diff --git a/CMakeLists.txt b/CMakeLists.txt
index 15ff9a0a..263942e3 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -106,8 +106,11 @@ set (SOURCES
     src/visited.c
     src/visited.h
     # Audio playback:
+    src/audio/buf.c
+    src/audio/buf.h
     src/audio/player.c
     src/audio/player.h
+    src/audio/stb_vorbis.c
     # User interface:
     src/ui/color.c
     src/ui/color.h
diff --git a/src/audio/buf.c b/src/audio/buf.c
new file mode 100644
index 00000000..e61164d4
--- /dev/null
+++ b/src/audio/buf.c
@@ -0,0 +1,107 @@
+/* Copyright 2020 Jaakko Keränen <jaakko.keranen@iki.fi>
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are met:
+
+1. Redistributions of source code must retain the above copyright notice, this
+   list of conditions and the following disclaimer.
+2. Redistributions in binary form must reproduce the above copyright notice,
+   this list of conditions and the following disclaimer in the documentation
+   and/or other materials provided with the distribution.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
+ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR
+ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */
+
+#include "buf.h"
+
+iDefineTypeConstruction(InputBuf)
+
+void init_InputBuf(iInputBuf *d) {
+    init_Mutex(&d->mtx);
+    init_Condition(&d->changed);
+    init_Block(&d->data, 0);
+    d->isComplete = iTrue;
+}
+
+void deinit_InputBuf(iInputBuf *d) {
+    deinit_Block(&d->data);
+    deinit_Condition(&d->changed);
+    deinit_Mutex(&d->mtx);
+}
+
+size_t size_InputBuf(const iInputBuf *d) {
+    return size_Block(&d->data);
+}
+
+/*----------------------------------------------------------------------------------------------*/
+
+iDefineTypeConstructionArgs(SampleBuf, (SDL_AudioFormat format, size_t numChannels, size_t count),
+                            format, numChannels, count)
+
+void init_SampleBuf(iSampleBuf *d, SDL_AudioFormat format, size_t numChannels, size_t count) {
+    d->format      = format;
+    d->numChannels = numChannels;
+    d->sampleSize  = SDL_AUDIO_BITSIZE(format) / 8 * numChannels;
+    d->count       = count + 1; /* considered empty if head==tail */
+    d->data        = malloc(d->sampleSize * d->count);
+    d->head        = 0;
+    d->tail        = 0;
+    init_Condition(&d->moreNeeded);
+}
+
+void deinit_SampleBuf(iSampleBuf *d) {
+    deinit_Condition(&d->moreNeeded);
+    free(d->data);
+}
+
+size_t size_SampleBuf(const iSampleBuf *d) {
+    return d->head - d->tail;
+}
+
+size_t vacancy_SampleBuf(const iSampleBuf *d) {
+    return d->count - size_SampleBuf(d) - 1;
+}
+
+iBool isFull_SampleBuf(const iSampleBuf *d) {
+    return vacancy_SampleBuf(d) == 0;
+}
+
+void write_SampleBuf(iSampleBuf *d, const void *samples, const size_t n) {
+    iAssert(n <= vacancy_SampleBuf(d));
+    const size_t headPos = d->head % d->count;
+    const size_t avail   = d->count - headPos;
+    if (n > avail) {
+        const char *in = samples;
+        memcpy(ptr_SampleBuf_(d, headPos), in, d->sampleSize * avail);
+        in += d->sampleSize * avail;
+        memcpy(ptr_SampleBuf_(d, 0), in, d->sampleSize * (n - avail));
+    }
+    else {
+        memcpy(ptr_SampleBuf_(d, headPos), samples, d->sampleSize * n);
+    }
+    d->head += n;
+}
+
+void read_SampleBuf(iSampleBuf *d, const size_t n, void *samples_out) {
+    iAssert(n <= size_SampleBuf(d));
+    const size_t tailPos = d->tail % d->count;
+    const size_t avail   = d->count - tailPos;
+    if (n > avail) {
+        char *out = samples_out;
+        memcpy(out, ptr_SampleBuf_(d, tailPos), d->sampleSize * avail);
+        out += d->sampleSize * avail;
+        memcpy(out, ptr_SampleBuf_(d, 0), d->sampleSize * (n - avail));
+    }
+    else {
+        memcpy(samples_out, ptr_SampleBuf_(d, tailPos), d->sampleSize * n);
+    }
+    d->tail += n;
+}
diff --git a/src/audio/buf.h b/src/audio/buf.h
new file mode 100644
index 00000000..de123481
--- /dev/null
+++ b/src/audio/buf.h
@@ -0,0 +1,74 @@
+/* Copyright 2020 Jaakko Keränen <jaakko.keranen@iki.fi>
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are met:
+
+1. Redistributions of source code must retain the above copyright notice, this
+   list of conditions and the following disclaimer.
+2. Redistributions in binary form must reproduce the above copyright notice,
+   this list of conditions and the following disclaimer in the documentation
+   and/or other materials provided with the distribution.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
+ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR
+ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */
+
+#pragma once
+
+#include "the_Foundation/block.h"
+#include "the_Foundation/mutex.h"
+
+#include <SDL_audio.h>
+
+iDeclareType(InputBuf)
+iDeclareType(SampleBuf)
+
+#if !defined (AUDIO_S24LSB)
+#   define AUDIO_S24LSB     0x8018  /* 24-bit integer samples */
+#endif
+#if !defined (AUDIO_F64LSB)
+#   define AUDIO_F64LSB     0x8140  /* 64-bit floating point samples */
+#endif
+
+struct Impl_InputBuf {
+    iMutex     mtx;
+    iCondition changed;
+    iBlock     data;
+    iBool      isComplete;
+};
+
+iDeclareTypeConstruction(InputBuf)
+
+size_t  size_InputBuf   (const iInputBuf *);
+
+/*----------------------------------------------------------------------------------------------*/
+
+struct Impl_SampleBuf {
+    SDL_AudioFormat format;
+    uint8_t         numChannels;
+    uint8_t         sampleSize; /* as bytes; one sample includes values for all channels */
+    void *          data;
+    size_t          count;
+    size_t          head, tail;
+    iCondition      moreNeeded;
+};
+
+iDeclareTypeConstructionArgs(SampleBuf, SDL_AudioFormat format, size_t numChannels, size_t count)
+
+size_t  size_SampleBuf      (const iSampleBuf *);
+iBool   isFull_SampleBuf    (const iSampleBuf *);
+size_t  vacancy_SampleBuf   (const iSampleBuf *);
+
+iLocalDef void *ptr_SampleBuf_(iSampleBuf *d, size_t pos) {
+    return ((char *) d->data) + (d->sampleSize * pos);
+}
+
+void    write_SampleBuf     (iSampleBuf *, const void *samples, const size_t n);
+void    read_SampleBuf      (iSampleBuf *, const size_t n, void *samples_out);
diff --git a/src/audio/player.c b/src/audio/player.c
index 07f41f01..0825dabd 100644
--- a/src/audio/player.c
+++ b/src/audio/player.c
@@ -21,146 +21,38 @@ ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
 SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */
 
 #include "player.h"
+#include "buf.h"
+
+#define STB_VORBIS_HEADER_ONLY
+#include "stb_vorbis.c"
 
 #include <the_Foundation/buffer.h>
 #include <the_Foundation/thread.h>
 #include <SDL_audio.h>
 
