/* DO NOT EDIT! This file is generated by sdlgenaudiocvt.pl */ /* Simple DirectMedia Layer Copyright (C) 1997-2013 Sam Lantinga This software is provided 'as-is', without any express or implied warranty. In no event will the authors be held liable for any damages arising from the use of this software. Permission is granted to anyone to use this software for any purpose, including commercial applications, and to alter it and redistribute it freely, subject to the following restrictions: 1. The origin of this software must not be misrepresented; you must not claim that you wrote the original software. If you use this software in a product, an acknowledgment in the product documentation would be appreciated but is not required. 2. Altered source versions must be plainly marked as such, and must not be misrepresented as being the original software. 3. This notice may not be removed or altered from any source distribution. */ #include "SDL_config.h" #include "SDL_audio.h" #include "SDL_audio_c.h" #ifndef DEBUG_CONVERT #define DEBUG_CONVERT 0 #endif /* If you can guarantee your data and need space, you can eliminate code... */ /* Just build the arbitrary resamplers if you're saving code space. */ #ifndef LESS_RESAMPLERS #define LESS_RESAMPLERS 0 #endif /* Don't build any resamplers if you're REALLY saving code space. */ #ifndef NO_RESAMPLERS #define NO_RESAMPLERS 0 #endif /* Don't build any type converters if you're saving code space. */ #ifndef NO_CONVERTERS #define NO_CONVERTERS 0 #endif /* *INDENT-OFF* */ #define DIVBY127 0.0078740157480315f #define DIVBY32767 3.05185094759972e-05f #define DIVBY2147483647 4.6566128752458e-10f #if !NO_CONVERTERS static void SDLCALL SDL_Convert_U8_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_S8.\n"); #endif src = (const Uint8 *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, ++src, ++dst) { const Sint8 val = ((*src) ^ 0x80); *dst = ((Sint8) val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_U8_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_U16LSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Uint16 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Uint16 val = (((Uint16) *src) << 8); *dst = SDL_SwapLE16(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_U8_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_S16LSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint16 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Sint16 val = (((Sint16) ((*src) ^ 0x80)) << 8); *dst = ((Sint16) SDL_SwapLE16(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_U8_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_U16MSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Uint16 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Uint16 val = (((Uint16) *src) << 8); *dst = SDL_SwapBE16(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_U8_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_S16MSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint16 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Sint16 val = (((Sint16) ((*src) ^ 0x80)) << 8); *dst = ((Sint16) SDL_SwapBE16(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_U8_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_S32LSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 4)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((*src) ^ 0x80)) << 24); *dst = ((Sint32) SDL_SwapLE32(val)); } cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_U8_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_S32MSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 4)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((*src) ^ 0x80)) << 24); *dst = ((Sint32) SDL_SwapBE32(val)); } cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_U8_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_F32LSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const float val = ((((float) *src) * DIVBY127) - 1.0f); *dst = SDL_SwapFloatLE(val); } cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } static void SDLCALL SDL_Convert_U8_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U8 to AUDIO_F32MSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const float val = ((((float) *src) * DIVBY127) - 1.0f); *dst = SDL_SwapFloatBE(val); } cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_S8_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_U8.\n"); #endif src = (const Uint8 *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, ++src, ++dst) { const Uint8 val = ((((Sint8) *src)) ^ 0x80); *dst = val; } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_S8_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_U16LSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Uint16 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Uint16 val = (((Uint16) ((((Sint8) *src)) ^ 0x80)) << 8); *dst = SDL_SwapLE16(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_S8_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_S16LSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint16 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Sint16 val = (((Sint16) ((Sint8) *src)) << 8); *dst = ((Sint16) SDL_SwapLE16(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_S8_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_U16MSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Uint16 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Uint16 val = (((Uint16) ((((Sint8) *src)) ^ 0x80)) << 8); *dst = SDL_SwapBE16(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_S8_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_S16MSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint16 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Sint16 val = (((Sint16) ((Sint8) *src)) << 8); *dst = ((Sint16) SDL_SwapBE16(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_S8_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_S32LSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 4)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((Sint8) *src)) << 24); *dst = ((Sint32) SDL_SwapLE32(val)); } cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_S8_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_S32MSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 4)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((Sint8) *src)) << 24); *dst = ((Sint32) SDL_SwapBE32(val)); } cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_S8_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_F32LSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const float val = (((float) ((Sint8) *src)) * DIVBY127); *dst = SDL_SwapFloatLE(val); } cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } static void SDLCALL SDL_Convert_S8_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint8 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S8 to AUDIO_F32MSB.\n"); #endif src = ((const Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 4)) - 1; for (i = cvt->len_cvt / sizeof (Uint8); i; --i, --src, --dst) { const float val = (((float) ((Sint8) *src)) * DIVBY127); *dst = SDL_SwapFloatBE(val); } cvt->len_cvt *= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_U16LSB_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_U8.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint8 val = ((Uint8) (SDL_SwapLE16(*src) >> 8)); *dst = val; } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_U16LSB_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_S8.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint8 val = ((Sint8) (((SDL_SwapLE16(*src)) ^ 0x8000) >> 8)); *dst = ((Sint8) val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_U16LSB_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_S16LSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint16 val = ((SDL_SwapLE16(*src)) ^ 0x8000); *dst = ((Sint16) SDL_SwapLE16(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_U16LSB_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_U16MSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint16 val = SDL_SwapLE16(*src); *dst = SDL_SwapBE16(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_U16LSB_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_S16MSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint16 val = ((SDL_SwapLE16(*src)) ^ 0x8000); *dst = ((Sint16) SDL_SwapBE16(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_U16LSB_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_S32LSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((SDL_SwapLE16(*src)) ^ 0x8000)) << 16); *dst = ((Sint32) SDL_SwapLE32(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_U16LSB_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_S32MSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((SDL_SwapLE16(*src)) ^ 0x8000)) << 16); *dst = ((Sint32) SDL_SwapBE32(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_U16LSB_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_F32LSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const float val = ((((float) SDL_SwapLE16(*src)) * DIVBY32767) - 1.0f); *dst = SDL_SwapFloatLE(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } static void SDLCALL SDL_Convert_U16LSB_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16LSB to AUDIO_F32MSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const float val = ((((float) SDL_SwapLE16(*src)) * DIVBY32767) - 1.0f); *dst = SDL_SwapFloatBE(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_S16LSB_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_U8.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint8 val = ((Uint8) (((((Sint16) SDL_SwapLE16(*src))) ^ 0x8000) >> 8)); *dst = val; } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_S16LSB_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_S8.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint8 val = ((Sint8) (((Sint16) SDL_SwapLE16(*src)) >> 8)); *dst = ((Sint8) val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_S16LSB_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_U16LSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint16 val = ((((Sint16) SDL_SwapLE16(*src))) ^ 0x8000); *dst = SDL_SwapLE16(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_S16LSB_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_U16MSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint16 val = ((((Sint16) SDL_SwapLE16(*src))) ^ 0x8000); *dst = SDL_SwapBE16(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_S16LSB_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_S16MSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) SDL_SwapLE16(*src)); *dst = ((Sint16) SDL_SwapBE16(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_S16LSB_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_S32LSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((Sint16) SDL_SwapLE16(*src))) << 16); *dst = ((Sint32) SDL_SwapLE32(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_S16LSB_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_S32MSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((Sint16) SDL_SwapLE16(*src))) << 16); *dst = ((Sint32) SDL_SwapBE32(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_S16LSB_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_F32LSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const float val = (((float) ((Sint16) SDL_SwapLE16(*src))) * DIVBY32767); *dst = SDL_SwapFloatLE(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } static void SDLCALL SDL_Convert_S16LSB_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16LSB to AUDIO_F32MSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const float val = (((float) ((Sint16) SDL_SwapLE16(*src))) * DIVBY32767); *dst = SDL_SwapFloatBE(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_U16MSB_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_U8.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint8 val = ((Uint8) (SDL_SwapBE16(*src) >> 8)); *dst = val; } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_U16MSB_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_S8.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint8 val = ((Sint8) (((SDL_SwapBE16(*src)) ^ 0x8000) >> 8)); *dst = ((Sint8) val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_U16MSB_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_U16LSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint16 val = SDL_SwapBE16(*src); *dst = SDL_SwapLE16(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_U16MSB_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_S16LSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint16 val = ((SDL_SwapBE16(*src)) ^ 0x8000); *dst = ((Sint16) SDL_SwapLE16(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_U16MSB_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_S16MSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint16 val = ((SDL_SwapBE16(*src)) ^ 0x8000); *dst = ((Sint16) SDL_SwapBE16(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_U16MSB_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_S32LSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((SDL_SwapBE16(*src)) ^ 0x8000)) << 16); *dst = ((Sint32) SDL_SwapLE32(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_U16MSB_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_S32MSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((SDL_SwapBE16(*src)) ^ 0x8000)) << 16); *dst = ((Sint32) SDL_SwapBE32(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_U16MSB_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_F32LSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const float val = ((((float) SDL_SwapBE16(*src)) * DIVBY32767) - 1.0f); *dst = SDL_SwapFloatLE(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } static void SDLCALL SDL_Convert_U16MSB_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_U16MSB to AUDIO_F32MSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const float val = ((((float) SDL_SwapBE16(*src)) * DIVBY32767) - 1.0f); *dst = SDL_SwapFloatBE(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_S16MSB_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_U8.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint8 val = ((Uint8) (((((Sint16) SDL_SwapBE16(*src))) ^ 0x8000) >> 8)); *dst = val; } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_S16MSB_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_S8.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint8 val = ((Sint8) (((Sint16) SDL_SwapBE16(*src)) >> 8)); *dst = ((Sint8) val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_S16MSB_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_U16LSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint16 val = ((((Sint16) SDL_SwapBE16(*src))) ^ 0x8000); *dst = SDL_SwapLE16(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_S16MSB_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_S16LSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) SDL_SwapBE16(*src)); *dst = ((Sint16) SDL_SwapLE16(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_S16MSB_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_U16MSB.\n"); #endif src = (const Uint16 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, ++src, ++dst) { const Uint16 val = ((((Sint16) SDL_SwapBE16(*src))) ^ 0x8000); *dst = SDL_SwapBE16(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_S16MSB_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_S32LSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((Sint16) SDL_SwapBE16(*src))) << 16); *dst = ((Sint32) SDL_SwapLE32(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_S16MSB_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_S32MSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((Sint32 *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const Sint32 val = (((Sint32) ((Sint16) SDL_SwapBE16(*src))) << 16); *dst = ((Sint32) SDL_SwapBE32(val)); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_S16MSB_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_F32LSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const float val = (((float) ((Sint16) SDL_SwapBE16(*src))) * DIVBY32767); *dst = SDL_SwapFloatLE(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } static void SDLCALL SDL_Convert_S16MSB_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint16 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S16MSB to AUDIO_F32MSB.\n"); #endif src = ((const Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; dst = ((float *) (cvt->buf + cvt->len_cvt * 2)) - 1; for (i = cvt->len_cvt / sizeof (Uint16); i; --i, --src, --dst) { const float val = (((float) ((Sint16) SDL_SwapBE16(*src))) * DIVBY32767); *dst = SDL_SwapFloatBE(val); } cvt->len_cvt *= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_S32LSB_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_U8.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Uint8 val = ((Uint8) (((((Sint32) SDL_SwapLE32(*src))) ^ 0x80000000) >> 24)); *dst = val; } cvt->len_cvt /= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_S32LSB_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_S8.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Sint8 val = ((Sint8) (((Sint32) SDL_SwapLE32(*src)) >> 24)); *dst = ((Sint8) val); } cvt->len_cvt /= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_S32LSB_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_U16LSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Uint16 val = ((Uint16) (((((Sint32) SDL_SwapLE32(*src))) ^ 0x80000000) >> 16)); *dst = SDL_SwapLE16(val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_S32LSB_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_S16LSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) (((Sint32) SDL_SwapLE32(*src)) >> 16)); *dst = ((Sint16) SDL_SwapLE16(val)); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_S32LSB_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_U16MSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Uint16 val = ((Uint16) (((((Sint32) SDL_SwapLE32(*src))) ^ 0x80000000) >> 16)); *dst = SDL_SwapBE16(val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_S32LSB_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_S16MSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) (((Sint32) SDL_SwapLE32(*src)) >> 16)); *dst = ((Sint16) SDL_SwapBE16(val)); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_S32LSB_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_S32MSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Sint32 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Sint32 val = ((Sint32) SDL_SwapLE32(*src)); *dst = ((Sint32) SDL_SwapBE32(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_S32LSB_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_F32LSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (float *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const float val = (((float) ((Sint32) SDL_SwapLE32(*src))) * DIVBY2147483647); *dst = SDL_SwapFloatLE(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } static void SDLCALL SDL_Convert_S32LSB_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32LSB to AUDIO_F32MSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (float *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const float val = (((float) ((Sint32) SDL_SwapLE32(*src))) * DIVBY2147483647); *dst = SDL_SwapFloatBE(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_S32MSB_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_U8.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Uint8 val = ((Uint8) (((((Sint32) SDL_SwapBE32(*src))) ^ 0x80000000) >> 24)); *dst = val; } cvt->len_cvt /= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_S32MSB_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_S8.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Sint8 val = ((Sint8) (((Sint32) SDL_SwapBE32(*src)) >> 24)); *dst = ((Sint8) val); } cvt->len_cvt /= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_S32MSB_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_U16LSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Uint16 val = ((Uint16) (((((Sint32) SDL_SwapBE32(*src))) ^ 0x80000000) >> 16)); *dst = SDL_SwapLE16(val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_S32MSB_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_S16LSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) (((Sint32) SDL_SwapBE32(*src)) >> 16)); *dst = ((Sint16) SDL_SwapLE16(val)); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_S32MSB_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_U16MSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Uint16 val = ((Uint16) (((((Sint32) SDL_SwapBE32(*src))) ^ 0x80000000) >> 16)); *dst = SDL_SwapBE16(val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_S32MSB_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_S16MSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) (((Sint32) SDL_SwapBE32(*src)) >> 16)); *dst = ((Sint16) SDL_SwapBE16(val)); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_S32MSB_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_S32LSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (Sint32 *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const Sint32 val = ((Sint32) SDL_SwapBE32(*src)); *dst = ((Sint32) SDL_SwapLE32(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_S32MSB_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_F32LSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (float *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const float val = (((float) ((Sint32) SDL_SwapBE32(*src))) * DIVBY2147483647); *dst = SDL_SwapFloatLE(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } static void SDLCALL SDL_Convert_S32MSB_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const Uint32 *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_S32MSB to AUDIO_F32MSB.\n"); #endif src = (const Uint32 *) cvt->buf; dst = (float *) cvt->buf; for (i = cvt->len_cvt / sizeof (Uint32); i; --i, ++src, ++dst) { const float val = (((float) ((Sint32) SDL_SwapBE32(*src))) * DIVBY2147483647); *dst = SDL_SwapFloatBE(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_F32LSB_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_U8.\n"); #endif src = (const float *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Uint8 val = ((Uint8) ((SDL_SwapFloatLE(*src) + 1.0f) * 127.0f)); *dst = val; } cvt->len_cvt /= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_F32LSB_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_S8.\n"); #endif src = (const float *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint8 val = ((Sint8) (SDL_SwapFloatLE(*src) * 127.0f)); *dst = ((Sint8) val); } cvt->len_cvt /= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_F32LSB_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_U16LSB.\n"); #endif src = (const float *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Uint16 val = ((Uint16) ((SDL_SwapFloatLE(*src) + 1.0f) * 32767.0f)); *dst = SDL_SwapLE16(val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_F32LSB_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_S16LSB.\n"); #endif src = (const float *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) (SDL_SwapFloatLE(*src) * 32767.0f)); *dst = ((Sint16) SDL_SwapLE16(val)); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_F32LSB_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_U16MSB.\n"); #endif src = (const float *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Uint16 val = ((Uint16) ((SDL_SwapFloatLE(*src) + 1.0f) * 32767.0f)); *dst = SDL_SwapBE16(val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_F32LSB_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_S16MSB.\n"); #endif src = (const float *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) (SDL_SwapFloatLE(*src) * 32767.0f)); *dst = ((Sint16) SDL_SwapBE16(val)); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_F32LSB_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_S32LSB.\n"); #endif src = (const float *) cvt->buf; dst = (Sint32 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint32 val = ((Sint32) (SDL_SwapFloatLE(*src) * 2147483647.0)); *dst = ((Sint32) SDL_SwapLE32(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_F32LSB_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_S32MSB.