src/audio/SDL_audiocvt.c
author Sam Lantinga <slouken@libsdl.org>
Mon, 08 Dec 2008 00:27:32 +0000
changeset 2859 99210400e8b9
parent 2768 26861c61142a
child 2878 10c319ce07fb
permissions -rw-r--r--
Updated copyright date
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/*
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    SDL - Simple DirectMedia Layer
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    Copyright (C) 1997-2009 Sam Lantinga
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    This library is free software; you can redistribute it and/or
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    modify it under the terms of the GNU Lesser General Public
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    License as published by the Free Software Foundation; either
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    version 2.1 of the License, or (at your option) any later version.
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    This library is distributed in the hope that it will be useful,
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    but WITHOUT ANY WARRANTY; without even the implied warranty of
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    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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    Lesser General Public License for more details.
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    You should have received a copy of the GNU Lesser General Public
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    License along with this library; if not, write to the Free Software
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    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
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    Sam Lantinga
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    slouken@libsdl.org
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*/
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#include "SDL_config.h"
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/* Functions for audio drivers to perform runtime conversion of audio format */
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#include "SDL_audio.h"
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#include "SDL_audio_c.h"
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#include "../libm/math.h"
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//#define DEBUG_CONVERT
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/* These are fractional multiplication routines. That is, their inputs
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   are two numbers in the range [-1, 1) and the result falls in that
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   same range. The output is the same size as the inputs, i.e.
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   32-bit x 32-bit = 32-bit.
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 */
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/* We hope here that the right shift includes sign extension */
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#ifdef SDL_HAS_64BIT_Type
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#define SDL_FixMpy32(a, b) ((((Sint64)a * (Sint64)b) >> 31) & 0xffffffff)
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#else
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/* If we don't have the 64-bit type, do something more complicated. See http://www.8052.com/mul16.phtml or http://www.cs.uaf.edu/~cs301/notes/Chapter5/node5.html */
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#define SDL_FixMpy32(a, b) ((((Sint64)a * (Sint64)b) >> 31) & 0xffffffff)
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#endif
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#define SDL_FixMpy16(a, b) ((((Sint32)a * (Sint32)b) >> 14) & 0xffff)
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#define SDL_FixMpy8(a, b) ((((Sint16)a * (Sint16)b) >> 7) & 0xff)
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/* This macro just makes the floating point filtering code not have to be a special case. */
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#define SDL_FloatMpy(a, b) (a * b)
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/* These macros take floating point numbers in the range [-1.0f, 1.0f) and
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   represent them as fixed-point numbers in that same range. There's no
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   checking that the floating point argument is inside the appropriate range.
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 */
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#define SDL_Make_1_7(a) (Sint8)(a * 128.0f)
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#define SDL_Make_1_15(a) (Sint16)(a * 32768.0f)
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#define SDL_Make_1_31(a) (Sint32)(a * 2147483648.0f)
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#define SDL_Make_2_6(a) (Sint8)(a * 64.0f)
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#define SDL_Make_2_14(a) (Sint16)(a * 16384.0f)
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#define SDL_Make_2_30(a) (Sint32)(a * 1073741824.0f)
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/* Effectively mix right and left channels into a single channel */
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static void SDLCALL
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SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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    int i;
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    Sint32 sample;
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#ifdef DEBUG_CONVERT
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    fprintf(stderr, "Converting to mono\n");
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#endif
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    switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
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    case AUDIO_U8:
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        {
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            Uint8 *src, *dst;
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            src = cvt->buf;
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            dst = cvt->buf;
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            for (i = cvt->len_cvt / 2; i; --i) {
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                sample = src[0] + src[1];
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                *dst = (Uint8) (sample / 2);
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                src += 2;
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                dst += 1;
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            }
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        }
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        break;
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    case AUDIO_S8:
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        {
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            Sint8 *src, *dst;
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            src = (Sint8 *) cvt->buf;
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            dst = (Sint8 *) cvt->buf;
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            for (i = cvt->len_cvt / 2; i; --i) {
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                sample = src[0] + src[1];
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                *dst = (Sint8) (sample / 2);
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                src += 2;
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                dst += 1;
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            }
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        }
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        break;
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    case AUDIO_U16:
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        {
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            Uint8 *src, *dst;
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            src = cvt->buf;
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            dst = cvt->buf;
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            if (SDL_AUDIO_ISBIGENDIAN(format)) {
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                for (i = cvt->len_cvt / 4; i; --i) {
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                    sample = (Uint16) ((src[0] << 8) | src[1]) +
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                        (Uint16) ((src[2] << 8) | src[3]);
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                    sample /= 2;
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                    dst[1] = (sample & 0xFF);
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                    sample >>= 8;
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                    dst[0] = (sample & 0xFF);
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                    src += 4;
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                    dst += 2;
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                }
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            } else {
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                for (i = cvt->len_cvt / 4; i; --i) {
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                    sample = (Uint16) ((src[1] << 8) | src[0]) +
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                        (Uint16) ((src[3] << 8) | src[2]);
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                    sample /= 2;
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                    dst[0] = (sample & 0xFF);
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                    sample >>= 8;
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                    dst[1] = (sample & 0xFF);
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                    src += 4;
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                    dst += 2;
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                }
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            }
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        }
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        break;
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    case AUDIO_S16:
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        {
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            Uint8 *src, *dst;
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            src = cvt->buf;
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            dst = cvt->buf;
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            if (SDL_AUDIO_ISBIGENDIAN(format)) {
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                for (i = cvt->len_cvt / 4; i; --i) {
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                    sample = (Sint16) ((src[0] << 8) | src[1]) +
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                        (Sint16) ((src[2] << 8) | src[3]);
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                    sample /= 2;
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                    dst[1] = (sample & 0xFF);
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                    sample >>= 8;
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                    dst[0] = (sample & 0xFF);
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                    src += 4;
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                    dst += 2;
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                }
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            } else {
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                for (i = cvt->len_cvt / 4; i; --i) {
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                    sample = (Sint16) ((src[1] << 8) | src[0]) +
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                        (Sint16) ((src[3] << 8) | src[2]);
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                    sample /= 2;
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                    dst[0] = (sample & 0xFF);
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                    sample >>= 8;
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                    dst[1] = (sample & 0xFF);
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                    src += 4;
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                    dst += 2;
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                }
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            }
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        }
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        break;
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    case AUDIO_S32:
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        {
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            const Uint32 *src = (const Uint32 *) cvt->buf;
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            Uint32 *dst = (Uint32 *) cvt->buf;
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            if (SDL_AUDIO_ISBIGENDIAN(format)) {
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                for (i = cvt->len_cvt / 8; i; --i, src += 2) {
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                    const Sint64 added =
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                        (((Sint64) (Sint32) SDL_SwapBE32(src[0])) +
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                         ((Sint64) (Sint32) SDL_SwapBE32(src[1])));
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                    *(dst++) = SDL_SwapBE32((Uint32) ((Sint32) (added / 2)));
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                }
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            } else {
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                for (i = cvt->len_cvt / 8; i; --i, src += 2) {
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                    const Sint64 added =
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                        (((Sint64) (Sint32) SDL_SwapLE32(src[0])) +
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                         ((Sint64) (Sint32) SDL_SwapLE32(src[1])));
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                    *(dst++) = SDL_SwapLE32((Uint32) ((Sint32) (added / 2)));
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                }