-iDeclareType(InputBuf)
-
-#if !defined (AUDIO_S24LSB)
-#   define AUDIO_S24LSB     0x8018  /* 24-bit integer samples */
-#endif
-#if !defined (AUDIO_F64LSB)
-#   define AUDIO_F64LSB     0x8140  /* 64-bit floating point samples */
-#endif
-
-struct Impl_InputBuf {
-    iMutex     mtx;
-    iCondition changed;
-    iBlock     data;
-    iBool      isComplete;
-};
-
-void init_InputBuf(iInputBuf *d) {
-    init_Mutex(&d->mtx);
-    init_Condition(&d->changed);
-    init_Block(&d->data, 0);
-    d->isComplete = iTrue;
-}
-
-void deinit_InputBuf(iInputBuf *d) {
-    deinit_Block(&d->data);
-    deinit_Condition(&d->changed);
-    deinit_Mutex(&d->mtx);
-}
-
-size_t size_InputBuf(const iInputBuf *d) {
-    return size_Block(&d->data);
-}
-
-iDefineTypeConstruction(InputBuf)
-
-/*----------------------------------------------------------------------------------------------*/
-
-iDeclareType(SampleBuf)
-
-struct Impl_SampleBuf {
-    SDL_AudioFormat format;
-    uint8_t         numChannels;
-    uint8_t         sampleSize; /* as bytes; one sample includes values for all channels */
-    void *          data;
-    size_t          count;
-    size_t          head, tail;
-    iCondition      moreNeeded;
-};
-
-void init_SampleBuf(iSampleBuf *d, SDL_AudioFormat format, size_t numChannels, size_t count) {
-    d->format      = format;
-    d->numChannels = numChannels;
-    d->sampleSize  = SDL_AUDIO_BITSIZE(format) / 8 * numChannels;
-    d->count       = count + 1; /* considered empty if head==tail */
-    d->data        = malloc(d->sampleSize * d->count);
-    d->head        = 0;
-    d->tail        = 0;
-    init_Condition(&d->moreNeeded);
-}
-
-void deinit_SampleBuf(iSampleBuf *d) {
-    deinit_Condition(&d->moreNeeded);
-    free(d->data);
-}
-
-size_t size_SampleBuf(const iSampleBuf *d) {
-    return d->head - d->tail;
-}
-
-size_t vacancy_SampleBuf(const iSampleBuf *d) {
-    return d->count - size_SampleBuf(d) - 1;
-}
-
-iBool isFull_SampleBuf(const iSampleBuf *d) {
-    return vacancy_SampleBuf(d) == 0;
-}
-
-iLocalDef void *ptr_SampleBuf_(iSampleBuf *d, size_t pos) {
-    return ((char *) d->data) + (d->sampleSize * pos);
-}
-
-void write_SampleBuf(iSampleBuf *d, const void *samples, const size_t n) {
-    iAssert(n <= vacancy_SampleBuf(d));
-    const size_t headPos = d->head % d->count;
-    const size_t avail   = d->count - headPos;
-    if (n > avail) {
-        const char *in = samples;
-        memcpy(ptr_SampleBuf_(d, headPos), in, d->sampleSize * avail);
-        in += d->sampleSize * avail;
-        memcpy(ptr_SampleBuf_(d, 0), in, d->sampleSize * (n - avail));
-    }
-    else {
-        memcpy(ptr_SampleBuf_(d, headPos), samples, d->sampleSize * n);
-    }
-    d->head += n;
-}
-
-void read_SampleBuf(iSampleBuf *d, const size_t n, void *samples_out) {
-    iAssert(n <= size_SampleBuf(d));
-    const size_t tailPos = d->tail % d->count;
-    const size_t avail   = d->count - tailPos;
-    if (n > avail) {
-        char *out = samples_out;
-        memcpy(out, ptr_SampleBuf_(d, tailPos), d->sampleSize * avail);
-        out += d->sampleSize * avail;
-        memcpy(out, ptr_SampleBuf_(d, 0), d->sampleSize * (n - avail));
-    }
-    else {
-        memcpy(samples_out, ptr_SampleBuf_(d, tailPos), d->sampleSize * n);
-    }
-    d->tail += n;
-}
-
 /*----------------------------------------------------------------------------------------------*/
 
 iDeclareType(ContentSpec)
 
-struct Impl_ContentSpec {
-    SDL_AudioFormat inputFormat;
-    SDL_AudioSpec   output;
-    size_t          totalInputSize;
-    uint64_t        totalSamples;
-    iRanges         wavData;
-};
-
-iDeclareType(Decoder)
-
 enum iDecoderType {
     none_DecoderType,
     wav_DecoderType,
-    mpeg_DecoderType,
     vorbis_DecoderType,
+    mpeg_DecoderType,
     midi_DecoderType,
 };
 
+struct Impl_ContentSpec {
+    enum iDecoderType type;
+    SDL_AudioFormat   inputFormat;
+    SDL_AudioSpec     output;
+    size_t            totalInputSize;
+    uint64_t          totalSamples;
+    iRanges           dataRange;
+};
+
+iDeclareType(Decoder)
+
 struct Impl_Decoder {
     enum iDecoderType type;
     float             gain;
@@ -176,12 +68,12 @@ struct Impl_Decoder {
     iRanges           wavData;
 };
 
-enum iDecoderParseStatus {
-    ok_DecoderParseStatus,
-    needMoreInput_DecoderParseStatus,
+enum iDecoderStatus {
+    ok_DecoderStatus,
+    needMoreInput_DecoderStatus,
 };
 
-static enum iDecoderParseStatus parseWav_Decoder_(iDecoder *d, iRanges inputRange) {
+static enum iDecoderStatus decodeWav_Decoder_(iDecoder *d, iRanges inputRange) {
     const uint8_t numChannels     = d->output.numChannels;
     const size_t  inputSampleSize = numChannels * SDL_AUDIO_BITSIZE(d->inputFormat) / 8;
     const size_t  vacancy         = vacancy_SampleBuf(&d->output);
@@ -189,11 +81,11 @@ static enum iDecoderParseStatus parseWav_Decoder_(iDecoder *d, iRanges inputRang
     const size_t  avail =
         iMin(inputRange.end - inputBytePos, d->wavData.end - inputBytePos) / inputSampleSize;
     if (avail == 0) {
-        return needMoreInput_DecoderParseStatus;
+        return needMoreInput_DecoderStatus;
     }
     const size_t n = iMin(vacancy, avail);
     if (n == 0) {
-        return ok_DecoderParseStatus;
+        return ok_DecoderStatus;
     }
     void *samples = malloc(inputSampleSize * n);
     /* Get a copy of the input for mixing. */ {
@@ -259,7 +151,7 @@ static enum iDecoderParseStatus parseWav_Decoder_(iDecoder *d, iRanges inputRang
     iGuardMutex(&d->outputMutex, write_SampleBuf(&d->output, samples, n));
     d->currentSample += n;
     free(samples);
-    return ok_DecoderParseStatus;
+    return ok_DecoderStatus;
 }
 
 static iThreadResult run_Decoder_(iThread *thread) {
@@ -273,17 +165,17 @@ static iThreadResult run_Decoder_(iThread *thread) {
         iAssert(inputRange.start <= inputRange.end);
         if (!d->type) break;
         /* Have data to work on and a place to save output? */
-        enum iDecoderParseStatus status = ok_DecoderParseStatus;
+        enum iDecoderStatus status = ok_DecoderStatus;
         if (!isEmpty_Range(&inputRange)) {
             switch (d->type) {
                 case wav_DecoderType:
-                    status = parseWav_Decoder_(d, inputRange);
+                    status = decodeWav_Decoder_(d, inputRange);
                     break;
                 default:
                     break;
             }
         }
-        if (status == needMoreInput_DecoderParseStatus) {
+        if (status == needMoreInput_DecoderStatus) {
             lock_Mutex(&d->input->mtx);
             if (size_InputBuf(d->input) == inputSize) {
                 wait_Condition(&d->input->changed, &d->input->mtx);
@@ -302,10 +194,10 @@ static iThreadResult run_Decoder_(iThread *thread) {
 }
 
 void init_Decoder(iDecoder *d, iInputBuf *input, const iContentSpec *spec) {
-    d->type        = wav_DecoderType;
+    d->type        = spec->type;
     d->gain        = 0.5f;
     d->input       = input;
-    d->inputPos    = spec->wavData.start;
+    d->inputPos    = spec->dataRange.start;
     d->inputFormat = spec->inputFormat;
     d->totalInputSize = spec->totalInputSize;
     init_SampleBuf(&d->output,
@@ -330,16 +222,22 @@ void deinit_Decoder(iDecoder *d) {
     deinit_SampleBuf(&d->output);
 }
 
+static void start_Decoder_(iDecoder *d) {
+    if (!d->thread && d->type != none_DecoderType) {
+    }
+}
+
 iDefineTypeConstructionArgs(Decoder, (iInputBuf *input, const iContentSpec *spec),
                             input, spec)
 
 /*----------------------------------------------------------------------------------------------*/
 
 struct Impl_Player {
-    SDL_AudioSpec spec;
+    SDL_AudioSpec     spec;
     SDL_AudioDeviceID device;
-    iInputBuf *data;
-    iDecoder *decoder;
+    iString           mime;
+    iInputBuf *       data;
+    iDecoder *        decoder;
 };
 
 iDefineTypeConstruction(Player)
@@ -358,8 +256,8 @@ static iContentSpec contentSpec_Player_(const iPlayer *d) {
     const size_t dataSize = size_InputBuf(d->data);
     iBuffer *buf = iClob(new_Buffer());
     open_Buffer(buf, &d->data->data);
-    enum iDecoderType decType = wav_DecoderType; /* TODO: from MIME */
-    if (decType == wav_DecoderType && dataSize >= 44) {
+    content.type = wav_DecoderType; /* TODO: from MIME */
+    if (content.type == wav_DecoderType && dataSize >= 44) {
         /* Read the RIFF/WAVE header. */
         iStream *is = stream_Buffer(buf);
         char magic[4];
@@ -428,8 +326,8 @@ static iContentSpec contentSpec_Player_(const iPlayer *d) {
                 }
             }
             else if (memcmp(magic, "data", 4) == 0) {
-                content.wavData      = (iRanges){ pos_Stream(is), pos_Stream(is) + size };
-                content.totalSamples = (uint64_t) size_Range(&content.wavData) / blockAlign;
+                content.dataRange    = (iRanges){ pos_Stream(is), pos_Stream(is) + size };
+                content.totalSamples = (uint64_t) size_Range(&content.dataRange) / blockAlign;
                 break;
             }
             else {
@@ -462,6 +360,7 @@ static void writeOutputSamples_Player_(void *plr, Uint8 *stream, int len) {
 
 void init_Player(iPlayer *d) {
     iZap(d->spec);
+    init_String(&d->mime);
     d->device  = 0;
     d->decoder = NULL;
     d->data    = new_InputBuf();
@@ -470,6 +369,7 @@ void init_Player(iPlayer *d) {
 void deinit_Player(iPlayer *d) {
     stop_Player(d);
     delete_InputBuf(d->data);
+    deinit_String(&d->mime);
 }
 
 iBool isStarted_Player(const iPlayer *d) {
@@ -481,13 +381,14 @@ iBool isPaused_Player(const iPlayer *d) {
     return SDL_GetAudioDeviceStatus(d->device) == SDL_AUDIO_PAUSED;
 }
 
