gosu 0.14.6.pre1 → 1.0.0.pre2

data/dependencies/SDL_sound/SDL_sound_vorbis.c

@@ -0,0 +1,223 @@
+/**
+ * SDL_sound; A sound processing toolkit.
+ *
+ * Please see the file LICENSE.txt in the source's root directory.
+ *
+ *  This file written by Ryan C. Gordon.
+ */
+
+/*
+ * Ogg Vorbis decoder for SDL_sound.
+ *
+ * This driver handles .OGG audio files, and uses Sean Barrett's excellent
+ *  stb_vorbis for the heavy lifting. Since stb_vorbis is a single,
+ *  public domain C header file, it's just compiled into this source and
+ *  needs no external dependencies.
+ *
+ * stb_vorbis homepage: https://nothings.org/stb_vorbis/
+ */
+
+#define __SDL_SOUND_INTERNAL__
+#include "SDL_sound_internal.h"
+
+#if SOUND_SUPPORTS_VORBIS
+
+/* Configure and include stb_vorbis for compiling... */
+#define STB_VORBIS_NO_STDIO 1
+#define STB_VORBIS_NO_CRT 1
+#define STB_VORBIS_NO_PUSHDATA_API 1
+#define STB_VORBIS_MAX_CHANNELS 6
+#define STBV_CDECL
+#define STB_FORCEINLINE SDL_FORCE_INLINE
+#if SDL_BYTEORDER == SDL_BIG_ENDIAN
+#define STB_VORBIS_BIG_ENDIAN 1
+#endif
+
+#if !defined(__clang_analyzer__)
+#ifdef assert
+#undef assert
+#endif
+#ifdef memset
+#undef memset
+#endif
+#ifdef memcpy
+#undef memcpy
+#endif
+#ifdef alloca
+#undef alloca
+#endif
+#define assert SDL_assert
+#define memset SDL_memset
+#define memcmp SDL_memcmp
+#define memcpy SDL_memcpy
+#define qsort SDL_qsort
+#define pow SDL_pow
+#define floor SDL_floor
+#define malloc SDL_malloc
+#define realloc SDL_realloc
+#define free SDL_free
+#define alloca(x) ((void *) SDL_stack_alloc(Uint8, (x)))
+#define dealloca(x) SDL_stack_free((x))
+#define ldexp(v, e) SDL_scalbn((v), (e))
+#define abs(x) SDL_abs(x)
+#define cos(x) SDL_cos(x)
+#define sin(x) SDL_sin(x)
+#define log(x) SDL_log(x)
+#endif
+#include "stb_vorbis.h"
+
+static const char *vorbis_error_string(const int err)
+{
+    switch (err)
+    {
+        case VORBIS__no_error: return NULL;
+        case VORBIS_need_more_data: return "VORBIS: need more data";
+        case VORBIS_invalid_api_mixing: return "VORBIS: can't mix API modes";
+        case VORBIS_outofmem: return "VORBIS: out of memory";
+        case VORBIS_feature_not_supported: return "VORBIS: feature not supported";
+        case VORBIS_too_many_channels: return "VORBIS: too many channels";
+        case VORBIS_seek_without_length: return "VORBIS: can't seek in unknown length stream";
+        case VORBIS_unexpected_eof: return "VORBIS: unexpected eof";
+        case VORBIS_seek_invalid: return "VORBIS: invalid seek";
+        case VORBIS_invalid_setup: return "VORBIS: invalid setup";
+        case VORBIS_invalid_stream: return "VORBIS: invalid stream";
+        case VORBIS_missing_capture_pattern: return "VORBIS: missing capture pattern";
+        case VORBIS_invalid_stream_structure_version: return "VORBIS: invalid stream structure version";
+        case VORBIS_continued_packet_flag_invalid: return "VORBIS: continued packet flag invalid";
+        case VORBIS_incorrect_stream_serial_number: return "VORBIS: incorrect stream serial number";
+        case VORBIS_invalid_first_page: return "VORBIS: invalid first page";
+        case VORBIS_bad_packet_type: return "VORBIS: bad packet type";
+        case VORBIS_cant_find_last_page: return "VORBIS: can't find last page";
+        case VORBIS_seek_failed: return "VORBIS: seek failed";
+        default: break;
+    } /* switch */
+
+    return "VORBIS: unknown error";
+} /* vorbis_error_string */
+
+static int VORBIS_init(void)
+{
+    return 1;  /* always succeeds. */
+} /* VORBIS_init */
+
+static void VORBIS_quit(void)
+{
+    /* it's a no-op. */
+} /* VORBIS_quit */
+
+static int VORBIS_open(Sound_Sample *sample, const char *ext)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    SDL_RWops *rw = internal->rw;
+    int err = 0;
+    stb_vorbis *stb = stb_vorbis_open_rwops(rw, 0, &err, NULL);
+    unsigned int num_frames;
+