\n"); #endif src = (const float *) cvt->buf; dst = (Sint32 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint32 val = ((Sint32) (SDL_SwapFloatLE(*src) * 2147483647.0)); *dst = ((Sint32) SDL_SwapBE32(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_F32LSB_to_F32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32LSB to AUDIO_F32MSB.\n"); #endif src = (const float *) cvt->buf; dst = (float *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const float val = SDL_SwapFloatLE(*src); *dst = SDL_SwapFloatBE(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32MSB); } } static void SDLCALL SDL_Convert_F32MSB_to_U8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Uint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_U8.\n"); #endif src = (const float *) cvt->buf; dst = (Uint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Uint8 val = ((Uint8) ((SDL_SwapFloatBE(*src) + 1.0f) * 127.0f)); *dst = val; } cvt->len_cvt /= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U8); } } static void SDLCALL SDL_Convert_F32MSB_to_S8(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint8 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_S8.\n"); #endif src = (const float *) cvt->buf; dst = (Sint8 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint8 val = ((Sint8) (SDL_SwapFloatBE(*src) * 127.0f)); *dst = ((Sint8) val); } cvt->len_cvt /= 4; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S8); } } static void SDLCALL SDL_Convert_F32MSB_to_U16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_U16LSB.\n"); #endif src = (const float *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Uint16 val = ((Uint16) ((SDL_SwapFloatBE(*src) + 1.0f) * 32767.0f)); *dst = SDL_SwapLE16(val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16LSB); } } static void SDLCALL SDL_Convert_F32MSB_to_S16LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_S16LSB.\n"); #endif src = (const float *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) (SDL_SwapFloatBE(*src) * 32767.0f)); *dst = ((Sint16) SDL_SwapLE16(val)); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16LSB); } } static void SDLCALL SDL_Convert_F32MSB_to_U16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Uint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_U16MSB.\n"); #endif src = (const float *) cvt->buf; dst = (Uint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Uint16 val = ((Uint16) ((SDL_SwapFloatBE(*src) + 1.0f) * 32767.0f)); *dst = SDL_SwapBE16(val); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_U16MSB); } } static void SDLCALL SDL_Convert_F32MSB_to_S16MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint16 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_S16MSB.\n"); #endif src = (const float *) cvt->buf; dst = (Sint16 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint16 val = ((Sint16) (SDL_SwapFloatBE(*src) * 32767.0f)); *dst = ((Sint16) SDL_SwapBE16(val)); } cvt->len_cvt /= 2; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S16MSB); } } static void SDLCALL SDL_Convert_F32MSB_to_S32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_S32LSB.\n"); #endif src = (const float *) cvt->buf; dst = (Sint32 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint32 val = ((Sint32) (SDL_SwapFloatBE(*src) * 2147483647.0)); *dst = ((Sint32) SDL_SwapLE32(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32LSB); } } static void SDLCALL SDL_Convert_F32MSB_to_S32MSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; Sint32 *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_S32MSB.\n"); #endif src = (const float *) cvt->buf; dst = (Sint32 *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const Sint32 val = ((Sint32) (SDL_SwapFloatBE(*src) * 2147483647.0)); *dst = ((Sint32) SDL_SwapBE32(val)); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_S32MSB); } } static void SDLCALL SDL_Convert_F32MSB_to_F32LSB(SDL_AudioCVT * cvt, SDL_AudioFormat format) { int i; const float *src; float *dst; #if DEBUG_CONVERT fprintf(stderr, "Converting AUDIO_F32MSB to AUDIO_F32LSB.\n"); #endif src = (const float *) cvt->buf; dst = (float *) cvt->buf; for (i = cvt->len_cvt / sizeof (float); i; --i, ++src, ++dst) { const float val = SDL_SwapFloatBE(*src); *dst = SDL_SwapFloatLE(val); } if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, AUDIO_F32LSB); } } #endif /* !NO_CONVERTERS */ const SDL_AudioTypeFilters sdl_audio_type_filters[] = { #if !NO_CONVERTERS { AUDIO_U8, AUDIO_S8, SDL_Convert_U8_to_S8 }, { AUDIO_U8, AUDIO_U16LSB, SDL_Convert_U8_to_U16LSB }, { AUDIO_U8, AUDIO_S16LSB, SDL_Convert_U8_to_S16LSB }, { AUDIO_U8, AUDIO_U16MSB, SDL_Convert_U8_to_U16MSB }, { AUDIO_U8, AUDIO_S16MSB, SDL_Convert_U8_to_S16MSB }, { AUDIO_U8, AUDIO_S32LSB, SDL_Convert_U8_to_S32LSB }, { AUDIO_U8, AUDIO_S32MSB, SDL_Convert_U8_to_S32MSB }, { AUDIO_U8, AUDIO_F32LSB, SDL_Convert_U8_to_F32LSB }, { AUDIO_U8, AUDIO_F32MSB, SDL_Convert_U8_to_F32MSB }, { AUDIO_S8, AUDIO_U8, SDL_Convert_S8_to_U8 }, { AUDIO_S8, AUDIO_U16LSB, SDL_Convert_S8_to_U16LSB }, { AUDIO_S8, AUDIO_S16LSB, SDL_Convert_S8_to_S16LSB }, { AUDIO_S8, AUDIO_U16MSB, SDL_Convert_S8_to_U16MSB }, { AUDIO_S8, AUDIO_S16MSB, SDL_Convert_S8_to_S16MSB }, { AUDIO_S8, AUDIO_S32LSB, SDL_Convert_S8_to_S32LSB }, { AUDIO_S8, AUDIO_S32MSB, SDL_Convert_S8_to_S32MSB }, { AUDIO_S8, AUDIO_F32LSB, SDL_Convert_S8_to_F32LSB }, { AUDIO_S8, AUDIO_F32MSB, SDL_Convert_S8_to_F32MSB }, { AUDIO_U16LSB, AUDIO_U8, SDL_Convert_U16LSB_to_U8 }, { AUDIO_U16LSB, AUDIO_S8, SDL_Convert_U16LSB_to_S8 }, { AUDIO_U16LSB, AUDIO_S16LSB, SDL_Convert_U16LSB_to_S16LSB }, { AUDIO_U16LSB, AUDIO_U16MSB, SDL_Convert_U16LSB_to_U16MSB }, { AUDIO_U16LSB, AUDIO_S16MSB, SDL_Convert_U16LSB_to_S16MSB }, { AUDIO_U16LSB, AUDIO_S32LSB, SDL_Convert_U16LSB_to_S32LSB }, { AUDIO_U16LSB, AUDIO_S32MSB, SDL_Convert_U16LSB_to_S32MSB }, { AUDIO_U16LSB, AUDIO_F32LSB, SDL_Convert_U16LSB_to_F32LSB }, { AUDIO_U16LSB, AUDIO_F32MSB, SDL_Convert_U16LSB_to_F32MSB }, { AUDIO_S16LSB, AUDIO_U8, SDL_Convert_S16LSB_to_U8 }, { AUDIO_S16LSB, AUDIO_S8, SDL_Convert_S16LSB_to_S8 }, { AUDIO_S16LSB, AUDIO_U16LSB, SDL_Convert_S16LSB_to_U16LSB }, { AUDIO_S16LSB, AUDIO_U16MSB, SDL_Convert_S16LSB_to_U16MSB }, { AUDIO_S16LSB, AUDIO_S16MSB, SDL_Convert_S16LSB_to_S16MSB }, { AUDIO_S16LSB, AUDIO_S32LSB, SDL_Convert_S16LSB_to_S32LSB }, { AUDIO_S16LSB, AUDIO_S32MSB, SDL_Convert_S16LSB_to_S32MSB }, { AUDIO_S16LSB, AUDIO_F32LSB, SDL_Convert_S16LSB_to_F32LSB }, { AUDIO_S16LSB, AUDIO_F32MSB, SDL_Convert_S16LSB_to_F32MSB }, { AUDIO_U16MSB, AUDIO_U8, SDL_Convert_U16MSB_to_U8 }, { AUDIO_U16MSB, AUDIO_S8, SDL_Convert_U16MSB_to_S8 }, { AUDIO_U16MSB, AUDIO_U16LSB, SDL_Convert_U16MSB_to_U16LSB }, { AUDIO_U16MSB, AUDIO_S16LSB, SDL_Convert_U16MSB_to_S16LSB }, { AUDIO_U16MSB, AUDIO_S16MSB, SDL_Convert_U16MSB_to_S16MSB }, { AUDIO_U16MSB, AUDIO_S32LSB, SDL_Convert_U16MSB_to_S32LSB }, { AUDIO_U16MSB, AUDIO_S32MSB, SDL_Convert_U16MSB_to_S32MSB }, { AUDIO_U16MSB, AUDIO_F32LSB, SDL_Convert_U16MSB_to_F32LSB }, { AUDIO_U16MSB, AUDIO_F32MSB, SDL_Convert_U16MSB_to_F32MSB }, { AUDIO_S16MSB, AUDIO_U8, SDL_Convert_S16MSB_to_U8 }, { AUDIO_S16MSB, AUDIO_S8, SDL_Convert_S16MSB_to_S8 }, { AUDIO_S16MSB, AUDIO_U16LSB, SDL_Convert_S16MSB_to_U16LSB }, { AUDIO_S16MSB, AUDIO_S16LSB, SDL_Convert_S16MSB_to_S16LSB }, { AUDIO_S16MSB, AUDIO_U16MSB, SDL_Convert_S16MSB_to_U16MSB }, { AUDIO_S16MSB, AUDIO_S32LSB, SDL_Convert_S16MSB_to_S32LSB }, { AUDIO_S16MSB, AUDIO_S32MSB, SDL_Convert_S16MSB_to_S32MSB }, { AUDIO_S16MSB, AUDIO_F32LSB, SDL_Convert_S16MSB_to_F32LSB }, { AUDIO_S16MSB, AUDIO_F32MSB, SDL_Convert_S16MSB_to_F32MSB }, { AUDIO_S32LSB, AUDIO_U8, SDL_Convert_S32LSB_to_U8 }, { AUDIO_S32LSB, AUDIO_S8, SDL_Convert_S32LSB_to_S8 }, { AUDIO_S32LSB, AUDIO_U16LSB, SDL_Convert_S32LSB_to_U16LSB }, { AUDIO_S32LSB, AUDIO_S16LSB, SDL_Convert_S32LSB_to_S16LSB }, { AUDIO_S32LSB, AUDIO_U16MSB, SDL_Convert_S32LSB_to_U16MSB }, { AUDIO_S32LSB, AUDIO_S16MSB, SDL_Convert_S32LSB_to_S16MSB }, { AUDIO_S32LSB, AUDIO_S32MSB, SDL_Convert_S32LSB_to_S32MSB }, { AUDIO_S32LSB, AUDIO_F32LSB, SDL_Convert_S32LSB_to_F32LSB }, { AUDIO_S32LSB, AUDIO_F32MSB, SDL_Convert_S32LSB_to_F32MSB }, { AUDIO_S32MSB, AUDIO_U8, SDL_Convert_S32MSB_to_U8 }, { AUDIO_S32MSB, AUDIO_S8, SDL_Convert_S32MSB_to_S8 }, { AUDIO_S32MSB, AUDIO_U16LSB, SDL_Convert_S32MSB_to_U16LSB }, { AUDIO_S32MSB, AUDIO_S16LSB, SDL_Convert_S32MSB_to_S16LSB }, { AUDIO_S32MSB, AUDIO_U16MSB, SDL_Convert_S32MSB_to_U16MSB }, { AUDIO_S32MSB, AUDIO_S16MSB, SDL_Convert_S32MSB_to_S16MSB }, { AUDIO_S32MSB, AUDIO_S32LSB, SDL_Convert_S32MSB_to_S32LSB }, { AUDIO_S32MSB, AUDIO_F32LSB, SDL_Convert_S32MSB_to_F32LSB }, { AUDIO_S32MSB, AUDIO_F32MSB, SDL_Convert_S32MSB_to_F32MSB }, { AUDIO_F32LSB, AUDIO_U8, SDL_Convert_F32LSB_to_U8 }, { AUDIO_F32LSB, AUDIO_S8, SDL_Convert_F32LSB_to_S8 }, { AUDIO_F32LSB, AUDIO_U16LSB, SDL_Convert_F32LSB_to_U16LSB }, { AUDIO_F32LSB, AUDIO_S16LSB, SDL_Convert_F32LSB_to_S16LSB }, { AUDIO_F32LSB, AUDIO_U16MSB, SDL_Convert_F32LSB_to_U16MSB }, { AUDIO_F32LSB, AUDIO_S16MSB, SDL_Convert_F32LSB_to_S16MSB }, { AUDIO_F32LSB, AUDIO_S32LSB, SDL_Convert_F32LSB_to_S32LSB }, { AUDIO_F32LSB, AUDIO_S32MSB, SDL_Convert_F32LSB_to_S32MSB }, { AUDIO_F32LSB, AUDIO_F32MSB, SDL_Convert_F32LSB_to_F32MSB }, { AUDIO_F32MSB, AUDIO_U8, SDL_Convert_F32MSB_to_U8 }, { AUDIO_F32MSB, AUDIO_S8, SDL_Convert_F32MSB_to_S8 }, { AUDIO_F32MSB, AUDIO_U16LSB, SDL_Convert_F32MSB_to_U16LSB }, { AUDIO_F32MSB, AUDIO_S16LSB, SDL_Convert_F32MSB_to_S16LSB }, { AUDIO_F32MSB, AUDIO_U16MSB, SDL_Convert_F32MSB_to_U16MSB }, { AUDIO_F32MSB, AUDIO_S16MSB, SDL_Convert_F32MSB_to_S16MSB }, { AUDIO_F32MSB, AUDIO_S32LSB, SDL_Convert_F32MSB_to_S32LSB }, { AUDIO_F32MSB, AUDIO_S32MSB, SDL_Convert_F32MSB_to_S32MSB }, { AUDIO_F32MSB, AUDIO_F32LSB, SDL_Convert_F32MSB_to_F32LSB }, #endif /* !NO_CONVERTERS */ { 0, 0, NULL } }; #if !NO_RESAMPLERS static void SDLCALL SDL_Upsample_U8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U8, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 16; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 1; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint8 *target = ((const Uint8 *) cvt->buf); Uint8 sample0 = src[0]; Uint8 last_sample0 = sample0; while (dst >= target) { dst[0] = sample0; dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U8, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 16; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Uint8 sample0 = src[0]; Uint8 last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = sample0; dst++; sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U8, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 2; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint8 *target = ((const Uint8 *) cvt->buf); Uint8 sample1 = src[1]; Uint8 sample0 = src[0]; Uint8 last_sample1 = sample1; Uint8 last_sample0 = sample0; while (dst >= target) { dst[1] = sample1; dst[0] = sample0; dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (Uint8) ((((Sint16) src[1]) + ((Sint16) last_sample1)) >> 1); sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U8, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Uint8 sample0 = src[0]; Uint8 sample1 = src[1]; Uint8 last_sample0 = sample0; Uint8 last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = sample0; dst[1] = sample1; dst += 2; sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); sample1 = (Uint8) ((((Sint16) src[1]) + ((Sint16) last_sample1)) >> 1); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U8, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 4; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint8 *target = ((const Uint8 *) cvt->buf); Uint8 sample3 = src[3]; Uint8 sample2 = src[2]; Uint8 sample1 = src[1]; Uint8 sample0 = src[0]; Uint8 last_sample3 = sample3; Uint8 last_sample2 = sample2; Uint8 last_sample1 = sample1; Uint8 last_sample0 = sample0; while (dst >= target) { dst[3] = sample3; dst[2] = sample2; dst[1] = sample1; dst[0] = sample0; dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (Uint8) ((((Sint16) src[3]) + ((Sint16) last_sample3)) >> 1); sample2 = (Uint8) ((((Sint16) src[2]) + ((Sint16) last_sample2)) >> 1); sample1 = (Uint8) ((((Sint16) src[1]) + ((Sint16) last_sample1)) >> 1); sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U8, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Uint8 sample0 = src[0]; Uint8 sample1 = src[1]; Uint8 sample2 = src[2]; Uint8 sample3 = src[3]; Uint8 last_sample0 = sample0; Uint8 last_sample1 = sample1; Uint8 last_sample2 = sample2; Uint8 last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = sample0; dst[1] = sample1; dst[2] = sample2; dst[3] = sample3; dst += 4; sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); sample1 = (Uint8) ((((Sint16) src[1]) + ((Sint16) last_sample1)) >> 1); sample2 = (Uint8) ((((Sint16) src[2]) + ((Sint16) last_sample2)) >> 1); sample3 = (Uint8) ((((Sint16) src[3]) + ((Sint16) last_sample3)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U8, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 96; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 6; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint8 *target = ((const Uint8 *) cvt->buf); Uint8 sample5 = src[5]; Uint8 sample4 = src[4]; Uint8 sample3 = src[3]; Uint8 sample2 = src[2]; Uint8 sample1 = src[1]; Uint8 sample0 = src[0]; Uint8 last_sample5 = sample5; Uint8 last_sample4 = sample4; Uint8 last_sample3 = sample3; Uint8 last_sample2 = sample2; Uint8 last_sample1 = sample1; Uint8 last_sample0 = sample0; while (dst >= target) { dst[5] = sample5; dst[4] = sample4; dst[3] = sample3; dst[2] = sample2; dst[1] = sample1; dst[0] = sample0; dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (Uint8) ((((Sint16) src[5]) + ((Sint16) last_sample5)) >> 1); sample4 = (Uint8) ((((Sint16) src[4]) + ((Sint16) last_sample4)) >> 1); sample3 = (Uint8) ((((Sint16) src[3]) + ((Sint16) last_sample3)) >> 1); sample2 = (Uint8) ((((Sint16) src[2]) + ((Sint16) last_sample2)) >> 1); sample1 = (Uint8) ((((Sint16) src[1]) + ((Sint16) last_sample1)) >> 1); sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U8, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 96; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Uint8 sample0 = src[0]; Uint8 sample1 = src[1]; Uint8 sample2 = src[2]; Uint8 sample3 = src[3]; Uint8 sample4 = src[4]; Uint8 sample5 = src[5]; Uint8 last_sample0 = sample0; Uint8 last_sample1 = sample1; Uint8 last_sample2 = sample2; Uint8 last_sample3 = sample3; Uint8 last_sample4 = sample4; Uint8 last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = sample0; dst[1] = sample1; dst[2] = sample2; dst[3] = sample3; dst[4] = sample4; dst[5] = sample5; dst += 6; sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); sample1 = (Uint8) ((((Sint16) src[1]) + ((Sint16) last_sample1)) >> 1); sample2 = (Uint8) ((((Sint16) src[2]) + ((Sint16) last_sample2)) >> 1); sample3 = (Uint8) ((((Sint16) src[3]) + ((Sint16) last_sample3)) >> 1); sample4 = (Uint8) ((((Sint16) src[4]) + ((Sint16) last_sample4)) >> 1); sample5 = (Uint8) ((((Sint16) src[5]) + ((Sint16) last_sample5)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U8, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 8; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint8 *target = ((const Uint8 *) cvt->buf); Uint8 sample7 = src[7]; Uint8 sample6 = src[6]; Uint8 sample5 = src[5]; Uint8 sample4 = src[4]; Uint8 sample3 = src[3]; Uint8 sample2 = src[2]; Uint8 sample1 = src[1]; Uint8 sample0 = src[0]; Uint8 last_sample7 = sample7; Uint8 last_sample6 = sample6; Uint8 last_sample5 = sample5; Uint8 last_sample4 = sample4; Uint8 last_sample3 = sample3; Uint8 last_sample2 = sample2; Uint8 last_sample1 = sample1; Uint8 last_sample0 = sample0; while (dst >= target) { dst[7] = sample7; dst[6] = sample6; dst[5] = sample5; dst[4] = sample4; dst[3] = sample3; dst[2] = sample2; dst[1] = sample1; dst[0] = sample0; dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (Uint8) ((((Sint16) src[7]) + ((Sint16) last_sample7)) >> 1); sample6 = (Uint8) ((((Sint16) src[6]) + ((Sint16) last_sample6)) >> 1); sample5 = (Uint8) ((((Sint16) src[5]) + ((Sint16) last_sample5)) >> 1); sample4 = (Uint8) ((((Sint16) src[4]) + ((Sint16) last_sample4)) >> 1); sample3 = (Uint8) ((((Sint16) src[3]) + ((Sint16) last_sample3)) >> 1); sample2 = (Uint8) ((((Sint16) src[2]) + ((Sint16) last_sample2)) >> 1); sample1 = (Uint8) ((((Sint16) src[1]) + ((Sint16) last_sample1)) >> 1); sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U8, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Uint8 sample0 = src[0]; Uint8 sample1 = src[1]; Uint8 sample2 = src[2]; Uint8 sample3 = src[3]; Uint8 sample4 = src[4]; Uint8 sample5 = src[5]; Uint8 sample6 = src[6]; Uint8 sample7 = src[7]; Uint8 last_sample0 = sample0; Uint8 last_sample1 = sample1; Uint8 last_sample2 = sample2; Uint8 last_sample3 = sample3; Uint8 last_sample4 = sample4; Uint8 last_sample5 = sample5; Uint8 last_sample6 = sample6; Uint8 last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = sample0; dst[1] = sample1; dst[2] = sample2; dst[3] = sample3; dst[4] = sample4; dst[5] = sample5; dst[6] = sample6; dst[7] = sample7; dst += 8; sample0 = (Uint8) ((((Sint16) src[0]) + ((Sint16) last_sample0)) >> 1); sample1 = (Uint8) ((((Sint16) src[1]) + ((Sint16) last_sample1)) >> 1); sample2 = (Uint8) ((((Sint16) src[2]) + ((Sint16) last_sample2)) >> 1); sample3 = (Uint8) ((((Sint16) src[3]) + ((Sint16) last_sample3)) >> 1); sample4 = (Uint8) ((((Sint16) src[4]) + ((Sint16) last_sample4)) >> 1); sample5 = (Uint8) ((((Sint16) src[5]) + ((Sint16) last_sample5)) >> 1); sample6 = (Uint8) ((((Sint16) src[6]) + ((Sint16) last_sample6)) >> 1); sample7 = (Uint8) ((((Sint16) src[7]) + ((Sint16) last_sample7)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S8, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 16; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 1; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint8 sample0 = ((Sint8) src[0]); Sint8 last_sample0 = sample0; while (dst >= target) { dst[0] = ((Sint8) sample0); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S8, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 16; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint8 sample0 = ((Sint8) src[0]); Sint8 last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint8) sample0); dst++; sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S8, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 2; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint8 sample1 = ((Sint8) src[1]); Sint8 sample0 = ((Sint8) src[0]); Sint8 last_sample1 = sample1; Sint8 last_sample0 = sample0; while (dst >= target) { dst[1] = ((Sint8) sample1); dst[0] = ((Sint8) sample0); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (Sint8) ((((Sint16) ((Sint8) src[1])) + ((Sint16) last_sample1)) >> 1); sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S8, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint8 sample0 = ((Sint8) src[0]); Sint8 sample1 = ((Sint8) src[1]); Sint8 last_sample0 = sample0; Sint8 last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint8) sample0); dst[1] = ((Sint8) sample1); dst += 2; sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); sample1 = (Sint8) ((((Sint16) ((Sint8) src[1])) + ((Sint16) last_sample1)) >> 1); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S8, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 4; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint8 sample3 = ((Sint8) src[3]); Sint8 sample2 = ((Sint8) src[2]); Sint8 sample1 = ((Sint8) src[1]); Sint8 sample0 = ((Sint8) src[0]); Sint8 last_sample3 = sample3; Sint8 last_sample2 = sample2; Sint8 last_sample1 = sample1; Sint8 last_sample0 = sample0; while (dst >= target) { dst[3] = ((Sint8) sample3); dst[2] = ((Sint8) sample2); dst[1] = ((Sint8) sample1); dst[0] = ((Sint8) sample0); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (Sint8) ((((Sint16) ((Sint8) src[3])) + ((Sint16) last_sample3)) >> 1); sample2 = (Sint8) ((((Sint16) ((Sint8) src[2])) + ((Sint16) last_sample2)) >> 1); sample1 = (Sint8) ((((Sint16) ((Sint8) src[1])) + ((Sint16) last_sample1)) >> 1); sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S8, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint8 sample0 = ((Sint8) src[0]); Sint8 sample1 = ((Sint8) src[1]); Sint8 sample2 = ((Sint8) src[2]); Sint8 sample3 = ((Sint8) src[3]); Sint8 last_sample0 = sample0; Sint8 last_sample1 = sample1; Sint8 last_sample2 = sample2; Sint8 last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint8) sample0); dst[1] = ((Sint8) sample1); dst[2] = ((Sint8) sample2); dst[3] = ((Sint8) sample3); dst += 4; sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); sample1 = (Sint8) ((((Sint16) ((Sint8) src[1])) + ((Sint16) last_sample1)) >> 1); sample2 = (Sint8) ((((Sint16) ((Sint8) src[2])) + ((Sint16) last_sample2)) >> 1); sample3 = (Sint8) ((((Sint16) ((Sint8) src[3])) + ((Sint16) last_sample3)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S8, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 96; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 6; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint8 sample5 = ((Sint8) src[5]); Sint8 sample4 = ((Sint8) src[4]); Sint8 sample3 = ((Sint8) src[3]); Sint8 sample2 = ((Sint8) src[2]); Sint8 sample1 = ((Sint8) src[1]); Sint8 sample0 = ((Sint8) src[0]); Sint8 last_sample5 = sample5; Sint8 last_sample4 = sample4; Sint8 last_sample3 = sample3; Sint8 last_sample2 = sample2; Sint8 last_sample1 = sample1; Sint8 last_sample0 = sample0; while (dst >= target) { dst[5] = ((Sint8) sample5); dst[4] = ((Sint8) sample4); dst[3] = ((Sint8) sample3); dst[2] = ((Sint8) sample2); dst[1] = ((Sint8) sample1); dst[0] = ((Sint8) sample0); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (Sint8) ((((Sint16) ((Sint8) src[5])) + ((Sint16) last_sample5)) >> 1); sample4 = (Sint8) ((((Sint16) ((Sint8) src[4])) + ((Sint16) last_sample4)) >> 1); sample3 = (Sint8) ((((Sint16) ((Sint8) src[3])) + ((Sint16) last_sample3)) >> 1); sample2 = (Sint8) ((((Sint16) ((Sint8) src[2])) + ((Sint16) last_sample2)) >> 1); sample1 = (Sint8) ((((Sint16) ((Sint8) src[1])) + ((Sint16) last_sample1)) >> 1); sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S8, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 96; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint8 sample0 = ((Sint8) src[0]); Sint8 sample1 = ((Sint8) src[1]); Sint8 sample2 = ((Sint8) src[2]); Sint8 sample3 = ((Sint8) src[3]); Sint8 sample4 = ((Sint8) src[4]); Sint8 sample5 = ((Sint8) src[5]); Sint8 last_sample0 = sample0; Sint8 last_sample1 = sample1; Sint8 last_sample2 = sample2; Sint8 last_sample3 = sample3; Sint8 last_sample4 = sample4; Sint8 last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint8) sample0); dst[1] = ((Sint8) sample1); dst[2] = ((Sint8) sample2); dst[3] = ((Sint8) sample3); dst[4] = ((Sint8) sample4); dst[5] = ((Sint8) sample5); dst += 6; sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); sample1 = (Sint8) ((((Sint16) ((Sint8) src[1])) + ((Sint16) last_sample1)) >> 1); sample2 = (Sint8) ((((Sint16) ((Sint8) src[2])) + ((Sint16) last_sample2)) >> 1); sample3 = (Sint8) ((((Sint16) ((Sint8) src[3])) + ((Sint16) last_sample3)) >> 1); sample4 = (Sint8) ((((Sint16) ((Sint8) src[4])) + ((Sint16) last_sample4)) >> 1); sample5 = (Sint8) ((((Sint16) ((Sint8) src[5])) + ((Sint16) last_sample5)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S8, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 8; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint8 sample7 = ((Sint8) src[7]); Sint8 sample6 = ((Sint8) src[6]); Sint8 sample5 = ((Sint8) src[5]); Sint8 sample4 = ((Sint8) src[4]); Sint8 sample3 = ((Sint8) src[3]); Sint8 sample2 = ((Sint8) src[2]); Sint8 sample1 = ((Sint8) src[1]); Sint8 sample0 = ((Sint8) src[0]); Sint8 last_sample7 = sample7; Sint8 last_sample6 = sample6; Sint8 last_sample5 = sample5; Sint8 last_sample4 = sample4; Sint8 last_sample3 = sample3; Sint8 last_sample2 = sample2; Sint8 last_sample1 = sample1; Sint8 last_sample0 = sample0; while (dst >= target) { dst[7] = ((Sint8) sample7); dst[6] = ((Sint8) sample6); dst[5] = ((Sint8) sample5); dst[4] = ((Sint8) sample4); dst[3] = ((Sint8) sample3); dst[2] = ((Sint8) sample2); dst[1] = ((Sint8) sample1); dst[0] = ((Sint8) sample0); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (Sint8) ((((Sint16) ((Sint8) src[7])) + ((Sint16) last_sample7)) >> 1); sample6 = (Sint8) ((((Sint16) ((Sint8) src[6])) + ((Sint16) last_sample6)) >> 