-void setFormatHint_Player(iPlayer *d, const char *hint) {
-}
-
-void updateSourceData_Player(iPlayer *d, const iBlock *data, enum iPlayerUpdate update) {
+void updateSourceData_Player(iPlayer *d, const iString *mimeType, const iBlock *data,
+                             enum iPlayerUpdate update) {
     /* TODO: Add MIME as argument */
     iInputBuf *input = d->data;
     lock_Mutex(&input->mtx);
+    if (mimeType) {
+        set_String(&d->mime, mimeType);
+    }
     switch (update) {
         case replace_PlayerUpdate:
             set_Block(&input->data, data);
@@ -515,7 +416,7 @@ iBool start_Player(iPlayer *d) {
     if (isStarted_Player(d)) {
         return iFalse;
     }
-    iContentSpec content  = contentSpec_Player_(d);
+    iContentSpec content    = contentSpec_Player_(d);
     content.output.callback = writeOutputSamples_Player_;
     content.output.userdata = d;
     d->device = SDL_OpenAudioDevice(NULL, SDL_FALSE /* playback */, &content.output, &d->spec, 0);
diff --git a/src/audio/player.h b/src/audio/player.h
index fe6717b0..720f2d78 100644
--- a/src/audio/player.h
+++ b/src/audio/player.h
@@ -33,8 +33,8 @@ enum iPlayerUpdate {
     complete_PlayerUpdate,
 };
 
-void    setFormatHint_Player    (iPlayer *, const char *hint);
-void    updateSourceData_Player (iPlayer *, const iBlock *data, enum iPlayerUpdate update);
+void    updateSourceData_Player (iPlayer *, const iString *mimeType, const iBlock *data,
+                                 enum iPlayerUpdate update);
 
