alphatau 0.1.0

data/ext/alphatau/native/sdl/src/audio/SDL_wave.c

@@ -0,0 +1,694 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "../SDL_internal.h"
+
+/* Microsoft WAVE file loading routines */
+
+#include "SDL_audio.h"
+#include "SDL_wave.h"
+
+
+static int ReadChunk(SDL_RWops * src, Chunk * chunk);
+
+struct MS_ADPCM_decodestate
+{
+    Uint8 hPredictor;
+    Uint16 iDelta;
+    Sint16 iSamp1;
+    Sint16 iSamp2;
+};
+static struct MS_ADPCM_decoder
+{
+    WaveFMT wavefmt;
+    Uint16 wSamplesPerBlock;
+    Uint16 wNumCoef;
+    Sint16 aCoeff[7][2];
+    /* * * */
+    struct MS_ADPCM_decodestate state[2];
+} MS_ADPCM_state;
+
+static int
+InitMS_ADPCM(WaveFMT * format)
+{
+    Uint8 *rogue_feel;
+    int i;
+
+    /* Set the rogue pointer to the MS_ADPCM specific data */
+    MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
+    MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
+    MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
+    MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
+    MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
+    MS_ADPCM_state.wavefmt.bitspersample =
+        SDL_SwapLE16(format->bitspersample);
+    rogue_feel = (Uint8 *) format + sizeof(*format);
+    if (sizeof(*format) == 16) {
+        /* const Uint16 extra_info = ((rogue_feel[1] << 8) | rogue_feel[0]); */
+        rogue_feel += sizeof(Uint16);
+    }
+    MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1] << 8) | rogue_feel[0]);
+    rogue_feel += sizeof(Uint16);
+    MS_ADPCM_state.wNumCoef = ((rogue_feel[1] << 8) | rogue_feel[0]);
+    rogue_feel += sizeof(Uint16);
+    if (MS_ADPCM_state.wNumCoef != 7) {
+        SDL_SetError("Unknown set of MS_ADPCM coefficients");
+        return (-1);
+    }
+    for (i = 0; i < MS_ADPCM_state.wNumCoef; ++i) {
+        MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1] << 8) | rogue_feel[0]);
+        rogue_feel += sizeof(Uint16);
+        MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1] << 8) | rogue_feel[0]);
+        rogue_feel += sizeof(Uint16);
+    }
+    return (0);
+}
+
+static Sint32
+MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state,
+                Uint8 nybble, Sint16 * coeff)
+{
+    const Sint32 max_audioval = ((1 << (16 - 1)) - 1);
+    const Sint32 min_audioval = -(1 << (16 - 1));
+    const Sint32 adaptive[] = {
+        230, 230, 230, 230, 307, 409, 512, 614,
+        768, 614, 512, 409, 307, 230, 230, 230
+    };
+    Sint32 new_sample, delta;
+
+    new_sample = ((state->iSamp1 * coeff[0]) +
+                  (state->iSamp2 * coeff[1])) / 256;
+    if (nybble & 0x08) {
+        new_sample += state->iDelta * (nybble - 0x10);
+    } else {
+        new_sample += state->iDelta * nybble;
+    }
+    if (new_sample < min_audioval) {
+        new_sample = min_audioval;
+    } else if (new_sample > max_audioval) {
+        new_sample = max_audioval;
+    }
+    delta = ((Sint32) state->iDelta * adaptive[nybble]) / 256;
+    if (delta < 16) {
+        delta = 16;
+    }
+    state->iDelta = (Uint16) delta;
+    state->iSamp2 = state->iSamp1;
+    state->iSamp1 = (Sint16) new_sample;
+    return (new_sample);
+}
+
+static int
+MS_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
+{
+    struct MS_ADPCM_decodestate *state[2];
+    Uint8 *freeable, *encoded, *decoded;
+    Sint32 encoded_len, samplesleft;
+    Sint8 nybble;
+    Uint8 stereo;
+    Sint16 *coeff[2];
+    Sint32 new_sample;
+
+    /* Allocate the proper sized output buffer */
+    encoded_len = *audio_len;
+    encoded = *audio_buf;
+    freeable = *audio_buf;
+    *audio_len = (encoded_len / MS_ADPCM_state.wavefmt.blockalign) *