1); sample5 = (Sint8) ((((Sint16) ((Sint8) src[5])) + ((Sint16) last_sample5)) >> 1); sample4 = (Sint8) ((((Sint16) ((Sint8) src[4])) + ((Sint16) last_sample4)) >> 1); sample3 = (Sint8) ((((Sint16) ((Sint8) src[3])) + ((Sint16) last_sample3)) >> 1); sample2 = (Sint8) ((((Sint16) ((Sint8) src[2])) + ((Sint16) last_sample2)) >> 1); sample1 = (Sint8) ((((Sint16) ((Sint8) src[1])) + ((Sint16) last_sample1)) >> 1); sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S8, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint8 sample0 = ((Sint8) src[0]); Sint8 sample1 = ((Sint8) src[1]); Sint8 sample2 = ((Sint8) src[2]); Sint8 sample3 = ((Sint8) src[3]); Sint8 sample4 = ((Sint8) src[4]); Sint8 sample5 = ((Sint8) src[5]); Sint8 sample6 = ((Sint8) src[6]); Sint8 sample7 = ((Sint8) src[7]); Sint8 last_sample0 = sample0; Sint8 last_sample1 = sample1; Sint8 last_sample2 = sample2; Sint8 last_sample3 = sample3; Sint8 last_sample4 = sample4; Sint8 last_sample5 = sample5; Sint8 last_sample6 = sample6; Sint8 last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint8) sample0); dst[1] = ((Sint8) sample1); dst[2] = ((Sint8) sample2); dst[3] = ((Sint8) sample3); dst[4] = ((Sint8) sample4); dst[5] = ((Sint8) sample5); dst[6] = ((Sint8) sample6); dst[7] = ((Sint8) sample7); dst += 8; sample0 = (Sint8) ((((Sint16) ((Sint8) src[0])) + ((Sint16) last_sample0)) >> 1); sample1 = (Sint8) ((((Sint16) ((Sint8) src[1])) + ((Sint16) last_sample1)) >> 1); sample2 = (Sint8) ((((Sint16) ((Sint8) src[2])) + ((Sint16) last_sample2)) >> 1); sample3 = (Sint8) ((((Sint16) ((Sint8) src[3])) + ((Sint16) last_sample3)) >> 1); sample4 = (Sint8) ((((Sint16) ((Sint8) src[4])) + ((Sint16) last_sample4)) >> 1); sample5 = (Sint8) ((((Sint16) ((Sint8) src[5])) + ((Sint16) last_sample5)) >> 1); sample6 = (Sint8) ((((Sint16) ((Sint8) src[6])) + ((Sint16) last_sample6)) >> 1); sample7 = (Sint8) ((((Sint16) ((Sint8) src[7])) + ((Sint16) last_sample7)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16LSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 last_sample0 = sample0; while (dst >= target) { dst[0] = SDL_SwapLE16(sample0); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16LSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapLE16(sample0); dst++; sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16LSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample1 = SDL_SwapLE16(src[1]); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 last_sample1 = sample1; Uint16 last_sample0 = sample0; while (dst >= target) { dst[1] = SDL_SwapLE16(sample1); dst[0] = SDL_SwapLE16(sample0); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (Uint16) ((((Sint32) SDL_SwapLE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16LSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 sample1 = SDL_SwapLE16(src[1]); Uint16 last_sample0 = sample0; Uint16 last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapLE16(sample0); dst[1] = SDL_SwapLE16(sample1); dst += 2; sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapLE16(src[1])) + ((Sint32) last_sample1)) >> 1); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16LSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample3 = SDL_SwapLE16(src[3]); Uint16 sample2 = SDL_SwapLE16(src[2]); Uint16 sample1 = SDL_SwapLE16(src[1]); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 last_sample3 = sample3; Uint16 last_sample2 = sample2; Uint16 last_sample1 = sample1; Uint16 last_sample0 = sample0; while (dst >= target) { dst[3] = SDL_SwapLE16(sample3); dst[2] = SDL_SwapLE16(sample2); dst[1] = SDL_SwapLE16(sample1); dst[0] = SDL_SwapLE16(sample0); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (Uint16) ((((Sint32) SDL_SwapLE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapLE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapLE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16LSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 sample1 = SDL_SwapLE16(src[1]); Uint16 sample2 = SDL_SwapLE16(src[2]); Uint16 sample3 = SDL_SwapLE16(src[3]); Uint16 last_sample0 = sample0; Uint16 last_sample1 = sample1; Uint16 last_sample2 = sample2; Uint16 last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapLE16(sample0); dst[1] = SDL_SwapLE16(sample1); dst[2] = SDL_SwapLE16(sample2); dst[3] = SDL_SwapLE16(sample3); dst += 4; sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapLE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapLE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapLE16(src[3])) + ((Sint32) last_sample3)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16LSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 192; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample5 = SDL_SwapLE16(src[5]); Uint16 sample4 = SDL_SwapLE16(src[4]); Uint16 sample3 = SDL_SwapLE16(src[3]); Uint16 sample2 = SDL_SwapLE16(src[2]); Uint16 sample1 = SDL_SwapLE16(src[1]); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 last_sample5 = sample5; Uint16 last_sample4 = sample4; Uint16 last_sample3 = sample3; Uint16 last_sample2 = sample2; Uint16 last_sample1 = sample1; Uint16 last_sample0 = sample0; while (dst >= target) { dst[5] = SDL_SwapLE16(sample5); dst[4] = SDL_SwapLE16(sample4); dst[3] = SDL_SwapLE16(sample3); dst[2] = SDL_SwapLE16(sample2); dst[1] = SDL_SwapLE16(sample1); dst[0] = SDL_SwapLE16(sample0); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (Uint16) ((((Sint32) SDL_SwapLE16(src[5])) + ((Sint32) last_sample5)) >> 1); sample4 = (Uint16) ((((Sint32) SDL_SwapLE16(src[4])) + ((Sint32) last_sample4)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapLE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapLE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapLE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16LSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 192; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 sample1 = SDL_SwapLE16(src[1]); Uint16 sample2 = SDL_SwapLE16(src[2]); Uint16 sample3 = SDL_SwapLE16(src[3]); Uint16 sample4 = SDL_SwapLE16(src[4]); Uint16 sample5 = SDL_SwapLE16(src[5]); Uint16 last_sample0 = sample0; Uint16 last_sample1 = sample1; Uint16 last_sample2 = sample2; Uint16 last_sample3 = sample3; Uint16 last_sample4 = sample4; Uint16 last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapLE16(sample0); dst[1] = SDL_SwapLE16(sample1); dst[2] = SDL_SwapLE16(sample2); dst[3] = SDL_SwapLE16(sample3); dst[4] = SDL_SwapLE16(sample4); dst[5] = SDL_SwapLE16(sample5); dst += 6; sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapLE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapLE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapLE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample4 = (Uint16) ((((Sint32) SDL_SwapLE16(src[4])) + ((Sint32) last_sample4)) >> 1); sample5 = (Uint16) ((((Sint32) SDL_SwapLE16(src[5])) + ((Sint32) last_sample5)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16LSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample7 = SDL_SwapLE16(src[7]); Uint16 sample6 = SDL_SwapLE16(src[6]); Uint16 sample5 = SDL_SwapLE16(src[5]); Uint16 sample4 = SDL_SwapLE16(src[4]); Uint16 sample3 = SDL_SwapLE16(src[3]); Uint16 sample2 = SDL_SwapLE16(src[2]); Uint16 sample1 = SDL_SwapLE16(src[1]); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 last_sample7 = sample7; Uint16 last_sample6 = sample6; Uint16 last_sample5 = sample5; Uint16 last_sample4 = sample4; Uint16 last_sample3 = sample3; Uint16 last_sample2 = sample2; Uint16 last_sample1 = sample1; Uint16 last_sample0 = sample0; while (dst >= target) { dst[7] = SDL_SwapLE16(sample7); dst[6] = SDL_SwapLE16(sample6); dst[5] = SDL_SwapLE16(sample5); dst[4] = SDL_SwapLE16(sample4); dst[3] = SDL_SwapLE16(sample3); dst[2] = SDL_SwapLE16(sample2); dst[1] = SDL_SwapLE16(sample1); dst[0] = SDL_SwapLE16(sample0); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (Uint16) ((((Sint32) SDL_SwapLE16(src[7])) + ((Sint32) last_sample7)) >> 1); sample6 = (Uint16) ((((Sint32) SDL_SwapLE16(src[6])) + ((Sint32) last_sample6)) >> 1); sample5 = (Uint16) ((((Sint32) SDL_SwapLE16(src[5])) + ((Sint32) last_sample5)) >> 1); sample4 = (Uint16) ((((Sint32) SDL_SwapLE16(src[4])) + ((Sint32) last_sample4)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapLE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapLE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapLE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16LSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapLE16(src[0]); Uint16 sample1 = SDL_SwapLE16(src[1]); Uint16 sample2 = SDL_SwapLE16(src[2]); Uint16 sample3 = SDL_SwapLE16(src[3]); Uint16 sample4 = SDL_SwapLE16(src[4]); Uint16 sample5 = SDL_SwapLE16(src[5]); Uint16 sample6 = SDL_SwapLE16(src[6]); Uint16 sample7 = SDL_SwapLE16(src[7]); Uint16 last_sample0 = sample0; Uint16 last_sample1 = sample1; Uint16 last_sample2 = sample2; Uint16 last_sample3 = sample3; Uint16 last_sample4 = sample4; Uint16 last_sample5 = sample5; Uint16 last_sample6 = sample6; Uint16 last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapLE16(sample0); dst[1] = SDL_SwapLE16(sample1); dst[2] = SDL_SwapLE16(sample2); dst[3] = SDL_SwapLE16(sample3); dst[4] = SDL_SwapLE16(sample4); dst[5] = SDL_SwapLE16(sample5); dst[6] = SDL_SwapLE16(sample6); dst[7] = SDL_SwapLE16(sample7); dst += 8; sample0 = (Uint16) ((((Sint32) SDL_SwapLE16(src[0])) + ((Sint32) last_sample0)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapLE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapLE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapLE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample4 = (Uint16) ((((Sint32) SDL_SwapLE16(src[4])) + ((Sint32) last_sample4)) >> 1); sample5 = (Uint16) ((((Sint32) SDL_SwapLE16(src[5])) + ((Sint32) last_sample5)) >> 1); sample6 = (Uint16) ((((Sint32) SDL_SwapLE16(src[6])) + ((Sint32) last_sample6)) >> 1); sample7 = (Uint16) ((((Sint32) SDL_SwapLE16(src[7])) + ((Sint32) last_sample7)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16LSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 last_sample0 = sample0; while (dst >= target) { dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16LSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst++; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16LSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample1 = ((Sint16) SDL_SwapLE16(src[1])); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 last_sample1 = sample1; Sint16 last_sample0 = sample0; while (dst >= target) { dst[1] = ((Sint16) SDL_SwapLE16(sample1)); dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16LSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 sample1 = ((Sint16) SDL_SwapLE16(src[1])); Sint16 last_sample0 = sample0; Sint16 last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst[1] = ((Sint16) SDL_SwapLE16(sample1)); dst += 2; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[1]))) + ((Sint32) last_sample1)) >> 1); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16LSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample3 = ((Sint16) SDL_SwapLE16(src[3])); Sint16 sample2 = ((Sint16) SDL_SwapLE16(src[2])); Sint16 sample1 = ((Sint16) SDL_SwapLE16(src[1])); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 last_sample3 = sample3; Sint16 last_sample2 = sample2; Sint16 last_sample1 = sample1; Sint16 last_sample0 = sample0; while (dst >= target) { dst[3] = ((Sint16) SDL_SwapLE16(sample3)); dst[2] = ((Sint16) SDL_SwapLE16(sample2)); dst[1] = ((Sint16) SDL_SwapLE16(sample1)); dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16LSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 sample1 = ((Sint16) SDL_SwapLE16(src[1])); Sint16 sample2 = ((Sint16) SDL_SwapLE16(src[2])); Sint16 sample3 = ((Sint16) SDL_SwapLE16(src[3])); Sint16 last_sample0 = sample0; Sint16 last_sample1 = sample1; Sint16 last_sample2 = sample2; Sint16 last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst[1] = ((Sint16) SDL_SwapLE16(sample1)); dst[2] = ((Sint16) SDL_SwapLE16(sample2)); dst[3] = ((Sint16) SDL_SwapLE16(sample3)); dst += 4; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[3]))) + ((Sint32) last_sample3)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16LSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 192; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample5 = ((Sint16) SDL_SwapLE16(src[5])); Sint16 sample4 = ((Sint16) SDL_SwapLE16(src[4])); Sint16 sample3 = ((Sint16) SDL_SwapLE16(src[3])); Sint16 sample2 = ((Sint16) SDL_SwapLE16(src[2])); Sint16 sample1 = ((Sint16) SDL_SwapLE16(src[1])); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 last_sample5 = sample5; Sint16 last_sample4 = sample4; Sint16 last_sample3 = sample3; Sint16 last_sample2 = sample2; Sint16 last_sample1 = sample1; Sint16 last_sample0 = sample0; while (dst >= target) { dst[5] = ((Sint16) SDL_SwapLE16(sample5)); dst[4] = ((Sint16) SDL_SwapLE16(sample4)); dst[3] = ((Sint16) SDL_SwapLE16(sample3)); dst[2] = ((Sint16) SDL_SwapLE16(sample2)); dst[1] = ((Sint16) SDL_SwapLE16(sample1)); dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[5]))) + ((Sint32) last_sample5)) >> 1); sample4 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[4]))) + ((Sint32) last_sample4)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16LSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 192; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 sample1 = ((Sint16) SDL_SwapLE16(src[1])); Sint16 sample2 = ((Sint16) SDL_SwapLE16(src[2])); Sint16 sample3 = ((Sint16) SDL_SwapLE16(src[3])); Sint16 sample4 = ((Sint16) SDL_SwapLE16(src[4])); Sint16 sample5 = ((Sint16) SDL_SwapLE16(src[5])); Sint16 last_sample0 = sample0; Sint16 last_sample1 = sample1; Sint16 last_sample2 = sample2; Sint16 last_sample3 = sample3; Sint16 last_sample4 = sample4; Sint16 last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst[1] = ((Sint16) SDL_SwapLE16(sample1)); dst[2] = ((Sint16) SDL_SwapLE16(sample2)); dst[3] = ((Sint16) SDL_SwapLE16(sample3)); dst[4] = ((Sint16) SDL_SwapLE16(sample4)); dst[5] = ((Sint16) SDL_SwapLE16(sample5)); dst += 6; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample4 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[4]))) + ((Sint32) last_sample4)) >> 1); sample5 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[5]))) + ((Sint32) last_sample5)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16LSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample7 = ((Sint16) SDL_SwapLE16(src[7])); Sint16 sample6 = ((Sint16) SDL_SwapLE16(src[6])); Sint16 sample5 = ((Sint16) SDL_SwapLE16(src[5])); Sint16 sample4 = ((Sint16) SDL_SwapLE16(src[4])); Sint16 sample3 = ((Sint16) SDL_SwapLE16(src[3])); Sint16 sample2 = ((Sint16) SDL_SwapLE16(src[2])); Sint16 sample1 = ((Sint16) SDL_SwapLE16(src[1])); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 last_sample7 = sample7; Sint16 last_sample6 = sample6; Sint16 last_sample5 = sample5; Sint16 last_sample4 = sample4; Sint16 last_sample3 = sample3; Sint16 last_sample2 = sample2; Sint16 last_sample1 = sample1; Sint16 last_sample0 = sample0; while (dst >= target) { dst[7] = ((Sint16) SDL_SwapLE16(sample7)); dst[6] = ((Sint16) SDL_SwapLE16(sample6)); dst[5] = ((Sint16) SDL_SwapLE16(sample5)); dst[4] = ((Sint16) SDL_SwapLE16(sample4)); dst[3] = ((Sint16) SDL_SwapLE16(sample3)); dst[2] = ((Sint16) SDL_SwapLE16(sample2)); dst[1] = ((Sint16) SDL_SwapLE16(sample1)); dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[7]))) + ((Sint32) last_sample7)) >> 1); sample6 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[6]))) + ((Sint32) last_sample6)) >> 1); sample5 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[5]))) + ((Sint32) last_sample5)) >> 1); sample4 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[4]))) + ((Sint32) last_sample4)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16LSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapLE16(src[0])); Sint16 sample1 = ((Sint16) SDL_SwapLE16(src[1])); Sint16 sample2 = ((Sint16) SDL_SwapLE16(src[2])); Sint16 sample3 = ((Sint16) SDL_SwapLE16(src[3])); Sint16 sample4 = ((Sint16) SDL_SwapLE16(src[4])); Sint16 sample5 = ((Sint16) SDL_SwapLE16(src[5])); Sint16 sample6 = ((Sint16) SDL_SwapLE16(src[6])); Sint16 sample7 = ((Sint16) SDL_SwapLE16(src[7])); Sint16 last_sample0 = sample0; Sint16 last_sample1 = sample1; Sint16 last_sample2 = sample2; Sint16 last_sample3 = sample3; Sint16 last_sample4 = sample4; Sint16 last_sample5 = sample5; Sint16 last_sample6 = sample6; Sint16 last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapLE16(sample0)); dst[1] = ((Sint16) SDL_SwapLE16(sample1)); dst[2] = ((Sint16) SDL_SwapLE16(sample2)); dst[3] = ((Sint16) SDL_SwapLE16(sample3)); dst[4] = ((Sint16) SDL_SwapLE16(sample4)); dst[5] = ((Sint16) SDL_SwapLE16(sample5)); dst[6] = ((Sint16) SDL_SwapLE16(sample6)); dst[7] = ((Sint16) SDL_SwapLE16(sample7)); dst += 8; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[0]))) + ((Sint32) last_sample0)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample4 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[4]))) + ((Sint32) last_sample4)) >> 1); sample5 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[5]))) + ((Sint32) last_sample5)) >> 1); sample6 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[6]))) + ((Sint32) last_sample6)) >> 1); sample7 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapLE16(src[7]))) + ((Sint32) last_sample7)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16MSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 last_sample0 = sample0; while (dst >= target) { dst[0] = SDL_SwapBE16(sample0); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16MSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapBE16(sample0); dst++; sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16MSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample1 = SDL_SwapBE16(src[1]); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 last_sample1 = sample1; Uint16 last_sample0 = sample0; while (dst >= target) { dst[1] = SDL_SwapBE16(sample1); dst[0] = SDL_SwapBE16(sample0); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (Uint16) ((((Sint32) SDL_SwapBE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16MSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 sample1 = SDL_SwapBE16(src[1]); Uint16 last_sample0 = sample0; Uint16 last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapBE16(sample0); dst[1] = SDL_SwapBE16(sample1); dst += 2; sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapBE16(src[1])) + ((Sint32) last_sample1)) >> 1); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16MSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample3 = SDL_SwapBE16(src[3]); Uint16 sample2 = SDL_SwapBE16(src[2]); Uint16 sample1 = SDL_SwapBE16(src[1]); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 last_sample3 = sample3; Uint16 last_sample2 = sample2; Uint16 last_sample1 = sample1; Uint16 last_sample0 = sample0; while (dst >= target) { dst[3] = SDL_SwapBE16(sample3); dst[2] = SDL_SwapBE16(sample2); dst[1] = SDL_SwapBE16(sample1); dst[0] = SDL_SwapBE16(sample0); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (Uint16) ((((Sint32) SDL_SwapBE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapBE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapBE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16MSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 sample1 = SDL_SwapBE16(src[1]); Uint16 sample2 = SDL_SwapBE16(src[2]); Uint16 sample3 = SDL_SwapBE16(src[3]); Uint16 last_sample0 = sample0; Uint16 last_sample1 = sample1; Uint16 last_sample2 = sample2; Uint16 last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapBE16(sample0); dst[1] = SDL_SwapBE16(sample1); dst[2] = SDL_SwapBE16(sample2); dst[3] = SDL_SwapBE16(sample3); dst += 4; sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapBE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapBE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapBE16(src[3])) + ((Sint32) last_sample3)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16MSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 192; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample5 = SDL_SwapBE16(src[5]); Uint16 sample4 = SDL_SwapBE16(src[4]); Uint16 sample3 = SDL_SwapBE16(src[3]); Uint16 sample2 = SDL_SwapBE16(src[2]); Uint16 sample1 = SDL_SwapBE16(src[1]); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 last_sample5 = sample5; Uint16 last_sample4 = sample4; Uint16 last_sample3 = sample3; Uint16 last_sample2 = sample2; Uint16 last_sample1 = sample1; Uint16 last_sample0 = sample0; while (dst >= target) { dst[5] = SDL_SwapBE16(sample5); dst[4] = SDL_SwapBE16(sample4); dst[3] = SDL_SwapBE16(sample3); dst[2] = SDL_SwapBE16(sample2); dst[1] = SDL_SwapBE16(sample1); dst[0] = SDL_SwapBE16(sample0); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (Uint16) ((((Sint32) SDL_SwapBE16(src[5])) + ((Sint32) last_sample5)) >> 1); sample4 = (Uint16) ((((Sint32) SDL_SwapBE16(src[4])) + ((Sint32) last_sample4)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapBE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapBE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapBE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16MSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 192; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 sample1 = SDL_SwapBE16(src[1]); Uint16 sample2 = SDL_SwapBE16(src[2]); Uint16 sample3 = SDL_SwapBE16(src[3]); Uint16 sample4 = SDL_SwapBE16(src[4]); Uint16 sample5 = SDL_SwapBE16(src[5]); Uint16 last_sample0 = sample0; Uint16 last_sample1 = sample1; Uint16 last_sample2 = sample2; Uint16 last_sample3 = sample3; Uint16 last_sample4 = sample4; Uint16 last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapBE16(sample0); dst[1] = SDL_SwapBE16(sample1); dst[2] = SDL_SwapBE16(sample2); dst[3] = SDL_SwapBE16(sample3); dst[4] = SDL_SwapBE16(sample4); dst[5] = SDL_SwapBE16(sample5); dst += 6; sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapBE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapBE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapBE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample4 = (Uint16) ((((Sint32) SDL_SwapBE16(src[4])) + ((Sint32) last_sample4)) >> 1); sample5 = (Uint16) ((((Sint32) SDL_SwapBE16(src[5])) + ((Sint32) last_sample5)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_U16MSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint16 *target = ((const Uint16 *) cvt->buf); Uint16 sample7 = SDL_SwapBE16(src[7]); Uint16 sample6 = SDL_SwapBE16(src[6]); Uint16 sample5 = SDL_SwapBE16(src[5]); Uint16 sample4 = SDL_SwapBE16(src[4]); Uint16 sample3 = SDL_SwapBE16(src[3]); Uint16 sample2 = SDL_SwapBE16(src[2]); Uint16 sample1 = SDL_SwapBE16(src[1]); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 last_sample7 = sample7; Uint16 last_sample6 = sample6; Uint16 last_sample5 = sample5; Uint16 last_sample4 = sample4; Uint16 last_sample3 = sample3; Uint16 last_sample2 = sample2; Uint16 last_sample1 = sample1; Uint16 last_sample0 = sample0; while (dst >= target) { dst[7] = SDL_SwapBE16(sample7); dst[6] = SDL_SwapBE16(sample6); dst[5] = SDL_SwapBE16(sample5); dst[4] = SDL_SwapBE16(sample4); dst[3] = SDL_SwapBE16(sample3); dst[2] = SDL_SwapBE16(sample2); dst[1] = SDL_SwapBE16(sample1); dst[0] = SDL_SwapBE16(sample0); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (Uint16) ((((Sint32) SDL_SwapBE16(src[7])) + ((Sint32) last_sample7)) >> 1); sample6 = (Uint16) ((((Sint32) SDL_SwapBE16(src[6])) + ((Sint32) last_sample6)) >> 1); sample5 = (Uint16) ((((Sint32) SDL_SwapBE16(src[5])) + ((Sint32) last_sample5)) >> 1); sample4 = (Uint16) ((((Sint32) SDL_SwapBE16(src[4])) + ((Sint32) last_sample4)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapBE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapBE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapBE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_U16MSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Uint16 sample0 = SDL_SwapBE16(src[0]); Uint16 sample1 = SDL_SwapBE16(src[1]); Uint16 sample2 = SDL_SwapBE16(src[2]); Uint16 sample3 = SDL_SwapBE16(src[3]); Uint16 sample4 = SDL_SwapBE16(src[4]); Uint16 sample5 = SDL_SwapBE16(src[5]); Uint16 sample6 = SDL_SwapBE16(src[6]); Uint16 sample7 = SDL_SwapBE16(src[7]); Uint16 last_sample0 = sample0; Uint16 last_sample1 = sample1; Uint16 last_sample2 = sample2; Uint16 last_sample3 = sample3; Uint16 last_sample4 = sample4; Uint16 last_sample5 = sample5; Uint16 last_sample6 = sample6; Uint16 last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapBE16(sample0); dst[1] = SDL_SwapBE16(sample1); dst[2] = SDL_SwapBE16(sample2); dst[3] = SDL_SwapBE16(sample3); dst[4] = SDL_SwapBE16(sample4); dst[5] = SDL_SwapBE16(sample5); dst[6] = SDL_SwapBE16(sample6); dst[7] = SDL_SwapBE16(sample7); dst += 8; sample0 = (Uint16) ((((Sint32) SDL_SwapBE16(src[0])) + ((Sint32) last_sample0)) >> 1); sample1 = (Uint16) ((((Sint32) SDL_SwapBE16(src[1])) + ((Sint32) last_sample1)) >> 1); sample2 = (Uint16) ((((Sint32) SDL_SwapBE16(src[2])) + ((Sint32) last_sample2)) >> 1); sample3 = (Uint16) ((((Sint32) SDL_SwapBE16(src[3])) + ((Sint32) last_sample3)) >> 1); sample4 = (Uint16) ((((Sint32) SDL_SwapBE16(src[4])) + ((Sint32) last_sample4)) >> 1); sample5 = (Uint16) ((((Sint32) SDL_SwapBE16(src[5])) + ((Sint32) last_sample5)) >> 1); sample6 = (Uint16) ((((Sint32) SDL_SwapBE16(src[6])) + ((Sint32) last_sample6)) >> 1); sample7 = (Uint16) ((((Sint32) SDL_SwapBE16(src[7])) + ((Sint32) last_sample7)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16MSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 last_sample0 = sample0; while (dst >= target) { dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16MSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 32; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst++; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16MSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample1 = ((Sint16) SDL_SwapBE16(src[1])); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 last_sample1 = sample1; Sint16 last_sample0 = sample0; while (dst >= target) { dst[1] = ((Sint16) SDL_SwapBE16(sample1)); dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16MSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 sample1 = ((Sint16) SDL_SwapBE16(src[1])); Sint16 last_sample0 = sample0; Sint16 last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst[1] = ((Sint16) SDL_SwapBE16(sample1)); dst += 2; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[1]))) + ((Sint32) last_sample1)) >> 1); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16MSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample3 = ((Sint16) SDL_SwapBE16(src[3])); Sint16 sample2 = ((Sint16) SDL_SwapBE16(src[2])); Sint16 sample1 = ((Sint16) SDL_SwapBE16(src[1])); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 last_sample3 = sample3; Sint16 last_sample2 = sample2; Sint16 last_sample1 = sample1; Sint16 last_sample0 = sample0; while (dst >= target) { dst[3] = ((Sint16) SDL_SwapBE16(sample3)); dst[2] = ((Sint16) SDL_SwapBE16(sample2)); dst[1] = ((Sint16) SDL_SwapBE16(sample1)); dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16MSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 sample1 = ((Sint16) SDL_SwapBE16(src[1])); Sint16 sample2 = ((Sint16) SDL_SwapBE16(src[2])); Sint16 sample3 = ((Sint16) SDL_SwapBE16(src[3])); Sint16 last_sample0 = sample0; Sint16 last_sample1 = sample1; Sint16 last_sample2 = sample2; Sint16 last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst[1] = ((Sint16) SDL_SwapBE16(sample1)); dst[2] = ((Sint16) SDL_SwapBE16(sample2)); dst[3] = ((Sint16) SDL_SwapBE16(sample3)); dst += 4; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[3]))) + ((Sint32) last_sample3)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16MSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 192; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample5 = ((Sint16) SDL_SwapBE16(src[5])); Sint16 sample4 = ((Sint16) SDL_SwapBE16(src[4])); Sint16 sample3 = ((Sint16) SDL_SwapBE16(src[3])); Sint16 sample2 = ((Sint16) SDL_SwapBE16(src[2])); Sint16 sample1 = ((Sint16) SDL_SwapBE16(src[1])); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 last_sample5 = sample5; Sint16 last_sample4 = sample4; Sint16 last_sample3 = sample3; Sint16 last_sample2 = sample2; Sint16 last_sample1 = sample1; Sint16 last_sample0 = sample0; while (dst >= target) { dst[5] = ((Sint16) SDL_SwapBE16(sample5)); dst[4] = ((Sint16) SDL_SwapBE16(sample4)); dst[3] = ((Sint16) SDL_SwapBE16(sample3)); dst[2] = ((Sint16) SDL_SwapBE16(sample2)); dst[1] = ((Sint16) SDL_SwapBE16(sample1)); dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[5]))) + ((Sint32) last_sample5)) >> 1); sample4 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[4]))) + ((Sint32) last_sample4)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16MSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 192; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 sample1 = ((Sint16) SDL_SwapBE16(src[1])); Sint16 sample2 = ((Sint16) SDL_SwapBE16(src[2])); Sint16 sample3 = ((Sint16) SDL_SwapBE16(src[3])); Sint16 sample4 = ((Sint16) SDL_SwapBE16(src[4])); Sint16 sample5 = ((Sint16) SDL_SwapBE16(src[5])); Sint16 last_sample0 = sample0; Sint16 last_sample1 = sample1; Sint16 last_sample2 = sample2; Sint16 last_sample3 = sample3; Sint16 last_sample4 = sample4; Sint16 last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst[1] = ((Sint16) SDL_SwapBE16(sample1)); dst[2] = ((Sint16) SDL_SwapBE16(sample2)); dst[3] = ((Sint16) SDL_SwapBE16(sample3)); dst[4] = ((Sint16) SDL_SwapBE16(sample4)); dst[5] = ((Sint16) SDL_SwapBE16(sample5)); dst += 6; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample4 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[4]))) + ((Sint32) last_sample4)) >> 1); sample5 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[5]))) + ((Sint32) last_sample5)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S16MSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint16 sample7 = ((Sint16) SDL_SwapBE16(src[7])); Sint16 sample6 = ((Sint16) SDL_SwapBE16(src[6])); Sint16 sample5 = ((Sint16) SDL_SwapBE16(src[5])); Sint16 sample4 = ((Sint16) SDL_SwapBE16(src[4])); Sint16 sample3 = ((Sint16) SDL_SwapBE16(src[3])); Sint16 sample2 = ((Sint16) SDL_SwapBE16(src[2])); Sint16 sample1 = ((Sint16) SDL_SwapBE16(src[1])); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 last_sample7 = sample7; Sint16 last_sample6 = sample6; Sint16 last_sample5 = sample5; Sint16 last_sample4 = sample4; Sint16 last_sample3 = sample3; Sint16 last_sample2 = sample2; Sint16 last_sample1 = sample1; Sint16 last_sample0 = sample0; while (dst >= target) { dst[7] = ((Sint16) SDL_SwapBE16(sample7)); dst[6] = ((Sint16) SDL_SwapBE16(sample6)); dst[5] = ((Sint16) SDL_SwapBE16(sample5)); dst[4] = ((Sint16) SDL_SwapBE16(sample4)); dst[3] = ((Sint16) SDL_SwapBE16(sample3)); dst[2] = ((Sint16) SDL_SwapBE16(sample2)); dst[1] = ((Sint16) SDL_SwapBE16(sample1)); dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[7]))) + ((Sint32) last_sample7)) >> 1); sample6 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[6]))) + ((Sint32) last_sample6)) >> 1); sample5 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[5]))) + ((Sint32) last_sample5)) >> 1); sample4 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[4]))) + ((Sint32) last_sample4)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S16MSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint16 sample0 = ((Sint16) SDL_SwapBE16(src[0])); Sint16 sample1 = ((Sint16) SDL_SwapBE16(src[1])); Sint16 sample2 = ((Sint16) SDL_SwapBE16(src[2])); Sint16 sample3 = ((Sint16) SDL_SwapBE16(src[3])); Sint16 sample4 = ((Sint16) SDL_SwapBE16(src[4])); Sint16 sample5 = ((Sint16) SDL_SwapBE16(src[5])); Sint16 sample6 = ((Sint16) SDL_SwapBE16(src[6])); Sint16 sample7 = ((Sint16) SDL_SwapBE16(src[7])); Sint16 last_sample0 = sample0; Sint16 last_sample1 = sample1; Sint16 last_sample2 = sample2; Sint16 last_sample3 = sample3; Sint16 last_sample4 = sample4; Sint16 last_sample5 = sample5; Sint16 last_sample6 = sample6; Sint16 last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint16) SDL_SwapBE16(sample0)); dst[1] = ((Sint16) SDL_SwapBE16(sample1)); dst[2] = ((Sint16) SDL_SwapBE16(sample2)); dst[3] = ((Sint16) SDL_SwapBE16(sample3)); dst[4] = ((Sint16) SDL_SwapBE16(sample4)); dst[5] = ((Sint16) SDL_SwapBE16(sample5)); dst[6] = ((Sint16) SDL_SwapBE16(sample6)); dst[7] = ((Sint16) SDL_SwapBE16(sample7)); dst += 8; sample0 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[0]))) + ((Sint32) last_sample0)) >> 1); sample1 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[1]))) + ((Sint32) last_sample1)) >> 1); sample2 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[2]))) + ((Sint32) last_sample2)) >> 1); sample3 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[3]))) + ((Sint32) last_sample3)) >> 1); sample4 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[4]))) + ((Sint32) last_sample4)) >> 1); sample5 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[5]))) + ((Sint32) last_sample5)) >> 1); sample6 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[6]))) + ((Sint32) last_sample6)) >> 1); sample7 = (Sint16) ((((Sint32) ((Sint16) SDL_SwapBE16(src[7]))) + ((Sint32) last_sample7)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32LSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 last_sample0 = sample0; while (dst >= target) { dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32LSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst++; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32LSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample1 = ((Sint32) SDL_SwapLE32(src[1])); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 last_sample1 = sample1; Sint32 last_sample0 = sample0; while (dst >= target) { dst[1] = ((Sint32) SDL_SwapLE32(sample1)); dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32LSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 sample1 = ((Sint32) SDL_SwapLE32(src[1])); Sint32 last_sample0 = sample0; Sint32 last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst[1] = ((Sint32) SDL_SwapLE32(sample1)); dst += 2; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[1]))) + ((Sint64) last_sample1)) >> 1); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32LSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample3 = ((Sint32) SDL_SwapLE32(src[3])); Sint32 sample2 = ((Sint32) SDL_SwapLE32(src[2])); Sint32 sample1 = ((Sint32) SDL_SwapLE32(src[1])); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 last_sample3 = sample3; Sint32 last_sample2 = sample2; Sint32 last_sample1 = sample1; Sint32 last_sample0 = sample0; while (dst >= target) { dst[3] = ((Sint32) SDL_SwapLE32(sample3)); dst[2] = ((Sint32) SDL_SwapLE32(sample2)); dst[1] = ((Sint32) SDL_SwapLE32(sample1)); dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32LSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 sample1 = ((Sint32) SDL_SwapLE32(src[1])); Sint32 sample2 = ((Sint32) SDL_SwapLE32(src[2])); Sint32 sample3 = ((Sint32) SDL_SwapLE32(src[3])); Sint32 last_sample0 = sample0; Sint32 last_sample1 = sample1; Sint32 last_sample2 = sample2; Sint32 last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst[1] = ((Sint32) SDL_SwapLE32(sample1)); dst[2] = ((Sint32) SDL_SwapLE32(sample2)); dst[3] = ((Sint32) SDL_SwapLE32(sample3)); dst += 4; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[3]))) + ((Sint64) last_sample3)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32LSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 384; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample5 = ((Sint32) SDL_SwapLE32(src[5])); Sint32 sample4 = ((Sint32) SDL_SwapLE32(src[4])); Sint32 sample3 = ((Sint32) SDL_SwapLE32(src[3])); Sint32 sample2 = ((Sint32) SDL_SwapLE32(src[2])); Sint32 sample1 = ((Sint32) SDL_SwapLE32(src[1])); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 last_sample5 = sample5; Sint32 last_sample4 = sample4; Sint32 last_sample3 = sample3; Sint32 last_sample2 = sample2; Sint32 last_sample1 = sample1; Sint32 last_sample0 = sample0; while (dst >= target) { dst[5] = ((Sint32) SDL_SwapLE32(sample5)); dst[4] = ((Sint32) SDL_SwapLE32(sample4)); dst[3] = ((Sint32) SDL_SwapLE32(sample3)); dst[2] = ((Sint32) SDL_SwapLE32(sample2)); dst[1] = ((Sint32) SDL_SwapLE32(sample1)); dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[5]))) + ((Sint64) last_sample5)) >> 1); sample4 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[4]))) + ((Sint64) last_sample4)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32LSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 384; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 sample1 = ((Sint32) SDL_SwapLE32(src[1])); Sint32 sample2 = ((Sint32) SDL_SwapLE32(src[2])); Sint32 sample3 = ((Sint32) SDL_SwapLE32(src[3])); Sint32 sample4 = ((Sint32) SDL_SwapLE32(src[4])); Sint32 sample5 = ((Sint32) SDL_SwapLE32(src[5])); Sint32 last_sample0 = sample0; Sint32 last_sample1 = sample1; Sint32 last_sample2 = sample2; Sint32 last_sample3 = sample3; Sint32 last_sample4 = sample4; Sint32 last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst[1] = ((Sint32) SDL_SwapLE32(sample1)); dst[2] = ((Sint32) SDL_SwapLE32(sample2)); dst[3] = ((Sint32) SDL_SwapLE32(sample3)); dst[4] = ((Sint32) SDL_SwapLE32(sample4)); dst[5] = ((Sint32) SDL_SwapLE32(sample5)); dst += 6; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample4 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[4]))) + ((Sint64) last_sample4)) >> 1); sample5 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[5]))) + ((Sint64) last_sample5)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32LSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 512; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample7 = ((Sint32) SDL_SwapLE32(src[7])); Sint32 sample6 = ((Sint32) SDL_SwapLE32(src[6])); Sint32 sample5 = ((Sint32) SDL_SwapLE32(src[5])); Sint32 sample4 = ((Sint32) SDL_SwapLE32(src[4])); Sint32 sample3 = ((Sint32) SDL_SwapLE32(src[3])); Sint32 sample2 = ((Sint32) SDL_SwapLE32(src[2])); Sint32 sample1 = ((Sint32) SDL_SwapLE32(src[1])); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 last_sample7 = sample7; Sint32 last_sample6 = sample6; Sint32 last_sample5 = sample5; Sint32 last_sample4 = sample4; Sint32 last_sample3 = sample3; Sint32 last_sample2 = sample2; Sint32 last_sample1 = sample1; Sint32 last_sample0 = sample0; while (dst >= target) { dst[7] = ((Sint32) SDL_SwapLE32(sample7)); dst[6] = ((Sint32) SDL_SwapLE32(sample6)); dst[5] = ((Sint32) SDL_SwapLE32(sample5)); dst[4] = ((Sint32) SDL_SwapLE32(sample4)); dst[3] = ((Sint32) SDL_SwapLE32(sample3)); dst[2] = ((Sint32) SDL_SwapLE32(sample2)); dst[1] = ((Sint32) SDL_SwapLE32(sample1)); dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[7]))) + ((Sint64) last_sample7)) >> 1); sample6 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[6]))) + ((Sint64) last_sample6)) >> 1); sample5 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[5]))) + ((Sint64) last_sample5)) >> 1); sample4 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[4]))) + ((Sint64) last_sample4)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32LSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 512; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapLE32(src[0])); Sint32 sample1 = ((Sint32) SDL_SwapLE32(src[1])); Sint32 sample2 = ((Sint32) SDL_SwapLE32(src[2])); Sint32 sample3 = ((Sint32) SDL_SwapLE32(src[3])); Sint32 sample4 = ((Sint32) SDL_SwapLE32(src[4])); Sint32 sample5 = ((Sint32) SDL_SwapLE32(src[5])); Sint32 sample6 = ((Sint32) SDL_SwapLE32(src[6])); Sint32 sample7 = ((Sint32) SDL_SwapLE32(src[7])); Sint32 last_sample0 = sample0; Sint32 last_sample1 = sample1; Sint32 last_sample2 = sample2; Sint32 last_sample3 = sample3; Sint32 last_sample4 = sample4; Sint32 last_sample5 = sample5; Sint32 last_sample6 = sample6; Sint32 last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapLE32(sample0)); dst[1] = ((Sint32) SDL_SwapLE32(sample1)); dst[2] = ((Sint32) SDL_SwapLE32(sample2)); dst[3] = ((Sint32) SDL_SwapLE32(sample3)); dst[4] = ((Sint32) SDL_SwapLE32(sample4)); dst[5] = ((Sint32) SDL_SwapLE32(sample5)); dst[6] = ((Sint32) SDL_SwapLE32(sample6)); dst[7] = ((Sint32) SDL_SwapLE32(sample7)); dst += 8; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[0]))) + ((Sint64) last_sample0)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample4 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[4]))) + ((Sint64) last_sample4)) >> 1); sample5 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[5]))) + ((Sint64) last_sample5)) >> 1); sample6 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[6]))) + ((Sint64) last_sample6)) >> 1); sample7 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapLE32(src[7]))) + ((Sint64) last_sample7)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32MSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 last_sample0 = sample0; while (dst >= target) { dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32MSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst++; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32MSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample1 = ((Sint32) SDL_SwapBE32(src[1])); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 last_sample1 = sample1; Sint32 last_sample0 = sample0; while (dst >= target) { dst[1] = ((Sint32) SDL_SwapBE32(sample1)); dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32MSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 sample1 = ((Sint32) SDL_SwapBE32(src[1])); Sint32 last_sample0 = sample0; Sint32 last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst[1] = ((Sint32) SDL_SwapBE32(sample1)); dst += 2; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[1]))) + ((Sint64) last_sample1)) >> 1); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32MSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample3 = ((Sint32) SDL_SwapBE32(src[3])); Sint32 sample2 = ((Sint32) SDL_SwapBE32(src[2])); Sint32 sample1 = ((Sint32) SDL_SwapBE32(src[1])); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 last_sample3 = sample3; Sint32 last_sample2 = sample2; Sint32 last_sample1 = sample1; Sint32 last_sample0 = sample0; while (dst >= target) { dst[3] = ((Sint32) SDL_SwapBE32(sample3)); dst[2] = ((Sint32) SDL_SwapBE32(sample2)); dst[1] = ((Sint32) SDL_SwapBE32(sample1)); dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32MSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 sample1 = ((Sint32) SDL_SwapBE32(src[1])); Sint32 sample2 = ((Sint32) SDL_SwapBE32(src[2])); Sint32 sample3 = ((Sint32) SDL_SwapBE32(src[3])); Sint32 last_sample0 = sample0; Sint32 last_sample1 = sample1; Sint32 last_sample2 = sample2; Sint32 last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst[1] = ((Sint32) SDL_SwapBE32(sample1)); dst[2] = ((Sint32) SDL_SwapBE32(sample2)); dst[3] = ((Sint32) SDL_SwapBE32(sample3)); dst += 4; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[3]))) + ((Sint64) last_sample3)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32MSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 384; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample5 = ((Sint32) SDL_SwapBE32(src[5])); Sint32 sample4 = ((Sint32) SDL_SwapBE32(src[4])); Sint32 sample3 = ((Sint32) SDL_SwapBE32(src[3])); Sint32 sample2 = ((Sint32) SDL_SwapBE32(src[2])); Sint32 sample1 = ((Sint32) SDL_SwapBE32(src[1])); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 last_sample5 = sample5; Sint32 last_sample4 = sample4; Sint32 last_sample3 = sample3; Sint32 last_sample2 = sample2; Sint32 last_sample1 = sample1; Sint32 last_sample0 = sample0; while (dst >= target) { dst[5] = ((Sint32) SDL_SwapBE32(sample5)); dst[4] = ((Sint32) SDL_SwapBE32(sample4)); dst[3] = ((Sint32) SDL_SwapBE32(sample3)); dst[2] = ((Sint32) SDL_SwapBE32(sample2)); dst[1] = ((Sint32) SDL_SwapBE32(sample1)); dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[5]))) + ((Sint64) last_sample5)) >> 1); sample4 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[4]))) + ((Sint64) last_sample4)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32MSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 384; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 sample1 = ((Sint32) SDL_SwapBE32(src[1])); Sint32 sample2 = ((Sint32) SDL_SwapBE32(src[2])); Sint32 sample3 = ((Sint32) SDL_SwapBE32(src[3])); Sint32 sample4 = ((Sint32) SDL_SwapBE32(src[4])); Sint32 sample5 = ((Sint32) SDL_SwapBE32(src[5])); Sint32 last_sample0 = sample0; Sint32 last_sample1 = sample1; Sint32 last_sample2 = sample2; Sint32 last_sample3 = sample3; Sint32 last_sample4 = sample4; Sint32 last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst[1] = ((Sint32) SDL_SwapBE32(sample1)); dst[2] = ((Sint32) SDL_SwapBE32(sample2)); dst[3] = ((Sint32) SDL_SwapBE32(sample3)); dst[4] = ((Sint32) SDL_SwapBE32(sample4)); dst[5] = ((Sint32) SDL_SwapBE32(sample5)); dst += 6; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample4 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[4]))) + ((Sint64) last_sample4)) >> 1); sample5 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[5]))) + ((Sint64) last_sample5)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_S32MSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 512; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint32 sample7 = ((Sint32) SDL_SwapBE32(src[7])); Sint32 sample6 = ((Sint32) SDL_SwapBE32(src[6])); Sint32 sample5 = ((Sint32) SDL_SwapBE32(src[5])); Sint32 sample4 = ((Sint32) SDL_SwapBE32(src[4])); Sint32 sample3 = ((Sint32) SDL_SwapBE32(src[3])); Sint32 sample2 = ((Sint32) SDL_SwapBE32(src[2])); Sint32 sample1 = ((Sint32) SDL_SwapBE32(src[1])); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 last_sample7 = sample7; Sint32 last_sample6 = sample6; Sint32 last_sample5 = sample5; Sint32 last_sample4 = sample4; Sint32 last_sample3 = sample3; Sint32 last_sample2 = sample2; Sint32 last_sample1 = sample1; Sint32 last_sample0 = sample0; while (dst >= target) { dst[7] = ((Sint32) SDL_SwapBE32(sample7)); dst[6] = ((Sint32) SDL_SwapBE32(sample6)); dst[5] = ((Sint32) SDL_SwapBE32(sample5)); dst[4] = ((Sint32) SDL_SwapBE32(sample4)); dst[3] = ((Sint32) SDL_SwapBE32(sample3)); dst[2] = ((Sint32) SDL_SwapBE32(sample2)); dst[1] = ((Sint32) SDL_SwapBE32(sample1)); dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[7]))) + ((Sint64) last_sample7)) >> 1); sample6 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[6]))) + ((Sint64) last_sample6)) >> 1); sample5 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[5]))) + ((Sint64) last_sample5)) >> 1); sample4 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[4]))) + ((Sint64) last_sample4)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_S32MSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 512; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint32 sample0 = ((Sint32) SDL_SwapBE32(src[0])); Sint32 sample1 = ((Sint32) SDL_SwapBE32(src[1])); Sint32 sample2 = ((Sint32) SDL_SwapBE32(src[2])); Sint32 sample3 = ((Sint32) SDL_SwapBE32(src[3])); Sint32 sample4 = ((Sint32) SDL_SwapBE32(src[4])); Sint32 sample5 = ((Sint32) SDL_SwapBE32(src[5])); Sint32 sample6 = ((Sint32) SDL_SwapBE32(src[6])); Sint32 sample7 = ((Sint32) SDL_SwapBE32(src[7])); Sint32 last_sample0 = sample0; Sint32 last_sample1 = sample1; Sint32 last_sample2 = sample2; Sint32 last_sample3 = sample3; Sint32 last_sample4 = sample4; Sint32 last_sample5 = sample5; Sint32 last_sample6 = sample6; Sint32 last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = ((Sint32) SDL_SwapBE32(sample0)); dst[1] = ((Sint32) SDL_SwapBE32(sample1)); dst[2] = ((Sint32) SDL_SwapBE32(sample2)); dst[3] = ((Sint32) SDL_SwapBE32(sample3)); dst[4] = ((Sint32) SDL_SwapBE32(sample4)); dst[5] = ((Sint32) SDL_SwapBE32(sample5)); dst[6] = ((Sint32) SDL_SwapBE32(sample6)); dst[7] = ((Sint32) SDL_SwapBE32(sample7)); dst += 8; sample0 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[0]))) + ((Sint64) last_sample0)) >> 1); sample1 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[1]))) + ((Sint64) last_sample1)) >> 1); sample2 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[2]))) + ((Sint64) last_sample2)) >> 1); sample3 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[3]))) + ((Sint64) last_sample3)) >> 1); sample4 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[4]))) + ((Sint64) last_sample4)) >> 1); sample5 