src/audio/SDL_audiocvt.c
author Aaron Wishnick <schnarf@gmail.com>
Wed, 18 Jun 2008 18:55:50 +0000
branchgsoc2008_audio_resampling
changeset 2656 dd74182b3c3c
parent 2655 b8e736c8a5a8
child 2657 29306e52dab8
permissions -rw-r--r--
Began implementing IIR and FIR filters, and got zero stuffing and sample discarding working.

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2006 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"
#include <math.h>

/* Functions for audio drivers to perform runtime conversion of audio format */

#include "SDL_audio.h"
#include "SDL_audio_c.h"

#define DEBUG_CONVERT

/* Effectively mix right and left channels into a single channel */
static void SDLCALL
SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;
    Sint32 sample;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting to mono\n");
#endif
    switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
    case AUDIO_U8:
        {
            Uint8 *src, *dst;

            src = cvt->buf;
            dst = cvt->buf;
            for (i = cvt->len_cvt / 2; i; --i) {
                sample = src[0] + src[1];
                *dst = (Uint8) (sample / 2);
                src += 2;
                dst += 1;
            }
        }
        break;

    case AUDIO_S8:
        {
            Sint8 *src, *dst;

            src = (Sint8 *) cvt->buf;
            dst = (Sint8 *) cvt->buf;
            for (i = cvt->len_cvt / 2; i; --i) {
                sample = src[0] + src[1];
                *dst = (Sint8) (sample / 2);
                src += 2;
                dst += 1;
            }
        }
        break;

    case AUDIO_U16:
        {
            Uint8 *src, *dst;

            src = cvt->buf;
            dst = cvt->buf;
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 4; i; --i) {
                    sample = (Uint16) ((src[0] << 8) | src[1]) +
                        (Uint16) ((src[2] << 8) | src[3]);
                    sample /= 2;
                    dst[1] = (sample & 0xFF);
                    sample >>= 8;
                    dst[0] = (sample & 0xFF);
                    src += 4;
                    dst += 2;
                }
            } else {
                for (i = cvt->len_cvt / 4; i; --i) {
                    sample = (Uint16) ((src[1] << 8) | src[0]) +
                        (Uint16) ((src[3] << 8) | src[2]);
                    sample /= 2;
                    dst[0] = (sample & 0xFF);
                    sample >>= 8;
                    dst[1] = (sample & 0xFF);
                    src += 4;
                    dst += 2;
                }
            }
        }
        break;

    case AUDIO_S16:
        {
            Uint8 *src, *dst;

            src = cvt->buf;
            dst = cvt->buf;
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 4; i; --i) {
                    sample = (Sint16) ((src[0] << 8) | src[1]) +
                        (Sint16) ((src[2] << 8) | src[3]);
                    sample /= 2;
                    dst[1] = (sample & 0xFF);
                    sample >>= 8;
                    dst[0] = (sample & 0xFF);
                    src += 4;
                    dst += 2;
                }
            } else {
                for (i = cvt->len_cvt / 4; i; --i) {
                    sample = (Sint16) ((src[1] << 8) | src[0]) +
                        (Sint16) ((src[3] << 8) | src[2]);
                    sample /= 2;
                    dst[0] = (sample & 0xFF);
                    sample >>= 8;
                    dst[1] = (sample & 0xFF);
                    src += 4;
                    dst += 2;
                }
            }
        }
        break;

    case AUDIO_S32:
        {
            const Uint32 *src = (const Uint32 *) cvt->buf;
            Uint32 *dst = (Uint32 *) cvt->buf;
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 8; i; --i, src += 2) {
                    const Sint64 added =
                        (((Sint64) (Sint32) SDL_SwapBE32(src[0])) +
                         ((Sint64) (Sint32) SDL_SwapBE32(src[1])));
                    *(dst++) = SDL_SwapBE32((Uint32) ((Sint32) (added / 2)));
                }
            } else {
                for (i = cvt->len_cvt / 8; i; --i, src += 2) {
                    const Sint64 added =
                        (((Sint64) (Sint32) SDL_SwapLE32(src[0])) +
                         ((Sint64) (Sint32) SDL_SwapLE32(src[1])));
                    *(dst++) = SDL_SwapLE32((Uint32) ((Sint32) (added / 2)));
                }
            }
        }
        break;

    case AUDIO_F32:
        {
            const float *src = (const float *) cvt->buf;
            float *dst = (float *) cvt->buf;
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 8; i; --i, src += 2) {
                    const float src1 = SDL_SwapFloatBE(src[0]);
                    const float src2 = SDL_SwapFloatBE(src[1]);
                    const double added = ((double) src1) + ((double) src2);
                    const float halved = (float) (added * 0.5);
                    *(dst++) = SDL_SwapFloatBE(halved);
                }
            } else {
                for (i = cvt->len_cvt / 8; i; --i, src += 2) {
                    const float src1 = SDL_SwapFloatLE(src[0]);
                    const float src2 = SDL_SwapFloatLE(src[1]);
                    const double added = ((double) src1) + ((double) src2);
                    const float halved = (float) (added * 0.5);
                    *(dst++) = SDL_SwapFloatLE(halved);
                }
            }
        }
        break;
    }

    cvt->len_cvt /= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Discard top 4 channels */
static void SDLCALL
SDL_ConvertStrip(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting down from 6 channels to stereo\n");
#endif

#define strip_chans_6_to_2(type) \
    { \
        const type *src = (const type *) cvt->buf; \
        type *dst = (type *) cvt->buf; \
        for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
            dst[0] = src[0]; \
            dst[1] = src[1]; \
            src += 6; \
            dst += 2; \
        } \
    }

    /* this function only cares about typesize, and data as a block of bits. */
    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        strip_chans_6_to_2(Uint8);
        break;
    case 16:
        strip_chans_6_to_2(Uint16);
        break;
    case 32:
        strip_chans_6_to_2(Uint32);
        break;
    }

#undef strip_chans_6_to_2

    cvt->len_cvt /= 3;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Discard top 2 channels of 6 */
static void SDLCALL
SDL_ConvertStrip_2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting 6 down to quad\n");
#endif

#define strip_chans_6_to_4(type) \
    { \
        const type *src = (const type *) cvt->buf; \
        type *dst = (type *) cvt->buf; \
        for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
            dst[0] = src[0]; \
            dst[1] = src[1]; \
            dst[2] = src[2]; \
            dst[3] = src[3]; \
            src += 6; \
            dst += 4; \
        } \
    }

