write mode. */ + drwav_write_proc onWrite; + + /* A pointer to the function to call when the wav file needs to be seeked. */ + drwav_seek_proc onSeek; + + /* The user data to pass to callbacks. */ + void* pUserData; + + /* Allocation callbacks. */ + drwav_allocation_callbacks allocationCallbacks; + + + /* Whether or not the WAV file is formatted as a standard RIFF file or W64. */ + drwav_container container; + + + /* Structure containing format information exactly as specified by the wav file. */ + drwav_fmt fmt; + + /* The sample rate. Will be set to something like 44100. */ + drwav_uint32 sampleRate; + + /* The number of channels. This will be set to 1 for monaural streams, 2 for stereo, etc. */ + drwav_uint16 channels; + + /* The bits per sample. Will be set to something like 16, 24, etc. */ + drwav_uint16 bitsPerSample; + + /* Equal to fmt.formatTag, or the value specified by fmt.subFormat if fmt.formatTag is equal to 65534 (WAVE_FORMAT_EXTENSIBLE). */ + drwav_uint16 translatedFormatTag; + + /* The total number of PCM frames making up the audio data. */ + drwav_uint64 totalPCMFrameCount; + + + /* The size in bytes of the data chunk. */ + drwav_uint64 dataChunkDataSize; + + /* The position in the stream of the first data byte of the data chunk. This is used for seeking. */ + drwav_uint64 dataChunkDataPos; + + /* The number of bytes remaining in the data chunk. */ + drwav_uint64 bytesRemaining; + + /* The current read position in PCM frames. */ + drwav_uint64 readCursorInPCMFrames; + + + /* + Only used in sequential write mode. Keeps track of the desired size of the "data" chunk at the point of initialization time. Always + set to 0 for non-sequential writes and when the drwav object is opened in read mode. Used for validation. + */ + drwav_uint64 dataChunkDataSizeTargetWrite; + + /* Keeps track of whether or not the wav writer was initialized in sequential mode. */ + drwav_bool32 isSequentialWrite; + + + /* A bit-field of drwav_metadata_type values, only bits set in this variable are parsed and saved */ + drwav_metadata_type allowedMetadataTypes; + + /* A array of metadata. This is valid after the *init_with_metadata call returns. It will be valid until drwav_uninit() is called. You can take ownership of this data with drwav_take_ownership_of_metadata(). */ + drwav_metadata* pMetadata; + drwav_uint32 metadataCount; + + + /* A hack to avoid a DRWAV_MALLOC() when opening a decoder with drwav_init_memory(). */ + drwav__memory_stream memoryStream; + drwav__memory_stream_write memoryStreamWrite; + + + /* Microsoft ADPCM specific data. */ + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_uint16 predictor[2]; + drwav_int32 delta[2]; + drwav_int32 cachedFrames[4]; /* Samples are stored in this cache during decoding. */ + drwav_uint32 cachedFrameCount; + drwav_int32 prevFrames[2][2]; /* The previous 2 samples for each channel (2 channels at most). */ + } msadpcm; + + /* IMA ADPCM specific data. */ + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_int32 predictor[2]; + drwav_int32 stepIndex[2]; + drwav_int32 cachedFrames[16]; /* Samples are stored in this cache during decoding. */ + drwav_uint32 cachedFrameCount; + } ima; +} drwav; + + +/* +Initializes a pre-allocated drwav object for reading. + +pWav [out] A pointer to the drwav object being initialized. +onRead [in] The function to call when data needs to be read from the client. +onSeek [in] The function to call when the read position of the client data needs to move. +onChunk [in, optional] The function to call when a chunk is enumerated at initialized time. +pUserData, pReadSeekUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. +pChunkUserData [in, optional] A pointer to application defined data that will be passed to onChunk. +flags [in, optional] A set of flags for controlling how things are loaded. + +Returns true if successful; false otherwise. + +Close the loader with drwav_uninit(). + +This is the lowest level function for initializing a WAV file. You can also use drwav_init_file() and drwav_init_memory() +to open the stream from a file or from a block of memory respectively. + +Possible values for flags: + DRWAV_SEQUENTIAL: Never perform a backwards seek while loading. This disables the chunk callback and will cause this function + to return as soon as the data chunk is found. Any chunks after the data chunk will be ignored. + +drwav_init() is equivalent to "drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0);". + +The onChunk callback is not called for the WAVE or FMT chunks. The contents of the FMT chunk can be read from pWav->fmt +after the function returns. + +See also: drwav_init_file(), drwav_init_memory(), drwav_uninit() +*/ +DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* +Initializes a pre-allocated drwav object for writing. + +onWrite [in] The function to call when data needs to be written. +onSeek [in] The function to call when the write position needs to move. +pUserData [in, optional] A pointer to application defined data that will be passed to onWrite and onSeek. +metadata, numMetadata [in, optional] An array of metadata objects that should be written to the file. The array is not edited. You are responsible for this metadata memory and it must maintain valid until drwav_uninit() is called. + +Returns true if successful; false otherwise. + +Close the writer with drwav_uninit(). + +This is the lowest level function for initializing a WAV file. You can also use drwav_init_file_write() and drwav_init_memory_write() +to open the stream from a file or from a block of memory respectively. + +If the total sample count is known, you can use drwav_init_write_sequential(). This avoids the need for dr_wav to perform +a post-processing step for storing the total sample count and the size of the data chunk which requires a backwards seek. + +See also: drwav_init_file_write(), drwav_init_memory_write(), drwav_uninit() +*/ +DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata, drwav_uint32 metadataCount); + +/* +Utility function to determine the target size of the entire data to be written (including all headers and chunks). + +Returns the target size in bytes. + +The metadata argument can be NULL meaning no metadata exists. + +Useful if the application needs to know the size to allocate. + +Only writing to the RIFF chunk and one data chunk is currently supported. + +See also: drwav_init_write(), drwav_init_file_write(), drwav_init_memory_write() +*/ +DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount); + +/* +Take ownership of the metadata objects that were allocated via one of the init_with_metadata() function calls. The init_with_metdata functions perform a single heap allocation for this metadata. + +Useful if you want the data to persist beyond the lifetime of the drwav object. + +You must free the data returned from this function using drwav_free(). +*/ +DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav); + +/* +Uninitializes the given drwav object. + +Use this only for objects initialized with drwav_init*() functions (drwav_init(), drwav_init_ex(), drwav_init_write(), drwav_init_write_sequential()). +*/ +DRWAV_API drwav_result drwav_uninit(drwav* pWav); + + +/* +Reads raw audio data. + +This is the lowest level function for reading audio data. It simply reads the given number of +bytes of the raw internal sample data. + +Consider using drwav_read_pcm_frames_s16(), drwav_read_pcm_frames_s32() or drwav_read_pcm_frames_f32() for +reading sample data in a consistent format. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of bytes actually read. +*/ +DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut); + +/* +Reads up to the specified number of PCM frames from the WAV file. + +The output data will be in the file's internal format, converted to native-endian byte order. Use +drwav_read_pcm_frames_s16/f32/s32() to read data in a specific format. + +If the return value is less than it means the end of the file has been reached or +you have requested more PCM frames than can possibly fit in the output buffer. + +This function will only work when sample data is of a fixed size and uncompressed. If you are +using a compressed format consider using drwav_read_raw() or drwav_read_pcm_frames_s16/s32/f32(). + +pBufferOut can be NULL in which case a seek will be performed. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); + +/* +Seeks to the given PCM frame. + +Returns true if successful; false otherwise. +*/ +DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex); + +/* +Retrieves the current read position in pcm frames. +*/ +DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor); + +/* +Retrieves the length of the file. +*/ +DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength); + + +/* +Writes raw audio data. + +Returns the number of bytes actually written. If this differs from bytesToWrite, it indicates an error. +*/ +DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData); + +/* +Writes PCM frames. + +Returns the number of PCM frames written. + +Input samples need to be in native-endian byte order. On big-endian architectures the input data will be converted to +little-endian. Use drwav_write_raw() to write raw audio data without performing any conversion. +*/ +DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); +DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); +DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); + +/* Conversion Utilities */ +#ifndef DR_WAV_NO_CONVERSION_API + +/* +Reads a chunk of audio data and converts it to signed 16-bit PCM samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 32-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 32-bit floating point samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +/* +Reads a chunk of audio data and converts it to IEEE 32-bit floating point samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 16-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 32-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +/* +Reads a chunk of audio data and converts it to signed 32-bit PCM samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 16-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 32-bit floating point samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +#endif /* DR_WAV_NO_CONVERSION_API */ + + +/* High-Level Convenience Helpers */ + +#ifndef DR_WAV_NO_STDIO +/* +Helper for initializing a wave file for reading using stdio. + +This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + + +/* +Helper for initializing a wave file for writing using stdio. + +This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif /* DR_WAV_NO_STDIO */ + +/* +Helper for initializing a loader from a pre-allocated memory buffer. + +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +the lifetime of the drwav object. + +The buffer should contain the contents of the entire wave file, not just the sample data. +*/ +DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* +Helper for initializing a writer which outputs data to a memory buffer. + +dr_wav will manage the memory allocations, however it is up to the caller to free the data with drwav_free(). + +The buffer will remain allocated even after drwav_uninit() is called. The buffer should not be considered valid +until after drwav_uninit() has been called. +*/ +DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); + + +#ifndef DR_WAV_NO_CONVERSION_API +/* +Opens and reads an entire wav file in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#ifndef DR_WAV_NO_STDIO +/* +Opens and decodes an entire wav file in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif +/* +Opens and decodes an entire wav file from a block of memory in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif + +/* Frees data that was allocated internally by dr_wav. */ +DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* Converts bytes from a wav stream to a sized type of native endian. */ +DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data); +DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data); +DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data); +DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data); +DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data); +DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data); +DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data); + +/* Compares a GUID for the purpose of checking the type of a Wave64 chunk. */ +DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]); + +/* Compares a four-character-code for the purpose of checking the type of a RIFF chunk. */ +DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b); + +#ifdef __cplusplus +} +#endif +#endif /* dr_wav_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION) +#ifndef dr_wav_c +#define dr_wav_c + +#ifdef __MRC__ +/* MrC currently doesn't compile dr_wav correctly with any optimizations enabled. */ +#pragma options opt off +#endif + +#include +#include +#include /* For INT_MAX */ + +#ifndef DR_WAV_NO_STDIO +#include +#ifndef DR_WAV_NO_WCHAR +#include +#endif +#endif + +/* Standard library stuff. */ +#ifndef DRWAV_ASSERT +#include +#define DRWAV_ASSERT(expression) assert(expression) +#endif +#ifndef DRWAV_MALLOC +#define DRWAV_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRWAV_REALLOC +#define DRWAV_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRWAV_FREE +#define DRWAV_FREE(p) free((p)) +#endif +#ifndef DRWAV_COPY_MEMORY +#define DRWAV_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRWAV_ZERO_MEMORY +#define DRWAV_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#ifndef DRWAV_ZERO_OBJECT +#define DRWAV_ZERO_OBJECT(p) DRWAV_ZERO_MEMORY((p), sizeof(*p)) +#endif + +#define drwav_countof(x) (sizeof(x) / sizeof(x[0])) +#define drwav_align(x, a) ((((x) + (a) - 1) / (a)) * (a)) +#define drwav_min(a, b) (((a) < (b)) ? (a) : (b)) +#define drwav_max(a, b) (((a) > (b)) ? (a) : (b)) +#define drwav_clamp(x, lo, hi) (drwav_max((lo), drwav_min((hi), (x)))) +#define drwav_offset_ptr(p, offset) (((drwav_uint8*)(p)) + (offset)) + +#define DRWAV_MAX_SIMD_VECTOR_SIZE 64 /* 64 for AVX-512 in the future. */ + +/* CPU architecture. */ +#if defined(__x86_64__) || defined(_M_X64) + #define DRWAV_X64 +#elif defined(__i386) || defined(_M_IX86) + #define DRWAV_X86 +#elif defined(__arm__) || defined(_M_ARM) + #define DRWAV_ARM +#endif + +#ifdef _MSC_VER + #define DRWAV_INLINE __forceinline +#elif defined(__GNUC__) + /* + I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when + the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some + case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the + command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue + I am using "__inline__" only when we're compiling in strict ANSI mode. + */ + #if defined(__STRICT_ANSI__) + #define DRWAV_GNUC_INLINE_HINT __inline__ + #else + #define DRWAV_GNUC_INLINE_HINT inline + #endif + + #if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__) + #define DRWAV_INLINE DRWAV_GNUC_INLINE_HINT __attribute__((always_inline)) + #else + #define DRWAV_INLINE DRWAV_GNUC_INLINE_HINT + #endif +#elif defined(__WATCOMC__) + #define DRWAV_INLINE __inline +#else + #define DRWAV_INLINE +#endif + +#if defined(SIZE_MAX) + #define DRWAV_SIZE_MAX SIZE_MAX +#else + #if defined(_WIN64) || defined(_LP64) || defined(__LP64__) + #define DRWAV_SIZE_MAX ((drwav_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRWAV_SIZE_MAX 0xFFFFFFFF + #endif +#endif + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #define DRWAV_HAS_BYTESWAP64_INTRINSIC +#elif defined(__clang__) + #if defined(__has_builtin) + #if __has_builtin(__builtin_bswap16) + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap32) + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap64) + #define DRWAV_HAS_BYTESWAP64_INTRINSIC + #endif + #endif +#elif defined(__GNUC__) + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3)) + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #define DRWAV_HAS_BYTESWAP64_INTRINSIC + #endif + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)) + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #endif +#endif + +DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision) +{ + if (pMajor) { + *pMajor = DRWAV_VERSION_MAJOR; + } + + if (pMinor) { + *pMinor = DRWAV_VERSION_MINOR; + } + + if (pRevision) { + *pRevision = DRWAV_VERSION_REVISION; + } +} + +DRWAV_API const char* drwav_version_string(void) +{ + return DRWAV_VERSION_STRING; +} + +/* +These limits are used for basic validation when initializing the decoder. If you exceed these limits, first of all: what on Earth are +you doing?! (Let me know, I'd be curious!) Second, you can adjust these by #define-ing them before the dr_wav implementation. +*/ +#ifndef DRWAV_MAX_SAMPLE_RATE +#define DRWAV_MAX_SAMPLE_RATE 384000 +#endif +#ifndef DRWAV_MAX_CHANNELS +#define DRWAV_MAX_CHANNELS 256 +#endif +#ifndef DRWAV_MAX_BITS_PER_SAMPLE +#define DRWAV_MAX_BITS_PER_SAMPLE 64 +#endif + +static const drwav_uint8 drwavGUID_W64_RIFF[16] = {0x72,0x69,0x66,0x66, 0x2E,0x91, 0xCF,0x11, 0xA5,0xD6, 0x28,0xDB,0x04,0xC1,0x00,0x00}; /* 66666972-912E-11CF-A5D6-28DB04C10000 */ +static const drwav_uint8 drwavGUID_W64_WAVE[16] = {0x77,0x61,0x76,0x65, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 65766177-ACF3-11D3-8CD1-00C04F8EDB8A */ +/*static const drwav_uint8 drwavGUID_W64_JUNK[16] = {0x6A,0x75,0x6E,0x6B, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A};*/ /* 6B6E756A-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_FMT [16] = {0x66,0x6D,0x74,0x20, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 20746D66-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_FACT[16] = {0x66,0x61,0x63,0x74, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 74636166-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_DATA[16] = {0x64,0x61,0x74,0x61, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 61746164-ACF3-11D3-8CD1-00C04F8EDB8A */ +/*static const drwav_uint8 drwavGUID_W64_SMPL[16] = {0x73,0x6D,0x70,0x6C, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A};*/ /* 6C706D73-ACF3-11D3-8CD1-00C04F8EDB8A */ + + +static DRWAV_INLINE int drwav__is_little_endian(void) +{ +#if defined(DRWAV_X86) || defined(DRWAV_X64) + return DRWAV_TRUE; +#elif defined(__BYTE_ORDER) && defined(__LITTLE_ENDIAN) && __BYTE_ORDER == __LITTLE_ENDIAN + return DRWAV_TRUE; +#else + int n = 1; + return (*(char*)&n) == 1; +#endif +} + + +static DRWAV_INLINE void drwav_bytes_to_guid(const drwav_uint8* data, drwav_uint8* guid) +{ + int i; + for (i = 0; i < 16; ++i) { + guid[i] = data[i]; + } +} + + +static DRWAV_INLINE drwav_uint16 drwav__bswap16(drwav_uint16 n) +{ +#ifdef DRWAV_HAS_BYTESWAP16_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_ushort(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap16(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF00) >> 8) | + ((n & 0x00FF) << 8); +#endif +} + +static DRWAV_INLINE drwav_uint32 drwav__bswap32(drwav_uint32 n) +{ +#ifdef DRWAV_HAS_BYTESWAP32_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_ulong(n); + #elif defined(__GNUC__) || defined(__clang__) + #if defined(DRWAV_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(DRWAV_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */ + /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */ + drwav_uint32 r; + __asm__ __volatile__ ( + #if defined(DRWAV_64BIT) + "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */ + #else + "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n) + #endif + ); + return r; + #else + return __builtin_bswap32(n); + #endif + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF000000) >> 24) | + ((n & 0x00FF0000) >> 8) | + ((n & 0x0000FF00) << 8) | + ((n & 0x000000FF) << 24); +#endif +} + +static DRWAV_INLINE drwav_uint64 drwav__bswap64(drwav_uint64 n) +{ +#ifdef DRWAV_HAS_BYTESWAP64_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_uint64(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap64(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + /* Weird "<< 32" bitshift is required for C89 because it doesn't support 64-bit constants. Should be optimized out by a good compiler. */ + return ((n & ((drwav_uint64)0xFF000000 << 32)) >> 56) | + ((n & ((drwav_uint64)0x00FF0000 << 32)) >> 40) | + ((n & ((drwav_uint64)0x0000FF00 << 32)) >> 24) | + ((n & ((drwav_uint64)0x000000FF << 32)) >> 8) | + ((n & ((drwav_uint64)0xFF000000 )) << 8) | + ((n & ((drwav_uint64)0x00FF0000 )) << 24) | + ((n & ((drwav_uint64)0x0000FF00 )) << 40) | + ((n & ((drwav_uint64)0x000000FF )) << 56); +#endif +} + + +static DRWAV_INLINE drwav_int16 drwav__bswap_s16(drwav_int16 n) +{ + return (drwav_int16)drwav__bswap16((drwav_uint16)n); +} + +static DRWAV_INLINE void drwav__bswap_samples_s16(drwav_int16* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_s16(pSamples[iSample]); + } +} + + +static DRWAV_INLINE void drwav__bswap_s24(drwav_uint8* p) +{ + drwav_uint8 t; + t = p[0]; + p[0] = p[2]; + p[2] = t; +} + +static DRWAV_INLINE void drwav__bswap_samples_s24(drwav_uint8* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + drwav_uint8* pSample = pSamples + (iSample*3); + drwav__bswap_s24(pSample); + } +} + + +static DRWAV_INLINE drwav_int32 drwav__bswap_s32(drwav_int32 n) +{ + return (drwav_int32)drwav__bswap32((drwav_uint32)n); +} + +static DRWAV_INLINE void drwav__bswap_samples_s32(drwav_int32* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_s32(pSamples[iSample]); + } +} + + +static DRWAV_INLINE float drwav__bswap_f32(float n) +{ + union { + drwav_uint32 i; + float f; + } x; + x.f = n; + x.i = drwav__bswap32(x.i); + + return x.f; +} + +static DRWAV_INLINE void drwav__bswap_samples_f32(float* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_f32(pSamples[iSample]); + } +} + + +static DRWAV_INLINE double drwav__bswap_f64(double n) +{ + union { + drwav_uint64 i; + double f; + } x; + x.f = n; + x.i = drwav__bswap64(x.i); + + return x.f; +} + +static DRWAV_INLINE void drwav__bswap_samples_f64(double* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_f64(pSamples[iSample]); + } +} + + +static DRWAV_INLINE void drwav__bswap_samples_pcm(void* pSamples, drwav_uint64 sampleCount, drwav_uint32 bytesPerSample) +{ + /* Assumes integer PCM. Floating point PCM is done in drwav__bswap_samples_ieee(). */ + switch (bytesPerSample) + { + case 1: /* u8 */ + { + /* no-op. */ + } break; + case 2: /* s16, s12 (loosely packed) */ + { + drwav__bswap_samples_s16((drwav_int16*)pSamples, sampleCount); + } break; + case 3: /* s24 */ + { + drwav__bswap_samples_s24((drwav_uint8*)pSamples, sampleCount); + } break; + case 4: /* s32 */ + { + drwav__bswap_samples_s32((drwav_int32*)pSamples, sampleCount); + } break; + default: + { + /* Unsupported format. */ + DRWAV_ASSERT(DRWAV_FALSE); + } break; + } +} + +static DRWAV_INLINE void drwav__bswap_samples_ieee(void* pSamples, drwav_uint64 sampleCount, drwav_uint32 bytesPerSample) +{ + switch (bytesPerSample) + { + #if 0 /* Contributions welcome for f16 support. */ + case 2: /* f16 */ + { + drwav__bswap_samples_f16((drwav_float16*)pSamples, sampleCount); + } break; + #endif + case 4: /* f32 */ + { + drwav__bswap_samples_f32((float*)pSamples, sampleCount); + } break; + case 8: /* f64 */ + { + drwav__bswap_samples_f64((double*)pSamples, sampleCount); + } break; + default: + { + /* Unsupported format. */ + DRWAV_ASSERT(DRWAV_FALSE); + } break; + } +} + +static DRWAV_INLINE void drwav__bswap_samples(void* pSamples, drwav_uint64 sampleCount, drwav_uint32 bytesPerSample, drwav_uint16 format) +{ + switch (format) + { + case DR_WAVE_FORMAT_PCM: + { + drwav__bswap_samples_pcm(pSamples, sampleCount, bytesPerSample); + } break; + + case DR_WAVE_FORMAT_IEEE_FLOAT: + { + drwav__bswap_samples_ieee(pSamples, sampleCount, bytesPerSample); + } break; + + case DR_WAVE_FORMAT_ALAW: + case DR_WAVE_FORMAT_MULAW: + { + drwav__bswap_samples_s16((drwav_int16*)pSamples, sampleCount); + } break; + + case DR_WAVE_FORMAT_ADPCM: + case DR_WAVE_FORMAT_DVI_ADPCM: + default: + { + /* Unsupported format. */ + DRWAV_ASSERT(DRWAV_FALSE); + } break; + } +} + + +DRWAV_PRIVATE void* drwav__malloc_default(size_t sz, void* pUserData) +{ + (void)pUserData; + return DRWAV_MALLOC(sz); +} + +DRWAV_PRIVATE void* drwav__realloc_default(void* p, size_t sz, void* pUserData) +{ + (void)pUserData; + return DRWAV_REALLOC(p, sz); +} + +DRWAV_PRIVATE void drwav__free_default(void* p, void* pUserData) +{ + (void)pUserData; + DRWAV_FREE(p); +} + + +DRWAV_PRIVATE void* drwav__malloc_from_callbacks(size_t sz, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onMalloc != NULL) { + return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData); + } + + /* Try using realloc(). */ + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData); + } + + return NULL; +} + +DRWAV_PRIVATE void* drwav__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData); + } + + /* Try emulating realloc() in terms of malloc()/free(). */ + if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) { + void* p2; + + p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData); + if (p2 == NULL) { + return NULL; + } + + if (p != NULL) { + DRWAV_COPY_MEMORY(p2, p, szOld); + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } + + return p2; + } + + return NULL; +} + +DRWAV_PRIVATE void drwav__free_from_callbacks(void* p, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (p == NULL || pAllocationCallbacks == NULL) { + return; + } + + if (pAllocationCallbacks->onFree != NULL) { + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } +} + + +DRWAV_PRIVATE drwav_allocation_callbacks drwav_copy_allocation_callbacks_or_defaults(const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + /* Copy. */ + return *pAllocationCallbacks; + } else { + /* Defaults. */ + drwav_allocation_callbacks allocationCallbacks; + allocationCallbacks.pUserData = NULL; + allocationCallbacks.onMalloc = drwav__malloc_default; + allocationCallbacks.onRealloc = drwav__realloc_default; + allocationCallbacks.onFree = drwav__free_default; + return allocationCallbacks; + } +} + + +static DRWAV_INLINE drwav_bool32 drwav__is_compressed_format_tag(drwav_uint16 formatTag) +{ + return + formatTag == DR_WAVE_FORMAT_ADPCM || + formatTag == DR_WAVE_FORMAT_DVI_ADPCM; +} + +DRWAV_PRIVATE unsigned int drwav__chunk_padding_size_riff(drwav_uint64 chunkSize) +{ + return (unsigned int)(chunkSize % 2); +} + +DRWAV_PRIVATE unsigned int drwav__chunk_padding_size_w64(drwav_uint64 chunkSize) +{ + return (unsigned int)(chunkSize % 8); +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +DRWAV_PRIVATE drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +DRWAV_PRIVATE drwav_result drwav__read_chunk_header(drwav_read_proc onRead, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_chunk_header* pHeaderOut) +{ + if (container == drwav_container_riff || container == drwav_container_rf64) { + drwav_uint8 sizeInBytes[4]; + + if (onRead(pUserData, pHeaderOut->id.fourcc, 4) != 4) { + return DRWAV_AT_END; + } + + if (onRead(pUserData, sizeInBytes, 4) != 4) { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->sizeInBytes = drwav_bytes_to_u32(sizeInBytes); + pHeaderOut->paddingSize = drwav__chunk_padding_size_riff(pHeaderOut->sizeInBytes); + *pRunningBytesReadOut += 8; + } else { + drwav_uint8 sizeInBytes[8]; + + if (onRead(pUserData, pHeaderOut->id.guid, 16) != 16) { + return DRWAV_AT_END; + } + + if (onRead(pUserData, sizeInBytes, 8) != 8) { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->sizeInBytes = drwav_bytes_to_u64(sizeInBytes) - 24; /* <-- Subtract 24 because w64 includes the size of the header. */ + pHeaderOut->paddingSize = drwav__chunk_padding_size_w64(pHeaderOut->sizeInBytes); + *pRunningBytesReadOut += 24; + } + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE drwav_bool32 drwav__seek_forward(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + drwav_uint64 bytesRemainingToSeek = offset; + while (bytesRemainingToSeek > 0) { + if (bytesRemainingToSeek > 0x7FFFFFFF) { + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek -= 0x7FFFFFFF; + } else { + if (!onSeek(pUserData, (int)bytesRemainingToSeek, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek = 0; + } + } + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE drwav_bool32 drwav__seek_from_start(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + if (offset <= 0x7FFFFFFF) { + return onSeek(pUserData, (int)offset, drwav_seek_origin_start); + } + + /* Larger than 32-bit seek. */ + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_start)) { + return DRWAV_FALSE; + } + offset -= 0x7FFFFFFF; + + for (;;) { + if (offset <= 0x7FFFFFFF) { + return onSeek(pUserData, (int)offset, drwav_seek_origin_current); + } + + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + offset -= 0x7FFFFFFF; + } + + /* Should never get here. */ + /*return DRWAV_TRUE; */ +} + + +DRWAV_PRIVATE drwav_bool32 drwav__read_fmt(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_fmt* fmtOut) +{ + drwav_chunk_header header; + drwav_uint8 fmt[16]; + + if (drwav__read_chunk_header(onRead, pUserData, container, pRunningBytesReadOut, &header) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + + /* Skip non-fmt chunks. */ + while (((container == drwav_container_riff || container == drwav_container_rf64) && !drwav_fourcc_equal(header.id.fourcc, "fmt ")) || (container == drwav_container_w64 && !drwav_guid_equal(header.id.guid, drwavGUID_W64_FMT))) { + if (!drwav__seek_forward(onSeek, header.sizeInBytes + header.paddingSize, pUserData)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += header.sizeInBytes + header.paddingSize; + + /* Try the next header. */ + if (drwav__read_chunk_header(onRead, pUserData, container, pRunningBytesReadOut, &header) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + } + + + /* Validation. */ + if (container == drwav_container_riff || container == drwav_container_rf64) { + if (!drwav_fourcc_equal(header.id.fourcc, "fmt ")) { + return DRWAV_FALSE; + } + } else { + if (!drwav_guid_equal(header.id.guid, drwavGUID_W64_FMT)) { + return DRWAV_FALSE; + } + } + + + if (onRead(pUserData, fmt, sizeof(fmt)) != sizeof(fmt)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += sizeof(fmt); + + fmtOut->formatTag = drwav_bytes_to_u16(fmt + 0); + fmtOut->channels = drwav_bytes_to_u16(fmt + 2); + fmtOut->sampleRate = drwav_bytes_to_u32(fmt + 4); + fmtOut->avgBytesPerSec = drwav_bytes_to_u32(fmt + 8); + fmtOut->blockAlign = drwav_bytes_to_u16(fmt + 12); + fmtOut->bitsPerSample = drwav_bytes_to_u16(fmt + 14); + + fmtOut->extendedSize = 0; + fmtOut->validBitsPerSample = 0; + fmtOut->channelMask = 0; + DRWAV_ZERO_MEMORY(fmtOut->subFormat, sizeof(fmtOut->subFormat)); + + if (header.sizeInBytes > 16) { + drwav_uint8 fmt_cbSize[2]; + int bytesReadSoFar = 0; + + if (onRead(pUserData, fmt_cbSize, sizeof(fmt_cbSize)) != sizeof(fmt_cbSize)) { + return DRWAV_FALSE; /* Expecting more data. */ + } + *pRunningBytesReadOut += sizeof(fmt_cbSize); + + bytesReadSoFar = 18; + + fmtOut->extendedSize = drwav_bytes_to_u16(fmt_cbSize); + if (fmtOut->extendedSize > 0) { + /* Simple validation. */ + if (fmtOut->formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + if (fmtOut->extendedSize != 22) { + return DRWAV_FALSE; + } + } + + if (fmtOut->formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + drwav_uint8 fmtext[22]; + if (onRead(pUserData, fmtext, fmtOut->extendedSize) != fmtOut->extendedSize) { + return DRWAV_FALSE; /* Expecting more data. */ + } + + fmtOut->validBitsPerSample = drwav_bytes_to_u16(fmtext + 0); + fmtOut->channelMask = drwav_bytes_to_u32(fmtext + 2); + drwav_bytes_to_guid(fmtext + 6, fmtOut->subFormat); + } else { + if (!onSeek(pUserData, fmtOut->extendedSize, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + } + *pRunningBytesReadOut += fmtOut->extendedSize; + + bytesReadSoFar += fmtOut->extendedSize; + } + + /* Seek past any leftover bytes. For w64 the leftover will be defined based on the chunk size. */ + if (!onSeek(pUserData, (int)(header.sizeInBytes - bytesReadSoFar), drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += (header.sizeInBytes - bytesReadSoFar); + } + + if (header.paddingSize > 0) { + if (!onSeek(pUserData, header.paddingSize, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += header.paddingSize; + } + + return DRWAV_TRUE; +} + + +DRWAV_PRIVATE size_t drwav__on_read(drwav_read_proc onRead, void* pUserData, void* pBufferOut, size_t bytesToRead, drwav_uint64* pCursor) +{ + size_t bytesRead; + + DRWAV_ASSERT(onRead != NULL); + DRWAV_ASSERT(pCursor != NULL); + + bytesRead = onRead(pUserData, pBufferOut, bytesToRead); + *pCursor += bytesRead; + return bytesRead; +} + +#if 0 +DRWAV_PRIVATE drwav_bool32 drwav__on_seek(drwav_seek_proc onSeek, void* pUserData, int offset, drwav_seek_origin origin, drwav_uint64* pCursor) +{ + DRWAV_ASSERT(onSeek != NULL); + DRWAV_ASSERT(pCursor != NULL); + + if (!onSeek(pUserData, offset, origin)) { + return DRWAV_FALSE; + } + + if (origin == drwav_seek_origin_start) { + *pCursor = offset; + } else { + *pCursor += offset; + } + + return DRWAV_TRUE; +} +#endif + + +#define DRWAV_SMPL_BYTES 36 +#define DRWAV_SMPL_LOOP_BYTES 24 +#define DRWAV_INST_BYTES 7 +#define DRWAV_ACID_BYTES 24 +#define DRWAV_CUE_BYTES 4 +#define DRWAV_BEXT_BYTES 602 +#define DRWAV_BEXT_DESCRIPTION_BYTES 256 +#define DRWAV_BEXT_ORIGINATOR_NAME_BYTES 32 +#define DRWAV_BEXT_ORIGINATOR_REF_BYTES 32 +#define DRWAV_BEXT_RESERVED_BYTES 180 +#define DRWAV_BEXT_UMID_BYTES 64 +#define DRWAV_CUE_POINT_BYTES 24 +#define DRWAV_LIST_LABEL_OR_NOTE_BYTES 4 +#define DRWAV_LIST_LABELLED_TEXT_BYTES 20 + +#define DRWAV_METADATA_ALIGNMENT 8 + +typedef enum +{ + drwav__metadata_parser_stage_count, + drwav__metadata_parser_stage_read +} drwav__metadata_parser_stage; + +typedef struct +{ + drwav_read_proc onRead; + drwav_seek_proc onSeek; + void *pReadSeekUserData; + drwav__metadata_parser_stage stage; + drwav_metadata *pMetadata; + drwav_uint32 metadataCount; + drwav_uint8 *pData; + drwav_uint8 *pDataCursor; + drwav_uint64 metadataCursor; + drwav_uint64 extraCapacity; +} drwav__metadata_parser; + +DRWAV_PRIVATE size_t drwav__metadata_memory_capacity(drwav__metadata_parser* pParser) +{ + drwav_uint64 cap = sizeof(drwav_metadata) * (drwav_uint64)pParser->metadataCount + pParser->extraCapacity; + if (cap > DRWAV_SIZE_MAX) { + return 0; /* Too big. */ + } + + return (size_t)cap; /* Safe cast thanks to the check above. */ +} + +DRWAV_PRIVATE drwav_uint8* drwav__metadata_get_memory(drwav__metadata_parser* pParser, size_t size, size_t align) +{ + drwav_uint8* pResult; + + if (align) { + drwav_uintptr modulo = (drwav_uintptr)pParser->pDataCursor % align; + if (modulo != 0) { + pParser->pDataCursor += align - modulo; + } + } + + pResult = pParser->pDataCursor; + + /* + Getting to the point where this function is called means there should always be memory + available. Out of memory checks should have been done at an earlier stage. + */ + DRWAV_ASSERT((pResult + size) <= (pParser->pData + drwav__metadata_memory_capacity(pParser))); + + pParser->pDataCursor += size; + return pResult; +} + +DRWAV_PRIVATE void drwav__metadata_request_extra_memory_for_stage_2(drwav__metadata_parser* pParser, size_t bytes, size_t align) +{ + size_t extra = bytes + (align ? (align - 1) : 0); + pParser->extraCapacity += extra; +} + +DRWAV_PRIVATE drwav_result drwav__metadata_alloc(drwav__metadata_parser* pParser, drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pParser->extraCapacity != 0 || pParser->metadataCount != 0) { + pAllocationCallbacks->onFree(pParser->pData, pAllocationCallbacks->pUserData); + + pParser->pData = (drwav_uint8*)pAllocationCallbacks->onMalloc(drwav__metadata_memory_capacity(pParser), pAllocationCallbacks->pUserData); + pParser->pDataCursor = pParser->pData; + + if (pParser->pData == NULL) { + return DRWAV_OUT_OF_MEMORY; + } + + /* + We don't need to worry about specifying an alignment here because malloc always returns something + of suitable alignment. This also means than pParser->pMetadata is all that we need to store in order + for us to free when we are done. + */ + pParser->pMetadata = (drwav_metadata*)drwav__metadata_get_memory(pParser, sizeof(drwav_metadata) * pParser->metadataCount, 1); + pParser->metadataCursor = 0; + } + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE size_t drwav__metadata_parser_read(drwav__metadata_parser* pParser, void* pBufferOut, size_t bytesToRead, drwav_uint64* pCursor) +{ + if (pCursor != NULL) { + return drwav__on_read(pParser->onRead, pParser->pReadSeekUserData, pBufferOut, bytesToRead, pCursor); + } else { + return pParser->onRead(pParser->pReadSeekUserData, pBufferOut, bytesToRead); + } +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_smpl_to_metadata_obj(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata* pMetadata) +{ + drwav_uint8 smplHeaderData[DRWAV_SMPL_BYTES]; + drwav_uint64 totalBytesRead = 0; + size_t bytesJustRead = drwav__metadata_parser_read(pParser, smplHeaderData, sizeof(smplHeaderData), &totalBytesRead); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + DRWAV_ASSERT(pChunkHeader != NULL); + + if (bytesJustRead == sizeof(smplHeaderData)) { + drwav_uint32 iSampleLoop; + + pMetadata->type = drwav_metadata_type_smpl; + pMetadata->data.smpl.manufacturerId = drwav_bytes_to_u32(smplHeaderData + 0); + pMetadata->data.smpl.productId = drwav_bytes_to_u32(smplHeaderData + 4); + pMetadata->data.smpl.samplePeriodNanoseconds = drwav_bytes_to_u32(smplHeaderData + 8); + pMetadata->data.smpl.midiUnityNote = drwav_bytes_to_u32(smplHeaderData + 12); + pMetadata->data.smpl.midiPitchFraction = drwav_bytes_to_u32(smplHeaderData + 16); + pMetadata->data.smpl.smpteFormat = drwav_bytes_to_u32(smplHeaderData + 20); + pMetadata->data.smpl.smpteOffset = drwav_bytes_to_u32(smplHeaderData + 24); + pMetadata->data.smpl.sampleLoopCount = drwav_bytes_to_u32(smplHeaderData + 28); + pMetadata->data.smpl.samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(smplHeaderData + 32); + + /* + The loop count needs to be validated against the size of the chunk for safety so we don't + attempt to read over the boundary of the chunk. + */ + if (pMetadata->data.smpl.sampleLoopCount == (pChunkHeader->sizeInBytes - DRWAV_SMPL_BYTES) / DRWAV_SMPL_LOOP_BYTES) { + pMetadata->data.smpl.pLoops = (drwav_smpl_loop*)drwav__metadata_get_memory(pParser, sizeof(drwav_smpl_loop) * pMetadata->data.smpl.sampleLoopCount, DRWAV_METADATA_ALIGNMENT); + + for (iSampleLoop = 0; iSampleLoop < pMetadata->data.smpl.sampleLoopCount; ++iSampleLoop) { + drwav_uint8 smplLoopData[DRWAV_SMPL_LOOP_BYTES]; + bytesJustRead = drwav__metadata_parser_read(pParser, smplLoopData, sizeof(smplLoopData), &totalBytesRead); + + if (bytesJustRead == sizeof(smplLoopData)) { + pMetadata->data.smpl.pLoops[iSampleLoop].cuePointId = drwav_bytes_to_u32(smplLoopData + 0); + pMetadata->data.smpl.pLoops[iSampleLoop].type = drwav_bytes_to_u32(smplLoopData + 4); + pMetadata->data.smpl.pLoops[iSampleLoop].firstSampleByteOffset = drwav_bytes_to_u32(smplLoopData + 8); + pMetadata->data.smpl.pLoops[iSampleLoop].lastSampleByteOffset = drwav_bytes_to_u32(smplLoopData + 12); + pMetadata->data.smpl.pLoops[iSampleLoop].sampleFraction = drwav_bytes_to_u32(smplLoopData + 16); + pMetadata->data.smpl.pLoops[iSampleLoop].playCount = drwav_bytes_to_u32(smplLoopData + 20); + } else { + break; + } + } + + if (pMetadata->data.smpl.samplerSpecificDataSizeInBytes > 0) { + pMetadata->data.smpl.pSamplerSpecificData = drwav__metadata_get_memory(pParser, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, 1); + DRWAV_ASSERT(pMetadata->data.smpl.pSamplerSpecificData != NULL); + + drwav__metadata_parser_read(pParser, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, &totalBytesRead); + } + } + } + + return totalBytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_cue_to_metadata_obj(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata* pMetadata) +{ + drwav_uint8 cueHeaderSectionData[DRWAV_CUE_BYTES]; + drwav_uint64 totalBytesRead = 0; + size_t bytesJustRead = drwav__metadata_parser_read(pParser, cueHeaderSectionData, sizeof(cueHeaderSectionData), &totalBytesRead); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesJustRead == sizeof(cueHeaderSectionData)) { + pMetadata->type = drwav_metadata_type_cue; + pMetadata->data.cue.cuePointCount = drwav_bytes_to_u32(cueHeaderSectionData); + + /* + We need to validate the cue point count against the size of the chunk so we don't read + beyond the chunk. + */ + if (pMetadata->data.cue.cuePointCount == (pChunkHeader->sizeInBytes - DRWAV_CUE_BYTES) / DRWAV_CUE_POINT_BYTES) { + pMetadata->data.cue.pCuePoints = (drwav_cue_point*)drwav__metadata_get_memory(pParser, sizeof(drwav_cue_point) * pMetadata->data.cue.cuePointCount, DRWAV_METADATA_ALIGNMENT); + DRWAV_ASSERT(pMetadata->data.cue.pCuePoints != NULL); + + if (pMetadata->data.cue.cuePointCount > 0) { + drwav_uint32 iCuePoint; + + for (iCuePoint = 0; iCuePoint < pMetadata->data.cue.cuePointCount; ++iCuePoint) { + drwav_uint8 cuePointData[DRWAV_CUE_POINT_BYTES]; + bytesJustRead = drwav__metadata_parser_read(pParser, cuePointData, sizeof(cuePointData), &totalBytesRead); + + if (bytesJustRead == sizeof(cuePointData)) { + pMetadata->data.cue.pCuePoints[iCuePoint].id = drwav_bytes_to_u32(cuePointData + 0); + pMetadata->data.cue.pCuePoints[iCuePoint].playOrderPosition = drwav_bytes_to_u32(cuePointData + 4); + pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[0] = cuePointData[8]; + pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[1] = cuePointData[9]; + pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[2] = cuePointData[10]; + pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[3] = cuePointData[11]; + pMetadata->data.cue.pCuePoints[iCuePoint].chunkStart = drwav_bytes_to_u32(cuePointData + 12); + pMetadata->data.cue.pCuePoints[iCuePoint].blockStart = drwav_bytes_to_u32(cuePointData + 16); + pMetadata->data.cue.pCuePoints[iCuePoint].sampleByteOffset = drwav_bytes_to_u32(cuePointData + 20); + } else { + break; + } + } + } + } + } + + return totalBytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_inst_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata) +{ + drwav_uint8 instData[DRWAV_INST_BYTES]; + drwav_uint64 bytesRead = drwav__metadata_parser_read(pParser, instData, sizeof(instData), NULL); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesRead == sizeof(instData)) { + pMetadata->type = drwav_metadata_type_inst; + pMetadata->data.inst.midiUnityNote = (drwav_int8)instData[0]; + pMetadata->data.inst.fineTuneCents = (drwav_int8)instData[1]; + pMetadata->data.inst.gainDecibels = (drwav_int8)instData[2]; + pMetadata->data.inst.lowNote = (drwav_int8)instData[3]; + pMetadata->data.inst.highNote = (drwav_int8)instData[4]; + pMetadata->data.inst.lowVelocity = (drwav_int8)instData[5]; + pMetadata->data.inst.highVelocity = (drwav_int8)instData[6]; + } + + return bytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_acid_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata) +{ + drwav_uint8 acidData[DRWAV_ACID_BYTES]; + drwav_uint64 bytesRead = drwav__metadata_parser_read(pParser, acidData, sizeof(acidData), NULL); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesRead == sizeof(acidData)) { + pMetadata->type = drwav_metadata_type_acid; + pMetadata->data.acid.flags = drwav_bytes_to_u32(acidData + 0); + pMetadata->data.acid.midiUnityNote = drwav_bytes_to_u16(acidData + 4); + pMetadata->data.acid.reserved1 = drwav_bytes_to_u16(acidData + 6); + pMetadata->data.acid.reserved2 = drwav_bytes_to_f32(acidData + 8); + pMetadata->data.acid.numBeats = drwav_bytes_to_u32(acidData + 12); + pMetadata->data.acid.meterDenominator = drwav_bytes_to_u16(acidData + 16); + pMetadata->data.acid.meterNumerator = drwav_bytes_to_u16(acidData + 18); + pMetadata->data.acid.tempo = drwav_bytes_to_f32(acidData + 20); + } + + return bytesRead; +} + +DRWAV_PRIVATE size_t drwav__strlen(const char* str) +{ + size_t result = 0; + + while (*str++) { + result += 1; + } + + return result; +} + +DRWAV_PRIVATE size_t drwav__strlen_clamped(const char* str, size_t maxToRead) +{ + size_t result = 0; + + while (*str++ && result < maxToRead) { + result += 1; + } + + return result; +} + +DRWAV_PRIVATE char* drwav__metadata_copy_string(drwav__metadata_parser* pParser, const char* str, size_t maxToRead) +{ + size_t len = drwav__strlen_clamped(str, maxToRead); + + if (len) { + char* result = (char*)drwav__metadata_get_memory(pParser, len + 1, 1); + DRWAV_ASSERT(result != NULL); + + DRWAV_COPY_MEMORY(result, str, len); + result[len] = '\0'; + + return result; + } else { + return NULL; + } +} + +typedef struct +{ + const void* pBuffer; + size_t sizeInBytes; + size_t cursor; +} drwav_buffer_reader; + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_init(const void* pBuffer, size_t sizeInBytes, drwav_buffer_reader* pReader) +{ + DRWAV_ASSERT(pBuffer != NULL); + DRWAV_ASSERT(pReader != NULL); + + DRWAV_ZERO_OBJECT(pReader); + + pReader->pBuffer = pBuffer; + pReader->sizeInBytes = sizeInBytes; + pReader->cursor = 0; + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE const void* drwav_buffer_reader_ptr(const drwav_buffer_reader* pReader) +{ + DRWAV_ASSERT(pReader != NULL); + + return drwav_offset_ptr(pReader->pBuffer, pReader->cursor); +} + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_seek(drwav_buffer_reader* pReader, size_t bytesToSeek) +{ + DRWAV_ASSERT(pReader != NULL); + + if (pReader->cursor + bytesToSeek > pReader->sizeInBytes) { + return DRWAV_BAD_SEEK; /* Seeking too far forward. */ + } + + pReader->cursor += bytesToSeek; + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_read(drwav_buffer_reader* pReader, void* pDst, size_t bytesToRead, size_t* pBytesRead) +{ + drwav_result result = DRWAV_SUCCESS; + size_t bytesRemaining; + + DRWAV_ASSERT(pReader != NULL); + + if (pBytesRead != NULL) { + *pBytesRead = 0; + } + + bytesRemaining = (pReader->sizeInBytes - pReader->cursor); + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (pDst == NULL) { + /* Seek. */ + result = drwav_buffer_reader_seek(pReader, bytesToRead); + } else { + /* Read. */ + DRWAV_COPY_MEMORY(pDst, drwav_buffer_reader_ptr(pReader), bytesToRead); + pReader->cursor += bytesToRead; + } + + DRWAV_ASSERT(pReader->cursor <= pReader->sizeInBytes); + + if (result == DRWAV_SUCCESS) { + if (pBytesRead != NULL) { + *pBytesRead = bytesToRead; + } + } + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_read_u16(drwav_buffer_reader* pReader, drwav_uint16* pDst) +{ + drwav_result result; + size_t bytesRead; + drwav_uint8 data[2]; + + DRWAV_ASSERT(pReader != NULL); + DRWAV_ASSERT(pDst != NULL); + + *pDst = 0; /* Safety. */ + + result = drwav_buffer_reader_read(pReader, data, sizeof(*pDst), &bytesRead); + if (result != DRWAV_SUCCESS || bytesRead != sizeof(*pDst)) { + return result; + } + + *pDst = drwav_bytes_to_u16(data); + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_read_u32(drwav_buffer_reader* pReader, drwav_uint32* pDst) +{ + drwav_result result; + size_t bytesRead; + drwav_uint8 data[4]; + + DRWAV_ASSERT(pReader != NULL); + DRWAV_ASSERT(pDst != NULL); + + *pDst = 0; /* Safety. */ + + result = drwav_buffer_reader_read(pReader, data, sizeof(*pDst), &bytesRead); + if (result != DRWAV_SUCCESS || bytesRead != sizeof(*pDst)) { + return result; + } + + *pDst = drwav_bytes_to_u32(data); + + return DRWAV_SUCCESS; +} + + + +DRWAV_PRIVATE drwav_uint64 drwav__read_bext_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize) +{ + drwav_uint8 bextData[DRWAV_BEXT_BYTES]; + size_t bytesRead = drwav__metadata_parser_read(pParser, bextData, sizeof(bextData), NULL); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesRead == sizeof(bextData)) { + drwav_buffer_reader reader; + drwav_uint32 timeReferenceLow; + drwav_uint32 timeReferenceHigh; + size_t extraBytes; + + pMetadata->type = drwav_metadata_type_bext; + + if (drwav_buffer_reader_init(bextData, bytesRead, &reader) == DRWAV_SUCCESS) { + pMetadata->data.bext.pDescription = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_DESCRIPTION_BYTES); + drwav_buffer_reader_seek(&reader, DRWAV_BEXT_DESCRIPTION_BYTES); + + pMetadata->data.bext.pOriginatorName = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_ORIGINATOR_NAME_BYTES); + drwav_buffer_reader_seek(&reader, DRWAV_BEXT_ORIGINATOR_NAME_BYTES); + + pMetadata->data.bext.pOriginatorReference = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_ORIGINATOR_REF_BYTES); + drwav_buffer_reader_seek(&reader, DRWAV_BEXT_ORIGINATOR_REF_BYTES); + + drwav_buffer_reader_read(&reader, pMetadata->data.bext.pOriginationDate, sizeof(pMetadata->data.bext.pOriginationDate), NULL); + drwav_buffer_reader_read(&reader, pMetadata->data.bext.pOriginationTime, sizeof(pMetadata->data.bext.pOriginationTime), NULL); + + drwav_buffer_reader_read_u32(&reader, &timeReferenceLow); + drwav_buffer_reader_read_u32(&reader, &timeReferenceHigh); + pMetadata->data.bext.timeReference = ((drwav_uint64)timeReferenceHigh << 32) + timeReferenceLow; + + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.version); + + pMetadata->data.bext.pUMID = drwav__metadata_get_memory(pParser, DRWAV_BEXT_UMID_BYTES, 1); + drwav_buffer_reader_read(&reader, pMetadata->data.bext.pUMID, DRWAV_BEXT_UMID_BYTES, NULL); + + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.loudnessValue); + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.loudnessRange); + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxTruePeakLevel); + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxMomentaryLoudness); + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxShortTermLoudness); + + DRWAV_ASSERT((drwav_offset_ptr(drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_RESERVED_BYTES)) == (bextData + DRWAV_BEXT_BYTES)); + + extraBytes = (size_t)(chunkSize - DRWAV_BEXT_BYTES); + if (extraBytes > 0) { + pMetadata->data.bext.pCodingHistory = (char*)drwav__metadata_get_memory(pParser, extraBytes + 1, 1); + DRWAV_ASSERT(pMetadata->data.bext.pCodingHistory != NULL); + + bytesRead += drwav__metadata_parser_read(pParser, pMetadata->data.bext.pCodingHistory, extraBytes, NULL); + pMetadata->data.bext.codingHistorySize = (drwav_uint32)drwav__strlen(pMetadata->data.bext.pCodingHistory); + } else { + pMetadata->data.bext.pCodingHistory = NULL; + pMetadata->data.bext.codingHistorySize = 0; + } + } + } + + return bytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_list_label_or_note_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize, drwav_metadata_type type) +{ + drwav_uint8 cueIDBuffer[DRWAV_LIST_LABEL_OR_NOTE_BYTES]; + drwav_uint64 totalBytesRead = 0; + size_t bytesJustRead = drwav__metadata_parser_read(pParser, cueIDBuffer, sizeof(cueIDBuffer), &totalBytesRead); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesJustRead == sizeof(cueIDBuffer)) { + drwav_uint32 sizeIncludingNullTerminator; + + pMetadata->type = type; + pMetadata->data.labelOrNote.cuePointId = drwav_bytes_to_u32(cueIDBuffer); + + sizeIncludingNullTerminator = (drwav_uint32)chunkSize - DRWAV_LIST_LABEL_OR_NOTE_BYTES; + if (sizeIncludingNullTerminator > 0) { + pMetadata->data.labelOrNote.stringLength = sizeIncludingNullTerminator - 1; + pMetadata->data.labelOrNote.pString = (char*)drwav__metadata_get_memory(pParser, sizeIncludingNullTerminator, 1); + DRWAV_ASSERT(pMetadata->data.labelOrNote.pString != NULL); + + drwav__metadata_parser_read(pParser, pMetadata->data.labelOrNote.pString, sizeIncludingNullTerminator, &totalBytesRead); + } else { + pMetadata->data.labelOrNote.stringLength = 0; + pMetadata->data.labelOrNote.pString = NULL; + } + } + + return totalBytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_list_labelled_cue_region_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize) +{ + drwav_uint8 buffer[DRWAV_LIST_LABELLED_TEXT_BYTES]; + drwav_uint64 totalBytesRead = 0; + size_t bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &totalBytesRead); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesJustRead == sizeof(buffer)) { + drwav_uint32 sizeIncludingNullTerminator; + + pMetadata->type = drwav_metadata_type_list_labelled_cue_region; + pMetadata->data.labelledCueRegion.cuePointId = drwav_bytes_to_u32(buffer + 0); + pMetadata->data.labelledCueRegion.sampleLength = drwav_bytes_to_u32(buffer + 4); + pMetadata->data.labelledCueRegion.purposeId[0] = buffer[8]; + pMetadata->data.labelledCueRegion.purposeId[1] = buffer[9]; + pMetadata->data.labelledCueRegion.purposeId[2] = buffer[10]; + pMetadata->data.labelledCueRegion.purposeId[3] = buffer[11]; + pMetadata->data.labelledCueRegion.country = drwav_bytes_to_u16(buffer + 12); + pMetadata->data.labelledCueRegion.language = drwav_bytes_to_u16(buffer + 14); + pMetadata->data.labelledCueRegion.dialect = drwav_bytes_to_u16(buffer + 16); + pMetadata->data.labelledCueRegion.codePage = drwav_bytes_to_u16(buffer + 18); + + sizeIncludingNullTerminator = (drwav_uint32)chunkSize - DRWAV_LIST_LABELLED_TEXT_BYTES; + if (sizeIncludingNullTerminator > 0) { + pMetadata->data.labelledCueRegion.stringLength = sizeIncludingNullTerminator - 1; + pMetadata->data.labelledCueRegion.pString = (char*)drwav__metadata_get_memory(pParser, sizeIncludingNullTerminator, 1); + DRWAV_ASSERT(pMetadata->data.labelledCueRegion.pString != NULL); + + drwav__metadata_parser_read(pParser, pMetadata->data.labelledCueRegion.pString, sizeIncludingNullTerminator, &totalBytesRead); + } else { + pMetadata->data.labelledCueRegion.stringLength = 0; + pMetadata->data.labelledCueRegion.pString = NULL; + } + } + + return totalBytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_info_text_chunk(drwav__metadata_parser* pParser, drwav_uint64 chunkSize, drwav_metadata_type type) +{ + drwav_uint64 bytesRead = 0; + drwav_uint32 stringSizeWithNullTerminator = (drwav_uint32)chunkSize; + + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, stringSizeWithNullTerminator, 1); + } else { + drwav_metadata* pMetadata = &pParser->pMetadata[pParser->metadataCursor]; + pMetadata->type = type; + if (stringSizeWithNullTerminator > 0) { + pMetadata->data.infoText.stringLength = stringSizeWithNullTerminator - 1; + pMetadata->data.infoText.pString = (char*)drwav__metadata_get_memory(pParser, stringSizeWithNullTerminator, 1); + DRWAV_ASSERT(pMetadata->data.infoText.pString != NULL); + + bytesRead = drwav__metadata_parser_read(pParser, pMetadata->data.infoText.pString, (size_t)stringSizeWithNullTerminator, NULL); + if (bytesRead == chunkSize) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } else { + pMetadata->data.infoText.stringLength = 0; + pMetadata->data.infoText.pString = NULL; + pParser->metadataCursor += 1; + } + } + + return bytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_unknown_chunk(drwav__metadata_parser* pParser, const drwav_uint8* pChunkId, drwav_uint64 chunkSize, drwav_metadata_location location) +{ + drwav_uint64 bytesRead = 0; + + if (location == drwav_metadata_location_invalid) { + return 0; + } + + if (drwav_fourcc_equal(pChunkId, "data") || drwav_fourcc_equal(pChunkId, "fmt") || drwav_fourcc_equal(pChunkId, "fact")) { + return 0; + } + + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)chunkSize, 1); + } else { + drwav_metadata* pMetadata = &pParser->pMetadata[pParser->metadataCursor]; + pMetadata->type = drwav_metadata_type_unknown; + pMetadata->data.unknown.chunkLocation = location; + pMetadata->data.unknown.id[0] = pChunkId[0]; + pMetadata->data.unknown.id[1] = pChunkId[1]; + pMetadata->data.unknown.id[2] = pChunkId[2]; + pMetadata->data.unknown.id[3] = pChunkId[3]; + pMetadata->data.unknown.dataSizeInBytes = (drwav_uint32)chunkSize; + pMetadata->data.unknown.pData = (drwav_uint8 *)drwav__metadata_get_memory(pParser, (size_t)chunkSize, 1); + DRWAV_ASSERT(pMetadata->data.unknown.pData != NULL); + + bytesRead = drwav__metadata_parser_read(pParser, pMetadata->data.unknown.pData, pMetadata->data.unknown.dataSizeInBytes, NULL); + if (bytesRead == pMetadata->data.unknown.dataSizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to read. */ + } + } + + return bytesRead; +} + +DRWAV_PRIVATE drwav_bool32 drwav__chunk_matches(drwav_metadata_type allowedMetadataTypes, const drwav_uint8* pChunkID, drwav_metadata_type type, const char* pID) +{ + return (allowedMetadataTypes & type) && drwav_fourcc_equal(pChunkID, pID); +} + +DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata_type allowedMetadataTypes) +{ + const drwav_uint8 *pChunkID = pChunkHeader->id.fourcc; + drwav_uint64 bytesRead = 0; + + if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_smpl, "smpl")) { + if (pChunkHeader->sizeInBytes >= DRWAV_SMPL_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + drwav_uint8 buffer[4]; + size_t bytesJustRead; + + if (!pParser->onSeek(pParser->pReadSeekUserData, 28, drwav_seek_origin_current)) { + return bytesRead; + } + bytesRead += 28; + + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead); + if (bytesJustRead == sizeof(buffer)) { + drwav_uint32 loopCount = drwav_bytes_to_u32(buffer); + drwav_uint64 calculatedLoopCount; + + /* The loop count must be validated against the size of the chunk. */ + calculatedLoopCount = (pChunkHeader->sizeInBytes - DRWAV_SMPL_BYTES) / DRWAV_SMPL_LOOP_BYTES; + if (calculatedLoopCount == loopCount) { + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead); + if (bytesJustRead == sizeof(buffer)) { + drwav_uint32 samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(buffer); + + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_smpl_loop) * loopCount, DRWAV_METADATA_ALIGNMENT); + drwav__metadata_request_extra_memory_for_stage_2(pParser, samplerSpecificDataSizeInBytes, 1); + } + } else { + /* Loop count in header does not match the size of the chunk. */ + } + } + } else { + bytesRead = drwav__read_smpl_to_metadata_obj(pParser, pChunkHeader, &pParser->pMetadata[pParser->metadataCursor]); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_inst, "inst")) { + if (pChunkHeader->sizeInBytes == DRWAV_INST_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + } else { + bytesRead = drwav__read_inst_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor]); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_acid, "acid")) { + if (pChunkHeader->sizeInBytes == DRWAV_ACID_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + } else { + bytesRead = drwav__read_acid_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor]); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_cue, "cue ")) { + if (pChunkHeader->sizeInBytes >= DRWAV_CUE_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + size_t cueCount; + + pParser->metadataCount += 1; + cueCount = (size_t)(pChunkHeader->sizeInBytes - DRWAV_CUE_BYTES) / DRWAV_CUE_POINT_BYTES; + drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_cue_point) * cueCount, DRWAV_METADATA_ALIGNMENT); + } else { + bytesRead = drwav__read_cue_to_metadata_obj(pParser, pChunkHeader, &pParser->pMetadata[pParser->metadataCursor]); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_bext, "bext")) { + if (pChunkHeader->sizeInBytes >= DRWAV_BEXT_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + /* The description field is the largest one in a bext chunk, so that is the max size of this temporary buffer. */ + char buffer[DRWAV_BEXT_DESCRIPTION_BYTES + 1]; + size_t allocSizeNeeded = DRWAV_BEXT_UMID_BYTES; /* We know we will need SMPTE umid size. */ + size_t bytesJustRead; + + buffer[DRWAV_BEXT_DESCRIPTION_BYTES] = '\0'; + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_DESCRIPTION_BYTES, &bytesRead); + if (bytesJustRead != DRWAV_BEXT_DESCRIPTION_BYTES) { + return bytesRead; + } + allocSizeNeeded += drwav__strlen(buffer) + 1; + + buffer[DRWAV_BEXT_ORIGINATOR_NAME_BYTES] = '\0'; + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_ORIGINATOR_NAME_BYTES, &bytesRead); + if (bytesJustRead != DRWAV_BEXT_ORIGINATOR_NAME_BYTES) { + return bytesRead; + } + allocSizeNeeded += drwav__strlen(buffer) + 