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[5]))) + ((Sint64) last_sample5)) >> 1); sample6 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[6]))) + ((Sint64) last_sample6)) >> 1); sample7 = (Sint32) ((((Sint64) ((Sint32) SDL_SwapBE32(src[7]))) + ((Sint64) last_sample7)) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32LSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 1; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1; const float *target = ((const float *) cvt->buf); float sample0 = SDL_SwapFloatLE(src[0]); float last_sample0 = sample0; while (dst >= target) { dst[0] = SDL_SwapFloatLE(sample0); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32LSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatLE(src[0]); float last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatLE(sample0); dst++; sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32LSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2; const float *target = ((const float *) cvt->buf); float sample1 = SDL_SwapFloatLE(src[1]); float sample0 = SDL_SwapFloatLE(src[0]); float last_sample1 = sample1; float last_sample0 = sample0; while (dst >= target) { dst[1] = SDL_SwapFloatLE(sample1); dst[0] = SDL_SwapFloatLE(sample0); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (float) ((((double) SDL_SwapFloatLE(src[1])) + ((double) last_sample1)) * 0.5); sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32LSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatLE(src[0]); float sample1 = SDL_SwapFloatLE(src[1]); float last_sample0 = sample0; float last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatLE(sample0); dst[1] = SDL_SwapFloatLE(sample1); dst += 2; sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatLE(src[1])) + ((double) last_sample1)) * 0.5); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32LSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4; const float *target = ((const float *) cvt->buf); float sample3 = SDL_SwapFloatLE(src[3]); float sample2 = SDL_SwapFloatLE(src[2]); float sample1 = SDL_SwapFloatLE(src[1]); float sample0 = SDL_SwapFloatLE(src[0]); float last_sample3 = sample3; float last_sample2 = sample2; float last_sample1 = sample1; float last_sample0 = sample0; while (dst >= target) { dst[3] = SDL_SwapFloatLE(sample3); dst[2] = SDL_SwapFloatLE(sample2); dst[1] = SDL_SwapFloatLE(sample1); dst[0] = SDL_SwapFloatLE(sample0); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (float) ((((double) SDL_SwapFloatLE(src[3])) + ((double) last_sample3)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatLE(src[2])) + ((double) last_sample2)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatLE(src[1])) + ((double) last_sample1)) * 0.5); sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32LSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatLE(src[0]); float sample1 = SDL_SwapFloatLE(src[1]); float sample2 = SDL_SwapFloatLE(src[2]); float sample3 = SDL_SwapFloatLE(src[3]); float last_sample0 = sample0; float last_sample1 = sample1; float last_sample2 = sample2; float last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatLE(sample0); dst[1] = SDL_SwapFloatLE(sample1); dst[2] = SDL_SwapFloatLE(sample2); dst[3] = SDL_SwapFloatLE(sample3); dst += 4; sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatLE(src[1])) + ((double) last_sample1)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatLE(src[2])) + ((double) last_sample2)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatLE(src[3])) + ((double) last_sample3)) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32LSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 384; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 6; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6; const float *target = ((const float *) cvt->buf); float sample5 = SDL_SwapFloatLE(src[5]); float sample4 = SDL_SwapFloatLE(src[4]); float sample3 = SDL_SwapFloatLE(src[3]); float sample2 = SDL_SwapFloatLE(src[2]); float sample1 = SDL_SwapFloatLE(src[1]); float sample0 = SDL_SwapFloatLE(src[0]); float last_sample5 = sample5; float last_sample4 = sample4; float last_sample3 = sample3; float last_sample2 = sample2; float last_sample1 = sample1; float last_sample0 = sample0; while (dst >= target) { dst[5] = SDL_SwapFloatLE(sample5); dst[4] = SDL_SwapFloatLE(sample4); dst[3] = SDL_SwapFloatLE(sample3); dst[2] = SDL_SwapFloatLE(sample2); dst[1] = SDL_SwapFloatLE(sample1); dst[0] = SDL_SwapFloatLE(sample0); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (float) ((((double) SDL_SwapFloatLE(src[5])) + ((double) last_sample5)) * 0.5); sample4 = (float) ((((double) SDL_SwapFloatLE(src[4])) + ((double) last_sample4)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatLE(src[3])) + ((double) last_sample3)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatLE(src[2])) + ((double) last_sample2)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatLE(src[1])) + ((double) last_sample1)) * 0.5); sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32LSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 384; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatLE(src[0]); float sample1 = SDL_SwapFloatLE(src[1]); float sample2 = SDL_SwapFloatLE(src[2]); float sample3 = SDL_SwapFloatLE(src[3]); float sample4 = SDL_SwapFloatLE(src[4]); float sample5 = SDL_SwapFloatLE(src[5]); float last_sample0 = sample0; float last_sample1 = sample1; float last_sample2 = sample2; float last_sample3 = sample3; float last_sample4 = sample4; float last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatLE(sample0); dst[1] = SDL_SwapFloatLE(sample1); dst[2] = SDL_SwapFloatLE(sample2); dst[3] = SDL_SwapFloatLE(sample3); dst[4] = SDL_SwapFloatLE(sample4); dst[5] = SDL_SwapFloatLE(sample5); dst += 6; sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatLE(src[1])) + ((double) last_sample1)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatLE(src[2])) + ((double) last_sample2)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatLE(src[3])) + ((double) last_sample3)) * 0.5); sample4 = (float) ((((double) SDL_SwapFloatLE(src[4])) + ((double) last_sample4)) * 0.5); sample5 = (float) ((((double) SDL_SwapFloatLE(src[5])) + ((double) last_sample5)) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32LSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 512; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 8; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8; const float *target = ((const float *) cvt->buf); float sample7 = SDL_SwapFloatLE(src[7]); float sample6 = SDL_SwapFloatLE(src[6]); float sample5 = SDL_SwapFloatLE(src[5]); float sample4 = SDL_SwapFloatLE(src[4]); float sample3 = SDL_SwapFloatLE(src[3]); float sample2 = SDL_SwapFloatLE(src[2]); float sample1 = SDL_SwapFloatLE(src[1]); float sample0 = SDL_SwapFloatLE(src[0]); float last_sample7 = sample7; float last_sample6 = sample6; float last_sample5 = sample5; float last_sample4 = sample4; float last_sample3 = sample3; float last_sample2 = sample2; float last_sample1 = sample1; float last_sample0 = sample0; while (dst >= target) { dst[7] = SDL_SwapFloatLE(sample7); dst[6] = SDL_SwapFloatLE(sample6); dst[5] = SDL_SwapFloatLE(sample5); dst[4] = SDL_SwapFloatLE(sample4); dst[3] = SDL_SwapFloatLE(sample3); dst[2] = SDL_SwapFloatLE(sample2); dst[1] = SDL_SwapFloatLE(sample1); dst[0] = SDL_SwapFloatLE(sample0); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (float) ((((double) SDL_SwapFloatLE(src[7])) + ((double) last_sample7)) * 0.5); sample6 = (float) ((((double) SDL_SwapFloatLE(src[6])) + ((double) last_sample6)) * 0.5); sample5 = (float) ((((double) SDL_SwapFloatLE(src[5])) + ((double) last_sample5)) * 0.5); sample4 = (float) ((((double) SDL_SwapFloatLE(src[4])) + ((double) last_sample4)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatLE(src[3])) + ((double) last_sample3)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatLE(src[2])) + ((double) last_sample2)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatLE(src[1])) + ((double) last_sample1)) * 0.5); sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32LSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 512; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatLE(src[0]); float sample1 = SDL_SwapFloatLE(src[1]); float sample2 = SDL_SwapFloatLE(src[2]); float sample3 = SDL_SwapFloatLE(src[3]); float sample4 = SDL_SwapFloatLE(src[4]); float sample5 = SDL_SwapFloatLE(src[5]); float sample6 = SDL_SwapFloatLE(src[6]); float sample7 = SDL_SwapFloatLE(src[7]); float last_sample0 = sample0; float last_sample1 = sample1; float last_sample2 = sample2; float last_sample3 = sample3; float last_sample4 = sample4; float last_sample5 = sample5; float last_sample6 = sample6; float last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatLE(sample0); dst[1] = SDL_SwapFloatLE(sample1); dst[2] = SDL_SwapFloatLE(sample2); dst[3] = SDL_SwapFloatLE(sample3); dst[4] = SDL_SwapFloatLE(sample4); dst[5] = SDL_SwapFloatLE(sample5); dst[6] = SDL_SwapFloatLE(sample6); dst[7] = SDL_SwapFloatLE(sample7); dst += 8; sample0 = (float) ((((double) SDL_SwapFloatLE(src[0])) + ((double) last_sample0)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatLE(src[1])) + ((double) last_sample1)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatLE(src[2])) + ((double) last_sample2)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatLE(src[3])) + ((double) last_sample3)) * 0.5); sample4 = (float) ((((double) SDL_SwapFloatLE(src[4])) + ((double) last_sample4)) * 0.5); sample5 = (float) ((((double) SDL_SwapFloatLE(src[5])) + ((double) last_sample5)) * 0.5); sample6 = (float) ((((double) SDL_SwapFloatLE(src[6])) + ((double) last_sample6)) * 0.5); sample7 = (float) ((((double) SDL_SwapFloatLE(src[7])) + ((double) last_sample7)) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32MSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 1; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1; const float *target = ((const float *) cvt->buf); float sample0 = SDL_SwapFloatBE(src[0]); float last_sample0 = sample0; while (dst >= target) { dst[0] = SDL_SwapFloatBE(sample0); dst--; eps += srcsize; if ((eps << 1) >= dstsize) { src--; sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32MSB, 1 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 64; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatBE(src[0]); float last_sample0 = sample0; while (dst < target) { src++; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatBE(sample0); dst++; sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); last_sample0 = sample0; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32MSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2; const float *target = ((const float *) cvt->buf); float sample1 = SDL_SwapFloatBE(src[1]); float sample0 = SDL_SwapFloatBE(src[0]); float last_sample1 = sample1; float last_sample0 = sample0; while (dst >= target) { dst[1] = SDL_SwapFloatBE(sample1); dst[0] = SDL_SwapFloatBE(sample0); dst -= 2; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 2; sample1 = (float) ((((double) SDL_SwapFloatBE(src[1])) + ((double) last_sample1)) * 0.5); sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32MSB, 2 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 128; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatBE(src[0]); float sample1 = SDL_SwapFloatBE(src[1]); float last_sample0 = sample0; float last_sample1 = sample1; while (dst < target) { src += 2; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatBE(sample0); dst[1] = SDL_SwapFloatBE(sample1); dst += 2; sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatBE(src[1])) + ((double) last_sample1)) * 0.5); last_sample0 = sample0; last_sample1 = sample1; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32MSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4; const float *target = ((const float *) cvt->buf); float sample3 = SDL_SwapFloatBE(src[3]); float sample2 = SDL_SwapFloatBE(src[2]); float sample1 = SDL_SwapFloatBE(src[1]); float sample0 = SDL_SwapFloatBE(src[0]); float last_sample3 = sample3; float last_sample2 = sample2; float last_sample1 = sample1; float last_sample0 = sample0; while (dst >= target) { dst[3] = SDL_SwapFloatBE(sample3); dst[2] = SDL_SwapFloatBE(sample2); dst[1] = SDL_SwapFloatBE(sample1); dst[0] = SDL_SwapFloatBE(sample0); dst -= 4; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 4; sample3 = (float) ((((double) SDL_SwapFloatBE(src[3])) + ((double) last_sample3)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatBE(src[2])) + ((double) last_sample2)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatBE(src[1])) + ((double) last_sample1)) * 0.5); sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32MSB, 4 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 256; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatBE(src[0]); float sample1 = SDL_SwapFloatBE(src[1]); float sample2 = SDL_SwapFloatBE(src[2]); float sample3 = SDL_SwapFloatBE(src[3]); float last_sample0 = sample0; float last_sample1 = sample1; float last_sample2 = sample2; float last_sample3 = sample3; while (dst < target) { src += 4; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatBE(sample0); dst[1] = SDL_SwapFloatBE(sample1); dst[2] = SDL_SwapFloatBE(sample2); dst[3] = SDL_SwapFloatBE(sample3); dst += 4; sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatBE(src[1])) + ((double) last_sample1)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatBE(src[2])) + ((double) last_sample2)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatBE(src[3])) + ((double) last_sample3)) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32MSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 384; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 6; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6; const float *target = ((const float *) cvt->buf); float sample5 = SDL_SwapFloatBE(src[5]); float sample4 = SDL_SwapFloatBE(src[4]); float sample3 = SDL_SwapFloatBE(src[3]); float sample2 = SDL_SwapFloatBE(src[2]); float sample1 = SDL_SwapFloatBE(src[1]); float sample0 = SDL_SwapFloatBE(src[0]); float last_sample5 = sample5; float last_sample4 = sample4; float last_sample3 = sample3; float last_sample2 = sample2; float last_sample1 = sample1; float last_sample0 = sample0; while (dst >= target) { dst[5] = SDL_SwapFloatBE(sample5); dst[4] = SDL_SwapFloatBE(sample4); dst[3] = SDL_SwapFloatBE(sample3); dst[2] = SDL_SwapFloatBE(sample2); dst[1] = SDL_SwapFloatBE(sample1); dst[0] = SDL_SwapFloatBE(sample0); dst -= 6; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 6; sample5 = (float) ((((double) SDL_SwapFloatBE(src[5])) + ((double) last_sample5)) * 0.5); sample4 = (float) ((((double) SDL_SwapFloatBE(src[4])) + ((double) last_sample4)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatBE(src[3])) + ((double) last_sample3)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatBE(src[2])) + ((double) last_sample2)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatBE(src[1])) + ((double) last_sample1)) * 0.5); sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32MSB, 6 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 384; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatBE(src[0]); float sample1 = SDL_SwapFloatBE(src[1]); float sample2 = SDL_SwapFloatBE(src[2]); float sample3 = SDL_SwapFloatBE(src[3]); float sample4 = SDL_SwapFloatBE(src[4]); float sample5 = SDL_SwapFloatBE(src[5]); float last_sample0 = sample0; float last_sample1 = sample1; float last_sample2 = sample2; float last_sample3 = sample3; float last_sample4 = sample4; float last_sample5 = sample5; while (dst < target) { src += 6; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatBE(sample0); dst[1] = SDL_SwapFloatBE(sample1); dst[2] = SDL_SwapFloatBE(sample2); dst[3] = SDL_SwapFloatBE(sample3); dst[4] = SDL_SwapFloatBE(sample4); dst[5] = SDL_SwapFloatBE(sample5); dst += 6; sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatBE(src[1])) + ((double) last_sample1)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatBE(src[2])) + ((double) last_sample2)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatBE(src[3])) + ((double) last_sample3)) * 0.5); sample4 = (float) ((((double) SDL_SwapFloatBE(src[4])) + ((double) last_sample4)) * 0.5); sample5 = (float) ((((double) SDL_SwapFloatBE(src[5])) + ((double) last_sample5)) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample arbitrary (x%f) AUDIO_F32MSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 512; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = ((float *) (cvt->buf + dstsize)) - 8; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8; const float *target = ((const float *) cvt->buf); float sample7 = SDL_SwapFloatBE(src[7]); float sample6 = SDL_SwapFloatBE(src[6]); float sample5 = SDL_SwapFloatBE(src[5]); float sample4 = SDL_SwapFloatBE(src[4]); float sample3 = SDL_SwapFloatBE(src[3]); float sample2 = SDL_SwapFloatBE(src[2]); float sample1 = SDL_SwapFloatBE(src[1]); float sample0 = SDL_SwapFloatBE(src[0]); float last_sample7 = sample7; float last_sample6 = sample6; float last_sample5 = sample5; float last_sample4 = sample4; float last_sample3 = sample3; float last_sample2 = sample2; float last_sample1 = sample1; float last_sample0 = sample0; while (dst >= target) { dst[7] = SDL_SwapFloatBE(sample7); dst[6] = SDL_SwapFloatBE(sample6); dst[5] = SDL_SwapFloatBE(sample5); dst[4] = SDL_SwapFloatBE(sample4); dst[3] = SDL_SwapFloatBE(sample3); dst[2] = SDL_SwapFloatBE(sample2); dst[1] = SDL_SwapFloatBE(sample1); dst[0] = SDL_SwapFloatBE(sample0); dst -= 8; eps += srcsize; if ((eps << 1) >= dstsize) { src -= 8; sample7 = (float) ((((double) SDL_SwapFloatBE(src[7])) + ((double) last_sample7)) * 0.5); sample6 = (float) ((((double) SDL_SwapFloatBE(src[6])) + ((double) last_sample6)) * 0.5); sample5 = (float) ((((double) SDL_SwapFloatBE(src[5])) + ((double) last_sample5)) * 0.5); sample4 = (float) ((((double) SDL_SwapFloatBE(src[4])) + ((double) last_sample4)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatBE(src[3])) + ((double) last_sample3)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatBE(src[2])) + ((double) last_sample2)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatBE(src[1])) + ((double) last_sample1)) * 0.5); sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; eps -= dstsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample arbitrary (x%f) AUDIO_F32MSB, 8 channels.\n", cvt->rate_incr); #endif const int srcsize = cvt->len_cvt - 512; const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr); register int eps = 0; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); float sample0 = SDL_SwapFloatBE(src[0]); float sample1 = SDL_SwapFloatBE(src[1]); float sample2 = SDL_SwapFloatBE(src[2]); float sample3 = SDL_SwapFloatBE(src[3]); float sample4 = SDL_SwapFloatBE(src[4]); float sample5 = SDL_SwapFloatBE(src[5]); float sample6 = SDL_SwapFloatBE(src[6]); float sample7 = SDL_SwapFloatBE(src[7]); float last_sample0 = sample0; float last_sample1 = sample1; float last_sample2 = sample2; float last_sample3 = sample3; float last_sample4 = sample4; float last_sample5 = sample5; float last_sample6 = sample6; float last_sample7 = sample7; while (dst < target) { src += 8; eps += dstsize; if ((eps << 1) >= srcsize) { dst[0] = SDL_SwapFloatBE(sample0); dst[1] = SDL_SwapFloatBE(sample1); dst[2] = SDL_SwapFloatBE(sample2); dst[3] = SDL_SwapFloatBE(sample3); dst[4] = SDL_SwapFloatBE(sample4); dst[5] = SDL_SwapFloatBE(sample5); dst[6] = SDL_SwapFloatBE(sample6); dst[7] = SDL_SwapFloatBE(sample7); dst += 8; sample0 = (float) ((((double) SDL_SwapFloatBE(src[0])) + ((double) last_sample0)) * 0.5); sample1 = (float) ((((double) SDL_SwapFloatBE(src[1])) + ((double) last_sample1)) * 0.5); sample2 = (float) ((((double) SDL_SwapFloatBE(src[2])) + ((double) last_sample2)) * 0.5); sample3 = (float) ((((double) SDL_SwapFloatBE(src[3])) + ((double) last_sample3)) * 0.5); sample4 = (float) ((((double) SDL_SwapFloatBE(src[4])) + ((double) last_sample4)) * 0.5); sample5 = (float) ((((double) SDL_SwapFloatBE(src[5])) + ((double) last_sample5)) * 0.5); sample6 = (float) ((((double) SDL_SwapFloatBE(src[6])) + ((double) last_sample6)) * 0.5); sample7 = (float) ((((double) SDL_SwapFloatBE(src[7])) + ((double) last_sample7)) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; eps -= srcsize; } } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } #if !LESS_RESAMPLERS static void SDLCALL SDL_Upsample_U8_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U8, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 1 * 2; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample0 = (Sint16) src[0]; src--; dst[1] = (Uint8) ((sample0 + last_sample0) >> 1); dst[0] = (Uint8) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U8, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; src += 2; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U8, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 1 * 4; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample0 = (Sint16) src[0]; src--; dst[3] = (Uint8) ((sample0 + (3 * last_sample0)) >> 2); dst[2] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) (((3 * sample0) + last_sample0) >> 2); dst[0] = (Uint8) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U8, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; src += 4; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U8, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 2 * 2; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample0 = (Sint16) src[0]; src -= 2; dst[3] = (Uint8) ((sample1 + last_sample1) >> 1); dst[2] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) sample1; dst[0] = (Uint8) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U8, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; Sint16 last_sample1 = (Sint16) src[1]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; const Sint16 sample1 = (Sint16) src[1]; src += 4; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U8, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 2 * 4; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample0 = (Sint16) src[0]; src -= 2; dst[7] = (Uint8) ((sample1 + (3 * last_sample1)) >> 2); dst[6] = (Uint8) ((sample0 + (3 * last_sample0)) >> 2); dst[5] = (Uint8) ((sample1 + last_sample1) >> 1); dst[4] = (Uint8) ((sample0 + last_sample0) >> 1); dst[3] = (Uint8) (((3 * sample1) + last_sample1) >> 2); dst[2] = (Uint8) (((3 * sample0) + last_sample0) >> 2); dst[1] = (Uint8) sample1; dst[0] = (Uint8) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U8, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; Sint16 last_sample1 = (Sint16) src[1]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; const Sint16 sample1 = (Sint16) src[1]; src += 8; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U8, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 4 * 2; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample0 = (Sint16) src[0]; src -= 4; dst[7] = (Uint8) ((sample3 + last_sample3) >> 1); dst[6] = (Uint8) ((sample2 + last_sample2) >> 1); dst[5] = (Uint8) ((sample1 + last_sample1) >> 1); dst[4] = (Uint8) ((sample0 + last_sample0) >> 1); dst[3] = (Uint8) sample3; dst[2] = (Uint8) sample2; dst[1] = (Uint8) sample1; dst[0] = (Uint8) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U8, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample3 = (Sint16) src[3]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample3 = (Sint16) src[3]; src += 8; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) ((sample1 + last_sample1) >> 1); dst[2] = (Uint8) ((sample2 + last_sample2) >> 1); dst[3] = (Uint8) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U8, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 4 * 4; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample0 = (Sint16) src[0]; src -= 4; dst[15] = (Uint8) ((sample3 + (3 * last_sample3)) >> 2); dst[14] = (Uint8) ((sample2 + (3 * last_sample2)) >> 2); dst[13] = (Uint8) ((sample1 + (3 * last_sample1)) >> 2); dst[12] = (Uint8) ((sample0 + (3 * last_sample0)) >> 2); dst[11] = (Uint8) ((sample3 + last_sample3) >> 1); dst[10] = (Uint8) ((sample2 + last_sample2) >> 1); dst[9] = (Uint8) ((sample1 + last_sample1) >> 1); dst[8] = (Uint8) ((sample0 + last_sample0) >> 1); dst[7] = (Uint8) (((3 * sample3) + last_sample3) >> 