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            }
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        }
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        break;
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    case AUDIO_F32:
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        {
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            const float *src = (const float *) cvt->buf;
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            float *dst = (float *) cvt->buf;
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            if (SDL_AUDIO_ISBIGENDIAN(format)) {
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                for (i = cvt->len_cvt / 8; i; --i, src += 2) {
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                    const float src1 = SDL_SwapFloatBE(src[0]);
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                    const float src2 = SDL_SwapFloatBE(src[1]);
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                    const double added = ((double) src1) + ((double) src2);
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                    const float halved = (float) (added * 0.5);
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                    *(dst++) = SDL_SwapFloatBE(halved);
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                }
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            } else {
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                for (i = cvt->len_cvt / 8; i; --i, src += 2) {
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                    const float src1 = SDL_SwapFloatLE(src[0]);
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                    const float src2 = SDL_SwapFloatLE(src[1]);
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                    const double added = ((double) src1) + ((double) src2);
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                    const float halved = (float) (added * 0.5);
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                    *(dst++) = SDL_SwapFloatLE(halved);
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                }
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            }
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        }
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        break;
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    }
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    cvt->len_cvt /= 2;
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    if (cvt->filters[++cvt->filter_index]) {
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        cvt->filters[cvt->filter_index] (cvt, format);
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    }
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}
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/* Discard top 4 channels */
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static void SDLCALL
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SDL_ConvertStrip(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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    int i;
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#ifdef DEBUG_CONVERT
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    fprintf(stderr, "Converting down from 6 channels to stereo\n");
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#endif
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#define strip_chans_6_to_2(type) \
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    { \
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        const type *src = (const type *) cvt->buf; \
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        type *dst = (type *) cvt->buf; \
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        for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
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            dst[0] = src[0]; \
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            dst[1] = src[1]; \
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            src += 6; \
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            dst += 2; \
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        } \
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    }
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    /* this function only cares about typesize, and data as a block of bits. */
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    switch (SDL_AUDIO_BITSIZE(format)) {
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    case 8:
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        strip_chans_6_to_2(Uint8);
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        break;
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    case 16:
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        strip_chans_6_to_2(Uint16);
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        break;
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    case 32:
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        strip_chans_6_to_2(Uint32);
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        break;
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    }
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#undef strip_chans_6_to_2
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    cvt->len_cvt /= 3;
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    if (cvt->filters[++cvt->filter_index]) {
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        cvt->filters[cvt->filter_index] (cvt, format);
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    }
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}
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/* Discard top 2 channels of 6 */
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static void SDLCALL
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SDL_ConvertStrip_2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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{
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    int i;
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#ifdef DEBUG_CONVERT
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    fprintf(stderr, "Converting 6 down to quad\n");
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#endif
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#define strip_chans_6_to_4(type) \
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    { \
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        const type *src = (const type *) cvt->buf; \
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        type *dst = (type *) cvt->buf; \
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        for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
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            dst[0] = src[0]; \
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            dst[1] = src[1]; \
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            dst[2] = src[2]; \
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            dst[3] = src[3]; \
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            src += 6; \
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            dst += 4; \
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        } \
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    }
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    /* this function only cares about typesize, and data as a block of bits. */
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    switch (SDL_AUDIO_BITSIZE(format)) {
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    case 8:
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        strip_chans_6_to_4(Uint8);
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        break;
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    case 16:
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        strip_chans_6_to_4(Uint16);
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        break;
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    case 32:
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        strip_chans_6_to_4(Uint32);
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        break;
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   301
    }
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   302
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   303
#undef strip_chans_6_to_4
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   304
icculus@1982
   305
    cvt->len_cvt /= 6;
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   306
    cvt->len_cvt *= 4;
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   307
    if (cvt->filters[++cvt->filter_index]) {
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   308
        cvt->filters[cvt->filter_index] (cvt, format);
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   309
    }
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   310
}
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   311
slouken@0
   312
/* Duplicate a mono channel to both stereo channels */
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   313
static void SDLCALL
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   314
SDL_ConvertStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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   315
{
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   316
    int i;
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   317
slouken@0
   318
#ifdef DEBUG_CONVERT
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   319
    fprintf(stderr, "Converting to stereo\n");
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   320
#endif
slouken@0
   321
slouken@1985
   322
#define dup_chans_1_to_2(type) \
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   323
    { \
icculus@1982
   324
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
icculus@1982
   325
        type *dst = (type *) (cvt->buf + cvt->len_cvt * 2); \
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   326
        for (i = cvt->len_cvt / 2; i; --i, --src) { \
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   327
            const type val = *src; \
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   328
            dst -= 2; \
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   329
            dst[0] = dst[1] = val; \
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   330
        } \
icculus@1982
   331
    }
slouken@0
   332
icculus@1982
   333
    /* this function only cares about typesize, and data as a block of bits. */
icculus@1982
   334
    switch (SDL_AUDIO_BITSIZE(format)) {
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   335
    case 8:
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   336
        dup_chans_1_to_2(Uint8);
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   337
        break;
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   338
    case 16:
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   339
        dup_chans_1_to_2(Uint16);
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        break;
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   341
    case 32:
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   342
        dup_chans_1_to_2(Uint32);
slouken@1985
   343
        break;
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   344
    }
icculus@1982
   345
slouken@1985
   346
#undef dup_chans_1_to_2
icculus@1982
   347
slouken@1895
   348
    cvt->len_cvt *= 2;
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   349
    if (cvt->filters[++cvt->filter_index]) {
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   350
        cvt->filters[cvt->filter_index] (cvt, format);
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   351
    }
slouken@0
   352
}
slouken@0
   353
slouken@942
   354
slouken@942
   355
/* Duplicate a stereo channel to a pseudo-5.1 stream */
icculus@1982
   356
static void SDLCALL
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   357
SDL_ConvertSurround(SDL_AudioCVT * cvt, SDL_AudioFormat format)
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   358
{
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   359
    int i;
slouken@942
   360
slouken@942
   361
#ifdef DEBUG_CONVERT
slouken@1895
   362
    fprintf(stderr, "Converting stereo to surround\n");
slouken@942
   363
#endif
slouken@942
   364
slouken@1985
   365
    switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
slouken@1895
   366
    case AUDIO_U8:
slouken@1895
   367
        {
slouken@1895
   368
            Uint8 *src, *dst, lf, rf, ce;
slouken@942
   369
slouken@1895
   370
            src = (Uint8 *) (cvt->buf + cvt->len_cvt);
slouken@1895
   371
            dst = (Uint8 *) (cvt->buf + cvt->len_cvt * 3);
slouken@1895
   372
            for (i = cvt->len_cvt; i; --i) {
slouken@1895
   373
                dst -= 6;
slouken@1895
   374
                src -= 2;
slouken@1895
   375
                lf = src[0];
slouken@1895
   376
                rf = src[1];
slouken@1895
   377
                ce = (lf / 2) + (rf / 2);
slouken@1895
   378
                dst[0] = lf;
slouken@1895
   379
                dst[1] = rf;
slouken@1895
   380
                dst[2] = lf - ce;
slouken@1895
   381
                dst[3] = rf - ce;
slouken@1895
   382
                dst[4] = ce;
slouken@1895
   383
                dst[5] = ce;
slouken@1895
   384
            }
slouken@1895
   385
        }
slouken@1895
   386
        break;
slouken@942
   387
slouken@1895
   388
    case AUDIO_S8:
slouken@1895
   389
        {
slouken@1895
   390
            Sint8 *src, *dst, lf, rf, ce;
slouken@942
   391
slouken@1895
   392
            src = (Sint8 *) cvt->buf + cvt->len_cvt;
slouken@1895
   393
            dst = (Sint8 *) cvt->buf + cvt->len_cvt * 3;
slouken@1895
   394
            for (i = cvt->len_cvt; i; --i) {
slouken@1895
   395
                dst -= 6;
slouken@1895
   396
                src -= 2;
slouken@1895
   397
                lf = src[0];
slouken@1895
   398
                rf = src[1];
slouken@1895
   399
                ce = (lf / 2) + (rf / 2);
slouken@1895
   400
                dst[0] = lf;
slouken@1895
   401
                dst[1] = rf;
slouken@1895
   402
                dst[2] = lf - ce;
slouken@1895
   403
                dst[3] = rf - ce;
slouken@1895
   404
                dst[4] = ce;
slouken@1895
   405
                dst[5] = ce;
slouken@1895
   406
            }
slouken@1895
   407
        }
slouken@1895
   408
        break;
slouken@942
   409
slouken@1895
   410
    case AUDIO_U16:
slouken@1895
   411
        {
slouken@1895
   412
            Uint8 *src, *dst;
slouken@1895
   413
            Uint16 lf, rf, ce, lr, rr;
slouken@942
   414
slouken@1895
   415
            src = cvt->buf + cvt->len_cvt;
slouken@1895
   416
            dst = cvt->buf + cvt->len_cvt * 3;
slouken@942
   417
icculus@1982
   418
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
slouken@1895
