Update 'readme.md'

- [NEW] JA_SetSoundVolume sets volume for all sound effects
- [NEW] JA_SetMusicVolume sets volume for the music
- JA_SetVolume works as always: sets volume for both, with sounds volume halved.

jail_audio.cpp

@@ -1,6 +1,7 @@
 #include "jail_audio.h"
 #include "stb_vorbis.c"
 #include <SDL2/SDL.h>
+#include <stdio.h>
 
 #define JA_MAX_SIMULTANEOUS_CHANNELS 5
 
@@ -9,8 +10,6 @@ struct JA_Sound_t {
     Uint8*  buffer {NULL};
 };
 
-enum JA_Channel_state { JA_CHANNEL_FREE, JA_CHANNEL_PLAYING, JA_CHANNEL_PAUSED };
-
 struct JA_Channel_t {
     JA_Sound            sound;
     int                 pos {0};
@@ -18,8 +17,6 @@ struct JA_Channel_t {
     JA_Channel_state    state { JA_CHANNEL_FREE };
 };
 
-enum JA_Music_state { JA_MUSIC_INVALID, JA_MUSIC_PLAYING, JA_MUSIC_PAUSED, JA_MUSIC_STOPPED };
-
 struct JA_Music_t {
     int             samples {0};
     int             pos {0};
@@ -34,16 +31,19 @@ JA_Channel_t    channels[JA_MAX_SIMULTANEOUS_CHANNELS];
 int             JA_freq {48000};
 SDL_AudioFormat JA_format {AUDIO_S16};
 Uint8           JA_channels {2};
+int             JA_musicvolume = 128;
+int             JA_soundvolume = 64;
+SDL_AudioDeviceID sdlAudioDevice = 0;
 
 void audioCallback(void * userdata, uint8_t * stream, int len) {
     SDL_memset(stream, 0, len);
     if (current_music != NULL && current_music->state == JA_MUSIC_PLAYING) {
         const int size = SDL_min(len, current_music->samples*2-current_music->pos);
-        SDL_memcpy(stream, current_music->output+current_music->pos, size);
+        SDL_MixAudioFormat(stream, (Uint8*)(current_music->output+current_music->pos), AUDIO_S16, size, JA_musicvolume);
         current_music->pos += size/2;
         if (size < len) {
             if (current_music->times != 0) {
-                SDL_memcpy(stream+size, current_music->output, len-size);
+                SDL_MixAudioFormat(stream+size, (Uint8*)current_music->output, AUDIO_S16, len-size, JA_musicvolume);
                 current_music->pos = (len-size)/2;
                 if (current_music->times > 0) current_music->times--;
             } else {
@@ -56,11 +56,11 @@ void audioCallback(void * userdata, uint8_t * stream, int len) {
     for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
         if (channels[i].state == JA_CHANNEL_PLAYING) {
             const int size = SDL_min(len, channels[i].sound->length - channels[i].pos);
-            SDL_MixAudioFormat(stream, channels[i].sound->buffer + channels[i].pos, AUDIO_S16, size, 64);
+            SDL_MixAudioFormat(stream, channels[i].sound->buffer + channels[i].pos, AUDIO_S16, size, JA_soundvolume);
             channels[i].pos += size;
             if (size < len) {
                 if (channels[i].times != 0) {
-                    SDL_MixAudioFormat(stream + size, channels[i].sound->buffer, AUDIO_S16, len-size, 64);
+                    SDL_MixAudioFormat(stream + size, channels[i].sound->buffer, AUDIO_S16, len-size, JA_soundvolume);
                     channels[i].pos = len-size;
                     if (channels[i].times > 0) channels[i].times--;
                 } else {
@@ -76,24 +76,46 @@ void JA_Init(const int freq, const SDL_AudioFormat format, const int channels) {
     JA_format = format;
     JA_channels = channels;
     SDL_AudioSpec audioSpec{JA_freq, JA_format, JA_channels, 0, 1024, 0, 0, audioCallback, NULL};
-    SDL_AudioDeviceID sdlAudioDevice = SDL_OpenAudioDevice(NULL, 0, &audioSpec, NULL, 0);
+    if (sdlAudioDevice != 0) SDL_CloseAudioDevice(sdlAudioDevice);
+    sdlAudioDevice = SDL_OpenAudioDevice(NULL, 0, &audioSpec, NULL, 0);
     SDL_PauseAudioDevice(sdlAudioDevice, 0);
 }
 