+        MS_ADPCM_state.wSamplesPerBlock *
+        MS_ADPCM_state.wavefmt.channels * sizeof(Sint16);
+    *audio_buf = (Uint8 *) SDL_malloc(*audio_len);
+    if (*audio_buf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    decoded = *audio_buf;
+
+    /* Get ready... Go! */
+    stereo = (MS_ADPCM_state.wavefmt.channels == 2);
+    state[0] = &MS_ADPCM_state.state[0];
+    state[1] = &MS_ADPCM_state.state[stereo];
+    while (encoded_len >= MS_ADPCM_state.wavefmt.blockalign) {
+        /* Grab the initial information for this block */
+        state[0]->hPredictor = *encoded++;
+        if (stereo) {
+            state[1]->hPredictor = *encoded++;
+        }
+        state[0]->iDelta = ((encoded[1] << 8) | encoded[0]);
+        encoded += sizeof(Sint16);
+        if (stereo) {
+            state[1]->iDelta = ((encoded[1] << 8) | encoded[0]);
+            encoded += sizeof(Sint16);
+        }
+        state[0]->iSamp1 = ((encoded[1] << 8) | encoded[0]);
+        encoded += sizeof(Sint16);
+        if (stereo) {
+            state[1]->iSamp1 = ((encoded[1] << 8) | encoded[0]);
+            encoded += sizeof(Sint16);
+        }
+        state[0]->iSamp2 = ((encoded[1] << 8) | encoded[0]);
+        encoded += sizeof(Sint16);
+        if (stereo) {
+            state[1]->iSamp2 = ((encoded[1] << 8) | encoded[0]);
+            encoded += sizeof(Sint16);
+        }
+        coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor];
+        coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor];
+
+        /* Store the two initial samples we start with */
+        decoded[0] = state[0]->iSamp2 & 0xFF;
+        decoded[1] = state[0]->iSamp2 >> 8;
+        decoded += 2;
+        if (stereo) {
+            decoded[0] = state[1]->iSamp2 & 0xFF;
+            decoded[1] = state[1]->iSamp2 >> 8;
+            decoded += 2;
+        }
+        decoded[0] = state[0]->iSamp1 & 0xFF;
+        decoded[1] = state[0]->iSamp1 >> 8;
+        decoded += 2;
+        if (stereo) {
+            decoded[0] = state[1]->iSamp1 & 0xFF;
+            decoded[1] = state[1]->iSamp1 >> 8;
+            decoded += 2;
+        }
+
+        /* Decode and store the other samples in this block */
+        samplesleft = (MS_ADPCM_state.wSamplesPerBlock - 2) *
+            MS_ADPCM_state.wavefmt.channels;
+        while (samplesleft > 0) {
+            nybble = (*encoded) >> 4;
+            new_sample = MS_ADPCM_nibble(state[0], nybble, coeff[0]);
+            decoded[0] = new_sample & 0xFF;
+            new_sample >>= 8;
+            decoded[1] = new_sample & 0xFF;
+            decoded += 2;
+
+            nybble = (*encoded) & 0x0F;
+            new_sample = MS_ADPCM_nibble(state[1], nybble, coeff[1]);
+            decoded[0] = new_sample & 0xFF;
+            new_sample >>= 8;
+            decoded[1] = new_sample & 0xFF;
+            decoded += 2;
+
+            ++encoded;
+            samplesleft -= 2;
+        }
+        encoded_len -= MS_ADPCM_state.wavefmt.blockalign;
+    }
+    SDL_free(freeable);
+    return (0);
+}
+
+struct IMA_ADPCM_decodestate
+{
+    Sint32 sample;
+    Sint8 index;
+};
+static struct IMA_ADPCM_decoder
+{
+    WaveFMT wavefmt;
+    Uint16 wSamplesPerBlock;
+    /* * * */
+    struct IMA_ADPCM_decodestate state[2];
+} IMA_ADPCM_state;
+
+static int
+InitIMA_ADPCM(WaveFMT * format)
+{
+    Uint8 *rogue_feel;
+
+    /* Set the rogue pointer to the IMA_ADPCM specific data */
+    IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
+    IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
+    IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