   419
                for (i = cvt->len_cvt / 4; i; --i) {
slouken@1895
   420
                    dst -= 12;
slouken@1895
   421
                    src -= 4;
slouken@1895
   422
                    lf = (Uint16) ((src[0] << 8) | src[1]);
slouken@1895
   423
                    rf = (Uint16) ((src[2] << 8) | src[3]);
slouken@1895
   424
                    ce = (lf / 2) + (rf / 2);
slouken@1895
   425
                    rr = lf - ce;
slouken@1895
   426
                    lr = rf - ce;
slouken@1895
   427
                    dst[1] = (lf & 0xFF);
slouken@1895
   428
                    dst[0] = ((lf >> 8) & 0xFF);
slouken@1895
   429
                    dst[3] = (rf & 0xFF);
slouken@1895
   430
                    dst[2] = ((rf >> 8) & 0xFF);
slouken@942
   431
slouken@1895
   432
                    dst[1 + 4] = (lr & 0xFF);
slouken@1895
   433
                    dst[0 + 4] = ((lr >> 8) & 0xFF);
slouken@1895
   434
                    dst[3 + 4] = (rr & 0xFF);
slouken@1895
   435
                    dst[2 + 4] = ((rr >> 8) & 0xFF);
slouken@942
   436
slouken@1895
   437
                    dst[1 + 8] = (ce & 0xFF);
slouken@1895
   438
                    dst[0 + 8] = ((ce >> 8) & 0xFF);
slouken@1895
   439
                    dst[3 + 8] = (ce & 0xFF);
slouken@1895
   440
                    dst[2 + 8] = ((ce >> 8) & 0xFF);
slouken@1895
   441
                }
slouken@1895
   442
            } else {
slouken@1895
   443
                for (i = cvt->len_cvt / 4; i; --i) {
slouken@1895
   444
                    dst -= 12;
slouken@1895
   445
                    src -= 4;
slouken@1895
   446
                    lf = (Uint16) ((src[1] << 8) | src[0]);
slouken@1895
   447
                    rf = (Uint16) ((src[3] << 8) | src[2]);
slouken@1895
   448
                    ce = (lf / 2) + (rf / 2);
slouken@1895
   449
                    rr = lf - ce;
slouken@1895
   450
                    lr = rf - ce;
slouken@1895
   451
                    dst[0] = (lf & 0xFF);
slouken@1895
   452
                    dst[1] = ((lf >> 8) & 0xFF);
slouken@1895
   453
                    dst[2] = (rf & 0xFF);
slouken@1895
   454
                    dst[3] = ((rf >> 8) & 0xFF);
slouken@942
   455
slouken@1895
   456
                    dst[0 + 4] = (lr & 0xFF);
slouken@1895
   457
                    dst[1 + 4] = ((lr >> 8) & 0xFF);
slouken@1895
   458
                    dst[2 + 4] = (rr & 0xFF);
slouken@1895
   459
                    dst[3 + 4] = ((rr >> 8) & 0xFF);
slouken@942
   460
slouken@1895
   461
                    dst[0 + 8] = (ce & 0xFF);
slouken@1895
   462
                    dst[1 + 8] = ((ce >> 8) & 0xFF);
slouken@1895
   463
                    dst[2 + 8] = (ce & 0xFF);
slouken@1895
   464
                    dst[3 + 8] = ((ce >> 8) & 0xFF);
slouken@1895
   465
                }
slouken@1895
   466
            }
slouken@1895
   467
        }
slouken@1895
   468
        break;
slouken@942
   469
slouken@1895
   470
    case AUDIO_S16:
slouken@1895
   471
        {
slouken@1895
   472
            Uint8 *src, *dst;
slouken@1895
   473
            Sint16 lf, rf, ce, lr, rr;
slouken@942
   474
slouken@1895
   475
            src = cvt->buf + cvt->len_cvt;
slouken@1895
   476
            dst = cvt->buf + cvt->len_cvt * 3;
slouken@942
   477
icculus@1982
   478
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
slouken@1895
   479
                for (i = cvt->len_cvt / 4; i; --i) {
slouken@1895
   480
                    dst -= 12;
slouken@1895
   481
                    src -= 4;
slouken@1895
   482
                    lf = (Sint16) ((src[0] << 8) | src[1]);
slouken@1895
   483
                    rf = (Sint16) ((src[2] << 8) | src[3]);
slouken@1895
   484
                    ce = (lf / 2) + (rf / 2);
slouken@1895
   485
                    rr = lf - ce;
slouken@1895
   486
                    lr = rf - ce;
slouken@1895
   487
                    dst[1] = (lf & 0xFF);
slouken@1895
   488
                    dst[0] = ((lf >> 8) & 0xFF);
slouken@1895
   489
                    dst[3] = (rf & 0xFF);
slouken@1895
   490
                    dst[2] = ((rf >> 8) & 0xFF);
slouken@942
   491
slouken@1895
   492
                    dst[1 + 4] = (lr & 0xFF);
slouken@1895
   493
                    dst[0 + 4] = ((lr >> 8) & 0xFF);
slouken@1895
   494
                    dst[3 + 4] = (rr & 0xFF);
slouken@1895
   495
                    dst[2 + 4] = ((rr >> 8) & 0xFF);
slouken@942
   496
slouken@1895
   497
                    dst[1 + 8] = (ce & 0xFF);
slouken@1895
   498
                    dst[0 + 8] = ((ce >> 8) & 0xFF);
slouken@1895
   499
                    dst[3 + 8] = (ce & 0xFF);
slouken@1895
   500
                    dst[2 + 8] = ((ce >> 8) & 0xFF);
slouken@1895
   501
                }
slouken@1895
   502
            } else {
slouken@1895
   503
                for (i = cvt->len_cvt / 4; i; --i) {
slouken@1895
   504
                    dst -= 12;
slouken@1895
   505
                    src -= 4;
slouken@1895
   506
                    lf = (Sint16) ((src[1] << 8) | src[0]);
slouken@1895
   507
                    rf = (Sint16) ((src[3] << 8) | src[2]);
slouken@1895
   508
                    ce = (lf / 2) + (rf / 2);
slouken@1895
   509
                    rr = lf - ce;
slouken@1895
   510
                    lr = rf - ce;
slouken@1895
   511
                    dst[0] = (lf & 0xFF);
slouken@1895
   512
                    dst[1] = ((lf >> 8) & 0xFF);
slouken@1895
   513
                    dst[2] = (rf & 0xFF);
slouken@1895
   514
                    dst[3] = ((rf >> 8) & 0xFF);
slouken@942
   515
slouken@1895
   516
                    dst[0 + 4] = (lr & 0xFF);
slouken@1895
   517
                    dst[1 + 4] = ((lr >> 8) & 0xFF);
slouken@1895
   518
                    dst[2 + 4] = (rr & 0xFF);
slouken@1895
   519
                    dst[3 + 4] = ((rr >> 8) & 0xFF);
slouken@942
   520
slouken@1895
   521
                    dst[0 + 8] = (ce & 0xFF);
slouken@1895
   522
                    dst[1 + 8] = ((ce >> 8) & 0xFF);
slouken@1895
   523
                    dst[2 + 8] = (ce & 0xFF);
slouken@1895
   524
                    dst[3 + 8] = ((ce >> 8) & 0xFF);
slouken@1895
   525
                }
slouken@1895
   526
            }
slouken@1895
   527
        }
slouken@1895
   528
        break;
icculus@1982
   529
icculus@1982
   530
    case AUDIO_S32:
icculus@1982
   531
        {
icculus@1982
   532
            Sint32 lf, rf, ce;
icculus@1982
   533
            const Uint32 *src = (const Uint32 *) cvt->buf + cvt->len_cvt;
icculus@1982
   534
            Uint32 *dst = (Uint32 *) cvt->buf + cvt->len_cvt * 3;
icculus@1982
   535
icculus@1982
   536
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
icculus@1982
   537
                for (i = cvt->len_cvt / 8; i; --i) {
icculus@1982
   538
                    dst -= 6;
icculus@1982
   539
                    src -= 2;
icculus@1982
   540
                    lf = (Sint32) SDL_SwapBE32(src[0]);
icculus@1982
   541
                    rf = (Sint32) SDL_SwapBE32(src[1]);
icculus@1982
   542
                    ce = (lf / 2) + (rf / 2);
icculus@1982
   543
                    dst[0] = SDL_SwapBE32((Uint32) lf);
icculus@1982
   544
                    dst[1] = SDL_SwapBE32((Uint32) rf);
icculus@1982
   545
                    dst[2] = SDL_SwapBE32((Uint32) (lf - ce));
icculus@1982
   546
                    dst[3] = SDL_SwapBE32((Uint32) (rf - ce));
icculus@1982
   547
                    dst[4] = SDL_SwapBE32((Uint32) ce);
icculus@1982
   548
                    dst[5] = SDL_SwapBE32((Uint32) ce);
icculus@1982
   549
                }
icculus@1982
   550
            } else {
icculus@1982
   551
                for (i = cvt->len_cvt / 8; i; --i) {
icculus@1982
   552
                    dst -= 6;
icculus@1982
   553
                    src -= 2;
icculus@1982
   554
                    lf = (Sint32) SDL_SwapLE32(src[0]);
icculus@1982
   555
                    rf = (Sint32) SDL_SwapLE32(src[1]);
icculus@1982
   556
                    ce = (lf / 2) + (rf / 2);
icculus@1982
   557
                    dst[0] = src[0];
icculus@1982
   558
                    dst[1] = src[1];
icculus@1982
   559
                    dst[2] = SDL_SwapLE32((Uint32) (lf - ce));
icculus@1982
   560
                    dst[3] = SDL_SwapLE32((Uint32) (rf - ce));
icculus@1982
   561
                    dst[4] = SDL_SwapLE32((Uint32) ce);
icculus@1982
   562
                    dst[5] = SDL_SwapLE32((Uint32) ce);
icculus@1982
   563
                }
icculus@1982
   564
            }
icculus@1982
   565
        }
icculus@1982
   566
        break;
icculus@1982
   567
icculus@1982
   568
    case AUDIO_F32:
icculus@1982
   569
        {
icculus@1982
   570
            float lf, rf, ce;
icculus@2014
   571
            const float *src = (const float *) cvt->buf + cvt->len_cvt;
icculus@2014
   572
            float *dst = (float *) cvt->buf + cvt->len_cvt * 3;
icculus@1982
   573
icculus@1982
   574
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
icculus@1982
   575
                for (i = cvt->len_cvt / 8; i; --i) {
icculus@1982
   576
                    dst -= 6;
icculus@1982
   577
                    src -= 2;
icculus@2014
   578
                    lf = SDL_SwapFloatBE(src[0]);
icculus@2014
   579
                    rf = SDL_SwapFloatBE(src[1]);
icculus@1982
   580
                    ce = (lf * 0.5f) + (rf * 0.5f);
icculus@1982
   581
                    dst[0] = src[0];
icculus@1982
   582
                    dst[1] = src[1];
icculus@2014
   583
                    dst[2] = SDL_SwapFloatBE(lf - ce);
icculus@2014
   584
                    dst[3] = SDL_SwapFloatBE(rf - ce);
icculus@2014
   585
                    dst[4] = dst[5] = SDL_SwapFloatBE(ce);
icculus@1982
   586
                }
icculus@1982
   587
            } else {
icculus@1982
   588
                for (i = cvt->len_cvt / 8; i; --i) {
icculus@1982
   589
                    dst -= 6;
icculus@1982
   590
                    src -= 2;
icculus@2014
   591
                    lf = SDL_SwapFloatLE(src[0]);
icculus@2014
   592
                    rf = SDL_SwapFloatLE(src[1]);
icculus@1982
   593
                    ce = (lf * 0.5f) + (rf * 0.5f);
icculus@1982
   594
                    dst[0] = src[0];
icculus@1982
   595
                    dst[1] = src[1];
icculus@2014
   596
                    dst[2] = SDL_SwapFloatLE(lf - ce);
icculus@2014
   597
                    dst[3] = SDL_SwapFloatLE(rf - ce);
icculus@2014
   598
                    dst[4] = dst[5] = SDL_SwapFloatLE(ce);
icculus@1982
   599
                }
icculus@1982
   600
            }
icculus@1982
   601
        }
icculus@1982
   602
        break;
icculus@1982
   603
slouken@1895
   604
    }
slouken@1895
   605
    cvt->len_cvt *= 3;
slouken@1895
   606
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   607
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   608
    }
slouken@942
   609
}
slouken@942
   610
slouken@942
   611
slouken@942
   612
/* Duplicate a stereo channel to a pseudo-4.0 stream */
icculus@1982
   613
static void SDLCALL
icculus@1982
   614
SDL_ConvertSurround_4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
   615
{
slouken@1895
   616
    int i;
slouken@942
   617
slouken@942
   618
#ifdef DEBUG_CONVERT
slouken@1895
   619
    fprintf(stderr, "Converting stereo to quad\n");
slouken@942
   620
#endif
slouken@942
   621
slouken@1985
   622
    switch (format & (SDL_AUDIO_MASK_SIGNED | SDL_AUDIO_MASK_BITSIZE)) {
slouken@1895
   623
    case AUDIO_U8:
slouken@1895
   624
        {
slouken@1895
   625
            Uint8 *src, *dst, lf, rf, ce;
slouken@942
   626
slouken@1895
   627
            src = (Uint8 *) (cvt->buf + cvt->len_cvt);
slouken@1895
   628
            dst = (Uint8 *) (cvt->buf + cvt->len_cvt * 2);
slouken@1895
   629
            for (i = cvt->len_cvt; i; --i) {
slouken@1895
   630
                dst -= 4;
slouken@1895
   631
                src -= 2;
slouken@1895
   632
                lf = src[0];
slouken@1895
   633
                rf = src[1];
slouken@1895
   634
                ce = (lf / 2) + (rf / 2);
slouken@1895
   635
                dst[0] = lf;
slouken@1895
   636
                dst[1] = rf;
slouken@1895
   637
                dst[2] = lf - ce;
slouken@1895
   638
                dst[3] = rf - ce;
slouken@1895
   639
            }
slouken@1895
   640
        }
slouken@1895
   641
        break;
slouken@942
   642
slouken@1895
   643
    case AUDIO_S8:
slouken@1895
   644
        {
slouken@1895
   645
            Sint8 *src, *dst, lf, rf, ce;
slouken@942
   646
slouken@1895
   647
            src = (Sint8 *) cvt->buf + cvt->len_cvt;
slouken@1895
   648
            dst = (Sint8 *) cvt->buf + cvt->len_cvt * 2;
slouken@1895
   649
            for (i = cvt->len_cvt; i; --i) {
slouken@1895
   650
                dst -= 4;
slouken@1895
   651
                src -= 2;
slouken@1895
   652
                lf = src[0];
slouken@1895
   653
                rf = src[1];
slouken@1895
   654
                ce = (lf / 2) + (rf / 2);
slouken@1895
   655
                dst[0] = lf;
slouken@1895
   656
                dst[1] = rf;
slouken@1895
   657
                dst[2] = lf - ce;
slouken@1895
   658
                dst[3] = rf - ce;
slouken@1895
   659
            }
slouken@1895
   660
        }
slouken@1895
   661
        break;
slouken@942
   662
slouken@1895
   663
    case AUDIO_U16:
slouken@1895
   664
        {
slouken@1895
   665
            Uint8 *src, *dst;
slouken@1895
   666
            Uint16 lf, rf, ce, lr, rr;
slouken@942
   667
slouken@1895
   668
            src = cvt->buf + cvt->len_cvt;
slouken@1895
   669
            dst = cvt->buf + cvt->len_cvt * 2;
slouken@942
   670
icculus@1982
   671
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
slouken@1895
   672
                for (i = cvt->len_cvt / 4; i; --i) {
slouken@1895
   673
                    dst -= 8;
slouken@1895
   674
                    src -= 4;
slouken@1895
   675
                    lf = (Uint16) ((src[0] << 8) | src[1]);
slouken@1895
   676
                    rf = (Uint16) ((src[2] << 8) | src[3]);
slouken@1895
   677
                    ce = (lf / 2) + (rf / 2);
slouken@1895
   678
                    rr = lf - ce;
slouken@1895
   679
                    lr = rf - ce;
slouken@1895
   680
                    dst[1] = (lf & 0xFF);
slouken@1895
   681
                    dst[0] = ((lf >> 8) & 0xFF);
slouken@1895
   682
                    dst[3] = (rf & 0xFF);
slouken@1895
   683
                    dst[2] = ((rf >> 8) & 0xFF);
slouken@942
   684
slouken@1895
   685
                    dst[1 + 4] = (lr & 0xFF);
slouken@1895
   686
                    dst[0 + 4] = ((lr >> 8) & 0xFF);
slouken@1895
   687
                    dst[3 + 4] = (rr & 0xFF);
slouken@1895
   688
                    dst[2 + 4] = ((rr >> 8) & 0xFF);
slouken@1895
   689
                }
slouken@1895
   690
            } else {
slouken@1895
   691
                for (i = cvt->len_cvt / 4; i; --i) {
slouken@1895
   692
                    dst -= 8;
slouken@1895
   693
                    src -= 4;
slouken@1895
   694
                    lf = (Uint16) ((src[1] << 8) | src[0]);
slouken@1895
   695