 iBool   start_Player            (iPlayer *);
 void    setPaused_Player        (iPlayer *, iBool isPaused);
diff --git a/src/audio/stb_vorbis.c b/src/audio/stb_vorbis.c
new file mode 100644
index 00000000..a8cbfa6c
--- /dev/null
+++ b/src/audio/stb_vorbis.c
@@ -0,0 +1,5563 @@
+// Ogg Vorbis audio decoder - v1.20 - public domain
+// http://nothings.org/stb_vorbis/
+//
+// Original version written by Sean Barrett in 2007.
+//
+// Originally sponsored by RAD Game Tools. Seeking implementation
+// sponsored by Phillip Bennefall, Marc Andersen, Aaron Baker,
+// Elias Software, Aras Pranckevicius, and Sean Barrett.
+//
+// LICENSE
+//
+//   See end of file for license information.
+//
+// Limitations:
+//
+//   - floor 0 not supported (used in old ogg vorbis files pre-2004)
+//   - lossless sample-truncation at beginning ignored
+//   - cannot concatenate multiple vorbis streams
+//   - sample positions are 32-bit, limiting seekable 192Khz
+//       files to around 6 hours (Ogg supports 64-bit)
+//
+// Feature contributors:
+//    Dougall Johnson (sample-exact seeking)
+//
+// Bugfix/warning contributors:
+//    Terje Mathisen     Niklas Frykholm     Andy Hill
+//    Casey Muratori     John Bolton         Gargaj
+//    Laurent Gomila     Marc LeBlanc        Ronny Chevalier
+//    Bernhard Wodo      Evan Balster        github:alxprd
+//    Tom Beaumont       Ingo Leitgeb        Nicolas Guillemot
+//    Phillip Bennefall  Rohit               Thiago Goulart
+//    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
+//
+// Partial history:
+//    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.
+//    1.17    - 2019-07-08 - fix CVE-2019-13217..CVE-2019-13223 (by ForAllSecure)
+//    1.16    - 2019-03-04 - fix warnings
+//    1.15    - 2019-02-07 - explicit failure if Ogg Skeleton data is found
+//    1.14    - 2018-02-11 - delete bogus dealloca usage
+//    1.13    - 2018-01-29 - fix truncation of last frame (hopefully)
+//    1.12    - 2017-11-21 - limit residue begin/end to blocksize/2 to avoid large temp allocs in bad/corrupt files
+//    1.11    - 2017-07-23 - fix MinGW compilation
+//    1.10    - 2017-03-03 - more robust seeking; fix negative ilog(); clear error in open_memory
+//    1.09    - 2016-04-04 - back out 'truncation of last frame' fix from previous version
+//    1.08    - 2016-04-02 - warnings; setup memory leaks; truncation of last frame
+//    1.07    - 2015-01-16 - fixes for crashes on invalid files; warning fixes; const
+//    1.06    - 2015-08-31 - full, correct support for seeking API (Dougall Johnson)
+//                           some crash fixes when out of memory or with corrupt files
+//                           fix some inappropriately signed shifts
+//    1.05    - 2015-04-19 - don't define __forceinline if it's redundant
+//    1.04    - 2014-08-27 - fix missing const-correct case in API
+//    1.03    - 2014-08-07 - warning fixes
+//    1.02    - 2014-07-09 - declare qsort comparison as explicitly _cdecl in Windows
+//    1.01    - 2014-06-18 - fix stb_vorbis_get_samples_float (interleaved was correct)
+//    1.0     - 2014-05-26 - fix memory leaks; fix warnings; fix bugs in >2-channel;
+//                           (API change) report sample rate for decode-full-file funcs
+//
+// See end of file for full version history.
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+//  HEADER BEGINS HERE
+//
+
+#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H
+#define STB_VORBIS_INCLUDE_STB_VORBIS_H
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+#define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+///////////   THREAD SAFETY
+
+// Individual stb_vorbis* handles are not thread-safe; you cannot decode from
+// them from multiple threads at the same time. However, you can have multiple
+// stb_vorbis* handles and decode from them independently in multiple thrads.
+
+
+///////////   MEMORY ALLOCATION
+
+// normally stb_vorbis uses malloc() to allocate memory at startup,
+// and alloca() to allocate temporary memory during a frame on the
+// stack. (Memory consumption will depend on the amount of setup
+// data in the file and how you set the compile flags for speed
+// vs. size. In my test files the maximal-size usage is ~150KB.)
+//
+// You can modify the wrapper functions in the source (setup_malloc,
+// setup_temp_malloc, temp_malloc) to change this behavior, or you
+// can use a simpler allocation model: you pass in a buffer from
+// which stb_vorbis will allocate _all_ its memory (including the
+// temp memory). "open" may fail with a VORBIS_outofmem if you
+// do not pass in enough data; there is no way to determine how
+// much you do need except to succeed (at which point you can
+// query get_info to find the exact amount required. yes I know
+// this is lame).
+//
+// If you pass in a non-NULL buffer of the type below, allocation
+// will occur from it as described above. Otherwise just pass NULL
+// to use malloc()/alloca()
+
+typedef struct
+{
+   char *alloc_buffer;
+   int   alloc_buffer_length_in_bytes;
+} stb_vorbis_alloc;
+
+
+///////////   FUNCTIONS USEABLE WITH ALL INPUT MODES
+
+typedef struct stb_vorbis stb_vorbis;
+
+typedef struct
+{
+   unsigned int sample_rate;
+   int channels;
+
+   unsigned int setup_memory_required;
+   unsigned int setup_temp_memory_required;
+   unsigned int temp_memory_required;
+
+   int max_frame_size;
+} stb_vorbis_info;
+
+typedef struct
+{
+   char *vendor;
+
+   int comment_list_length;
+   char **comment_list;
+} stb_vorbis_comment;
+
+// get general information about the file
+extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f);
+
+// get ogg comments
+extern stb_vorbis_comment stb_vorbis_get_comment(stb_vorbis *f);
+
+// get the last error detected (clears it, too)
+extern int stb_vorbis_get_error(stb_vorbis *f);
+
+// close an ogg vorbis file and free all memory in use
+extern void stb_vorbis_close(stb_vorbis *f);
+
+// this function returns the offset (in samples) from the beginning of the
+// file that will be returned by the next decode, if it is known, or -1
+// otherwise. after a flush_pushdata() call, this may take a while before
+// it becomes valid again.
+// NOT WORKING YET after a seek with PULLDATA API
+extern int stb_vorbis_get_sample_offset(stb_vorbis *f);
+
+// returns the current seek point within the file, or offset from the beginning
+// of the memory buffer. In pushdata mode it returns 0.
+extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f);
+
+///////////   PUSHDATA API
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+
+// this API allows you to get blocks of data from any source and hand
+// them to stb_vorbis. you have to buffer them; stb_vorbis will tell
+// you how much it used, and you have to give it the rest next time;
+// and stb_vorbis may not have enough data to work with and you will
+// need to give it the same data again PLUS more. Note that the Vorbis
+// specification does not bound the size of an individual frame.
+
+extern stb_vorbis *stb_vorbis_open_pushdata(
+         const unsigned char * datablock, int datablock_length_in_bytes,
+         int *datablock_memory_consumed_in_bytes,
+         int *error,
+         const stb_vorbis_alloc *alloc_buffer);
+// create a vorbis decoder by passing in the initial data block containing
+//    the ogg&vorbis headers (you don't need to do parse them, just provide
+//    the first N bytes of the file--you're told if it's not enough, see below)
+// on success, returns an stb_vorbis *, does not set error, returns the amount of
+//    data parsed/consumed on this call in *datablock_memory_consumed_in_bytes;
+// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed
+// if returns NULL and *error is VORBIS_need_more_data, then the input block was
+//       incomplete and you need to pass in a larger block from the start of the file
+
+extern int stb_vorbis_decode_frame_pushdata(
+         stb_vorbis *f,
+         const unsigned char *datablock, int datablock_length_in_bytes,
+         int *channels,             // place to write number of float * buffers
+         float ***output,           // place to write float ** array of float * buffers
+         int *samples               // place to write number of output samples
+     );
+// decode a frame of audio sample data if possible from the passed-in data block
+//
+// return value: number of bytes we used from datablock
+//
+// possible cases:
+//     0 bytes used, 0 samples output (need more data)
+//     N bytes used, 0 samples output (resynching the stream, keep going)
+//     N bytes used, M samples output (one frame of data)
+// note that after opening a file, you will ALWAYS get one N-bytes,0-sample
+// frame, because Vorbis always "discards" the first frame.
+//
+// Note that on resynch, stb_vorbis will rarely consume all of the buffer,
+// instead only datablock_length_in_bytes-3 or less. This is because it wants
+// to avoid missing parts of a page header if they cross a datablock boundary,
+// without writing state-machiney code to record a partial detection.
+//
+// The number of channels returned are stored in *channels (which can be
+// NULL--it is always the same as the number of channels reported by
+// get_info). *output will contain an array of float* buffers, one per
+// 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.
+
+extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
+// inform stb_vorbis that your next datablock will not be contiguous with
+// previous ones (e.g. you've seeked in the data); future attempts to decode
+// frames will cause stb_vorbis to resynchronize (as noted above), and
+// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it
+// will begin decoding the _next_ frame.
+//
+// if you want to seek using pushdata, you need to seek in your file, then
+// call stb_vorbis_flush_pushdata(), then start calling decoding, then once
+// decoding is returning you data, call stb_vorbis_get_sample_offset, and
+// if you don't like the result, seek your file again and repeat.
+#endif
+
+
+//////////   PULLING INPUT API
+
+#ifndef STB_VORBIS_NO_PULLDATA_API
+// This API assumes stb_vorbis is allowed to pull data from a source--
+// either a block of memory containing the _entire_ vorbis stream, or a
+// FILE * that you or it create, or possibly some other reading mechanism
+// if you go modify the source to replace the FILE * case with some kind