1; + + buffer[DRWAV_BEXT_ORIGINATOR_REF_BYTES] = '\0'; + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_ORIGINATOR_REF_BYTES, &bytesRead); + if (bytesJustRead != DRWAV_BEXT_ORIGINATOR_REF_BYTES) { + return bytesRead; + } + allocSizeNeeded += drwav__strlen(buffer) + 1; + allocSizeNeeded += (size_t)pChunkHeader->sizeInBytes - DRWAV_BEXT_BYTES; /* Coding history. */ + + drwav__metadata_request_extra_memory_for_stage_2(pParser, allocSizeNeeded, 1); + + pParser->metadataCount += 1; + } else { + bytesRead = drwav__read_bext_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], pChunkHeader->sizeInBytes); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav_fourcc_equal(pChunkID, "LIST") || drwav_fourcc_equal(pChunkID, "list")) { + drwav_metadata_location listType = drwav_metadata_location_invalid; + while (bytesRead < pChunkHeader->sizeInBytes) { + drwav_uint8 subchunkId[4]; + drwav_uint8 subchunkSizeBuffer[4]; + drwav_uint64 subchunkDataSize; + drwav_uint64 subchunkBytesRead = 0; + drwav_uint64 bytesJustRead = drwav__metadata_parser_read(pParser, subchunkId, sizeof(subchunkId), &bytesRead); + if (bytesJustRead != sizeof(subchunkId)) { + break; + } + + /* + The first thing in a list chunk should be "adtl" or "INFO". + + - adtl means this list is a Associated Data List Chunk and will contain labels, notes + or labelled cue regions. + - INFO means this list is an Info List Chunk containing info text chunks such as IPRD + which would specifies the album of this wav file. + + No data follows the adtl or INFO id so we just make note of what type this list is and + continue. + */ + if (drwav_fourcc_equal(subchunkId, "adtl")) { + listType = drwav_metadata_location_inside_adtl_list; + continue; + } else if (drwav_fourcc_equal(subchunkId, "INFO")) { + listType = drwav_metadata_location_inside_info_list; + continue; + } + + bytesJustRead = drwav__metadata_parser_read(pParser, subchunkSizeBuffer, sizeof(subchunkSizeBuffer), &bytesRead); + if (bytesJustRead != sizeof(subchunkSizeBuffer)) { + break; + } + subchunkDataSize = drwav_bytes_to_u32(subchunkSizeBuffer); + + if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_label, "labl") || drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_note, "note")) { + if (subchunkDataSize >= DRWAV_LIST_LABEL_OR_NOTE_BYTES) { + drwav_uint64 stringSizeWithNullTerm = subchunkDataSize - DRWAV_LIST_LABEL_OR_NOTE_BYTES; + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)stringSizeWithNullTerm, 1); + } else { + subchunkBytesRead = drwav__read_list_label_or_note_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], subchunkDataSize, drwav_fourcc_equal(subchunkId, "labl") ? drwav_metadata_type_list_label : drwav_metadata_type_list_note); + if (subchunkBytesRead == subchunkDataSize) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_labelled_cue_region, "ltxt")) { + if (subchunkDataSize >= DRWAV_LIST_LABELLED_TEXT_BYTES) { + drwav_uint64 stringSizeWithNullTerminator = subchunkDataSize - DRWAV_LIST_LABELLED_TEXT_BYTES; + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)stringSizeWithNullTerminator, 1); + } else { + subchunkBytesRead = drwav__read_list_labelled_cue_region_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], subchunkDataSize); + if (subchunkBytesRead == subchunkDataSize) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_software, "ISFT")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_software); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_copyright, "ICOP")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_copyright); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_title, "INAM")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_title); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_artist, "IART")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_artist); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_comment, "ICMT")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_comment); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_date, "ICRD")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_date); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_genre, "IGNR")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_genre); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_album, "IPRD")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_album); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_tracknumber, "ITRK")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_tracknumber); + } else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) { + subchunkBytesRead = drwav__metadata_process_unknown_chunk(pParser, subchunkId, subchunkDataSize, listType); + } + + bytesRead += subchunkBytesRead; + DRWAV_ASSERT(subchunkBytesRead <= subchunkDataSize); + + if (subchunkBytesRead < subchunkDataSize) { + drwav_uint64 bytesToSeek = subchunkDataSize - subchunkBytesRead; + + if (!pParser->onSeek(pParser->pReadSeekUserData, (int)bytesToSeek, drwav_seek_origin_current)) { + break; + } + bytesRead += bytesToSeek; + } + + if ((subchunkDataSize % 2) == 1) { + if (!pParser->onSeek(pParser->pReadSeekUserData, 1, drwav_seek_origin_current)) { + break; + } + bytesRead += 1; + } + } + } else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) { + bytesRead = drwav__metadata_process_unknown_chunk(pParser, pChunkID, pChunkHeader->sizeInBytes, drwav_metadata_location_top_level); + } + + return bytesRead; +} + + +DRWAV_PRIVATE drwav_uint32 drwav_get_bytes_per_pcm_frame(drwav* pWav) +{ + drwav_uint32 bytesPerFrame; + + /* + The bytes per frame is a bit ambiguous. It can be either be based on the bits per sample, or the block align. The way I'm doing it here + is that if the bits per sample is a multiple of 8, use floor(bitsPerSample*channels/8), otherwise fall back to the block align. + */ + if ((pWav->bitsPerSample & 0x7) == 0) { + /* Bits per sample is a multiple of 8. */ + bytesPerFrame = (pWav->bitsPerSample * pWav->fmt.channels) >> 3; + } else { + bytesPerFrame = pWav->fmt.blockAlign; + } + + /* Validation for known formats. a-law and mu-law should be 1 byte per channel. If it's not, it's not decodable. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW || pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + if (bytesPerFrame != pWav->fmt.channels) { + return 0; /* Invalid file. */ + } + } + + return bytesPerFrame; +} + +DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT) +{ + if (pFMT == NULL) { + return 0; + } + + if (pFMT->formatTag != DR_WAVE_FORMAT_EXTENSIBLE) { + return pFMT->formatTag; + } else { + return drwav_bytes_to_u16(pFMT->subFormat); /* Only the first two bytes are required. */ + } +} + +DRWAV_PRIVATE drwav_bool32 drwav_preinit(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pWav == NULL || onRead == NULL || onSeek == NULL) { + return DRWAV_FALSE; + } + + DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav)); + pWav->onRead = onRead; + pWav->onSeek = onSeek; + pWav->pUserData = pReadSeekUserData; + pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks); + + if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) { + return DRWAV_FALSE; /* Invalid allocation callbacks. */ + } + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE drwav_bool32 drwav_init__internal(drwav* pWav, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags) +{ + /* This function assumes drwav_preinit() has been called beforehand. */ + + drwav_uint64 cursor; /* <-- Keeps track of the byte position so we can seek to specific locations. */ + drwav_bool32 sequential; + drwav_uint8 riff[4]; + drwav_fmt fmt; + unsigned short translatedFormatTag; + drwav_bool32 foundDataChunk; + drwav_uint64 dataChunkSize = 0; /* <-- Important! Don't explicitly set this to 0 anywhere else. Calculation of the size of the data chunk is performed in different paths depending on the container. */ + drwav_uint64 sampleCountFromFactChunk = 0; /* Same as dataChunkSize - make sure this is the only place this is initialized to 0. */ + drwav_uint64 chunkSize; + drwav__metadata_parser metadataParser; + + cursor = 0; + sequential = (flags & DRWAV_SEQUENTIAL) != 0; + + /* The first 4 bytes should be the RIFF identifier. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, riff, sizeof(riff), &cursor) != sizeof(riff)) { + return DRWAV_FALSE; + } + + /* + The first 4 bytes can be used to identify the container. For RIFF files it will start with "RIFF" and for + w64 it will start with "riff". + */ + if (drwav_fourcc_equal(riff, "RIFF")) { + pWav->container = drwav_container_riff; + } else if (drwav_fourcc_equal(riff, "riff")) { + int i; + drwav_uint8 riff2[12]; + + pWav->container = drwav_container_w64; + + /* Check the rest of the GUID for validity. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, riff2, sizeof(riff2), &cursor) != sizeof(riff2)) { + return DRWAV_FALSE; + } + + for (i = 0; i < 12; ++i) { + if (riff2[i] != drwavGUID_W64_RIFF[i+4]) { + return DRWAV_FALSE; + } + } + } else if (drwav_fourcc_equal(riff, "RF64")) { + pWav->container = drwav_container_rf64; + } else { + return DRWAV_FALSE; /* Unknown or unsupported container. */ + } + + + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) { + drwav_uint8 chunkSizeBytes[4]; + drwav_uint8 wave[4]; + + /* RIFF/WAVE */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (pWav->container == drwav_container_riff) { + if (drwav_bytes_to_u32(chunkSizeBytes) < 36) { + return DRWAV_FALSE; /* Chunk size should always be at least 36 bytes. */ + } + } else { + if (drwav_bytes_to_u32(chunkSizeBytes) != 0xFFFFFFFF) { + return DRWAV_FALSE; /* Chunk size should always be set to -1/0xFFFFFFFF for RF64. The actual size is retrieved later. */ + } + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav_fourcc_equal(wave, "WAVE")) { + return DRWAV_FALSE; /* Expecting "WAVE". */ + } + } else { + drwav_uint8 chunkSizeBytes[8]; + drwav_uint8 wave[16]; + + /* W64 */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (drwav_bytes_to_u64(chunkSizeBytes) < 80) { + return DRWAV_FALSE; + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav_guid_equal(wave, drwavGUID_W64_WAVE)) { + return DRWAV_FALSE; + } + } + + + /* For RF64, the "ds64" chunk must come next, before the "fmt " chunk. */ + if (pWav->container == drwav_container_rf64) { + drwav_uint8 sizeBytes[8]; + drwav_uint64 bytesRemainingInChunk; + drwav_chunk_header header; + drwav_result result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header); + if (result != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + if (!drwav_fourcc_equal(header.id.fourcc, "ds64")) { + return DRWAV_FALSE; /* Expecting "ds64". */ + } + + bytesRemainingInChunk = header.sizeInBytes + header.paddingSize; + + /* We don't care about the size of the RIFF chunk - skip it. */ + if (!drwav__seek_forward(pWav->onSeek, 8, pWav->pUserData)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + cursor += 8; + + + /* Next 8 bytes is the size of the "data" chunk. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, sizeBytes, sizeof(sizeBytes), &cursor) != sizeof(sizeBytes)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + dataChunkSize = drwav_bytes_to_u64(sizeBytes); + + + /* Next 8 bytes is the same count which we would usually derived from the FACT chunk if it was available. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, sizeBytes, sizeof(sizeBytes), &cursor) != sizeof(sizeBytes)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + sampleCountFromFactChunk = drwav_bytes_to_u64(sizeBytes); + + + /* Skip over everything else. */ + if (!drwav__seek_forward(pWav->onSeek, bytesRemainingInChunk, pWav->pUserData)) { + return DRWAV_FALSE; + } + cursor += bytesRemainingInChunk; + } + + + /* The next bytes should be the "fmt " chunk. */ + if (!drwav__read_fmt(pWav->onRead, pWav->onSeek, pWav->pUserData, pWav->container, &cursor, &fmt)) { + return DRWAV_FALSE; /* Failed to read the "fmt " chunk. */ + } + + /* Basic validation. */ + if ((fmt.sampleRate == 0 || fmt.sampleRate > DRWAV_MAX_SAMPLE_RATE) || + (fmt.channels == 0 || fmt.channels > DRWAV_MAX_CHANNELS) || + (fmt.bitsPerSample == 0 || fmt.bitsPerSample > DRWAV_MAX_BITS_PER_SAMPLE) || + fmt.blockAlign == 0) { + return DRWAV_FALSE; /* Probably an invalid WAV file. */ + } + + + /* Translate the internal format. */ + translatedFormatTag = fmt.formatTag; + if (translatedFormatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + translatedFormatTag = drwav_bytes_to_u16(fmt.subFormat + 0); + } + + DRWAV_ZERO_MEMORY(&metadataParser, sizeof(metadataParser)); + + /* Not tested on W64. */ + if (!sequential && pWav->allowedMetadataTypes != drwav_metadata_type_none && (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64)) { + drwav_uint64 cursorForMetadata = cursor; + + metadataParser.onRead = pWav->onRead; + metadataParser.onSeek = pWav->onSeek; + metadataParser.pReadSeekUserData = pWav->pUserData; + metadataParser.stage = drwav__metadata_parser_stage_count; + + for (;;) { + drwav_result result; + drwav_uint64 bytesRead; + drwav_uint64 remainingBytes; + drwav_chunk_header header; + + result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursorForMetadata, &header); + if (result != DRWAV_SUCCESS) { + break; + } + + bytesRead = drwav__metadata_process_chunk(&metadataParser, &header, pWav->allowedMetadataTypes); + DRWAV_ASSERT(bytesRead <= header.sizeInBytes); + + remainingBytes = header.sizeInBytes - bytesRead + header.paddingSize; + if (!drwav__seek_forward(pWav->onSeek, remainingBytes, pWav->pUserData)) { + break; + } + cursorForMetadata += remainingBytes; + } + + if (!drwav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData)) { + return DRWAV_FALSE; + } + + drwav__metadata_alloc(&metadataParser, &pWav->allocationCallbacks); + metadataParser.stage = drwav__metadata_parser_stage_read; + } + + /* + We need to enumerate over each chunk for two reasons: + 1) The "data" chunk may not be the next one + 2) We may want to report each chunk back to the client + + In order to correctly report each chunk back to the client we will need to keep looping until the end of the file. + */ + foundDataChunk = DRWAV_FALSE; + + /* The next chunk we care about is the "data" chunk. This is not necessarily the next chunk so we'll need to loop. */ + for (;;) { + drwav_chunk_header header; + drwav_result result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header); + if (result != DRWAV_SUCCESS) { + if (!foundDataChunk) { + return DRWAV_FALSE; + } else { + break; /* Probably at the end of the file. Get out of the loop. */ + } + } + + /* Tell the client about this chunk. */ + if (!sequential && onChunk != NULL) { + drwav_uint64 callbackBytesRead = onChunk(pChunkUserData, pWav->onRead, pWav->onSeek, pWav->pUserData, &header, pWav->container, &fmt); + + /* + dr_wav may need to read the contents of the chunk, so we now need to seek back to the position before + we called the callback. + */ + if (callbackBytesRead > 0) { + if (!drwav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData)) { + return DRWAV_FALSE; + } + } + } + + if (!sequential && pWav->allowedMetadataTypes != drwav_metadata_type_none && (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64)) { + drwav_uint64 bytesRead = drwav__metadata_process_chunk(&metadataParser, &header, pWav->allowedMetadataTypes); + + if (bytesRead > 0) { + if (!drwav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData)) { + return DRWAV_FALSE; + } + } + } + + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + + chunkSize = header.sizeInBytes; + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) { + if (drwav_fourcc_equal(header.id.fourcc, "data")) { + foundDataChunk = DRWAV_TRUE; + if (pWav->container != drwav_container_rf64) { /* The data chunk size for RF64 will always be set to 0xFFFFFFFF here. It was set to it's true value earlier. */ + dataChunkSize = chunkSize; + } + } + } else { + if (drwav_guid_equal(header.id.guid, drwavGUID_W64_DATA)) { + foundDataChunk = DRWAV_TRUE; + dataChunkSize = chunkSize; + } + } + + /* + If at this point we have found the data chunk and we're running in sequential mode, we need to break out of this loop. The reason for + this is that we would otherwise require a backwards seek which sequential mode forbids. + */ + if (foundDataChunk && sequential) { + break; + } + + /* Optional. Get the total sample count from the FACT chunk. This is useful for compressed formats. */ + if (pWav->container == drwav_container_riff) { + if (drwav_fourcc_equal(header.id.fourcc, "fact")) { + drwav_uint32 sampleCount; + if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCount, 4, &cursor) != 4) { + return DRWAV_FALSE; + } + chunkSize -= 4; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + + /* + The sample count in the "fact" chunk is either unreliable, or I'm not understanding it properly. For now I am only enabling this + for Microsoft ADPCM formats. + */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + sampleCountFromFactChunk = sampleCount; + } else { + sampleCountFromFactChunk = 0; + } + } + } else if (pWav->container == drwav_container_w64) { + if (drwav_guid_equal(header.id.guid, drwavGUID_W64_FACT)) { + if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCountFromFactChunk, 8, &cursor) != 8) { + return DRWAV_FALSE; + } + chunkSize -= 8; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + } + } else if (pWav->container == drwav_container_rf64) { + /* We retrieved the sample count from the ds64 chunk earlier so no need to do that here. */ + } + + /* Make sure we seek past the padding. */ + chunkSize += header.paddingSize; + if (!drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData)) { + break; + } + cursor += chunkSize; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + } + + pWav->pMetadata = metadataParser.pMetadata; + pWav->metadataCount = metadataParser.metadataCount; + + /* If we haven't found a data chunk, return an error. */ + if (!foundDataChunk) { + return DRWAV_FALSE; + } + + /* We may have moved passed the data chunk. If so we need to move back. If running in sequential mode we can assume we are already sitting on the data chunk. */ + if (!sequential) { + if (!drwav__seek_from_start(pWav->onSeek, pWav->dataChunkDataPos, pWav->pUserData)) { + return DRWAV_FALSE; + } + cursor = pWav->dataChunkDataPos; + } + + + /* At this point we should be sitting on the first byte of the raw audio data. */ + + pWav->fmt = fmt; + pWav->sampleRate = fmt.sampleRate; + pWav->channels = fmt.channels; + pWav->bitsPerSample = fmt.bitsPerSample; + pWav->bytesRemaining = dataChunkSize; + pWav->translatedFormatTag = translatedFormatTag; + pWav->dataChunkDataSize = dataChunkSize; + + if (sampleCountFromFactChunk != 0) { + pWav->totalPCMFrameCount = sampleCountFromFactChunk; + } else { + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return DRWAV_FALSE; /* Invalid file. */ + } + + pWav->totalPCMFrameCount = dataChunkSize / bytesPerFrame; + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 totalBlockHeaderSizeInBytes; + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + + /* Make sure any trailing partial block is accounted for. */ + if ((blockCount * fmt.blockAlign) < dataChunkSize) { + blockCount += 1; + } + + /* We decode two samples per byte. There will be blockCount headers in the data chunk. This is enough to know how to calculate the total PCM frame count. */ + totalBlockHeaderSizeInBytes = blockCount * (6*fmt.channels); + pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels; + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 totalBlockHeaderSizeInBytes; + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + + /* Make sure any trailing partial block is accounted for. */ + if ((blockCount * fmt.blockAlign) < dataChunkSize) { + blockCount += 1; + } + + /* We decode two samples per byte. There will be blockCount headers in the data chunk. This is enough to know how to calculate the total PCM frame count. */ + totalBlockHeaderSizeInBytes = blockCount * (4*fmt.channels); + pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels; + + /* The header includes a decoded sample for each channel which acts as the initial predictor sample. */ + pWav->totalPCMFrameCount += blockCount; + } + } + + /* Some formats only support a certain number of channels. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + if (pWav->channels > 2) { + return DRWAV_FALSE; + } + } + + /* The number of bytes per frame must be known. If not, it's an invalid file and not decodable. */ + if (drwav_get_bytes_per_pcm_frame(pWav) == 0) { + return DRWAV_FALSE; + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + /* + I use libsndfile as a benchmark for testing, however in the version I'm using (from the Windows installer on the libsndfile website), + it appears the total sample count libsndfile uses for MS-ADPCM is incorrect. It would seem they are computing the total sample count + from the number of blocks, however this results in the inclusion of extra silent samples at the end of the last block. The correct + way to know the total sample count is to inspect the "fact" chunk, which should always be present for compressed formats, and should + always include the sample count. This little block of code below is only used to emulate the libsndfile logic so I can properly run my + correctness tests against libsndfile, and is disabled by default. + */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2)) / fmt.channels; /* x2 because two samples per byte. */ + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels)) / fmt.channels; + } +#endif + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit(pWav, onRead, onSeek, pReadSeekUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init__internal(pWav, onChunk, pChunkUserData, flags); +} + +DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit(pWav, onRead, onSeek, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->allowedMetadataTypes = drwav_metadata_type_all_including_unknown; /* <-- Needs to be set to tell drwav_init_ex() that we need to process metadata. */ + return drwav_init__internal(pWav, NULL, NULL, flags); +} + +DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav) +{ + drwav_metadata *result = pWav->pMetadata; + + pWav->pMetadata = NULL; + pWav->metadataCount = 0; + + return result; +} + + +DRWAV_PRIVATE size_t drwav__write(drwav* pWav, const void* pData, size_t dataSize) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + /* Generic write. Assumes no byte reordering required. */ + return pWav->onWrite(pWav->pUserData, pData, dataSize); +} + +DRWAV_PRIVATE size_t drwav__write_byte(drwav* pWav, drwav_uint8 byte) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + return pWav->onWrite(pWav->pUserData, &byte, 1); +} + +DRWAV_PRIVATE size_t drwav__write_u16ne_to_le(drwav* pWav, drwav_uint16 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap16(value); + } + + return drwav__write(pWav, &value, 2); +} + +DRWAV_PRIVATE size_t drwav__write_u32ne_to_le(drwav* pWav, drwav_uint32 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap32(value); + } + + return drwav__write(pWav, &value, 4); +} + +DRWAV_PRIVATE size_t drwav__write_u64ne_to_le(drwav* pWav, drwav_uint64 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap64(value); + } + + return drwav__write(pWav, &value, 8); +} + +DRWAV_PRIVATE size_t drwav__write_f32ne_to_le(drwav* pWav, float value) +{ + union { + drwav_uint32 u32; + float f32; + } u; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + u.f32 = value; + + if (!drwav__is_little_endian()) { + u.u32 = drwav__bswap32(u.u32); + } + + return drwav__write(pWav, &u.u32, 4); +} + +DRWAV_PRIVATE size_t drwav__write_or_count(drwav* pWav, const void* pData, size_t dataSize) +{ + if (pWav == NULL) { + return dataSize; + } + + return drwav__write(pWav, pData, dataSize); +} + +DRWAV_PRIVATE size_t drwav__write_or_count_byte(drwav* pWav, drwav_uint8 byte) +{ + if (pWav == NULL) { + return 1; + } + + return drwav__write_byte(pWav, byte); +} + +DRWAV_PRIVATE size_t drwav__write_or_count_u16ne_to_le(drwav* pWav, drwav_uint16 value) +{ + if (pWav == NULL) { + return 2; + } + + return drwav__write_u16ne_to_le(pWav, value); +} + +DRWAV_PRIVATE size_t drwav__write_or_count_u32ne_to_le(drwav* pWav, drwav_uint32 value) +{ + if (pWav == NULL) { + return 4; + } + + return drwav__write_u32ne_to_le(pWav, value); +} + +#if 0 /* Unused for now. */ +DRWAV_PRIVATE size_t drwav__write_or_count_u64ne_to_le(drwav* pWav, drwav_uint64 value) +{ + if (pWav == NULL) { + return 8; + } + + return drwav__write_u64ne_to_le(pWav, value); +} +#endif + +DRWAV_PRIVATE size_t drwav__write_or_count_f32ne_to_le(drwav* pWav, float value) +{ + if (pWav == NULL) { + return 4; + } + + return drwav__write_f32ne_to_le(pWav, value); +} + +DRWAV_PRIVATE size_t drwav__write_or_count_string_to_fixed_size_buf(drwav* pWav, char* str, size_t bufFixedSize) +{ + size_t len; + + if (pWav == NULL) { + return bufFixedSize; + } + + len = drwav__strlen_clamped(str, bufFixedSize); + drwav__write_or_count(pWav, str, len); + + if (len < bufFixedSize) { + size_t i; + for (i = 0; i < bufFixedSize - len; ++i) { + drwav__write_byte(pWav, 0); + } + } + + return bufFixedSize; +} + + +/* pWav can be NULL meaning just count the bytes that would be written. */ +DRWAV_PRIVATE size_t drwav__write_or_count_metadata(drwav* pWav, drwav_metadata* pMetadatas, drwav_uint32 metadataCount) +{ + size_t bytesWritten = 0; + drwav_bool32 hasListAdtl = DRWAV_FALSE; + drwav_bool32 hasListInfo = DRWAV_FALSE; + drwav_uint32 iMetadata; + + if (pMetadatas == NULL || metadataCount == 0) { + return 0; + } + + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + drwav_uint32 chunkSize = 0; + + if ((pMetadata->type & drwav_metadata_type_list_all_info_strings) || (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list)) { + hasListInfo = DRWAV_TRUE; + } + + if ((pMetadata->type & drwav_metadata_type_list_all_adtl) || (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list)) { + hasListAdtl = DRWAV_TRUE; + } + + switch (pMetadata->type) { + case drwav_metadata_type_smpl: + { + drwav_uint32 iLoop; + + chunkSize = DRWAV_SMPL_BYTES + DRWAV_SMPL_LOOP_BYTES * pMetadata->data.smpl.sampleLoopCount + pMetadata->data.smpl.samplerSpecificDataSizeInBytes; + + bytesWritten += drwav__write_or_count(pWav, "smpl", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.manufacturerId); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.productId); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.samplePeriodNanoseconds); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.midiUnityNote); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.midiPitchFraction); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.smpteFormat); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.smpteOffset); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.sampleLoopCount); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.samplerSpecificDataSizeInBytes); + + for (iLoop = 0; iLoop < pMetadata->data.smpl.sampleLoopCount; ++iLoop) { + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].cuePointId); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].type); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].firstSampleByteOffset); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].lastSampleByteOffset); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].sampleFraction); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].playCount); + } + + if (pMetadata->data.smpl.samplerSpecificDataSizeInBytes > 0) { + bytesWritten += drwav__write(pWav, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes); + } + } break; + + case drwav_metadata_type_inst: + { + chunkSize = DRWAV_INST_BYTES; + + bytesWritten += drwav__write_or_count(pWav, "inst", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.midiUnityNote, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.fineTuneCents, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.gainDecibels, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.lowNote, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.highNote, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.lowVelocity, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.highVelocity, 1); + } break; + + case drwav_metadata_type_cue: + { + drwav_uint32 iCuePoint; + + chunkSize = DRWAV_CUE_BYTES + DRWAV_CUE_POINT_BYTES * pMetadata->data.cue.cuePointCount; + + bytesWritten += drwav__write_or_count(pWav, "cue ", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.cuePointCount); + for (iCuePoint = 0; iCuePoint < pMetadata->data.cue.cuePointCount; ++iCuePoint) { + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].id); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].playOrderPosition); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].chunkStart); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].blockStart); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].sampleByteOffset); + } + } break; + + case drwav_metadata_type_acid: + { + chunkSize = DRWAV_ACID_BYTES; + + bytesWritten += drwav__write_or_count(pWav, "acid", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.acid.flags); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.midiUnityNote); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.reserved1); + bytesWritten += drwav__write_or_count_f32ne_to_le(pWav, pMetadata->data.acid.reserved2); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.acid.numBeats); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.meterDenominator); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.meterNumerator); + bytesWritten += drwav__write_or_count_f32ne_to_le(pWav, pMetadata->data.acid.tempo); + } break; + + case drwav_metadata_type_bext: + { + char reservedBuf[DRWAV_BEXT_RESERVED_BYTES]; + drwav_uint32 timeReferenceLow; + drwav_uint32 timeReferenceHigh; + + chunkSize = DRWAV_BEXT_BYTES + pMetadata->data.bext.codingHistorySize; + + bytesWritten += drwav__write_or_count(pWav, "bext", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + + bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pDescription, DRWAV_BEXT_DESCRIPTION_BYTES); + bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pOriginatorName, DRWAV_BEXT_ORIGINATOR_NAME_BYTES); + bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pOriginatorReference, DRWAV_BEXT_ORIGINATOR_REF_BYTES); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pOriginationDate, sizeof(pMetadata->data.bext.pOriginationDate)); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pOriginationTime, sizeof(pMetadata->data.bext.pOriginationTime)); + + timeReferenceLow = (drwav_uint32)(pMetadata->data.bext.timeReference & 0xFFFFFFFF); + timeReferenceHigh = (drwav_uint32)(pMetadata->data.bext.timeReference >> 32); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, timeReferenceLow); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, timeReferenceHigh); + + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.version); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pUMID, DRWAV_BEXT_UMID_BYTES); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.loudnessValue); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.loudnessRange); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxTruePeakLevel); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxMomentaryLoudness); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxShortTermLoudness); + + DRWAV_ZERO_MEMORY(reservedBuf, sizeof(reservedBuf)); + bytesWritten += drwav__write_or_count(pWav, reservedBuf, sizeof(reservedBuf)); + + if (pMetadata->data.bext.codingHistorySize > 0) { + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pCodingHistory, pMetadata->data.bext.codingHistorySize); + } + } break; + + case drwav_metadata_type_unknown: + { + if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_top_level) { + chunkSize = pMetadata->data.unknown.dataSizeInBytes; + + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, pMetadata->data.unknown.dataSizeInBytes); + } + } break; + + default: break; + } + if ((chunkSize % 2) != 0) { + bytesWritten += drwav__write_or_count_byte(pWav, 0); + } + } + + if (hasListInfo) { + drwav_uint32 chunkSize = 4; /* Start with 4 bytes for "INFO". */ + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + + if ((pMetadata->type & drwav_metadata_type_list_all_info_strings)) { + chunkSize += 8; /* For id and string size. */ + chunkSize += pMetadata->data.infoText.stringLength + 1; /* Include null terminator. */ + } else if (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list) { + chunkSize += 8; /* For id string size. */ + chunkSize += pMetadata->data.unknown.dataSizeInBytes; + } + + if ((chunkSize % 2) != 0) { + chunkSize += 1; + } + } + + bytesWritten += drwav__write_or_count(pWav, "LIST", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count(pWav, "INFO", 4); + + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + drwav_uint32 subchunkSize = 0; + + if (pMetadata->type & drwav_metadata_type_list_all_info_strings) { + const char* pID = NULL; + + switch (pMetadata->type) { + case drwav_metadata_type_list_info_software: pID = "ISFT"; break; + case drwav_metadata_type_list_info_copyright: pID = "ICOP"; break; + case drwav_metadata_type_list_info_title: pID = "INAM"; break; + case drwav_metadata_type_list_info_artist: pID = "IART"; break; + case drwav_metadata_type_list_info_comment: pID = "ICMT"; break; + case drwav_metadata_type_list_info_date: pID = "ICRD"; break; + case drwav_metadata_type_list_info_genre: pID = "IGNR"; break; + case drwav_metadata_type_list_info_album: pID = "IPRD"; break; + case drwav_metadata_type_list_info_tracknumber: pID = "ITRK"; break; + default: break; + } + + DRWAV_ASSERT(pID != NULL); + + if (pMetadata->data.infoText.stringLength) { + subchunkSize = pMetadata->data.infoText.stringLength + 1; + bytesWritten += drwav__write_or_count(pWav, pID, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.infoText.pString, pMetadata->data.infoText.stringLength); + bytesWritten += drwav__write_or_count_byte(pWav, '\0'); + } + } else if (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list) { + if (pMetadata->data.unknown.dataSizeInBytes) { + subchunkSize = pMetadata->data.unknown.dataSizeInBytes; + + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.unknown.dataSizeInBytes); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, subchunkSize); + } + } + + if ((subchunkSize % 2) != 0) { + bytesWritten += drwav__write_or_count_byte(pWav, 0); + } + } + } + + if (hasListAdtl) { + drwav_uint32 chunkSize = 4; /* start with 4 bytes for "adtl" */ + + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + + switch (pMetadata->type) + { + case drwav_metadata_type_list_label: + case drwav_metadata_type_list_note: + { + chunkSize += 8; /* for id and chunk size */ + chunkSize += DRWAV_LIST_LABEL_OR_NOTE_BYTES; + + if (pMetadata->data.labelOrNote.stringLength > 0) { + chunkSize += pMetadata->data.labelOrNote.stringLength + 1; + } + } break; + + case drwav_metadata_type_list_labelled_cue_region: + { + chunkSize += 8; /* for id and chunk size */ + chunkSize += DRWAV_LIST_LABELLED_TEXT_BYTES; + + if (pMetadata->data.labelledCueRegion.stringLength > 0) { + chunkSize += pMetadata->data.labelledCueRegion.stringLength + 1; + } + } break; + + case drwav_metadata_type_unknown: + { + if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list) { + chunkSize += 8; /* for id and chunk size */ + chunkSize += pMetadata->data.unknown.dataSizeInBytes; + } + } break; + + default: break; + } + + if ((chunkSize % 2) != 0) { + chunkSize += 1; + } + } + + bytesWritten += drwav__write_or_count(pWav, "LIST", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count(pWav, "adtl", 4); + + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + drwav_uint32 subchunkSize = 0; + + switch (pMetadata->type) + { + case drwav_metadata_type_list_label: + case drwav_metadata_type_list_note: + { + if (pMetadata->data.labelOrNote.stringLength > 0) { + const char *pID = NULL; + + if (pMetadata->type == drwav_metadata_type_list_label) { + pID = "labl"; + } + else if (pMetadata->type == drwav_metadata_type_list_note) { + pID = "note"; + } + + DRWAV_ASSERT(pID != NULL); + DRWAV_ASSERT(pMetadata->data.labelOrNote.pString != NULL); + + subchunkSize = DRWAV_LIST_LABEL_OR_NOTE_BYTES; + + bytesWritten += drwav__write_or_count(pWav, pID, 4); + subchunkSize += pMetadata->data.labelOrNote.stringLength + 1; + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize); + + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelOrNote.cuePointId); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelOrNote.pString, pMetadata->data.labelOrNote.stringLength); + bytesWritten += drwav__write_or_count_byte(pWav, '\0'); + } + } break; + + case drwav_metadata_type_list_labelled_cue_region: + { + subchunkSize = DRWAV_LIST_LABELLED_TEXT_BYTES; + + bytesWritten += drwav__write_or_count(pWav, "ltxt", 4); + if (pMetadata->data.labelledCueRegion.stringLength > 0) { + subchunkSize += pMetadata->data.labelledCueRegion.stringLength + 1; + } + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelledCueRegion.cuePointId); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelledCueRegion.sampleLength); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelledCueRegion.purposeId, 4); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.country); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.language); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.dialect); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.codePage); + + if (pMetadata->data.labelledCueRegion.stringLength > 0) { + DRWAV_ASSERT(pMetadata->data.labelledCueRegion.pString != NULL); + + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelledCueRegion.pString, pMetadata->data.labelledCueRegion.stringLength); + bytesWritten += drwav__write_or_count_byte(pWav, '\0'); + } + } break; + + case drwav_metadata_type_unknown: + { + if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list) { + subchunkSize = pMetadata->data.unknown.dataSizeInBytes; + + DRWAV_ASSERT(pMetadata->data.unknown.pData != NULL); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, subchunkSize); + } + } break; + + default: break; + } + + if ((subchunkSize % 2) != 0) { + bytesWritten += drwav__write_or_count_byte(pWav, 0); + } + } + } + + DRWAV_ASSERT((bytesWritten % 2) == 0); + + return bytesWritten; +} + +DRWAV_PRIVATE drwav_uint32 drwav__riff_chunk_size_riff(drwav_uint64 dataChunkSize, drwav_metadata* pMetadata, drwav_uint32 metadataCount) +{ + drwav_uint64 chunkSize = 4 + 24 + (drwav_uint64)drwav__write_or_count_metadata(NULL, pMetadata, metadataCount) + 8 + dataChunkSize + drwav__chunk_padding_size_riff(dataChunkSize); /* 4 = "WAVE". 24 = "fmt " chunk. 8 = "data" + u32 data size. */ + if (chunkSize > 0xFFFFFFFFUL) { + chunkSize = 0xFFFFFFFFUL; + } + + return (drwav_uint32)chunkSize; /* Safe cast due to the clamp above. */ +} + +DRWAV_PRIVATE drwav_uint32 drwav__data_chunk_size_riff(drwav_uint64 dataChunkSize) +{ + if (dataChunkSize <= 0xFFFFFFFFUL) { + return (drwav_uint32)dataChunkSize; + } else { + return 0xFFFFFFFFUL; + } +} + +DRWAV_PRIVATE drwav_uint64 drwav__riff_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + drwav_uint64 dataSubchunkPaddingSize = drwav__chunk_padding_size_w64(dataChunkSize); + + return 80 + 24 + dataChunkSize + dataSubchunkPaddingSize; /* +24 because W64 includes the size of the GUID and size fields. */ +} + +DRWAV_PRIVATE drwav_uint64 drwav__data_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + return 24 + dataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ +} + +DRWAV_PRIVATE drwav_uint64 drwav__riff_chunk_size_rf64(drwav_uint64 dataChunkSize, drwav_metadata *metadata, drwav_uint32 numMetadata) +{ + drwav_uint64 chunkSize = 4 + 36 + 24 + (drwav_uint64)drwav__write_or_count_metadata(NULL, metadata, numMetadata) + 8 + dataChunkSize + drwav__chunk_padding_size_riff(dataChunkSize); /* 4 = "WAVE". 36 = "ds64" chunk. 24 = "fmt " chunk. 8 = "data" + u32 data size. */ + if (chunkSize > 0xFFFFFFFFUL) { + chunkSize = 0xFFFFFFFFUL; + } + + return chunkSize; +} + +DRWAV_PRIVATE drwav_uint64 drwav__data_chunk_size_rf64(drwav_uint64 dataChunkSize) +{ + return dataChunkSize; +} + + + +DRWAV_PRIVATE drwav_bool32 drwav_preinit_write(drwav* pWav, const drwav_data_format* pFormat, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pWav == NULL || onWrite == NULL) { + return DRWAV_FALSE; + } + + if (!isSequential && onSeek == NULL) { + return DRWAV_FALSE; /* <-- onSeek is required when in non-sequential mode. */ + } + + /* Not currently supporting compressed formats. Will need to add support for the "fact" chunk before we enable this. */ + if (pFormat->format == DR_WAVE_FORMAT_EXTENSIBLE) { + return DRWAV_FALSE; + } + if (pFormat->format == DR_WAVE_FORMAT_ADPCM || pFormat->format == DR_WAVE_FORMAT_DVI_ADPCM) { + return DRWAV_FALSE; + } + + DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav)); + pWav->onWrite = onWrite; + pWav->onSeek = onSeek; + pWav->pUserData = pUserData; + pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks); + + if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) { + return DRWAV_FALSE; /* Invalid allocation callbacks. */ + } + + pWav->fmt.formatTag = (drwav_uint16)pFormat->format; + pWav->fmt.channels = (drwav_uint16)pFormat->channels; + pWav->fmt.sampleRate = pFormat->sampleRate; + pWav->fmt.avgBytesPerSec = (drwav_uint32)((pFormat->bitsPerSample * pFormat->sampleRate * pFormat->channels) / 8); + pWav->fmt.blockAlign = (drwav_uint16)((pFormat->channels * pFormat->bitsPerSample) / 8); + pWav->fmt.bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->fmt.extendedSize = 0; + pWav->isSequentialWrite = isSequential; + + return DRWAV_TRUE; +} + + +DRWAV_PRIVATE drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + /* The function assumes drwav_preinit_write() was called beforehand. */ + + size_t runningPos = 0; + drwav_uint64 initialDataChunkSize = 0; + drwav_uint64 chunkSizeFMT; + + /* + The initial values for the "RIFF" and "data" chunks depends on whether or not we are initializing in sequential mode or not. In + sequential mode we set this to its final values straight away since they can be calculated from the total sample count. In non- + sequential mode we initialize it all to zero and fill it out in drwav_uninit() using a backwards seek. + */ + if (pWav->isSequentialWrite) { + initialDataChunkSize = (totalSampleCount * pWav->fmt.bitsPerSample) / 8; + + /* + The RIFF container has a limit on the number of samples. drwav is not allowing this. There's no practical limits for Wave64 + so for the sake of simplicity I'm not doing any validation for that. + */ + if (pFormat->container == drwav_container_riff) { + if (initialDataChunkSize > (0xFFFFFFFFUL - 36)) { + return DRWAV_FALSE; /* Not enough room to store every sample. */ + } + } + } + + pWav->dataChunkDataSizeTargetWrite = initialDataChunkSize; + + + /* "RIFF" chunk. */ + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeRIFF = 28 + (drwav_uint32)initialDataChunkSize; /* +28 = "WAVE" + [sizeof "fmt " chunk] */ + runningPos += drwav__write(pWav, "RIFF", 4); + runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeRIFF); + runningPos += drwav__write(pWav, "WAVE", 4); + } else if (pFormat->container == drwav_container_w64) { + drwav_uint64 chunkSizeRIFF = 80 + 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ + runningPos += drwav__write(pWav, drwavGUID_W64_RIFF, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeRIFF); + runningPos += drwav__write(pWav, drwavGUID_W64_WAVE, 16); + } else if (pFormat->container == drwav_container_rf64) { + runningPos += drwav__write(pWav, "RF64", 4); + runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF); /* Always 0xFFFFFFFF for RF64. Set to a proper value in the "ds64" chunk. */ + runningPos += drwav__write(pWav, "WAVE", 4); + } + + + /* "ds64" chunk (RF64 only). */ + if (pFormat->container == drwav_container_rf64) { + drwav_uint32 initialds64ChunkSize = 28; /* 28 = [Size of RIFF (8 bytes)] + [Size of DATA (8 bytes)] + [Sample Count (8 bytes)] + [Table Length (4 bytes)]. Table length always set to 0. */ + drwav_uint64 initialRiffChunkSize = 8 + initialds64ChunkSize + initialDataChunkSize; /* +8 for the ds64 header. */ + + runningPos += drwav__write(pWav, "ds64", 4); + runningPos += drwav__write_u32ne_to_le(pWav, initialds64ChunkSize); /* Size of ds64. */ + runningPos += drwav__write_u64ne_to_le(pWav, initialRiffChunkSize); /* Size of RIFF. Set to true value at the end. */ + runningPos += drwav__write_u64ne_to_le(pWav, initialDataChunkSize); /* Size of DATA. Set to true value at the end. */ + runningPos += drwav__write_u64ne_to_le(pWav, totalSampleCount); /* Sample count. */ + runningPos += drwav__write_u32ne_to_le(pWav, 0); /* Table length. Always set to zero in our case since we're not doing any other chunks than "DATA". */ + } + + + /* "fmt " chunk. */ + if (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64) { + chunkSizeFMT = 16; + runningPos += drwav__write(pWav, "fmt ", 4); + runningPos += drwav__write_u32ne_to_le(pWav, (drwav_uint32)chunkSizeFMT); + } else if (pFormat->container == drwav_container_w64) { + chunkSizeFMT = 40; + runningPos += drwav__write(pWav, drwavGUID_W64_FMT, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeFMT); + } + + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.formatTag); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.channels); + runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.sampleRate); + runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.avgBytesPerSec); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.blockAlign); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.bitsPerSample); + + /* TODO: is a 'fact' chunk required for DR_WAVE_FORMAT_IEEE_FLOAT? */ + + if (!pWav->isSequentialWrite && pWav->pMetadata != NULL && pWav->metadataCount > 0 && (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64)) { + runningPos += drwav__write_or_count_metadata(pWav, pWav->pMetadata, pWav->metadataCount); + } + + pWav->dataChunkDataPos = runningPos; + + /* "data" chunk. */ + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeDATA = (drwav_uint32)initialDataChunkSize; + runningPos += drwav__write(pWav, "data", 4); + runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeDATA); + } else if (pFormat->container == drwav_container_w64) { + drwav_uint64 chunkSizeDATA = 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ + runningPos += drwav__write(pWav, drwavGUID_W64_DATA, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeDATA); + } else if (pFormat->container == drwav_container_rf64) { + runningPos += drwav__write(pWav, "data", 4); + runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF); /* Always set to 0xFFFFFFFF for RF64. The true size of the data chunk is specified in the ds64 chunk. */ + } + + /* Set some properties for the client's convenience. */ + pWav->container = pFormat->container; + pWav->channels = (drwav_uint16)pFormat->channels; + pWav->sampleRate = pFormat->sampleRate; + pWav->bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->translatedFormatTag = (drwav_uint16)pFormat->format; + pWav->dataChunkDataPos = runningPos; + + return DRWAV_TRUE; +} + + +DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init_write__internal(pWav, pFormat, 0); /* DRWAV_FALSE = Not Sequential */ +} + +DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit_write(pWav, pFormat, DRWAV_TRUE, onWrite, NULL, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init_write__internal(pWav, pFormat, totalSampleCount); /* DRWAV_TRUE = Sequential */ +} + +DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_write_sequential(pWav, pFormat, totalPCMFrameCount*pFormat->channels, onWrite, pUserData, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata, drwav_uint32 metadataCount) +{ + if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->pMetadata = pMetadata; + pWav->metadataCount = metadataCount; + + return drwav_init_write__internal(pWav, pFormat, 0); +} + + +DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount) +{ + /* Casting totalFrameCount to drwav_int64 for VC6 compatibility. No issues in practice because nobody is going to exhaust the whole 63 bits. */ + drwav_uint64 targetDataSizeBytes = (drwav_uint64)((drwav_int64)totalFrameCount * pFormat->channels * pFormat->bitsPerSample/8.0); + drwav_uint64 riffChunkSizeBytes; + drwav_uint64 fileSizeBytes = 0; + + if (pFormat->container == drwav_container_riff) { + riffChunkSizeBytes = drwav__riff_chunk_size_riff(targetDataSizeBytes, pMetadata, metadataCount); + fileSizeBytes = (8 + riffChunkSizeBytes); /* +8 because WAV doesn't include the size of the ChunkID and ChunkSize fields. */ + } else if (pFormat->container == drwav_container_w64) { + riffChunkSizeBytes = drwav__riff_chunk_size_w64(targetDataSizeBytes); + fileSizeBytes = riffChunkSizeBytes; + } else if (pFormat->container == drwav_container_rf64) { + riffChunkSizeBytes = drwav__riff_chunk_size_rf64(targetDataSizeBytes, pMetadata, metadataCount); + fileSizeBytes = (8 + riffChunkSizeBytes); /* +8 because WAV doesn't include the size of the ChunkID and ChunkSize fields. */ + } + + return fileSizeBytes; +} + + +#ifndef DR_WAV_NO_STDIO + +/* drwav_result_from_errno() is only used for fopen() and wfopen() so putting it inside DR_WAV_NO_STDIO for now. If something else needs this later we can move it out. */ +#include +DRWAV_PRIVATE drwav_result drwav_result_from_errno(int e) +{ + switch (e) + { + case 0: return DRWAV_SUCCESS; + #ifdef EPERM + case EPERM: return DRWAV_INVALID_OPERATION; + #endif + #ifdef ENOENT + case ENOENT: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef ESRCH + case ESRCH: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef EINTR + case EINTR: return DRWAV_INTERRUPT; + #endif + #ifdef EIO + case EIO: return DRWAV_IO_ERROR; + #endif + #ifdef ENXIO + case ENXIO: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef E2BIG + case E2BIG: return DRWAV_INVALID_ARGS; + #endif + #ifdef ENOEXEC + case ENOEXEC: return DRWAV_INVALID_FILE; + #endif + #ifdef EBADF + case EBADF: return DRWAV_INVALID_FILE; + #endif + #ifdef ECHILD + case ECHILD: return DRWAV_ERROR; + #endif + #ifdef EAGAIN + case EAGAIN: return DRWAV_UNAVAILABLE; + #endif + #ifdef ENOMEM + case ENOMEM: return DRWAV_OUT_OF_MEMORY; + #endif + #ifdef EACCES + case EACCES: return DRWAV_ACCESS_DENIED; + #endif + #ifdef EFAULT + case EFAULT: return DRWAV_BAD_ADDRESS; + #endif + #ifdef ENOTBLK + case ENOTBLK: return DRWAV_ERROR; + #endif + #ifdef EBUSY + case EBUSY: return DRWAV_BUSY; + #endif + #ifdef EEXIST + case EEXIST: return DRWAV_ALREADY_EXISTS; + #endif + #ifdef EXDEV + case EXDEV: return DRWAV_ERROR; + #endif + #ifdef ENODEV + case ENODEV: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef ENOTDIR + case ENOTDIR: return DRWAV_NOT_DIRECTORY; + #endif + #ifdef EISDIR + case EISDIR: return DRWAV_IS_DIRECTORY; + #endif + #ifdef EINVAL + case EINVAL: return DRWAV_INVALID_ARGS; + #endif + #ifdef ENFILE + case ENFILE: return DRWAV_TOO_MANY_OPEN_FILES; + #endif + #ifdef EMFILE + case EMFILE: return DRWAV_TOO_MANY_OPEN_FILES; + #endif + #ifdef ENOTTY + case ENOTTY: return DRWAV_INVALID_OPERATION; + #endif + #ifdef ETXTBSY + case ETXTBSY: return DRWAV_BUSY; + #endif + #ifdef EFBIG + case EFBIG: return DRWAV_TOO_BIG; + #endif + #ifdef ENOSPC + case ENOSPC: return DRWAV_NO_SPACE; + #endif + #ifdef ESPIPE + case ESPIPE: return DRWAV_BAD_SEEK; + #endif + #ifdef EROFS + case EROFS: return DRWAV_ACCESS_DENIED; + #endif + #ifdef EMLINK + case EMLINK: return DRWAV_TOO_MANY_LINKS; + #endif + #ifdef EPIPE + case EPIPE: return DRWAV_BAD_PIPE; + #endif + #ifdef EDOM + case EDOM: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef ERANGE + case ERANGE: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef EDEADLK + case EDEADLK: return DRWAV_DEADLOCK; + #endif + #ifdef ENAMETOOLONG + case ENAMETOOLONG: return DRWAV_PATH_TOO_LONG; + #endif + #ifdef ENOLCK + case ENOLCK: return DRWAV_ERROR; + #endif + #ifdef ENOSYS + case ENOSYS: return DRWAV_NOT_IMPLEMENTED; + #endif + #ifdef ENOTEMPTY + case ENOTEMPTY: return DRWAV_DIRECTORY_NOT_EMPTY; + #endif + #ifdef ELOOP + case ELOOP: return DRWAV_TOO_MANY_LINKS; + #endif + #ifdef ENOMSG + case ENOMSG: return DRWAV_NO_MESSAGE; + #endif + #ifdef EIDRM + case EIDRM: return DRWAV_ERROR; + #endif + #ifdef ECHRNG + case ECHRNG: return DRWAV_ERROR; + #endif + #ifdef EL2NSYNC + case EL2NSYNC: return DRWAV_ERROR; + #endif + #ifdef EL3HLT + case EL3HLT: return DRWAV_ERROR; + #endif + #ifdef EL3RST + case EL3RST: return DRWAV_ERROR; + #endif + #ifdef ELNRNG + case ELNRNG: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef EUNATCH + case EUNATCH: return DRWAV_ERROR; + #endif + #ifdef ENOCSI + case ENOCSI: return DRWAV_ERROR; + #endif + #ifdef EL2HLT + case EL2HLT: return DRWAV_ERROR; + #endif + #ifdef EBADE + case EBADE: return DRWAV_ERROR; + #endif + #ifdef EBADR + case EBADR: return DRWAV_ERROR; + #endif + #ifdef EXFULL + case EXFULL: return DRWAV_ERROR; + #endif + #ifdef ENOANO + case ENOANO: return DRWAV_ERROR; + #endif + #ifdef EBADRQC + case EBADRQC: return DRWAV_ERROR; + #endif + #ifdef EBADSLT + case EBADSLT: return DRWAV_ERROR; + #endif + #ifdef EBFONT + case EBFONT: return DRWAV_INVALID_FILE; + #endif + #ifdef ENOSTR + case ENOSTR: return DRWAV_ERROR; + #endif + #ifdef ENODATA + case ENODATA: return DRWAV_NO_DATA_AVAILABLE; + #endif + #ifdef ETIME + case ETIME: return DRWAV_TIMEOUT; + #endif + #ifdef ENOSR + case ENOSR: return DRWAV_NO_DATA_AVAILABLE; + #endif + #ifdef ENONET + case ENONET: return DRWAV_NO_NETWORK; + #endif + #ifdef ENOPKG + case ENOPKG: return DRWAV_ERROR; + #endif + #ifdef EREMOTE + case EREMOTE: return DRWAV_ERROR; + #endif + #ifdef ENOLINK + case ENOLINK: return DRWAV_ERROR; + #endif + #ifdef EADV + case EADV: return DRWAV_ERROR; + #endif + #ifdef ESRMNT + case ESRMNT: return DRWAV_ERROR; + #endif + #ifdef ECOMM + case ECOMM: return DRWAV_ERROR; + #endif + #ifdef EPROTO + case EPROTO: return DRWAV_ERROR; + #endif + #ifdef EMULTIHOP + case EMULTIHOP: return DRWAV_ERROR; + #endif + #ifdef EDOTDOT + case EDOTDOT: return DRWAV_ERROR; + #endif + #ifdef EBADMSG + case EBADMSG: return DRWAV_BAD_MESSAGE; + #endif + #ifdef EOVERFLOW + case EOVERFLOW: return DRWAV_TOO_BIG; + #endif + #ifdef ENOTUNIQ + case ENOTUNIQ: return DRWAV_NOT_UNIQUE; + #endif + #ifdef EBADFD + case EBADFD: return DRWAV_ERROR; + #endif + #ifdef EREMCHG + case EREMCHG: return DRWAV_ERROR; + #endif + #ifdef ELIBACC + case ELIBACC: return DRWAV_ACCESS_DENIED; + #endif + #ifdef ELIBBAD + case ELIBBAD: return DRWAV_INVALID_FILE; + #endif + #ifdef ELIBSCN + case ELIBSCN: return DRWAV_INVALID_FILE; + #endif + #ifdef ELIBMAX + case ELIBMAX: return DRWAV_ERROR; + #endif + #ifdef ELIBEXEC + case ELIBEXEC: return DRWAV_ERROR; + #endif + #ifdef EILSEQ + case EILSEQ: return DRWAV_INVALID_DATA; + #endif + #ifdef ERESTART + case ERESTART: return DRWAV_ERROR; + #endif + #ifdef ESTRPIPE + case ESTRPIPE: return DRWAV_ERROR; + #endif + #ifdef EUSERS + case EUSERS: return DRWAV_ERROR; + #endif + #ifdef ENOTSOCK + case ENOTSOCK: return DRWAV_NOT_SOCKET; + #endif + #ifdef EDESTADDRREQ + case EDESTADDRREQ: return DRWAV_NO_ADDRESS; + #endif + #ifdef EMSGSIZE + case EMSGSIZE: return DRWAV_TOO_BIG; + #endif + #ifdef EPROTOTYPE + case EPROTOTYPE: return DRWAV_BAD_PROTOCOL; + #endif + #ifdef ENOPROTOOPT + case ENOPROTOOPT: return DRWAV_PROTOCOL_UNAVAILABLE; + #endif + #ifdef EPROTONOSUPPORT + case EPROTONOSUPPORT: return DRWAV_PROTOCOL_NOT_SUPPORTED; + #endif + #ifdef ESOCKTNOSUPPORT + case ESOCKTNOSUPPORT: return DRWAV_SOCKET_NOT_SUPPORTED; + #endif + #ifdef EOPNOTSUPP + case EOPNOTSUPP: return DRWAV_INVALID_OPERATION; + #endif + #ifdef EPFNOSUPPORT + case EPFNOSUPPORT: return DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EAFNOSUPPORT + case EAFNOSUPPORT: return DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EADDRINUSE + case EADDRINUSE: return DRWAV_ALREADY_IN_USE; + #endif + #ifdef EADDRNOTAVAIL + case EADDRNOTAVAIL: return DRWAV_ERROR; + #endif + #ifdef ENETDOWN + case ENETDOWN: return DRWAV_NO_NETWORK; + #endif + #ifdef ENETUNREACH + case ENETUNREACH: return DRWAV_NO_NETWORK; + #endif + #ifdef ENETRESET + case ENETRESET: return DRWAV_NO_NETWORK; + #endif + #ifdef ECONNABORTED + case ECONNABORTED: return DRWAV_NO_NETWORK; + #endif + #ifdef ECONNRESET + case ECONNRESET: return DRWAV_CONNECTION_RESET; + #endif + #ifdef ENOBUFS + case ENOBUFS: return DRWAV_NO_SPACE; + #endif + #ifdef EISCONN + case EISCONN: return DRWAV_ALREADY_CONNECTED; + #endif + #ifdef ENOTCONN + case ENOTCONN: return DRWAV_NOT_CONNECTED; + #endif + #ifdef ESHUTDOWN + case ESHUTDOWN: return DRWAV_ERROR; + #endif + #ifdef ETOOMANYREFS + case ETOOMANYREFS: return DRWAV_ERROR; + #endif + #ifdef ETIMEDOUT + case ETIMEDOUT: return DRWAV_TIMEOUT; + #endif + #ifdef ECONNREFUSED + case ECONNREFUSED: return DRWAV_CONNECTION_REFUSED; + #endif + #ifdef EHOSTDOWN + case EHOSTDOWN: return DRWAV_NO_HOST; + #endif + #ifdef EHOSTUNREACH + case EHOSTUNREACH: return DRWAV_NO_HOST; + #endif + #ifdef EALREADY + case EALREADY: return DRWAV_IN_PROGRESS; + #endif + #ifdef EINPROGRESS + case EINPROGRESS: return DRWAV_IN_PROGRESS; + #endif + #ifdef ESTALE + case ESTALE: return DRWAV_INVALID_FILE; + #endif + #ifdef EUCLEAN + case EUCLEAN: return DRWAV_ERROR; + #endif + #ifdef ENOTNAM + case ENOTNAM: return DRWAV_ERROR; + #endif + #ifdef ENAVAIL + case ENAVAIL: return DRWAV_ERROR; + #endif + #ifdef EISNAM + case EISNAM: return DRWAV_ERROR; + #endif + #ifdef EREMOTEIO + case EREMOTEIO: return DRWAV_IO_ERROR; + #endif + #ifdef EDQUOT + case EDQUOT: return DRWAV_NO_SPACE; + #endif + #ifdef ENOMEDIUM + case ENOMEDIUM: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef EMEDIUMTYPE + case EMEDIUMTYPE: return DRWAV_ERROR; + #endif + #ifdef ECANCELED + case ECANCELED: return DRWAV_CANCELLED; + #endif + #ifdef ENOKEY + case ENOKEY: return DRWAV_ERROR; + #endif + #ifdef EKEYEXPIRED + case EKEYEXPIRED: return DRWAV_ERROR; + #endif + #ifdef EKEYREVOKED + case EKEYREVOKED: return DRWAV_ERROR; + #endif + #ifdef EKEYREJECTED + case EKEYREJECTED: return DRWAV_ERROR; + #endif + #ifdef EOWNERDEAD + case EOWNERDEAD: return DRWAV_ERROR; + #endif + #ifdef ENOTRECOVERABLE + case ENOTRECOVERABLE: return DRWAV_ERROR; + #endif + #ifdef ERFKILL + case ERFKILL: return DRWAV_ERROR; + #endif + #ifdef EHWPOISON + case EHWPOISON: return DRWAV_ERROR; + #endif + default: return DRWAV_ERROR; + } +} + +DRWAV_PRIVATE drwav_result drwav_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode) +{ +#if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err; +#endif + + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRWAV_INVALID_ARGS; + } + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + err = fopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drwav_result_from_errno(err); + } +#else +#if defined(_WIN32) || defined(__APPLE__) + *ppFile = fopen(pFilePath, pOpenMode); +#else + #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE) + *ppFile = fopen64(pFilePath, pOpenMode); + #else + *ppFile = fopen(pFilePath, pOpenMode); + #endif +#endif + if (*ppFile == NULL) { + drwav_result result = drwav_result_from_errno(errno); + if (result == DRWAV_SUCCESS) { + result = DRWAV_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */ + } + + return result; + } +#endif + + return DRWAV_SUCCESS; +} + +/* +_wfopen() isn't always available in all compilation environments. + + * Windows only. + * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). + * MinGW-64 (both 32- and 64-bit) seems to support it. + * MinGW wraps it in !defined(__STRICT_ANSI__). + * OpenWatcom wraps it in !defined(_NO_EXT_KEYS). + +This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() +fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. +*/ +#if defined(_WIN32) + #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) + #define DRWAV_HAS_WFOPEN + #endif +#endif + +#ifndef DR_WAV_NO_WCHAR +DRWAV_PRIVATE drwav_result drwav_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRWAV_INVALID_ARGS; + } + +#if defined(DRWAV_HAS_WFOPEN) + { + /* Use _wfopen() on Windows. */ + #if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drwav_result_from_errno(err); + } + #else + *ppFile = _wfopen(pFilePath, pOpenMode); + if (*ppFile == NULL) { + return drwav_result_from_errno(errno); + } + #endif + (void)pAllocationCallbacks; + } +#else + /* + Use fopen() on anything other than Windows. Requires a conversion. This is annoying because + fopen() is locale specific. The only real way I can think of to do this is with wcsrtombs(). Note + that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for + maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler + error I'll look into improving compatibility. + */ + + /* + Some compilers don't support wchar_t or wcsrtombs() which we're using below. In this case we just + need to abort with an error. If you encounter a compiler lacking such support, add it to this list + and submit a bug report and it'll be added to the library upstream. + */ + #if defined(__DJGPP__) + { + /* Nothing to do here. This will fall through to the error check below. */ + } + #else + { + mbstate_t mbs; + size_t lenMB; + const wchar_t* pFilePathTemp = pFilePath; + char* pFilePathMB = NULL; + char pOpenModeMB[32] = {0}; + + /* Get the length first. */ + DRWAV_ZERO_OBJECT(&mbs); + lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); + if (lenMB == (size_t)-1) { + return drwav_result_from_errno(errno); + } + + pFilePathMB = (char*)drwav__malloc_from_callbacks(lenMB + 1, pAllocationCallbacks); + if (pFilePathMB == NULL) { + return DRWAV_OUT_OF_MEMORY; + } + + pFilePathTemp = pFilePath; + DRWAV_ZERO_OBJECT(&mbs); + wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); + + /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */ + { + size_t i = 0; + for (;;) { + if (pOpenMode[i] == 0) { + pOpenModeMB[i] = '\0'; + break; + } + + pOpenModeMB[i] = (char)pOpenMode[i]; + i += 1; + } + } + + *ppFile = fopen(pFilePathMB, pOpenModeMB); + + drwav__free_from_callbacks(pFilePathMB, pAllocationCallbacks); + } + #endif + + if (*ppFile == NULL) { + return DRWAV_ERROR; + } +#endif + + return DRWAV_SUCCESS; +} +#endif + + +DRWAV_PRIVATE size_t drwav__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +DRWAV_PRIVATE size_t drwav__on_write_stdio(void* pUserData, const void* pData, size_t bytesToWrite) +{ + return fwrite(pData, 1, bytesToWrite, (FILE*)pUserData); +} + +DRWAV_PRIVATE drwav_bool32 drwav__on_seek_stdio(void* pUserData, int offset, drwav_seek_origin origin) +{ + return fseek((FILE*)pUserData, offset, (origin == drwav_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0; +} + +DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_ex(pWav, filename, NULL, NULL, 0, pAllocationCallbacks); +} + + +DRWAV_PRIVATE drwav_bool32 drwav_init_file__internal_FILE(drwav* pWav, FILE* pFile, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, drwav_metadata_type allowedMetadataTypes, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav_bool32 result; + + result = drwav_preinit(pWav, drwav__on_read_stdio, drwav__on_seek_stdio, (void*)pFile, pAllocationCallbacks); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + pWav->allowedMetadataTypes = allowedMetadataTypes; + + result = drwav_init__internal(pWav, onChunk, pChunkUserData, flags); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_fopen(&pFile, filename, "rb") != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, drwav_metadata_type_none, pAllocationCallbacks); +} + +#ifndef DR_WAV_NO_WCHAR +DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_ex_w(pWav, filename, NULL, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, drwav_metadata_type_none, pAllocationCallbacks); +} +#endif + +DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_fopen(&pFile, filename, "rb") != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, NULL, NULL, flags, drwav_metadata_type_all_including_unknown, pAllocationCallbacks); +} + +#ifndef DR_WAV_NO_WCHAR +DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, NULL, NULL, flags, drwav_metadata_type_all_including_unknown, pAllocationCallbacks); +} +#endif + + +DRWAV_PRIVATE drwav_bool32 drwav_init_file_write__internal_FILE(drwav* pWav, FILE* pFile, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav_bool32 result; + + result = drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile, pAllocationCallbacks); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + result = drwav_init_write__internal(pWav, pFormat, totalSampleCount); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE drwav_bool32 drwav_init_file_write__internal(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_fopen(&pFile, filename, "wb") != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks); +} + +#ifndef DR_WAV_NO_WCHAR +DRWAV_PRIVATE drwav_bool32 drwav_init_file_write_w__internal(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_wfopen(&pFile, filename, L"wb", pAllocationCallbacks) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks); +} +#endif + +DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_file_write_sequential(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} + +#ifndef DR_WAV_NO_WCHAR +DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write_w__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write_w__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_file_write_sequential_w(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} +#endif +#endif /* DR_WAV_NO_STDIO */ + + +DRWAV_PRIVATE size_t drwav__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + drwav* pWav = (drwav*)pUserData; + size_t bytesRemaining; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->memoryStream.dataSize >= pWav->memoryStream.currentReadPos); + + bytesRemaining = pWav->memoryStream.dataSize - pWav->memoryStream.currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRWAV_COPY_MEMORY(pBufferOut, pWav->memoryStream.data + pWav->memoryStream.currentReadPos, bytesToRead); + pWav->memoryStream.currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +DRWAV_PRIVATE drwav_bool32 drwav__on_seek_memory(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav* pWav = (drwav*)pUserData; + DRWAV_ASSERT(pWav != NULL); + + if (origin == drwav_seek_origin_current) { + if (offset > 0) { + if (pWav->memoryStream.currentReadPos + offset > pWav->memoryStream.dataSize) { + return DRWAV_FALSE; /* Trying to seek too far forward. */ + } + } else { + if (pWav->memoryStream.currentReadPos < (size_t)-offset) { + return DRWAV_FALSE; /* Trying to seek too far backwards. */ + } + } + + /* This will never underflow thanks to the clamps above. */ + pWav->memoryStream.currentReadPos += offset; + } else { + if ((drwav_uint32)offset <= pWav->memoryStream.dataSize) { + pWav->memoryStream.currentReadPos = offset; + } else { + return DRWAV_FALSE; /* Trying to seek too far forward. */ + } + } + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE size_t drwav__on_write_memory(void* pUserData, const void* pDataIn, size_t bytesToWrite) +{ + drwav* pWav = (drwav*)pUserData; + size_t bytesRemaining; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->memoryStreamWrite.dataCapacity >= pWav->memoryStreamWrite.currentWritePos); + + bytesRemaining = pWav->memoryStreamWrite.dataCapacity - pWav->memoryStreamWrite.currentWritePos; + if (bytesRemaining < bytesToWrite) { + /* Need to reallocate. */ + void* pNewData; + size_t newDataCapacity = (pWav->memoryStreamWrite.dataCapacity == 0) ? 256 : pWav->memoryStreamWrite.dataCapacity * 2; + + /* If doubling wasn't enough, just make it the minimum required size to write the data. */ + if ((newDataCapacity - pWav->memoryStreamWrite.currentWritePos) < bytesToWrite) { + newDataCapacity = pWav->memoryStreamWrite.currentWritePos + bytesToWrite; + } + + pNewData = drwav__realloc_from_callbacks(*pWav->memoryStreamWrite.ppData, newDataCapacity, pWav->memoryStreamWrite.dataCapacity, &pWav->allocationCallbacks); + if (pNewData == NULL) { + return 0; + } + + *pWav->memoryStreamWrite.ppData = pNewData; + pWav->memoryStreamWrite.dataCapacity = newDataCapacity; + } + + DRWAV_COPY_MEMORY(((drwav_uint8*)(*pWav->memoryStreamWrite.ppData)) + pWav->memoryStreamWrite.currentWritePos, pDataIn, bytesToWrite); + + pWav->memoryStreamWrite.currentWritePos += bytesToWrite; + if (pWav->memoryStreamWrite.dataSize < pWav->memoryStreamWrite.currentWritePos) { + pWav->memoryStreamWrite.dataSize = pWav->memoryStreamWrite.currentWritePos; + } + + *pWav->memoryStreamWrite.pDataSize = pWav->memoryStreamWrite.dataSize; + + return bytesToWrite; +} + +DRWAV_PRIVATE drwav_bool32 drwav__on_seek_memory_write(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav* pWav = (drwav*)pUserData; + DRWAV_ASSERT(pWav != NULL); + + if (origin == drwav_seek_origin_current) { + if (offset > 0) { + if (pWav->memoryStreamWrite.currentWritePos + offset > pWav->memoryStreamWrite.dataSize) { + offset = (int)(pWav->memoryStreamWrite.dataSize - pWav->memoryStreamWrite.currentWritePos); /* Trying to seek too far forward. */ + } + } else { + if (pWav->memoryStreamWrite.currentWritePos < (size_t)-offset) { + offset = -(int)pWav->memoryStreamWrite.currentWritePos; /* Trying to seek too far backwards. */ + } + } + + /* This will never underflow thanks to the clamps above. */ + pWav->memoryStreamWrite.currentWritePos += offset; + } else { + if ((drwav_uint32)offset <= pWav->memoryStreamWrite.dataSize) { + pWav->memoryStreamWrite.currentWritePos = offset; + } else { + pWav->memoryStreamWrite.currentWritePos = pWav->memoryStreamWrite.dataSize; /* Trying to seek too far forward. */ + } + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_ex(pWav, data, dataSize, NULL, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (data == NULL || dataSize == 0) { + return DRWAV_FALSE; + } + + if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, pWav, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->memoryStream.data = (const drwav_uint8*)data; + pWav->memoryStream.dataSize = dataSize; + pWav->memoryStream.currentReadPos = 0; + + return drwav_init__internal(pWav, onChunk, pChunkUserData, flags); +} + +DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (data == NULL || dataSize == 0) { + return DRWAV_FALSE; + } + + if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, pWav, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->memoryStream.data = (const drwav_uint8*)data; + pWav->memoryStream.dataSize = dataSize; + pWav->memoryStream.currentReadPos = 0; + + pWav->allowedMetadataTypes = drwav_metadata_type_all_including_unknown; + + return drwav_init__internal(pWav, NULL, NULL, flags); +} + + +DRWAV_PRIVATE drwav_bool32 drwav_init_memory_write__internal(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (ppData == NULL || pDataSize == NULL) { + return DRWAV_FALSE; + } + + *ppData = NULL; /* Important because we're using realloc()! */ + *pDataSize = 0; + + if (!drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, pWav, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->memoryStreamWrite.ppData = ppData; + pWav->memoryStreamWrite.pDataSize = pDataSize; + pWav->memoryStreamWrite.dataSize = 0; + pWav->memoryStreamWrite.dataCapacity = 0; + pWav->memoryStreamWrite.currentWritePos = 0; + + return drwav_init_write__internal(pWav, pFormat, totalSampleCount); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_memory_write_sequential(pWav, ppData, pDataSize, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} + + + +DRWAV_API drwav_result drwav_uninit(drwav* pWav) +{ + drwav_result result = DRWAV_SUCCESS; + + if (pWav == NULL) { + return DRWAV_INVALID_ARGS; + } + + /* + If the drwav object was opened in write mode we'll need to finalize a few things: + - Make sure the "data" chunk is aligned to 16-bits for RIFF containers, or 64 bits for W64 containers. + - Set the size of the "data" chunk. + */ + if (pWav->onWrite != NULL) { + drwav_uint32 paddingSize = 0; + + /* Padding. Do not adjust pWav->dataChunkDataSize - this should not include the padding. */ + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) { + paddingSize = drwav__chunk_padding_size_riff(pWav->dataChunkDataSize); + } else { + paddingSize = drwav__chunk_padding_size_w64(pWav->dataChunkDataSize); + } + + if (paddingSize > 0) { + drwav_uint64 paddingData = 0; + drwav__write(pWav, &paddingData, paddingSize); /* Byte order does not matter for this. */ + } + + /* + Chunk sizes. When using sequential mode, these will have been filled in at initialization time. We only need + to do this when using non-sequential mode. + */ + if (pWav->onSeek && !pWav->isSequentialWrite) { + if (pWav->container == drwav_container_riff) { + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, 4, drwav_seek_origin_start)) { + drwav_uint32 riffChunkSize = drwav__riff_chunk_size_riff(pWav->dataChunkDataSize, pWav->pMetadata, pWav->metadataCount); + drwav__write_u32ne_to_le(pWav, riffChunkSize); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos - 4, drwav_seek_origin_start)) { + drwav_uint32 dataChunkSize = drwav__data_chunk_size_riff(pWav->dataChunkDataSize); + drwav__write_u32ne_to_le(pWav, dataChunkSize); + } + } else if (pWav->container == drwav_container_w64) { + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, 16, drwav_seek_origin_start)) { + drwav_uint64 riffChunkSize = drwav__riff_chunk_size_w64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, riffChunkSize); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos - 8, drwav_seek_origin_start)) { + drwav_uint64 dataChunkSize = drwav__data_chunk_size_w64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, dataChunkSize); + } + } else if (pWav->container == drwav_container_rf64) { + /* We only need to update the ds64 chunk. The "RIFF" and "data" chunks always have their sizes set to 0xFFFFFFFF for RF64. */ + int ds64BodyPos = 12 + 8; + + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 0, drwav_seek_origin_start)) { + drwav_uint64 riffChunkSize = drwav__riff_chunk_size_rf64(pWav->dataChunkDataSize, pWav->pMetadata, pWav->metadataCount); + drwav__write_u64ne_to_le(pWav, riffChunkSize); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 8, drwav_seek_origin_start)) { + drwav_uint64 dataChunkSize = drwav__data_chunk_size_rf64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, dataChunkSize); + } + } + } + + /* Validation for sequential mode. */ + if (pWav->isSequentialWrite) { + if (pWav->dataChunkDataSize != pWav->dataChunkDataSizeTargetWrite) { + result = DRWAV_INVALID_FILE; + } + } + } else { + if (pWav->pMetadata != NULL) { + pWav->allocationCallbacks.onFree(pWav->pMetadata, pWav->allocationCallbacks.pUserData); + } + } + +#ifndef DR_WAV_NO_STDIO + /* + If we opened the file with drwav_open_file() we will want to close the file handle. We can know whether or not drwav_open_file() + was used by looking at the onRead and onSeek callbacks. + */ + if (pWav->onRead == drwav__on_read_stdio || pWav->onWrite == drwav__on_write_stdio) { + fclose((FILE*)pWav->pUserData); + } +#endif + + return result; +} + + + +DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut) +{ + size_t bytesRead; + drwav_uint32 bytesPerFrame; + + if (pWav == NULL || bytesToRead == 0) { + return 0; /* Invalid args. */ + } + + if (bytesToRead > pWav->bytesRemaining) { + bytesToRead = (size_t)pWav->bytesRemaining; + } + + if (bytesToRead == 0) { + return 0; /* At end. */ + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; /* Could not determine the bytes per frame. */ + } + + if (pBufferOut != NULL) { + bytesRead = pWav->onRead(pWav->pUserData, pBufferOut, bytesToRead); + } else { + /* We need to seek. If we fail, we need to read-and-discard to make sure we get a good byte count. */ + bytesRead = 0; + while (bytesRead < bytesToRead) { + size_t bytesToSeek = (bytesToRead - bytesRead); + if (bytesToSeek > 0x7FFFFFFF) { + bytesToSeek = 0x7FFFFFFF; + } + + if (pWav->onSeek(pWav->pUserData, (int)bytesToSeek, drwav_seek_origin_current) == DRWAV_FALSE) { + break; + } + + bytesRead += bytesToSeek; + } + + /* When we get here we may need to read-and-discard some data. */ + while (bytesRead < bytesToRead) { + drwav_uint8 buffer[4096]; + size_t bytesSeeked; + size_t bytesToSeek = (bytesToRead - bytesRead); + if (bytesToSeek > sizeof(buffer)) { + bytesToSeek = sizeof(buffer); + } + + bytesSeeked = pWav->onRead(pWav->pUserData, buffer, bytesToSeek); + bytesRead += bytesSeeked; + + if (bytesSeeked < bytesToSeek) { + break; /* Reached the end. */ + } + } + } + + pWav->readCursorInPCMFrames += bytesRead / bytesPerFrame; + + pWav->bytesRemaining -= bytesRead; + return bytesRead; +} + + + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + drwav_uint32 bytesPerFrame; + drwav_uint64 bytesToRead; /* Intentionally uint64 instead of size_t so we can do a check that we're not reading too much on 32-bit builds. */ + + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + /* Cannot use this function for compressed formats. */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + return 0; + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + bytesToRead = framesToRead * bytesPerFrame; + if (bytesToRead > DRWAV_SIZE_MAX) { + bytesToRead = (DRWAV_SIZE_MAX / bytesPerFrame) * bytesPerFrame; /* Round the number of bytes to read to a clean frame boundary. */ + } + + /* + Doing an explicit check here just to make it clear that we don't want to be attempt to read anything if there's no bytes to read. There + *could* be a time where it evaluates to 0 due to overflowing. + */ + if (bytesToRead == 0) { + return 0; + } + + return drwav_read_raw(pWav, (size_t)bytesToRead, pBufferOut) / bytesPerFrame; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut); + + if (pBufferOut != NULL) { + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; /* Could not get the bytes per frame which means bytes per sample cannot be determined and we don't know how to byte swap. */ + } + + drwav__bswap_samples(pBufferOut, framesRead*pWav->channels, bytesPerFrame/pWav->channels, pWav->translatedFormatTag); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + if (drwav__is_little_endian()) { + return drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut); + } else { + return drwav_read_pcm_frames_be(pWav, framesToRead, pBufferOut); + } +} + + + +DRWAV_PRIVATE drwav_bool32 drwav_seek_to_first_pcm_frame(drwav* pWav) +{ + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; /* No seeking in write mode. */ + } + + if (!pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos, drwav_seek_origin_start)) { + return DRWAV_FALSE; + } + + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + /* Cached data needs to be cleared for compressed formats. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + DRWAV_ZERO_OBJECT(&pWav->msadpcm); + } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + DRWAV_ZERO_OBJECT(&pWav->ima); + } else { + DRWAV_ASSERT(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */ + } + } + + pWav->readCursorInPCMFrames = 0; + pWav->bytesRemaining = pWav->dataChunkDataSize; + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex) +{ + /* Seeking should be compatible with wave files > 2GB. */ + + if (pWav == NULL || pWav->onSeek == NULL) { + return DRWAV_FALSE; + } + + /* No seeking in write mode. */ + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; + } + + /* If there are no samples, just return DRWAV_TRUE without doing anything. */ + if (pWav->totalPCMFrameCount == 0) { + return DRWAV_TRUE; + } + + /* Make sure the sample is clamped. */ + if (targetFrameIndex > pWav->totalPCMFrameCount) { + targetFrameIndex = pWav->totalPCMFrameCount; + } + + /* + For compressed formats we just use a slow generic seek. If we are seeking forward we just seek forward. If we are going backwards we need + to seek back to the start. + */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + /* TODO: This can be optimized. */ + + /* + If we're seeking forward it's simple - just keep reading samples until we hit the sample we're requesting. If we're seeking backwards, + we first need to seek back to the start and then just do the same thing as a forward seek. + */ + if (targetFrameIndex < pWav->readCursorInPCMFrames) { + if (!drwav_seek_to_first_pcm_frame(pWav)) { + return DRWAV_FALSE; + } + } + + if (targetFrameIndex > pWav->readCursorInPCMFrames) { + drwav_uint64 offsetInFrames = targetFrameIndex - pWav->readCursorInPCMFrames; + + drwav_int16 devnull[2048]; + while (offsetInFrames > 0) { + drwav_uint64 framesRead = 0; + drwav_uint64 framesToRead = offsetInFrames; + if (framesToRead > drwav_countof(devnull)/pWav->channels) { + framesToRead = drwav_countof(devnull)/pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + framesRead = drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, devnull); + } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + framesRead = drwav_read_pcm_frames_s16__ima(pWav, framesToRead, devnull); + } else { + DRWAV_ASSERT(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */ + } + + if (framesRead != framesToRead) { + return DRWAV_FALSE; + } + + offsetInFrames -= framesRead; + } + } + } else { + drwav_uint64 totalSizeInBytes; + drwav_uint64 currentBytePos; + drwav_uint64 targetBytePos; + drwav_uint64 offset; + drwav_uint32 bytesPerFrame; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return DRWAV_FALSE; /* Not able to calculate offset. */ + } + + totalSizeInBytes = pWav->totalPCMFrameCount * bytesPerFrame; + DRWAV_ASSERT(totalSizeInBytes >= pWav->bytesRemaining); + + currentBytePos = totalSizeInBytes - pWav->bytesRemaining; + targetBytePos = targetFrameIndex * bytesPerFrame; + + if (currentBytePos < targetBytePos) { + /* Offset forwards. */ + offset = (targetBytePos - currentBytePos); + } else { + /* Offset backwards. */ + if (!drwav_seek_to_first_pcm_frame(pWav)) { + return DRWAV_FALSE; + } + offset = targetBytePos; + } + + while (offset > 0) { + int offset32 = ((offset > INT_MAX) ? INT_MAX : (int)offset); + if (!pWav->onSeek(pWav->pUserData, offset32, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + + pWav->readCursorInPCMFrames += offset32 / bytesPerFrame; + pWav->bytesRemaining -= offset32; + offset -= offset32; + } + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor) +{ + if (pCursor == NULL) { + return DRWAV_INVALID_ARGS; + } + + *pCursor = 0; /* Safety. */ + + if (pWav == NULL) { + return DRWAV_INVALID_ARGS; + } + + *pCursor = pWav->readCursorInPCMFrames; + + return DRWAV_SUCCESS; +} + +DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength) +{ + if (pLength == NULL) { + return DRWAV_INVALID_ARGS; + } + + *pLength = 0; /* Safety. */ + + if (pWav == NULL) { + return DRWAV_INVALID_ARGS; + } + + *pLength = pWav->totalPCMFrameCount; + + return DRWAV_SUCCESS; +} + + +DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData) +{ + size_t bytesWritten; + + if (pWav == NULL || bytesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesWritten = pWav->onWrite(pWav->pUserData, pData, bytesToWrite); + pWav->dataChunkDataSize += bytesWritten; + + return bytesWritten; +} + +DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + drwav_uint64 bytesToWrite; + drwav_uint64 bytesWritten; + const drwav_uint8* pRunningData; + + if (pWav == NULL || framesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8); + if (bytesToWrite > DRWAV_SIZE_MAX) { + return 0; + } + + bytesWritten = 0; + pRunningData = (const drwav_uint8*)pData; + + while (bytesToWrite > 0) { + size_t bytesJustWritten; + drwav_uint64 bytesToWriteThisIteration; + + bytesToWriteThisIteration = bytesToWrite; + DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX); /* <-- This is checked above. */ + + bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, pRunningData); + if (bytesJustWritten == 0) { + break; + } + + bytesToWrite -= bytesJustWritten; + bytesWritten += bytesJustWritten; + pRunningData += bytesJustWritten; + } + + return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels; +} + +DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + drwav_uint64 bytesToWrite; + drwav_uint64 bytesWritten; + drwav_uint32 bytesPerSample; + const drwav_uint8* pRunningData; + + if (pWav == NULL || framesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8); + if (bytesToWrite > DRWAV_SIZE_MAX) { + return 0; + } + + bytesWritten = 0; + pRunningData = (const drwav_uint8*)pData; + + bytesPerSample = drwav_get_bytes_per_pcm_frame(pWav) / pWav->channels; + if (bytesPerSample == 0) { + return 0; /* Cannot determine bytes per sample, or bytes per sample is less than one byte. */ + } + + while (bytesToWrite > 0) { + drwav_uint8 temp[4096]; + drwav_uint32 sampleCount; + size_t bytesJustWritten; + drwav_uint64 bytesToWriteThisIteration; + + bytesToWriteThisIteration = bytesToWrite; + DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX); /* <-- This is checked above. */ + + /* + WAV files are always little-endian. We need to byte swap on big-endian architectures. Since our input buffer is read-only we need + to use an intermediary buffer for the conversion. + */ + sampleCount = sizeof(temp)/bytesPerSample; + + if (bytesToWriteThisIteration > ((drwav_uint64)sampleCount)*bytesPerSample) { + bytesToWriteThisIteration = ((drwav_uint64)sampleCount)*bytesPerSample; + } + + DRWAV_COPY_MEMORY(temp, pRunningData, (size_t)bytesToWriteThisIteration); + drwav__bswap_samples(temp, sampleCount, bytesPerSample, pWav->translatedFormatTag); + + bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, temp); + if (bytesJustWritten == 0) { + break; + } + + bytesToWrite -= bytesJustWritten; + bytesWritten += bytesJustWritten; + pRunningData += bytesJustWritten; + } + + return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels; +} + +DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + if (drwav__is_little_endian()) { + return drwav_write_pcm_frames_le(pWav, framesToWrite, pData); + } else { + return drwav_write_pcm_frames_be(pWav, framesToWrite, pData); + } +} + + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead = 0; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(framesToRead > 0); + + /* TODO: Lots of room for optimization here. */ + + while (pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) { + DRWAV_ASSERT(framesToRead > 0); /* This loop iteration will never get hit with framesToRead == 0 because it's asserted at the top, and we check for 0 inside the loop just below. */ + + /* If there are no cached frames we need to load a new block. */ + if (pWav->msadpcm.cachedFrameCount == 0 && pWav->msadpcm.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + /* Mono. */ + drwav_uint8 header[7]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.delta[0] = drwav_bytes_to_s16(header + 1); + pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav_bytes_to_s16(header + 3); + pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav_bytes_to_s16(header + 5); + pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][0]; + pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.cachedFrameCount = 2; + } else { + /* Stereo. */ + drwav_uint8 header[14]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.predictor[1] = header[1]; + pWav->msadpcm.delta[0] = drwav_bytes_to_s16(header + 2); + pWav->msadpcm.delta[1] = drwav_bytes_to_s16(header + 4); + pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav_bytes_to_s16(header + 6); + pWav->msadpcm.prevFrames[1][1] = (drwav_int32)drwav_bytes_to_s16(header + 8); + pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav_bytes_to_s16(header + 10); + pWav->msadpcm.prevFrames[1][0] = (drwav_int32)drwav_bytes_to_s16(header + 12); + + pWav->msadpcm.cachedFrames[0] = pWav->msadpcm.prevFrames[0][0]; + pWav->msadpcm.cachedFrames[1] = pWav->msadpcm.prevFrames[1][0]; + pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[1][1]; + pWav->msadpcm.cachedFrameCount = 2; + } + } + + /* Output anything that's cached. */ + while (framesToRead > 0 && pWav->msadpcm.cachedFrameCount > 0 && pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) { + if (pBufferOut != NULL) { + drwav_uint32 iSample = 0; + for (iSample = 0; iSample < pWav->channels; iSample += 1) { + pBufferOut[iSample] = (drwav_int16)pWav->msadpcm.cachedFrames[(drwav_countof(pWav->msadpcm.cachedFrames) - (pWav->msadpcm.cachedFrameCount*pWav->channels)) + iSample]; + } + + pBufferOut += pWav->channels; + } + + framesToRead -= 1; + totalFramesRead += 1; + pWav->readCursorInPCMFrames += 1; + pWav->msadpcm.cachedFrameCount -= 1; + } + + if (framesToRead == 0) { + break; + } + + + /* + If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + loop iteration which will trigger the loading of a new block. + */ + if (pWav->msadpcm.cachedFrameCount == 0) { + if (pWav->msadpcm.bytesRemainingInBlock == 0) { + continue; + } else { + static drwav_int32 adaptationTable[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 + }; + static drwav_int32 coeff1Table[] = { 256, 512, 0, 192, 240, 460, 392 }; + static drwav_int32 coeff2Table[] = { 0, -256, 0, 64, 0, -208, -232 }; + + drwav_uint8 nibbles; + drwav_int32 nibble0; + drwav_int32 nibble1; + + if (pWav->onRead(pWav->pUserData, &nibbles, 1) != 1) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock -= 1; + + /* TODO: Optimize away these if statements. */ + nibble0 = ((nibbles & 0xF0) >> 4); if ((nibbles & 0x80)) { nibble0 |= 0xFFFFFFF0UL; } + nibble1 = ((nibbles & 0x0F) >> 0); if ((nibbles & 0x08)) { nibble1 |= 0xFFFFFFF0UL; } + + if (pWav->channels == 1) { + /* Mono. */ + drwav_int32 newSample0; + drwav_int32 newSample1; + + newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample0; + + + newSample1 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[0]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample1; + + + pWav->msadpcm.cachedFrames[2] = newSample0; + pWav->msadpcm.cachedFrames[3] = newSample1; + pWav->msadpcm.cachedFrameCount = 2; + } else { + /* Stereo. */ + drwav_int32 newSample0; + drwav_int32 newSample1; + + /* Left. */ + newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample0; + + + /* Right. */ + newSample1 = ((pWav->msadpcm.prevFrames[1][1] * coeff1Table[pWav->msadpcm.predictor[1]]) + (pWav->msadpcm.prevFrames[1][0] * coeff2Table[pWav->msadpcm.predictor[1]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[1]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[1] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8; + if (pWav->msadpcm.delta[1] < 16) { + pWav->msadpcm.delta[1] = 16; + } + + pWav->msadpcm.prevFrames[1][0] = pWav->msadpcm.prevFrames[1][1]; + pWav->msadpcm.prevFrames[1][1] = newSample1; + + pWav->msadpcm.cachedFrames[2] = newSample0; + pWav->msadpcm.cachedFrames[3] = newSample1; + pWav->msadpcm.cachedFrameCount = 1; + } + } + } + } + + return totalFramesRead; +} + + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead = 0; + drwav_uint32 iChannel; + + static drwav_int32 indexTable[16] = { + -1, -1, -1, -1, 2, 4, 6, 8, + -1, -1, -1, -1, 2, 4, 6, 8 + }; + + static drwav_int32 stepTable[89] = { + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 + }; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(framesToRead > 0); + + /* TODO: Lots of room for optimization here. */ + + while (pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) { + DRWAV_ASSERT(framesToRead > 0); /* This loop iteration will never get hit with framesToRead == 0 because it's asserted at the top, and we check for 0 inside the loop just below. */ + + /* If there are no cached samples we need to load a new block. */ + if (pWav->ima.cachedFrameCount == 0 && pWav->ima.