                    rf = (Uint16) ((src[3] << 8) | src[2]);
slouken@1895
   696
                    ce = (lf / 2) + (rf / 2);
slouken@1895
   697
                    rr = lf - ce;
slouken@1895
   698
                    lr = rf - ce;
slouken@1895
   699
                    dst[0] = (lf & 0xFF);
slouken@1895
   700
                    dst[1] = ((lf >> 8) & 0xFF);
slouken@1895
   701
                    dst[2] = (rf & 0xFF);
slouken@1895
   702
                    dst[3] = ((rf >> 8) & 0xFF);
slouken@942
   703
slouken@1895
   704
                    dst[0 + 4] = (lr & 0xFF);
slouken@1895
   705
                    dst[1 + 4] = ((lr >> 8) & 0xFF);
slouken@1895
   706
                    dst[2 + 4] = (rr & 0xFF);
slouken@1895
   707
                    dst[3 + 4] = ((rr >> 8) & 0xFF);
slouken@1895
   708
                }
slouken@1895
   709
            }
slouken@1895
   710
        }
slouken@1895
   711
        break;
slouken@942
   712
slouken@1895
   713
    case AUDIO_S16:
slouken@1895
   714
        {
slouken@1895
   715
            Uint8 *src, *dst;
slouken@1895
   716
            Sint16 lf, rf, ce, lr, rr;
slouken@942
   717
slouken@1895
   718
            src = cvt->buf + cvt->len_cvt;
slouken@1895
   719
            dst = cvt->buf + cvt->len_cvt * 2;
slouken@942
   720
icculus@1982
   721
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
slouken@1895
   722
                for (i = cvt->len_cvt / 4; i; --i) {
slouken@1895
   723
                    dst -= 8;
slouken@1895
   724
                    src -= 4;
slouken@1895
   725
                    lf = (Sint16) ((src[0] << 8) | src[1]);
slouken@1895
   726
                    rf = (Sint16) ((src[2] << 8) | src[3]);
slouken@1895
   727
                    ce = (lf / 2) + (rf / 2);
slouken@1895
   728
                    rr = lf - ce;
slouken@1895
   729
                    lr = rf - ce;
slouken@1895
   730
                    dst[1] = (lf & 0xFF);
slouken@1895
   731
                    dst[0] = ((lf >> 8) & 0xFF);
slouken@1895
   732
                    dst[3] = (rf & 0xFF);
slouken@1895
   733
                    dst[2] = ((rf >> 8) & 0xFF);
slouken@942
   734
slouken@1895
   735
                    dst[1 + 4] = (lr & 0xFF);
slouken@1895
   736
                    dst[0 + 4] = ((lr >> 8) & 0xFF);
slouken@1895
   737
                    dst[3 + 4] = (rr & 0xFF);
slouken@1895
   738
                    dst[2 + 4] = ((rr >> 8) & 0xFF);
slouken@1895
   739
                }
slouken@1895
   740
            } else {
slouken@1895
   741
                for (i = cvt->len_cvt / 4; i; --i) {
slouken@1895
   742
                    dst -= 8;
slouken@1895
   743
                    src -= 4;
slouken@1895
   744
                    lf = (Sint16) ((src[1] << 8) | src[0]);
slouken@1895
   745
                    rf = (Sint16) ((src[3] << 8) | src[2]);
slouken@1895
   746
                    ce = (lf / 2) + (rf / 2);
slouken@1895
   747
                    rr = lf - ce;
slouken@1895
   748
                    lr = rf - ce;
slouken@1895
   749
                    dst[0] = (lf & 0xFF);
slouken@1895
   750
                    dst[1] = ((lf >> 8) & 0xFF);
slouken@1895
   751
                    dst[2] = (rf & 0xFF);
slouken@1895
   752
                    dst[3] = ((rf >> 8) & 0xFF);
slouken@942
   753
slouken@1895
   754
                    dst[0 + 4] = (lr & 0xFF);
slouken@1895
   755
                    dst[1 + 4] = ((lr >> 8) & 0xFF);
slouken@1895
   756
                    dst[2 + 4] = (rr & 0xFF);
slouken@1895
   757
                    dst[3 + 4] = ((rr >> 8) & 0xFF);
slouken@1895
   758
                }
slouken@1895
   759
            }
slouken@1895
   760
        }
slouken@1895
   761
        break;
slouken@942
   762
icculus@1982
   763
    case AUDIO_S32:
icculus@1982
   764
        {
icculus@1982
   765
            const Uint32 *src = (const Uint32 *) (cvt->buf + cvt->len_cvt);
icculus@1982
   766
            Uint32 *dst = (Uint32 *) (cvt->buf + cvt->len_cvt * 2);
icculus@1982
   767
            Sint32 lf, rf, ce;
slouken@942
   768
icculus@1982
   769
            if (SDL_AUDIO_ISBIGENDIAN(format)) {
icculus@1982
   770
                for (i = cvt->len_cvt / 8; i; --i) {
icculus@1982
   771
                    dst -= 4;
icculus@1982
   772
                    src -= 2;
icculus@1982
   773
                    lf = (Sint32) SDL_SwapBE32(src[0]);
icculus@1982
   774
                    rf = (Sint32) SDL_SwapBE32(src[1]);
icculus@1982
   775
                    ce = (lf / 2) + (rf / 2);
icculus@1982
   776
                    dst[0] = src[0];
icculus@1982
   777
                    dst[1] = src[1];
icculus@1982
   778
                    dst[2] = SDL_SwapBE32((Uint32) (lf - ce));
icculus@1982
   779
                    dst[3] = SDL_SwapBE32((Uint32) (rf - ce));
icculus@1982
   780
                }
icculus@1982
   781
            } else {
icculus@1982
   782
                for (i = cvt->len_cvt / 8; i; --i) {
icculus@1982
   783
                    dst -= 4;
icculus@1982
   784
                    src -= 2;
icculus@1982
   785
                    lf = (Sint32) SDL_SwapLE32(src[0]);
icculus@1982
   786
                    rf = (Sint32) SDL_SwapLE32(src[1]);
icculus@1982
   787
                    ce = (lf / 2) + (rf / 2);
icculus@1982
   788
                    dst[0] = src[0];
icculus@1982
   789
                    dst[1] = src[1];
icculus@1982
   790
                    dst[2] = SDL_SwapLE32((Uint32) (lf - ce));
icculus@1982
   791
                    dst[3] = SDL_SwapLE32((Uint32) (rf - ce));
icculus@1982
   792
                }
icculus@1982
   793
            }
slouken@1895
   794
        }
slouken@1895
   795
        break;
slouken@1895
   796
    }
slouken@1895
   797
    cvt->len_cvt *= 2;
slouken@1895
   798
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   799
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   800
    }
slouken@0
   801
}
slouken@0
   802
icculus@1982
   803
/* Convert rate up by multiple of 2 */
icculus@1982
   804
static void SDLCALL
icculus@1982
   805
SDL_RateMUL2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
icculus@1982
   806
{
icculus@1982
   807
    int i;
icculus@1982
   808
icculus@1982
   809
#ifdef DEBUG_CONVERT
icculus@1982
   810
    fprintf(stderr, "Converting audio rate * 2 (mono)\n");
icculus@1982
   811
#endif
icculus@1982
   812
slouken@1985
   813
#define mul2_mono(type) { \
icculus@1982
   814
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
icculus@1982
   815
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
icculus@1982
   816
        for (i = cvt->len_cvt / sizeof (type); i; --i) { \
icculus@1982
   817
            src--; \
icculus@1982
   818
            dst[-1] = dst[-2] = src[0]; \
icculus@1982
   819
            dst -= 2; \
icculus@1982
   820
        } \
icculus@1982
   821
    }
icculus@1982
   822
icculus@1982
   823
    switch (SDL_AUDIO_BITSIZE(format)) {
icculus@1982
   824
    case 8:
icculus@1982
   825
        mul2_mono(Uint8);
icculus@1982
   826
        break;
icculus@1982
   827
    case 16:
icculus@1982
   828
        mul2_mono(Uint16);
icculus@1982
   829
        break;
icculus@1982
   830
    case 32:
icculus@1982
   831
        mul2_mono(Uint32);
icculus@1982
   832
        break;
icculus@1982
   833
    }
icculus@1982
   834
slouken@1985
   835
#undef mul2_mono
icculus@1982
   836
icculus@1982
   837
    cvt->len_cvt *= 2;
icculus@1982
   838
    if (cvt->filters[++cvt->filter_index]) {
icculus@1982
   839
        cvt->filters[cvt->filter_index] (cvt, format);
icculus@1982
   840
    }
icculus@1982
   841
}
icculus@1982
   842
slouken@942
   843
slouken@942
   844
/* Convert rate up by multiple of 2, for stereo */
icculus@1982
   845
static void SDLCALL
icculus@1982
   846
SDL_RateMUL2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
   847
{
slouken@1895
   848
    int i;
slouken@942
   849
slouken@942
   850
#ifdef DEBUG_CONVERT
icculus@1982
   851
    fprintf(stderr, "Converting audio rate * 2 (stereo)\n");
slouken@942
   852
#endif
icculus@1982
   853
slouken@1985
   854
#define mul2_stereo(type) { \
icculus@1982
   855
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
icculus@1982
   856
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
icculus@1982
   857
        for (i = cvt->len_cvt / (sizeof (type) * 2); i; --i) { \
icculus@1982
   858
            const type r = src[-1]; \
icculus@1982
   859
            const type l = src[-2]; \
icculus@1982
   860
            src -= 2; \
icculus@1982
   861
            dst[-1] = r; \
icculus@1982
   862
            dst[-2] = l; \
icculus@1982
   863
            dst[-3] = r; \
icculus@1982
   864
            dst[-4] = l; \
icculus@1982
   865
            dst -= 4; \
icculus@1982
   866
        } \
icculus@1982
   867
    }
icculus@1982
   868
icculus@1982
   869
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
   870
    case 8:
icculus@1982
   871
        mul2_stereo(Uint8);
slouken@1895
   872
        break;
slouken@1895
   873
    case 16:
icculus@1982
   874
        mul2_stereo(Uint16);
icculus@1982
   875
        break;
icculus@1982
   876
    case 32:
icculus@1982
   877
        mul2_stereo(Uint32);
slouken@1895
   878
        break;
slouken@1895
   879
    }
icculus@1982
   880
slouken@1985
   881
#undef mul2_stereo
icculus@1982
   882
slouken@1895
   883
    cvt->len_cvt *= 2;
slouken@1895
   884
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   885
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   886
    }
slouken@942
   887
}
slouken@942
   888
slouken@942
   889
/* Convert rate up by multiple of 2, for quad */
icculus@1982
   890
static void SDLCALL
icculus@1982
   891
SDL_RateMUL2_c4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
   892
{
slouken@1895
   893
    int i;
slouken@942
   894
slouken@942
   895
#ifdef DEBUG_CONVERT
icculus@1982
   896
    fprintf(stderr, "Converting audio rate * 2 (quad)\n");
slouken@942
   897
#endif
icculus@1982
   898
slouken@1985
   899
#define mul2_quad(type) { \
icculus@1982
   900
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
icculus@1982
   901
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
icculus@1982
   902
        for (i = cvt->len_cvt / (sizeof (type) * 4); i; --i) { \
icculus@1982
   903
            const type c1 = src[-1]; \
icculus@1982
   904
            const type c2 = src[-2]; \
icculus@1982
   905
            const type c3 = src[-3]; \
icculus@1982
   906
            const type c4 = src[-4]; \
icculus@1982
   907
            src -= 4; \
icculus@1982
   908
            dst[-1] = c1; \
icculus@1982
   909
            dst[-2] = c2; \
icculus@1982
   910
            dst[-3] = c3; \
icculus@1982
   911
            dst[-4] = c4; \
icculus@1982
   912
            dst[-5] = c1; \
icculus@1982
   913
            dst[-6] = c2; \
icculus@1982
   914
            dst[-7] = c3; \
icculus@1982
   915
            dst[-8] = c4; \
icculus@1982
   916
            dst -= 8; \
icculus@1982
   917
        } \
icculus@1982
   918
    }
icculus@1982
   919
icculus@1982
   920
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
   921
    case 8:
icculus@1982
   922
        mul2_quad(Uint8);
slouken@1895
   923
        break;
slouken@1895
   924
    case 16:
icculus@1982
   925
        mul2_quad(Uint16);
icculus@1982
   926
        break;
icculus@1982
   927
    case 32:
icculus@1982
   928
        mul2_quad(Uint32);
slouken@1895
   929
        break;
slouken@1895
   930
    }
icculus@1982
   931
slouken@1985
   932
#undef mul2_quad
icculus@1982
   933
slouken@1895
   934
    cvt->len_cvt *= 2;
slouken@1895
   935
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   936
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   937
    }
slouken@942
   938
}
slouken@942
   939
slouken@942
   940
slouken@942
   941
/* Convert rate up by multiple of 2, for 5.1 */
icculus@1982
   942
static void SDLCALL
icculus@1982
   943
SDL_RateMUL2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
   944
{
slouken@1895
   945
    int i;
slouken@942
   946
slouken@942
   947
#ifdef DEBUG_CONVERT
icculus@1982
   948
    fprintf(stderr, "Converting audio rate * 2 (six channels)\n");
slouken@942
   949
#endif
icculus@1982
   950
slouken@1985
   951
#define mul2_chansix(type) { \
icculus@1982
   952
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
icculus@1982
   953
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * 2)); \
icculus@1982
   954
        for (i = cvt->len_cvt / (sizeof (type) * 6); i; --i) { \
icculus@1982
   955
            const type c1 = src[-1]; \
icculus@1982
   956
            const type c2 = src[-2]; \
icculus@1982
   957
            const type c3 = src[-3]; \
icculus@1982
   958
            const type c4 = src[-4]; \
icculus@1982
   959
            const type c5 = src[-5]; \
icculus@1982
   960
            const type c6 = src[-6]; \
icculus@1982
   961
            src -= 6; \
icculus@1982
   962
            dst[-1] = c1; \
icculus@1982
   963
            dst[-2] = c2; \
icculus@1982
   964
            dst[-3] = c3; \
icculus@1982
   965
            dst[-4] = c4; \
icculus@1982
   966
            dst[-5] = c5; \
icculus@1982
   967
            dst[-6] = c6; \
icculus@1982
   968
            dst[-7] = c1; \
icculus@1982
   969
            dst[-8] = c2; \
icculus@1982
   970
            dst[-9] = c3; \
icculus@1982
   971
            dst[-10] = c4; \
icculus@1982
   972
            dst[-11] = c5; \
icculus@1982
   973
            dst[-12] = c6; \
icculus@1982
   974
            dst -= 12; \
icculus@1982
   975
        } \
icculus@1982
   976
    }
icculus@1982
   977
icculus@1982
   978
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
   979
    case 8:
icculus@1982
   980
        mul2_chansix(Uint8);
slouken@1895
   981
        break;
slouken@1895
   982
    case 16:
icculus@1982
   983
        mul2_chansix(Uint16);
icculus@1982
   984
        break;
icculus@1982
   985
    case 32:
icculus@1982
   986
        mul2_chansix(Uint32);
slouken@1895
   987
        break;
slouken@1895
   988
    }
icculus@1982
   989
slouken@1985
   990
#undef mul2_chansix
icculus@1982
   991
slouken@1895
   992
    cvt->len_cvt *= 2;
slouken@1895
   993
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
   994
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
   995
    }
slouken@942
   996
}
slouken@942
   997
slouken@0
   998
/* Convert rate down by multiple of 2 */
icculus@1982
   999
static void SDLCALL
icculus@1982
  1000
SDL_RateDIV2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@0
  1001
{
slouken@1895
  1002
    int i;
slouken@0
  1003
slouken@0
  1004
#ifdef DEBUG_CONVERT
icculus@1982
  1005
    fprintf(stderr, "Converting audio rate / 2 (mono)\n");
slouken@0
  1006
#endif
icculus@1982
  1007
slouken@1985
  1008
#define div2_mono(type) { \
icculus@1982
  1009
        const type *src = (const type *) cvt->buf; \
icculus@1982
  1010
        type *dst = (type *) cvt->buf; \
icculus@1982
  1011
        for (i = cvt->len_cvt / (sizeof (type) * 2); i; --i) { \
icculus@1982
  1012
            dst[0] = src[0]; \
icculus@1982
  1013
            src += 2; \
icculus@1982
  1014
            dst++; \
icculus@1982
  1015
        } \
icculus@1982
  1016
    }
icculus@1982
  1017
icculus@1982
  1018
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