+// of callback to your code. (But if you don't support seeking, you may
+// just want to go ahead and use pushdata.)
+
+#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+extern int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output);
+#endif
+#if !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+extern int stb_vorbis_decode_memory(const unsigned char *mem, int len, int *channels, int *sample_rate, short **output);
+#endif
+// decode an entire file and output the data interleaved into a malloc()ed
+// buffer stored in *output. The return value is the number of samples
+// decoded, or -1 if the file could not be opened or was not an ogg vorbis file.
+// When you're done with it, just free() the pointer returned in *output.
+
+extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len,
+                                  int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from an ogg vorbis stream in memory (note
+// this must be the entire stream!). on failure, returns NULL and sets *error
+
+#ifndef STB_VORBIS_NO_STDIO
+extern stb_vorbis * stb_vorbis_open_filename(const char *filename,
+                                  int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from a filename via fopen(). on failure,
+// returns NULL and sets *error (possibly to VORBIS_file_open_failure).
+
+extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close,
+                                  int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+// the _current_ seek point (ftell). on failure, returns NULL and sets *error.
+// note that stb_vorbis must "own" this stream; if you seek it in between
+// calls to stb_vorbis, it will become confused. Moreover, if you attempt to
+// perform stb_vorbis_seek_*() operations on this file, it will assume it
+// owns the _entire_ rest of the file after the start point. Use the next
+// function, stb_vorbis_open_file_section(), to limit it.
+
+extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close,
+                int *error, const stb_vorbis_alloc *alloc_buffer, unsigned int len);
+// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+// the _current_ seek point (ftell); the stream will be of length 'len' bytes.
+// on failure, returns NULL and sets *error. note that stb_vorbis must "own"
+// this stream; if you seek it in between calls to stb_vorbis, it will become
+// confused.
+#endif
+
+extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number);
+extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number);
+// these functions seek in the Vorbis file to (approximately) 'sample_number'.
+// after calling seek_frame(), the next call to get_frame_*() will include
+// the specified sample. after calling stb_vorbis_seek(), the next call to
+// stb_vorbis_get_samples_* will start with the specified sample. If you
+// do not need to seek to EXACTLY the target sample when using get_samples_*,
+// you can also use seek_frame().
+
+extern int stb_vorbis_seek_start(stb_vorbis *f);
+// this function is equivalent to stb_vorbis_seek(f,0)
+
+extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f);
+extern float        stb_vorbis_stream_length_in_seconds(stb_vorbis *f);
+// these functions return the total length of the vorbis stream
+
+extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output);
+// decode the next frame and return the number of samples. the number of
+// channels returned are stored in *channels (which can be NULL--it is always
+// the same as the number of channels reported by get_info). *output will
+// contain an array of float* buffers, one per channel. These outputs will
+// be overwritten on the next call to stb_vorbis_get_frame_*.
+//
+// You generally should not intermix calls to stb_vorbis_get_frame_*()
+// and stb_vorbis_get_samples_*(), since the latter calls the former.
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts);
+extern int stb_vorbis_get_frame_short            (stb_vorbis *f, int num_c, short **buffer, int num_samples);
+#endif
+// decode the next frame and return the number of *samples* per channel.
+// Note that for interleaved data, you pass in the number of shorts (the
+// size of your array), but the return value is the number of samples per
+// channel, not the total number of samples.
+//
+// The data is coerced to the number of channels you request according to the
+// channel coercion rules (see below). You must pass in the size of your
+// buffer(s) so that stb_vorbis will not overwrite the end of the buffer.
+// The maximum buffer size needed can be gotten from get_info(); however,
+// the Vorbis I specification implies an absolute maximum of 4096 samples
+// per channel.
+
+// Channel coercion rules:
+//    Let M be the number of channels requested, and N the number of channels present,
+//    and Cn be the nth channel; let stereo L be the sum of all L and center channels,
+//    and stereo R be the sum of all R and center channels (channel assignment from the
+//    vorbis spec).
+//        M    N       output
+//        1    k      sum(Ck) for all k
+//        2    *      stereo L, stereo R
+//        k    l      k > l, the first l channels, then 0s
+//        k    l      k <= l, the first k channels
+//    Note that this is not _good_ surround etc. mixing at all! It's just so
+//    you get something useful.
+
+extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats);
+extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples);
+// gets num_samples samples, not necessarily on a frame boundary--this requires
+// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES.
+// Returns the number of samples stored per channel; it may be less than requested
+// at the end of the file. If there are no more samples in the file, returns 0.
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts);
+extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples);
+#endif
+// gets num_samples samples, not necessarily on a frame boundary--this requires
+// buffering so you have to supply the buffers. Applies the coercion rules above
+// to produce 'channels' channels. Returns the number of samples stored per channel;
+// it may be less than requested at the end of the file. If there are no more
+// samples in the file, returns 0.
+
+#endif
+
+////////   ERROR CODES
+
+enum STBVorbisError
+{
+   VORBIS__no_error,
+
+   VORBIS_need_more_data=1,             // not a real error
+
+   VORBIS_invalid_api_mixing,           // can't mix API modes
+   VORBIS_outofmem,                     // not enough memory
+   VORBIS_feature_not_supported,        // uses floor 0
+   VORBIS_too_many_channels,            // STB_VORBIS_MAX_CHANNELS is too small
+   VORBIS_file_open_failure,            // fopen() failed
+   VORBIS_seek_without_length,          // can't seek in unknown-length file
+
+   VORBIS_unexpected_eof=10,            // file is truncated?
+   VORBIS_seek_invalid,                 // seek past EOF
+
+   // decoding errors (corrupt/invalid stream) -- you probably
+   // don't care about the exact details of these
+
+   // vorbis errors:
+   VORBIS_invalid_setup=20,
+   VORBIS_invalid_stream,
+
+   // ogg errors:
+   VORBIS_missing_capture_pattern=30,
+   VORBIS_invalid_stream_structure_version,
+   VORBIS_continued_packet_flag_invalid,
+   VORBIS_incorrect_stream_serial_number,
+   VORBIS_invalid_first_page,
+   VORBIS_bad_packet_type,
+   VORBIS_cant_find_last_page,
+   VORBIS_seek_failed,
+   VORBIS_ogg_skeleton_not_supported
+};
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H
+//
+//  HEADER ENDS HERE
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#ifndef STB_VORBIS_HEADER_ONLY
+
+// global configuration settings (e.g. set these in the project/makefile),
+// or just set them in this file at the top (although ideally the first few
+// should be visible when the header file is compiled too, although it's not
+// crucial)
+
+// STB_VORBIS_NO_PUSHDATA_API
+//     does not compile the code for the various stb_vorbis_*_pushdata()
+//     functions
+// #define STB_VORBIS_NO_PUSHDATA_API
+
+// STB_VORBIS_NO_PULLDATA_API
+//     does not compile the code for the non-pushdata APIs
+// #define STB_VORBIS_NO_PULLDATA_API
+
+// STB_VORBIS_NO_STDIO
+//     does not compile the code for the APIs that use FILE *s internally
+//     or externally (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_STDIO
+
+// STB_VORBIS_NO_INTEGER_CONVERSION
+//     does not compile the code for converting audio sample data from
+//     float to integer (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_INTEGER_CONVERSION
+
+// STB_VORBIS_NO_FAST_SCALED_FLOAT
+//      does not use a fast float-to-int trick to accelerate float-to-int on
+//      most platforms which requires endianness be defined correctly.
+//#define STB_VORBIS_NO_FAST_SCALED_FLOAT
+
+
+// STB_VORBIS_MAX_CHANNELS [number]
+//     globally define this to the maximum number of channels you need.
+//     The spec does not put a restriction on channels except that
+//     the count is stored in a byte, so 255 is the hard limit.
+//     Reducing this saves about 16 bytes per value, so using 16 saves
+//     (255-16)*16 or around 4KB. Plus anything other memory usage
+//     I forgot to account for. Can probably go as low as 8 (7.1 audio),
+//     6 (5.1 audio), or 2 (stereo only).
+#ifndef STB_VORBIS_MAX_CHANNELS
+#define STB_VORBIS_MAX_CHANNELS    16  // enough for anyone?
+#endif
+
+// STB_VORBIS_PUSHDATA_CRC_COUNT [number]
+//     after a flush_pushdata(), stb_vorbis begins scanning for the
+//     next valid page, without backtracking. when it finds something
+//     that looks like a page, it streams through it and verifies its
+//     CRC32. Should that validation fail, it keeps scanning. But it's
+//     possible that _while_ streaming through to check the CRC32 of
+//     one candidate page, it sees another candidate page. This #define
+//     determines how many "overlapping" candidate pages it can search
+//     at once. Note that "real" pages are typically ~4KB to ~8KB, whereas
+//     garbage pages could be as big as 64KB, but probably average ~16KB.
+//     So don't hose ourselves by scanning an apparent 64KB page and
+//     missing a ton of real ones in the interim; so minimum of 2
+#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT
+#define STB_VORBIS_PUSHDATA_CRC_COUNT  4