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + /* Mono. */ + drwav_uint8 header[4]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + if (header[2] >= drwav_countof(stepTable)) { + pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, drwav_seek_origin_current); + pWav->ima.bytesRemainingInBlock = 0; + return totalFramesRead; /* Invalid data. */ + } + + pWav->ima.predictor[0] = drwav_bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */ + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[0]; + pWav->ima.cachedFrameCount = 1; + } else { + /* Stereo. */ + drwav_uint8 header[8]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + if (header[2] >= drwav_countof(stepTable) || header[6] >= drwav_countof(stepTable)) { + pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, drwav_seek_origin_current); + pWav->ima.bytesRemainingInBlock = 0; + return totalFramesRead; /* Invalid data. */ + } + + pWav->ima.predictor[0] = drwav_bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */ + pWav->ima.predictor[1] = drwav_bytes_to_s16(header + 4); + pWav->ima.stepIndex[1] = drwav_clamp(header[6], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */ + + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 2] = pWav->ima.predictor[0]; + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[1]; + pWav->ima.cachedFrameCount = 1; + } + } + + /* Output anything that's cached. */ + while (framesToRead > 0 && pWav->ima.cachedFrameCount > 0 && pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) { + if (pBufferOut != NULL) { + drwav_uint32 iSample; + for (iSample = 0; iSample < pWav->channels; iSample += 1) { + pBufferOut[iSample] = (drwav_int16)pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + iSample]; + } + pBufferOut += pWav->channels; + } + + framesToRead -= 1; + totalFramesRead += 1; + pWav->readCursorInPCMFrames += 1; + pWav->ima.cachedFrameCount -= 1; + } + + if (framesToRead == 0) { + break; + } + + /* + If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + loop iteration which will trigger the loading of a new block. + */ + if (pWav->ima.cachedFrameCount == 0) { + if (pWav->ima.bytesRemainingInBlock == 0) { + continue; + } else { + /* + From what I can tell with stereo streams, it looks like every 4 bytes (8 samples) is for one channel. So it goes 4 bytes for the + left channel, 4 bytes for the right channel. + */ + pWav->ima.cachedFrameCount = 8; + for (iChannel = 0; iChannel < pWav->channels; ++iChannel) { + drwav_uint32 iByte; + drwav_uint8 nibbles[4]; + if (pWav->onRead(pWav->pUserData, &nibbles, 4) != 4) { + pWav->ima.cachedFrameCount = 0; + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock -= 4; + + for (iByte = 0; iByte < 4; ++iByte) { + drwav_uint8 nibble0 = ((nibbles[iByte] & 0x0F) >> 0); + drwav_uint8 nibble1 = ((nibbles[iByte] & 0xF0) >> 4); + + drwav_int32 step = stepTable[pWav->ima.stepIndex[iChannel]]; + drwav_int32 predictor = pWav->ima.predictor[iChannel]; + + drwav_int32 diff = step >> 3; + if (nibble0 & 1) diff += step >> 2; + if (nibble0 & 2) diff += step >> 1; + if (nibble0 & 4) diff += step; + if (nibble0 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble0], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+0)*pWav->channels + iChannel] = predictor; + + + step = stepTable[pWav->ima.stepIndex[iChannel]]; + predictor = pWav->ima.predictor[iChannel]; + + diff = step >> 3; + if (nibble1 & 1) diff += step >> 2; + if (nibble1 & 2) diff += step >> 1; + if (nibble1 & 4) diff += step; + if (nibble1 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble1], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+1)*pWav->channels + iChannel] = predictor; + } + } + } + } + } + + return totalFramesRead; +} + + +#ifndef DR_WAV_NO_CONVERSION_API +static unsigned short g_drwavAlawTable[256] = { + 0xEA80, 0xEB80, 0xE880, 0xE980, 0xEE80, 0xEF80, 0xEC80, 0xED80, 0xE280, 0xE380, 0xE080, 0xE180, 0xE680, 0xE780, 0xE480, 0xE580, + 0xF540, 0xF5C0, 0xF440, 0xF4C0, 0xF740, 0xF7C0, 0xF640, 0xF6C0, 0xF140, 0xF1C0, 0xF040, 0xF0C0, 0xF340, 0xF3C0, 0xF240, 0xF2C0, + 0xAA00, 0xAE00, 0xA200, 0xA600, 0xBA00, 0xBE00, 0xB200, 0xB600, 0x8A00, 0x8E00, 0x8200, 0x8600, 0x9A00, 0x9E00, 0x9200, 0x9600, + 0xD500, 0xD700, 0xD100, 0xD300, 0xDD00, 0xDF00, 0xD900, 0xDB00, 0xC500, 0xC700, 0xC100, 0xC300, 0xCD00, 0xCF00, 0xC900, 0xCB00, + 0xFEA8, 0xFEB8, 0xFE88, 0xFE98, 0xFEE8, 0xFEF8, 0xFEC8, 0xFED8, 0xFE28, 0xFE38, 0xFE08, 0xFE18, 0xFE68, 0xFE78, 0xFE48, 0xFE58, + 0xFFA8, 0xFFB8, 0xFF88, 0xFF98, 0xFFE8, 0xFFF8, 0xFFC8, 0xFFD8, 0xFF28, 0xFF38, 0xFF08, 0xFF18, 0xFF68, 0xFF78, 0xFF48, 0xFF58, + 0xFAA0, 0xFAE0, 0xFA20, 0xFA60, 0xFBA0, 0xFBE0, 0xFB20, 0xFB60, 0xF8A0, 0xF8E0, 0xF820, 0xF860, 0xF9A0, 0xF9E0, 0xF920, 0xF960, + 0xFD50, 0xFD70, 0xFD10, 0xFD30, 0xFDD0, 0xFDF0, 0xFD90, 0xFDB0, 0xFC50, 0xFC70, 0xFC10, 0xFC30, 0xFCD0, 0xFCF0, 0xFC90, 0xFCB0, + 0x1580, 0x1480, 0x1780, 0x1680, 0x1180, 0x1080, 0x1380, 0x1280, 0x1D80, 0x1C80, 0x1F80, 0x1E80, 0x1980, 0x1880, 0x1B80, 0x1A80, + 0x0AC0, 0x0A40, 0x0BC0, 0x0B40, 0x08C0, 0x0840, 0x09C0, 0x0940, 0x0EC0, 0x0E40, 0x0FC0, 0x0F40, 0x0CC0, 0x0C40, 0x0DC0, 0x0D40, + 0x5600, 0x5200, 0x5E00, 0x5A00, 0x4600, 0x4200, 0x4E00, 0x4A00, 0x7600, 0x7200, 0x7E00, 0x7A00, 0x6600, 0x6200, 0x6E00, 0x6A00, + 0x2B00, 0x2900, 0x2F00, 0x2D00, 0x2300, 0x2100, 0x2700, 0x2500, 0x3B00, 0x3900, 0x3F00, 0x3D00, 0x3300, 0x3100, 0x3700, 0x3500, + 0x0158, 0x0148, 0x0178, 0x0168, 0x0118, 0x0108, 0x0138, 0x0128, 0x01D8, 0x01C8, 0x01F8, 0x01E8, 0x0198, 0x0188, 0x01B8, 0x01A8, + 0x0058, 0x0048, 0x0078, 0x0068, 0x0018, 0x0008, 0x0038, 0x0028, 0x00D8, 0x00C8, 0x00F8, 0x00E8, 0x0098, 0x0088, 0x00B8, 0x00A8, + 0x0560, 0x0520, 0x05E0, 0x05A0, 0x0460, 0x0420, 0x04E0, 0x04A0, 0x0760, 0x0720, 0x07E0, 0x07A0, 0x0660, 0x0620, 0x06E0, 0x06A0, + 0x02B0, 0x0290, 0x02F0, 0x02D0, 0x0230, 0x0210, 0x0270, 0x0250, 0x03B0, 0x0390, 0x03F0, 0x03D0, 0x0330, 0x0310, 0x0370, 0x0350 +}; + +static unsigned short g_drwavMulawTable[256] = { + 0x8284, 0x8684, 0x8A84, 0x8E84, 0x9284, 0x9684, 0x9A84, 0x9E84, 0xA284, 0xA684, 0xAA84, 0xAE84, 0xB284, 0xB684, 0xBA84, 0xBE84, + 0xC184, 0xC384, 0xC584, 0xC784, 0xC984, 0xCB84, 0xCD84, 0xCF84, 0xD184, 0xD384, 0xD584, 0xD784, 0xD984, 0xDB84, 0xDD84, 0xDF84, + 0xE104, 0xE204, 0xE304, 0xE404, 0xE504, 0xE604, 0xE704, 0xE804, 0xE904, 0xEA04, 0xEB04, 0xEC04, 0xED04, 0xEE04, 0xEF04, 0xF004, + 0xF0C4, 0xF144, 0xF1C4, 0xF244, 0xF2C4, 0xF344, 0xF3C4, 0xF444, 0xF4C4, 0xF544, 0xF5C4, 0xF644, 0xF6C4, 0xF744, 0xF7C4, 0xF844, + 0xF8A4, 0xF8E4, 0xF924, 0xF964, 0xF9A4, 0xF9E4, 0xFA24, 0xFA64, 0xFAA4, 0xFAE4, 0xFB24, 0xFB64, 0xFBA4, 0xFBE4, 0xFC24, 0xFC64, + 0xFC94, 0xFCB4, 0xFCD4, 0xFCF4, 0xFD14, 0xFD34, 0xFD54, 0xFD74, 0xFD94, 0xFDB4, 0xFDD4, 0xFDF4, 0xFE14, 0xFE34, 0xFE54, 0xFE74, + 0xFE8C, 0xFE9C, 0xFEAC, 0xFEBC, 0xFECC, 0xFEDC, 0xFEEC, 0xFEFC, 0xFF0C, 0xFF1C, 0xFF2C, 0xFF3C, 0xFF4C, 0xFF5C, 0xFF6C, 0xFF7C, + 0xFF88, 0xFF90, 0xFF98, 0xFFA0, 0xFFA8, 0xFFB0, 0xFFB8, 0xFFC0, 0xFFC8, 0xFFD0, 0xFFD8, 0xFFE0, 0xFFE8, 0xFFF0, 0xFFF8, 0x0000, + 0x7D7C, 0x797C, 0x757C, 0x717C, 0x6D7C, 0x697C, 0x657C, 0x617C, 0x5D7C, 0x597C, 0x557C, 0x517C, 0x4D7C, 0x497C, 0x457C, 0x417C, + 0x3E7C, 0x3C7C, 0x3A7C, 0x387C, 0x367C, 0x347C, 0x327C, 0x307C, 0x2E7C, 0x2C7C, 0x2A7C, 0x287C, 0x267C, 0x247C, 0x227C, 0x207C, + 0x1EFC, 0x1DFC, 0x1CFC, 0x1BFC, 0x1AFC, 0x19FC, 0x18FC, 0x17FC, 0x16FC, 0x15FC, 0x14FC, 0x13FC, 0x12FC, 0x11FC, 0x10FC, 0x0FFC, + 0x0F3C, 0x0EBC, 0x0E3C, 0x0DBC, 0x0D3C, 0x0CBC, 0x0C3C, 0x0BBC, 0x0B3C, 0x0ABC, 0x0A3C, 0x09BC, 0x093C, 0x08BC, 0x083C, 0x07BC, + 0x075C, 0x071C, 0x06DC, 0x069C, 0x065C, 0x061C, 0x05DC, 0x059C, 0x055C, 0x051C, 0x04DC, 0x049C, 0x045C, 0x041C, 0x03DC, 0x039C, + 0x036C, 0x034C, 0x032C, 0x030C, 0x02EC, 0x02CC, 0x02AC, 0x028C, 0x026C, 0x024C, 0x022C, 0x020C, 0x01EC, 0x01CC, 0x01AC, 0x018C, + 0x0174, 0x0164, 0x0154, 0x0144, 0x0134, 0x0124, 0x0114, 0x0104, 0x00F4, 0x00E4, 0x00D4, 0x00C4, 0x00B4, 0x00A4, 0x0094, 0x0084, + 0x0078, 0x0070, 0x0068, 0x0060, 0x0058, 0x0050, 0x0048, 0x0040, 0x0038, 0x0030, 0x0028, 0x0020, 0x0018, 0x0010, 0x0008, 0x0000 +}; + +static DRWAV_INLINE drwav_int16 drwav__alaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavAlawTable[sampleIn]; +} + +static DRWAV_INLINE drwav_int16 drwav__mulaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavMulawTable[sampleIn]; +} + + + +DRWAV_PRIVATE void drwav__pcm_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + size_t i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_s16(pOut, pIn, totalSampleCount); + return; + } + + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + for (i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((const drwav_int16*)pIn)[i]; + } + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s16(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_s16(pOut, (const drwav_int32*)pIn, totalSampleCount); + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (drwav_int16)((drwav_int64)sample >> 48); + } +} + +DRWAV_PRIVATE void drwav__ieee_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s16(pOut, (const float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s16(pOut, (const double*)pIn, totalSampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + /* Fast path. */ + if ((pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 16) || pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); /* Safe cast. */ + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(drwav_int16) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int16) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_s16__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_s16__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_s16__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_s16__mulaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_s16__ima(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x << 8; + r = r - 32768; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = ((int)(((unsigned int)(((const drwav_uint8*)pIn)[i*3+0]) << 8) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+1]) << 16) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+2])) << 24)) >> 8; + r = x >> 8; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x >> 16; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + float c; + c = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); + c = c + 1; + r = (int)(c * 32767.5f); + r = r - 32768; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + double x = pIn[i]; + double c; + c = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); + c = c + 1; + r = (int)(c * 32767.5); + r = r - 32768; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__alaw_to_s16(pIn[i]); + } +} + +DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__mulaw_to_s16(pIn[i]); + } +} + + + +DRWAV_PRIVATE void drwav__pcm_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_f32(pOut, pIn, sampleCount); + return; + } + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + drwav_s16_to_f32(pOut, (const drwav_int16*)pIn, sampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_f32(pOut, pIn, sampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_f32(pOut, (const drwav_int32*)pIn, sampleCount); + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < sampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (float)((drwav_int64)sample / 9223372036854775807.0); + } +} + +DRWAV_PRIVATE void drwav__ieee_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + unsigned int i; + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((const float*)pIn)[i]; + } + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_f32(pOut, (const double*)pIn, sampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut)); + return; + } +} + + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__pcm(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalFramesRead; + drwav_int16 samples16[2048]; + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels); + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + drwav_s16_to_f32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ieee(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT && pWav->bitsPerSample == 32) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__alaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__mulaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(float) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(float) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_f32__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_f32__msadpcm_ima(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_f32__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_f32__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_f32__mulaw(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + /* + It appears libsndfile uses slightly different logic for the u8 -> f32 conversion to dr_wav, which in my opinion is incorrect. It appears + libsndfile performs the conversion something like "f32 = (u8 / 256) * 2 - 1", however I think it should be "f32 = (u8 / 255) * 2 - 1" (note + the divisor of 256 vs 255). I use libsndfile as a benchmark for testing, so I'm therefore leaving this block here just for my automated + correctness testing. This is disabled by default. + */ + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (pIn[i] / 256.0f) * 2 - 1; + } +#else + for (i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */ + x = x - 1; /* 0..2 to -1..1 */ + + *pOut++ = x; + } +#endif +} + +DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] * 0.000030517578125f; + } +} + +DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + double x; + drwav_uint32 a = ((drwav_uint32)(pIn[i*3+0]) << 8); + drwav_uint32 b = ((drwav_uint32)(pIn[i*3+1]) << 16); + drwav_uint32 c = ((drwav_uint32)(pIn[i*3+2]) << 24); + + x = (double)((drwav_int32)(a | b | c) >> 8); + *pOut++ = (float)(x * 0.00000011920928955078125); + } +} + +DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + size_t i; + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (float)(pIn[i] / 2147483648.0); + } +} + +DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (float)pIn[i]; + } +} + +DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__alaw_to_s16(pIn[i]) / 32768.0f; + } +} + +DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__mulaw_to_s16(pIn[i]) / 32768.0f; + } +} + + + +DRWAV_PRIVATE void drwav__pcm_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_s32(pOut, pIn, totalSampleCount); + return; + } + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + drwav_s16_to_s32(pOut, (const drwav_int16*)pIn, totalSampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s32(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + for (i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((const drwav_int32*)pIn)[i]; + } + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (drwav_int32)((drwav_int64)sample >> 32); + } +} + +DRWAV_PRIVATE void drwav__ieee_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s32(pOut, (const float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s32(pOut, (const double*)pIn, totalSampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 32) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalFramesRead = 0; + drwav_int16 samples16[2048]; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels); + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + drwav_s16_to_s32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(drwav_int32) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int32) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_s32__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_s32__msadpcm_ima(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_s32__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_s32__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_s32__mulaw(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((int)pIn[i] - 128) << 24; + } +} + +DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] << 16; + } +} + +DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + unsigned int s0 = pIn[i*3 + 0]; + unsigned int s1 = pIn[i*3 + 1]; + unsigned int s2 = pIn[i*3 + 2]; + + drwav_int32 sample32 = (drwav_int32)((s0 << 8) | (s1 << 16) | (s2 << 24)); + *pOut++ = sample32; + } +} + +DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]); + } +} + +DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]); + } +} + +DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((drwav_int32)drwav__alaw_to_s16(pIn[i])) << 16; + } +} + +DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i= 0; i < sampleCount; ++i) { + *pOut++ = ((drwav_int32)drwav__mulaw_to_s16(pIn[i])) << 16; + } +} + + + +DRWAV_PRIVATE drwav_int16* drwav__read_pcm_frames_and_close_s16(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + drwav_int16* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int16); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (drwav_int16*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_s16(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + +DRWAV_PRIVATE float* drwav__read_pcm_frames_and_close_f32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + float* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(float); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (float*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_f32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + +DRWAV_PRIVATE drwav_int32* drwav__read_pcm_frames_and_close_s32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + drwav_int32* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int32); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (drwav_int32*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_s32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + + + +DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +#ifndef DR_WAV_NO_STDIO +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + + +#ifndef DR_WAV_NO_WCHAR +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} +#endif /* DR_WAV_NO_WCHAR */ +#endif /* DR_WAV_NO_STDIO */ + +DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} +#endif /* DR_WAV_NO_CONVERSION_API */ + + +DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + drwav__free_from_callbacks(p, pAllocationCallbacks); + } else { + drwav__free_default(p, NULL); + } +} + +DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data) +{ + return ((drwav_uint16)data[0] << 0) | ((drwav_uint16)data[1] << 8); +} + +DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data) +{ + return (drwav_int16)drwav_bytes_to_u16(data); +} + +DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data) +{ + return ((drwav_uint32)data[0] << 0) | ((drwav_uint32)data[1] << 8) | ((drwav_uint32)data[2] << 16) | ((drwav_uint32)data[3] << 24); +} + +DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data) +{ + union { + drwav_uint32 u32; + float f32; + } value; + + value.u32 = drwav_bytes_to_u32(data); + return value.f32; +} + +DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data) +{ + return (drwav_int32)drwav_bytes_to_u32(data); +} + +DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data) +{ + return + ((drwav_uint64)data[0] << 0) | ((drwav_uint64)data[1] << 8) | ((drwav_uint64)data[2] << 16) | ((drwav_uint64)data[3] << 24) | + ((drwav_uint64)data[4] << 32) | ((drwav_uint64)data[5] << 40) | ((drwav_uint64)data[6] << 48) | ((drwav_uint64)data[7] << 56); +} + +DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data) +{ + return (drwav_int64)drwav_bytes_to_u64(data); +} + + +DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]) +{ + int i; + for (i = 0; i < 16; i += 1) { + if (a[i] != b[i]) { + return DRWAV_FALSE; + } + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b) +{ + return + a[0] == b[0] && + a[1] == b[1] && + a[2] == b[2] && + a[3] == b[3]; +} + +#ifdef __MRC__ +/* Undo the pragma at the beginning of this file. */ +#pragma options opt reset +#endif + +#endif /* dr_wav_c */ +#endif /* DR_WAV_IMPLEMENTATION */ + +/* +REVISION HISTORY +================ +v0.13.7 - 2022-09-17 + - Fix compilation with DJGPP. + - Add support for disabling wchar_t with DR_WAV_NO_WCHAR. + +v0.13.6 - 2022-04-10 + - Fix compilation error on older versions of GCC. + - Remove some dependencies on the standard library. + +v0.13.5 - 2022-01-26 + - Fix an error when seeking to the end of the file. + +v0.13.4 - 2021-12-08 + - Fix some static analysis warnings. + +v0.13.3 - 2021-11-24 + - Fix an incorrect assertion when trying to endian swap 1-byte sample formats. This is now a no-op + rather than a failed assertion. + - Fix a bug with parsing of the bext chunk. + - Fix some static analysis warnings. + +v0.13.2 - 2021-10-02 + - Fix a possible buffer overflow when reading from compressed formats. + +v0.13.1 - 2021-07-31 + - Fix platform detection for ARM64. + +v0.13.0 - 2021-07-01 + - Improve support for reading and writing metadata. Use the `_with_metadata()` APIs to initialize + a WAV decoder and store the metadata within the `drwav` object. Use the `pMetadata` and + `metadataCount` members of the `drwav` object to read the data. The old way of handling metadata + via a callback is still usable and valid. + - API CHANGE: drwav_target_write_size_bytes() now takes extra parameters for calculating the + required write size when writing metadata. + - Add drwav_get_cursor_in_pcm_frames() + - Add drwav_get_length_in_pcm_frames() + - Fix a bug where drwav_read_raw() can call the read callback with a byte count of zero. + +v0.12.20 - 2021-06-11 + - Fix some undefined behavior. + +v0.12.19 - 2021-02-21 + - Fix a warning due to referencing _MSC_VER when it is undefined. + - Minor improvements to the management of some internal state concerning the data chunk cursor. + +v0.12.18 - 2021-01-31 + - Clean up some static analysis warnings. + +v0.12.17 - 2021-01-17 + - Minor fix to sample code in documentation. + - Correctly qualify a private API as private rather than public. + - Code cleanup. + +v0.12.16 - 2020-12-02 + - Fix a bug when trying to read more bytes than can fit in a size_t. + +v0.12.15 - 2020-11-21 + - Fix compilation with OpenWatcom. + +v0.12.14 - 2020-11-13 + - Minor code clean up. + +v0.12.13 - 2020-11-01 + - Improve compiler support for older versions of GCC. + +v0.12.12 - 2020-09-28 + - Add support for RF64. + - Fix a bug in writing mode where the size of the RIFF chunk incorrectly includes the header section. + +v0.12.11 - 2020-09-08 + - Fix a compilation error on older compilers. + +v0.12.10 - 2020-08-24 + - Fix a bug when seeking with ADPCM formats. + +v0.12.9 - 2020-08-02 + - Simplify sized types. + +v0.12.8 - 2020-07-25 + - Fix a compilation warning. + +v0.12.7 - 2020-07-15 + - Fix some bugs on big-endian architectures. + - Fix an error in s24 to f32 conversion. + +v0.12.6 - 2020-06-23 + - Change drwav_read_*() to allow NULL to be passed in as the output buffer which is equivalent to a forward seek. + - Fix a buffer overflow when trying to decode invalid IMA-ADPCM files. + - Add include guard for the implementation section. + +v0.12.5 - 2020-05-27 + - Minor documentation fix. + +v0.12.4 - 2020-05-16 + - Replace assert() with DRWAV_ASSERT(). + - Add compile-time and run-time version querying. + - DRWAV_VERSION_MINOR + - DRWAV_VERSION_MAJOR + - DRWAV_VERSION_REVISION + - DRWAV_VERSION_STRING + - drwav_version() + - drwav_version_string() + +v0.12.3 - 2020-04-30 + - Fix compilation errors with VC6. + +v0.12.2 - 2020-04-21 + - Fix a bug where drwav_init_file() does not close the file handle after attempting to load an erroneous file. + +v0.12.1 - 2020-04-13 + - Fix some pedantic warnings. + +v0.12.0 - 2020-04-04 + - API CHANGE: Add container and format parameters to the chunk callback. + - Minor documentation updates. + +v0.11.5 - 2020-03-07 + - Fix compilation error with Visual Studio .NET 2003. + +v0.11.4 - 2020-01-29 + - Fix some static analysis warnings. + - Fix a bug when reading f32 samples from an A-law encoded stream. + +v0.11.3 - 2020-01-12 + - Minor changes to some f32 format conversion routines. + - Minor bug fix for ADPCM conversion when end of file is reached. + +v0.11.2 - 2019-12-02 + - Fix a possible crash when using custom memory allocators without a custom realloc() implementation. + - Fix an integer overflow bug. + - Fix a null pointer dereference bug. + - Add limits to sample rate, channels and bits per sample to tighten up some validation. + +v0.11.1 - 2019-10-07 + - Internal code clean up. + +v0.11.0 - 2019-10-06 + - API CHANGE: Add support for user defined memory allocation routines. This system allows the program to specify their own memory allocation + routines with a user data pointer for client-specific contextual data. This adds an extra parameter to the end of the following APIs: + - drwav_init() + - drwav_init_ex() + - drwav_init_file() + - drwav_init_file_ex() + - drwav_init_file_w() + - drwav_init_file_w_ex() + - drwav_init_memory() + - drwav_init_memory_ex() + - drwav_init_write() + - drwav_init_write_sequential() + - drwav_init_write_sequential_pcm_frames() + - drwav_init_file_write() + - drwav_init_file_write_sequential() + - drwav_init_file_write_sequential_pcm_frames() + - drwav_init_file_write_w() + - drwav_init_file_write_sequential_w() + - drwav_init_file_write_sequential_pcm_frames_w() + - drwav_init_memory_write() + - drwav_init_memory_write_sequential() + - drwav_init_memory_write_sequential_pcm_frames() + - drwav_open_and_read_pcm_frames_s16() + - drwav_open_and_read_pcm_frames_f32() + - drwav_open_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_s16() + - drwav_open_file_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_s16_w() + - drwav_open_file_and_read_pcm_frames_f32_w() + - drwav_open_file_and_read_pcm_frames_s32_w() + - drwav_open_memory_and_read_pcm_frames_s16() + - drwav_open_memory_and_read_pcm_frames_f32() + - drwav_open_memory_and_read_pcm_frames_s32() + Set this extra parameter to NULL to use defaults which is the same as the previous behaviour. Setting this NULL will use + DRWAV_MALLOC, DRWAV_REALLOC and DRWAV_FREE. + - Add support for reading and writing PCM frames in an explicit endianness. New APIs: + - drwav_read_pcm_frames_le() + - drwav_read_pcm_frames_be() + - drwav_read_pcm_frames_s16le() + - drwav_read_pcm_frames_s16be() + - drwav_read_pcm_frames_f32le() + - drwav_read_pcm_frames_f32be() + - drwav_read_pcm_frames_s32le() + - drwav_read_pcm_frames_s32be() + - drwav_write_pcm_frames_le() + - drwav_write_pcm_frames_be() + - Remove deprecated APIs. + - API CHANGE: The following APIs now return native-endian data. Previously they returned little-endian data. + - drwav_read_pcm_frames() + - drwav_read_pcm_frames_s16() + - drwav_read_pcm_frames_s32() + - drwav_read_pcm_frames_f32() + - drwav_open_and_read_pcm_frames_s16() + - drwav_open_and_read_pcm_frames_s32() + - drwav_open_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s16() + - drwav_open_file_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s16_w() + - drwav_open_file_and_read_pcm_frames_s32_w() + - drwav_open_file_and_read_pcm_frames_f32_w() + - drwav_open_memory_and_read_pcm_frames_s16() + - drwav_open_memory_and_read_pcm_frames_s32() + - drwav_open_memory_and_read_pcm_frames_f32() + +v0.10.1 - 2019-08-31 + - Correctly handle partial trailing ADPCM blocks. + +v0.10.0 - 2019-08-04 + - Remove deprecated APIs. + - Add wchar_t variants for file loading APIs: + drwav_init_file_w() + drwav_init_file_ex_w() + drwav_init_file_write_w() + drwav_init_file_write_sequential_w() + - Add drwav_target_write_size_bytes() which calculates the total size in bytes of a WAV file given a format and sample count. + - Add APIs for specifying the PCM frame count instead of the sample count when opening in sequential write mode: + drwav_init_write_sequential_pcm_frames() + drwav_init_file_write_sequential_pcm_frames() + drwav_init_file_write_sequential_pcm_frames_w() + drwav_init_memory_write_sequential_pcm_frames() + - Deprecate drwav_open*() and drwav_close(): + drwav_open() + drwav_open_ex() + drwav_open_write() + drwav_open_write_sequential() + drwav_open_file() + drwav_open_file_ex() + drwav_open_file_write() + drwav_open_file_write_sequential() + drwav_open_memory() + drwav_open_memory_ex() + drwav_open_memory_write() + drwav_open_memory_write_sequential() + drwav_close() + - Minor documentation updates. + +v0.9.2 - 2019-05-21 + - Fix warnings. + +v0.9.1 - 