+    BAIL_IF_MACRO(!stb, vorbis_error_string(err), 0);
+
+    SNDDBG(("VORBIS: Accepting data stream.\n"));
+
+    internal->decoder_private = stb;
+    sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
+    sample->actual.format = AUDIO_F32SYS;
+    sample->actual.channels = stb->channels;
+    sample->actual.rate = stb->sample_rate;
+    num_frames = stb_vorbis_stream_length_in_samples(stb);
+    if (!num_frames)
+        internal->total_time = -1;
+    else
+    {
+        const unsigned int rate = stb->sample_rate;
+        internal->total_time = (num_frames / rate) * 1000;
+        internal->total_time += (num_frames % rate) * 1000 / rate;
+    } /* else */
+
+    return 1; /* we'll handle this data. */
+} /* VORBIS_open */
+
+
+static void VORBIS_close(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    stb_vorbis *stb = (stb_vorbis *) internal->decoder_private;
+    stb_vorbis_close(stb);
+} /* VORBIS_close */
+
+
+static Uint32 VORBIS_read(Sound_Sample *sample)
+{
+    Uint32 retval;
+    int rc;
+    int err;
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    stb_vorbis *stb = (stb_vorbis *) internal->decoder_private;
+    const int channels = (int) sample->actual.channels;
+    const int want_samples = (int) (internal->buffer_size / sizeof (float));
+
+    stb_vorbis_get_error(stb);  /* clear any error state */
+    rc = stb_vorbis_get_samples_float_interleaved(stb, channels, (float *) internal->buffer, want_samples);
+    retval = (Uint32) (rc * channels * sizeof (float));  /* rc == number of sample frames read */
+    err = stb_vorbis_get_error(stb);
+
+    if (retval == 0)
+    {
+        if (!err)
+            sample->flags |= SOUND_SAMPLEFLAG_EOF;
+        else
+        {
+            sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+            /* !!! FIXME: should I set this? __Sound_SetError(vorbis_error_string(err)); */
+        } /* else */
+    } /* if */
+
+    else if (retval < internal->buffer_size)
+        sample->flags |= SOUND_SAMPLEFLAG_EAGAIN;
+
+    return retval;
+} /* VORBIS_read */
+
+
+static int VORBIS_rewind(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    stb_vorbis *stb = (stb_vorbis *) internal->decoder_private;
+    BAIL_IF_MACRO(!stb_vorbis_seek_start(stb), vorbis_error_string(stb_vorbis_get_error(stb)), 0);
+    return 1;
+} /* VORBIS_rewind */
+
+
+static int VORBIS_seek(Sound_Sample *sample, Uint32 ms)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    stb_vorbis *stb = (stb_vorbis *) internal->decoder_private;
+    const float frames_per_ms = ((float) sample->actual.rate) / 1000.0f;
+    const Uint32 frame_offset = (Uint32) (frames_per_ms * ((float) ms));
+    const unsigned int sampnum = (unsigned int) frame_offset;
+    BAIL_IF_MACRO(!stb_vorbis_seek(stb, sampnum), vorbis_error_string(stb_vorbis_get_error(stb)), 0);
+    return 1;
+} /* VORBIS_seek */
+
+
+static const char *extensions_vorbis[] = { "OGG", NULL };
+const Sound_DecoderFunctions __Sound_DecoderFunctions_VORBIS =
+{
+    {
+        extensions_vorbis,
+        "Ogg Vorbis audio",
+        "Ryan C. Gordon <icculus@icculus.org>",
+        "https://icculus.org/SDL_sound/"
+    },
+
+    VORBIS_init,       /*   init() method */
+    VORBIS_quit,       /*   quit() method */
+    VORBIS_open,       /*   open() method */
+    VORBIS_close,      /*  close() method */
+    VORBIS_read,       /*   read() method */
+    VORBIS_rewind,     /* rewind() method */
+    VORBIS_seek        /*   seek() method */
+};
+
+#endif /* SOUND_SUPPORTS_VORBIS */
+
+/* end of SDL_sound_vorbis.c ... */
+

data/dependencies/SDL_sound/SDL_sound_wav.c

@@ -0,0 +1,783 @@
+/**
+ * SDL_sound; A sound processing toolkit.
+ *
+ * Please see the file LICENSE.txt in the source's root directory.
+ *
+ *  This file written by Ryan C. Gordon.
+ */
+
+/*
+ * WAV decoder for SDL_sound.
+ *
+ * This driver handles Microsoft .WAVs, in as many of the thousands of
+ *  variations as we can.