    /* this function only cares about typesize, and data as a block of bits. */
    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        strip_chans_6_to_4(Uint8);
        break;
    case 16:
        strip_chans_6_to_4(Uint16);
        break;
    case 32:
        strip_chans_6_to_4(Uint32);
        break;
    }

#undef strip_chans_6_to_4

    cvt->len_cvt /= 6;
    cvt->len_cvt *= 4;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

/* Duplicate a mono channel to both stereo channels */
static void SDLCALL
SDL_ConvertStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting to stereo\n");
#endif

#define dup_chans_1_to_2(type) \
    { \
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
        type *dst = (type *) (cvt->buf + cvt->len_cvt * 2); \
        for (i = cvt->len_cvt / 2; i; --i, --src) { \
            const type val = *src; \
            dst -= 2; \
            dst[0] = dst[1] = val; \
        } \
    }

    /* this function only cares about typesize, and data as a block of bits. */
    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        dup_chans_1_to_2(Uint8);
        break;
    case 16:
        dup_chans_1_to_2(Uint16);
        break;
    case 32:
        dup_chans_1_to_2(Uint32);
        break;
    }

#undef dup_chans_1_to_2

    cvt->len_cvt *= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Duplicate a stereo channel to a pseudo-5.1 stream */
static void SDLCALL
SDL_ConvertSurround(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting stereo to surround\n");
#endif

    switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
    case AUDIO_U8:
        {
            Uint8 *src, *dst, lf, rf, ce;

            src = (Uint8 *) (cvt->buf + cvt->len_cvt);
            dst = (Uint8 *) (cvt->buf + cvt->len_cvt * 3);
            for (i = cvt->len_cvt; i; --i) {
                dst -= 6;
                src -= 2;
                lf = src[0];
                rf = src[1];
                ce = (lf / 2) + (rf / 2);
                dst[0] = lf;
                dst[1] = rf;
                dst[2] = lf - ce;
                dst[3] = rf - ce;
                dst[4] = ce;
                dst[5] = ce;
            }
        }
        break;

    case AUDIO_S8:
        {
            Sint8 *src, *dst, lf, rf, ce;

            src = (Sint8 *) cvt->buf + cvt->len_cvt;
            dst = (Sint8 *) cvt->buf + cvt->len_cvt * 3;
            for (i = cvt->len_cvt; i; --i) {
                dst -= 6;
                src -= 2;
                lf = src[0];
                rf = src[1];
                ce = (lf / 2) + (rf / 2);
                dst[0] = lf;
                dst[1] = rf;
                dst[2] = lf - ce;
                dst[3] = rf - ce;
                dst[4] = ce;
                dst[5] = ce;
            }
        }
        break;

    case AUDIO_U16:
        {
            Uint8 *src, *dst;
            Uint16 lf, rf, ce, lr, rr;

            src = cvt->buf + cvt->len_cvt;
            dst = cvt->buf + cvt->len_cvt * 3;

            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 4; i; --i) {
                    dst -= 12;
                    src -= 4;
                    lf = (Uint16) ((src[0] << 8) | src[1]);
                    rf = (Uint16) ((src[2] << 8) | src[3]);
                    ce = (lf / 2) + (rf / 2);
                    rr = lf - ce;
                    lr = rf - ce;
                    dst[1] = (lf & 0xFF);
                    dst[0] = ((lf >> 8) & 0xFF);
                    dst[3] = (rf & 0xFF);
                    dst[2] = ((rf >> 8) & 0xFF);

                    dst[1 + 4] = (lr & 0xFF);
                    dst[0 + 4] = ((lr >> 8) & 0xFF);
                    dst[3 + 4] = (rr & 0xFF);
                    dst[2 + 4] = ((rr >> 8) & 0xFF);

                    dst[1 + 8] = (ce & 0xFF);
                    dst[0 + 8] = ((ce >> 8) & 0xFF);
                    dst[3 + 8] = (ce & 0xFF);
                    dst[2 + 8] = ((ce >> 8) & 0xFF);
                }
            } else {
                for (i = cvt->len_cvt / 4; i; --i) {
                    dst -= 12;
                    src -= 4;
                    lf = (Uint16) ((src[1] << 8) | src[0]);
                    rf = (Uint16) ((src[3] << 8) | src[2]);
                    ce = (lf / 2) + (rf / 2);
                    rr = lf - ce;
                    lr = rf - ce;
                    dst[0] = (lf & 0xFF);
                    dst[1] = ((lf >> 8) & 0xFF);
                    dst[2] = (rf & 0xFF);
                    dst[3] = ((rf >> 8) & 0xFF);

                    dst[0 + 4] = (lr & 0xFF);
                    dst[1 + 4] = ((lr >> 8) & 0xFF);
                    dst[2 + 4] = (rr & 0xFF);
                    dst[3 + 4] = ((rr >> 8) & 0xFF);

                    dst[0 + 8] = (ce & 0xFF);
                    dst[1 + 8] = ((ce >> 8) & 0xFF);
                    dst[2 + 8] = (ce & 0xFF);
                    dst[3 + 8] = ((ce >> 8) & 0xFF);
                }
            }
        }
        break;

    case AUDIO_S16:
        {
            Uint8 *src, *dst;
            Sint16 lf, rf, ce, lr, rr;

            src = cvt->buf + cvt->len_cvt;
            dst = cvt->buf + cvt->len_cvt * 3;

            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 4; i; --i) {
                    dst -= 12;
                    src -= 4;
                    lf = (Sint16) ((src[0] << 8) | src[1]);
                    rf = (Sint16) ((src[2] << 8) | src[3]);
                    ce = (lf / 2) + (rf / 2);
                    rr = lf - ce;
                    lr = rf - ce;
                    dst[1] = (lf & 0xFF);
                    dst[0] = ((lf >> 8) & 0xFF);
                    dst[3] = (rf & 0xFF);
                    dst[2] = ((rf >> 8) & 0xFF);

                    dst[1 + 4] = (lr & 0xFF);
                    dst[0 + 4] = ((lr >> 8) & 0xFF);
                    dst[3 + 4] = (rr & 0xFF);
                    dst[2 + 4] = ((rr >> 8) & 0xFF);

                    dst[1 + 8] = (ce & 0xFF);
                    dst[0 + 8] = ((ce >> 8) & 0xFF);
                    dst[3 + 8] = (ce & 0xFF);
                    dst[2 + 8] = ((ce >> 8) & 0xFF);
                }
            } else {
                for (i = cvt->len_cvt / 4; i; --i) {
                    dst -= 12;
                    src -= 4;
                    lf = (Sint16) ((src[1] << 8) | src[0]);
                    rf = (Sint16) ((src[3] << 8) | src[2]);
                    ce = (lf / 2) + (rf / 2);
                    rr = lf - ce;
                    lr = rf - ce;
                    dst[0] = (lf & 0xFF);
                    dst[1] = ((lf >> 8) & 0xFF);
                    dst[2] = (rf & 0xFF);
                    dst[3] = ((rf >> 8) & 0xFF);