  1019
    case 8:
icculus@1982
  1020
        div2_mono(Uint8);
slouken@1895
  1021
        break;
slouken@1895
  1022
    case 16:
icculus@1982
  1023
        div2_mono(Uint16);
icculus@1982
  1024
        break;
icculus@1982
  1025
    case 32:
icculus@1982
  1026
        div2_mono(Uint32);
slouken@1895
  1027
        break;
slouken@1895
  1028
    }
icculus@1982
  1029
slouken@1985
  1030
#undef div2_mono
icculus@1982
  1031
slouken@1895
  1032
    cvt->len_cvt /= 2;
slouken@1895
  1033
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
  1034
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
  1035
    }
slouken@0
  1036
}
slouken@0
  1037
slouken@942
  1038
slouken@942
  1039
/* Convert rate down by multiple of 2, for stereo */
icculus@1982
  1040
static void SDLCALL
icculus@1982
  1041
SDL_RateDIV2_c2(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
  1042
{
slouken@1895
  1043
    int i;
slouken@942
  1044
slouken@942
  1045
#ifdef DEBUG_CONVERT
icculus@1982
  1046
    fprintf(stderr, "Converting audio rate / 2 (stereo)\n");
slouken@942
  1047
#endif
icculus@1982
  1048
slouken@1985
  1049
#define div2_stereo(type) { \
icculus@1982
  1050
        const type *src = (const type *) cvt->buf; \
icculus@1982
  1051
        type *dst = (type *) cvt->buf; \
icculus@1982
  1052
        for (i = cvt->len_cvt / (sizeof (type) * 4); i; --i) { \
icculus@1982
  1053
            dst[0] = src[0]; \
icculus@1982
  1054
            dst[1] = src[1]; \
icculus@1982
  1055
            src += 4; \
icculus@1982
  1056
            dst += 2; \
icculus@1982
  1057
        } \
icculus@1982
  1058
    }
icculus@1982
  1059
icculus@1982
  1060
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
  1061
    case 8:
icculus@1982
  1062
        div2_stereo(Uint8);
slouken@1895
  1063
        break;
slouken@1895
  1064
    case 16:
icculus@1982
  1065
        div2_stereo(Uint16);
icculus@1982
  1066
        break;
icculus@1982
  1067
    case 32:
icculus@1982
  1068
        div2_stereo(Uint32);
slouken@1895
  1069
        break;
slouken@1895
  1070
    }
icculus@1982
  1071
slouken@1985
  1072
#undef div2_stereo
icculus@1982
  1073
slouken@1895
  1074
    cvt->len_cvt /= 2;
slouken@1895
  1075
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
  1076
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
  1077
    }
slouken@942
  1078
}
slouken@942
  1079
slouken@942
  1080
slouken@942
  1081
/* Convert rate down by multiple of 2, for quad */
icculus@1982
  1082
static void SDLCALL
icculus@1982
  1083
SDL_RateDIV2_c4(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
  1084
{
slouken@1895
  1085
    int i;
slouken@942
  1086
slouken@942
  1087
#ifdef DEBUG_CONVERT
icculus@1982
  1088
    fprintf(stderr, "Converting audio rate / 2 (quad)\n");
slouken@942
  1089
#endif
icculus@1982
  1090
slouken@1985
  1091
#define div2_quad(type) { \
icculus@1982
  1092
        const type *src = (const type *) cvt->buf; \
icculus@1982
  1093
        type *dst = (type *) cvt->buf; \
icculus@1982
  1094
        for (i = cvt->len_cvt / (sizeof (type) * 8); i; --i) { \
icculus@1982
  1095
            dst[0] = src[0]; \
icculus@1982
  1096
            dst[1] = src[1]; \
icculus@1982
  1097
            dst[2] = src[2]; \
icculus@1982
  1098
            dst[3] = src[3]; \
icculus@1982
  1099
            src += 8; \
icculus@1982
  1100
            dst += 4; \
icculus@1982
  1101
        } \
icculus@1982
  1102
    }
icculus@1982
  1103
icculus@1982
  1104
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
  1105
    case 8:
icculus@1982
  1106
        div2_quad(Uint8);
slouken@1895
  1107
        break;
slouken@1895
  1108
    case 16:
icculus@1982
  1109
        div2_quad(Uint16);
icculus@1982
  1110
        break;
icculus@1982
  1111
    case 32:
icculus@1982
  1112
        div2_quad(Uint32);
slouken@1895
  1113
        break;
slouken@1895
  1114
    }
icculus@1982
  1115
slouken@1985
  1116
#undef div2_quad
icculus@1982
  1117
slouken@1895
  1118
    cvt->len_cvt /= 2;
slouken@1895
  1119
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
  1120
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
  1121
    }
slouken@942
  1122
}
slouken@942
  1123
slouken@942
  1124
/* Convert rate down by multiple of 2, for 5.1 */
icculus@1982
  1125
static void SDLCALL
icculus@1982
  1126
SDL_RateDIV2_c6(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@942
  1127
{
slouken@1895
  1128
    int i;
slouken@942
  1129
slouken@942
  1130
#ifdef DEBUG_CONVERT
icculus@1982
  1131
    fprintf(stderr, "Converting audio rate / 2 (six channels)\n");
slouken@942
  1132
#endif
icculus@1982
  1133
slouken@1985
  1134
#define div2_chansix(type) { \
icculus@1982
  1135
        const type *src = (const type *) cvt->buf; \
icculus@1982
  1136
        type *dst = (type *) cvt->buf; \
icculus@1982
  1137
        for (i = cvt->len_cvt / (sizeof (type) * 12); i; --i) { \
icculus@1982
  1138
            dst[0] = src[0]; \
icculus@1982
  1139
            dst[1] = src[1]; \
icculus@1982
  1140
            dst[2] = src[2]; \
icculus@1982
  1141
            dst[3] = src[3]; \
icculus@1982
  1142
            dst[4] = src[4]; \
icculus@1982
  1143
            dst[5] = src[5]; \
icculus@1982
  1144
            src += 12; \
icculus@1982
  1145
            dst += 6; \
icculus@1982
  1146
        } \
icculus@1982
  1147
    }
icculus@1982
  1148
icculus@1982
  1149
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
  1150
    case 8:
icculus@1982
  1151
        div2_chansix(Uint8);
slouken@1895
  1152
        break;
slouken@1895
  1153
    case 16:
icculus@1982
  1154
        div2_chansix(Uint16);
icculus@1982
  1155
        break;
icculus@1982
  1156
    case 32:
icculus@1982
  1157
        div2_chansix(Uint32);
slouken@1895
  1158
        break;
slouken@1895
  1159
    }
icculus@1982
  1160
slouken@1985
  1161
#undef div_chansix
icculus@1982
  1162
slouken@1895
  1163
    cvt->len_cvt /= 2;
slouken@1895
  1164
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
  1165
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
  1166
    }
slouken@942
  1167
}
slouken@942
  1168
slouken@0
  1169
/* Very slow rate conversion routine */
icculus@1982
  1170
static void SDLCALL
icculus@1982
  1171
SDL_RateSLOW(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@0
  1172
{
slouken@1895
  1173
    double ipos;
slouken@1895
  1174
    int i, clen;
slouken@0
  1175
slouken@0
  1176
#ifdef DEBUG_CONVERT
slouken@1895
  1177
    fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0 / cvt->rate_incr);
slouken@0
  1178
#endif
slouken@1895
  1179
    clen = (int) ((double) cvt->len_cvt / cvt->rate_incr);
slouken@1895
  1180
    if (cvt->rate_incr > 1.0) {
icculus@1982
  1181
        switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
  1182
        case 8:
slouken@1895
  1183
            {
slouken@1895
  1184
                Uint8 *output;
slouken@0
  1185
slouken@1895
  1186
                output = cvt->buf;
slouken@1895
  1187
                ipos = 0.0;
slouken@1895
  1188
                for (i = clen; i; --i) {
slouken@1895
  1189
                    *output = cvt->buf[(int) ipos];
slouken@1895
  1190
                    ipos += cvt->rate_incr;
slouken@1895
  1191
                    output += 1;
slouken@1895
  1192
                }
slouken@1895
  1193
            }
slouken@1895
  1194
            break;
slouken@0
  1195
slouken@1895
  1196
        case 16:
slouken@1895
  1197
            {
slouken@1895
  1198
                Uint16 *output;
slouken@0
  1199
slouken@1895
  1200
                clen &= ~1;
slouken@1895
  1201
                output = (Uint16 *) cvt->buf;
slouken@1895
  1202
                ipos = 0.0;
slouken@1895
  1203
                for (i = clen / 2; i; --i) {
slouken@1895
  1204
                    *output = ((Uint16 *) cvt->buf)[(int) ipos];
slouken@1895
  1205
                    ipos += cvt->rate_incr;
slouken@1895
  1206
                    output += 1;
slouken@1895
  1207
                }
slouken@1895
  1208
            }
slouken@1895
  1209
            break;
icculus@1982
  1210
icculus@1982
  1211
        case 32:
icculus@1982
  1212
            {
icculus@1982
  1213
                /* !!! FIXME: need 32-bit converter here! */
slouken@2130
  1214
#ifdef DEBUG_CONVERT
icculus@1982
  1215
                fprintf(stderr, "FIXME: need 32-bit converter here!\n");
slouken@2130
  1216
#endif
icculus@1982
  1217
            }
slouken@1895
  1218
        }
slouken@1895
  1219
    } else {
icculus@1982
  1220
        switch (SDL_AUDIO_BITSIZE(format)) {
slouken@1895
  1221
        case 8:
slouken@1895
  1222
            {
slouken@1895
  1223
                Uint8 *output;
slouken@0
  1224
slouken@1895
  1225
                output = cvt->buf + clen;
slouken@1895
  1226
                ipos = (double) cvt->len_cvt;
slouken@1895
  1227
                for (i = clen; i; --i) {
slouken@1895
  1228
                    ipos -= cvt->rate_incr;
slouken@1895
  1229
                    output -= 1;
slouken@1895
  1230
                    *output = cvt->buf[(int) ipos];
slouken@1895
  1231
                }
slouken@1895
  1232
            }
slouken@1895
  1233
            break;
slouken@0
  1234
slouken@1895
  1235
        case 16:
slouken@1895
  1236
            {
slouken@1895
  1237
                Uint16 *output;
slouken@0
  1238
slouken@1895
  1239
                clen &= ~1;
slouken@1895
  1240
                output = (Uint16 *) (cvt->buf + clen);
slouken@1895
  1241
                ipos = (double) cvt->len_cvt / 2;
slouken@1895
  1242
                for (i = clen / 2; i; --i) {
slouken@1895
  1243
                    ipos -= cvt->rate_incr;
slouken@1895
  1244
                    output -= 1;
slouken@1895
  1245
                    *output = ((Uint16 *) cvt->buf)[(int) ipos];
slouken@1895
  1246
                }
slouken@1895
  1247
            }
slouken@1895
  1248
            break;
icculus@1982
  1249
icculus@1982
  1250
        case 32:
icculus@1982
  1251
            {
icculus@1982
  1252
                /* !!! FIXME: need 32-bit converter here! */
slouken@2130
  1253
#ifdef DEBUG_CONVERT
icculus@1982
  1254
                fprintf(stderr, "FIXME: need 32-bit converter here!\n");
slouken@2130
  1255
#endif
icculus@1982
  1256
            }
slouken@1895
  1257
        }
slouken@1895
  1258
    }
icculus@1982
  1259
slouken@1895
  1260
    cvt->len_cvt = clen;
slouken@1895
  1261
    if (cvt->filters[++cvt->filter_index]) {
slouken@1895
  1262
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@1895
  1263
    }
slouken@0
  1264
}
slouken@0
  1265
slouken@1895
  1266
int
slouken@1895
  1267
SDL_ConvertAudio(SDL_AudioCVT * cvt)
slouken@0
  1268
{
slouken@1895
  1269
    /* Make sure there's data to convert */
slouken@1895
  1270
    if (cvt->buf == NULL) {
slouken@1895
  1271
        SDL_SetError("No buffer allocated for conversion");
slouken@1895
  1272
        return (-1);
slouken@1895
  1273
    }
slouken@1895
  1274
    /* Return okay if no conversion is necessary */
slouken@1895
  1275
    cvt->len_cvt = cvt->len;
slouken@1895
  1276
    if (cvt->filters[0] == NULL) {
slouken@1895
  1277
        return (0);
slouken@1895
  1278
    }
slouken@0
  1279
slouken@1895
  1280
    /* Set up the conversion and go! */
slouken@1895
  1281
    cvt->filter_index = 0;
slouken@1895
  1282
    cvt->filters[0] (cvt, cvt->src_format);
slouken@1895
  1283
    return (0);
slouken@0
  1284
}
slouken@0
  1285
icculus@1982
  1286
icculus@1982
  1287
static SDL_AudioFilter
icculus@1982
  1288
SDL_HandTunedTypeCVT(SDL_AudioFormat src_fmt, SDL_AudioFormat dst_fmt)
icculus@1982
  1289
{
icculus@1982
  1290
    /*
icculus@1982
  1291
     * Fill in any future conversions that are specialized to a
icculus@1982
  1292
     *  processor, platform, compiler, or library here.
icculus@1982
  1293
     */
icculus@1982
  1294
slouken@1985
  1295
    return NULL;                /* no specialized converter code available. */
icculus@1982
  1296
}
icculus@1982
  1297
icculus@1982
  1298
icculus@1982
  1299
/*
icculus@1982
  1300
 * Find a converter between two data types. We try to select a hand-tuned
icculus@1982
  1301
 *  asm/vectorized/optimized function first, and then fallback to an
icculus@1982
  1302
 *  autogenerated function that is customized to convert between two
icculus@1982
  1303
 *  specific data types.
icculus@1982
  1304
 */
icculus@1982
  1305
static int
icculus@1982
  1306
SDL_BuildAudioTypeCVT(SDL_AudioCVT * cvt,
icculus@1982
  1307
                      SDL_AudioFormat src_fmt, SDL_AudioFormat dst_fmt)
icculus@1982
  1308
{
icculus@1982
  1309
    if (src_fmt != dst_fmt) {
icculus@1982
  1310
        const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
icculus@1982
  1311
        const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
icculus@1982
  1312
        SDL_AudioFilter filter = SDL_HandTunedTypeCVT(src_fmt, dst_fmt);
icculus@1982
  1313
icculus@1982
  1314
        /* No hand-tuned converter? Try the autogenerated ones. */
icculus@1982
  1315
        if (filter == NULL) {
icculus@1982
  1316
            int i;
icculus@1982
  1317
            for (i = 0; sdl_audio_type_filters[i].filter != NULL; i++) {
icculus@1982
  1318
                const SDL_AudioTypeFilters *filt = &sdl_audio_type_filters[i];
icculus@1982
  1319
                if ((filt->src_fmt == src_fmt) && (filt->dst_fmt == dst_fmt)) {
icculus@1982
  1320
                    filter = filt->filter;
icculus@1982
  1321
                    break;
icculus@1982
  1322
                }
icculus@1982
  1323
            }
icculus@1982
  1324
icculus@1982
  1325
            if (filter == NULL) {
slouken@1985
  1326
                return -1;      /* Still no matching converter?! */
icculus@1982
  1327
            }
icculus@1982
  1328
        }
icculus@1982
  1329
icculus@1982
  1330
        /* Update (cvt) with filter details... */
icculus@1982
  1331
        cvt->filters[cvt->filter_index++] = filter;
icculus@1982
  1332
        if (src_bitsize < dst_bitsize) {
icculus@1982
  1333
            const int mult = (dst_bitsize / src_bitsize);
icculus@1982
  1334
            cvt->len_mult *= mult;
icculus@1982
  1335
            cvt->len_ratio *= mult;
icculus@1982
  1336
        } else if (src_bitsize > dst_bitsize) {
icculus@1982
  1337
            cvt->len_ratio /= (src_bitsize / dst_bitsize);
icculus@1982
  1338
        }
icculus@1982
  1339
slouken@1985
  1340
        return 1;               /* added a converter. */
icculus@1982
  1341
    }
icculus@1982
  1342
slouken@1985
  1343
    return 0;                   /* no conversion necessary. */
icculus@1982
  1344
}
icculus@1982
  1345
slouken@2716
  1346
/* Generate the necessary IIR lowpass coefficients for resampling.
slouken@2716
  1347
   Assume that the SDL_AudioCVT struct is already set up with
slouken@2716
  1348
   the correct values for len_mult and len_div, and use the
slouken@2716
  1349
   type of dst_format. Also assume the buffer is allocated.
slouken@2716
  1350
   Note the buffer needs to be 6 units long.
slouken@2716
  1351
   For now, use RBJ's cookbook coefficients. It might be more
slouken@2716
  1352
   optimal to create a Butterworth filter, but this is more difficult.