+ */
+
+#define __SDL_SOUND_INTERNAL__
+#include "SDL_sound_internal.h"
+
+#if SOUND_SUPPORTS_WAV
+
+/* Better than SDL_ReadLE16, since you can detect i/o errors... */
+static SDL_INLINE int read_le16(SDL_RWops *rw, Uint16 *ui16)
+{
+    int rc = SDL_RWread(rw, ui16, sizeof (Uint16), 1);
+    BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0);
+    *ui16 = SDL_SwapLE16(*ui16);
+    return 1;
+} /* read_le16 */
+
+
+/* Better than SDL_ReadLE32, since you can detect i/o errors... */
+static SDL_INLINE int read_le32(SDL_RWops *rw, Uint32 *ui32)
+{
+    int rc = SDL_RWread(rw, ui32, sizeof (Uint32), 1);
+    BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0);
+    *ui32 = SDL_SwapLE32(*ui32);
+    return 1;
+} /* read_le32 */
+
+static SDL_INLINE int read_le16s(SDL_RWops *rw, Sint16 *si16)
+{
+    return read_le16(rw, (Uint16 *) si16);
+} /* read_le16s */
+
+static SDL_INLINE int read_le32s(SDL_RWops *rw, Sint32 *si32)
+{
+    return read_le32(rw, (Uint32 *) si32);
+} /* read_le32s */
+
+
+/* This is just cleaner on the caller's end... */
+static SDL_INLINE int read_uint8(SDL_RWops *rw, Uint8 *ui8)
+{
+    int rc = SDL_RWread(rw, ui8, sizeof (Uint8), 1);
+    BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0);
+    return 1;
+} /* read_uint8 */
+
+
+    /* Chunk management code... */
+
+#define riffID 0x46464952  /* "RIFF", in ascii. */
+#define waveID 0x45564157  /* "WAVE", in ascii. */
+#define factID 0x74636166  /* "fact", in ascii. */
+
+
+/*****************************************************************************
+ * The FORMAT chunk...                                                       *
+ *****************************************************************************/
+
+#define fmtID  0x20746D66  /* "fmt ", in ascii. */
+
+#define FMT_NORMAL 0x0001    /* Uncompressed waveform data.     */
+#define FMT_ADPCM  0x0002    /* ADPCM compressed waveform data. */
+
+typedef struct
+{
+    Sint16 iCoef1;
+    Sint16 iCoef2;
+} ADPCMCOEFSET;
+
+typedef struct
+{
+    Uint8 bPredictor;
+    Uint16 iDelta;
+    Sint16 iSamp1;
+    Sint16 iSamp2;
+} ADPCMBLOCKHEADER;
+
+typedef struct S_WAV_FMT_T
+{
+    Uint32 chunkID;
+    Sint32 chunkSize;
+    Sint16 wFormatTag;
+    Uint16 wChannels;
+    Uint32 dwSamplesPerSec;
+    Uint32 dwAvgBytesPerSec;
+    Uint16 wBlockAlign;
+    Uint16 wBitsPerSample;
+
+    Uint32 next_chunk_offset;
+    
+    Uint32 sample_frame_size;
+    Uint32 data_starting_offset;
+    Uint32 total_bytes;
+
+    void (*free)(struct S_WAV_FMT_T *fmt);
+    Uint32 (*read_sample)(Sound_Sample *sample);
+    int (*rewind_sample)(Sound_Sample *sample);
+    int (*seek_sample)(Sound_Sample *sample, Uint32 ms);
+
+    union
+    {
+        struct
+        {
+            Uint16 cbSize;
+            Uint16 wSamplesPerBlock;
+            Uint16 wNumCoef;
+            ADPCMCOEFSET *aCoef;
+            ADPCMBLOCKHEADER *blockheaders;
+            Uint32 samples_left_in_block;
+            int nibble_state;
+            Sint8 nibble;
+        } adpcm;
+
+        /* put other format-specific data here... */
+    } fmt;
+} fmt_t;
+
+
+/*
+ * Read in a fmt_t from disk. This makes this process safe regardless of
+ *  the processor's byte order or how the fmt_t structure is packed.
+ * Note that the union "fmt" is not read in here; that is handled as 
+ *  needed in the read_fmt_* functions.