                    dst[0 + 4] = (lr & 0xFF);
                    dst[1 + 4] = ((lr >> 8) & 0xFF);
                    dst[2 + 4] = (rr & 0xFF);
                    dst[3 + 4] = ((rr >> 8) & 0xFF);

                    dst[0 + 8] = (ce & 0xFF);
                    dst[1 + 8] = ((ce >> 8) & 0xFF);
                    dst[2 + 8] = (ce & 0xFF);
                    dst[3 + 8] = ((ce >> 8) & 0xFF);
                }
            }
        }
        break;

    case AUDIO_S32:
        {
            Sint32 lf, rf, ce;
            const Uint32 *src = (const Uint32 *) cvt->buf + cvt->len_cvt;
            Uint32 *dst = (Uint32 *) cvt->buf + cvt->len_cvt * 3;

            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 8; i; --i) {
                    dst -= 6;
                    src -= 2;
                    lf = (Sint32) SDL_SwapBE32(src[0]);
                    rf = (Sint32) SDL_SwapBE32(src[1]);
                    ce = (lf / 2) + (rf / 2);
                    dst[0] = SDL_SwapBE32((Uint32) lf);
                    dst[1] = SDL_SwapBE32((Uint32) rf);
                    dst[2] = SDL_SwapBE32((Uint32) (lf - ce));
                    dst[3] = SDL_SwapBE32((Uint32) (rf - ce));
                    dst[4] = SDL_SwapBE32((Uint32) ce);
                    dst[5] = SDL_SwapBE32((Uint32) ce);
                }
            } else {
                for (i = cvt->len_cvt / 8; i; --i) {
                    dst -= 6;
                    src -= 2;
                    lf = (Sint32) SDL_SwapLE32(src[0]);
                    rf = (Sint32) SDL_SwapLE32(src[1]);
                    ce = (lf / 2) + (rf / 2);
                    dst[0] = src[0];
                    dst[1] = src[1];
                    dst[2] = SDL_SwapLE32((Uint32) (lf - ce));
                    dst[3] = SDL_SwapLE32((Uint32) (rf - ce));
                    dst[4] = SDL_SwapLE32((Uint32) ce);
                    dst[5] = SDL_SwapLE32((Uint32) ce);
                }
            }
        }
        break;

    case AUDIO_F32:
        {
            float lf, rf, ce;
            const float *src = (const float *) cvt->buf + cvt->len_cvt;
            float *dst = (float *) cvt->buf + cvt->len_cvt * 3;

            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 8; i; --i) {
                    dst -= 6;
                    src -= 2;
                    lf = SDL_SwapFloatBE(src[0]);
                    rf = SDL_SwapFloatBE(src[1]);
                    ce = (lf * 0.5f) + (rf * 0.5f);
                    dst[0] = src[0];
                    dst[1] = src[1];
                    dst[2] = SDL_SwapFloatBE(lf - ce);
                    dst[3] = SDL_SwapFloatBE(rf - ce);
                    dst[4] = dst[5] = SDL_SwapFloatBE(ce);
                }
            } else {
                for (i = cvt->len_cvt / 8; i; --i) {
                    dst -= 6;
                    src -= 2;
                    lf = SDL_SwapFloatLE(src[0]);
                    rf = SDL_SwapFloatLE(src[1]);
                    ce = (lf * 0.5f) + (rf * 0.5f);
                    dst[0] = src[0];
                    dst[1] = src[1];
                    dst[2] = SDL_SwapFloatLE(lf - ce);
                    dst[3] = SDL_SwapFloatLE(rf - ce);
                    dst[4] = dst[5] = SDL_SwapFloatLE(ce);
                }
            }
        }
        break;

    }
    cvt->len_cvt *= 3;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Duplicate a stereo channel to a pseudo-4.0 stream */
static void SDLCALL
SDL_ConvertSurround_4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting stereo to quad\n");
#endif

    switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
    case AUDIO_U8:
        {
            Uint8 *src, *dst, lf, rf, ce;

            src = (Uint8 *) (cvt->buf + cvt->len_cvt);
            dst = (Uint8 *) (cvt->buf + cvt->len_cvt * 2);
            for (i = cvt->len_cvt; i; --i) {
                dst -= 4;
                src -= 2;
                lf = src[0];
                rf = src[1];
                ce = (lf / 2) + (rf / 2);
                dst[0] = lf;
                dst[1] = rf;
                dst[2] = lf - ce;
                dst[3] = rf - ce;
            }
        }
        break;

    case AUDIO_S8:
        {
            Sint8 *src, *dst, lf, rf, ce;

            src = (Sint8 *) cvt->buf + cvt->len_cvt;
            dst = (Sint8 *) cvt->buf + cvt->len_cvt * 2;
            for (i = cvt->len_cvt; i; --i) {
                dst -= 4;
                src -= 2;
                lf = src[0];
                rf = src[1];
                ce = (lf / 2) + (rf / 2);
                dst[0] = lf;
                dst[1] = rf;
                dst[2] = lf - ce;
                dst[3] = rf - ce;
            }
        }
        break;

    case AUDIO_U16:
        {
            Uint8 *src, *dst;
            Uint16 lf, rf, ce, lr, rr;

            src = cvt->buf + cvt->len_cvt;
            dst = cvt->buf + cvt->len_cvt * 2;

