src/audio/SDL_wave.c

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2011 Sam Lantinga

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Lesser General Public
    License as published by the Free Software Foundation; either
    version 2.1 of the License, or (at your option) any later version.

    This library is distributed in the hope that it will be useful,
    but WITHOUT ANY WARRANTY; without even the implied warranty of
    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
    Lesser General Public License for more details.

    You should have received a copy of the GNU Lesser General Public
    License along with this library; if not, write to the Free Software
    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA

    Sam Lantinga
    slouken@libsdl.org
*/
#include "SDL_config.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;
    Uint16 extra_info;
    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) {
        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, 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) {
        SDL_Error(SDL_ENOMEM);
        return (-1);
    }
    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;
    Uint16 extra_info;

    /* 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) {
        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 */
    step = step_table[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];
    if (state->index > 88) {
        state->index = 88;
    } else if (state->index < 0) {
        state->index = 0;
    }

    /* 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 %d channels",
                     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) {
        SDL_Error(SDL_ENOMEM);
        return (-1);
    }
    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);
}

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;

    /* 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 {
        if (chunk.data != NULL) {
            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));

    /* 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 MP3_CODE:
        SDL_SetError("MPEG Layer 3 data not supported",
                     SDL_SwapLE16(format->encoding));
        was_error = 1;
        goto done;
    default:
        SDL_SetError("Unknown WAVE data format: 0x%.4x",
                     SDL_SwapLE16(format->encoding));
        was_error = 1;
        goto done;
    }
    SDL_memset(spec, 0, (sizeof *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 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 {
        if (*audio_buf != NULL) {
            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;
        }
    }

    /* 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:
    if (format != NULL) {
        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)
{
    if (audio_buf != NULL) {
        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) {
        SDL_Error(SDL_ENOMEM);
        return (-1);
    }
    if (SDL_RWread(src, chunk->data, chunk->length, 1) != 1) {
        SDL_Error(SDL_EFREAD);
        SDL_free(chunk->data);
        chunk->data = NULL;
        return (-1);
    }
    return (chunk->length);
}

/* vi: set ts=4 sw=4 expandtab: */