2); dst[6] = (Uint8) (((3 * sample2) + last_sample2) >> 2); dst[5] = (Uint8) (((3 * sample1) + last_sample1) >> 2); dst[4] = (Uint8) (((3 * sample0) + last_sample0) >> 2); dst[3] = (Uint8) sample3; dst[2] = (Uint8) sample2; dst[1] = (Uint8) sample1; dst[0] = (Uint8) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U8, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample3 = (Sint16) src[3]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample3 = (Sint16) src[3]; src += 16; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) ((sample1 + last_sample1) >> 1); dst[2] = (Uint8) ((sample2 + last_sample2) >> 1); dst[3] = (Uint8) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U8, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 6 * 2; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample5 = (Sint16) src[5]; Sint16 last_sample4 = (Sint16) src[4]; Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample5 = (Sint16) src[5]; const Sint16 sample4 = (Sint16) src[4]; const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample0 = (Sint16) src[0]; src -= 6; dst[11] = (Uint8) ((sample5 + last_sample5) >> 1); dst[10] = (Uint8) ((sample4 + last_sample4) >> 1); dst[9] = (Uint8) ((sample3 + last_sample3) >> 1); dst[8] = (Uint8) ((sample2 + last_sample2) >> 1); dst[7] = (Uint8) ((sample1 + last_sample1) >> 1); dst[6] = (Uint8) ((sample0 + last_sample0) >> 1); dst[5] = (Uint8) sample5; dst[4] = (Uint8) sample4; dst[3] = (Uint8) sample3; dst[2] = (Uint8) sample2; dst[1] = (Uint8) sample1; dst[0] = (Uint8) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U8, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample4 = (Sint16) src[4]; Sint16 last_sample5 = (Sint16) src[5]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample4 = (Sint16) src[4]; const Sint16 sample5 = (Sint16) src[5]; src += 12; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) ((sample1 + last_sample1) >> 1); dst[2] = (Uint8) ((sample2 + last_sample2) >> 1); dst[3] = (Uint8) ((sample3 + last_sample3) >> 1); dst[4] = (Uint8) ((sample4 + last_sample4) >> 1); dst[5] = (Uint8) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U8, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 6 * 4; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample5 = (Sint16) src[5]; Sint16 last_sample4 = (Sint16) src[4]; Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample5 = (Sint16) src[5]; const Sint16 sample4 = (Sint16) src[4]; const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample0 = (Sint16) src[0]; src -= 6; dst[23] = (Uint8) ((sample5 + (3 * last_sample5)) >> 2); dst[22] = (Uint8) ((sample4 + (3 * last_sample4)) >> 2); dst[21] = (Uint8) ((sample3 + (3 * last_sample3)) >> 2); dst[20] = (Uint8) ((sample2 + (3 * last_sample2)) >> 2); dst[19] = (Uint8) ((sample1 + (3 * last_sample1)) >> 2); dst[18] = (Uint8) ((sample0 + (3 * last_sample0)) >> 2); dst[17] = (Uint8) ((sample5 + last_sample5) >> 1); dst[16] = (Uint8) ((sample4 + last_sample4) >> 1); dst[15] = (Uint8) ((sample3 + last_sample3) >> 1); dst[14] = (Uint8) ((sample2 + last_sample2) >> 1); dst[13] = (Uint8) ((sample1 + last_sample1) >> 1); dst[12] = (Uint8) ((sample0 + last_sample0) >> 1); dst[11] = (Uint8) (((3 * sample5) + last_sample5) >> 2); dst[10] = (Uint8) (((3 * sample4) + last_sample4) >> 2); dst[9] = (Uint8) (((3 * sample3) + last_sample3) >> 2); dst[8] = (Uint8) (((3 * sample2) + last_sample2) >> 2); dst[7] = (Uint8) (((3 * sample1) + last_sample1) >> 2); dst[6] = (Uint8) (((3 * sample0) + last_sample0) >> 2); dst[5] = (Uint8) sample5; dst[4] = (Uint8) sample4; dst[3] = (Uint8) sample3; dst[2] = (Uint8) sample2; dst[1] = (Uint8) sample1; dst[0] = (Uint8) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U8, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample4 = (Sint16) src[4]; Sint16 last_sample5 = (Sint16) src[5]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample4 = (Sint16) src[4]; const Sint16 sample5 = (Sint16) src[5]; src += 24; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) ((sample1 + last_sample1) >> 1); dst[2] = (Uint8) ((sample2 + last_sample2) >> 1); dst[3] = (Uint8) ((sample3 + last_sample3) >> 1); dst[4] = (Uint8) ((sample4 + last_sample4) >> 1); dst[5] = (Uint8) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U8, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 8 * 2; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample7 = (Sint16) src[7]; Sint16 last_sample6 = (Sint16) src[6]; Sint16 last_sample5 = (Sint16) src[5]; Sint16 last_sample4 = (Sint16) src[4]; Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample7 = (Sint16) src[7]; const Sint16 sample6 = (Sint16) src[6]; const Sint16 sample5 = (Sint16) src[5]; const Sint16 sample4 = (Sint16) src[4]; const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample0 = (Sint16) src[0]; src -= 8; dst[15] = (Uint8) ((sample7 + last_sample7) >> 1); dst[14] = (Uint8) ((sample6 + last_sample6) >> 1); dst[13] = (Uint8) ((sample5 + last_sample5) >> 1); dst[12] = (Uint8) ((sample4 + last_sample4) >> 1); dst[11] = (Uint8) ((sample3 + last_sample3) >> 1); dst[10] = (Uint8) ((sample2 + last_sample2) >> 1); dst[9] = (Uint8) ((sample1 + last_sample1) >> 1); dst[8] = (Uint8) ((sample0 + last_sample0) >> 1); dst[7] = (Uint8) sample7; dst[6] = (Uint8) sample6; dst[5] = (Uint8) sample5; dst[4] = (Uint8) sample4; dst[3] = (Uint8) sample3; dst[2] = (Uint8) sample2; dst[1] = (Uint8) sample1; dst[0] = (Uint8) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U8, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample4 = (Sint16) src[4]; Sint16 last_sample5 = (Sint16) src[5]; Sint16 last_sample6 = (Sint16) src[6]; Sint16 last_sample7 = (Sint16) src[7]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample4 = (Sint16) src[4]; const Sint16 sample5 = (Sint16) src[5]; const Sint16 sample6 = (Sint16) src[6]; const Sint16 sample7 = (Sint16) src[7]; src += 16; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) ((sample1 + last_sample1) >> 1); dst[2] = (Uint8) ((sample2 + last_sample2) >> 1); dst[3] = (Uint8) ((sample3 + last_sample3) >> 1); dst[4] = (Uint8) ((sample4 + last_sample4) >> 1); dst[5] = (Uint8) ((sample5 + last_sample5) >> 1); dst[6] = (Uint8) ((sample6 + last_sample6) >> 1); dst[7] = (Uint8) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U8_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U8, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 8 * 4; const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint8 *target = ((const Uint8 *) cvt->buf); Sint16 last_sample7 = (Sint16) src[7]; Sint16 last_sample6 = (Sint16) src[6]; Sint16 last_sample5 = (Sint16) src[5]; Sint16 last_sample4 = (Sint16) src[4]; Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample0 = (Sint16) src[0]; while (dst >= target) { const Sint16 sample7 = (Sint16) src[7]; const Sint16 sample6 = (Sint16) src[6]; const Sint16 sample5 = (Sint16) src[5]; const Sint16 sample4 = (Sint16) src[4]; const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample0 = (Sint16) src[0]; src -= 8; dst[31] = (Uint8) ((sample7 + (3 * last_sample7)) >> 2); dst[30] = (Uint8) ((sample6 + (3 * last_sample6)) >> 2); dst[29] = (Uint8) ((sample5 + (3 * last_sample5)) >> 2); dst[28] = (Uint8) ((sample4 + (3 * last_sample4)) >> 2); dst[27] = (Uint8) ((sample3 + (3 * last_sample3)) >> 2); dst[26] = (Uint8) ((sample2 + (3 * last_sample2)) >> 2); dst[25] = (Uint8) ((sample1 + (3 * last_sample1)) >> 2); dst[24] = (Uint8) ((sample0 + (3 * last_sample0)) >> 2); dst[23] = (Uint8) ((sample7 + last_sample7) >> 1); dst[22] = (Uint8) ((sample6 + last_sample6) >> 1); dst[21] = (Uint8) ((sample5 + last_sample5) >> 1); dst[20] = (Uint8) ((sample4 + last_sample4) >> 1); dst[19] = (Uint8) ((sample3 + last_sample3) >> 1); dst[18] = (Uint8) ((sample2 + last_sample2) >> 1); dst[17] = (Uint8) ((sample1 + last_sample1) >> 1); dst[16] = (Uint8) ((sample0 + last_sample0) >> 1); dst[15] = (Uint8) (((3 * sample7) + last_sample7) >> 2); dst[14] = (Uint8) (((3 * sample6) + last_sample6) >> 2); dst[13] = (Uint8) (((3 * sample5) + last_sample5) >> 2); dst[12] = (Uint8) (((3 * sample4) + last_sample4) >> 2); dst[11] = (Uint8) (((3 * sample3) + last_sample3) >> 2); dst[10] = (Uint8) (((3 * sample2) + last_sample2) >> 2); dst[9] = (Uint8) (((3 * sample1) + last_sample1) >> 2); dst[8] = (Uint8) (((3 * sample0) + last_sample0) >> 2); dst[7] = (Uint8) sample7; dst[6] = (Uint8) sample6; dst[5] = (Uint8) sample5; dst[4] = (Uint8) sample4; dst[3] = (Uint8) sample3; dst[2] = (Uint8) sample2; dst[1] = (Uint8) sample1; dst[0] = (Uint8) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U8_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U8, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint8 *dst = (Uint8 *) cvt->buf; const Uint8 *src = (Uint8 *) cvt->buf; const Uint8 *target = (const Uint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) src[0]; Sint16 last_sample1 = (Sint16) src[1]; Sint16 last_sample2 = (Sint16) src[2]; Sint16 last_sample3 = (Sint16) src[3]; Sint16 last_sample4 = (Sint16) src[4]; Sint16 last_sample5 = (Sint16) src[5]; Sint16 last_sample6 = (Sint16) src[6]; Sint16 last_sample7 = (Sint16) src[7]; while (dst < target) { const Sint16 sample0 = (Sint16) src[0]; const Sint16 sample1 = (Sint16) src[1]; const Sint16 sample2 = (Sint16) src[2]; const Sint16 sample3 = (Sint16) src[3]; const Sint16 sample4 = (Sint16) src[4]; const Sint16 sample5 = (Sint16) src[5]; const Sint16 sample6 = (Sint16) src[6]; const Sint16 sample7 = (Sint16) src[7]; src += 32; dst[0] = (Uint8) ((sample0 + last_sample0) >> 1); dst[1] = (Uint8) ((sample1 + last_sample1) >> 1); dst[2] = (Uint8) ((sample2 + last_sample2) >> 1); dst[3] = (Uint8) ((sample3 + last_sample3) >> 1); dst[4] = (Uint8) ((sample4 + last_sample4) >> 1); dst[5] = (Uint8) ((sample5 + last_sample5) >> 1); dst[6] = (Uint8) ((sample6 + last_sample6) >> 1); dst[7] = (Uint8) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S8, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 1 * 2; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src--; dst[1] = (Sint8) ((sample0 + last_sample0) >> 1); dst[0] = (Sint8) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S8, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src += 2; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S8, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 1 * 4; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src--; dst[3] = (Sint8) ((sample0 + (3 * last_sample0)) >> 2); dst[2] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) (((3 * sample0) + last_sample0) >> 2); dst[0] = (Sint8) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S8, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src += 4; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S8, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 2 * 2; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src -= 2; dst[3] = (Sint8) ((sample1 + last_sample1) >> 1); dst[2] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) sample1; dst[0] = (Sint8) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S8, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); src += 4; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S8, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 2 * 4; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src -= 2; dst[7] = (Sint8) ((sample1 + (3 * last_sample1)) >> 2); dst[6] = (Sint8) ((sample0 + (3 * last_sample0)) >> 2); dst[5] = (Sint8) ((sample1 + last_sample1) >> 1); dst[4] = (Sint8) ((sample0 + last_sample0) >> 1); dst[3] = (Sint8) (((3 * sample1) + last_sample1) >> 2); dst[2] = (Sint8) (((3 * sample0) + last_sample0) >> 2); dst[1] = (Sint8) sample1; dst[0] = (Sint8) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S8, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); src += 8; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S8, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 4 * 2; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src -= 4; dst[7] = (Sint8) ((sample3 + last_sample3) >> 1); dst[6] = (Sint8) ((sample2 + last_sample2) >> 1); dst[5] = (Sint8) ((sample1 + last_sample1) >> 1); dst[4] = (Sint8) ((sample0 + last_sample0) >> 1); dst[3] = (Sint8) sample3; dst[2] = (Sint8) sample2; dst[1] = (Sint8) sample1; dst[0] = (Sint8) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S8, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); src += 8; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) ((sample1 + last_sample1) >> 1); dst[2] = (Sint8) ((sample2 + last_sample2) >> 1); dst[3] = (Sint8) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S8, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 4 * 4; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src -= 4; dst[15] = (Sint8) ((sample3 + (3 * last_sample3)) >> 2); dst[14] = (Sint8) ((sample2 + (3 * last_sample2)) >> 2); dst[13] = (Sint8) ((sample1 + (3 * last_sample1)) >> 2); dst[12] = (Sint8) ((sample0 + (3 * last_sample0)) >> 2); dst[11] = (Sint8) ((sample3 + last_sample3) >> 1); dst[10] = (Sint8) ((sample2 + last_sample2) >> 1); dst[9] = (Sint8) ((sample1 + last_sample1) >> 1); dst[8] = (Sint8) ((sample0 + last_sample0) >> 1); dst[7] = (Sint8) (((3 * sample3) + last_sample3) >> 2); dst[6] = (Sint8) (((3 * sample2) + last_sample2) >> 2); dst[5] = (Sint8) (((3 * sample1) + last_sample1) >> 2); dst[4] = (Sint8) (((3 * sample0) + last_sample0) >> 2); dst[3] = (Sint8) sample3; dst[2] = (Sint8) sample2; dst[1] = (Sint8) sample1; dst[0] = (Sint8) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S8, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); src += 16; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) ((sample1 + last_sample1) >> 1); dst[2] = (Sint8) ((sample2 + last_sample2) >> 1); dst[3] = (Sint8) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S8, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 6 * 2; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample5 = (Sint16) ((Sint8) src[5]); Sint16 last_sample4 = (Sint16) ((Sint8) src[4]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample5 = (Sint16) ((Sint8) src[5]); const Sint16 sample4 = (Sint16) ((Sint8) src[4]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src -= 6; dst[11] = (Sint8) ((sample5 + last_sample5) >> 1); dst[10] = (Sint8) ((sample4 + last_sample4) >> 1); dst[9] = (Sint8) ((sample3 + last_sample3) >> 1); dst[8] = (Sint8) ((sample2 + last_sample2) >> 1); dst[7] = (Sint8) ((sample1 + last_sample1) >> 1); dst[6] = (Sint8) ((sample0 + last_sample0) >> 1); dst[5] = (Sint8) sample5; dst[4] = (Sint8) sample4; dst[3] = (Sint8) sample3; dst[2] = (Sint8) sample2; dst[1] = (Sint8) sample1; dst[0] = (Sint8) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S8, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample4 = (Sint16) ((Sint8) src[4]); Sint16 last_sample5 = (Sint16) ((Sint8) src[5]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample4 = (Sint16) ((Sint8) src[4]); const Sint16 sample5 = (Sint16) ((Sint8) src[5]); src += 12; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) ((sample1 + last_sample1) >> 1); dst[2] = (Sint8) ((sample2 + last_sample2) >> 1); dst[3] = (Sint8) ((sample3 + last_sample3) >> 1); dst[4] = (Sint8) ((sample4 + last_sample4) >> 1); dst[5] = (Sint8) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S8, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 6 * 4; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample5 = (Sint16) ((Sint8) src[5]); Sint16 last_sample4 = (Sint16) ((Sint8) src[4]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample5 = (Sint16) ((Sint8) src[5]); const Sint16 sample4 = (Sint16) ((Sint8) src[4]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src -= 6; dst[23] = (Sint8) ((sample5 + (3 * last_sample5)) >> 2); dst[22] = (Sint8) ((sample4 + (3 * last_sample4)) >> 2); dst[21] = (Sint8) ((sample3 + (3 * last_sample3)) >> 2); dst[20] = (Sint8) ((sample2 + (3 * last_sample2)) >> 2); dst[19] = (Sint8) ((sample1 + (3 * last_sample1)) >> 2); dst[18] = (Sint8) ((sample0 + (3 * last_sample0)) >> 2); dst[17] = (Sint8) ((sample5 + last_sample5) >> 1); dst[16] = (Sint8) ((sample4 + last_sample4) >> 1); dst[15] = (Sint8) ((sample3 + last_sample3) >> 1); dst[14] = (Sint8) ((sample2 + last_sample2) >> 1); dst[13] = (Sint8) ((sample1 + last_sample1) >> 1); dst[12] = (Sint8) ((sample0 + last_sample0) >> 1); dst[11] = (Sint8) (((3 * sample5) + last_sample5) >> 2); dst[10] = (Sint8) (((3 * sample4) + last_sample4) >> 2); dst[9] = (Sint8) (((3 * sample3) + last_sample3) >> 2); dst[8] = (Sint8) (((3 * sample2) + last_sample2) >> 2); dst[7] = (Sint8) (((3 * sample1) + last_sample1) >> 2); dst[6] = (Sint8) (((3 * sample0) + last_sample0) >> 2); dst[5] = (Sint8) sample5; dst[4] = (Sint8) sample4; dst[3] = (Sint8) sample3; dst[2] = (Sint8) sample2; dst[1] = (Sint8) sample1; dst[0] = (Sint8) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S8, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample4 = (Sint16) ((Sint8) src[4]); Sint16 last_sample5 = (Sint16) ((Sint8) src[5]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample4 = (Sint16) ((Sint8) src[4]); const Sint16 sample5 = (Sint16) ((Sint8) src[5]); src += 24; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) ((sample1 + last_sample1) >> 1); dst[2] = (Sint8) ((sample2 + last_sample2) >> 1); dst[3] = (Sint8) ((sample3 + last_sample3) >> 1); dst[4] = (Sint8) ((sample4 + last_sample4) >> 1); dst[5] = (Sint8) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S8, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 8 * 2; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample7 = (Sint16) ((Sint8) src[7]); Sint16 last_sample6 = (Sint16) ((Sint8) src[6]); Sint16 last_sample5 = (Sint16) ((Sint8) src[5]); Sint16 last_sample4 = (Sint16) ((Sint8) src[4]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample7 = (Sint16) ((Sint8) src[7]); const Sint16 sample6 = (Sint16) ((Sint8) src[6]); const Sint16 sample5 = (Sint16) ((Sint8) src[5]); const Sint16 sample4 = (Sint16) ((Sint8) src[4]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src -= 8; dst[15] = (Sint8) ((sample7 + last_sample7) >> 1); dst[14] = (Sint8) ((sample6 + last_sample6) >> 1); dst[13] = (Sint8) ((sample5 + last_sample5) >> 1); dst[12] = (Sint8) ((sample4 + last_sample4) >> 1); dst[11] = (Sint8) ((sample3 + last_sample3) >> 1); dst[10] = (Sint8) ((sample2 + last_sample2) >> 1); dst[9] = (Sint8) ((sample1 + last_sample1) >> 1); dst[8] = (Sint8) ((sample0 + last_sample0) >> 1); dst[7] = (Sint8) sample7; dst[6] = (Sint8) sample6; dst[5] = (Sint8) sample5; dst[4] = (Sint8) sample4; dst[3] = (Sint8) sample3; dst[2] = (Sint8) sample2; dst[1] = (Sint8) sample1; dst[0] = (Sint8) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S8, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample4 = (Sint16) ((Sint8) src[4]); Sint16 last_sample5 = (Sint16) ((Sint8) src[5]); Sint16 last_sample6 = (Sint16) ((Sint8) src[6]); Sint16 last_sample7 = (Sint16) ((Sint8) src[7]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample4 = (Sint16) ((Sint8) src[4]); const Sint16 sample5 = (Sint16) ((Sint8) src[5]); const Sint16 sample6 = (Sint16) ((Sint8) src[6]); const Sint16 sample7 = (Sint16) ((Sint8) src[7]); src += 16; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) ((sample1 + last_sample1) >> 1); dst[2] = (Sint8) ((sample2 + last_sample2) >> 1); dst[3] = (Sint8) ((sample3 + last_sample3) >> 1); dst[4] = (Sint8) ((sample4 + last_sample4) >> 1); dst[5] = (Sint8) ((sample5 + last_sample5) >> 1); dst[6] = (Sint8) ((sample6 + last_sample6) >> 1); dst[7] = (Sint8) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S8_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S8, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 8 * 4; const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint8 *target = ((const Sint8 *) cvt->buf); Sint16 last_sample7 = (Sint16) ((Sint8) src[7]); Sint16 last_sample6 = (Sint16) ((Sint8) src[6]); Sint16 last_sample5 = (Sint16) ((Sint8) src[5]); Sint16 last_sample4 = (Sint16) ((Sint8) src[4]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); while (dst >= target) { const Sint16 sample7 = (Sint16) ((Sint8) src[7]); const Sint16 sample6 = (Sint16) ((Sint8) src[6]); const Sint16 sample5 = (Sint16) ((Sint8) src[5]); const Sint16 sample4 = (Sint16) ((Sint8) src[4]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample0 = (Sint16) ((Sint8) src[0]); src -= 8; dst[31] = (Sint8) ((sample7 + (3 * last_sample7)) >> 2); dst[30] = (Sint8) ((sample6 + (3 * last_sample6)) >> 2); dst[29] = (Sint8) ((sample5 + (3 * last_sample5)) >> 2); dst[28] = (Sint8) ((sample4 + (3 * last_sample4)) >> 2); dst[27] = (Sint8) ((sample3 + (3 * last_sample3)) >> 2); dst[26] = (Sint8) ((sample2 + (3 * last_sample2)) >> 2); dst[25] = (Sint8) ((sample1 + (3 * last_sample1)) >> 2); dst[24] = (Sint8) ((sample0 + (3 * last_sample0)) >> 2); dst[23] = (Sint8) ((sample7 + last_sample7) >> 1); dst[22] = (Sint8) ((sample6 + last_sample6) >> 1); dst[21] = (Sint8) ((sample5 + last_sample5) >> 1); dst[20] = (Sint8) ((sample4 + last_sample4) >> 1); dst[19] = (Sint8) ((sample3 + last_sample3) >> 1); dst[18] = (Sint8) ((sample2 + last_sample2) >> 1); dst[17] = (Sint8) ((sample1 + last_sample1) >> 1); dst[16] = (Sint8) ((sample0 + last_sample0) >> 1); dst[15] = (Sint8) (((3 * sample7) + last_sample7) >> 2); dst[14] = (Sint8) (((3 * sample6) + last_sample6) >> 2); dst[13] = (Sint8) (((3 * sample5) + last_sample5) >> 2); dst[12] = (Sint8) (((3 * sample4) + last_sample4) >> 2); dst[11] = (Sint8) (((3 * sample3) + last_sample3) >> 2); dst[10] = (Sint8) (((3 * sample2) + last_sample2) >> 2); dst[9] = (Sint8) (((3 * sample1) + last_sample1) >> 2); dst[8] = (Sint8) (((3 * sample0) + last_sample0) >> 2); dst[7] = (Sint8) sample7; dst[6] = (Sint8) sample6; dst[5] = (Sint8) sample5; dst[4] = (Sint8) sample4; dst[3] = (Sint8) sample3; dst[2] = (Sint8) sample2; dst[1] = (Sint8) sample1; dst[0] = (Sint8) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S8_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S8, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint8 *dst = (Sint8 *) cvt->buf; const Sint8 *src = (Sint8 *) cvt->buf; const Sint8 *target = (const Sint8 *) (cvt->buf + dstsize); Sint16 last_sample0 = (Sint16) ((Sint8) src[0]); Sint16 last_sample1 = (Sint16) ((Sint8) src[1]); Sint16 last_sample2 = (Sint16) ((Sint8) src[2]); Sint16 last_sample3 = (Sint16) ((Sint8) src[3]); Sint16 last_sample4 = (Sint16) ((Sint8) src[4]); Sint16 last_sample5 = (Sint16) ((Sint8) src[5]); Sint16 last_sample6 = (Sint16) ((Sint8) src[6]); Sint16 last_sample7 = (Sint16) ((Sint8) src[7]); while (dst < target) { const Sint16 sample0 = (Sint16) ((Sint8) src[0]); const Sint16 sample1 = (Sint16) ((Sint8) src[1]); const Sint16 sample2 = (Sint16) ((Sint8) src[2]); const Sint16 sample3 = (Sint16) ((Sint8) src[3]); const Sint16 sample4 = (Sint16) ((Sint8) src[4]); const Sint16 sample5 = (Sint16) ((Sint8) src[5]); const Sint16 sample6 = (Sint16) ((Sint8) src[6]); const Sint16 sample7 = (Sint16) ((Sint8) src[7]); src += 32; dst[0] = (Sint8) ((sample0 + last_sample0) >> 1); dst[1] = (Sint8) ((sample1 + last_sample1) >> 1); dst[2] = (Sint8) ((sample2 + last_sample2) >> 1); dst[3] = (Sint8) ((sample3 + last_sample3) >> 1); dst[4] = (Sint8) ((sample4 + last_sample4) >> 1); dst[5] = (Sint8) ((sample5 + last_sample5) >> 1); dst[6] = (Sint8) ((sample6 + last_sample6) >> 1); dst[7] = (Sint8) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src--; dst[1] = (Uint16) ((sample0 + last_sample0) >> 1); dst[0] = (Uint16) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src += 2; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src--; dst[3] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[2] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[0] = (Uint16) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src += 4; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src -= 2; dst[3] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); src += 4; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src -= 2; dst[7] = (Uint16) ((sample1 + (3 * last_sample1)) >> 2); dst[6] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[5] = (Uint16) ((sample1 + last_sample1) >> 1); dst[4] = (Uint16) ((sample0 + last_sample0) >> 1); dst[3] = (Uint16) (((3 * sample1) + last_sample1) >> 