            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 4; i; --i) {
                    dst -= 8;
                    src -= 4;
                    lf = (Uint16) ((src[0] << 8) | src[1]);
                    rf = (Uint16) ((src[2] << 8) | src[3]);
                    ce = (lf / 2) + (rf / 2);
                    rr = lf - ce;
                    lr = rf - ce;
                    dst[1] = (lf & 0xFF);
                    dst[0] = ((lf >> 8) & 0xFF);
                    dst[3] = (rf & 0xFF);
                    dst[2] = ((rf >> 8) & 0xFF);

                    dst[1 + 4] = (lr & 0xFF);
                    dst[0 + 4] = ((lr >> 8) & 0xFF);
                    dst[3 + 4] = (rr & 0xFF);
                    dst[2 + 4] = ((rr >> 8) & 0xFF);
                }
            } else {
                for (i = cvt->len_cvt / 4; i; --i) {
                    dst -= 8;
                    src -= 4;
                    lf = (Uint16) ((src[1] << 8) | src[0]);
                    rf = (Uint16) ((src[3] << 8) | src[2]);
                    ce = (lf / 2) + (rf / 2);
                    rr = lf - ce;
                    lr = rf - ce;
                    dst[0] = (lf & 0xFF);
                    dst[1] = ((lf >> 8) & 0xFF);
                    dst[2] = (rf & 0xFF);
                    dst[3] = ((rf >> 8) & 0xFF);

                    dst[0 + 4] = (lr & 0xFF);
                    dst[1 + 4] = ((lr >> 8) & 0xFF);
                    dst[2 + 4] = (rr & 0xFF);
                    dst[3 + 4] = ((rr >> 8) & 0xFF);
                }
            }
        }
        break;

    case AUDIO_S16:
        {
            Uint8 *src, *dst;
            Sint16 lf, rf, ce, lr, rr;

            src = cvt->buf + cvt->len_cvt;
            dst = cvt->buf + cvt->len_cvt * 2;

            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 4; i; --i) {
                    dst -= 8;
                    src -= 4;
                    lf = (Sint16) ((src[0] << 8) | src[1]);
                    rf = (Sint16) ((src[2] << 8) | src[3]);
                    ce = (lf / 2) + (rf / 2);
                    rr = lf - ce;
                    lr = rf - ce;
                    dst[1] = (lf & 0xFF);
                    dst[0] = ((lf >> 8) & 0xFF);
                    dst[3] = (rf & 0xFF);
                    dst[2] = ((rf >> 8) & 0xFF);

                    dst[1 + 4] = (lr & 0xFF);
                    dst[0 + 4] = ((lr >> 8) & 0xFF);
                    dst[3 + 4] = (rr & 0xFF);
                    dst[2 + 4] = ((rr >> 8) & 0xFF);
                }
            } else {
                for (i = cvt->len_cvt / 4; i; --i) {
                    dst -= 8;
                    src -= 4;
                    lf = (Sint16) ((src[1] << 8) | src[0]);
                    rf = (Sint16) ((src[3] << 8) | src[2]);
                    ce = (lf / 2) + (rf / 2);
                    rr = lf - ce;
                    lr = rf - ce;
                    dst[0] = (lf & 0xFF);
                    dst[1] = ((lf >> 8) & 0xFF);
                    dst[2] = (rf & 0xFF);
                    dst[3] = ((rf >> 8) & 0xFF);

                    dst[0 + 4] = (lr & 0xFF);
                    dst[1 + 4] = ((lr >> 8) & 0xFF);
                    dst[2 + 4] = (rr & 0xFF);
                    dst[3 + 4] = ((rr >> 8) & 0xFF);
                }
            }
        }
        break;

    case AUDIO_S32:
        {
            const Uint32 *src = (const Uint32 *) (cvt->buf + cvt->len_cvt);
            Uint32 *dst = (Uint32 *) (cvt->buf + cvt->len_cvt * 2);
            Sint32 lf, rf, ce;

            if (SDL_AUDIO_ISBIGENDIAN(format)) {
                for (i = cvt->len_cvt / 8; i; --i) {
                    dst -= 4;
                    src -= 2;
                    lf = (Sint32) SDL_SwapBE32(src[0]);
                    rf = (Sint32) SDL_SwapBE32(src[1]);
                    ce = (lf / 2) + (rf / 2);
                    dst[0] = src[0];
                    dst[1] = src[1];
                    dst[2] = SDL_SwapBE32((Uint32) (lf - ce));
                    dst[3] = SDL_SwapBE32((Uint32) (rf - ce));
                }
            } else {
                for (i = cvt->len_cvt / 8; i; --i) {
                    dst -= 4;
                    src -= 2;
                    lf = (Sint32) SDL_SwapLE32(src[0]);
                    rf = (Sint32) SDL_SwapLE32(src[1]);
                    ce = (lf / 2) + (rf / 2);
                    dst[0] = src[0];
                    dst[1] = src[1];
                    dst[2] = SDL_SwapLE32((Uint32) (lf - ce));
                    dst[3] = SDL_SwapLE32((Uint32) (rf - ce));
                }
            }
        }
        break;
    }
    cvt->len_cvt *= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

/* Convert rate up by multiple of 2 */
static void SDLCALL
SDL_RateMUL2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate * 2 (mono)\n");
#endif

#define mul2_mono(type) { \
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
        for (i = cvt->len_cvt / sizeof (type); i; --i) { \
            src--; \
            dst[-1] = dst[-2] = src[0]; \
            dst -= 2; \
        } \
    }

    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        mul2_mono(Uint8);
        break;
    case 16:
        mul2_mono(Uint16);
        break;
    case 32:
        mul2_mono(Uint32);
        break;
    }

#undef mul2_mono

    cvt->len_cvt *= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Convert rate up by multiple of 2, for stereo */
static void SDLCALL
SDL_RateMUL2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate * 2 (stereo)\n");
#endif

#define mul2_stereo(type) { \
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
        for (i = cvt->len_cvt / (sizeof (type) * 2); i; --i) { \
            const type r = src[-1]; \
            const type l = src[-2]; \
            src -= 2; \
            dst[-1] = r; \
            dst[-2] = l; \
            dst[-3] = r; \
            dst[-4] = l; \
            dst -= 4; \
        } \
    }

    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        mul2_stereo(Uint8);
        break;
    case 16:
        mul2_stereo(Uint16);
        break;
    case 32:
        mul2_stereo(Uint32);
        break;
    }

#undef mul2_stereo

    cvt->len_cvt *= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

/* Convert rate up by multiple of 2, for quad */
static void SDLCALL
SDL_RateMUL2_c4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate * 2 (quad)\n");
#endif

#define mul2_quad(type) { \
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
        for (i = cvt->len_cvt / (sizeof (type) * 4); i; --i) { \
            const type c1 = src[-1]; \
            const type c2 = src[-2]; \
            const type c3 = src[-3]; \
            const type c4 = src[-4]; \
            src -= 4; \
            dst[-1] = c1; \
            dst[-2] = c2; \
            dst[-3] = c3; \
            dst[-4] = c4; \
            dst[-5] = c1; \
            dst[-6] = c2; \
            dst[-7] = c3; \
            dst[-8] = c4; \
            dst -= 8; \
        } \
    }

    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        mul2_quad(Uint8);
        break;
    case 16:
        mul2_quad(Uint16);
        break;
    case 32:
        mul2_quad(Uint32);
        break;
    }