+ */
+static int read_fmt_chunk(SDL_RWops *rw, fmt_t *fmt)
+{
+    /* skip reading the chunk ID, since it was already read at this point... */
+    fmt->chunkID = fmtID;
+
+    BAIL_IF_MACRO(!read_le32s(rw, &fmt->chunkSize), NULL, 0);
+    BAIL_IF_MACRO(fmt->chunkSize < 16, "WAV: Invalid chunk size", 0);
+    fmt->next_chunk_offset = SDL_RWtell(rw) + fmt->chunkSize;
+    
+    BAIL_IF_MACRO(!read_le16s(rw, &fmt->wFormatTag), NULL, 0);
+    BAIL_IF_MACRO(!read_le16(rw, &fmt->wChannels), NULL, 0);
+    BAIL_IF_MACRO(!read_le32(rw, &fmt->dwSamplesPerSec), NULL, 0);
+    BAIL_IF_MACRO(!read_le32(rw, &fmt->dwAvgBytesPerSec), NULL, 0);
+    BAIL_IF_MACRO(!read_le16(rw, &fmt->wBlockAlign), NULL, 0);
+    BAIL_IF_MACRO(!read_le16(rw, &fmt->wBitsPerSample), NULL, 0);
+
+    return 1;
+} /* read_fmt_chunk */
+
+
+
+/*****************************************************************************
+ * The DATA chunk...                                                         *
+ *****************************************************************************/
+
+#define dataID 0x61746164  /* "data", in ascii. */
+
+typedef struct
+{
+    Uint32 chunkID;
+    Sint32 chunkSize;
+    /* Then, (chunkSize) bytes of waveform data... */
+} data_t;
+
+
+/*
+ * Read in a data_t from disk. This makes this process safe regardless of
+ *  the processor's byte order or how the fmt_t structure is packed.
+ */
+static int read_data_chunk(SDL_RWops *rw, data_t *data)
+{
+    /* skip reading the chunk ID, since it was already read at this point... */
+    data->chunkID = dataID;
+    BAIL_IF_MACRO(!read_le32s(rw, &data->chunkSize), NULL, 0);
+    return 1;
+} /* read_data_chunk */
+
+
+
+
+/*****************************************************************************
+ * this is what we store in our internal->decoder_private field...           *
+ *****************************************************************************/
+
+typedef struct
+{
+    fmt_t *fmt;
+    Sint32 bytesLeft;
+} wav_t;
+
+
+
+
+/*****************************************************************************
+ * Normal, uncompressed waveform handler...                                  *
+ *****************************************************************************/
+
+/*
+ * Sound_Decode() lands here for uncompressed WAVs...
+ */
+static Uint32 read_sample_fmt_normal(Sound_Sample *sample)
+{
+    Uint32 retval;
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    Uint32 max = (internal->buffer_size < (Uint32) w->bytesLeft) ?
+                    internal->buffer_size : (Uint32) w->bytesLeft;
+
+    SDL_assert(max > 0);
+
+        /*
+         * We don't actually do any decoding, so we read the wav data
+         *  directly into the internal buffer...
+         */
+    retval = SDL_RWread(internal->rw, internal->buffer, 1, max);
+
+    w->bytesLeft -= retval;
+
+        /* Make sure the read went smoothly... */
+    if ((retval == 0) || (w->bytesLeft == 0))
+        sample->flags |= SOUND_SAMPLEFLAG_EOF;
+
+    else if (retval == -1)
+        sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+
+        /* (next call this EAGAIN may turn into an EOF or error.) */
+    else if (retval < internal->buffer_size)
+        sample->flags |= SOUND_SAMPLEFLAG_EAGAIN;
+
+    return retval;
+} /* read_sample_fmt_normal */
+
+
+static int seek_sample_fmt_normal(Sound_Sample *sample, Uint32 ms)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    fmt_t *fmt = w->fmt;
+    int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
+    int pos = (int) (fmt->data_starting_offset + offset);
+    int rc = SDL_RWseek(internal->rw, pos, SEEK_SET);
+    BAIL_IF_MACRO(rc != pos, ERR_IO_ERROR, 0);
+    w->bytesLeft = fmt->total_bytes - offset;
+    return 1;  /* success. */
+} /* seek_sample_fmt_normal */
+
+
+static int rewind_sample_fmt_normal(Sound_Sample *sample)
+{
+    /* no-op. */
+    return 1;
+} /* rewind_sample_fmt_normal */
+
+
+static int read_fmt_normal(SDL_RWops *rw, fmt_t *fmt)
+{
+    /* (don't need to read more from the RWops...) */
+    fmt->free = NULL;
+    fmt->read_sample = read_sample_fmt_normal;
+    fmt->rewind_sample = rewind_sample_fmt_normal;
+    fmt->seek_sample = seek_sample_fmt_normal;
+    return 1;
+} /* read_fmt_normal */
+
+
+
+/*****************************************************************************