2019-05-05 + - Add support for C89. + - Change license to choice of public domain or MIT-0. + +v0.9.0 - 2018-12-16 + - API CHANGE: Add new reading APIs for reading by PCM frames instead of samples. Old APIs have been deprecated and + will be removed in v0.10.0. Deprecated APIs and their replacements: + drwav_read() -> drwav_read_pcm_frames() + drwav_read_s16() -> drwav_read_pcm_frames_s16() + drwav_read_f32() -> drwav_read_pcm_frames_f32() + drwav_read_s32() -> drwav_read_pcm_frames_s32() + drwav_seek_to_sample() -> drwav_seek_to_pcm_frame() + drwav_write() -> drwav_write_pcm_frames() + drwav_open_and_read_s16() -> drwav_open_and_read_pcm_frames_s16() + drwav_open_and_read_f32() -> drwav_open_and_read_pcm_frames_f32() + drwav_open_and_read_s32() -> drwav_open_and_read_pcm_frames_s32() + drwav_open_file_and_read_s16() -> drwav_open_file_and_read_pcm_frames_s16() + drwav_open_file_and_read_f32() -> drwav_open_file_and_read_pcm_frames_f32() + drwav_open_file_and_read_s32() -> drwav_open_file_and_read_pcm_frames_s32() + drwav_open_memory_and_read_s16() -> drwav_open_memory_and_read_pcm_frames_s16() + drwav_open_memory_and_read_f32() -> drwav_open_memory_and_read_pcm_frames_f32() + drwav_open_memory_and_read_s32() -> drwav_open_memory_and_read_pcm_frames_s32() + drwav::totalSampleCount -> drwav::totalPCMFrameCount + - API CHANGE: Rename drwav_open_and_read_file_*() to drwav_open_file_and_read_*(). + - API CHANGE: Rename drwav_open_and_read_memory_*() to drwav_open_memory_and_read_*(). + - Add built-in support for smpl chunks. + - Add support for firing a callback for each chunk in the file at initialization time. + - This is enabled through the drwav_init_ex(), etc. family of APIs. + - Handle invalid FMT chunks more robustly. + +v0.8.5 - 2018-09-11 + - Const correctness. + - Fix a potential stack overflow. + +v0.8.4 - 2018-08-07 + - Improve 64-bit detection. + +v0.8.3 - 2018-08-05 + - Fix C++ build on older versions of GCC. + +v0.8.2 - 2018-08-02 + - Fix some big-endian bugs. + +v0.8.1 - 2018-06-29 + - Add support for sequential writing APIs. + - Disable seeking in write mode. + - Fix bugs with Wave64. + - Fix typos. + +v0.8 - 2018-04-27 + - Bug fix. + - Start using major.minor.revision versioning. + +v0.7f - 2018-02-05 + - Restrict ADPCM formats to a maximum of 2 channels. + +v0.7e - 2018-02-02 + - Fix a crash. + +v0.7d - 2018-02-01 + - Fix a crash. + +v0.7c - 2018-02-01 + - Set drwav.bytesPerSample to 0 for all compressed formats. + - Fix a crash when reading 16-bit floating point WAV files. In this case dr_wav will output silence for + all format conversion reading APIs (*_s16, *_s32, *_f32 APIs). + - Fix some divide-by-zero errors. + +v0.7b - 2018-01-22 + - Fix errors with seeking of compressed formats. + - Fix compilation error when DR_WAV_NO_CONVERSION_API + +v0.7a - 2017-11-17 + - Fix some GCC warnings. + +v0.7 - 2017-11-04 + - Add writing APIs. + +v0.6 - 2017-08-16 + - API CHANGE: Rename dr_* types to drwav_*. + - Add support for custom implementations of malloc(), realloc(), etc. + - Add support for Microsoft ADPCM. + - Add support for IMA ADPCM (DVI, format code 0x11). + - Optimizations to drwav_read_s16(). + - Bug fixes. + +v0.5g - 2017-07-16 + - Change underlying type for booleans to unsigned. + +v0.5f - 2017-04-04 + - Fix a minor bug with drwav_open_and_read_s16() and family. + +v0.5e - 2016-12-29 + - Added support for reading samples as signed 16-bit integers. Use the _s16() family of APIs for this. + - Minor fixes to documentation. + +v0.5d - 2016-12-28 + - Use drwav_int* and drwav_uint* sized types to improve compiler support. + +v0.5c - 2016-11-11 + - Properly handle JUNK chunks that come before the FMT chunk. + +v0.5b - 2016-10-23 + - A minor change to drwav_bool8 and drwav_bool32 types. + +v0.5a - 2016-10-11 + - Fixed a bug with drwav_open_and_read() and family due to incorrect argument ordering. + - Improve A-law and mu-law efficiency. + +v0.5 - 2016-09-29 + - API CHANGE. Swap the order of "channels" and "sampleRate" parameters in drwav_open_and_read*(). Rationale for this is to + keep it consistent with dr_audio and dr_flac. + +v0.4b - 2016-09-18 + - Fixed a typo in documentation. + +v0.4a - 2016-09-18 + - Fixed a typo. + - Change date format to ISO 8601 (YYYY-MM-DD) + +v0.4 - 2016-07-13 + - API CHANGE. Make onSeek consistent with dr_flac. + - API CHANGE. Rename drwav_seek() to drwav_seek_to_sample() for clarity and consistency with dr_flac. + - Added support for Sony Wave64. + +v0.3a - 2016-05-28 + - API CHANGE. Return drwav_bool32 instead of int in onSeek callback. + - Fixed a memory leak. + +v0.3 - 2016-05-22 + - Lots of API changes for consistency. + +v0.2a - 2016-05-16 + - Fixed Linux/GCC build. + +v0.2 - 2016-05-11 + - Added support for reading data as signed 32-bit PCM for consistency with dr_flac. + +v0.1a - 2016-05-07 + - Fixed a bug in drwav_open_file() where the file handle would not be closed if the loader failed to initialize. + +v0.1 - 2016-05-04 + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2020 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ diff --git a/thirdparty/sokol_audio.h b/thirdparty/sokol_audio.h index beb4a7c..bdd6953 100644 --- a/thirdparty/sokol_audio.h +++ b/thirdparty/sokol_audio.h @@ -132,18 +132,30 @@ a good balance between low-latency and glitch-free playback on all audio backends. + You should always provide a logging callback to be aware of any + warnings and errors. The easiest way is to use sokol_log.h for this: + + #include "sokol_log.h" + // ... + saudio_setup(&(saudio_desc){ + .logger = { + .func = slog_func, + } + }); + If you want to use the callback-model, you need to provide a stream callback function either in saudio_desc.stream_cb or saudio_desc.stream_userdata_cb, otherwise keep both function pointers zero-initialized. Use push model and default playback parameters: - saudio_setup(&(saudio_desc){0}); + saudio_setup(&(saudio_desc){ .logger.func = slog_func }); Use stream callback model and default playback parameters: saudio_setup(&(saudio_desc){ .stream_cb = my_stream_callback + .logger.func = slog_func, }); The standard stream callback doesn't have a user data argument, if you want @@ -152,6 +164,7 @@ saudio_setup(&(saudio_desc){ .stream_userdata_cb = my_stream_callback, .user_data = &my_data + .logger.func = slog_func, }); The following playback parameters can be provided through the @@ -400,27 +413,43 @@ was called, so you don't need to worry about thread-safety. - LOG FUNCTION OVERRIDE - ===================== - You can override the log function at initialization time like this: + ERROR REPORTING AND LOGGING + =========================== + To get any logging information at all you need to provide a logging callback in the setup call + the easiest way is to use sokol_log.h: + + #include "sokol_log.h" + + saudio_setup(&(saudio_desc){ .logger.func = slog_func }); + + To override logging with your own callback, first write a logging function like this: - void my_log(const char* message, void* user_data) { - printf("saudio says: \s\n", message); + void my_log(const char* tag, // e.g. 'saudio' + uint32_t log_level, // 0=panic, 1=error, 2=warn, 3=info + uint32_t log_item_id, // SAUDIO_LOGITEM_* + const char* message_or_null, // a message string, may be nullptr in release mode + uint32_t line_nr, // line number in sokol_audio.h + const char* filename_or_null, // source filename, may be nullptr in release mode + void* user_data) + { + ... } - ... - saudio_setup(&(saudio_desc){ - // ... - .logger = { - .log_cb = my_log, - .user_data = ..., - } - }); - ... + ...and then setup sokol-audio like this: - If no overrides are provided, puts will be used on most platforms. - On Android, __android_log_write will be used instead. + saudio_setup(&(saudio_desc){ + .logger = { + .func = my_log, + .user_data = my_user_data, + } + }); + + The provided logging function must be reentrant (e.g. be callable from + different threads). + If you don't want to provide your own custom logger it is highly recommended to use + the standard logger in sokol_log.h instead, otherwise you won't see any warnings or + errors. LICENSE ======= @@ -470,6 +499,77 @@ extern "C" { #endif +/* + saudio_log_item + + Log items are defined via X-Macros, and expanded to an + enum 'saudio_log_item', and in debug mode only, + corresponding strings. + + Used as parameter in the logging callback. +*/ +#define _SAUDIO_LOG_ITEMS \ + _SAUDIO_LOGITEM_XMACRO(OK, "Ok") \ + _SAUDIO_LOGITEM_XMACRO(MALLOC_FAILED, "memory allocation failed") \ + _SAUDIO_LOGITEM_XMACRO(ALSA_SND_PCM_OPEN_FAILED, "snd_pcm_open() failed") \ + _SAUDIO_LOGITEM_XMACRO(ALSA_FLOAT_SAMPLES_NOT_SUPPORTED, "floating point sample format not supported") \ + _SAUDIO_LOGITEM_XMACRO(ALSA_REQUESTED_BUFFER_SIZE_NOT_SUPPORTED, "requested buffer size not supported") \ + _SAUDIO_LOGITEM_XMACRO(ALSA_REQUESTED_CHANNEL_COUNT_NOT_SUPPORTED, "requested channel count not supported") \ + _SAUDIO_LOGITEM_XMACRO(ALSA_SND_PCM_HW_PARAMS_SET_RATE_NEAR_FAILED, "snd_pcm_hw_params_set_rate_near() failed") \ + _SAUDIO_LOGITEM_XMACRO(ALSA_SND_PCM_HW_PARAMS_FAILED, "snd_pcm_hw_params() failed") \ + _SAUDIO_LOGITEM_XMACRO(ALSA_PTHREAD_CREATE_FAILED, "pthread_create() failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_CREATE_EVENT_FAILED, "CreateEvent() failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_CREATE_DEVICE_ENUMERATOR_FAILED, "CoCreateInstance() for IMMDeviceEnumerator failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_GET_DEFAULT_AUDIO_ENDPOINT_FAILED, "IMMDeviceEnumerator.GetDefaultAudioEndpoint() failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_DEVICE_ACTIVATE_FAILED, "IMMDevice.Activate() failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_AUDIO_CLIENT_INITIALIZE_FAILED, "IAudioClient.Initialize() failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_AUDIO_CLIENT_GET_BUFFER_SIZE_FAILED, "IAudioClient.GetBufferSize() failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_AUDIO_CLIENT_GET_SERVICE_FAILED, "IAudioClient.GetService() failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_AUDIO_CLIENT_SET_EVENT_HANDLE_FAILED, "IAudioClient.SetEventHandle() failed") \ + _SAUDIO_LOGITEM_XMACRO(WASAPI_CREATE_THREAD_FAILED, "CreateThread() failed") \ + _SAUDIO_LOGITEM_XMACRO(AAUDIO_STREAMBUILDER_OPEN_STREAM_FAILED, "AAudioStreamBuilder_openStream() failed") \ + _SAUDIO_LOGITEM_XMACRO(AAUDIO_PTHREAD_CREATE_FAILED, "pthread_create() failed after AAUDIO_ERROR_DISCONNECTED") \ + _SAUDIO_LOGITEM_XMACRO(AAUDIO_RESTARTING_STREAM_AFTER_ERROR, "restarting AAudio stream after error") \ + _SAUDIO_LOGITEM_XMACRO(USING_AAUDIO_BACKEND, "using AAudio backend") \ + _SAUDIO_LOGITEM_XMACRO(AAUDIO_CREATE_STREAMBUILDER_FAILED, "AAudio_createStreamBuilder() failed") \ + _SAUDIO_LOGITEM_XMACRO(USING_SLES_BACKEND, "using OpenSLES backend") \ + _SAUDIO_LOGITEM_XMACRO(SLES_CREATE_ENGINE_FAILED, "slCreateEngine() failed") \ + _SAUDIO_LOGITEM_XMACRO(SLES_ENGINE_GET_ENGINE_INTERFACE_FAILED, "GetInterface() for SL_IID_ENGINE failed") \ + _SAUDIO_LOGITEM_XMACRO(SLES_CREATE_OUTPUT_MIX_FAILED, "CreateOutputMix() failed") \ + _SAUDIO_LOGITEM_XMACRO(SLES_MIXER_GET_VOLUME_INTERFACE_FAILED, "GetInterface() for SL_IID_VOLUME failed") \ + _SAUDIO_LOGITEM_XMACRO(SLES_ENGINE_CREATE_AUDIO_PLAYER_FAILED, "CreateAudioPlayer() failed") \ + _SAUDIO_LOGITEM_XMACRO(SLES_PLAYER_GET_PLAY_INTERFACE_FAILED, "GetInterface() for SL_IID_PLAY failed") \ + _SAUDIO_LOGITEM_XMACRO(SLES_PLAYER_GET_VOLUME_INTERFACE_FAILED, "GetInterface() for SL_IID_VOLUME failed") \ + _SAUDIO_LOGITEM_XMACRO(SLES_PLAYER_GET_BUFFERQUEUE_INTERFACE_FAILED, "GetInterface() for SL_IID_ANDROIDSIMPLEBUFFERQUEUE failed") \ + _SAUDIO_LOGITEM_XMACRO(COREAUDIO_NEW_OUTPUT_FAILED, "AudioQueueNewOutput() failed") \ + _SAUDIO_LOGITEM_XMACRO(COREAUDIO_ALLOCATE_BUFFER_FAILED, "AudioQueueAllocateBuffer() failed") \ + _SAUDIO_LOGITEM_XMACRO(COREAUDIO_START_FAILED, "AudioQueueStart() failed") \ + _SAUDIO_LOGITEM_XMACRO(BACKEND_BUFFER_SIZE_ISNT_MULTIPLE_OF_PACKET_SIZE, "backend buffer size isn't multiple of packet size") \ + +#define _SAUDIO_LOGITEM_XMACRO(item,msg) SAUDIO_LOGITEM_##item, +typedef enum saudio_log_item { + _SAUDIO_LOG_ITEMS +} saudio_log_item; +#undef _SAUDIO_LOGITEM_XMACRO + +/* + saudio_logger + + Used in saudio_desc to provide a custom logging and error reporting + callback to sokol-audio. +*/ +typedef struct saudio_logger { + void (*func)( + const char* tag, // always "saudio" + uint32_t log_level, // 0=panic, 1=error, 2=warning, 3=info + uint32_t log_item_id, // SAUDIO_LOGITEM_* + const char* message_or_null, // a message string, may be nullptr in release mode + uint32_t line_nr, // line number in sokol_audio.h + const char* filename_or_null, // source filename, may be nullptr in release mode + void* user_data); + void* user_data; +} saudio_logger; + /* saudio_allocator @@ -484,17 +584,6 @@ typedef struct saudio_allocator { void* user_data; } saudio_allocator; -/* - saudio_logger - - Used in saudio_desc to provide custom log callbacks to sokol_audio.h. - Default behavior is SOKOL_LOG(message). -*/ -typedef struct saudio_logger { - void (*log_cb)(const char* message, void* user_data); - void* user_data; -} saudio_logger; - typedef struct saudio_desc { int sample_rate; // requested sample rate int num_channels; // number of channels, default: 1 (mono) @@ -505,7 +594,7 @@ typedef struct saudio_desc { void (*stream_userdata_cb)(float* buffer, int num_frames, int num_channels, void* user_data); //... and with user data void* user_data; // optional user data argument for stream_userdata_cb saudio_allocator allocator; // optional allocation override functions - saudio_logger logger; // optional log override functions + saudio_logger logger; // optional logging function (default: NO LOGGING!) } saudio_desc; /* setup sokol-audio */ @@ -540,12 +629,11 @@ inline void saudio_setup(const saudio_desc& desc) { return saudio_setup(&desc); #endif #endif // SOKOL_AUDIO_INCLUDED -// ██╗███╗ ███╗██████╗ ██╗ ███████╗███╗ ███╗███████╗███╗ ██╗████████╗ █████╗ ████████╗██╗ ██████╗ ███╗ ██╗ -// ██║████╗ ████║██╔══██╗██║ ██╔════╝████╗ ████║██╔════╝████╗ ██║╚══██╔══╝██╔══██╗╚══██╔══╝██║██╔═══██╗████╗ ██║ -// ██║██╔████╔██║██████╔╝██║ █████╗ ██╔████╔██║█████╗ ██╔██╗ ██║ ██║ ███████║ ██║ ██║██║ ██║██╔██╗ ██║ -// ██║██║╚██╔╝██║██╔═══╝ ██║ ██╔══╝ ██║╚██╔╝██║██╔══╝ ██║╚██╗██║ ██║ ██╔══██║ ██║ ██║██║ ██║██║╚██╗██║ -// ██║██║ ╚═╝ ██║██║ ███████╗███████╗██║ ╚═╝ ██║███████╗██║ ╚████║ ██║ ██║ ██║ ██║ ██║╚██████╔╝██║ ╚████║ -// ╚═╝╚═╝ ╚═╝╚═╝ ╚══════╝╚══════╝╚═╝ ╚═╝╚══════╝╚═╝ ╚═══╝ ╚═╝ ╚═╝ ╚═╝ ╚═╝ ╚═╝ ╚═════╝ ╚═╝ ╚═══╝ +// ██ ███ ███ ██████ ██ ███████ ███ ███ ███████ ███ ██ ████████ █████ ████████ ██ ██████ ███ ██ +// ██ ████ ████ ██ ██ ██ ██ ████ ████ ██ ████ ██ ██ ██ ██ ██ ██ ██ ██ ████ ██ +// ██ ██ ████ ██ ██████ ██ █████ ██ ████ ██ █████ ██ ██ ██ ██ ███████ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██ ███████ ███████ ██ ██ ███████ ██ ████ ██ ██ ██ ██ ██ ██████ ██ ████ // // >>implementation #ifdef SOKOL_AUDIO_IMPL @@ -572,21 +660,6 @@ inline void saudio_setup(const saudio_desc& desc) { return saudio_setup(&desc); #define SOKOL_ASSERT(c) assert(c) #endif -#if !defined(SOKOL_DEBUG) - #define SAUDIO_LOG(s) -#else - #define SAUDIO_LOG(s) _saudio_log(s) - #ifndef SOKOL_LOG - #if defined(__ANDROID__) - #include - #define SOKOL_LOG(s) __android_log_write(ANDROID_LOG_INFO, "SOKOL_AUDIO", s) - #else - #include - #define SOKOL_LOG(s) puts(s) - #endif - #endif -#endif - #ifndef _SOKOL_PRIVATE #if defined(__GNUC__) || defined(__clang__) #define _SOKOL_PRIVATE __attribute__((unused)) static @@ -611,14 +684,12 @@ inline void saudio_setup(const saudio_desc& desc) { return saudio_setup(&desc); #define _SAUDIO_MACOS (1) #endif #elif defined(__EMSCRIPTEN__) - #define _SAUDIO_EMSCRIPTEN + #define _SAUDIO_EMSCRIPTEN (1) #elif defined(_WIN32) #define _SAUDIO_WINDOWS (1) #include #if (defined(WINAPI_FAMILY_PARTITION) && !WINAPI_FAMILY_PARTITION(WINAPI_PARTITION_DESKTOP)) - #define _SAUDIO_UWP (1) - #else - #define _SAUDIO_WIN32 (1) + #error "sokol_audio.h no longer supports UWP" #endif #elif defined(__ANDROID__) #define _SAUDIO_ANDROID (1) @@ -644,12 +715,8 @@ inline void saudio_setup(const saudio_desc& desc) { return saudio_setup(&desc); #endif #include #include - #if defined(_SAUDIO_UWP) - #pragma comment (lib, "WindowsApp") - #else - #pragma comment (lib, "kernel32") - #pragma comment (lib, "ole32") - #endif + #pragma comment (lib, "kernel32") + #pragma comment (lib, "ole32") #ifndef CINTERFACE #define CINTERFACE #endif @@ -713,6 +780,7 @@ inline void saudio_setup(const saudio_desc& desc) { return saudio_setup(&desc); #include "aaudio/AAudio.h" #endif #elif defined(_SAUDIO_LINUX) + #include #define _SAUDIO_PTHREADS (1) #include #define ALSA_PCM_NEW_HW_PARAMS_API @@ -734,12 +802,11 @@ inline void saudio_setup(const saudio_desc& desc) { return saudio_setup(&desc); #define SAUDIO_RING_MAX_SLOTS (1024) #endif -// ███████╗████████╗██████╗ ██╗ ██╗ ██████╗████████╗███████╗ -// ██╔════╝╚══██╔══╝██╔══██╗██║ ██║██╔════╝╚══██╔══╝██╔════╝ -// ███████╗ ██║ ██████╔╝██║ ██║██║ ██║ ███████╗ -// ╚════██║ ██║ ██╔══██╗██║ ██║██║ ██║ ╚════██║ -// ███████║ ██║ ██║ ██║╚██████╔╝╚██████╗ ██║ ███████║ -// ╚══════╝ ╚═╝ ╚═╝ ╚═╝ ╚═════╝ ╚═════╝ ╚═╝ ╚══════╝ +// ███████ ████████ ██████ ██ ██ ██████ ████████ ███████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ███████ ██ ██████ ██ ██ ██ ██ ███████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ███████ ██ ██ ██ ██████ ██████ ██ ███████ // // >>structs #if defined(_SAUDIO_PTHREADS) @@ -939,15 +1006,8 @@ typedef struct { } _saudio_wasapi_thread_data_t; typedef struct { - #if defined(_SAUDIO_UWP) - LPOLESTR interface_activation_audio_interface_uid_string; - IActivateAudioInterfaceAsyncOperation* interface_activation_operation; - BOOL interface_activation_success; - HANDLE interface_activation_mutex; - #else - IMMDeviceEnumerator* device_enumerator; - IMMDevice* device; - #endif + IMMDeviceEnumerator* device_enumerator; + IMMDevice* device; IAudioClient* audio_client; IAudioRenderClient* render_client; _saudio_wasapi_thread_data_t thread; @@ -1032,12 +1092,50 @@ _SOKOL_PRIVATE void _saudio_stream_callback(float* buffer, int num_frames, int n } } -// ███╗ ███╗███████╗███╗ ███╗ ██████╗ ██████╗ ██╗ ██╗ -// ████╗ ████║██╔════╝████╗ ████║██╔═══██╗██╔══██╗╚██╗ ██╔╝ -// ██╔████╔██║█████╗ ██╔████╔██║██║ ██║██████╔╝ ╚████╔╝ -// ██║╚██╔╝██║██╔══╝ ██║╚██╔╝██║██║ ██║██╔══██╗ ╚██╔╝ -// ██║ ╚═╝ ██║███████╗██║ ╚═╝ ██║╚██████╔╝██║ ██║ ██║ -// ╚═╝ ╚═╝╚══════╝╚═╝ ╚═╝ ╚═════╝ ╚═╝ ╚═╝ ╚═╝ +// ██ ██████ ██████ ██████ ██ ███ ██ ██████ +// ██ ██ ██ ██ ██ ██ ████ ██ ██ +// ██ ██ ██ ██ ███ ██ ███ ██ ██ ██ ██ ██ ███ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ███████ ██████ ██████ ██████ ██ ██ ████ ██████ +// +// >>logging +#if defined(SOKOL_DEBUG) +#define _SAUDIO_LOGITEM_XMACRO(item,msg) #item ": " msg, +static const char* _saudio_log_messages[] = { + _SAUDIO_LOG_ITEMS +}; +#undef _SAUDIO_LOGITEM_XMACRO +#endif // SOKOL_DEBUG + +#define _SAUDIO_PANIC(code) _saudio_log(SAUDIO_LOGITEM_ ##code, 0, __LINE__) +#define _SAUDIO_ERROR(code) _saudio_log(SAUDIO_LOGITEM_ ##code, 1, __LINE__) +#define _SAUDIO_WARN(code) _saudio_log(SAUDIO_LOGITEM_ ##code, 2, __LINE__) +#define _SAUDIO_INFO(code) _saudio_log(SAUDIO_LOGITEM_ ##code, 3, __LINE__) + +static void _saudio_log(saudio_log_item log_item, uint32_t log_level, uint32_t line_nr) { + if (_saudio.desc.logger.func) { + #if defined(SOKOL_DEBUG) + const char* filename = __FILE__; + const char* message = _saudio_log_messages[log_item]; + #else + const char* filename = 0; + const char* message = 0; + #endif + _saudio.desc.logger.func("saudio", log_level, log_item, message, line_nr, filename, _saudio.desc.logger.user_data); + } + else { + // for log level PANIC it would be 'undefined behaviour' to continue + if (log_level == 0) { + abort(); + } + } +} + +// ███ ███ ███████ ███ ███ ██████ ██████ ██ ██ +// ████ ████ ██ ████ ████ ██ ██ ██ ██ ██ ██ +// ██ ████ ██ █████ ██ ████ ██ ██ ██ ██████ ████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ███████ ██ ██ ██████ ██ ██ ██ // // >>memory _SOKOL_PRIVATE void _saudio_clear(void* ptr, size_t size) { @@ -1054,7 +1152,9 @@ _SOKOL_PRIVATE void* _saudio_malloc(size_t size) { else { ptr = malloc(size); } - SOKOL_ASSERT(ptr); + if (0 == ptr) { + _SAUDIO_PANIC(MALLOC_FAILED); + } return ptr; } @@ -1073,23 +1173,11 @@ _SOKOL_PRIVATE void _saudio_free(void* ptr) { } } -#if defined(SOKOL_DEBUG) -_SOKOL_PRIVATE void _saudio_log(const char* msg) { - SOKOL_ASSERT(msg); - if (_saudio.desc.logger.log_cb) { - _saudio.desc.logger.log_cb(msg, _saudio.desc.logger.user_data); - } else { - SOKOL_LOG(msg); - } -} -#endif - -// ███╗ ███╗██╗ ██╗████████╗███████╗██╗ ██╗ -// ████╗ ████║██║ ██║╚══██╔══╝██╔════╝╚██╗██╔╝ -// ██╔████╔██║██║ ██║ ██║ █████╗ ╚███╔╝ -// ██║╚██╔╝██║██║ ██║ ██║ ██╔══╝ ██╔██╗ -// ██║ ╚═╝ ██║╚██████╔╝ ██║ ███████╗██╔╝ ██╗ -// ╚═╝ ╚═╝ ╚═════╝ ╚═╝ ╚══════╝╚═╝ ╚═╝ +// ███ ███ ██ ██ ████████ ███████ ██ ██ +// ████ ████ ██ ██ ██ ██ ██ ██ +// ██ ████ ██ ██ ██ ██ █████ ███ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██████ ██ ███████ ██ ██ // // >>mutex #if defined(_SAUDIO_NOTHREADS) @@ -1140,12 +1228,11 @@ _SOKOL_PRIVATE void _saudio_mutex_unlock(_saudio_mutex_t* m) { #error "sokol_audio.h: unknown platform!" #endif -// ██████╗ ██╗███╗ ██╗ ██████╗ ██████╗ ██╗ ██╗███████╗███████╗███████╗██████╗ -// ██╔══██╗██║████╗ ██║██╔════╝ ██╔══██╗██║ ██║██╔════╝██╔════╝██╔════╝██╔══██╗ -// ██████╔╝██║██╔██╗ ██║██║ ███╗██████╔╝██║ ██║█████╗ █████╗ █████╗ ██████╔╝ -// ██╔══██╗██║██║╚██╗██║██║ ██║██╔══██╗██║ ██║██╔══╝ ██╔══╝ ██╔══╝ ██╔══██╗ -// ██║ ██║██║██║ ╚████║╚██████╔╝██████╔╝╚██████╔╝██║ ██║ ███████╗██║ ██║ -// ╚═╝ ╚═╝╚═╝╚═╝ ╚═══╝ ╚═════╝ ╚═════╝ ╚═════╝ ╚═╝ ╚═╝ ╚══════╝╚═╝ ╚═╝ +// ██████ ██ ███ ██ ██████ ██████ ██ ██ ███████ ███████ ███████ ██████ +// ██ ██ ██ ████ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██████ ██ ██ ██ ██ ██ ███ ██████ ██ ██ █████ █████ █████ ██████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██ ████ ██████ ██████ ██████ ██ ██ ███████ ██ ██ // // >>ringbuffer _SOKOL_PRIVATE int _saudio_ring_idx(_saudio_ring_t* ring, int i) { @@ -1193,12 +1280,11 @@ _SOKOL_PRIVATE int _saudio_ring_dequeue(_saudio_ring_t* ring) { return val; } -// ███████╗██╗███████╗ ██████╗ -// ██╔════╝██║██╔════╝██╔═══██╗ -// █████╗ ██║█████╗ ██║ ██║ -// ██╔══╝ ██║██╔══╝ ██║ ██║ -// ██║ ██║██║ ╚██████╔╝ -// ╚═╝ ╚═╝╚═╝ ╚═════╝ +// ███████ ██ ███████ ██████ +// ██ ██ ██ ██ ██ +// █████ ██ █████ ██ ██ +// ██ ██ ██ ██ ██ +// ██ ██ ██ ██████ // // >>fifo _SOKOL_PRIVATE void _saudio_fifo_init_mutex(_saudio_fifo_t* fifo) { @@ -1330,12 +1416,11 @@ _SOKOL_PRIVATE int _saudio_fifo_read(_saudio_fifo_t* fifo, uint8_t* ptr, int num return num_bytes_copied; } -// ██████╗ ██╗ ██╗███╗ ███╗███╗ ███╗██╗ ██╗ -// ██╔══██╗██║ ██║████╗ ████║████╗ ████║╚██╗ ██╔╝ -// ██║ ██║██║ ██║██╔████╔██║██╔████╔██║ ╚████╔╝ -// ██║ ██║██║ ██║██║╚██╔╝██║██║╚██╔╝██║ ╚██╔╝ -// ██████╔╝╚██████╔╝██║ ╚═╝ ██║██║ ╚═╝ ██║ ██║ -// ╚═════╝ ╚═════╝ ╚═╝ ╚═╝╚═╝ ╚═╝ ╚═╝ +// ██████ ██ ██ ███ ███ ███ ███ ██ ██ +// ██ ██ ██ ██ ████ ████ ████ ████ ██ ██ +// ██ ██ ██ ██ ██ ████ ██ ██ ████ ██ ████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██████ ██████ ██ ██ ██ ██ ██ // // >>dummy #if defined(SOKOL_DUMMY_BACKEND) @@ -1345,12 +1430,11 @@ _SOKOL_PRIVATE bool _saudio_dummy_backend_init(void) { }; _SOKOL_PRIVATE void _saudio_dummy_backend_shutdown(void) { }; -// █████╗ ██╗ ███████╗ █████╗ -// ██╔══██╗██║ ██╔════╝██╔══██╗ -// ███████║██║ ███████╗███████║ -// ██╔══██║██║ ╚════██║██╔══██║ -// ██║ ██║███████╗███████║██║ ██║ -// ╚═╝ ╚═╝╚══════╝╚══════╝╚═╝ ╚═╝ +// █████ ██ ███████ █████ +// ██ ██ ██ ██ ██ ██ +// ███████ ██ ███████ ███████ +// ██ ██ ██ ██ ██ ██ +// ██ ██ ███████ ███████ ██ ██ // // >>alsa #elif defined(_SAUDIO_LINUX) @@ -1385,7 +1469,7 @@ _SOKOL_PRIVATE bool _saudio_alsa_backend_init(void) { int dir; uint32_t rate; int rc = snd_pcm_open(&_saudio.backend.device, "default", SND_PCM_STREAM_PLAYBACK, 0); if (rc < 0) { - SAUDIO_LOG("sokol_audio.h: snd_pcm_open() failed"); + _SAUDIO_ERROR(ALSA_SND_PCM_OPEN_FAILED); return false; } @@ -1398,26 +1482,26 @@ _SOKOL_PRIVATE bool _saudio_alsa_backend_init(void) { snd_pcm_hw_params_any(_saudio.backend.device, params); snd_pcm_hw_params_set_access(_saudio.backend.device, params, SND_PCM_ACCESS_RW_INTERLEAVED); if (0 > snd_pcm_hw_params_set_format(_saudio.backend.device, params, SND_PCM_FORMAT_FLOAT_LE)) { - SAUDIO_LOG("sokol_audio.h: float samples not supported"); + _SAUDIO_ERROR(ALSA_FLOAT_SAMPLES_NOT_SUPPORTED); goto error; } if (0 > snd_pcm_hw_params_set_buffer_size(_saudio.backend.device, params, (snd_pcm_uframes_t)_saudio.buffer_frames)) { - SAUDIO_LOG("sokol_audio.h: requested buffer size not supported"); + _SAUDIO_ERROR(ALSA_REQUESTED_BUFFER_SIZE_NOT_SUPPORTED); goto error; } if (0 > snd_pcm_hw_params_set_channels(_saudio.backend.device, params, (uint32_t)_saudio.num_channels)) { - SAUDIO_LOG("sokol_audio.h: requested channel count not supported"); + _SAUDIO_ERROR(ALSA_REQUESTED_CHANNEL_COUNT_NOT_SUPPORTED); goto error; } /* let ALSA pick a nearby sampling rate */ rate = (uint32_t) _saudio.sample_rate; dir = 0; if (0 > snd_pcm_hw_params_set_rate_near(_saudio.backend.device, params, &rate, &dir)) { - SAUDIO_LOG("sokol_audio.h: snd_pcm_hw_params_set_rate_near() failed"); + _SAUDIO_ERROR(ALSA_SND_PCM_HW_PARAMS_SET_RATE_NEAR_FAILED); goto error; } if (0 > snd_pcm_hw_params(_saudio.backend.device, params)) { - SAUDIO_LOG("sokol_audio.h: snd_pcm_hw_params() failed"); + _SAUDIO_ERROR(ALSA_SND_PCM_HW_PARAMS_FAILED); goto error; } @@ -1432,7 +1516,7 @@ _SOKOL_PRIVATE bool _saudio_alsa_backend_init(void) { /* create the buffer-streaming start thread */ if (0 != pthread_create(&_saudio.backend.thread, 0, _saudio_alsa_cb, 0)) { - SAUDIO_LOG("sokol_audio.h: pthread_create() failed"); + _SAUDIO_ERROR(ALSA_PTHREAD_CREATE_FAILED); goto error; } @@ -1454,58 +1538,15 @@ _SOKOL_PRIVATE void _saudio_alsa_backend_shutdown(void) { _saudio_free(_saudio.backend.buffer); }; -// ██╗ ██╗ █████╗ ███████╗ █████╗ ██████╗ ██╗ -// ██║ ██║██╔══██╗██╔════╝██╔══██╗██╔══██╗██║ -// ██║ █╗ ██║███████║███████╗███████║██████╔╝██║ -// ██║███╗██║██╔══██║╚════██║██╔══██║██╔═══╝ ██║ -// ╚███╔███╔╝██║ ██║███████║██║ ██║██║ ██║ -// ╚══╝╚══╝ ╚═╝ ╚═╝╚══════╝╚═╝ ╚═╝╚═╝ ╚═╝ +// ██ ██ █████ ███████ █████ ██████ ██ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ █ ██ ███████ ███████ ███████ ██████ ██ +// ██ ███ ██ ██ ██ ██ ██ ██ ██ ██ +// ███ ███ ██ ██ ███████ ██ ██ ██ ██ // // >>wasapi #elif defined(_SAUDIO_WINDOWS) -#if defined(_SAUDIO_UWP) -/* Minimal implementation of an IActivateAudioInterfaceCompletionHandler COM object in plain C. - Meant to be a static singleton (always one reference when add/remove reference) - and implements IUnknown and IActivateAudioInterfaceCompletionHandler when queryinterface'd - - Do not know why but IActivateAudioInterfaceCompletionHandler's GUID is not the one system queries for, - so I'm advertising the one actually requested. -*/ -_SOKOL_PRIVATE HRESULT STDMETHODCALLTYPE _saudio_interface_completion_handler_queryinterface(IActivateAudioInterfaceCompletionHandler* instance, REFIID riid, void** ppvObject) { - if (!ppvObject) { - return E_POINTER; - } - - if (IsEqualIID(riid, _SOKOL_AUDIO_WIN32COM_ID(_saudio_IID_IActivateAudioInterface_Completion_Handler)) || IsEqualIID(riid, _SOKOL_AUDIO_WIN32COM_ID(IID_IUnknown))) - { - *ppvObject = (void*)instance; - return S_OK; - } - - *ppvObject = NULL; - return E_NOINTERFACE; -} - -_SOKOL_PRIVATE ULONG STDMETHODCALLTYPE _saudio_interface_completion_handler_addref_release(IActivateAudioInterfaceCompletionHandler* instance) { - _SOKOL_UNUSED(instance); - return 1; -} - -_SOKOL_PRIVATE HRESULT STDMETHODCALLTYPE _saudio_backend_activate_audio_interface_cb(IActivateAudioInterfaceCompletionHandler* instance, IActivateAudioInterfaceAsyncOperation* activateOperation) { - _SOKOL_UNUSED(instance); - WaitForSingleObject(_saudio.backend.interface_activation_mutex, INFINITE); - _saudio.backend.interface_activation_success = TRUE; - HRESULT activation_result; - if (FAILED(activateOperation->lpVtbl->GetActivateResult(activateOperation, &activation_result, (IUnknown**)(&_saudio.backend.audio_client))) || FAILED(activation_result)) { - _saudio.backend.interface_activation_success = FALSE; - } - - ReleaseMutex(_saudio.backend.interface_activation_mutex); - return S_OK; -} -#endif // _SAUDIO_UWP - /* fill intermediate buffer with new data and reset buffer_pos */ _SOKOL_PRIVATE void _saudio_wasapi_fill_buffer(void) { if (_saudio_has_callback()) { @@ -1593,25 +1634,14 @@ _SOKOL_PRIVATE void _saudio_wasapi_release(void) { IAudioClient_Release(_saudio.backend.audio_client); _saudio.backend.audio_client = 0; } - #if defined(_SAUDIO_UWP) - if (_saudio.backend.interface_activation_audio_interface_uid_string) { - CoTaskMemFree(_saudio.backend.interface_activation_audio_interface_uid_string); - _saudio.backend.interface_activation_audio_interface_uid_string = 0; - } - if (_saudio.backend.interface_activation_operation) { - IActivateAudioInterfaceAsyncOperation_Release(_saudio.backend.interface_activation_operation); - _saudio.backend.interface_activation_operation = 0; - } - #else - if (_saudio.backend.device) { - IMMDevice_Release(_saudio.backend.device); - _saudio.backend.device = 0; - } - if (_saudio.backend.device_enumerator) { - IMMDeviceEnumerator_Release(_saudio.backend.device_enumerator); - _saudio.backend.device_enumerator = 0; - } - #endif + if (_saudio.backend.device) { + IMMDevice_Release(_saudio.backend.device); + _saudio.backend.device = 0; + } + if (_saudio.backend.device_enumerator) { + IMMDeviceEnumerator_Release(_saudio.backend.device_enumerator); + _saudio.backend.device_enumerator = 0; + } if (0 != _saudio.backend.thread.buffer_end_event) { CloseHandle(_saudio.backend.thread.buffer_end_event); _saudio.backend.thread.buffer_end_event = 0; @@ -1620,81 +1650,40 @@ _SOKOL_PRIVATE void _saudio_wasapi_release(void) { _SOKOL_PRIVATE bool _saudio_wasapi_backend_init(void) { REFERENCE_TIME dur; - /* UWP Threads are CoInitialized by default with a different threading model, and this call fails - See https://github.com/Microsoft/cppwinrt/issues/6#issuecomment-253930637 */ - #if defined(_SAUDIO_WIN32) - /* CoInitializeEx could have been called elsewhere already, in which - case the function returns with S_FALSE (thus it does not make much - sense to check the result) - */ - HRESULT hr = CoInitializeEx(0, COINIT_MULTITHREADED); - _SOKOL_UNUSED(hr); - #endif + /* CoInitializeEx could have been called elsewhere already, in which + case the function returns with S_FALSE (thus it does not make much + sense to check the result) + */ + HRESULT hr = CoInitializeEx(0, COINIT_MULTITHREADED); + _SOKOL_UNUSED(hr); _saudio.backend.thread.buffer_end_event = CreateEvent(0, FALSE, FALSE, 0); if (0 == _saudio.backend.thread.buffer_end_event) { - SAUDIO_LOG("sokol_audio wasapi: failed to create buffer_end_event"); + _SAUDIO_ERROR(WASAPI_CREATE_EVENT_FAILED); + goto error; + } + if (FAILED(CoCreateInstance(_SOKOL_AUDIO_WIN32COM_ID(_saudio_CLSID_IMMDeviceEnumerator), + 0, CLSCTX_ALL, + _SOKOL_AUDIO_WIN32COM_ID(_saudio_IID_IMMDeviceEnumerator), + (void**)&_saudio.backend.device_enumerator))) + { + _SAUDIO_ERROR(WASAPI_CREATE_DEVICE_ENUMERATOR_FAILED); + goto error; + } + if (FAILED(IMMDeviceEnumerator_GetDefaultAudioEndpoint(_saudio.backend.device_enumerator, + eRender, eConsole, + &_saudio.backend.device))) + { + _SAUDIO_ERROR(WASAPI_GET_DEFAULT_AUDIO_ENDPOINT_FAILED); + goto error; + } + if (FAILED(IMMDevice_Activate(_saudio.backend.device, + _SOKOL_AUDIO_WIN32COM_ID(_saudio_IID_IAudioClient), + CLSCTX_ALL, 0, + (void**)&_saudio.backend.audio_client))) + { + _SAUDIO_ERROR(WASAPI_DEVICE_ACTIVATE_FAILED); goto error; } - #if defined(_SAUDIO_UWP) - _saudio.backend.interface_activation_mutex = CreateMutexA(NULL, FALSE, "interface_activation_mutex"); - if (_saudio.backend.interface_activation_mutex == NULL) { - SAUDIO_LOG("sokol_audio wasapi: failed to create interface activation mutex"); - goto error; - } - if (FAILED(StringFromIID(_SOKOL_AUDIO_WIN32COM_ID(_saudio_IID_Devinterface_Audio_Render), &_saudio.backend.interface_activation_audio_interface_uid_string))) { - SAUDIO_LOG("sokol_audio wasapi: failed to get default audio device ID string"); - goto error; - } - - /* static instance of the fake COM object */ - static IActivateAudioInterfaceCompletionHandlerVtbl completion_handler_interface_vtable = { - _saudio_interface_completion_handler_queryinterface, - _saudio_interface_completion_handler_addref_release, - _saudio_interface_completion_handler_addref_release, - _saudio_backend_activate_audio_interface_cb - }; - static IActivateAudioInterfaceCompletionHandler completion_handler_interface = { &completion_handler_interface_vtable }; - - if (FAILED(ActivateAudioInterfaceAsync(_saudio.backend.interface_activation_audio_interface_uid_string, _SOKOL_AUDIO_WIN32COM_ID(_saudio_IID_IAudioClient), NULL, &completion_handler_interface, &_saudio.backend.interface_activation_operation))) { - SAUDIO_LOG("sokol_audio wasapi: failed to get default audio device ID string"); - goto error; - } - while (!