#undef mul2_quad

    cvt->len_cvt *= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Convert rate up by multiple of 2, for 5.1 */
static void SDLCALL
SDL_RateMUL2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate * 2 (six channels)\n");
#endif

#define mul2_chansix(type) { \
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
        for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
            const type c1 = src[-1]; \
            const type c2 = src[-2]; \
            const type c3 = src[-3]; \
            const type c4 = src[-4]; \
            const type c5 = src[-5]; \
            const type c6 = src[-6]; \
            src -= 6; \
            dst[-1] = c1; \
            dst[-2] = c2; \
            dst[-3] = c3; \
            dst[-4] = c4; \
            dst[-5] = c5; \
            dst[-6] = c6; \
            dst[-7] = c1; \
            dst[-8] = c2; \
            dst[-9] = c3; \
            dst[-10] = c4; \
            dst[-11] = c5; \
            dst[-12] = c6; \
            dst -= 12; \
        } \
    }

    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        mul2_chansix(Uint8);
        break;
    case 16:
        mul2_chansix(Uint16);
        break;
    case 32:
        mul2_chansix(Uint32);
        break;
    }

#undef mul2_chansix

    cvt->len_cvt *= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

/* Convert rate down by multiple of 2 */
static void SDLCALL
SDL_RateDIV2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate / 2 (mono)\n");
#endif

#define div2_mono(type) { \
        const type *src = (const type *) cvt->buf; \
        type *dst = (type *) cvt->buf; \
        for (i = cvt->len_cvt / (sizeof (type) * 2); i; --i) { \
            dst[0] = src[0]; \
            src += 2; \
            dst++; \
        } \
    }

    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        div2_mono(Uint8);
        break;
    case 16:
        div2_mono(Uint16);
        break;
    case 32:
        div2_mono(Uint32);
        break;
    }

#undef div2_mono

    cvt->len_cvt /= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Convert rate down by multiple of 2, for stereo */
static void SDLCALL
SDL_RateDIV2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate / 2 (stereo)\n");
#endif

#define div2_stereo(type) { \
        const type *src = (const type *) cvt->buf; \
        type *dst = (type *) cvt->buf; \
        for (i = cvt->len_cvt / (sizeof (type) * 4); i; --i) { \
            dst[0] = src[0]; \
            dst[1] = src[1]; \
            src += 4; \
            dst += 2; \
        } \
    }

    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        div2_stereo(Uint8);
        break;
    case 16:
        div2_stereo(Uint16);
        break;
    case 32:
        div2_stereo(Uint32);
        break;
    }

#undef div2_stereo

    cvt->len_cvt /= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}


/* Convert rate down by multiple of 2, for quad */
static void SDLCALL
SDL_RateDIV2_c4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate / 2 (quad)\n");
#endif

#define div2_quad(type) { \
        const type *src = (const type *) cvt->buf; \
        type *dst = (type *) cvt->buf; \
        for (i = cvt->len_cvt / (sizeof (type) * 8); i; --i) { \
            dst[0] = src[0]; \
            dst[1] = src[1]; \
            dst[2] = src[2]; \
            dst[3] = src[3]; \
            src += 8; \
            dst += 4; \
        } \
    }

    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        div2_quad(Uint8);
        break;
    case 16:
        div2_quad(Uint16);
        break;
    case 32:
        div2_quad(Uint32);
        break;
    }

#undef div2_quad

    cvt->len_cvt /= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

/* Convert rate down by multiple of 2, for 5.1 */
static void SDLCALL
SDL_RateDIV2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate / 2 (six channels)\n");
#endif

#define div2_chansix(type) { \
        const type *src = (const type *) cvt->buf; \
        type *dst = (type *) cvt->buf; \
        for (i = cvt->len_cvt / (sizeof (type) * 12); i; --i) { \
            dst[0] = src[0]; \
            dst[1] = src[1]; \
            dst[2] = src[2]; \
            dst[3] = src[3]; \
            dst[4] = src[4]; \
            dst[5] = src[5]; \
            src += 12; \
            dst += 6; \
        } \
    }

    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        div2_chansix(Uint8);
        break;
    case 16:
        div2_chansix(Uint16);
        break;
    case 32:
        div2_chansix(Uint32);
        break;
    }

#undef div_chansix

    cvt->len_cvt /= 2;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

/* Very slow rate conversion routine */
static void SDLCALL
SDL_RateSLOW(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    double ipos;
    int i, clen;

#ifdef DEBUG_CONVERT
    fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0 / cvt->rate_incr);
#endif
    clen = (int) ((double) cvt->len_cvt / cvt->rate_incr);
    if (cvt->rate_incr > 1.0) {
        switch (SDL_AUDIO_BITSIZE(format)) {
        case 8:
            {
                Uint8 *output;

                output = cvt->buf;
                ipos = 0.0;
                for (i = clen; i; --i) {
                    *output = cvt->buf[(int) ipos];
                    ipos += cvt->rate_incr;
                    output += 1;
                }
            }
            break;

        case 16:
            {
                Uint16 *output;

                clen &= ~1;
                output = (Uint16 *) cvt->buf;
                ipos = 0.0;
                for (i = clen / 2; i; --i) {
                    *output = ((Uint16 *) cvt->buf)[(int) ipos];
                    ipos += cvt->rate_incr;
                    output += 1;
                }
            }
            break;

        case 32:
            {
                /* !!! FIXME: need 32-bit converter here! */
#ifdef DEBUG_CONVERT
                fprintf(stderr, "FIXME: need 32-bit converter here!\n");
#endif
            }
        }
    } else {
        switch (SDL_AUDIO_BITSIZE(format)) {
        case 8:
            {
                Uint8 *output;

                output = cvt->buf + clen;
                ipos = (double) cvt->len_cvt;
                for (i = clen; i; --i) {
                    ipos -= cvt->rate_incr;
                    output -= 1;
                    *output = cvt->buf[(int) ipos];
                }
            }
            break;

        case 16:
            {
                Uint16 *output;

                clen &= ~1;
                output = (Uint16 *) (cvt->buf + clen);
                ipos = (double) cvt->len_cvt / 2;
                for (i = clen / 2; i; --i) {
                    ipos -= cvt->rate_incr;
                    output -= 1;
                    *output = ((Uint16 *) cvt->buf)[(int) ipos];
                }
            }
            break;

        case 32:
            {
                /* !!! FIXME: need 32-bit converter here! */
#ifdef DEBUG_CONVERT
                fprintf(stderr, "FIXME: need 32-bit converter here!\n");
#endif
            }
        }
    }

    cvt->len_cvt = clen;
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

/* Perform proper resampling */
static void SDLCALL
SDL_Resample(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
    int i, j;