+ * ADPCM compression handler...                                              *
+ *****************************************************************************/
+
+#define FIXED_POINT_COEF_BASE      256
+#define FIXED_POINT_ADAPTION_BASE  256
+#define SMALLEST_ADPCM_DELTA       16
+
+
+static SDL_INLINE int read_adpcm_block_headers(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    SDL_RWops *rw = internal->rw;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    fmt_t *fmt = w->fmt;
+    ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
+    int i;
+    int max = fmt->wChannels;
+
+    if (w->bytesLeft < fmt->wBlockAlign)
+    {
+        sample->flags |= SOUND_SAMPLEFLAG_EOF;
+        return 0;
+    } /* if */
+
+    w->bytesLeft -= fmt->wBlockAlign;
+
+    for (i = 0; i < max; i++)
+        BAIL_IF_MACRO(!read_uint8(rw, &headers[i].bPredictor), NULL, 0);
+
+    for (i = 0; i < max; i++)
+        BAIL_IF_MACRO(!read_le16(rw, &headers[i].iDelta), NULL, 0);
+
+    for (i = 0; i < max; i++)
+        BAIL_IF_MACRO(!read_le16s(rw, &headers[i].iSamp1), NULL, 0);
+
+    for (i = 0; i < max; i++)
+        BAIL_IF_MACRO(!read_le16s(rw, &headers[i].iSamp2), NULL, 0);
+
+    fmt->fmt.adpcm.samples_left_in_block = fmt->fmt.adpcm.wSamplesPerBlock;
+    fmt->fmt.adpcm.nibble_state = 0;
+    return 1;
+} /* read_adpcm_block_headers */
+
+
+static SDL_INLINE void do_adpcm_nibble(Uint8 nib,
+                                       ADPCMBLOCKHEADER *header,
+                                       Sint32 lPredSamp)
+{
+	static const Sint32 max_audioval = ((1<<(16-1))-1);
+	static const Sint32 min_audioval = -(1<<(16-1));
+	static const Sint32 AdaptionTable[] =
+    {
+		230, 230, 230, 230, 307, 409, 512, 614,
+		768, 614, 512, 409, 307, 230, 230, 230
+	};
+
+    Sint32 lNewSamp;
+    Sint32 delta;
+
+    if (nib & 0x08)
+        lNewSamp = lPredSamp + (header->iDelta * (nib - 0x10));
+	else
+        lNewSamp = lPredSamp + (header->iDelta * nib);
+
+        /* clamp value... */
+    if (lNewSamp < min_audioval)
+        lNewSamp = min_audioval;
+    else if (lNewSamp > max_audioval)
+        lNewSamp = max_audioval;
+
+    delta = ((Sint32) header->iDelta * AdaptionTable[nib]) /
+              FIXED_POINT_ADAPTION_BASE;
+
+	if (delta < SMALLEST_ADPCM_DELTA)
+	    delta = SMALLEST_ADPCM_DELTA;
+
+    header->iDelta = delta;
+	header->iSamp2 = header->iSamp1;
+	header->iSamp1 = lNewSamp;
+} /* do_adpcm_nibble */
+
+
+static SDL_INLINE int decode_adpcm_sample_frame(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    fmt_t *fmt = w->fmt;
+    ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
+    SDL_RWops *rw = internal->rw;
+    int i;
+    int max = fmt->wChannels;
+    Sint32 delta;
+    Uint8 nib = fmt->fmt.adpcm.nibble;
+
+    for (i = 0; i < max; i++)
+    {
+        Uint8 byte;
+        Sint16 iCoef1 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef1;
+        Sint16 iCoef2 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef2;
+        Sint32 lPredSamp = ((headers[i].iSamp1 * iCoef1) +
+                            (headers[i].iSamp2 * iCoef2)) / 
+                             FIXED_POINT_COEF_BASE;
+
+        if (fmt->fmt.adpcm.nibble_state == 0)
+        {
+            BAIL_IF_MACRO(!read_uint8(rw, &nib), NULL, 0);
+            fmt->fmt.adpcm.nibble_state = 1;
+            do_adpcm_nibble(nib >> 4, &headers[i], lPredSamp);
+        } /* if */
+        else
+        {
+            fmt->fmt.adpcm.nibble_state = 0;
+            do_adpcm_nibble(nib & 0x0F, &headers[i], lPredSamp);
+        } /* else */
+    } /* for */
+
+    fmt->fmt.adpcm.nibble = nib;
+    return 1;
+} /* decode_adpcm_sample_frame */
+
+
+static SDL_INLINE void put_adpcm_sample_frame1(void *_buf, fmt_t *fmt)
+{
+    Uint16 *buf = (Uint16 *) _buf;
+    ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
+    int i;
+    for (i = 0; i < fmt->wChannels; i++)
+        *(buf++) = headers[i].iSamp1;
+} /* put_adpcm_sample_frame1 */
+
+
+static SDL_INLINE void put_adpcm_sample_frame2(void *_buf, fmt_t *fmt)
+{
+    Uint16 *buf = (Uint16 *) _buf;
+    ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
+    int i;
+    for (i = 0; i < fmt->wChannels; i++)
+        *(buf++) = headers[i].iSamp2;
+} /* put_adpcm_sample_frame2 */
+
+
+/*
+ * Sound_Decode() lands here for ADPCM-encoded WAVs...