+#endif
+
+// STB_VORBIS_FAST_HUFFMAN_LENGTH [number]
+//     sets the log size of the huffman-acceleration table.  Maximum
+//     supported value is 24. with larger numbers, more decodings are O(1),
+//     but the table size is larger so worse cache missing, so you'll have
+//     to probe (and try multiple ogg vorbis files) to find the sweet spot.
+#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH
+#define STB_VORBIS_FAST_HUFFMAN_LENGTH   10
+#endif
+
+// STB_VORBIS_FAST_BINARY_LENGTH [number]
+//     sets the log size of the binary-search acceleration table. this
+//     is used in similar fashion to the fast-huffman size to set initial
+//     parameters for the binary search
+
+// STB_VORBIS_FAST_HUFFMAN_INT
+//     The fast huffman tables are much more efficient if they can be
+//     stored as 16-bit results instead of 32-bit results. This restricts
+//     the codebooks to having only 65535 possible outcomes, though.
+//     (At least, accelerated by the huffman table.)
+#ifndef STB_VORBIS_FAST_HUFFMAN_INT
+#define STB_VORBIS_FAST_HUFFMAN_SHORT
+#endif
+
+// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+//     If the 'fast huffman' search doesn't succeed, then stb_vorbis falls
+//     back on binary searching for the correct one. This requires storing
+//     extra tables with the huffman codes in sorted order. Defining this
+//     symbol trades off space for speed by forcing a linear search in the
+//     non-fast case, except for "sparse" codebooks.
+// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+
+// STB_VORBIS_DIVIDES_IN_RESIDUE
+//     stb_vorbis precomputes the result of the scalar residue decoding
+//     that would otherwise require a divide per chunk. you can trade off
+//     space for time by defining this symbol.
+// #define STB_VORBIS_DIVIDES_IN_RESIDUE
+
+// STB_VORBIS_DIVIDES_IN_CODEBOOK
+//     vorbis VQ codebooks can be encoded two ways: with every case explicitly
+//     stored, or with all elements being chosen from a small range of values,
+//     and all values possible in all elements. By default, stb_vorbis expands
+//     this latter kind out to look like the former kind for ease of decoding,
+//     because otherwise an integer divide-per-vector-element is required to
+//     unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can
+//     trade off storage for speed.
+//#define STB_VORBIS_DIVIDES_IN_CODEBOOK
+
+#ifdef STB_VORBIS_CODEBOOK_SHORTS
+#error "STB_VORBIS_CODEBOOK_SHORTS is no longer supported as it produced incorrect results for some input formats"
+#endif
+
+// STB_VORBIS_DIVIDE_TABLE
+//     this replaces small integer divides in the floor decode loop with
+//     table lookups. made less than 1% difference, so disabled by default.
+
+// STB_VORBIS_NO_INLINE_DECODE
+//     disables the inlining of the scalar codebook fast-huffman decode.
+//     might save a little codespace; useful for debugging
+// #define STB_VORBIS_NO_INLINE_DECODE
+
+// STB_VORBIS_NO_DEFER_FLOOR
+//     Normally we only decode the floor without synthesizing the actual
+//     full curve. We can instead synthesize the curve immediately. This
+//     requires more memory and is very likely slower, so I don't think
+//     you'd ever want to do it except for debugging.
+// #define STB_VORBIS_NO_DEFER_FLOOR
+
+
+
+
+//////////////////////////////////////////////////////////////////////////////
+
+#ifdef STB_VORBIS_NO_PULLDATA_API
+   #define STB_VORBIS_NO_INTEGER_CONVERSION
+   #define STB_VORBIS_NO_STDIO
+#endif
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+   #define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+
+   // only need endianness for fast-float-to-int, which we don't
+   // use for pushdata
+
+   #ifndef STB_VORBIS_BIG_ENDIAN
+     #define STB_VORBIS_ENDIAN  0
+   #else
+     #define STB_VORBIS_ENDIAN  1
+   #endif
+
+#endif
+#endif
+
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifndef STB_VORBIS_NO_CRT
+   #include <stdlib.h>
+   #include <string.h>
+   #include <assert.h>
+   #include <math.h>
+
+   // find definition of alloca if it's not in stdlib.h:
+   #if defined(_MSC_VER) || defined(__MINGW32__)
+      #include <malloc.h>
+   #endif
+   #if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__) || defined(__NEWLIB__)
+      #include <alloca.h>
+   #endif
+#else // STB_VORBIS_NO_CRT
+   #define NULL 0
+   #define malloc(s)   0
+   #define free(s)     ((void) 0)
+   #define realloc(s)  0
+#endif // STB_VORBIS_NO_CRT
+
+#include <limits.h>
+
+#ifdef __MINGW32__
+   // eff you mingw:
+   //     "fixed":
+   //         http://sourceforge.net/p/mingw-w64/mailman/message/32882927/
+   //     "no that broke the build, reverted, who cares about C":
+   //         http://sourceforge.net/p/mingw-w64/mailman/message/32890381/
+   #ifdef __forceinline
+   #undef __forceinline
+   #endif
+   #define __forceinline
+   #ifndef alloca
+   #define alloca __builtin_alloca
+   #endif
+#elif !defined(_MSC_VER)
+   #if __GNUC__
+      #define __forceinline inline
+   #else
+      #define __forceinline
+   #endif
+#endif
+
+#if STB_VORBIS_MAX_CHANNELS > 256
+#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range"
+#endif
+
+#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24
+#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range"
+#endif
+
+
+#if 0
+#include <crtdbg.h>
+#define CHECK(f)   _CrtIsValidHeapPointer(f->channel_buffers[1])
+#else
+#define CHECK(f)   ((void) 0)
+#endif
+
+#define MAX_BLOCKSIZE_LOG  13   // from specification
+#define MAX_BLOCKSIZE      (1 << MAX_BLOCKSIZE_LOG)
+
+
+typedef unsigned char  uint8;
+typedef   signed char   int8;
+typedef unsigned short uint16;
+typedef   signed short  int16;
+typedef unsigned int   uint32;
+typedef   signed int    int32;
+
+#ifndef TRUE
+#define TRUE 1
+#define FALSE 0
+#endif
+
+typedef float codetype;
+
+// @NOTE
+//
+// Some arrays below are tagged "//varies", which means it's actually
+// a variable-sized piece of data, but rather than malloc I assume it's
+// small enough it's better to just allocate it all together with the
+// main thing
+//
+// Most of the variables are specified with the smallest size I could pack
+// them into. It might give better performance to make them all full-sized
+// integers. It should be safe to freely rearrange the structures or change
+// the sizes larger--nothing relies on silently truncating etc., nor the
+// order of variables.
+
+#define FAST_HUFFMAN_TABLE_SIZE   (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH)
+#define FAST_HUFFMAN_TABLE_MASK   (FAST_HUFFMAN_TABLE_SIZE - 1)
+
+typedef struct
+{
+   int dimensions, entries;
+   uint8 *codeword_lengths;
+   float  minimum_value;
+   float  delta_value;
+   uint8  value_bits;
+   uint8  lookup_type;
+   uint8  sequence_p;
+   uint8  sparse;
+   uint32 lookup_values;
+   codetype *multiplicands;
+   uint32 *codewords;
+   #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+    int16  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+   #else
+    int32  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+   #endif
+   uint32 *sorted_codewords;
+   int    *sorted_values;
+   int     sorted_entries;
+} Codebook;
+
+typedef struct
+{
+   uint8 order;
+   uint16 rate;
+   uint16 bark_map_size;
+   uint8 amplitude_bits;
+   uint8 amplitude_offset;
+   uint8 number_of_books;
+   uint8 book_list[16]; // varies
+} Floor0;
+
+typedef struct
+{
+   uint8 partitions;
+   uint8 partition_class_list[32]; // varies
+   uint8 class_dimensions[16]; // varies
+   uint8 class_subclasses[16]; // varies
+   uint8 class_masterbooks[16]; // varies
+   int16 subclass_books[16][8]; // varies
+   uint16 Xlist[31*8+2]; // varies
+   uint8 sorted_order[31*8+2];
+   uint8 neighbors[31*8+2][2];
+   uint8 floor1_multiplier;
+   uint8 rangebits;
+   int values;
+} Floor1;
+
+typedef union
+{
+   Floor0 floor0;
+   Floor1 floor1;
+} Floor;
+
+typedef struct
+{
+   uint32 begin, end;
+   uint32 part_size;
+   uint8 classifications;
+   uint8 classbook;
+   uint8 **classdata;
+   int16 (*residue_books)[8];
+} Residue;
+
+typedef struct
+{
+   uint8 magnitude;
+   uint8 angle;
+   uint8 mux;
+} MappingChannel;
+
+typedef struct
+{
+   uint16 coupling_steps;
+   MappingChannel *chan;
+   uint8  submaps;
+   uint8  submap_floor[15]; // varies
+   uint8  submap_residue[15]; // varies
+} Mapping;
+
+typedef struct
+{
+   uint8 blockflag;
+   uint8 mapping;
+   uint16 windowtype;
+   uint16 transformtype;
+} Mode;
+
+typedef struct
+{
+   uint32  goal_crc;    // expected crc if match
+   int     bytes_left;  // bytes left in packet
+   uint32  crc_so_far;  // running crc
+   int     bytes_done;  // bytes processed in _current_ chunk
+   uint32  sample_loc;  // granule pos encoded in page
+} CRCscan;
+
+typedef struct
+{
+   uint32 page_start, page_end;
+   uint32 last_decoded_sample;
+} ProbedPage;
+
+struct stb_vorbis
+{
+  // user-accessible info
+   unsigned int sample_rate;
+   int channels;
+
+   unsigned int setup_memory_required;
+   unsigned int temp_memory_required;
+   unsigned int setup_temp_memory_required;
+
+   char *vendor;
+   int comment_list_length;
+   char **comment_list;
+
+  // input config
+#ifndef STB_VORBIS_NO_STDIO
+   FILE *f;
+   uint32 f_start;
+   int close_on_free;
+#endif
+
+   uint8 *stream;
+   uint8 *stream_start;
+   uint8 *stream_end;
+
+   uint32 stream_len;
+
+   uint8  push_mode;
+
+   // the page to seek to when seeking to start, may be zero
+   uint32 first_audio_page_offset;
+
+   // p_first is the page on which the first audio packet ends
+   // (but not necessarily the page on which it starts)
+   ProbedPage p_first, p_last;
+
+  // memory management
+   stb_vorbis_alloc alloc;
+   int setup_offset;
+   int temp_offset;
+
+  // run-time results
+   int eof;
+   enum STBVorbisError error;
+
+  // user-useful data
+
+  // header info
+   int blocksize[2];
+   int blocksize_0, blocksize_1;
+   int codebook_count;
+   Codebook *codebooks;
+   int floor_count;
+   uint16 floor_types[64]; // varies
+   Floor *floor_config;
+   int residue_count;
+   uint16 residue_types[64]; // varies
+   Residue *residue_config;
+   int mapping_count;
+   Mapping *mapping;
+   int mode_count;
+   Mode mode_config[64];  // varies
+
+   uint32 total_samples;
+
+  // decode buffer
+   float *channel_buffers[STB_VORBIS_MAX_CHANNELS];
+   float *outputs        [STB_VORBIS_MAX_CHANNELS];