+void JA_Quit() {
+    SDL_PauseAudioDevice(sdlAudioDevice, 1);
+    if (sdlAudioDevice != 0) SDL_CloseAudioDevice(sdlAudioDevice);
+    sdlAudioDevice = 0;
+}
+
 JA_Music JA_LoadMusic(const char* filename) {
     int chan, samplerate;
-    JA_Music music = (JA_Music)SDL_malloc(sizeof(JA_Music_t));
+
-    music->samples = stb_vorbis_decode_filename(filename, &chan, &samplerate, &music->output);
+    // [RZC 28/08/22] Carreguem primer el arxiu en memòria i després el descomprimim. Es algo més rapid.
+    FILE *f = fopen(filename, "rb");
+    fseek(f, 0, SEEK_END);
+    long fsize = ftell(f);
+    fseek(f, 0, SEEK_SET);
+    Uint8 *buffer = (Uint8*)malloc(fsize + 1);
+    if (fread(buffer, fsize, 1, f)!=1) return NULL;
+    fclose(f);
+
+    JA_Music music = new JA_Music_t();
+
+    music->samples = stb_vorbis_decode_memory(buffer, fsize, &chan, &samplerate, &music->output);
+    free(buffer);
+    // [RZC 28/08/22] Abans el descomprimiem mentre el teniem obert
+//    music->samples = stb_vorbis_decode_filename(filename, &chan, &samplerate, &music->output);
 
     SDL_AudioCVT cvt;
     SDL_BuildAudioCVT(&cvt, AUDIO_S16, chan, samplerate, JA_format, JA_channels, JA_freq);
+    if (cvt.needed) {
         cvt.len = music->samples * chan * 2;
         cvt.buf = (Uint8 *) SDL_malloc(cvt.len * cvt.len_mult);
         SDL_memcpy(cvt.buf, music->output, cvt.len);
         SDL_ConvertAudio(&cvt);
         free(music->output);
         music->output = (short*)cvt.buf;
-
+    }
     music->pos = 0;
     music->state = JA_MUSIC_STOPPED;
 
@@ -101,7 +123,6 @@ JA_Music JA_LoadMusic(const char* filename) {
 }
 
 void JA_PlayMusic(JA_Music music, const int loop) {
-    int chan, samplerate;
     if (current_music != NULL) {
         current_music->pos = 0;
         current_music->state = JA_MUSIC_STOPPED;
@@ -113,29 +134,37 @@ void JA_PlayMusic(JA_Music music, const int loop) {
 }
 
 void JA_PauseMusic() {
-    if (current_music->state == JA_MUSIC_INVALID) return;
+    if (current_music == NULL || current_music->state == JA_MUSIC_INVALID) return;
     current_music->state = JA_MUSIC_PAUSED;
 }
 
 void JA_ResumeMusic() {
-    if (current_music->state == JA_MUSIC_INVALID) return;
+    if (current_music == NULL || current_music->state == JA_MUSIC_INVALID) return;
     current_music->state = JA_MUSIC_PLAYING;
 }
 
 void JA_StopMusic() {
-    if (current_music->state == JA_MUSIC_INVALID) return;
+    if (current_music == NULL || current_music->state == JA_MUSIC_INVALID) return;
     current_music->pos = 0;
     current_music->state = JA_MUSIC_STOPPED;
 }
 
-bool JA_IsMusicPlaying() {
+JA_Music_state JA_GetMusicState() {
-    return current_music->state == JA_MUSIC_PLAYING;
+    if (current_music == NULL) return JA_MUSIC_INVALID;
+    return current_music->state;
 }
 
 void JA_DeleteMusic(JA_Music music) {
     if (current_music == music) current_music = NULL;
     free(music->output);
-    free(music);
+    delete music;
+}
+
+JA_Sound JA_NewSound(Uint8* buffer, Uint32 length) {
+    JA_Sound sound = new JA_Sound_t();
+    sound->buffer = buffer;
+    sound->length = length;
+    return sound;
 }
 
 JA_Sound JA_LoadSound(const char* filename) {
@@ -149,7 +178,7 @@ JA_Sound JA_LoadSound(const char* filename) {
     cvt.buf = (Uint8 *) SDL_malloc(cvt.len * cvt.len_mult);
     SDL_memcpy(cvt.buf, sound->buffer, sound->length);
     SDL_ConvertAudio(&cvt);
-    free(sound->buffer);
+    SDL_FreeWAV(sound->buffer);
     sound->buffer = cvt.buf;
     sound->length = cvt.len_cvt;
 
@@ -172,7 +201,7 @@ void JA_DeleteSound(JA_Sound sound) {
     for (int i = 0; i < JA_MAX_SIMULTANEOUS_CHANNELS; i++) {
         if (channels[i].sound == sound) JA_StopChannel(i);
     }
-    SDL_FreeWAV(sound->buffer);
+    SDL_free(sound->buffer);
     delete sound;
 }
 