+    IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
+    IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
+    IMA_ADPCM_state.wavefmt.bitspersample =
+        SDL_SwapLE16(format->bitspersample);
+    rogue_feel = (Uint8 *) format + sizeof(*format);
+    if (sizeof(*format) == 16) {
+        /* const Uint16 extra_info = ((rogue_feel[1] << 8) | rogue_feel[0]); */
+        rogue_feel += sizeof(Uint16);
+    }
+    IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1] << 8) | rogue_feel[0]);
+    return (0);
+}
+
+static Sint32
+IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state, Uint8 nybble)
+{
+    const Sint32 max_audioval = ((1 << (16 - 1)) - 1);
+    const Sint32 min_audioval = -(1 << (16 - 1));
+    const int index_table[16] = {
+        -1, -1, -1, -1,
+        2, 4, 6, 8,
+        -1, -1, -1, -1,
+        2, 4, 6, 8
+    };
+    const Sint32 step_table[89] = {
+        7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31,
+        34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130,
+        143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408,
+        449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282,
+        1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
+        3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630,
+        9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350,
+        22385, 24623, 27086, 29794, 32767
+    };
+    Sint32 delta, step;
+
+    /* Compute difference and new sample value */
+    if (state->index > 88) {
+        state->index = 88;
+    } else if (state->index < 0) {
+        state->index = 0;
+    }
+    /* explicit cast to avoid gcc warning about using 'char' as array index */
+    step = step_table[(int)state->index];
+    delta = step >> 3;
+    if (nybble & 0x04)
+        delta += step;
+    if (nybble & 0x02)
+        delta += (step >> 1);
+    if (nybble & 0x01)
+        delta += (step >> 2);
+    if (nybble & 0x08)
+        delta = -delta;
+    state->sample += delta;
+
+    /* Update index value */
+    state->index += index_table[nybble];
+
+    /* Clamp output sample */
+    if (state->sample > max_audioval) {
+        state->sample = max_audioval;
+    } else if (state->sample < min_audioval) {
+        state->sample = min_audioval;
+    }
+    return (state->sample);
+}
+
+/* Fill the decode buffer with a channel block of data (8 samples) */
+static void
+Fill_IMA_ADPCM_block(Uint8 * decoded, Uint8 * encoded,
+                     int channel, int numchannels,
+                     struct IMA_ADPCM_decodestate *state)
+{
+    int i;
+    Sint8 nybble;
+    Sint32 new_sample;
+
+    decoded += (channel * 2);
+    for (i = 0; i < 4; ++i) {
+        nybble = (*encoded) & 0x0F;
+        new_sample = IMA_ADPCM_nibble(state, nybble);
+        decoded[0] = new_sample & 0xFF;
+        new_sample >>= 8;
+        decoded[1] = new_sample & 0xFF;
+        decoded += 2 * numchannels;
+
+        nybble = (*encoded) >> 4;
+        new_sample = IMA_ADPCM_nibble(state, nybble);
+        decoded[0] = new_sample & 0xFF;
+        new_sample >>= 8;
+        decoded[1] = new_sample & 0xFF;
+        decoded += 2 * numchannels;
+
+        ++encoded;
+    }
+}
+
+static int
+IMA_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
+{
+    struct IMA_ADPCM_decodestate *state;
+    Uint8 *freeable, *encoded, *decoded;
+    Sint32 encoded_len, samplesleft;
+    unsigned int c, channels;
+
+    /* Check to make sure we have enough variables in the state array */
+    channels = IMA_ADPCM_state.wavefmt.channels;
+    if (channels > SDL_arraysize(IMA_ADPCM_state.state)) {