+
+   float *previous_window[STB_VORBIS_MAX_CHANNELS];
+   int previous_length;
+
+   #ifndef STB_VORBIS_NO_DEFER_FLOOR
+   int16 *finalY[STB_VORBIS_MAX_CHANNELS];
+   #else
+   float *floor_buffers[STB_VORBIS_MAX_CHANNELS];
+   #endif
+
+   uint32 current_loc; // sample location of next frame to decode
+   int    current_loc_valid;
+
+  // per-blocksize precomputed data
+
+   // twiddle factors
+   float *A[2],*B[2],*C[2];
+   float *window[2];
+   uint16 *bit_reverse[2];
+
+  // current page/packet/segment streaming info
+   uint32 serial; // stream serial number for verification
+   int last_page;
+   int segment_count;
+   uint8 segments[255];
+   uint8 page_flag;
+   uint8 bytes_in_seg;
+   uint8 first_decode;
+   int next_seg;
+   int last_seg;  // flag that we're on the last segment
+   int last_seg_which; // what was the segment number of the last seg?
+   uint32 acc;
+   int valid_bits;
+   int packet_bytes;
+   int end_seg_with_known_loc;
+   uint32 known_loc_for_packet;
+   int discard_samples_deferred;
+   uint32 samples_output;
+
+  // push mode scanning
+   int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+   CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT];
+#endif
+
+  // sample-access
+   int channel_buffer_start;
+   int channel_buffer_end;
+};
+
+#if defined(STB_VORBIS_NO_PUSHDATA_API)
+   #define IS_PUSH_MODE(f)   FALSE
+#elif defined(STB_VORBIS_NO_PULLDATA_API)
+   #define IS_PUSH_MODE(f)   TRUE
+#else
+   #define IS_PUSH_MODE(f)   ((f)->push_mode)
+#endif
+
+typedef struct stb_vorbis vorb;
+
+static int error(vorb *f, enum STBVorbisError e)
+{
+   f->error = e;
+   if (!f->eof && e != VORBIS_need_more_data) {
+      f->error=e; // breakpoint for debugging
+   }
+   return 0;
+}
+
+
+// these functions are used for allocating temporary memory
+// while decoding. if you can afford the stack space, use
+// alloca(); otherwise, provide a temp buffer and it will
+// allocate out of those.
+
+#define array_size_required(count,size)  (count*(sizeof(void *)+(size)))
+
+#define temp_alloc(f,size)              (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size))
+#define temp_free(f,p)                  (void)0
+#define temp_alloc_save(f)              ((f)->temp_offset)
+#define temp_alloc_restore(f,p)         ((f)->temp_offset = (p))
+
+#define temp_block_array(f,count,size)  make_block_array(temp_alloc(f,array_size_required(count,size)), count, size)
+
+// given a sufficiently large block of memory, make an array of pointers to subblocks of it
+static void *make_block_array(void *mem, int count, int size)
+{
+   int i;
+   void ** p = (void **) mem;
+   char *q = (char *) (p + count);
+   for (i=0; i < count; ++i) {
+      p[i] = q;
+      q += size;
+   }
+   return p;
+}
+
+static void *setup_malloc(vorb *f, int sz)
+{
+   sz = (sz+7) & ~7; // round up to nearest 8 for alignment of future allocs.
+   f->setup_memory_required += sz;
+   if (f->alloc.alloc_buffer) {
+      void *p = (char *) f->alloc.alloc_buffer + f->setup_offset;
+      if (f->setup_offset + sz > f->temp_offset) return NULL;
+      f->setup_offset += sz;
+      return p;
+   }
+   return sz ? malloc(sz) : NULL;
+}
+
+static void setup_free(vorb *f, void *p)
+{
+   if (f->alloc.alloc_buffer) return; // do nothing; setup mem is a stack
+   free(p);
+}
+
+static void *setup_temp_malloc(vorb *f, int sz)
+{
+   sz = (sz+7) & ~7; // round up to nearest 8 for alignment of future allocs.
+   if (f->alloc.alloc_buffer) {
+      if (f->temp_offset - sz < f->setup_offset) return NULL;
+      f->temp_offset -= sz;
+      return (char *) f->alloc.alloc_buffer + f->temp_offset;
+   }
+   return malloc(sz);
+}
+
+static void setup_temp_free(vorb *f, void *p, int sz)
+{
+   if (f->alloc.alloc_buffer) {
+      f->temp_offset += (sz+7)&~7;
+      return;
+   }
+   free(p);
+}
+
+#define CRC32_POLY    0x04c11db7   // from spec
+
+static uint32 crc_table[256];
+static void crc32_init(void)
+{
+   int i,j;
+   uint32 s;
+   for(i=0; i < 256; i++) {
+      for (s=(uint32) i << 24, j=0; j < 8; ++j)
+         s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0);
+      crc_table[i] = s;
+   }
+}
+
+static __forceinline uint32 crc32_update(uint32 crc, uint8 byte)
+{
+   return (crc << 8) ^ crc_table[byte ^ (crc >> 24)];
+}
+
+
+// used in setup, and for huffman that doesn't go fast path
+static unsigned int bit_reverse(unsigned int n)
+{
+  n = ((n & 0xAAAAAAAA) >>  1) | ((n & 0x55555555) << 1);
+  n = ((n & 0xCCCCCCCC) >>  2) | ((n & 0x33333333) << 2);
+  n = ((n & 0xF0F0F0F0) >>  4) | ((n & 0x0F0F0F0F) << 4);
+  n = ((n & 0xFF00FF00) >>  8) | ((n & 0x00FF00FF) << 8);
+  return (n >> 16) | (n << 16);
+}
+
+static float square(float x)
+{
+   return x*x;
+}
+
+// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3
+// as required by the specification. fast(?) implementation from stb.h
+// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup
+static int ilog(int32 n)
+{
+   static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 };
+
+   if (n < 0) return 0; // signed n returns 0
+
+   // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29)
+   if (n < (1 << 14))
+        if (n < (1 <<  4))            return  0 + log2_4[n      ];
+        else if (n < (1 <<  9))       return  5 + log2_4[n >>  5];
+             else                     return 10 + log2_4[n >> 10];
+   else if (n < (1 << 24))
+             if (n < (1 << 19))       return 15 + log2_4[n >> 15];
+             else                     return 20 + log2_4[n >> 20];
+        else if (n < (1 << 29))       return 25 + log2_4[n >> 25];
+             else                     return 30 + log2_4[n >> 30];
+}
+
+#ifndef M_PI
+  #define M_PI  3.14159265358979323846264f  // from CRC
+#endif
+
+// code length assigned to a value with no huffman encoding
+#define NO_CODE   255
+
+/////////////////////// LEAF SETUP FUNCTIONS //////////////////////////
+//
+// these functions are only called at setup, and only a few times
+// per file
+
+static float float32_unpack(uint32 x)
+{
+   // from the specification
+   uint32 mantissa = x & 0x1fffff;
+   uint32 sign = x & 0x80000000;
+   uint32 exp = (x & 0x7fe00000) >> 21;
+   double res = sign ? -(double)mantissa : (double)mantissa;
+   return (float) ldexp((float)res, exp-788);
+}
+
+
+// zlib & jpeg huffman tables assume that the output symbols
+// can either be arbitrarily arranged, or have monotonically
+// increasing frequencies--they rely on the lengths being sorted;
+// this makes for a very simple generation algorithm.
+// vorbis allows a huffman table with non-sorted lengths. This
+// requires a more sophisticated construction, since symbols in
+// order do not map to huffman codes "in order".
+static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values)
+{
+   if (!c->sparse) {
+      c->codewords      [symbol] = huff_code;
+   } else {
+      c->codewords       [count] = huff_code;
+      c->codeword_lengths[count] = len;
+      values             [count] = symbol;
+   }
+}
+
+static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
+{
+   int i,k,m=0;
+   uint32 available[32];
+
+   memset(available, 0, sizeof(available));
+   // 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; }
+   // add to the list
+   add_entry(c, 0, k, m++, len[k], values);
+   // add all available leaves
+   for (i=1; i <= len[k]; ++i)
+      available[i] = 1U << (32-i);
+   // note that the above code treats the first case specially,
+   // but it's really the same as the following code, so they
+   // could probably be combined (except the initial code is 0,
+   // and I use 0 in available[] to mean 'empty')
+   for (i=k+1; i < n; ++i) {
+      uint32 res;
+      int z = len[i], y;
+      if (z == NO_CODE) continue;
+      // 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
+      // more than one free leaf at a given level, isn't totally
+      // trivial to prove, but it seems true and the assert never
+      // fires, so!
+      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));
+         }
+      }
+   }
+   return TRUE;
+}
+
+// accelerated huffman table allows fast O(1) match of all symbols
+// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH
+static void compute_accelerated_huffman(Codebook *c)
+{
+   int i, len;
+   for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i)
+      c->fast_huffman[i] = -1;
+
+   len = c->sparse ? c->sorted_entries : c->entries;
+   #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+   if (len > 32767) len = 32767; // largest possible value we can encode!
+   #endif
+   for (i=0; i < len; ++i) {
+      if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) {
+         uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i];
+         // set table entries for all bit combinations in the higher bits
+         while (z < FAST_HUFFMAN_TABLE_SIZE) {
+             c->fast_huffman[z] = i;
+             z += 1 << c->codeword_lengths[i];
+         }
+      }
+   }
+}
+
+#ifdef _MSC_VER
+#define STBV_CDECL __cdecl
+#else
+#define STBV_CDECL
+#endif
+
+static int STBV_CDECL uint32_compare(const void *p, const void *q)
+{
+   uint32 x = * (uint32 *) p;
+   uint32 y = * (uint32 *) q;
+   return x < y ? -1 : x > y;
+}
+
+static int include_in_sort(Codebook *c, uint8 len)
+{
+   if (c->sparse) { assert(len != NO_CODE); return TRUE; }
+   if (len == NO_CODE) return FALSE;
+   if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE;
+   return FALSE;
+}
+
+// if the fast table above doesn't work, we want to binary
+// search them... need to reverse the bits
+static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values)
+{
+   int i, len;
+   // build a list of all the entries
+   // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN.
+   // this is kind of a frivolous optimization--I don't see any performance improvement,
+   // but it's like 4 extra lines of code, so.
+   if (!c->sparse) {
+      int k = 0;
+      for (i=0; i < c->entries; ++i)
+         if (include_in_sort(c, lengths[i]))
+            c->sorted_codewords[k++] = bit_reverse(c->codewords[i]);
+      assert(k == c->sorted_entries);
+   } else {
+      for (i=0; i < c->sorted_entries; ++i)
+         c->sorted_codewords[i] = bit_reverse(c->codewords[i]);
+   }
+
+   qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare);
+   c->sorted_codewords[c->sorted_entries] = 0xffffffff;