slouken@2716
  1353
*/
slouken@2760
  1354
#if 0
slouken@2716
  1355
int
slouken@2716
  1356
SDL_BuildIIRLowpass(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@2716
  1357
{
slouken@2716
  1358
    float fc;                   /* cutoff frequency */
slouken@2716
  1359
    float coeff[6];             /* floating point iir coefficients b0, b1, b2, a0, a1, a2 */
slouken@2716
  1360
    float scale;
slouken@2716
  1361
    float w0, alpha, cosw0;
slouken@2716
  1362
    int i;
slouken@2716
  1363
slouken@2716
  1364
    /* The higher Q is, the higher CUTOFF can be. Need to find a good balance to avoid aliasing */
slouken@2716
  1365
    static const float Q = 5.0f;
slouken@2716
  1366
    static const float CUTOFF = 0.4f;
slouken@2716
  1367
slouken@2716
  1368
    fc = (cvt->len_mult >
slouken@2716
  1369
          cvt->len_div) ? CUTOFF / (float) cvt->len_mult : CUTOFF /
slouken@2716
  1370
        (float) cvt->len_div;
slouken@2716
  1371
slouken@2716
  1372
    w0 = 2.0f * M_PI * fc;
slouken@2716
  1373
    cosw0 = cosf(w0);
slouken@2728
  1374
    alpha = sinf(w0) / (2.0f * Q);
slouken@2716
  1375
slouken@2716
  1376
    /* Compute coefficients, normalizing by a0 */
slouken@2716
  1377
    scale = 1.0f / (1.0f + alpha);
slouken@2716
  1378
slouken@2716
  1379
    coeff[0] = (1.0f - cosw0) / 2.0f * scale;
slouken@2716
  1380
    coeff[1] = (1.0f - cosw0) * scale;
slouken@2716
  1381
    coeff[2] = coeff[0];
slouken@2716
  1382
slouken@2716
  1383
    coeff[3] = 1.0f;            /* a0 is normalized to 1 */
slouken@2716
  1384
    coeff[4] = -2.0f * cosw0 * scale;
slouken@2716
  1385
    coeff[5] = (1.0f - alpha) * scale;
slouken@2716
  1386
slouken@2716
  1387
    /* Copy the coefficients to the struct. If necessary, convert coefficients to fixed point, using the range (-2.0, 2.0) */
slouken@2716
  1388
#define convert_fixed(type, fix) { \
slouken@2716
  1389
            type *cvt_coeff = (type *)cvt->coeff; \
slouken@2716
  1390
            for(i = 0; i < 6; ++i) { \
slouken@2716
  1391
                cvt_coeff[i] = fix(coeff[i]); \
slouken@2716
  1392
            } \
slouken@2716
  1393
        }
slouken@2716
  1394
slouken@2716
  1395
    if (SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
slouken@2716
  1396
        float *cvt_coeff = (float *) cvt->coeff;
slouken@2716
  1397
        for (i = 0; i < 6; ++i) {
slouken@2716
  1398
            cvt_coeff[i] = coeff[i];
slouken@2716
  1399
        }
slouken@2716
  1400
    } else {
slouken@2716
  1401
        switch (SDL_AUDIO_BITSIZE(format)) {
slouken@2716
  1402
        case 8:
slouken@2716
  1403
            convert_fixed(Uint8, SDL_Make_2_6);
slouken@2716
  1404
            break;
slouken@2716
  1405
        case 16:
slouken@2716
  1406
            convert_fixed(Uint16, SDL_Make_2_14);
slouken@2716
  1407
            break;
slouken@2716
  1408
        case 32:
slouken@2716
  1409
            convert_fixed(Uint32, SDL_Make_2_30);
slouken@2716
  1410
            break;
slouken@2716
  1411
        }
slouken@2716
  1412
    }
slouken@2716
  1413
slouken@2716
  1414
#ifdef DEBUG_CONVERT
slouken@2716
  1415
#define debug_iir(type) { \
slouken@2716
  1416
            type *cvt_coeff = (type *)cvt->coeff; \
slouken@2716
  1417
            for(i = 0; i < 6; ++i) { \
slouken@2716
  1418
                printf("coeff[%u] = %f = 0x%x\n", i, coeff[i], cvt_coeff[i]); \
slouken@2716
  1419
            } \
slouken@2716
  1420
        }
slouken@2716
  1421
    if (SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
slouken@2716
  1422
        float *cvt_coeff = (float *) cvt->coeff;
slouken@2716
  1423
        for (i = 0; i < 6; ++i) {
slouken@2716
  1424
            printf("coeff[%u] = %f = %f\n", i, coeff[i], cvt_coeff[i]);
slouken@2716
  1425
        }
slouken@2716
  1426
    } else {
slouken@2716
  1427
        switch (SDL_AUDIO_BITSIZE(format)) {
slouken@2716
  1428
        case 8:
slouken@2716
  1429
            debug_iir(Uint8);
slouken@2716
  1430
            break;
slouken@2716
  1431
        case 16:
slouken@2716
  1432
            debug_iir(Uint16);
slouken@2716
  1433
            break;
slouken@2716
  1434
        case 32:
slouken@2716
  1435
            debug_iir(Uint32);
slouken@2716
  1436
            break;
slouken@2716
  1437
        }
slouken@2716
  1438
    }
slouken@2716
  1439
#undef debug_iir
slouken@2716
  1440
#endif
slouken@2716
  1441
slouken@2716
  1442
    /* Initialize the state buffer to all zeroes, and set initial position */
slouken@2728
  1443
    SDL_memset(cvt->state_buf, 0, 4 * SDL_AUDIO_BITSIZE(format) / 4);
slouken@2716
  1444
    cvt->state_pos = 0;
slouken@2716
  1445
#undef convert_fixed
slouken@2760
  1446
slouken@2765
  1447
    return 0;
slouken@2716
  1448
}
slouken@2760
  1449
#endif
slouken@2716
  1450
slouken@2716
  1451
/* Apply the lowpass IIR filter to the given SDL_AudioCVT struct */
slouken@2716
  1452
/* This was implemented because it would be much faster than the fir filter, 
slouken@2716
  1453
   but it doesn't seem to have a steep enough cutoff so we'd need several
slouken@2716
  1454
   cascaded biquads, which probably isn't a great idea. Therefore, this
slouken@2716
  1455
   function can probably be discarded.
slouken@2716
  1456
*/
slouken@2716
  1457
static void
slouken@2716
  1458
SDL_FilterIIR(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@2716
  1459
{
slouken@2716
  1460
    Uint32 i, n;
slouken@2716
  1461
slouken@2716
  1462
    /* TODO: Check that n is calculated right */
slouken@2716
  1463
    n = 8 * cvt->len_cvt / SDL_AUDIO_BITSIZE(format);
slouken@2716
  1464
slouken@2716
  1465
    /* Note that the coefficients are 2_x and the input is 1_x. Do we need to shift left at the end here? The right shift temp = buf[n] >> 1 needs to depend on whether the type is signed or not for sign extension. */
slouken@2716
  1466
    /* cvt->state_pos = 1: state[0] = x_n-1, state[1] = x_n-2, state[2] = y_n-1, state[3] - y_n-2 */
slouken@2716
  1467
#define iir_fix(type, mult) {\
slouken@2716
  1468
            type *coeff = (type *)cvt->coeff; \
slouken@2716
  1469
            type *state = (type *)cvt->state_buf; \
slouken@2716
  1470
            type *buf = (type *)cvt->buf; \
slouken@2716
  1471
            type temp; \
slouken@2716
  1472
            for(i = 0; i < n; ++i) { \
slouken@2716
  1473
                    temp = buf[i] >> 1; \
slouken@2716
  1474
                    if(cvt->state_pos) { \
slouken@2716
  1475
                        buf[i] = mult(coeff[0], temp) + mult(coeff[1], state[0]) + mult(coeff[2], state[1]) - mult(coeff[4], state[2]) - mult(coeff[5], state[3]); \
slouken@2716
  1476
                        state[1] = temp; \
slouken@2716
  1477
                        state[3] = buf[i]; \
slouken@2716
  1478
                        cvt->state_pos = 0; \
slouken@2716
  1479
                    } else { \
slouken@2716
  1480
                        buf[i] = mult(coeff[0], temp) + mult(coeff[1], state[1]) + mult(coeff[2], state[0]) - mult(coeff[4], state[3]) - mult(coeff[5], state[2]); \
slouken@2716
  1481
                        state[0] = temp; \
slouken@2716
  1482
                        state[2] = buf[i]; \
slouken@2716
  1483
                        cvt->state_pos = 1; \
slouken@2716
  1484
                    } \
slouken@2716
  1485
                } \
slouken@2716
  1486
        }
slouken@2716
  1487
/* Need to test to see if the previous method or this one is faster */
slouken@2716
  1488
/*#define iir_fix(type, mult) {\
slouken@2716
  1489
            type *coeff = (type *)cvt->coeff; \
slouken@2716
  1490
            type *state = (type *)cvt->state_buf; \
slouken@2716
  1491
            type *buf = (type *)cvt->buf; \
slouken@2716
  1492
            type temp; \
slouken@2716
  1493
            for(i = 0; i < n; ++i) { \
slouken@2716
  1494
                    temp = buf[i] >> 1; \
slouken@2716
  1495
                    buf[i] = mult(coeff[0], temp) + mult(coeff[1], state[0]) + mult(coeff[2], state[1]) - mult(coeff[4], state[2]) - mult(coeff[5], state[3]); \
slouken@2716
  1496
                    state[1] = state[0]; \
slouken@2716
  1497
                    state[0] = temp; \
slouken@2716
  1498
                    state[3] = state[2]; \
slouken@2716
  1499
                    state[2] = buf[i]; \
slouken@2716
  1500
                } \
slouken@2716
  1501
        }*/
slouken@2716
  1502
slouken@2716
  1503
    if (SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
slouken@2716
  1504
        float *coeff = (float *) cvt->coeff;
slouken@2716
  1505
        float *state = (float *) cvt->state_buf;
slouken@2716
  1506
        float *buf = (float *) cvt->buf;
slouken@2716
  1507
        float temp;
slouken@2716
  1508
slouken@2716
  1509
        for (i = 0; i < n; ++i) {
slouken@2716
  1510
            /* y[n] = b0 * x[n] + b1 * x[n-1] + b2 * x[n-2] - a1 * y[n-1] - a[2] * y[n-2] */
slouken@2716
  1511
            temp = buf[i];
slouken@2716
  1512
            if (cvt->state_pos) {
slouken@2716
  1513
                buf[i] =
slouken@2716
  1514
                    coeff[0] * buf[n] + coeff[1] * state[0] +
slouken@2716
  1515
                    coeff[2] * state[1] - coeff[4] * state[2] -
slouken@2716
  1516
                    coeff[5] * state[3];
slouken@2716
  1517
                state[1] = temp;
slouken@2716
  1518
                state[3] = buf[i];
slouken@2716
  1519
                cvt->state_pos = 0;
slouken@2716
  1520
            } else {
slouken@2716
  1521
                buf[i] =
slouken@2716
  1522
                    coeff[0] * buf[n] + coeff[1] * state[1] +
slouken@2716
  1523
                    coeff[2] * state[0] - coeff[4] * state[3] -
slouken@2716
  1524
                    coeff[5] * state[2];
slouken@2716
  1525
                state[0] = temp;
slouken@2716
  1526
                state[2] = buf[i];
slouken@2716
  1527
                cvt->state_pos = 1;
slouken@2716
  1528
            }
slouken@2716
  1529
        }
slouken@2716
  1530
    } else {
slouken@2716
  1531
        /* Treat everything as signed! */
slouken@2716
  1532
        switch (SDL_AUDIO_BITSIZE(format)) {
slouken@2716
  1533
        case 8:
slouken@2716
  1534
            iir_fix(Sint8, SDL_FixMpy8);
slouken@2716
  1535
            break;
slouken@2716
  1536
        case 16:
slouken@2716
  1537
            iir_fix(Sint16, SDL_FixMpy16);
slouken@2716
  1538
            break;
slouken@2716
  1539
        case 32:
slouken@2716
  1540
            iir_fix(Sint32, SDL_FixMpy32);
slouken@2716
  1541
            break;
slouken@2716
  1542
        }
slouken@2716
  1543
    }
slouken@2716
  1544
#undef iir_fix
slouken@2716
  1545
}
slouken@2716
  1546
slouken@2716
  1547
/* Apply the windowed sinc FIR filter to the given SDL_AudioCVT struct.
slouken@2716
  1548
*/
slouken@2716
  1549
static void
slouken@2716
  1550
SDL_FilterFIR(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@2716
  1551
{
slouken@2716
  1552
    int n = 8 * cvt->len_cvt / SDL_AUDIO_BITSIZE(format);
slouken@2716
  1553
    int m = cvt->len_sinc;
slouken@2716
  1554
    int i, j;
slouken@2716
  1555
slouken@2716
  1556
    /* 
slouken@2716
  1557
       Note: We can make a big optimization here by taking advantage
slouken@2716
  1558
       of the fact that the signal is zero stuffed, so we can do
slouken@2716
  1559
       significantly fewer multiplications and additions. However, this
slouken@2716
  1560
       depends on the zero stuffing ratio, so it may not pay off. This would
slouken@2716
  1561
       basically be a polyphase filter.
slouken@2716
  1562
     */
slouken@2716
  1563
    /* One other way to do this fast is to look at the fir filter from a different angle:
slouken@2716
  1564
       After we zero stuff, we have input of all zeroes, except for every len_mult
slouken@2716
  1565
       sample. If we choose a sinc length equal to len_mult, then the fir filter becomes
slouken@2716
  1566
       much more simple: we're just taking a windowed sinc, shifting it to start at each
slouken@2716
  1567
       len_mult sample, and scaling it by the value of that sample. If we do this, then
slouken@2716
  1568
       we don't even need to worry about the sample histories, and the inner loop here is
slouken@2716
  1569
       unnecessary. This probably sacrifices some quality but could really speed things up as well.