+ */
+static Uint32 read_sample_fmt_adpcm(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    fmt_t *fmt = w->fmt;
+    Uint32 bw = 0;
+
+    while (bw < internal->buffer_size)
+    {
+        /* write ongoing sample frame before reading more data... */
+        switch (fmt->fmt.adpcm.samples_left_in_block)
+        {
+            case 0:  /* need to read a new block... */
+                if (!read_adpcm_block_headers(sample))
+                {
+                    if ((sample->flags & SOUND_SAMPLEFLAG_EOF) == 0)
+                        sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+                    return bw;
+                } /* if */
+
+                /* only write first sample frame for now. */
+                put_adpcm_sample_frame2((Uint8 *) internal->buffer + bw, fmt);
+                fmt->fmt.adpcm.samples_left_in_block--;
+                bw += fmt->sample_frame_size;
+                break;
+
+            case 1:  /* output last sample frame of block... */
+                put_adpcm_sample_frame1((Uint8 *) internal->buffer + bw, fmt);
+                fmt->fmt.adpcm.samples_left_in_block--;
+                bw += fmt->sample_frame_size;
+                break;
+
+            default: /* output latest sample frame and read a new one... */
+                put_adpcm_sample_frame1((Uint8 *) internal->buffer + bw, fmt);
+                fmt->fmt.adpcm.samples_left_in_block--;
+                bw += fmt->sample_frame_size;
+
+                if (!decode_adpcm_sample_frame(sample))
+                {
+                    sample->flags |= SOUND_SAMPLEFLAG_ERROR;
+                    return bw;
+                } /* if */
+        } /* switch */
+    } /* while */
+
+    return bw;
+} /* read_sample_fmt_adpcm */
+
+
+/*
+ * Sound_FreeSample() lands here for ADPCM-encoded WAVs...
+ */
+static void free_fmt_adpcm(fmt_t *fmt)
+{
+    if (fmt->fmt.adpcm.aCoef != NULL)
+        SDL_free(fmt->fmt.adpcm.aCoef);
+
+    if (fmt->fmt.adpcm.blockheaders != NULL)
+        SDL_free(fmt->fmt.adpcm.blockheaders);
+} /* free_fmt_adpcm */
+
+
+static int rewind_sample_fmt_adpcm(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    w->fmt->fmt.adpcm.samples_left_in_block = 0;
+    return 1;
+} /* rewind_sample_fmt_adpcm */
+
+
+static int seek_sample_fmt_adpcm(Sound_Sample *sample, Uint32 ms)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    fmt_t *fmt = w->fmt;
+    Uint32 origsampsleft = fmt->fmt.adpcm.samples_left_in_block;
+    int origpos = SDL_RWtell(internal->rw);
+    int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
+    int bpb = (fmt->fmt.adpcm.wSamplesPerBlock * fmt->sample_frame_size);
+    int skipsize = (offset / bpb) * fmt->wBlockAlign;
+    int pos = skipsize + fmt->data_starting_offset;
+    int rc = SDL_RWseek(internal->rw, pos, SEEK_SET);
+    BAIL_IF_MACRO(rc != pos, ERR_IO_ERROR, 0);
+
+    /* The offset we need is in this block, so we need to decode to there. */
+    skipsize += (offset % bpb);
+    rc = (offset % bpb);  /* bytes into this block we need to decode */
+    if (!read_adpcm_block_headers(sample))
+    {
+        SDL_RWseek(internal->rw, origpos, SEEK_SET);  /* try to make sane. */
+        return 0;
+    } /* if */
+
+    /* first sample frame of block is a freebie. :) */
+    fmt->fmt.adpcm.samples_left_in_block--;
+    rc -= fmt->sample_frame_size;
+    while (rc > 0)
+    {
+        if (!decode_adpcm_sample_frame(sample))
+        {
+            SDL_RWseek(internal->rw, origpos, SEEK_SET);
+            fmt->fmt.adpcm.samples_left_in_block = origsampsleft;
+            return 0;
+        } /* if */
+
+        fmt->fmt.adpcm.samples_left_in_block--;
+        rc -= fmt->sample_frame_size;
+    } /* while */
+
+    w->bytesLeft = fmt->total_bytes - skipsize;
+    return 1;  /* success. */
+} /* seek_sample_fmt_adpcm */
+
+
+/*
+ * Read in the adpcm-specific info from disk. This makes this process
+ *  safe regardless of the processor's byte order or how the fmt_t 
+ *  structure is packed.