+        SDL_SetError("IMA ADPCM decoder can only handle %u channels",
+                     (unsigned int)SDL_arraysize(IMA_ADPCM_state.state));
+        return (-1);
+    }
+    state = IMA_ADPCM_state.state;
+
+    /* Allocate the proper sized output buffer */
+    encoded_len = *audio_len;
+    encoded = *audio_buf;
+    freeable = *audio_buf;
+    *audio_len = (encoded_len / IMA_ADPCM_state.wavefmt.blockalign) *
+        IMA_ADPCM_state.wSamplesPerBlock *
+        IMA_ADPCM_state.wavefmt.channels * sizeof(Sint16);
+    *audio_buf = (Uint8 *) SDL_malloc(*audio_len);
+    if (*audio_buf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    decoded = *audio_buf;
+
+    /* Get ready... Go! */
+    while (encoded_len >= IMA_ADPCM_state.wavefmt.blockalign) {
+        /* Grab the initial information for this block */
+        for (c = 0; c < channels; ++c) {
+            /* Fill the state information for this block */
+            state[c].sample = ((encoded[1] << 8) | encoded[0]);
+            encoded += 2;
+            if (state[c].sample & 0x8000) {
+                state[c].sample -= 0x10000;
+            }
+            state[c].index = *encoded++;
+            /* Reserved byte in buffer header, should be 0 */
+            if (*encoded++ != 0) {
+                /* Uh oh, corrupt data?  Buggy code? */ ;
+            }
+
+            /* Store the initial sample we start with */
+            decoded[0] = (Uint8) (state[c].sample & 0xFF);
+            decoded[1] = (Uint8) (state[c].sample >> 8);
+            decoded += 2;
+        }
+
+        /* Decode and store the other samples in this block */
+        samplesleft = (IMA_ADPCM_state.wSamplesPerBlock - 1) * channels;
+        while (samplesleft > 0) {
+            for (c = 0; c < channels; ++c) {
+                Fill_IMA_ADPCM_block(decoded, encoded,
+                                     c, channels, &state[c]);
+                encoded += 4;
+                samplesleft -= 8;
+            }
+            decoded += (channels * 8 * 2);
+        }
+        encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;
+    }
+    SDL_free(freeable);
+    return (0);
+}
+
+
+static int
+ConvertSint24ToSint32(Uint8 ** audio_buf, Uint32 * audio_len)
+{
+    const double DIVBY8388608 = 0.00000011920928955078125;
+    const Uint32 original_len = *audio_len;
+    const Uint32 samples = original_len / 3;
+    const Uint32 expanded_len = samples * sizeof (Uint32);
+    Uint8 *ptr = (Uint8 *) SDL_realloc(*audio_buf, expanded_len);
+    const Uint8 *src;
+    Uint32 *dst;
+    Uint32 i;
+
+    if (!ptr) {
+        return SDL_OutOfMemory();
+    }
+
+    *audio_buf = ptr;
+    *audio_len = expanded_len;
+
+    /* work from end to start, since we're expanding in-place. */
+    src = (ptr + original_len) - 3;
+    dst = ((Uint32 *) (ptr + expanded_len)) - 1;
+    for (i = 0; i < samples; i++) {
+        /* There's probably a faster way to do all this. */
+        const Sint32 converted = ((Sint32) ( (((Uint32) src[2]) << 24) |
+                                             (((Uint32) src[1]) << 16) |
+                                             (((Uint32) src[0]) << 8) )) >> 8;
+        const double scaled = (((double) converted) * DIVBY8388608);
+        src -= 3;
+        *(dst--) = (Sint32) (scaled * 2147483647.0);
+    }
+
+    return 0;
+}
+
+
+/* GUIDs that are used by WAVE_FORMAT_EXTENSIBLE */
+static const Uint8 extensible_pcm_guid[16] = { 1, 0, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113 };
+static const Uint8 extensible_ieee_guid[16] = { 3, 0, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113 };
+
+SDL_AudioSpec *
+SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
+               SDL_AudioSpec * spec, Uint8 ** audio_buf, Uint32 * audio_len)
+{
+    int was_error;
+    Chunk chunk;
+    int lenread;
+    int IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded;
+    int samplesize;
+
+    /* WAV magic header */
+    Uint32 RIFFchunk;
+    Uint32 wavelen = 0;
+    Uint32 WAVEmagic;
+    Uint32 headerDiff = 0;
+
+    /* FMT chunk */
+    WaveFMT *format = NULL;
+    WaveExtensibleFMT *ext = NULL;
+
+    SDL_zero(chunk);
+
+    /* Make sure we are passed a valid data source */
+    was_error = 0;
+    if (src == NULL) {
+        was_error = 1;
+        goto done;
+    }
+
+    /* Check the magic header */
+    RIFFchunk = SDL_ReadLE32(src);
+    wavelen = SDL_ReadLE32(src);
+    if (wavelen == WAVE) {      /* The RIFFchunk has already been read */
+        WAVEmagic = wavelen;
+        wavelen = RIFFchunk;
+        RIFFchunk = RIFF;
+    } else {
+        WAVEmagic = SDL_ReadLE32(src);
+    }
+    if ((RIFFchunk != RIFF) || (WAVEmagic != WAVE)) {
+        SDL_SetError("Unrecognized file type (not WAVE)");
+        was_error = 1;
+        goto done;
+    }
+    headerDiff += sizeof(Uint32);       /* for WAVE */
+
+    /* Read the audio data format chunk */
+    chunk.data = NULL;
+    do {
+        SDL_free(chunk.data);
+        chunk.data = NULL;
+        lenread = ReadChunk(src, &chunk);
+        if (lenread < 0) {
+            was_error = 1;
+            goto done;
+        }
+        /* 2 Uint32's for chunk header+len, plus the lenread */
+        headerDiff += lenread + 2 * sizeof(Uint32);
+    } while ((chunk.magic == FACT) || (chunk.magic == LIST) || (chunk.magic == BEXT) || (chunk.magic == JUNK));
+
+    /* Decode the audio data format */
+    format = (WaveFMT *) chunk.data;
+    if (chunk.magic != FMT) {
+        SDL_SetError("Complex WAVE files not supported");
+        was_error = 1;
+        goto done;
+    }
+    IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
+    switch (SDL_SwapLE16(format->encoding)) {
+    case PCM_CODE:
+        /* We can understand this */
+        break;
+    case IEEE_FLOAT_CODE:
+        IEEE_float_encoded = 1;
+        /* We can understand this */
+        break;
+    case MS_ADPCM_CODE:
+        /* Try to understand this */
+        if (InitMS_ADPCM(format) < 0) {
+            was_error = 1;
+            goto done;
+        }
+        MS_ADPCM_encoded = 1;
+        break;
+    case IMA_ADPCM_CODE:
+        /* Try to understand this */
+        if (InitIMA_ADPCM(format) < 0) {
+            was_error = 1;
+            goto done;
+        }
+        IMA_ADPCM_encoded = 1;
+        break;
+    case EXTENSIBLE_CODE:
+        /* note that this ignores channel masks, smaller valid bit counts
+           inside a larger container, and most subtypes. This is just enough
+           to get things that didn't really _need_ WAVE_FORMAT_EXTENSIBLE
+           to be useful working when they use this format flag. */
+        ext = (WaveExtensibleFMT *) format;
+        if (SDL_SwapLE16(ext->size) < 22) {
+            SDL_SetError("bogus extended .wav header");
+            was_error = 1;
+            goto done;
+        }
+        if (SDL_memcmp(ext->subformat, extensible_pcm_guid, 16) == 0) {
+            break;  /* cool. */
+        } else if (SDL_memcmp(ext->subformat, extensible_ieee_guid, 16) == 0) {
+            IEEE_float_encoded = 1;
+            break;
+        }
+        break;
+    case MP3_CODE:
+        SDL_SetError("MPEG Layer 3 data not supported");
+        was_error = 1;
+        goto done;
+    default:
+        SDL_SetError("Unknown WAVE data format: 0x%.4x",