slouken@2716
  1570
     */
slouken@2716
  1571
    /* We only calculate the values of samples which are 0 (mod len_div) because
slouken@2716
  1572
       those are the only ones used. All the other ones are discarded in the
slouken@2716
  1573
       third step of resampling. This is a huge speedup. As a warning, though,
slouken@2716
  1574
       if for some reason this is used elsewhere where there are no samples discarded,
slouken@2716
  1575
       the output will not be corrrect if len_div is not 1. To make this filter a
slouken@2716
  1576
       generic FIR filter, simply remove the if statement "if(i % cvt->len_div == 0)"
slouken@2716
  1577
       around the inner loop so that every sample is processed.
slouken@2716
  1578
     */
slouken@2716
  1579
    /* This is basically just a FIR filter. i.e. for input x_n and m coefficients,
slouken@2716
  1580
       y_n = x_n*sinc_0 + x_(n-1)*sinc_1 +  x_(n-2)*sinc_2 + ... + x_(n-m+1)*sinc_(m-1)
slouken@2716
  1581
     */
slouken@2716
  1582
#define filter_sinc(type, mult) { \
slouken@2716
  1583
            type *sinc = (type *)cvt->coeff; \
slouken@2716
  1584
            type *state = (type *)cvt->state_buf; \
slouken@2716
  1585
            type *buf = (type *)cvt->buf; \
slouken@2716
  1586
            for(i = 0; i < n; ++i) { \
slouken@2716
  1587
                state[cvt->state_pos] = buf[i]; \
slouken@2716
  1588
                buf[i] = 0; \
slouken@2716
  1589
                if( i % cvt->len_div == 0 ) { \
slouken@2716
  1590
                    for(j = 0; j < m;  ++j) { \
slouken@2716
  1591
                        buf[i] += mult(sinc[j], state[(cvt->state_pos + j) % m]); \
slouken@2716
  1592
                    } \
slouken@2716
  1593
                }\
slouken@2716
  1594
                cvt->state_pos = (cvt->state_pos + 1) % m; \
slouken@2716
  1595
            } \
slouken@2716
  1596
        }
slouken@2716
  1597
slouken@2716
  1598
    if (SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
slouken@2716
  1599
        filter_sinc(float, SDL_FloatMpy);
slouken@2716
  1600
    } else {
slouken@2716
  1601
        switch (SDL_AUDIO_BITSIZE(format)) {
slouken@2716
  1602
        case 8:
slouken@2716
  1603
            filter_sinc(Sint8, SDL_FixMpy8);
slouken@2716
  1604
            break;
slouken@2716
  1605
        case 16:
slouken@2716
  1606
            filter_sinc(Sint16, SDL_FixMpy16);
slouken@2716
  1607
            break;
slouken@2716
  1608
        case 32:
slouken@2716
  1609
            filter_sinc(Sint32, SDL_FixMpy32);
slouken@2716
  1610
            break;
slouken@2716
  1611
        }
slouken@2716
  1612
    }
slouken@2716
  1613
slouken@2716
  1614
#undef filter_sinc
slouken@2716
  1615
slouken@2716
  1616
}
slouken@2716
  1617
slouken@2716
  1618
/* Generate the necessary windowed sinc filter for resampling.
slouken@2716
  1619
   Assume that the SDL_AudioCVT struct is already set up with
slouken@2716
  1620
   the correct values for len_mult and len_div, and use the
slouken@2716
  1621
   type of dst_format. Also assume the buffer is allocated.
slouken@2716
  1622
   Note the buffer needs to be m+1 units long.
slouken@2716
  1623
*/
slouken@2716
  1624
int
slouken@2716
  1625
SDL_BuildWindowedSinc(SDL_AudioCVT * cvt, SDL_AudioFormat format,
slouken@2716
  1626
                      unsigned int m)
slouken@2716
  1627
{
slouken@2716
  1628
    float *fSinc;               /* floating point sinc buffer, to be converted to fixed point */
slouken@2716
  1629
    float fc;                   /* cutoff frequency */
slouken@2716
  1630
    float two_pi_fc, two_pi_over_m, four_pi_over_m, m_over_two;
slouken@2716
  1631
    float norm_sum, norm_fact;
slouken@2716
  1632
    unsigned int i;
slouken@2716
  1633
slouken@2716
  1634
    /* Check that the buffer is allocated */
slouken@2716
  1635
    if (cvt->coeff == NULL) {
slouken@2716
  1636
        return -1;
slouken@2716
  1637
    }
slouken@2716
  1638
slouken@2716
  1639
    /* Set the length */
slouken@2716
  1640
    cvt->len_sinc = m + 1;
slouken@2716
  1641
slouken@2716
  1642
    /* Allocate the floating point windowed sinc. */
slouken@2728
  1643
    fSinc = SDL_stack_alloc(float, (m + 1));
slouken@2716
  1644
    if (fSinc == NULL) {
slouken@2716
  1645
        return -1;
slouken@2716
  1646
    }
slouken@2716
  1647
slouken@2716
  1648
    /* Set up the filter parameters */
slouken@2716
  1649
    fc = (cvt->len_mult >
slouken@2716
  1650
          cvt->len_div) ? 0.5f / (float) cvt->len_mult : 0.5f /
slouken@2716
  1651
        (float) cvt->len_div;
slouken@2716
  1652
#ifdef DEBUG_CONVERT
slouken@2716
  1653
    printf("Lowpass cutoff frequency = %f\n", fc);
slouken@2716
  1654
#endif
slouken@2716
  1655
    two_pi_fc = 2.0f * M_PI * fc;
slouken@2716
  1656
    two_pi_over_m = 2.0f * M_PI / (float) m;
slouken@2716
  1657
    four_pi_over_m = 2.0f * two_pi_over_m;
slouken@2716
  1658
    m_over_two = (float) m / 2.0f;
slouken@2716
  1659
    norm_sum = 0.0f;
slouken@2716
  1660
slouken@2716
  1661
    for (i = 0; i <= m; ++i) {
slouken@2716
  1662
        if (i == m / 2) {
slouken@2716
  1663
            fSinc[i] = two_pi_fc;
slouken@2716
  1664
        } else {
slouken@2716
  1665
            fSinc[i] =
slouken@2716
  1666
                sinf(two_pi_fc * ((float) i - m_over_two)) / ((float) i -
slouken@2716
  1667
                                                              m_over_two);
slouken@2716
  1668
            /* Apply blackman window */
slouken@2716
  1669
            fSinc[i] *=
slouken@2716
  1670
                0.42f - 0.5f * cosf(two_pi_over_m * (float) i) +
slouken@2716
  1671
                0.08f * cosf(four_pi_over_m * (float) i);
slouken@2716
  1672
        }
slouken@2765
  1673
        norm_sum += fSinc[i] < 0 ? -fSinc[i] : fSinc[i];        /* fabs(fSinc[i]); */
slouken@2716
  1674
    }
slouken@2716
  1675
slouken@2716
  1676
    norm_fact = 1.0f / norm_sum;
slouken@2716
  1677
slouken@2716
  1678
#define convert_fixed(type, fix) { \
slouken@2716
  1679
        type *dst = (type *)cvt->coeff; \
slouken@2716
  1680
        for( i = 0; i <= m; ++i ) { \
slouken@2716
  1681
            dst[i] = fix(fSinc[i] * norm_fact); \
slouken@2716
  1682
        } \
slouken@2716
  1683
    }
slouken@2716
  1684
slouken@2716
  1685
    /* If we're using floating point, we only need to normalize */
slouken@2716
  1686
    if (SDL_AUDIO_ISFLOAT(format) && SDL_AUDIO_BITSIZE(format) == 32) {
slouken@2716
  1687
        float *fDest = (float *) cvt->coeff;
slouken@2716
  1688
        for (i = 0; i <= m; ++i) {
slouken@2716
  1689
            fDest[i] = fSinc[i] * norm_fact;
slouken@2716
  1690
        }
slouken@2716
  1691
    } else {
slouken@2716
  1692
        switch (SDL_AUDIO_BITSIZE(format)) {
slouken@2716
  1693
        case 8:
slouken@2716
  1694
            convert_fixed(Uint8, SDL_Make_1_7);
slouken@2716
  1695
            break;
slouken@2716
  1696
        case 16:
slouken@2716
  1697
            convert_fixed(Uint16, SDL_Make_1_15);
slouken@2716
  1698
            break;
slouken@2716
  1699
        case 32:
slouken@2716
  1700
            convert_fixed(Uint32, SDL_Make_1_31);
slouken@2716
  1701
            break;
slouken@2716
  1702
        }
slouken@2716
  1703
    }
slouken@2716
  1704
slouken@2716
  1705
    /* Initialize the state buffer to all zeroes, and set initial position */
slouken@2728
  1706
    SDL_memset(cvt->state_buf, 0,
slouken@2728
  1707
               cvt->len_sinc * SDL_AUDIO_BITSIZE(format) / 4);
slouken@2716
  1708
    cvt->state_pos = 0;
slouken@2716
  1709
slouken@2716
  1710
    /* Clean up */
slouken@2716
  1711
#undef convert_fixed
slouken@2728
  1712
    SDL_stack_free(fSinc);
slouken@2738
  1713
slouken@2738
  1714
    return 0;
slouken@2716
  1715
}
slouken@2716
  1716
slouken@2716
  1717
/* This is used to reduce the resampling ratio */
slouken@2728
  1718
static __inline__ int
slouken@2716
  1719
SDL_GCD(int a, int b)
slouken@2716
  1720
{
slouken@2716
  1721
    int temp;
slouken@2716
  1722
    while (b != 0) {
slouken@2716
  1723
        temp = a % b;
slouken@2716
  1724
        a = b;
slouken@2716
  1725
        b = temp;
slouken@2716
  1726
    }
slouken@2716
  1727
    return a;
slouken@2716
  1728
}
slouken@2716
  1729
slouken@2716
  1730
/* Perform proper resampling. This is pretty slow but it's the best-sounding method. */
slouken@2716
  1731
static void SDLCALL
slouken@2716
  1732
SDL_Resample(SDL_AudioCVT * cvt, SDL_AudioFormat format)
slouken@2716
  1733
{
slouken@2716
  1734
    int i, j;
slouken@2716
  1735
slouken@2716
  1736
#ifdef DEBUG_CONVERT
slouken@2716
  1737
    printf("Converting audio rate via proper resampling (mono)\n");
slouken@2716
  1738
#endif
slouken@2716
  1739
slouken@2716
  1740
#define zerostuff_mono(type) { \
slouken@2716
  1741
        const type *src = (const type *) (cvt->buf + cvt->len_cvt); \
slouken@2716
  1742
        type *dst = (type *) (cvt->buf + (cvt->len_cvt * cvt->len_mult)); \
slouken@2716
  1743
        for (i = cvt->len_cvt / sizeof (type); i; --i) { \
slouken@2716
  1744
            src--; \
slouken@2716
  1745
            dst[-1] = src[0]; \
slouken@2716
  1746
            for( j = -cvt->len_mult; j < -1; ++j ) { \
slouken@2716
  1747
                dst[j] = 0; \
slouken@2716
  1748
            } \
slouken@2716
  1749
            dst -= cvt->len_mult; \
slouken@2716
  1750
        } \
slouken@2716
  1751
    }
slouken@2716
  1752
slouken@2716
  1753
#define discard_mono(type) { \
slouken@2716
  1754
        const type *src = (const type *) (cvt->buf); \
slouken@2716
  1755
        type *dst = (type *) (cvt->buf); \
slouken@2716
  1756
        for (i = 0; i < (cvt->len_cvt / sizeof(type)) / cvt->len_div; ++i) { \
slouken@2716
  1757
            dst[0] = src[0]; \
slouken@2716
  1758
            src += cvt->len_div; \
slouken@2716
  1759
            ++dst; \
slouken@2716
  1760
        } \
slouken@2716
  1761
    }
slouken@2716
  1762
slouken@2716
  1763
    /* Step 1: Zero stuff the conversion buffer. This upsamples by a factor of len_mult,
slouken@2716
  1764
       creating aliasing at frequencies above the original nyquist frequency.
slouken@2716
  1765
     */
slouken@2716
  1766
#ifdef DEBUG_CONVERT
slouken@2716
  1767
    printf("Zero-stuffing by a factor of %u\n", cvt->len_mult);
slouken@2716
  1768
#endif
slouken@2716
  1769
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@2716
  1770
    case 8:
slouken@2716
  1771
        zerostuff_mono(Uint8);
slouken@2716
  1772
        break;
slouken@2716
  1773
    case 16:
slouken@2716
  1774
        zerostuff_mono(Uint16);
slouken@2716
  1775
        break;
slouken@2716
  1776
    case 32:
slouken@2716
  1777
        zerostuff_mono(Uint32);
slouken@2716
  1778
        break;
slouken@2716
  1779
    }
slouken@2716
  1780
slouken@2716
  1781
    cvt->len_cvt *= cvt->len_mult;
slouken@2716
  1782
slouken@2716
  1783
    /* Step 2: Use a windowed sinc FIR filter (lowpass filter) to remove the alias
slouken@2716
  1784
       frequencies. This is the slow part.