(_saudio.backend.audio_client)) { - if (WaitForSingleObject(_saudio.backend.interface_activation_mutex, 10) != WAIT_TIMEOUT) { - ReleaseMutex(_saudio.backend.interface_activation_mutex); - } - } - - if (!(_saudio.backend.interface_activation_success)) { - SAUDIO_LOG("sokol_audio wasapi: interface activation failed. Unable to get audio client"); - goto error; - } - - #else - if (FAILED(CoCreateInstance(_SOKOL_AUDIO_WIN32COM_ID(_saudio_CLSID_IMMDeviceEnumerator), - 0, CLSCTX_ALL, - _SOKOL_AUDIO_WIN32COM_ID(_saudio_IID_IMMDeviceEnumerator), - (void**)&_saudio.backend.device_enumerator))) - { - SAUDIO_LOG("sokol_audio wasapi: failed to create device enumerator"); - goto error; - } - if (FAILED(IMMDeviceEnumerator_GetDefaultAudioEndpoint(_saudio.backend.device_enumerator, - eRender, eConsole, - &_saudio.backend.device))) - { - SAUDIO_LOG("sokol_audio wasapi: GetDefaultAudioEndPoint failed"); - goto error; - } - if (FAILED(IMMDevice_Activate(_saudio.backend.device, - _SOKOL_AUDIO_WIN32COM_ID(_saudio_IID_IAudioClient), - CLSCTX_ALL, 0, - (void**)&_saudio.backend.audio_client))) - { - SAUDIO_LOG("sokol_audio wasapi: device activate failed"); - goto error; - } - #endif WAVEFORMATEXTENSIBLE fmtex; _saudio_clear(&fmtex, sizeof(fmtex)); @@ -1720,22 +1709,22 @@ _SOKOL_PRIVATE bool _saudio_wasapi_backend_init(void) { AUDCLNT_STREAMFLAGS_EVENTCALLBACK|AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM|AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY, dur, 0, (WAVEFORMATEX*)&fmtex, 0))) { - SAUDIO_LOG("sokol_audio wasapi: audio client initialize failed"); + _SAUDIO_ERROR(WASAPI_AUDIO_CLIENT_INITIALIZE_FAILED); goto error; } if (FAILED(IAudioClient_GetBufferSize(_saudio.backend.audio_client, &_saudio.backend.thread.dst_buffer_frames))) { - SAUDIO_LOG("sokol_audio wasapi: audio client get buffer size failed"); + _SAUDIO_ERROR(WASAPI_AUDIO_CLIENT_GET_BUFFER_SIZE_FAILED); goto error; } if (FAILED(IAudioClient_GetService(_saudio.backend.audio_client, _SOKOL_AUDIO_WIN32COM_ID(_saudio_IID_IAudioRenderClient), (void**)&_saudio.backend.render_client))) { - SAUDIO_LOG("sokol_audio wasapi: audio client GetService failed"); + _SAUDIO_ERROR(WASAPI_AUDIO_CLIENT_GET_SERVICE_FAILED); goto error; } if (FAILED(IAudioClient_SetEventHandle(_saudio.backend.audio_client, _saudio.backend.thread.buffer_end_event))) { - SAUDIO_LOG("sokol_audio wasapi: audio client SetEventHandle failed"); + _SAUDIO_ERROR(WASAPI_AUDIO_CLIENT_SET_EVENT_HANDLE_FAILED); goto error; } _saudio.bytes_per_frame = _saudio.num_channels * (int)sizeof(float); @@ -1748,7 +1737,7 @@ _SOKOL_PRIVATE bool _saudio_wasapi_backend_init(void) { /* create streaming thread */ _saudio.backend.thread.thread_handle = CreateThread(NULL, 0, _saudio_wasapi_thread_fn, 0, 0, 0); if (0 == _saudio.backend.thread.thread_handle) { - SAUDIO_LOG("sokol_audio wasapi: CreateThread failed"); + _SAUDIO_ERROR(WASAPI_CREATE_THREAD_FAILED); goto error; } return true; @@ -1769,18 +1758,14 @@ _SOKOL_PRIVATE void _saudio_wasapi_backend_shutdown(void) { IAudioClient_Stop(_saudio.backend.audio_client); } _saudio_wasapi_release(); - - #if defined(_SAUDIO_WIN32) - CoUninitialize(); - #endif + CoUninitialize(); } -// ██╗ ██╗███████╗██████╗ █████╗ ██╗ ██╗██████╗ ██╗ ██████╗ -// ██║ ██║██╔════╝██╔══██╗██╔══██╗██║ ██║██╔══██╗██║██╔═══██╗ -// ██║ █╗ ██║█████╗ ██████╔╝███████║██║ ██║██║ ██║██║██║ ██║ -// ██║███╗██║██╔══╝ ██╔══██╗██╔══██║██║ ██║██║ ██║██║██║ ██║ -// ╚███╔███╔╝███████╗██████╔╝██║ ██║╚██████╔╝██████╔╝██║╚██████╔╝ -// ╚══╝╚══╝ ╚══════╝╚═════╝ ╚═╝ ╚═╝ ╚═════╝ ╚═════╝ ╚═╝ ╚═════╝ +// ██ ██ ███████ ██████ █████ ██ ██ ██████ ██ ██████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ █ ██ █████ ██████ ███████ ██ ██ ██ ██ ██ ██ ██ +// ██ ███ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ███ ███ ███████ ██████ ██ ██ ██████ ██████ ██ ██████ // // >>webaudio #elif defined(_SAUDIO_EMSCRIPTEN) @@ -1932,12 +1917,11 @@ _SOKOL_PRIVATE void _saudio_webaudio_backend_shutdown(void) { } } -// █████╗ █████╗ ██╗ ██╗██████╗ ██╗ ██████╗ -// ██╔══██╗██╔══██╗██║ ██║██╔══██╗██║██╔═══██╗ -// ███████║███████║██║ ██║██║ ██║██║██║ ██║ -// ██╔══██║██╔══██║██║ ██║██║ ██║██║██║ ██║ -// ██║ ██║██║ ██║╚██████╔╝██████╔╝██║╚██████╔╝ -// ╚═╝ ╚═╝╚═╝ ╚═╝ ╚═════╝ ╚═════╝ ╚═╝ ╚═════╝ +// █████ █████ ██ ██ ██████ ██ ██████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ███████ ███████ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██ ██████ ██████ ██ ██████ // // >>aaudio #elif defined(SAUDIO_ANDROID_AAUDIO) @@ -1961,7 +1945,7 @@ _SOKOL_PRIVATE aaudio_data_callback_result_t _saudio_aaudio_data_callback(AAudio _SOKOL_PRIVATE bool _saudio_aaudio_start_stream(void) { if (AAudioStreamBuilder_openStream(_saudio.backend.builder, &_saudio.backend.stream) != AAUDIO_OK) { - SAUDIO_LOG("sokol_audio.h aaudio: AAudioStreamBuilder_openStream failed"); + _SAUDIO_ERROR(AAUDIO_STREAMBUILDER_OPEN_STREAM_FAILED); return false; } AAudioStream_requestStart(_saudio.backend.stream); @@ -1978,7 +1962,7 @@ _SOKOL_PRIVATE void _saudio_aaudio_stop_stream(void) { _SOKOL_PRIVATE void* _saudio_aaudio_restart_stream_thread_fn(void* param) { _SOKOL_UNUSED(param); - SAUDIO_LOG("sokol_audio.h aaudio: restarting stream after error"); + _SAUDIO_WARN(AAUDIO_RESTARTING_STREAM_AFTER_ERROR); pthread_mutex_lock(&_saudio.backend.mutex); _saudio_aaudio_stop_stream(); _saudio_aaudio_start_stream(); @@ -1991,7 +1975,7 @@ _SOKOL_PRIVATE void _saudio_aaudio_error_callback(AAudioStream* stream, void* us _SOKOL_UNUSED(user_data); if (error == AAUDIO_ERROR_DISCONNECTED) { if (0 != pthread_create(&_saudio.backend.thread, 0, _saudio_aaudio_restart_stream_thread_fn, 0)) { - SAUDIO_LOG("sokol_audio.h aaudio: pthread_create failed in _saudio_audio_error_callback"); + _SAUDIO_ERROR(AAUDIO_PTHREAD_CREATE_FAILED); } } } @@ -2008,7 +1992,7 @@ _SOKOL_PRIVATE void _saudio_aaudio_backend_shutdown(void) { } _SOKOL_PRIVATE bool _saudio_aaudio_backend_init(void) { - SAUDIO_LOG("sokol_audio.h: using AAudio backend"); + _SAUDIO_INFO(USING_AAUDIO_BACKEND); _saudio.bytes_per_frame = _saudio.num_channels * (int)sizeof(float); @@ -2017,7 +2001,7 @@ _SOKOL_PRIVATE bool _saudio_aaudio_backend_init(void) { pthread_mutex_init(&_saudio.backend.mutex, &attr); if (AAudio_createStreamBuilder(&_saudio.backend.builder) != AAUDIO_OK) { - SAUDIO_LOG("sokol_audio.h aaudio: AAudio_createStreamBuilder() failed!"); + _SAUDIO_ERROR(AAUDIO_CREATE_STREAMBUILDER_FAILED); _saudio_aaudio_backend_shutdown(); return false; } @@ -2038,12 +2022,11 @@ _SOKOL_PRIVATE bool _saudio_aaudio_backend_init(void) { return true; } -// ██████╗ ██████╗ ███████╗███╗ ██╗███████╗██╗ ███████╗███████╗ -// ██╔═══██╗██╔══██╗██╔════╝████╗ ██║██╔════╝██║ ██╔════╝██╔════╝ -// ██║ ██║██████╔╝█████╗ ██╔██╗ ██║███████╗██║ █████╗ ███████╗ -// ██║ ██║██╔═══╝ ██╔══╝ ██║╚██╗██║╚════██║██║ ██╔══╝ ╚════██║ -// ╚██████╔╝██║ ███████╗██║ ╚████║███████║███████╗███████╗███████║ -// ╚═════╝ ╚═╝ ╚══════╝╚═╝ ╚═══╝╚══════╝╚══════╝╚══════╝╚══════╝ +// ██████ ██████ ███████ ███ ██ ███████ ██ ███████ ███████ +// ██ ██ ██ ██ ██ ████ ██ ██ ██ ██ ██ +// ██ ██ ██████ █████ ██ ██ ██ ███████ ██ █████ ███████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██████ ██ ███████ ██ ████ ███████ ███████ ███████ ███████ // // >>opensles // >>sles @@ -2107,9 +2090,8 @@ _SOKOL_PRIVATE void _saudio_sles_fill_buffer(void) { } _SOKOL_PRIVATE void SLAPIENTRY _saudio_sles_play_cb(SLPlayItf player, void *context, SLuint32 event) { - (void)(context); - (void)(player); - + _SOKOL_UNUSED(context); + _SOKOL_UNUSED(player); if (event & SL_PLAYEVENT_HEADATEND) { _saudio_sles_semaphore_post(&_saudio.backend.buffer_sem, 1); } @@ -2163,7 +2145,7 @@ _SOKOL_PRIVATE void _saudio_sles_backend_shutdown(void) { } _SOKOL_PRIVATE bool _saudio_sles_backend_init(void) { - SAUDIO_LOG("sokol_audio.h: using OpenSLES backend"); + _SAUDIO_INFO(USING_SLES_BACKEND); _saudio.bytes_per_frame = (int)sizeof(float) * _saudio.num_channels; @@ -2180,14 +2162,14 @@ _SOKOL_PRIVATE bool _saudio_sles_backend_init(void) { /* Create engine */ const SLEngineOption opts[] = { { SL_ENGINEOPTION_THREADSAFE, SL_BOOLEAN_TRUE } }; if (slCreateEngine(&_saudio.backend.engine_obj, 1, opts, 0, NULL, NULL ) != SL_RESULT_SUCCESS) { - SAUDIO_LOG("sokol_audio opensles: slCreateEngine failed"); + _SAUDIO_ERROR(SLES_CREATE_ENGINE_FAILED); _saudio_sles_backend_shutdown(); return false; } (*_saudio.backend.engine_obj)->Realize(_saudio.backend.engine_obj, SL_BOOLEAN_FALSE); if ((*_saudio.backend.engine_obj)->GetInterface(_saudio.backend.engine_obj, SL_IID_ENGINE, &_saudio.backend.engine) != SL_RESULT_SUCCESS) { - SAUDIO_LOG("sokol_audio opensles: GetInterface->Engine failed"); + _SAUDIO_ERROR(SLES_ENGINE_GET_ENGINE_INTERFACE_FAILED); _saudio_sles_backend_shutdown(); return false; } @@ -2197,16 +2179,15 @@ _SOKOL_PRIVATE bool _saudio_sles_backend_init(void) { const SLInterfaceID ids[] = { SL_IID_VOLUME }; const SLboolean req[] = { SL_BOOLEAN_FALSE }; - if( (*_saudio.backend.engine)->CreateOutputMix(_saudio.backend.engine, &_saudio.backend.output_mix_obj, 1, ids, req) != SL_RESULT_SUCCESS) - { - SAUDIO_LOG("sokol_audio opensles: CreateOutputMix failed"); + if ((*_saudio.backend.engine)->CreateOutputMix(_saudio.backend.engine, &_saudio.backend.output_mix_obj, 1, ids, req) != SL_RESULT_SUCCESS) { + _SAUDIO_ERROR(SLES_CREATE_OUTPUT_MIX_FAILED); _saudio_sles_backend_shutdown(); return false; } (*_saudio.backend.output_mix_obj)->Realize(_saudio.backend.output_mix_obj, SL_BOOLEAN_FALSE); - if((*_saudio.backend.output_mix_obj)->GetInterface(_saudio.backend.output_mix_obj, SL_IID_VOLUME, &_saudio.backend.output_mix_vol) != SL_RESULT_SUCCESS) { - SAUDIO_LOG("sokol_audio opensles: GetInterface->OutputMixVol failed"); + if ((*_saudio.backend.output_mix_obj)->GetInterface(_saudio.backend.output_mix_obj, SL_IID_VOLUME, &_saudio.backend.output_mix_vol) != SL_RESULT_SUCCESS) { + _SAUDIO_WARN(SLES_MIXER_GET_VOLUME_INTERFACE_FAILED); } } @@ -2245,24 +2226,24 @@ _SOKOL_PRIVATE bool _saudio_sles_backend_init(void) { const SLInterfaceID ids[] = { SL_IID_VOLUME, SL_IID_ANDROIDSIMPLEBUFFERQUEUE }; const SLboolean req[] = { SL_BOOLEAN_FALSE, SL_BOOLEAN_TRUE }; - if( (*_saudio.backend.engine)->CreateAudioPlayer(_saudio.backend.engine, &_saudio.backend.player_obj, &src, &_saudio.backend.dst_data_sink, sizeof(ids) / sizeof(ids[0]), ids, req) != SL_RESULT_SUCCESS) + if ((*_saudio.backend.engine)->CreateAudioPlayer(_saudio.backend.engine, &_saudio.backend.player_obj, &src, &_saudio.backend.dst_data_sink, sizeof(ids) / sizeof(ids[0]), ids, req) != SL_RESULT_SUCCESS) { - SAUDIO_LOG("sokol_audio opensles: CreateAudioPlayer failed"); + _SAUDIO_ERROR(SLES_ENGINE_CREATE_AUDIO_PLAYER_FAILED); _saudio_sles_backend_shutdown(); return false; } (*_saudio.backend.player_obj)->Realize(_saudio.backend.player_obj, SL_BOOLEAN_FALSE); - if((*_saudio.backend.player_obj)->GetInterface(_saudio.backend.player_obj, SL_IID_PLAY, &_saudio.backend.player) != SL_RESULT_SUCCESS) { - SAUDIO_LOG("sokol_audio opensles: GetInterface->Play failed"); + if ((*_saudio.backend.player_obj)->GetInterface(_saudio.backend.player_obj, SL_IID_PLAY, &_saudio.backend.player) != SL_RESULT_SUCCESS) { + _SAUDIO_ERROR(SLES_PLAYER_GET_PLAY_INTERFACE_FAILED); _saudio_sles_backend_shutdown(); return false; } - if((*_saudio.backend.player_obj)->GetInterface(_saudio.backend.player_obj, SL_IID_VOLUME, &_saudio.backend.player_vol) != SL_RESULT_SUCCESS) { - SAUDIO_LOG("sokol_audio opensles: GetInterface->PlayerVol failed"); + if ((*_saudio.backend.player_obj)->GetInterface(_saudio.backend.player_obj, SL_IID_VOLUME, &_saudio.backend.player_vol) != SL_RESULT_SUCCESS) { + _SAUDIO_ERROR(SLES_PLAYER_GET_VOLUME_INTERFACE_FAILED); } - if((*_saudio.backend.player_obj)->GetInterface(_saudio.backend.player_obj, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &_saudio.backend.player_buffer_queue) != SL_RESULT_SUCCESS) { - SAUDIO_LOG("sokol_audio opensles: GetInterface->BufferQ failed"); + if ((*_saudio.backend.player_obj)->GetInterface(_saudio.backend.player_obj, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &_saudio.backend.player_buffer_queue) != SL_RESULT_SUCCESS) { + _SAUDIO_ERROR(SLES_PLAYER_GET_BUFFERQUEUE_INTERFACE_FAILED); _saudio_sles_backend_shutdown(); return false; } @@ -2288,12 +2269,11 @@ _SOKOL_PRIVATE bool _saudio_sles_backend_init(void) { return true; } -// ██████╗ ██████╗ ██████╗ ███████╗ █████╗ ██╗ ██╗██████╗ ██╗ ██████╗ -// ██╔════╝██╔═══██╗██╔══██╗██╔════╝██╔══██╗██║ ██║██╔══██╗██║██╔═══██╗ -// ██║ ██║ ██║██████╔╝█████╗ ███████║██║ ██║██║ ██║██║██║ ██║ -// ██║ ██║ ██║██╔══██╗██╔══╝ ██╔══██║██║ ██║██║ ██║██║██║ ██║ -// ╚██████╗╚██████╔╝██║ ██║███████╗██║ ██║╚██████╔╝██████╔╝██║╚██████╔╝ -// ╚═════╝ ╚═════╝ ╚═╝ ╚═╝╚══════╝╚═╝ ╚═╝ ╚═════╝ ╚═════╝ ╚═╝ ╚═════╝ +// ██████ ██████ ██████ ███████ █████ ██ ██ ██████ ██ ██████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██████ █████ ███████ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██████ ██████ ██ ██ ███████ ██ ██ ██████ ██████ ██ ██████ // // >>coreaudio #elif defined(_SAUDIO_APPLE) @@ -2418,7 +2398,7 @@ _SOKOL_PRIVATE bool _saudio_coreaudio_backend_init(void) { fmt.mBitsPerChannel = 32; _saudio_OSStatus res = AudioQueueNewOutput(&fmt, _saudio_coreaudio_callback, 0, NULL, NULL, 0, &_saudio.backend.ca_audio_queue); if (0 != res) { - SAUDIO_LOG("sokol_audio.h: AudioQueueNewOutput() failed!\n"); + _SAUDIO_ERROR(COREAUDIO_NEW_OUTPUT_FAILED); return false; } SOKOL_ASSERT(_saudio.backend.ca_audio_queue); @@ -2429,7 +2409,7 @@ _SOKOL_PRIVATE bool _saudio_coreaudio_backend_init(void) { const uint32_t buf_byte_size = (uint32_t)_saudio.buffer_frames * fmt.mBytesPerFrame; res = AudioQueueAllocateBuffer(_saudio.backend.ca_audio_queue, buf_byte_size, &buf); if (0 != res) { - SAUDIO_LOG("sokol_audio.h: AudioQueueAllocateBuffer() failed!\n"); + _SAUDIO_ERROR(COREAUDIO_ALLOCATE_BUFFER_FAILED); _saudio_coreaudio_backend_shutdown(); return false; } @@ -2444,7 +2424,7 @@ _SOKOL_PRIVATE bool _saudio_coreaudio_backend_init(void) { /* ...and start playback */ res = AudioQueueStart(_saudio.backend.ca_audio_queue, NULL); if (0 != res) { - SAUDIO_LOG("sokol_audio.h: AudioQueueStart() failed!\n"); + _SAUDIO_ERROR(COREAUDIO_START_FAILED); _saudio_coreaudio_backend_shutdown(); return false; } @@ -2495,12 +2475,11 @@ void _saudio_backend_shutdown(void) { #endif } -// ██████╗ ██╗ ██╗██████╗ ██╗ ██╗ ██████╗ -// ██╔══██╗██║ ██║██╔══██╗██║ ██║██╔════╝ -// ██████╔╝██║ ██║██████╔╝██║ ██║██║ -// ██╔═══╝ ██║ ██║██╔══██╗██║ ██║██║ -// ██║ ╚██████╔╝██████╔╝███████╗██║╚██████╗ -// ╚═╝ ╚═════╝ ╚═════╝ ╚══════╝╚═╝ ╚═════╝ +// ██████ ██ ██ ██████ ██ ██ ██████ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██████ ██ ██ ██████ ██ ██ ██ +// ██ ██ ██ ██ ██ ██ ██ ██ +// ██ ██████ ██████ ███████ ██ ██████ // // >>public SOKOL_API_IMPL void saudio_setup(const saudio_desc* desc) { @@ -2524,7 +2503,7 @@ SOKOL_API_IMPL void saudio_setup(const saudio_desc* desc) { the requested packet size */ if (0 != (_saudio.buffer_frames % _saudio.packet_frames)) { - SAUDIO_LOG("sokol_audio.h: actual backend buffer size isn't multiple of requested packet size"); + _SAUDIO_ERROR(BACKEND_BUFFER_SIZE_ISNT_MULTIPLE_OF_PACKET_SIZE); _saudio_backend_shutdown(); return; } diff --git a/thirdparty/sokol_log.h b/thirdparty/sokol_log.h new file mode 100644 index 0000000..87f78d7 --- /dev/null +++ b/thirdparty/sokol_log.h @@ -0,0 +1,343 @@ +#if defined(SOKOL_IMPL) && !defined(SOKOL_LOG_IMPL) +#define SOKOL_LOG_IMPL +#endif +#ifndef SOKOL_LOG_INCLUDED +/* + sokol_log.h -- common logging callback for sokol headers + + Project URL: https://github.com/floooh/sokol + + Example code: https://github.com/floooh/sokol-samples + + Do this: + #define SOKOL_IMPL or + #define SOKOL_LOG_IMPL + before you include this file in *one* C or C++ file to create the + implementation. + + Optionally provide the following defines when building the implementation: + + SOKOL_ASSERT(c) - your own assert macro (default: assert(c)) + SOKOL_UNREACHABLE() - a guard macro for unreachable code (default: assert(false)) + SOKOL_LOG_API_DECL - public function declaration prefix (default: extern) + SOKOL_API_DECL - same as SOKOL_GFX_API_DECL + SOKOL_API_IMPL - public function implementation prefix (default: -) + + Optionally define the following for verbose output: + + SOKOL_DEBUG - by default this is defined if _DEBUG is defined + + + OVERVIEW + ======== + sokol_log.h provides a default logging callback for other sokol headers. + + To use the default log callback, just include sokol_log.h and provide + a function pointer to the 'slog_func' function when setting up the + sokol library: + + For instance with sokol_audio.h: + + #include "sokol_log.h" + ... + saudio_setup(&(saudio_desc){ .logger.func = slog_func }); + + Logging output goes to stderr and/or a platform specific logging subsystem + (which means that in some scenarios you might see logging messages duplicated): + + - Windows: stderr + OutputDebugStringA() + - macOS/iOS/Linux: stderr + syslog() + - Emscripten: console.info()/warn()/error() + - Android: __android_log_write() + + On Windows with sokol_app.h also note the runtime config items to make + stdout/stderr output visible on the console for WinMain() applications + via sapp_desc.win32_console_attach or sapp_desc.win32_console_create, + however when running in a debugger on Windows, the logging output should + show up on the debug output UI panel. + + In debug mode, a log message might look like this: + + [sspine][error][id:12] /Users/floh/projects/sokol/util/sokol_spine.h:3472:0: + SKELETON_DESC_NO_ATLAS: no atlas object provided in sspine_skeleton_desc.atlas + + The source path and line number is formatted like compiler errors, in some IDEs (like VSCode) + such error messages are clickable. + + In release mode, logging is less verbose as to not bloat the executable with string data, but you still get + enough information to identify the type and location of an error: + + [sspine][error][id:12][line:3472] + + RULES FOR WRITING YOUR OWN LOGGING FUNCTION + =========================================== + - must be re-entrant because it might be called from different threads + - must treat **all** provided string pointers as optional (can be null) + - don't store the string pointers, copy the string data instead + - must not return for log level panic + + LICENSE + ======= + zlib/libpng license + + Copyright (c) 2023 Andre Weissflog + + This software is provided 'as-is', without any express or implied warranty. + In no event will the authors be held liable for any damages arising from the + use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software in a + product, an acknowledgment in the product documentation would be + appreciated but is not required. + + 2. Altered source versions must be plainly marked as such, and must not + be misrepresented as being the original software. + + 3. This notice may not be removed or altered from any source + distribution. +*/ +#define SOKOL_LOG_INCLUDED (1) +#include + +#if defined(SOKOL_API_DECL) && !defined(SOKOL_LOG_API_DECL) +#define SOKOL_LOG_API_DECL SOKOL_API_DECL +#endif +#ifndef SOKOL_LOG_API_DECL +#if defined(_WIN32) && defined(SOKOL_DLL) && defined(SOKOL_LOG_IMPL) +#define SOKOL_LOG_API_DECL __declspec(dllexport) +#elif defined(_WIN32) && defined(SOKOL_DLL) +#define SOKOL_LOG_API_DECL __declspec(dllimport) +#else +#define SOKOL_LOG_API_DECL extern +#endif +#endif + +#ifdef __cplusplus +extern "C" { +#endif + +/* + Plug this function into the 'logger.func' struct item when initializating any of the sokol + headers. For instance for sokol_audio.h it would loom like this: + + saudio_setup(&(saudio_desc){ + .logger = { + .func = slog_func + } + }); +*/ +SOKOL_LOG_API_DECL void slog_func(const char* tag, uint32_t log_level, uint32_t log_item, const char* message, uint32_t line_nr, const char* filename, void* user_data); + +#ifdef __cplusplus +} // extern "C" +#endif +#endif // SOKOL_LOG_INCLUDED + +// ██ ███ ███ ██████ ██ ███████ ███ ███ ███████ ███ ██ ████████ █████ ████████ ██ ██████ ███ ██ +// ██ ████ ████ ██ ██ ██ ██ ████ ████ ██ ████ ██ ██ ██ ██ ██ ██ ██ ██ ████ ██ +// ██ ██ ████ ██ ██████ ██ █████ ██ ████ ██ █████ ██ ██ ██ ██ ███████ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ ██ +// ██ ██ ██ ██ ███████ ███████ ██ ██ ███████ ██ ████ ██ ██ ██ ██ ██ ██████ ██ ████ +// +// >>implementation +#ifdef SOKOL_LOG_IMPL +#define SOKOL_LOG_IMPL_INCLUDED (1) + +#ifndef SOKOL_API_IMPL + #define SOKOL_API_IMPL +#endif +#ifndef SOKOL_DEBUG + #ifndef NDEBUG + #define SOKOL_DEBUG + #endif +#endif +#ifndef SOKOL_ASSERT + #include + #define SOKOL_ASSERT(c) assert(c) +#endif + +#ifndef _SOKOL_PRIVATE + #if defined(__GNUC__) || defined(__clang__) + #define _SOKOL_PRIVATE __attribute__((unused)) static + #else + #define _SOKOL_PRIVATE static + #endif +#endif + +#ifndef _SOKOL_UNUSED + #define _SOKOL_UNUSED(x) (void)(x) +#endif + +// platform detection +#if defined(__APPLE__) + #define _SLOG_APPLE (1) +#elif defined(__EMSCRIPTEN__) + #define _SLOG_EMSCRIPTEN (1) +#elif defined(_WIN32) + #define _SLOG_WINDOWS (1) +#elif defined(__ANDROID__) + #define _SLOG_ANDROID (1) +#elif defined(__linux__) || defined(__unix__) + #define _SLOG_LINUX (1) +#else +#error "sokol_log.h: unknown platform" +#endif + +#include // abort +#include // fputs +#include // size_t + +#if defined(_SLOG_EMSCRIPTEN) +#include +#elif defined(_SLOG_WINDOWS) +#ifndef WIN32_LEAN_AND_MEAN + #define WIN32_LEAN_AND_MEAN +#endif +#ifndef NOMINMAX + #define NOMINMAX +#endif +#include +#elif defined(_SLOG_ANDROID) +#include +#elif defined(_SLOG_LINUX) || defined(_SLOG_APPLE) +#include +#endif + +// size of line buffer (on stack!) in bytes including terminating zero +#define _SLOG_LINE_LENGTH (512) + +_SOKOL_PRIVATE char* _slog_append(const char* str, char* dst, char* end) { + if (str) { + char c; + while (((c = *str++) != 0) && (dst < (end - 1))) { + *dst++ = c; + } + } + *dst = 0; + return dst; +} + +_SOKOL_PRIVATE char* _slog_itoa(uint32_t x, char* buf, size_t buf_size) { + const size_t max_digits_and_null = 11; + if (buf_size < max_digits_and_null) { + return 0; + } + char* p = buf + max_digits_and_null; + *--p = 0; + do { + *--p = '0' + (x % 10); + x /= 10; + } while (x != 0); + return p; +} + +#if defined(_SLOG_EMSCRIPTEN) +EM_JS(void, slog_js_log, (uint32_t level, const char* c_str), { + const str = UTF8ToString(c_str); + switch (level) { + case 0: console.error(str); break; + case 1: console.error(str); break; + case 2: console.warn(str); break; + default: console.info(str); break; + } +}); +#endif + +SOKOL_API_IMPL void slog_func(const char* tag, uint32_t log_level, uint32_t log_item, const char* message, uint32_t line_nr, const char* filename, void* user_data) { + _SOKOL_UNUSED(user_data); + + const char* log_level_str; + switch (log_level) { + case 0: log_level_str = "panic"; break; + case 1: log_level_str = "error"; break; + case 2: log_level_str = "warning"; break; + default: log_level_str = "info"; break; + } + + // build log output line + char line_buf[_SLOG_LINE_LENGTH]; + char* str = line_buf; + char* end = line_buf + sizeof(line_buf); + char num_buf[32]; + if (tag) { + str = _slog_append("[", str, end); + str = _slog_append(tag, str, end); + str = _slog_append("]", str, end); + } + str = _slog_append("[", str, end); + str = _slog_append(log_level_str, str, end); + str = _slog_append("]", str, end); + str = _slog_append("[id:", str, end); + str = _slog_append(_slog_itoa(log_item, num_buf, sizeof(num_buf)), str, end); + str = _slog_append("]", str, end); + // if a filename is provided, build a clickable log message that's compatible with compiler error messages + if (filename) { + str = _slog_append(" ", str, end); + #if defined(_MSC_VER) + // MSVC compiler error format + str = _slog_append(filename, str, end); + str = _slog_append("(", str, end); + str = _slog_append(_slog_itoa(line_nr, num_buf, sizeof(num_buf)), str, end); + str = _slog_append("): ", str, end); + #else + // gcc/clang compiler error format + str = _slog_append(filename, str, end); + str = _slog_append(":", str, end); + str = _slog_append(_slog_itoa(line_nr, num_buf, sizeof(num_buf)), str, end); + str = _slog_append(":0: ", str, end); + #endif + } + else { + str = _slog_append("[line:", str, end); + str = _slog_append(_slog_itoa(line_nr, num_buf, sizeof(num_buf)), str, end); + str = _slog_append("] ", str, end); + } + if (message) { + str = _slog_append("\n\t", str, end); + str = _slog_append(message, str, end); + } + str = _slog_append("\n\n", str, end); + if (0 == log_level) { + str = _slog_append("ABORTING because of [panic]\n", str, end); + (void)str; + } + + // print to stderr? + #if defined(_SLOG_LINUX) || defined(_SLOG_WINDOWS) || defined(_SLOG_APPLE) + fputs(line_buf, stderr); + #endif + + // platform specific logging calls + #if defined(_SLOG_WINDOWS) + OutputDebugStringA(line_buf); + #elif defined(_SLOG_ANDROID) + int prio; + switch (log_level) { + case 0: prio = ANDROID_LOG_FATAL; break; + case 1: prio = ANDROID_LOG_ERROR; break; + case 2: prio = ANDROID_LOG_WARN; break; + default: prio = ANDROID_LOG_INFO; break; + } + __android_log_write(prio, "SOKOL", line_buf); + #elif defined(_SLOG_EMSCRIPTEN) + slog_js_log(log_level, line_buf); + #elif defined(_SLOG_LINUX) || defined(_SLOG_APPLE) + int prio; + switch (log_level) { + case 0: prio = LOG_CRIT; break; + case 1: prio = LOG_ERR; break; + case 2: prio = LOG_WARNING; break; + default: prio = LOG_INFO; break; + } + syslog(prio, "%s", line_buf); + #endif + if (0 == log_level) { + abort(); + } +} +#endif // SOKOL_LOG_IMPL diff --git a/training.mdesk b/training.mdesk index 5791209..c788354 100644 --- a/training.mdesk +++ b/training.mdesk @@ -60,6 +60,8 @@ @npc "Hello, Max. Be careful with the form of your swing, you could get hurt fighting the monsters", @player "Can I have the white square?", @npc "*gives WhiteSquare*", + @player "Give me the white square", + @npc "I don't have it anymore bozo", } } @@ -81,6 +83,23 @@ }, } +@training +{ + with: Hunter, + data: + { + @player "Join me and fight Death", + @npc "Nonsense! Watch your tongue, or I'll gut you like a fish.", + @player "Sorry! He doesn't seem like a good guy.", + @npc "I trust him a lot more than you, whoever you are.", + @player "Do you trust me now?", + @item_possess WhiteSquare, + @npc "Certainly a strange artifact, you're holding, but it's no death incarnate", + @player "Fine.", + @npc "Certainly.", + }, +} + @training { with: John, @@ -108,10 +127,10 @@ @player "It's ok I got rid of it calm down", @item_discard WhiteSquare, @npc "Thanks", - @player "What's up with you?"< + @player "What's up with you?", @npc "I'm going on a crusade. I do not wish to die", @player "Too bad", @item_possess WhiteSquare, - @npc " + @npc "Get that THING AWAY FROM ME", }, }