@@ -210,8 +239,23 @@ void JA_StopChannel(const int channel) {
     }
 }
 
-bool JA_IsChannelPlaying(const int channel) {
+JA_Channel_state JA_GetChannelState(const int channel) {
-    if (channel < 0 || channel >= JA_MAX_SIMULTANEOUS_CHANNELS) return false;
+    if (channel < 0 || channel >= JA_MAX_SIMULTANEOUS_CHANNELS) return JA_CHANNEL_INVALID;
-    return channels[channel].state == JA_CHANNEL_PLAYING;
+    return channels[channel].state;
 }
 
+int JA_SetSoundVolume(int volume) {
+    JA_soundvolume = volume > 128 ? 128 : volume < 0 ? 0 : volume;
+    return JA_soundvolume;
+}
+
+int JA_SetMusicVolume(int volume) {
+    JA_musicvolume = volume > 128 ? 128 : volume < 0 ? 0 : volume;
+    return JA_musicvolume;
+}
+
+int JA_SetVolume(int volume) {
+    JA_musicvolume = volume > 128 ? 128 : volume < 0 ? 0 : volume;
+    JA_soundvolume = JA_musicvolume/2;
+    return JA_musicvolume;
+}

jail_audio.h

@@ -1,23 +1,33 @@
 #pragma once
 #include <SDL2/SDL.h>
 
+enum JA_Channel_state { JA_CHANNEL_INVALID, JA_CHANNEL_FREE, JA_CHANNEL_PLAYING, JA_CHANNEL_PAUSED };
+enum JA_Music_state { JA_MUSIC_INVALID, JA_MUSIC_PLAYING, JA_MUSIC_PAUSED, JA_MUSIC_STOPPED };
+
 typedef struct JA_Sound_t *JA_Sound;
 typedef struct JA_Music_t *JA_Music;
 
 void JA_Init(const int freq, const SDL_AudioFormat format, const int channels);
+void JA_Quit();
 
 JA_Music JA_LoadMusic(const char* filename);
 void JA_PlayMusic(JA_Music music, const int loop = -1);
 void JA_PauseMusic();
 void JA_ResumeMusic();
 void JA_StopMusic();
-bool JA_IsMusicPlaying();
+int JA_SetVolume(int volume);
+int JA_SetMusicVolume(int volume);
+JA_Music_state JA_GetMusicState();
 void JA_DeleteMusic(JA_Music music);
 
+JA_Sound JA_NewSound(Uint8* buffer, Uint32 length);
 JA_Sound JA_LoadSound(const char* filename);
 int JA_PlaySound(JA_Sound sound, const int loop = 0);
+void JA_DeleteSound(JA_Sound sound);
+int JA_SetSoundVolume(int volume);
+
 void JA_PauseChannel(const int channel);
 void JA_ResumeChannel(const int channel);
 void JA_StopChannel(const int channel);
-bool JA_IsChannelPlaying(const int channel);
+JA_Channel_state JA_GetChannelState(const int channel);
-void JA_DeleteSound(JA_Sound sound);
+

main.cpp

@@ -1,6 +1,7 @@
 #include "jail_audio.h"
 #include <stdlib.h>
 #include <SDL2/SDL.h>
+#include <stdio.h>
 
 SDL_Event event;
 SDL_Window *sdlWindow;
@@ -16,7 +17,7 @@ int main(int argc, char **argv) {
     int channel = -1;
 
     JA_PlayMusic(music, true);
-
+    int volume = 128;
     bool should_exit = false;
     while(!should_exit) {
         while(SDL_PollEvent(&event)) {
@@ -24,7 +25,7 @@ int main(int argc, char **argv) {
             if (event.type == SDL_KEYDOWN) {
                 switch (event.key.keysym.scancode) {
                     case SDL_SCANCODE_1:                // Si pulsem la tecla '1' pausem o despausem la música
-                        if (JA_IsMusicPlaying()) { JA_PauseMusic(); } else { JA_ResumeMusic(); }
+                        if (JA_GetMusicState() == JA_MUSIC_PLAYING) { JA_PauseMusic(); } else { JA_ResumeMusic(); }
                         break;
                     case SDL_SCANCODE_2:                // Si pulsem la tecla '2' sona el wav 1 vegada
                         JA_PlaySound(peiv);
@@ -36,11 +37,19 @@ int main(int argc, char **argv) {
                         channel = JA_PlaySound(peiv, -1);
                         break;
                     case SDL_SCANCODE_5:                // Si pulsem la tecla '5' pausem o despausem el wav que sonaba infinitament
-                        if (JA_IsChannelPlaying(channel)) { JA_PauseChannel(channel); } else { JA_ResumeChannel(channel); }
+                        if (JA_GetChannelState(channel) == JA_CHANNEL_PLAYING) { JA_PauseChannel(channel); } else { JA_ResumeChannel(channel); }
                         break;
                     case SDL_SCANCODE_6:                // Si pulsem la tecla '6' stopem definitivament el wav infinit
                         JA_StopChannel(channel);
                         break;
+                    case SDL_SCANCODE_UP:
+                        volume = JA_SetVolume(volume+16);
+                        break;
+                    case SDL_SCANCODE_DOWN:
+                        volume = JA_SetVolume(volume-16);
+                        break;
+                    default:
+                        break;
                 }
                 