#ifdef DEBUG_CONVERT
    printf("Converting audio rate via proper resampling (mono)\n");
#endif

#define zerostuff_mono(type) { \
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * cvt->len_mult)); \
        for (i = cvt->len_cvt / sizeof (type); i; --i) { \
            src--; \
            dst[-1] = src[0]; \
			for( j = -cvt->len_mult; j < -1; ++j ) { \
				dst[j] = 0; \
			} \
            dst -= cvt->len_mult; \
        } \
    }
	
#define discard_mono(type) { \
        const type *src = (const type *) (cvt->buf); \
        type *dst = (type *) (cvt->buf); \
        for (i = 0; i < cvt->len_cvt / cvt->len_div / sizeof (type); ++i) { \
            dst[0] = src[0]; \
            src += cvt->len_div; \
            ++dst; \
        } \
    }

	// Step 1: Zero stuff the conversion buffer
#ifdef DEBUG_CONVERT
	printf("Zero-stuffing by a factor of %u\n", cvt->len_mult);
#endif
    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        zerostuff_mono(Uint8);
        break;
    case 16:
        zerostuff_mono(Uint16);
        break;
    case 32:
        zerostuff_mono(Uint32);
        break;
    }
	
	cvt->len_cvt *= cvt->len_mult;

	// Step 2: Use either a windowed sinc FIR filter or IIR lowpass filter to remove all alias frequencies
	
	// Step 3: Discard unnecessary samples
#ifdef DEBUG_CONVERT
	printf("Discarding samples by a factor of %u\n", cvt->len_div);
#endif
    switch (SDL_AUDIO_BITSIZE(format)) {
    case 8:
        discard_mono(Uint8);
        break;
    case 16:
        discard_mono(Uint16);
        break;
    case 32:
        discard_mono(Uint32);
        break;
    }
	
#undef zerostuff_mono
#undef discard_mono

    cvt->len_cvt /= cvt->len_div;
	
    if (cvt->filters[++cvt->filter_index]) {
        cvt->filters[cvt->filter_index] (cvt, format);
    }
}

int
SDL_ConvertAudio(SDL_AudioCVT * cvt)
{
    /* Make sure there's data to convert */
    if (cvt->buf == NULL) {
        SDL_SetError("No buffer allocated for conversion");
        return (-1);
    }
    /* Return okay if no conversion is necessary */
    cvt->len_cvt = cvt->len;
    if (cvt->filters[0] == NULL) {
        return (0);
    }

    /* Set up the conversion and go! */
    cvt->filter_index = 0;
    cvt->filters[0] (cvt, cvt->src_format);
    return (0);
}


static SDL_AudioFilter
SDL_HandTunedTypeCVT(SDL_AudioFormat src_fmt, SDL_AudioFormat dst_fmt)
{
    /*
     * Fill in any future conversions that are specialized to a
     *  processor, platform, compiler, or library here.
     */

    return NULL;                /* no specialized converter code available. */
}


/*
 * Find a converter between two data types. We try to select a hand-tuned
 *  asm/vectorized/optimized function first, and then fallback to an
 *  autogenerated function that is customized to convert between two
 *  specific data types.
 */
static int
SDL_BuildAudioTypeCVT(SDL_AudioCVT * cvt,
                      SDL_AudioFormat src_fmt, SDL_AudioFormat dst_fmt)
{
    if (src_fmt != dst_fmt) {
        const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
        const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
        SDL_AudioFilter filter = SDL_HandTunedTypeCVT(src_fmt, dst_fmt);

        /* No hand-tuned converter? Try the autogenerated ones. */
        if (filter == NULL) {
            int i;
            for (i = 0; sdl_audio_type_filters[i].filter != NULL; i++) {
                const SDL_AudioTypeFilters *filt = &sdl_audio_type_filters[i];
                if ((filt->src_fmt == src_fmt) && (filt->dst_fmt == dst_fmt)) {
                    filter = filt->filter;
                    break;
                }
            }

            if (filter == NULL) {
                return -1;      /* Still no matching converter?! */
            }
        }

        /* Update (cvt) with filter details... */
        cvt->filters[cvt->filter_index++] = filter;
        if (src_bitsize < dst_bitsize) {
            const int mult = (dst_bitsize / src_bitsize);
            cvt->len_mult *= mult;
            cvt->len_ratio *= mult;
        } else if (src_bitsize > dst_bitsize) {
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
        }

        return 1;               /* added a converter. */
    }

    return 0;                   /* no conversion necessary. */
}

/* Generate the necessary IIR lowpass coefficients for resampling.
   Assume that the SDL_AudioCVT struct is already set up with
   the correct values for len_mult and len_div, and use the
   type of dst_format. Also assume the buffer is allocated.
   Note the buffer needs to be 6 units long.
   For now, use RBJ's cookbook coefficients. It might be more
   optimal to create a Butterworth filter, but this is more difficult.
*/
int SDL_BuildIIRLowpass(SDL_AudioCVT * cvt, SDL_AudioFormat format) {
	float fc;			/* cutoff frequency */
	float coeff[6];		/* floating point iir coefficients b0, b1, b2, a0, a1, a2 */
	float scale;
	float w0, alpha, cosw0;
	
	/* The higher Q is, the higher CUTOFF can be. Need to find a good balance to avoid aliasing */
	static const float Q = 5.0f;
	static const float CUTOFF = 0.4f;
	
	fc = (cvt->len_mult > cvt->len_div) ? CUTOFF / (float)cvt->len_mult : CUTOFF / (float)cvt->len_div;
	
	w0 = 2.0f * M_PI * fc;
	cosw0 = cosf(w0);
	alpha = sin(w0) / (2.0f * Q);
	
	/* Compute coefficients, normalizing by a0 */
	scale = 1.0f / (1.0f + alpha);
	
	coeff[0] = (1.0f - cosw0) / 2.0f * scale;
	coeff[1] = (1.0f - cosw0) * scale;
	coeff[2] = coeff[0];
	
	coeff[3] = 1.0f;	/* a0 is normalized to 1 */
	coeff[4] = -2.0f * cosw0 * scale;
	coeff[5] = (1.0f - alpha) * scale;
	
	/* Copy the coefficients to the struct. If necessary, convert coefficients to fixed point, using the range (-2.0, 2.0) */
	if(SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
		float *cvt_coeff = (float *)cvt->coeff;
		int i;
		for(i = 0; i < 6; ++i) {
			cvt_coeff[i] = coeff[i];
		}
	} else {
	}
	