slouken@2716
  1785
     */
slouken@2716
  1786
    SDL_FilterFIR(cvt, format);
slouken@2716
  1787
slouken@2716
  1788
    /* Step 3: Now downsample by discarding samples. */
slouken@2716
  1789
slouken@2716
  1790
#ifdef DEBUG_CONVERT
slouken@2716
  1791
    printf("Discarding samples by a factor of %u\n", cvt->len_div);
slouken@2716
  1792
#endif
slouken@2716
  1793
    switch (SDL_AUDIO_BITSIZE(format)) {
slouken@2716
  1794
    case 8:
slouken@2716
  1795
        discard_mono(Uint8);
slouken@2716
  1796
        break;
slouken@2716
  1797
    case 16:
slouken@2716
  1798
        discard_mono(Uint16);
slouken@2716
  1799
        break;
slouken@2716
  1800
    case 32:
slouken@2716
  1801
        discard_mono(Uint32);
slouken@2716
  1802
        break;
slouken@2716
  1803
    }
slouken@2716
  1804
slouken@2716
  1805
#undef zerostuff_mono
slouken@2716
  1806
#undef discard_mono
slouken@2716
  1807
slouken@2716
  1808
    cvt->len_cvt /= cvt->len_div;
slouken@2716
  1809
slouken@2716
  1810
    if (cvt->filters[++cvt->filter_index]) {
slouken@2716
  1811
        cvt->filters[cvt->filter_index] (cvt, format);
slouken@2716
  1812
    }
slouken@2716
  1813
}
icculus@1982
  1814
icculus@1982
  1815
icculus@1982
  1816
/* Creates a set of audio filters to convert from one format to another.
icculus@1982
  1817
   Returns -1 if the format conversion is not supported, 0 if there's
icculus@1982
  1818
   no conversion needed, or 1 if the audio filter is set up.
slouken@0
  1819
*/
slouken@1895
  1820
slouken@1895
  1821
int
slouken@1895
  1822
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
icculus@1982
  1823
                  SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
icculus@1982
  1824
                  SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
slouken@0
  1825
{
icculus@1982
  1826
    /* there are no unsigned types over 16 bits, so catch this upfront. */
icculus@1982
  1827
    if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
icculus@1982
  1828
        return -1;
icculus@1982
  1829
    }
icculus@1982
  1830
    if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
icculus@1982
  1831
        return -1;
icculus@1982
  1832
    }
slouken@1985
  1833
#ifdef DEBUG_CONVERT
icculus@1982
  1834
    printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
slouken@1985
  1835
           src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
slouken@1985
  1836
#endif
icculus@1982
  1837
slouken@1895
  1838
    /* Start off with no conversion necessary */
icculus@1982
  1839
icculus@1982
  1840
    cvt->src_format = src_fmt;
icculus@1982
  1841
    cvt->dst_format = dst_fmt;
slouken@1895
  1842
    cvt->needed = 0;
slouken@1895
  1843
    cvt->filter_index = 0;
slouken@1895
  1844
    cvt->filters[0] = NULL;
slouken@1895
  1845
    cvt->len_mult = 1;
slouken@1895
  1846
    cvt->len_ratio = 1.0;
slouken@0
  1847
icculus@1982
  1848
    /* Convert data types, if necessary. Updates (cvt). */
icculus@1982
  1849
    if (SDL_BuildAudioTypeCVT(cvt, src_fmt, dst_fmt) == -1)
slouken@1985
  1850
        return -1;              /* shouldn't happen, but just in case... */
slouken@0
  1851
icculus@1982
  1852
    /* Channel conversion */
slouken@1895
  1853
    if (src_channels != dst_channels) {
slouken@1895
  1854
        if ((src_channels == 1) && (dst_channels > 1)) {
slouken@1895
  1855
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
slouken@1895
  1856
            cvt->len_mult *= 2;
slouken@1895
  1857
            src_channels = 2;
slouken@1895
  1858
            cvt->len_ratio *= 2;
slouken@1895
  1859
        }
slouken@1895
  1860
        if ((src_channels == 2) && (dst_channels == 6)) {
slouken@1895
  1861
            cvt->filters[cvt->filter_index++] = SDL_ConvertSurround;
slouken@1895
  1862
            src_channels = 6;
slouken@1895
  1863
            cvt->len_mult *= 3;
slouken@1895
  1864
            cvt->len_ratio *= 3;
slouken@1895
  1865
        }
slouken@1895
  1866
        if ((src_channels == 2) && (dst_channels == 4)) {
slouken@1895
  1867
            cvt->filters[cvt->filter_index++] = SDL_ConvertSurround_4;
slouken@1895
  1868
            src_channels = 4;
slouken@1895
  1869
            cvt->len_mult *= 2;
slouken@1895
  1870
            cvt->len_ratio *= 2;
slouken@1895
  1871
        }
slouken@1895
  1872
        while ((src_channels * 2) <= dst_channels) {
slouken@1895
  1873
            cvt->filters[cvt->filter_index++] = SDL_ConvertStereo;
slouken@1895
  1874
            cvt->len_mult *= 2;
slouken@1895
  1875
            src_channels *= 2;
slouken@1895
  1876
            cvt->len_ratio *= 2;
slouken@1895
  1877
        }
slouken@1895
  1878
        if ((src_channels == 6) && (dst_channels <= 2)) {
slouken@1895
  1879
            cvt->filters[cvt->filter_index++] = SDL_ConvertStrip;
slouken@1895
  1880
            src_channels = 2;
slouken@1895
  1881
            cvt->len_ratio /= 3;
slouken@1895
  1882
        }
slouken@1895
  1883
        if ((src_channels == 6) && (dst_channels == 4)) {
slouken@1895
  1884
            cvt->filters[cvt->filter_index++] = SDL_ConvertStrip_2;
slouken@1895
  1885
            src_channels = 4;
slouken@1895
  1886
            cvt->len_ratio /= 2;
slouken@1895
  1887
        }
slouken@1895
  1888
        /* This assumes that 4 channel audio is in the format:
slouken@1895
  1889
           Left {front/back} + Right {front/back}
slouken@1895
  1890
           so converting to L/R stereo works properly.
slouken@1895
  1891
         */
slouken@1895
  1892
        while (((src_channels % 2) == 0) &&
slouken@1895
  1893
               ((src_channels / 2) >= dst_channels)) {
slouken@1895
  1894
            cvt->filters[cvt->filter_index++] = SDL_ConvertMono;
slouken@1895
  1895
            src_channels /= 2;
slouken@1895
  1896
            cvt->len_ratio /= 2;
slouken@1895
  1897
        }
slouken@1895
  1898
        if (src_channels != dst_channels) {
slouken@1895
  1899
            /* Uh oh.. */ ;
slouken@1895
  1900
        }
slouken@1895
  1901
    }
slouken@0
  1902
slouken@1895
  1903
    /* Do rate conversion */
slouken@2716
  1904
    if (src_rate != dst_rate) {
slouken@2716
  1905
        int rate_gcd;
slouken@2716
  1906
        rate_gcd = SDL_GCD(src_rate, dst_rate);
slouken@2716
  1907
        cvt->len_mult = dst_rate / rate_gcd;
slouken@2716
  1908
        cvt->len_div = src_rate / rate_gcd;
slouken@2716
  1909
        cvt->len_ratio = (double) cvt->len_mult / (double) cvt->len_div;
slouken@2716
  1910
        cvt->filters[cvt->filter_index++] = SDL_Resample;
slouken@2716
  1911
        SDL_BuildWindowedSinc(cvt, dst_fmt, 768);
slouken@2716
  1912
    }
slouken@2716
  1913
slouken@2716
  1914
/*
slouken@1895
  1915
    cvt->rate_incr = 0.0;
slouken@1895
  1916
    if ((src_rate / 100) != (dst_rate / 100)) {
slouken@1895
  1917
        Uint32 hi_rate, lo_rate;
slouken@1895
  1918
        int len_mult;
slouken@1895
  1919
        double len_ratio;
icculus@1982
  1920
        SDL_AudioFilter rate_cvt = NULL;
slouken@1895
  1921
slouken@1895
  1922
        if (src_rate > dst_rate) {
slouken@1895
  1923
            hi_rate = src_rate;
slouken@1895
  1924
            lo_rate = dst_rate;
slouken@1895
  1925
            switch (src_channels) {
slouken@1895
  1926
            case 1:
slouken@1895
  1927
                rate_cvt = SDL_RateDIV2;
slouken@1895
  1928
                break;
slouken@1895
  1929
            case 2:
slouken@1895
  1930
                rate_cvt = SDL_RateDIV2_c2;
slouken@1895
  1931
                break;
slouken@1895
  1932
            case 4:
slouken@1895
  1933
                rate_cvt = SDL_RateDIV2_c4;
slouken@1895
  1934
                break;
slouken@1895
  1935
            case 6:
slouken@1895
  1936
                rate_cvt = SDL_RateDIV2_c6;
slouken@1895
  1937
                break;
slouken@1895
  1938
            default:
slouken@1895
  1939
                return -1;
slouken@1895
  1940
            }
slouken@1895
  1941
            len_mult = 1;
slouken@1895
  1942
            len_ratio = 0.5;
slouken@1895
  1943
        } else {
slouken@1895
  1944
            hi_rate = dst_rate;
slouken@1895
  1945
            lo_rate = src_rate;
slouken@1895
  1946
            switch (src_channels) {
slouken@1895
  1947
            case 1:
slouken@1895
  1948
                rate_cvt = SDL_RateMUL2;
slouken@1895
  1949
                break;
slouken@1895
  1950
            case 2:
slouken@1895
  1951
                rate_cvt = SDL_RateMUL2_c2;
slouken@1895
  1952
                break;
slouken@1895
  1953
            case 4:
slouken@1895
  1954
                rate_cvt = SDL_RateMUL2_c4;
slouken@1895
  1955
                break;
slouken@1895
  1956
            case 6:
slouken@1895
  1957
                rate_cvt = SDL_RateMUL2_c6;
slouken@1895
  1958
                break;
slouken@1895
  1959
            default:
slouken@1895
  1960
                return -1;
slouken@1895
  1961
            }
slouken@1895
  1962
            len_mult = 2;
slouken@1895
  1963
            len_ratio = 2.0;
slouken@2716
  1964
        }*/
slouken@2716
  1965
    /* If hi_rate = lo_rate*2^x then conversion is easy */
slouken@2716
  1966
    /*   while (((lo_rate * 2) / 100) <= (hi_rate / 100)) {
slouken@2716
  1967
       cvt->filters[cvt->filter_index++] = rate_cvt;
slouken@2716
  1968
       cvt->len_mult *= len_mult;
slouken@2716
  1969
       lo_rate *= 2;
slouken@2716
  1970
       cvt->len_ratio *= len_ratio;
slouken@2716
  1971
       } */
slouken@2716
  1972
    /* We may need a slow conversion here to finish up */
slouken@2716
  1973
    /*    if ((lo_rate / 100) != (hi_rate / 100)) {
slouken@2716
  1974
       #if 1 */
slouken@2716
  1975
    /* The problem with this is that if the input buffer is
slouken@2716
  1976
       say 1K, and the conversion rate is say 1.1, then the
slouken@2716
  1977
       output buffer is 1.1K, which may not be an acceptable
slouken@2716
  1978
       buffer size for the audio driver (not a power of 2)
slouken@2716
  1979
     */
slouken@2716
  1980
    /* For now, punt and hope the rate distortion isn't great.
slouken@2716
  1981
     */
slouken@2716
  1982
/*#else
slouken@1895
  1983
            if (src_rate < dst_rate) {
slouken@1895
  1984
                cvt->rate_incr = (double) lo_rate / hi_rate;
slouken@1895
  1985
                cvt->len_mult *= 2;
slouken@1895
  1986
                cvt->len_ratio /= cvt->rate_incr;
slouken@1895
  1987
            } else {
slouken@1895
  1988
                cvt->rate_incr = (double) hi_rate / lo_rate;
slouken@1895
  1989
                cvt->len_ratio *= cvt->rate_incr;
slouken@1895
  1990
            }
slouken@1895
  1991
            cvt->filters[cvt->filter_index++] = SDL_RateSLOW;
slouken@0
  1992
#endif
slouken@1895
  1993
        }
slouken@2716
  1994
    }*/
slouken@0
  1995
slouken@1895
  1996
    /* Set up the filter information */
slouken@1895
  1997
    if (cvt->filter_index != 0) {
slouken@1895
  1998
        cvt->needed = 1;
icculus@1982
  1999
        cvt->src_format = src_fmt;
icculus@1982
  2000
        cvt->dst_format = dst_fmt;
slouken@1895
  2001
        cvt->len = 0;
slouken@1895
  2002
        cvt->buf = NULL;
slouken@1895
  2003
        cvt->filters[cvt->filter_index] = NULL;
slouken@1895
  2004
    }
slouken@1895
  2005
    return (cvt->needed);
slouken@0
  2006
}
slouken@1895
  2007
slouken@2716
  2008
#undef SDL_FixMpy8
slouken@2716
  2009
#undef SDL_FixMpy16
slouken@2716
  2010
#undef SDL_FixMpy32
slouken@2716
  2011
#undef SDL_FloatMpy
slouken@2716
  2012
#undef SDL_Make_1_7
slouken@2716
  2013
#undef SDL_Make_1_15
slouken@2716
  2014
#undef SDL_Make_1_31
slouken@2716
  2015
#undef SDL_Make_2_6
slouken@2716
  2016
#undef SDL_Make_2_14
slouken@2716
  2017
#undef SDL_Make_2_30
slouken@2716
  2018
slouken@1895
  2019
/* vi: set ts=4 sw=4 expandtab: */