+
+   len = c->sparse ? c->sorted_entries : c->entries;
+   // now we need to indicate how they correspond; we could either
+   //   #1: sort a different data structure that says who they correspond to
+   //   #2: for each sorted entry, search the original list to find who corresponds
+   //   #3: for each original entry, find the sorted entry
+   // #1 requires extra storage, #2 is slow, #3 can use binary search!
+   for (i=0; i < len; ++i) {
+      int huff_len = c->sparse ? lengths[values[i]] : lengths[i];
+      if (include_in_sort(c,huff_len)) {
+         uint32 code = bit_reverse(c->codewords[i]);
+         int x=0, n=c->sorted_entries;
+         while (n > 1) {
+            // invariant: sc[x] <= code < sc[x+n]
+            int m = x + (n >> 1);
+            if (c->sorted_codewords[m] <= code) {
+               x = m;
+               n -= (n>>1);
+            } else {
+               n >>= 1;
+            }
+         }
+         assert(c->sorted_codewords[x] == code);
+         if (c->sparse) {
+            c->sorted_values[x] = values[i];
+            c->codeword_lengths[x] = huff_len;
+         } else {
+            c->sorted_values[x] = i;
+         }
+      }
+   }
+}
+
+// only run while parsing the header (3 times)
+static int vorbis_validate(uint8 *data)
+{
+   static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' };
+   return memcmp(data, vorbis, 6) == 0;
+}
+
+// called from setup only, once per code book
+// (formula implied by specification)
+static int lookup1_values(int entries, int dim)
+{
+   int r = (int) floor(exp((float) log((float) entries) / dim));
+   if ((int) floor(pow((float) r+1, dim)) <= entries)   // (int) cast for MinGW warning;
+      ++r;                                              // floor() to avoid _ftol() when non-CRT
+   if (pow((float) r+1, dim) <= entries)
+      return -1;
+   if ((int) floor(pow((float) r, dim)) > entries)
+      return -1;
+   return r;
+}
+
+// called twice per file
+static void compute_twiddle_factors(int n, float *A, float *B, float *C)
+{
+   int n4 = n >> 2, n8 = n >> 3;
+   int k,k2;
+
+   for (k=k2=0; k < n4; ++k,k2+=2) {
+      A[k2  ] = (float)  cos(4*k*M_PI/n);
+      A[k2+1] = (float) -sin(4*k*M_PI/n);
+      B[k2  ] = (float)  cos((k2+1)*M_PI/n/2) * 0.5f;
+      B[k2+1] = (float)  sin((k2+1)*M_PI/n/2) * 0.5f;
+   }
+   for (k=k2=0; k < n8; ++k,k2+=2) {
+      C[k2  ] = (float)  cos(2*(k2+1)*M_PI/n);
+      C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
+   }
+}
+
+static void compute_window(int n, float *window)
+{
+   int n2 = n >> 1, i;
+   for (i=0; i < n2; ++i)
+      window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI)));
+}
+
+static void compute_bitreverse(int n, uint16 *rev)
+{
+   int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+   int i, n8 = n >> 3;
+   for (i=0; i < n8; ++i)
+      rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2;
+}
+
+static int init_blocksize(vorb *f, int b, int n)
+{
+   int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3;
+   f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4);
+   if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem);
+   compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]);
+   f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   if (!f->window[b]) return error(f, VORBIS_outofmem);
+   compute_window(n, f->window[b]);
+   f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8);
+   if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem);
+   compute_bitreverse(n, f->bit_reverse[b]);
+   return TRUE;
+}
+
+static void neighbors(uint16 *x, int n, int *plow, int *phigh)
+{
+   int low = -1;
+   int high = 65536;
+   int i;
+   for (i=0; i < n; ++i) {
+      if (x[i] > low  && x[i] < x[n]) { *plow  = i; low = x[i]; }
+      if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; }
+   }
+}
+
+// this has been repurposed so y is now the original index instead of y
+typedef struct
+{
+   uint16 x,id;
+} stbv__floor_ordering;
+
+static int STBV_CDECL point_compare(const void *p, const void *q)
+{
+   stbv__floor_ordering *a = (stbv__floor_ordering *) p;
+   stbv__floor_ordering *b = (stbv__floor_ordering *) q;
+   return a->x < b->x ? -1 : a->x > b->x;
+}
+
+//
+/////////////////////// END LEAF SETUP FUNCTIONS //////////////////////////
+
+
+#if defined(STB_VORBIS_NO_STDIO)
+   #define USE_MEMORY(z)    TRUE
+#else
+   #define USE_MEMORY(z)    ((z)->stream)
+#endif
+
+static uint8 get8(vorb *z)
+{
+   if (USE_MEMORY(z)) {
+      if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; }
+      return *z->stream++;
+   }
+
+   #ifndef STB_VORBIS_NO_STDIO
+   {
+   int c = fgetc(z->f);
+   if (c == EOF) { z->eof = TRUE; return 0; }
+   return c;
+   }
+   #endif
+}
+
+static uint32 get32(vorb *f)
+{
+   uint32 x;
+   x = get8(f);
+   x += get8(f) << 8;
+   x += get8(f) << 16;
+   x += (uint32) get8(f) << 24;
+   return x;
+}
+
+static int getn(vorb *z, uint8 *data, int n)
+{
+   if (USE_MEMORY(z)) {
+      if (z->stream+n > z->stream_end) { z->eof = 1; return 0; }
+      memcpy(data, z->stream, n);
+      z->stream += n;
+      return 1;
+   }
+
+   #ifndef STB_VORBIS_NO_STDIO
+   if (fread(data, n, 1, z->f) == 1)
+      return 1;
+   else {
+      z->eof = 1;
+      return 0;
+   }
+   #endif
+}
+
+static void skip(vorb *z, int n)
+{
+   if (USE_MEMORY(z)) {
+      z->stream += n;
+      if (z->stream >= z->stream_end) z->eof = 1;
+      return;
+   }
+   #ifndef STB_VORBIS_NO_STDIO
+   {
+      long x = ftell(z->f);
+      fseek(z->f, x+n, SEEK_SET);
+   }
+   #endif
+}
+
+static int set_file_offset(stb_vorbis *f, unsigned int loc)
+{
+   #ifndef STB_VORBIS_NO_PUSHDATA_API
+   if (f->push_mode) return 0;
+   #endif
+   f->eof = 0;
+   if (USE_MEMORY(f)) {
+      if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) {
+         f->stream = f->stream_end;
+         f->eof = 1;
+         return 0;
+      } else {
+         f->stream = f->stream_start + loc;
+         return 1;
+      }
+   }
+   #ifndef STB_VORBIS_NO_STDIO
+   if (loc + f->f_start < loc || loc >= 0x80000000) {
+      loc = 0x7fffffff;
+      f->eof = 1;
+   } else {
+      loc += f->f_start;
+   }
+   if (!fseek(f->f, loc, SEEK_SET))
+      return 1;
+   f->eof = 1;
+   fseek(f->f, f->f_start, SEEK_END);
+   return 0;
+   #endif
+}
+
+
+static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 };
+
+static int capture_pattern(vorb *f)
+{
+   if (0x4f != get8(f)) return FALSE;
+   if (0x67 != get8(f)) return FALSE;
+   if (0x67 != get8(f)) return FALSE;
+   if (0x53 != get8(f)) return FALSE;
+   return TRUE;
+}
+
+#define PAGEFLAG_continued_packet   1
+#define PAGEFLAG_first_page         2
+#define PAGEFLAG_last_page          4
+
+static int start_page_no_capturepattern(vorb *f)
+{
+   uint32 loc0,loc1,n;
+   if (f->first_decode && !IS_PUSH_MODE(f)) {
+      f->p_first.page_start = stb_vorbis_get_file_offset(f) - 4;
+   }
+   // stream structure version
+   if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version);
+   // header flag
+   f->page_flag = get8(f);
+   // absolute granule position
+   loc0 = get32(f);
+   loc1 = get32(f);
+   // @TODO: validate loc0,loc1 as valid positions?
+   // stream serial number -- vorbis doesn't interleave, so discard
+   get32(f);
+   //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number);
+   // page sequence number
+   n = get32(f);
+   f->last_page = n;
+   // CRC32
+   get32(f);
+   // page_segments
+   f->segment_count = get8(f);
+   if (!getn(f, f->segments, f->segment_count))
+      return error(f, VORBIS_unexpected_eof);
+   // assume we _don't_ know any the sample position of any segments
+   f->end_seg_with_known_loc = -2;
+   if (loc0 != ~0U || loc1 != ~0U) {
+      int i;
+      // determine which packet is the last one that will complete
+      for (i=f->segment_count-1; i >= 0; --i)
+         if (f->segments[i] < 255)
+            break;
+      // 'i' is now the index of the _last_ segment of a packet that ends
+      if (i >= 0) {
+         f->end_seg_with_known_loc = i;
+         f->known_loc_for_packet   = loc0;
+      }
+   }
+   if (f->first_decode) {
+      int i,len;
+      len = 0;
+      for (i=0; i < f->segment_count; ++i)
+         len += f->segments[i];
+      len += 27 + f->segment_count;
+      f->p_first.page_end = f->p_first.page_start + len;
+      f->p_first.last_decoded_sample = loc0;
+   }
+   f->next_seg = 0;
+   return TRUE;
+}
+
+static int start_page(vorb *f)
+{
+   if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern);
+   return start_page_no_capturepattern(f);
+}
+
+static int start_packet(vorb *f)
+{
+   while (f->next_seg == -1) {
+      if (!start_page(f)) return FALSE;
+      if (f->page_flag & PAGEFLAG_continued_packet)
+         return error(f, VORBIS_continued_packet_flag_invalid);
+   }
+   f->last_seg = FALSE;
+   f->valid_bits = 0;
+   f->packet_bytes = 0;
+   f->bytes_in_seg = 0;
+   // f->next_seg is now valid
+   return TRUE;
+}
+
+static int maybe_start_packet(vorb *f)
+{
+   if (f->next_seg == -1) {
+      int x = get8(f);
+      if (f->eof) return FALSE; // EOF at page boundary is not an error!
+      if (0x4f != x      ) return error(f, VORBIS_missing_capture_pattern);
+      if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (!start_page_no_capturepattern(f)) return FALSE;
+      if (f->page_flag & PAGEFLAG_continued_packet) {
+         // set up enough state that we can read this packet if we want,
+         // e.g. during recovery
+         f->last_seg = FALSE;
+         f->bytes_in_seg = 0;
+         return error(f, VORBIS_continued_packet_flag_invalid);
+      }
+   }
+   return start_packet(f);
+}
+
+static int next_segment(vorb *f)
+{
+   int len;
+   if (f->last_seg) return 0;
+   if (f->next_seg == -1) {
+      f->last_seg_which = f->segment_count-1; // in case start_page fails
+      if (!start_page(f)) { f->last_seg = 1; return 0; }
+      if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid);
+   }
+   len = f->segments[f->next_seg++];
+   if (len < 255) {
+      f->last_seg = TRUE;
+      f->last_seg_which = f->next_seg-1;
+   }
+   if (f->next_seg >= f->segment_count)
+      f->next_seg = -1;
+   assert(f->bytes_in_seg == 0);
+   f->bytes_in_seg = len;
+   return len;
+}
+
+#define EOP    (-1)
+#define INVALID_BITS  (-1)
+
+static int get8_packet_raw(vorb *f)
+{
+   if (!f-