+                     SDL_SwapLE16(format->encoding));
+        was_error = 1;
+        goto done;
+    }
+    SDL_zerop(spec);
+    spec->freq = SDL_SwapLE32(format->frequency);
+
+    if (IEEE_float_encoded) {
+        if ((SDL_SwapLE16(format->bitspersample)) != 32) {
+            was_error = 1;
+        } else {
+            spec->format = AUDIO_F32;
+        }
+    } else {
+        switch (SDL_SwapLE16(format->bitspersample)) {
+        case 4:
+            if (MS_ADPCM_encoded || IMA_ADPCM_encoded) {
+                spec->format = AUDIO_S16;
+            } else {
+                was_error = 1;
+            }
+            break;
+        case 8:
+            spec->format = AUDIO_U8;
+            break;
+        case 16:
+            spec->format = AUDIO_S16;
+            break;
+        case 24:  /* convert this. */
+            spec->format = AUDIO_S32;
+            break;
+        case 32:
+            spec->format = AUDIO_S32;
+            break;
+        default:
+            was_error = 1;
+            break;
+        }
+    }
+
+    if (was_error) {
+        SDL_SetError("Unknown %d-bit PCM data format",
+                     SDL_SwapLE16(format->bitspersample));
+        goto done;
+    }
+    spec->channels = (Uint8) SDL_SwapLE16(format->channels);
+    spec->samples = 4096;       /* Good default buffer size */
+
+    /* Read the audio data chunk */
+    *audio_buf = NULL;
+    do {
+        SDL_free(*audio_buf);
+        *audio_buf = NULL;
+        lenread = ReadChunk(src, &chunk);
+        if (lenread < 0) {
+            was_error = 1;
+            goto done;
+        }
+        *audio_len = lenread;
+        *audio_buf = chunk.data;
+        if (chunk.magic != DATA)
+            headerDiff += lenread + 2 * sizeof(Uint32);
+    } while (chunk.magic != DATA);
+    headerDiff += 2 * sizeof(Uint32);   /* for the data chunk and len */
+
+    if (MS_ADPCM_encoded) {
+        if (MS_ADPCM_decode(audio_buf, audio_len) < 0) {
+            was_error = 1;
+            goto done;
+        }
+    }
+    if (IMA_ADPCM_encoded) {
+        if (IMA_ADPCM_decode(audio_buf, audio_len) < 0) {
+            was_error = 1;
+            goto done;
+        }
+    }
+
+    if (SDL_SwapLE16(format->bitspersample) == 24) {
+        if (ConvertSint24ToSint32(audio_buf, audio_len) < 0) {
+            was_error = 1;
+            goto done;
+        }
+    }
+
+    /* Don't return a buffer that isn't a multiple of samplesize */
+    samplesize = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels;
+    *audio_len &= ~(samplesize - 1);
+
+  done:
+    SDL_free(format);
+    if (src) {
+        if (freesrc) {
+            SDL_RWclose(src);
+        } else {
+            /* seek to the end of the file (given by the RIFF chunk) */
+            SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
+        }
+    }
+    if (was_error) {
+        spec = NULL;
+    }
+    return (spec);
+}
+
+/* Since the WAV memory is allocated in the shared library, it must also
+   be freed here.  (Necessary under Win32, VC++)
+ */
+void
+SDL_FreeWAV(Uint8 * audio_buf)
+{
+    SDL_free(audio_buf);
+}
+
+static int
+ReadChunk(SDL_RWops * src, Chunk * chunk)
+{
+    chunk->magic = SDL_ReadLE32(src);
+    chunk->length = SDL_ReadLE32(src);
+    chunk->data = (Uint8 *) SDL_malloc(chunk->length);
+    if (chunk->data == NULL) {
+        return SDL_OutOfMemory();
+    }
+    if (SDL_RWread(src, chunk->data, chunk->length, 1) != 1) {
+        SDL_free(chunk->data);
+        chunk->data = NULL;
+        return SDL_Error(SDL_EFREAD);
+    }
+    return (chunk->length);
+}
+
+/* vi: set ts=4 sw=4 expandtab: */