+ */
+static int read_fmt_adpcm(SDL_RWops *rw, fmt_t *fmt)
+{
+    size_t i;
+
+    SDL_memset(&fmt->fmt.adpcm, '\0', sizeof (fmt->fmt.adpcm));
+    fmt->free = free_fmt_adpcm;
+    fmt->read_sample = read_sample_fmt_adpcm;
+    fmt->rewind_sample = rewind_sample_fmt_adpcm;
+    fmt->seek_sample = seek_sample_fmt_adpcm;
+
+    BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.cbSize), NULL, 0);
+    BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wSamplesPerBlock), NULL, 0);
+    BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wNumCoef), NULL, 0);
+
+    /* fmt->free() is always called, so these malloc()s will be cleaned up. */
+
+    i = sizeof (ADPCMCOEFSET) * fmt->fmt.adpcm.wNumCoef;
+    fmt->fmt.adpcm.aCoef = (ADPCMCOEFSET *) SDL_malloc(i);
+    BAIL_IF_MACRO(fmt->fmt.adpcm.aCoef == NULL, ERR_OUT_OF_MEMORY, 0);
+
+    for (i = 0; i < fmt->fmt.adpcm.wNumCoef; i++)
+    {
+        BAIL_IF_MACRO(!read_le16s(rw, &fmt->fmt.adpcm.aCoef[i].iCoef1), NULL, 0);
+        BAIL_IF_MACRO(!read_le16s(rw, &fmt->fmt.adpcm.aCoef[i].iCoef2), NULL, 0);
+    } /* for */
+
+    i = sizeof (ADPCMBLOCKHEADER) * fmt->wChannels;
+    fmt->fmt.adpcm.blockheaders = (ADPCMBLOCKHEADER *) SDL_malloc(i);
+    BAIL_IF_MACRO(fmt->fmt.adpcm.blockheaders == NULL, ERR_OUT_OF_MEMORY, 0);
+
+    return 1;
+} /* read_fmt_adpcm */
+
+
+
+/*****************************************************************************
+ * Everything else...                                                        *
+ *****************************************************************************/
+
+static int WAV_init(void)
+{
+    return 1;  /* always succeeds. */
+} /* WAV_init */
+
+
+static void WAV_quit(void)
+{
+    /* it's a no-op. */
+} /* WAV_quit */
+
+
+static int read_fmt(SDL_RWops *rw, fmt_t *fmt)
+{
+    /* if it's in this switch statement, we support the format. */
+    switch (fmt->wFormatTag)
+    {
+        case FMT_NORMAL:
+            SNDDBG(("WAV: Appears to be uncompressed audio.\n"));
+            return read_fmt_normal(rw, fmt);
+
+        case FMT_ADPCM:
+            SNDDBG(("WAV: Appears to be ADPCM compressed audio.\n"));
+            return read_fmt_adpcm(rw, fmt);
+
+        /* add other types here. */
+
+        default:
+            SNDDBG(("WAV: Format 0x%X is unknown.\n",
+                    (unsigned int) fmt->wFormatTag));
+            BAIL_MACRO("WAV: Unsupported format", 0);
+    } /* switch */
+
+    SDL_assert(0);  /* shouldn't hit this point. */
+    return 0;
+} /* read_fmt */
+
+
+/*
+ * Locate a specific chunk in the WAVE file by ID...
+ */
+static int find_chunk(SDL_RWops *rw, Uint32 id)
+{
+    Sint32 siz = 0;
+    Uint32 _id = 0;
+    Uint32 pos = SDL_RWtell(rw);
+
+    while (1)
+    {
+        BAIL_IF_MACRO(!read_le32(rw, &_id), NULL, 0);
+        if (_id == id)
+            return 1;
+
+            /* skip ahead and see what next chunk is... */
+        BAIL_IF_MACRO(!read_le32s(rw, &siz), NULL, 0);
+        SDL_assert(siz >= 0);
+        pos += (sizeof (Uint32) * 2) + siz;
+        if (siz > 0)
+            BAIL_IF_MACRO(SDL_RWseek(rw, pos, SEEK_SET) != pos, NULL, 0);
+    } /* while */
+
+    return 0;  /* shouldn't hit this, but just in case... */
+} /* find_chunk */
+
+
+static int WAV_open_internal(Sound_Sample *sample, const char *ext, fmt_t *fmt)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    SDL_RWops *rw = internal->rw;
+    data_t d;
+    wav_t *w;
+    Uint32 pos;
+
+    BAIL_IF_MACRO(SDL_ReadLE32(rw) != riffID, "WAV: Not a RIFF file.", 0);
+    SDL_ReadLE32(rw);  /* throw the length away; we get this info later. */
+    BAIL_IF_MACRO(SDL_ReadLE32(rw) != waveID, "WAV: Not a WAVE file.", 0);
+    BAIL_IF_MACRO(!find_chunk(rw, fmtID), "WAV: No format chunk.", 0);
+    BAIL_IF_MACRO(!read_fmt_chunk(rw, fmt), "WAV: Can't read format chunk.", 0);
+