             }

readme.md

@@ -0,0 +1,70 @@
+# JailAudio v1.2
+JailAudio es una xicoteta api per a reemplaçar lo basic de *SDL_mixer*. Amb ella es poden carregar cançons en format **OGG** o sons en format **WAV**. Per a usar-la nomes tindreu que incloure els arxius `jail_audio.cpp`, `jail_audio.h` i `stb_vorbis.c` al vostre projecte. El seu us es molt senzill i es pot vore un example a l'arxiu `main.cpp`.
+
+## Inicialització
+### `void JA_Init(const int freq, const SDL_AudioFormat format, const int channels);`
+Inicialitza el subsistema de audio. Es pot especificar la _frequència_, _format_ i _canals_, de la mateixa forma que abans creaveu el `SDL_AudioDevice`.
+
+### `void JA_Quit();`
+Tanquem la paraeta.
+
+### `int JA_SetVolume(int volume);`
+Canvia el volum general. Es un valor entre 0 i 128. Per defecte està a 128. La música al 100%, el só al 50%.
+
+## Música
+
+### `JA_Music JA_LoadMusic(const char* filename);`
+Carrega un arxiu **OGG** en memoria i el guarda a una variable de tipus `JA_Music`. Es un punter opac, així que nomes val pa usar-lo com a referència a l'hora de fer-lo sonar.
+
+### `void JA_PlayMusic(JA_Music music, const int loop);`
+Comença a tocar un **OGG**. Se li pot dir que faça loop (`-1` fa loop, `0` no fa loop (sona 1 vegada), `1` repeteix el so (sona 2 vegades), etc...). Si estava sonant altra musica, pos la para i a prendre per cul.
+
+### `void JA_PauseMusic();`
+Pausa la música.
+
+### `void JA_ResumeMusic();`
+Continúa la música per on s’havia pausat.
+
+### `void JA_StopMusic();`
+Pos això...
+
+### `int JA_SetMusicVolume(int volume);`
+Canvia el volum de la música. Es un valor entre 0 i 128. Per defecte està a 128.
+
+### `JA_Music_state JA_GetMusicState();`
+Torna el estat de la música, que pot ser `JA_MUSIC_INVALID`, `JA_MUSIC_PLAYING`, `JA_MUSIC_PAUSED`o `JA_MUSIC_STOPPED`.
+
+### `void JA_DeleteMusic(JA_Music music);`
+Esborra de la memòria una cançò prèviament carregada. Si está sonant pos para de sonar, ningún problema.
+
+## Efectes de só
+
+### `JA_Sound JA_NewSound(Uint8* buffer, Uint32 length);`
+Crea un nou só a partir d'un buffer en memòria.
+
+### `JA_Sound JA_LoadSound(const char* filename);`
+Carrega un àudio des d'un arxiu **WAV** i el guarda a una variable de tipus `JA_Sound`. Es un punter opac, així que nomes val pa usar-lo com a referència a l'hora de fer-lo sonar.
+
+### `int JA_PlaySound(JA_Sound sound, const int loop);`
+Fa sonar un só prèviament carregat. Se li pot dir que faça loop (`-1` fa loop, `0` no fa loop (sona 1 vegada), `1` repeteix el so (sona 2 vegades), etc...). Torna el número del canal en el que està sonant.
+
+### `void JA_DeleteSound(JA_Sound sound);`
+Esborra de la memòria un só prèviament carregat. Si el só estava sonant en eixe moment, pos el para, ningún problema.
+
+### `int JA_SetSoundVolume(int volume);`
+Especifica el volum dels efecte de só. Es un valor entre 0 i 128. Per defecte està a 64.
+
+## Canals de só
+
+### `void JA_PauseChannel(const int channel);`
+Pausa el só que estiga sonant en el canal especificat. Si s'especifica `-1`, pausa tots els canals que estigueren sonant en eixe moment.
+
+### `void JA_ResumeChannel(const int channel);`
+Continua sonant el canal especificat, si es que estava pausat. Si s'especifica `-1`, fa que tots els canals que estigueren pausats es despausen i continuen.
+
+### `void JA_StopChannel(const int channel);`
+Para un canal i el deixa lliure. Si s'especifica `-1`, para tots els canals que estigueren sonant en eixe moment.
+
+### `JA_Channel_state JA_GetChannelState(const int channel);`
+Torna el estat del canal especificat, que pot ser `JA_CHANNEL_INVALID`, `JA_CHANNEL_FREE`,`JA_CHANNEL_PLAYING` o `JA_CHANNEL_PAUSED`.
+

stb_vorbis.c

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