	/* Initialize the state buffer to all zeroes, and set initial position */
	memset(cvt->state_buf, 0, 4 * SDL_AUDIO_BITSIZE(format) / 4);
	cvt->state_pos = 0;
}

/* Apply the lowpass IIR filter to the given SDL_AudioCVT struct */
int SDL_FilterIIR(SDL_AudioCVT * cvt, SDL_AudioFormat format) {
	int i, n;
	
	n = cvt->len_cvt / (SDL_AUDIO_BITSIZE(format) / 4);
	
	if(SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
		float *coeff = (float *)cvt->coeff;
		float *state = (float *)cvt->state_buf;
		float *buf = (float *)cvt->buf;
		float temp;

		
		for(i = 0; i < n; ++i) {
			/* y[n] = b0 * x[n] + b1 * x[n-1] + b2 * x[n-2] - a1 * y[n-1] - a[2] * y[n-2] */
			temp = buf[n];
			if( cvt->state_pos ) {
				buf[n] = coeff[0] * buf[n] + coeff[1] * state[0] + coeff[2] * state[1] - coeff[4] * state[2] - coeff[5] * state[3];
				state[1] = temp;
				state[3] = buf[n];
				cvt->state_pos = 0;
			} else {
				buf[n] = coeff[0] * buf[n] + coeff[1] * state[1] + coeff[2] * state[0] - coeff[4] * state[3] - coeff[5] * state[2];
				state[0] = temp;
				state[2] = buf[n];
				cvt->state_pos = 1;
			}
		}
	} else {
	}
}

/* Apply the windowed sinc FIR filter to the given SDL_AudioCVT struct */
int SDL_FilterFIR(SDL_AudioCVT * cvt, SDL_AudioFormat format) {
	int n = cvt->len_cvt / (SDL_AUDIO_BITSIZE(format) / 4);
	int m = cvt->len_sinc;
	int i, j;
	
	/* Note: this makes use of the symmetry of the sinc filter.
	   We can also make a big optimization here by taking advantage
	   of the fact that the signal is zero stuffed, so we can do
	   significantly fewer multiplications and additions.
	*/
#define filter_sinc(type, shift_bits) { \
			type *sinc = (type *)cvt->coeff; \
			type *state = (type *)cvt->state_buf; \
			type *buf = (type *)cvt->buf; \
			for(i = 0; i < n; ++i) { \
				state[cvt->state_pos++] = buf[i] >> shift_bits; \
				if(cvt->state_pos == m) cvt->state_pos = 0; \
				buf[i] = 0; \
				for(j = 0; j < m;  ++j) { \
					buf[i] += state[j] * sinc[j]; \
				} \
			} \
		}
	
	/* If it's floating point, we don't need to do any shifting */
	if(SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
		float *sinc = (float *)cvt->coeff;
		float *state = (float *)cvt->state_buf;
		float *buf = (float *)cvt->buf;
		
		for(i = 0; i < n; ++i) {
			state[cvt->state_pos++] = buf[i];
			if(cvt->state_pos == m) cvt->state_pos = 0;
			buf[i] = 0.0f;
			for(j = 0; j < m; ++j) {
				buf[i] += state[j] * sinc[j];
			}
		}
	} else {
		switch (SDL_AUDIO_BITSIZE(format)) {
			case 8:
				filter_sinc(Uint8, 4);
				break;
			case 16:
				filter_sinc(Uint16, 8);
				break;
			case 32:
				filter_sinc(Uint32, 16);
				break;
		}
	}
	
#undef filter_sinc
			
}

/* Generate the necessary windowed sinc filter for resampling.
   Assume that the SDL_AudioCVT struct is already set up with
   the correct values for len_mult and len_div, and use the
   type of dst_format. Also assume the buffer is allocated.
   Note the buffer needs to be m+1 units long.
*/
int
SDL_BuildWindowedSinc(SDL_AudioCVT * cvt, SDL_AudioFormat format, unsigned int m) {
	float fScale;		/* scale factor for fixed point */
	float *fSinc;		/* floating point sinc buffer, to be converted to fixed point */
	float fc;			/* cutoff frequency */
	float two_pi_fc, two_pi_over_m, four_pi_over_m, m_over_two;
	float norm_sum, norm_fact;
	unsigned int i;

	/* Check that the buffer is allocated */
	if( cvt->coeff == NULL ) {
		return -1;
	}

	/* Set the length */
	cvt->len_sinc = m;
	
	/* Allocate the floating point windowed sinc. */
	fSinc = (float *)malloc(m * sizeof(float));
	if( fSinc == NULL ) {
		return -1;
	}
	
	/* Set up the filter parameters */
	fc = (cvt->len_mult > cvt->len_div) ? 0.5f / (float)cvt->len_mult : 0.5f / (float)cvt->len_div;
	two_pi_fc = 2.0f * M_PI * fc;
	two_pi_over_m = 2.0f * M_PI / (float)m;
	four_pi_over_m = 2.0f * two_pi_over_m;
	m_over_two = (float)m / 2.0f;
	norm_sum = 0.0f;
	
	for(i = 0; i <= m; ++i ) {
		if( i == m/2 ) {
			fSinc[i] = two_pi_fc;
		} else {
			fSinc[i] = sinf(two_pi_fc * ((float)i - m_over_two)) / ((float)i - m_over_two);
			/* Apply blackman window */
			fSinc[i] *= 0.42f - 0.5f * cosf(two_pi_over_m * (float)i) + 0.08f * cosf(four_pi_over_m * (float)i);
		}
		norm_sum += abs(fSinc[i]);
	}

	/* Now normalize and convert to fixed point. We scale everything to half the precision
	   of whatever datatype we're using, for example, 16 bit data means we use 8 bits */
	
#define convert_fixed(type, size) { \
		norm_fact = size / norm_sum; \
		type *dst = (type *)cvt->coeff; \
		for( i = 0; i <= m; ++i ) { \
			dst[i] = (type)(fSinc[i] * norm_fact); \
		} \
	}
	
	/* If we're using floating point, we only need to normalize */
	if(SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
		float *fDest = (float *)cvt->coeff;
		norm_fact = 1.0f / norm_sum;
		for(i = 0; i <= m; ++i) {
			fDest[i] = fSinc[i] * norm_fact;
		}
	} else {
		switch (SDL_AUDIO_BITSIZE(format)) {
			case 8:
				convert_fixed(Uint8, 0x0e);
				break;
			case 16:
				convert_fixed(Uint16, 0xfe);
				break;
			case 32:
				convert_fixed(Uint32, 0xfffe);
				break;
		}
	}
	