+    /* !!! FIXME: need float32 format stuff, since it's not just wBitsPerSample. */
+
+    sample->actual.channels = (Uint8) fmt->wChannels;
+    sample->actual.rate = fmt->dwSamplesPerSec;
+    if (fmt->wBitsPerSample == 4)
+        sample->actual.format = AUDIO_S16SYS;
+    else if (fmt->wBitsPerSample == 8)
+        sample->actual.format = AUDIO_U8;
+    else if (fmt->wBitsPerSample == 16)
+        sample->actual.format = AUDIO_S16LSB;
+    else if (fmt->wBitsPerSample == 32)
+        sample->actual.format = AUDIO_S32LSB;
+    else
+    {
+        SNDDBG(("WAV: %d bits per sample!?\n", (int) fmt->wBitsPerSample));
+        BAIL_MACRO("WAV: Unsupported sample size.", 0);
+    } /* else */
+
+    BAIL_IF_MACRO(!read_fmt(rw, fmt), NULL, 0);
+    SDL_RWseek(rw, fmt->next_chunk_offset, SEEK_SET);
+    BAIL_IF_MACRO(!find_chunk(rw, dataID), "WAV: No data chunk.", 0);
+    BAIL_IF_MACRO(!read_data_chunk(rw, &d), "WAV: Can't read data chunk.", 0);
+
+    w = (wav_t *) SDL_malloc(sizeof(wav_t));
+    BAIL_IF_MACRO(w == NULL, ERR_OUT_OF_MEMORY, 0);
+    w->fmt = fmt;
+    fmt->total_bytes = w->bytesLeft = d.chunkSize;
+    fmt->data_starting_offset = SDL_RWtell(rw);
+    fmt->sample_frame_size = ( ((sample->actual.format & 0xFF) / 8) *
+                               sample->actual.channels );
+    internal->decoder_private = (void *) w;
+
+    internal->total_time = (fmt->total_bytes / fmt->dwAvgBytesPerSec) * 1000;
+    internal->total_time += (fmt->total_bytes % fmt->dwAvgBytesPerSec)
+                              *  1000 / fmt->dwAvgBytesPerSec;
+
+    sample->flags = SOUND_SAMPLEFLAG_NONE;
+    if (fmt->seek_sample != NULL)
+        sample->flags |= SOUND_SAMPLEFLAG_CANSEEK;
+
+    SNDDBG(("WAV: Accepting data stream.\n"));
+    return 1; /* we'll handle this data. */
+} /* WAV_open_internal */
+
+
+static int WAV_open(Sound_Sample *sample, const char *ext)
+{
+    int rc;
+
+    fmt_t *fmt = (fmt_t *) SDL_calloc(1, sizeof (fmt_t));
+    BAIL_IF_MACRO(fmt == NULL, ERR_OUT_OF_MEMORY, 0);
+
+    rc = WAV_open_internal(sample, ext, fmt);
+    if (!rc)
+    {
+        if (fmt->free != NULL)
+            fmt->free(fmt);
+        SDL_free(fmt);
+    } /* if */
+
+    return rc;
+} /* WAV_open */
+
+
+static void WAV_close(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+
+    if (w->fmt->free != NULL)
+        w->fmt->free(w->fmt);
+
+    SDL_free(w->fmt);
+    SDL_free(w);
+} /* WAV_close */
+
+
+static Uint32 WAV_read(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    return w->fmt->read_sample(sample);
+} /* WAV_read */
+
+
+static int WAV_rewind(Sound_Sample *sample)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    fmt_t *fmt = w->fmt;
+    int rc = SDL_RWseek(internal->rw, fmt->data_starting_offset, SEEK_SET);
+    BAIL_IF_MACRO(rc != fmt->data_starting_offset, ERR_IO_ERROR, 0);
+    w->bytesLeft = fmt->total_bytes;
+    return fmt->rewind_sample(sample);
+} /* WAV_rewind */
+
+
+static int WAV_seek(Sound_Sample *sample, Uint32 ms)
+{
+    Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque;
+    wav_t *w = (wav_t *) internal->decoder_private;
+    return w->fmt->seek_sample(sample, ms);
+} /* WAV_seek */
+
+
+static const char *extensions_wav[] = { "WAV", NULL };
+const Sound_DecoderFunctions __Sound_DecoderFunctions_WAV =
+{
+    {
+        extensions_wav,
+        "Microsoft WAVE audio format",
+        "Ryan C. Gordon <icculus@icculus.org>",
+        "https://icculus.org/SDL_sound/"
+    },
+
+    WAV_init,       /*   init() method */
+    WAV_quit,       /*   quit() method */
+    WAV_open,       /*   open() method */
+    WAV_close,      /*  close() method */
+    WAV_read,       /*   read() method */
+    WAV_rewind,     /* rewind() method */
+    WAV_seek        /*   seek() method */
+};
+
+#endif /* SOUND_SUPPORTS_WAV */
+
+/* end of SDL_sound_wav.c ... */
+