	/* Initialize the state buffer to all zeroes, and set initial position */
	memset(cvt->state_buf, 0, cvt->len_sinc * SDL_AUDIO_BITSIZE(format) / 4);
	cvt->state_pos = 0;
	
	/* Clean up */
#undef convert_fixed
	free(fSinc);
}

/* This is used to reduce the resampling ratio */
inline int SDL_GCD(int a, int b) {
	int temp;
	while(b != 0) {
		temp = a % b;
		a = b;
		b = temp;
	}
	return a;
}


/* Creates a set of audio filters to convert from one format to another.
   Returns -1 if the format conversion is not supported, 0 if there's
   no conversion needed, or 1 if the audio filter is set up.
*/

int
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
                  SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
                  SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
{
    /* there are no unsigned types over 16 bits, so catch this upfront. */
    if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
        return -1;
    }
    if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
        return -1;
    }
#ifdef DEBUG_CONVERT
    printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
           src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
#endif

    /* Start off with no conversion necessary */

    cvt->src_format = src_fmt;
    cvt->dst_format = dst_fmt;
    cvt->needed = 0;
    cvt->filter_index = 0;
    cvt->filters[0] = NULL;
    cvt->len_mult = 1;
    cvt->len_ratio = 1.0;

    /* Convert data types, if necessary. Updates (cvt). */
    if (SDL_BuildAudioTypeCVT(cvt, src_fmt, dst_fmt) == -1)
        return -1;              /* shouldn't happen, but just in case... */

    /* Channel conversion */
    if (src_channels != dst_channels) {
        if ((src_channels == 1) && (dst_channels > 1)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
            cvt->len_mult *= 2;
            src_channels = 2;
            cvt->len_ratio *= 2;
        }
        if ((src_channels == 2) && (dst_channels == 6)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertSurround;
            src_channels = 6;
            cvt->len_mult *= 3;
            cvt->len_ratio *= 3;
        }
        if ((src_channels == 2) && (dst_channels == 4)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertSurround_4;
            src_channels = 4;
            cvt->len_mult *= 2;
            cvt->len_ratio *= 2;
        }
        while ((src_channels * 2) <= dst_channels) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
            cvt->len_mult *= 2;
            src_channels *= 2;
            cvt->len_ratio *= 2;
        }
        if ((src_channels == 6) && (dst_channels <= 2)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertStrip;
            src_channels = 2;
            cvt->len_ratio /= 3;
        }
        if ((src_channels == 6) && (dst_channels == 4)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertStrip_2;
            src_channels = 4;
            cvt->len_ratio /= 2;
        }
        /* This assumes that 4 channel audio is in the format:
           Left {front/back} + Right {front/back}
           so converting to L/R stereo works properly.
         */
        while (((src_channels % 2) == 0) &&
               ((src_channels / 2) >= dst_channels)) {
            cvt->filters[cvt->filter_index++] = SDL_ConvertMono;
            src_channels /= 2;
            cvt->len_ratio /= 2;
        }
        if (src_channels != dst_channels) {
            /* Uh oh.. */ ;
        }
    }

    /* Do rate conversion */
	int rate_gcd;
	rate_gcd = SDL_GCD(src_rate, dst_rate);
	cvt->len_mult = dst_rate / rate_gcd;
	cvt->len_div = src_rate / rate_gcd;
	cvt->len_ratio = (double)cvt->len_mult / (double)cvt->len_div;
	cvt->filters[cvt->filter_index++] = SDL_Resample;
	
    /*cvt->rate_incr = 0.0;
    if ((src_rate / 100) != (dst_rate / 100)) {
        Uint32 hi_rate, lo_rate;
        int len_mult;
        double len_ratio;
        SDL_AudioFilter rate_cvt = NULL;

        if (src_rate > dst_rate) {
            hi_rate = src_rate;
            lo_rate = dst_rate;
            switch (src_channels) {
            case 1:
                rate_cvt = SDL_RateDIV2;
                break;
            case 2:
                rate_cvt = SDL_RateDIV2_c2;
                break;
            case 4:
                rate_cvt = SDL_RateDIV2_c4;
                break;
            case 6:
                rate_cvt = SDL_RateDIV2_c6;
                break;
            default:
                return -1;
            }
            len_mult = 1;
            len_ratio = 0.5;
        } else {
            hi_rate = dst_rate;
            lo_rate = src_rate;
            switch (src_channels) {
            case 1:
                rate_cvt = SDL_RateMUL2;
                break;
            case 2:
                rate_cvt = SDL_RateMUL2_c2;
                break;
            case 4:
                rate_cvt = SDL_RateMUL2_c4;
                break;
            case 6:
                rate_cvt = SDL_RateMUL2_c6;
                break;
            default:
                return -1;
            }
            len_mult = 2;
            len_ratio = 2.0;
        }*/
        /* If hi_rate = lo_rate*2^x then conversion is easy */
        /*while (((lo_rate * 2) / 100) <= (hi_rate / 100)) {
            cvt->filters[cvt->filter_index++] = rate_cvt;
            cvt->len_mult *= len_mult;
            lo_rate *= 2;
            cvt->len_ratio *= len_ratio;
        }*/
        /* We may need a slow conversion here to finish up */
        /*if ((lo_rate / 100) != (hi_rate / 100)) {*/
#if 1
            /* The problem with this is that if the input buffer is
               say 1K, and the conversion rate is say 1.1, then the
               output buffer is 1.1K, which may not be an acceptable
               buffer size for the audio driver (not a power of 2)
             */
            /* For now, punt and hope the rate distortion isn't great.
             */
#else
            if (src_rate < dst_rate) {
                cvt->rate_incr = (double) lo_rate / hi_rate;
                cvt->len_mult *= 2;
                cvt->len_ratio /= cvt->rate_incr;
            } else {
                cvt->rate_incr = (double) hi_rate / lo_rate;
                cvt->len_ratio *= cvt->rate_incr;
            }
            cvt->filters[cvt->filter_index++] = SDL_RateSLOW;
#endif
/*        }
    }*/

    /* Set up the filter information */
    if (cvt->filter_index != 0) {
        cvt->needed = 1;
        cvt->src_format = src_fmt;
        cvt->dst_format = dst_fmt;
        cvt->len = 0;
        cvt->buf = NULL;
        cvt->filters[cvt->filter_index] = NULL;
    }
    return (cvt->needed);
}

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