2); dst[2] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); src += 8; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src -= 4; dst[7] = (Uint16) ((sample3 + last_sample3) >> 1); dst[6] = (Uint16) ((sample2 + last_sample2) >> 1); dst[5] = (Uint16) ((sample1 + last_sample1) >> 1); dst[4] = (Uint16) ((sample0 + last_sample0) >> 1); dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); src += 8; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src -= 4; dst[15] = (Uint16) ((sample3 + (3 * last_sample3)) >> 2); dst[14] = (Uint16) ((sample2 + (3 * last_sample2)) >> 2); dst[13] = (Uint16) ((sample1 + (3 * last_sample1)) >> 2); dst[12] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[11] = (Uint16) ((sample3 + last_sample3) >> 1); dst[10] = (Uint16) ((sample2 + last_sample2) >> 1); dst[9] = (Uint16) ((sample1 + last_sample1) >> 1); dst[8] = (Uint16) ((sample0 + last_sample0) >> 1); dst[7] = (Uint16) (((3 * sample3) + last_sample3) >> 2); dst[6] = (Uint16) (((3 * sample2) + last_sample2) >> 2); dst[5] = (Uint16) (((3 * sample1) + last_sample1) >> 2); dst[4] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); src += 16; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample5 = (Sint32) SDL_SwapLE16(src[5]); Sint32 last_sample4 = (Sint32) SDL_SwapLE16(src[4]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample5 = (Sint32) SDL_SwapLE16(src[5]); const Sint32 sample4 = (Sint32) SDL_SwapLE16(src[4]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src -= 6; dst[11] = (Uint16) ((sample5 + last_sample5) >> 1); dst[10] = (Uint16) ((sample4 + last_sample4) >> 1); dst[9] = (Uint16) ((sample3 + last_sample3) >> 1); dst[8] = (Uint16) ((sample2 + last_sample2) >> 1); dst[7] = (Uint16) ((sample1 + last_sample1) >> 1); dst[6] = (Uint16) ((sample0 + last_sample0) >> 1); dst[5] = (Uint16) sample5; dst[4] = (Uint16) sample4; dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample4 = (Sint32) SDL_SwapLE16(src[4]); Sint32 last_sample5 = (Sint32) SDL_SwapLE16(src[5]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample4 = (Sint32) SDL_SwapLE16(src[4]); const Sint32 sample5 = (Sint32) SDL_SwapLE16(src[5]); src += 12; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); dst[4] = (Uint16) ((sample4 + last_sample4) >> 1); dst[5] = (Uint16) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample5 = (Sint32) SDL_SwapLE16(src[5]); Sint32 last_sample4 = (Sint32) SDL_SwapLE16(src[4]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample5 = (Sint32) SDL_SwapLE16(src[5]); const Sint32 sample4 = (Sint32) SDL_SwapLE16(src[4]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src -= 6; dst[23] = (Uint16) ((sample5 + (3 * last_sample5)) >> 2); dst[22] = (Uint16) ((sample4 + (3 * last_sample4)) >> 2); dst[21] = (Uint16) ((sample3 + (3 * last_sample3)) >> 2); dst[20] = (Uint16) ((sample2 + (3 * last_sample2)) >> 2); dst[19] = (Uint16) ((sample1 + (3 * last_sample1)) >> 2); dst[18] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[17] = (Uint16) ((sample5 + last_sample5) >> 1); dst[16] = (Uint16) ((sample4 + last_sample4) >> 1); dst[15] = (Uint16) ((sample3 + last_sample3) >> 1); dst[14] = (Uint16) ((sample2 + last_sample2) >> 1); dst[13] = (Uint16) ((sample1 + last_sample1) >> 1); dst[12] = (Uint16) ((sample0 + last_sample0) >> 1); dst[11] = (Uint16) (((3 * sample5) + last_sample5) >> 2); dst[10] = (Uint16) (((3 * sample4) + last_sample4) >> 2); dst[9] = (Uint16) (((3 * sample3) + last_sample3) >> 2); dst[8] = (Uint16) (((3 * sample2) + last_sample2) >> 2); dst[7] = (Uint16) (((3 * sample1) + last_sample1) >> 2); dst[6] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[5] = (Uint16) sample5; dst[4] = (Uint16) sample4; dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample4 = (Sint32) SDL_SwapLE16(src[4]); Sint32 last_sample5 = (Sint32) SDL_SwapLE16(src[5]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample4 = (Sint32) SDL_SwapLE16(src[4]); const Sint32 sample5 = (Sint32) SDL_SwapLE16(src[5]); src += 24; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); dst[4] = (Uint16) ((sample4 + last_sample4) >> 1); dst[5] = (Uint16) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample7 = (Sint32) SDL_SwapLE16(src[7]); Sint32 last_sample6 = (Sint32) SDL_SwapLE16(src[6]); Sint32 last_sample5 = (Sint32) SDL_SwapLE16(src[5]); Sint32 last_sample4 = (Sint32) SDL_SwapLE16(src[4]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample7 = (Sint32) SDL_SwapLE16(src[7]); const Sint32 sample6 = (Sint32) SDL_SwapLE16(src[6]); const Sint32 sample5 = (Sint32) SDL_SwapLE16(src[5]); const Sint32 sample4 = (Sint32) SDL_SwapLE16(src[4]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src -= 8; dst[15] = (Uint16) ((sample7 + last_sample7) >> 1); dst[14] = (Uint16) ((sample6 + last_sample6) >> 1); dst[13] = (Uint16) ((sample5 + last_sample5) >> 1); dst[12] = (Uint16) ((sample4 + last_sample4) >> 1); dst[11] = (Uint16) ((sample3 + last_sample3) >> 1); dst[10] = (Uint16) ((sample2 + last_sample2) >> 1); dst[9] = (Uint16) ((sample1 + last_sample1) >> 1); dst[8] = (Uint16) ((sample0 + last_sample0) >> 1); dst[7] = (Uint16) sample7; dst[6] = (Uint16) sample6; dst[5] = (Uint16) sample5; dst[4] = (Uint16) sample4; dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample4 = (Sint32) SDL_SwapLE16(src[4]); Sint32 last_sample5 = (Sint32) SDL_SwapLE16(src[5]); Sint32 last_sample6 = (Sint32) SDL_SwapLE16(src[6]); Sint32 last_sample7 = (Sint32) SDL_SwapLE16(src[7]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample4 = (Sint32) SDL_SwapLE16(src[4]); const Sint32 sample5 = (Sint32) SDL_SwapLE16(src[5]); const Sint32 sample6 = (Sint32) SDL_SwapLE16(src[6]); const Sint32 sample7 = (Sint32) SDL_SwapLE16(src[7]); src += 16; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); dst[4] = (Uint16) ((sample4 + last_sample4) >> 1); dst[5] = (Uint16) ((sample5 + last_sample5) >> 1); dst[6] = (Uint16) ((sample6 + last_sample6) >> 1); dst[7] = (Uint16) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16LSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample7 = (Sint32) SDL_SwapLE16(src[7]); Sint32 last_sample6 = (Sint32) SDL_SwapLE16(src[6]); Sint32 last_sample5 = (Sint32) SDL_SwapLE16(src[5]); Sint32 last_sample4 = (Sint32) SDL_SwapLE16(src[4]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); while (dst >= target) { const Sint32 sample7 = (Sint32) SDL_SwapLE16(src[7]); const Sint32 sample6 = (Sint32) SDL_SwapLE16(src[6]); const Sint32 sample5 = (Sint32) SDL_SwapLE16(src[5]); const Sint32 sample4 = (Sint32) SDL_SwapLE16(src[4]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); src -= 8; dst[31] = (Uint16) ((sample7 + (3 * last_sample7)) >> 2); dst[30] = (Uint16) ((sample6 + (3 * last_sample6)) >> 2); dst[29] = (Uint16) ((sample5 + (3 * last_sample5)) >> 2); dst[28] = (Uint16) ((sample4 + (3 * last_sample4)) >> 2); dst[27] = (Uint16) ((sample3 + (3 * last_sample3)) >> 2); dst[26] = (Uint16) ((sample2 + (3 * last_sample2)) >> 2); dst[25] = (Uint16) ((sample1 + (3 * last_sample1)) >> 2); dst[24] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[23] = (Uint16) ((sample7 + last_sample7) >> 1); dst[22] = (Uint16) ((sample6 + last_sample6) >> 1); dst[21] = (Uint16) ((sample5 + last_sample5) >> 1); dst[20] = (Uint16) ((sample4 + last_sample4) >> 1); dst[19] = (Uint16) ((sample3 + last_sample3) >> 1); dst[18] = (Uint16) ((sample2 + last_sample2) >> 1); dst[17] = (Uint16) ((sample1 + last_sample1) >> 1); dst[16] = (Uint16) ((sample0 + last_sample0) >> 1); dst[15] = (Uint16) (((3 * sample7) + last_sample7) >> 2); dst[14] = (Uint16) (((3 * sample6) + last_sample6) >> 2); dst[13] = (Uint16) (((3 * sample5) + last_sample5) >> 2); dst[12] = (Uint16) (((3 * sample4) + last_sample4) >> 2); dst[11] = (Uint16) (((3 * sample3) + last_sample3) >> 2); dst[10] = (Uint16) (((3 * sample2) + last_sample2) >> 2); dst[9] = (Uint16) (((3 * sample1) + last_sample1) >> 2); dst[8] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[7] = (Uint16) sample7; dst[6] = (Uint16) sample6; dst[5] = (Uint16) sample5; dst[4] = (Uint16) sample4; dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16LSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapLE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapLE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapLE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapLE16(src[3]); Sint32 last_sample4 = (Sint32) SDL_SwapLE16(src[4]); Sint32 last_sample5 = (Sint32) SDL_SwapLE16(src[5]); Sint32 last_sample6 = (Sint32) SDL_SwapLE16(src[6]); Sint32 last_sample7 = (Sint32) SDL_SwapLE16(src[7]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapLE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapLE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapLE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapLE16(src[3]); const Sint32 sample4 = (Sint32) SDL_SwapLE16(src[4]); const Sint32 sample5 = (Sint32) SDL_SwapLE16(src[5]); const Sint32 sample6 = (Sint32) SDL_SwapLE16(src[6]); const Sint32 sample7 = (Sint32) SDL_SwapLE16(src[7]); src += 32; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); dst[4] = (Uint16) ((sample4 + last_sample4) >> 1); dst[5] = (Uint16) ((sample5 + last_sample5) >> 1); dst[6] = (Uint16) ((sample6 + last_sample6) >> 1); dst[7] = (Uint16) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src--; dst[1] = (Sint16) ((sample0 + last_sample0) >> 1); dst[0] = (Sint16) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src += 2; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src--; dst[3] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[2] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[0] = (Sint16) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src += 4; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src -= 2; dst[3] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); src += 4; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src -= 2; dst[7] = (Sint16) ((sample1 + (3 * last_sample1)) >> 2); dst[6] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[5] = (Sint16) ((sample1 + last_sample1) >> 1); dst[4] = (Sint16) ((sample0 + last_sample0) >> 1); dst[3] = (Sint16) (((3 * sample1) + last_sample1) >> 2); dst[2] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); src += 8; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src -= 4; dst[7] = (Sint16) ((sample3 + last_sample3) >> 1); dst[6] = (Sint16) ((sample2 + last_sample2) >> 1); dst[5] = (Sint16) ((sample1 + last_sample1) >> 1); dst[4] = (Sint16) ((sample0 + last_sample0) >> 1); dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); src += 8; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src -= 4; dst[15] = (Sint16) ((sample3 + (3 * last_sample3)) >> 2); dst[14] = (Sint16) ((sample2 + (3 * last_sample2)) >> 2); dst[13] = (Sint16) ((sample1 + (3 * last_sample1)) >> 2); dst[12] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[11] = (Sint16) ((sample3 + last_sample3) >> 1); dst[10] = (Sint16) ((sample2 + last_sample2) >> 1); dst[9] = (Sint16) ((sample1 + last_sample1) >> 1); dst[8] = (Sint16) ((sample0 + last_sample0) >> 1); dst[7] = (Sint16) (((3 * sample3) + last_sample3) >> 2); dst[6] = (Sint16) (((3 * sample2) + last_sample2) >> 2); dst[5] = (Sint16) (((3 * sample1) + last_sample1) >> 2); dst[4] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); src += 16; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src -= 6; dst[11] = (Sint16) ((sample5 + last_sample5) >> 1); dst[10] = (Sint16) ((sample4 + last_sample4) >> 1); dst[9] = (Sint16) ((sample3 + last_sample3) >> 1); dst[8] = (Sint16) ((sample2 + last_sample2) >> 1); dst[7] = (Sint16) ((sample1 + last_sample1) >> 1); dst[6] = (Sint16) ((sample0 + last_sample0) >> 1); dst[5] = (Sint16) sample5; dst[4] = (Sint16) sample4; dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); src += 12; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); dst[4] = (Sint16) ((sample4 + last_sample4) >> 1); dst[5] = (Sint16) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src -= 6; dst[23] = (Sint16) ((sample5 + (3 * last_sample5)) >> 2); dst[22] = (Sint16) ((sample4 + (3 * last_sample4)) >> 2); dst[21] = (Sint16) ((sample3 + (3 * last_sample3)) >> 2); dst[20] = (Sint16) ((sample2 + (3 * last_sample2)) >> 2); dst[19] = (Sint16) ((sample1 + (3 * last_sample1)) >> 2); dst[18] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[17] = (Sint16) ((sample5 + last_sample5) >> 1); dst[16] = (Sint16) ((sample4 + last_sample4) >> 1); dst[15] = (Sint16) ((sample3 + last_sample3) >> 1); dst[14] = (Sint16) ((sample2 + last_sample2) >> 1); dst[13] = (Sint16) ((sample1 + last_sample1) >> 1); dst[12] = (Sint16) ((sample0 + last_sample0) >> 1); dst[11] = (Sint16) (((3 * sample5) + last_sample5) >> 2); dst[10] = (Sint16) (((3 * sample4) + last_sample4) >> 2); dst[9] = (Sint16) (((3 * sample3) + last_sample3) >> 2); dst[8] = (Sint16) (((3 * sample2) + last_sample2) >> 2); dst[7] = (Sint16) (((3 * sample1) + last_sample1) >> 2); dst[6] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[5] = (Sint16) sample5; dst[4] = (Sint16) sample4; dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); src += 24; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); dst[4] = (Sint16) ((sample4 + last_sample4) >> 1); dst[5] = (Sint16) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample7 = (Sint32) ((Sint16) SDL_SwapLE16(src[7])); Sint32 last_sample6 = (Sint32) ((Sint16) SDL_SwapLE16(src[6])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample7 = (Sint32) ((Sint16) SDL_SwapLE16(src[7])); const Sint32 sample6 = (Sint32) ((Sint16) SDL_SwapLE16(src[6])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src -= 8; dst[15] = (Sint16) ((sample7 + last_sample7) >> 1); dst[14] = (Sint16) ((sample6 + last_sample6) >> 1); dst[13] = (Sint16) ((sample5 + last_sample5) >> 1); dst[12] = (Sint16) ((sample4 + last_sample4) >> 1); dst[11] = (Sint16) ((sample3 + last_sample3) >> 1); dst[10] = (Sint16) ((sample2 + last_sample2) >> 1); dst[9] = (Sint16) ((sample1 + last_sample1) >> 1); dst[8] = (Sint16) ((sample0 + last_sample0) >> 1); dst[7] = (Sint16) sample7; dst[6] = (Sint16) sample6; dst[5] = (Sint16) sample5; dst[4] = (Sint16) sample4; dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); Sint32 last_sample6 = (Sint32) ((Sint16) SDL_SwapLE16(src[6])); Sint32 last_sample7 = (Sint32) ((Sint16) SDL_SwapLE16(src[7])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); const Sint32 sample6 = (Sint32) ((Sint16) SDL_SwapLE16(src[6])); const Sint32 sample7 = (Sint32) ((Sint16) SDL_SwapLE16(src[7])); src += 16; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); dst[4] = (Sint16) ((sample4 + last_sample4) >> 1); dst[5] = (Sint16) ((sample5 + last_sample5) >> 1); dst[6] = (Sint16) ((sample6 + last_sample6) >> 1); dst[7] = (Sint16) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16LSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample7 = (Sint32) ((Sint16) SDL_SwapLE16(src[7])); Sint32 last_sample6 = (Sint32) ((Sint16) SDL_SwapLE16(src[6])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); while (dst >= target) { const Sint32 sample7 = (Sint32) ((Sint16) SDL_SwapLE16(src[7])); const Sint32 sample6 = (Sint32) ((Sint16) SDL_SwapLE16(src[6])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); src -= 8; dst[31] = (Sint16) ((sample7 + (3 * last_sample7)) >> 2); dst[30] = (Sint16) ((sample6 + (3 * last_sample6)) >> 2); dst[29] = (Sint16) ((sample5 + (3 * last_sample5)) >> 2); dst[28] = (Sint16) ((sample4 + (3 * last_sample4)) >> 2); dst[27] = (Sint16) ((sample3 + (3 * last_sample3)) >> 2); dst[26] = (Sint16) ((sample2 + (3 * last_sample2)) >> 2); dst[25] = (Sint16) ((sample1 + (3 * last_sample1)) >> 2); dst[24] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[23] = (Sint16) ((sample7 + last_sample7) >> 1); dst[22] = (Sint16) ((sample6 + last_sample6) >> 1); dst[21] = (Sint16) ((sample5 + last_sample5) >> 1); dst[20] = (Sint16) ((sample4 + last_sample4) >> 1); dst[19] = (Sint16) ((sample3 + last_sample3) >> 1); dst[18] = (Sint16) ((sample2 + last_sample2) >> 1); dst[17] = (Sint16) ((sample1 + last_sample1) >> 1); dst[16] = (Sint16) ((sample0 + last_sample0) >> 1); dst[15] = (Sint16) (((3 * sample7) + last_sample7) >> 2); dst[14] = (Sint16) (((3 * sample6) + last_sample6) >> 2); dst[13] = (Sint16) (((3 * sample5) + last_sample5) >> 2); dst[12] = (Sint16) (((3 * sample4) + last_sample4) >> 2); dst[11] = (Sint16) (((3 * sample3) + last_sample3) >> 2); dst[10] = (Sint16) (((3 * sample2) + last_sample2) >> 2); dst[9] = (Sint16) (((3 * sample1) + last_sample1) >> 2); dst[8] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[7] = (Sint16) sample7; dst[6] = (Sint16) sample6; dst[5] = (Sint16) sample5; dst[4] = (Sint16) sample4; dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16LSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); Sint32 last_sample6 = (Sint32) ((Sint16) SDL_SwapLE16(src[6])); Sint32 last_sample7 = (Sint32) ((Sint16) SDL_SwapLE16(src[7])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapLE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapLE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapLE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapLE16(src[3])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapLE16(src[4])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapLE16(src[5])); const Sint32 sample6 = (Sint32) ((Sint16) SDL_SwapLE16(src[6])); const Sint32 sample7 = (Sint32) ((Sint16) SDL_SwapLE16(src[7])); src += 32; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); dst[4] = (Sint16) ((sample4 + last_sample4) >> 1); dst[5] = (Sint16) ((sample5 + last_sample5) >> 1); dst[6] = (Sint16) ((sample6 + last_sample6) >> 1); dst[7] = (Sint16) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src--; dst[1] = (Uint16) ((sample0 + last_sample0) >> 1); dst[0] = (Uint16) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src += 2; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src--; dst[3] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[2] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[0] = (Uint16) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src += 4; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src -= 2; dst[3] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); src += 4; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src -= 2; dst[7] = (Uint16) ((sample1 + (3 * last_sample1)) >> 2); dst[6] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[5] = (Uint16) ((sample1 + last_sample1) >> 1); dst[4] = (Uint16) ((sample0 + last_sample0) >> 1); dst[3] = (Uint16) (((3 * sample1) + last_sample1) >> 2); dst[2] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); src += 8; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src -= 4; dst[7] = (Uint16) ((sample3 + last_sample3) >> 1); dst[6] = (Uint16) ((sample2 + last_sample2) >> 1); dst[5] = (Uint16) ((sample1 + last_sample1) >> 1); dst[4] = (Uint16) ((sample0 + last_sample0) >> 1); dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); src += 8; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src -= 4; dst[15] = (Uint16) ((sample3 + (3 * last_sample3)) >> 2); dst[14] = (Uint16) ((sample2 + (3 * last_sample2)) >> 2); dst[13] = (Uint16) ((sample1 + (3 * last_sample1)) >> 2); dst[12] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[11] = (Uint16) ((sample3 + last_sample3) >> 1); dst[10] = (Uint16) ((sample2 + last_sample2) >> 1); dst[9] = (Uint16) ((sample1 + last_sample1) >> 1); dst[8] = (Uint16) ((sample0 + last_sample0) >> 1); dst[7] = (Uint16) (((3 * sample3) + last_sample3) >> 2); dst[6] = (Uint16) (((3 * sample2) + last_sample2) >> 2); dst[5] = (Uint16) (((3 * sample1) + last_sample1) >> 2); dst[4] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); src += 16; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample5 = (Sint32) SDL_SwapBE16(src[5]); Sint32 last_sample4 = (Sint32) SDL_SwapBE16(src[4]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample5 = (Sint32) SDL_SwapBE16(src[5]); const Sint32 sample4 = (Sint32) SDL_SwapBE16(src[4]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src -= 6; dst[11] = (Uint16) ((sample5 + last_sample5) >> 1); dst[10] = (Uint16) ((sample4 + last_sample4) >> 1); dst[9] = (Uint16) ((sample3 + last_sample3) >> 1); dst[8] = (Uint16) ((sample2 + last_sample2) >> 1); dst[7] = (Uint16) ((sample1 + last_sample1) >> 1); dst[6] = (Uint16) ((sample0 + last_sample0) >> 1); dst[5] = (Uint16) sample5; dst[4] = (Uint16) sample4; dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample4 = (Sint32) SDL_SwapBE16(src[4]); Sint32 last_sample5 = (Sint32) SDL_SwapBE16(src[5]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample4 = (Sint32) SDL_SwapBE16(src[4]); const Sint32 sample5 = (Sint32) SDL_SwapBE16(src[5]); src += 12; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); dst[4] = (Uint16) ((sample4 + last_sample4) >> 1); dst[5] = (Uint16) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample5 = (Sint32) SDL_SwapBE16(src[5]); Sint32 last_sample4 = (Sint32) SDL_SwapBE16(src[4]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample5 = (Sint32) SDL_SwapBE16(src[5]); const Sint32 sample4 = (Sint32) SDL_SwapBE16(src[4]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src -= 6; dst[23] = (Uint16) ((sample5 + (3 * last_sample5)) >> 2); dst[22] = (Uint16) ((sample4 + (3 * last_sample4)) >> 2); dst[21] = (Uint16) ((sample3 + (3 * last_sample3)) >> 2); dst[20] = (Uint16) ((sample2 + (3 * last_sample2)) >> 2); dst[19] = (Uint16) ((sample1 + (3 * last_sample1)) >> 2); dst[18] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[17] = (Uint16) ((sample5 + last_sample5) >> 1); dst[16] = (Uint16) ((sample4 + last_sample4) >> 1); dst[15] = (Uint16) ((sample3 + last_sample3) >> 1); dst[14] = (Uint16) ((sample2 + last_sample2) >> 1); dst[13] = (Uint16) ((sample1 + last_sample1) >> 1); dst[12] = (Uint16) ((sample0 + last_sample0) >> 1); dst[11] = (Uint16) (((3 * sample5) + last_sample5) >> 2); dst[10] = (Uint16) (((3 * sample4) + last_sample4) >> 2); dst[9] = (Uint16) (((3 * sample3) + last_sample3) >> 2); dst[8] = (Uint16) (((3 * sample2) + last_sample2) >> 2); dst[7] = (Uint16) (((3 * sample1) + last_sample1) >> 2); dst[6] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[5] = (Uint16) sample5; dst[4] = (Uint16) sample4; dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample4 = (Sint32) SDL_SwapBE16(src[4]); Sint32 last_sample5 = (Sint32) SDL_SwapBE16(src[5]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample4 = (Sint32) SDL_SwapBE16(src[4]); const Sint32 sample5 = (Sint32) SDL_SwapBE16(src[5]); src += 24; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); dst[4] = (Uint16) ((sample4 + last_sample4) >> 1); dst[5] = (Uint16) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_U16MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8 * 2; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample7 = (Sint32) SDL_SwapBE16(src[7]); Sint32 last_sample6 = (Sint32) SDL_SwapBE16(src[6]); Sint32 last_sample5 = (Sint32) SDL_SwapBE16(src[5]); Sint32 last_sample4 = (Sint32) SDL_SwapBE16(src[4]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample7 = (Sint32) SDL_SwapBE16(src[7]); const Sint32 sample6 = (Sint32) SDL_SwapBE16(src[6]); const Sint32 sample5 = (Sint32) SDL_SwapBE16(src[5]); const Sint32 sample4 = (Sint32) SDL_SwapBE16(src[4]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src -= 8; dst[15] = (Uint16) ((sample7 + last_sample7) >> 1); dst[14] = (Uint16) ((sample6 + last_sample6) >> 1); dst[13] = (Uint16) ((sample5 + last_sample5) >> 1); dst[12] = (Uint16) ((sample4 + last_sample4) >> 1); dst[11] = (Uint16) ((sample3 + last_sample3) >> 1); dst[10] = (Uint16) ((sample2 + last_sample2) >> 1); dst[9] = (Uint16) ((sample1 + last_sample1) >> 1); dst[8] = (Uint16) ((sample0 + last_sample0) >> 1); dst[7] = (Uint16) sample7; dst[6] = (Uint16) sample6; dst[5] = (Uint16) sample5; dst[4] = (Uint16) sample4; dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_U16MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample4 = (Sint32) SDL_SwapBE16(src[4]); Sint32 last_sample5 = (Sint32) SDL_SwapBE16(src[5]); Sint32 last_sample6 = (Sint32) SDL_SwapBE16(src[6]); Sint32 last_sample7 = (Sint32) SDL_SwapBE16(src[7]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample4 = (Sint32) SDL_SwapBE16(src[4]); const Sint32 sample5 = (Sint32) SDL_SwapBE16(src[5]); const Sint32 sample6 = (Sint32) SDL_SwapBE16(src[6]); const Sint32 sample7 = (Sint32) SDL_SwapBE16(src[7]); src += 16; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); dst[4] = (Uint16) ((sample4 + last_sample4) >> 1); dst[5] = (Uint16) ((sample5 + last_sample5) >> 1); dst[6] = (Uint16) ((sample6 + last_sample6) >> 1); dst[7] = (Uint16) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_U16MSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_U16MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8 * 4; const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Uint16 *target = ((const Uint16 *) cvt->buf); Sint32 last_sample7 = (Sint32) SDL_SwapBE16(src[7]); Sint32 last_sample6 = (Sint32) SDL_SwapBE16(src[6]); Sint32 last_sample5 = (Sint32) SDL_SwapBE16(src[5]); Sint32 last_sample4 = (Sint32) SDL_SwapBE16(src[4]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); while (dst >= target) { const Sint32 sample7 = (Sint32) SDL_SwapBE16(src[7]); const Sint32 sample6 = (Sint32) SDL_SwapBE16(src[6]); const Sint32 sample5 = (Sint32) SDL_SwapBE16(src[5]); const Sint32 sample4 = (Sint32) SDL_SwapBE16(src[4]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); src -= 8; dst[31] = (Uint16) ((sample7 + (3 * last_sample7)) >> 2); dst[30] = (Uint16) ((sample6 + (3 * last_sample6)) >> 2); dst[29] = (Uint16) ((sample5 + (3 * last_sample5)) >> 2); dst[28] = (Uint16) ((sample4 + (3 * last_sample4)) >> 2); dst[27] = (Uint16) ((sample3 + (3 * last_sample3)) >> 2); dst[26] = (Uint16) ((sample2 + (3 * last_sample2)) >> 2); dst[25] = (Uint16) ((sample1 + (3 * last_sample1)) >> 2); dst[24] = (Uint16) ((sample0 + (3 * last_sample0)) >> 2); dst[23] = (Uint16) ((sample7 + last_sample7) >> 1); dst[22] = (Uint16) ((sample6 + last_sample6) >> 1); dst[21] = (Uint16) ((sample5 + last_sample5) >> 1); dst[20] = (Uint16) ((sample4 + last_sample4) >> 1); dst[19] = (Uint16) ((sample3 + last_sample3) >> 1); dst[18] = (Uint16) ((sample2 + last_sample2) >> 1); dst[17] = (Uint16) ((sample1 + last_sample1) >> 1); dst[16] = (Uint16) ((sample0 + last_sample0) >> 1); dst[15] = (Uint16) (((3 * sample7) + last_sample7) >> 2); dst[14] = (Uint16) (((3 * sample6) + last_sample6) >> 2); dst[13] = (Uint16) (((3 * sample5) + last_sample5) >> 2); dst[12] = (Uint16) (((3 * sample4) + last_sample4) >> 2); dst[11] = (Uint16) (((3 * sample3) + last_sample3) >> 2); dst[10] = (Uint16) (((3 * sample2) + last_sample2) >> 2); dst[9] = (Uint16) (((3 * sample1) + last_sample1) >> 2); dst[8] = (Uint16) (((3 * sample0) + last_sample0) >> 2); dst[7] = (Uint16) sample7; dst[6] = (Uint16) sample6; dst[5] = (Uint16) sample5; dst[4] = (Uint16) sample4; dst[3] = (Uint16) sample3; dst[2] = (Uint16) sample2; dst[1] = (Uint16) sample1; dst[0] = (Uint16) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_U16MSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_U16MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Uint16 *dst = (Uint16 *) cvt->buf; const Uint16 *src = (Uint16 *) cvt->buf; const Uint16 *target = (const Uint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) SDL_SwapBE16(src[0]); Sint32 last_sample1 = (Sint32) SDL_SwapBE16(src[1]); Sint32 last_sample2 = (Sint32) SDL_SwapBE16(src[2]); Sint32 last_sample3 = (Sint32) SDL_SwapBE16(src[3]); Sint32 last_sample4 = (Sint32) SDL_SwapBE16(src[4]); Sint32 last_sample5 = (Sint32) SDL_SwapBE16(src[5]); Sint32 last_sample6 = (Sint32) SDL_SwapBE16(src[6]); Sint32 last_sample7 = (Sint32) SDL_SwapBE16(src[7]); while (dst < target) { const Sint32 sample0 = (Sint32) SDL_SwapBE16(src[0]); const Sint32 sample1 = (Sint32) SDL_SwapBE16(src[1]); const Sint32 sample2 = (Sint32) SDL_SwapBE16(src[2]); const Sint32 sample3 = (Sint32) SDL_SwapBE16(src[3]); const Sint32 sample4 = (Sint32) SDL_SwapBE16(src[4]); const Sint32 sample5 = (Sint32) SDL_SwapBE16(src[5]); const Sint32 sample6 = (Sint32) SDL_SwapBE16(src[6]); const Sint32 sample7 = (Sint32) SDL_SwapBE16(src[7]); src += 32; dst[0] = (Uint16) ((sample0 + last_sample0) >> 1); dst[1] = (Uint16) ((sample1 + last_sample1) >> 1); dst[2] = (Uint16) ((sample2 + last_sample2) >> 1); dst[3] = (Uint16) ((sample3 + last_sample3) >> 1); dst[4] = (Uint16) ((sample4 + last_sample4) >> 1); dst[5] = (Uint16) ((sample5 + last_sample5) >> 1); dst[6] = (Uint16) ((sample6 + last_sample6) >> 1); dst[7] = (Uint16) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src--; dst[1] = (Sint16) ((sample0 + last_sample0) >> 1); dst[0] = (Sint16) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src += 2; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src--; dst[3] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[2] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[0] = (Sint16) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src += 4; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src -= 2; dst[3] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); src += 4; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src -= 2; dst[7] = (Sint16) ((sample1 + (3 * last_sample1)) >> 2); dst[6] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[5] = (Sint16) ((sample1 + last_sample1) >> 1); dst[4] = (Sint16) ((sample0 + last_sample0) >> 1); dst[3] = (Sint16) (((3 * sample1) + last_sample1) >> 2); dst[2] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); src += 8; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src -= 4; dst[7] = (Sint16) ((sample3 + last_sample3) >> 1); dst[6] = (Sint16) ((sample2 + last_sample2) >> 1); dst[5] = (Sint16) ((sample1 + last_sample1) >> 1); dst[4] = (Sint16) ((sample0 + last_sample0) >> 1); dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); src += 8; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src -= 4; dst[15] = (Sint16) ((sample3 + (3 * last_sample3)) >> 2); dst[14] = (Sint16) ((sample2 + (3 * last_sample2)) >> 2); dst[13] = (Sint16) ((sample1 + (3 * last_sample1)) >> 2); dst[12] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[11] = (Sint16) ((sample3 + last_sample3) >> 1); dst[10] = (Sint16) ((sample2 + last_sample2) >> 1); dst[9] = (Sint16) ((sample1 + last_sample1) >> 1); dst[8] = (Sint16) ((sample0 + last_sample0) >> 1); dst[7] = (Sint16) (((3 * sample3) + last_sample3) >> 2); dst[6] = (Sint16) (((3 * sample2) + last_sample2) >> 2); dst[5] = (Sint16) (((3 * sample1) + last_sample1) >> 2); dst[4] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); src += 16; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src -= 6; dst[11] = (Sint16) ((sample5 + last_sample5) >> 1); dst[10] = (Sint16) ((sample4 + last_sample4) >> 1); dst[9] = (Sint16) ((sample3 + last_sample3) >> 1); dst[8] = (Sint16) ((sample2 + last_sample2) >> 1); dst[7] = (Sint16) ((sample1 + last_sample1) >> 1); dst[6] = (Sint16) ((sample0 + last_sample0) >> 1); dst[5] = (Sint16) sample5; dst[4] = (Sint16) sample4; dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); src += 12; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); dst[4] = (Sint16) ((sample4 + last_sample4) >> 1); dst[5] = (Sint16) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src -= 6; dst[23] = (Sint16) ((sample5 + (3 * last_sample5)) >> 2); dst[22] = (Sint16) ((sample4 + (3 * last_sample4)) >> 2); dst[21] = (Sint16) ((sample3 + (3 * last_sample3)) >> 2); dst[20] = (Sint16) ((sample2 + (3 * last_sample2)) >> 2); dst[19] = (Sint16) ((sample1 + (3 * last_sample1)) >> 2); dst[18] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[17] = (Sint16) ((sample5 + last_sample5) >> 1); dst[16] = (Sint16) ((sample4 + last_sample4) >> 1); dst[15] = (Sint16) ((sample3 + last_sample3) >> 1); dst[14] = (Sint16) ((sample2 + last_sample2) >> 1); dst[13] = (Sint16) ((sample1 + last_sample1) >> 1); dst[12] = (Sint16) ((sample0 + last_sample0) >> 1); dst[11] = (Sint16) (((3 * sample5) + last_sample5) >> 2); dst[10] = (Sint16) (((3 * sample4) + last_sample4) >> 2); dst[9] = (Sint16) (((3 * sample3) + last_sample3) >> 2); dst[8] = (Sint16) (((3 * sample2) + last_sample2) >> 2); dst[7] = (Sint16) (((3 * sample1) + last_sample1) >> 2); dst[6] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[5] = (Sint16) sample5; dst[4] = (Sint16) sample4; dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); src += 24; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); dst[4] = (Sint16) ((sample4 + last_sample4) >> 1); dst[5] = (Sint16) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S16MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8 * 2; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample7 = (Sint32) ((Sint16) SDL_SwapBE16(src[7])); Sint32 last_sample6 = (Sint32) ((Sint16) SDL_SwapBE16(src[6])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample7 = (Sint32) ((Sint16) SDL_SwapBE16(src[7])); const Sint32 sample6 = (Sint32) ((Sint16) SDL_SwapBE16(src[6])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src -= 8; dst[15] = (Sint16) ((sample7 + last_sample7) >> 1); dst[14] = (Sint16) ((sample6 + last_sample6) >> 1); dst[13] = (Sint16) ((sample5 + last_sample5) >> 1); dst[12] = (Sint16) ((sample4 + last_sample4) >> 1); dst[11] = (Sint16) ((sample3 + last_sample3) >> 1); dst[10] = (Sint16) ((sample2 + last_sample2) >> 1); dst[9] = (Sint16) ((sample1 + last_sample1) >> 1); dst[8] = (Sint16) ((sample0 + last_sample0) >> 1); dst[7] = (Sint16) sample7; dst[6] = (Sint16) sample6; dst[5] = (Sint16) sample5; dst[4] = (Sint16) sample4; dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S16MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); Sint32 last_sample6 = (Sint32) ((Sint16) SDL_SwapBE16(src[6])); Sint32 last_sample7 = (Sint32) ((Sint16) SDL_SwapBE16(src[7])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); const Sint32 sample6 = (Sint32) ((Sint16) SDL_SwapBE16(src[6])); const Sint32 sample7 = (Sint32) ((Sint16) SDL_SwapBE16(src[7])); src += 16; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); dst[4] = (Sint16) ((sample4 + last_sample4) >> 1); dst[5] = (Sint16) ((sample5 + last_sample5) >> 1); dst[6] = (Sint16) ((sample6 + last_sample6) >> 1); dst[7] = (Sint16) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S16MSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S16MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8 * 4; const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint16 *target = ((const Sint16 *) cvt->buf); Sint32 last_sample7 = (Sint32) ((Sint16) SDL_SwapBE16(src[7])); Sint32 last_sample6 = (Sint32) ((Sint16) SDL_SwapBE16(src[6])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); while (dst >= target) { const Sint32 sample7 = (Sint32) ((Sint16) SDL_SwapBE16(src[7])); const Sint32 sample6 = (Sint32) ((Sint16) SDL_SwapBE16(src[6])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); src -= 8; dst[31] = (Sint16) ((sample7 + (3 * last_sample7)) >> 2); dst[30] = (Sint16) ((sample6 + (3 * last_sample6)) >> 2); dst[29] = (Sint16) ((sample5 + (3 * last_sample5)) >> 2); dst[28] = (Sint16) ((sample4 + (3 * last_sample4)) >> 2); dst[27] = (Sint16) ((sample3 + (3 * last_sample3)) >> 2); dst[26] = (Sint16) ((sample2 + (3 * last_sample2)) >> 2); dst[25] = (Sint16) ((sample1 + (3 * last_sample1)) >> 2); dst[24] = (Sint16) ((sample0 + (3 * last_sample0)) >> 2); dst[23] = (Sint16) ((sample7 + last_sample7) >> 1); dst[22] = (Sint16) ((sample6 + last_sample6) >> 1); dst[21] = (Sint16) ((sample5 + last_sample5) >> 1); dst[20] = (Sint16) ((sample4 + last_sample4) >> 1); dst[19] = (Sint16) ((sample3 + last_sample3) >> 1); dst[18] = (Sint16) ((sample2 + last_sample2) >> 1); dst[17] = (Sint16) ((sample1 + last_sample1) >> 1); dst[16] = (Sint16) ((sample0 + last_sample0) >> 1); dst[15] = (Sint16) (((3 * sample7) + last_sample7) >> 2); dst[14] = (Sint16) (((3 * sample6) + last_sample6) >> 2); dst[13] = (Sint16) (((3 * sample5) + last_sample5) >> 2); dst[12] = (Sint16) (((3 * sample4) + last_sample4) >> 2); dst[11] = (Sint16) (((3 * sample3) + last_sample3) >> 2); dst[10] = (Sint16) (((3 * sample2) + last_sample2) >> 2); dst[9] = (Sint16) (((3 * sample1) + last_sample1) >> 2); dst[8] = (Sint16) (((3 * sample0) + last_sample0) >> 2); dst[7] = (Sint16) sample7; dst[6] = (Sint16) sample6; dst[5] = (Sint16) sample5; dst[4] = (Sint16) sample4; dst[3] = (Sint16) sample3; dst[2] = (Sint16) sample2; dst[1] = (Sint16) sample1; dst[0] = (Sint16) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S16MSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S16MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint16 *dst = (Sint16 *) cvt->buf; const Sint16 *src = (Sint16 *) cvt->buf; const Sint16 *target = (const Sint16 *) (cvt->buf + dstsize); Sint32 last_sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); Sint32 last_sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); Sint32 last_sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); Sint32 last_sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); Sint32 last_sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); Sint32 last_sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); Sint32 last_sample6 = (Sint32) ((Sint16) SDL_SwapBE16(src[6])); Sint32 last_sample7 = (Sint32) ((Sint16) SDL_SwapBE16(src[7])); while (dst < target) { const Sint32 sample0 = (Sint32) ((Sint16) SDL_SwapBE16(src[0])); const Sint32 sample1 = (Sint32) ((Sint16) SDL_SwapBE16(src[1])); const Sint32 sample2 = (Sint32) ((Sint16) SDL_SwapBE16(src[2])); const Sint32 sample3 = (Sint32) ((Sint16) SDL_SwapBE16(src[3])); const Sint32 sample4 = (Sint32) ((Sint16) SDL_SwapBE16(src[4])); const Sint32 sample5 = (Sint32) ((Sint16) SDL_SwapBE16(src[5])); const Sint32 sample6 = (Sint32) ((Sint16) SDL_SwapBE16(src[6])); const Sint32 sample7 = (Sint32) ((Sint16) SDL_SwapBE16(src[7])); src += 32; dst[0] = (Sint16) ((sample0 + last_sample0) >> 1); dst[1] = (Sint16) ((sample1 + last_sample1) >> 1); dst[2] = (Sint16) ((sample2 + last_sample2) >> 1); dst[3] = (Sint16) ((sample3 + last_sample3) >> 1); dst[4] = (Sint16) ((sample4 + last_sample4) >> 1); dst[5] = (Sint16) ((sample5 + last_sample5) >> 1); dst[6] = (Sint16) ((sample6 + last_sample6) >> 1); dst[7] = (Sint16) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src--; dst[1] = (Sint32) ((sample0 + last_sample0) >> 1); dst[0] = (Sint32) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src += 2; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src--; dst[3] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[2] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[0] = (Sint32) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src += 4; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src -= 2; dst[3] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); src += 4; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src -= 2; dst[7] = (Sint32) ((sample1 + (3 * last_sample1)) >> 2); dst[6] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[5] = (Sint32) ((sample1 + last_sample1) >> 1); dst[4] = (Sint32) ((sample0 + last_sample0) >> 1); dst[3] = (Sint32) (((3 * sample1) + last_sample1) >> 2); dst[2] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); src += 8; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src -= 4; dst[7] = (Sint32) ((sample3 + last_sample3) >> 1); dst[6] = (Sint32) ((sample2 + last_sample2) >> 1); dst[5] = (Sint32) ((sample1 + last_sample1) >> 1); dst[4] = (Sint32) ((sample0 + last_sample0) >> 1); dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); src += 8; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src -= 4; dst[15] = (Sint32) ((sample3 + (3 * last_sample3)) >> 2); dst[14] = (Sint32) ((sample2 + (3 * last_sample2)) >> 2); dst[13] = (Sint32) ((sample1 + (3 * last_sample1)) >> 2); dst[12] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[11] = (Sint32) ((sample3 + last_sample3) >> 1); dst[10] = (Sint32) ((sample2 + last_sample2) >> 1); dst[9] = (Sint32) ((sample1 + last_sample1) >> 1); dst[8] = (Sint32) ((sample0 + last_sample0) >> 1); dst[7] = (Sint32) (((3 * sample3) + last_sample3) >> 2); dst[6] = (Sint32) (((3 * sample2) + last_sample2) >> 2); dst[5] = (Sint32) (((3 * sample1) + last_sample1) >> 2); dst[4] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); src += 16; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src -= 6; dst[11] = (Sint32) ((sample5 + last_sample5) >> 1); dst[10] = (Sint32) ((sample4 + last_sample4) >> 1); dst[9] = (Sint32) ((sample3 + last_sample3) >> 1); dst[8] = (Sint32) ((sample2 + last_sample2) >> 1); dst[7] = (Sint32) ((sample1 + last_sample1) >> 1); dst[6] = (Sint32) ((sample0 + last_sample0) >> 1); dst[5] = (Sint32) sample5; dst[4] = (Sint32) sample4; dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); src += 12; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); dst[4] = (Sint32) ((sample4 + last_sample4) >> 1); dst[5] = (Sint32) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src -= 6; dst[23] = (Sint32) ((sample5 + (3 * last_sample5)) >> 2); dst[22] = (Sint32) ((sample4 + (3 * last_sample4)) >> 2); dst[21] = (Sint32) ((sample3 + (3 * last_sample3)) >> 2); dst[20] = (Sint32) ((sample2 + (3 * last_sample2)) >> 2); dst[19] = (Sint32) ((sample1 + (3 * last_sample1)) >> 2); dst[18] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[17] = (Sint32) ((sample5 + last_sample5) >> 1); dst[16] = (Sint32) ((sample4 + last_sample4) >> 1); dst[15] = (Sint32) ((sample3 + last_sample3) >> 1); dst[14] = (Sint32) ((sample2 + last_sample2) >> 1); dst[13] = (Sint32) ((sample1 + last_sample1) >> 1); dst[12] = (Sint32) ((sample0 + last_sample0) >> 1); dst[11] = (Sint32) (((3 * sample5) + last_sample5) >> 2); dst[10] = (Sint32) (((3 * sample4) + last_sample4) >> 2); dst[9] = (Sint32) (((3 * sample3) + last_sample3) >> 2); dst[8] = (Sint32) (((3 * sample2) + last_sample2) >> 2); dst[7] = (Sint32) (((3 * sample1) + last_sample1) >> 2); dst[6] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[5] = (Sint32) sample5; dst[4] = (Sint32) sample4; dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); src += 24; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); dst[4] = (Sint32) ((sample4 + last_sample4) >> 1); dst[5] = (Sint32) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample7 = (Sint64) ((Sint32) SDL_SwapLE32(src[7])); Sint64 last_sample6 = (Sint64) ((Sint32) SDL_SwapLE32(src[6])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample7 = (Sint64) ((Sint32) SDL_SwapLE32(src[7])); const Sint64 sample6 = (Sint64) ((Sint32) SDL_SwapLE32(src[6])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src -= 8; dst[15] = (Sint32) ((sample7 + last_sample7) >> 1); dst[14] = (Sint32) ((sample6 + last_sample6) >> 1); dst[13] = (Sint32) ((sample5 + last_sample5) >> 1); dst[12] = (Sint32) ((sample4 + last_sample4) >> 1); dst[11] = (Sint32) ((sample3 + last_sample3) >> 1); dst[10] = (Sint32) ((sample2 + last_sample2) >> 1); dst[9] = (Sint32) ((sample1 + last_sample1) >> 1); dst[8] = (Sint32) ((sample0 + last_sample0) >> 1); dst[7] = (Sint32) sample7; dst[6] = (Sint32) sample6; dst[5] = (Sint32) sample5; dst[4] = (Sint32) sample4; dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); Sint64 last_sample6 = (Sint64) ((Sint32) SDL_SwapLE32(src[6])); Sint64 last_sample7 = (Sint64) ((Sint32) SDL_SwapLE32(src[7])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); const Sint64 sample6 = (Sint64) ((Sint32) SDL_SwapLE32(src[6])); const Sint64 sample7 = (Sint64) ((Sint32) SDL_SwapLE32(src[7])); src += 16; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); dst[4] = (Sint32) ((sample4 + last_sample4) >> 1); dst[5] = (Sint32) ((sample5 + last_sample5) >> 1); dst[6] = (Sint32) ((sample6 + last_sample6) >> 1); dst[7] = (Sint32) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32LSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample7 = (Sint64) ((Sint32) SDL_SwapLE32(src[7])); Sint64 last_sample6 = (Sint64) ((Sint32) SDL_SwapLE32(src[6])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); while (dst >= target) { const Sint64 sample7 = (Sint64) ((Sint32) SDL_SwapLE32(src[7])); const Sint64 sample6 = (Sint64) ((Sint32) SDL_SwapLE32(src[6])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); src -= 8; dst[31] = (Sint32) ((sample7 + (3 * last_sample7)) >> 2); dst[30] = (Sint32) ((sample6 + (3 * last_sample6)) >> 2); dst[29] = (Sint32) ((sample5 + (3 * last_sample5)) >> 2); dst[28] = (Sint32) ((sample4 + (3 * last_sample4)) >> 2); dst[27] = (Sint32) ((sample3 + (3 * last_sample3)) >> 2); dst[26] = (Sint32) ((sample2 + (3 * last_sample2)) >> 2); dst[25] = (Sint32) ((sample1 + (3 * last_sample1)) >> 2); dst[24] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[23] = (Sint32) ((sample7 + last_sample7) >> 1); dst[22] = (Sint32) ((sample6 + last_sample6) >> 1); dst[21] = (Sint32) ((sample5 + last_sample5) >> 1); dst[20] = (Sint32) ((sample4 + last_sample4) >> 1); dst[19] = (Sint32) ((sample3 + last_sample3) >> 1); dst[18] = (Sint32) ((sample2 + last_sample2) >> 1); dst[17] = (Sint32) ((sample1 + last_sample1) >> 1); dst[16] = (Sint32) ((sample0 + last_sample0) >> 1); dst[15] = (Sint32) (((3 * sample7) + last_sample7) >> 2); dst[14] = (Sint32) (((3 * sample6) + last_sample6) >> 2); dst[13] = (Sint32) (((3 * sample5) + last_sample5) >> 2); dst[12] = (Sint32) (((3 * sample4) + last_sample4) >> 2); dst[11] = (Sint32) (((3 * sample3) + last_sample3) >> 2); dst[10] = (Sint32) (((3 * sample2) + last_sample2) >> 2); dst[9] = (Sint32) (((3 * sample1) + last_sample1) >> 2); dst[8] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[7] = (Sint32) sample7; dst[6] = (Sint32) sample6; dst[5] = (Sint32) sample5; dst[4] = (Sint32) sample4; dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32LSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); Sint64 last_sample6 = (Sint64) ((Sint32) SDL_SwapLE32(src[6])); Sint64 last_sample7 = (Sint64) ((Sint32) SDL_SwapLE32(src[7])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapLE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapLE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapLE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapLE32(src[3])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapLE32(src[4])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapLE32(src[5])); const Sint64 sample6 = (Sint64) ((Sint32) SDL_SwapLE32(src[6])); const Sint64 sample7 = (Sint64) ((Sint32) SDL_SwapLE32(src[7])); src += 32; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); dst[4] = (Sint32) ((sample4 + last_sample4) >> 1); dst[5] = (Sint32) ((sample5 + last_sample5) >> 1); dst[6] = (Sint32) ((sample6 + last_sample6) >> 1); dst[7] = (Sint32) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src--; dst[1] = (Sint32) ((sample0 + last_sample0) >> 1); dst[0] = (Sint32) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src += 2; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src--; dst[3] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[2] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[0] = (Sint32) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src += 4; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src -= 2; dst[3] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); src += 4; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src -= 2; dst[7] = (Sint32) ((sample1 + (3 * last_sample1)) >> 2); dst[6] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[5] = (Sint32) ((sample1 + last_sample1) >> 1); dst[4] = (Sint32) ((sample0 + last_sample0) >> 1); dst[3] = (Sint32) (((3 * sample1) + last_sample1) >> 2); dst[2] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); src += 8; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src -= 4; dst[7] = (Sint32) ((sample3 + last_sample3) >> 1); dst[6] = (Sint32) ((sample2 + last_sample2) >> 1); dst[5] = (Sint32) ((sample1 + last_sample1) >> 1); dst[4] = (Sint32) ((sample0 + last_sample0) >> 1); dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); src += 8; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src -= 4; dst[15] = (Sint32) ((sample3 + (3 * last_sample3)) >> 2); dst[14] = (Sint32) ((sample2 + (3 * last_sample2)) >> 2); dst[13] = (Sint32) ((sample1 + (3 * last_sample1)) >> 2); dst[12] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[11] = (Sint32) ((sample3 + last_sample3) >> 1); dst[10] = (Sint32) ((sample2 + last_sample2) >> 1); dst[9] = (Sint32) ((sample1 + last_sample1) >> 1); dst[8] = (Sint32) ((sample0 + last_sample0) >> 1); dst[7] = (Sint32) (((3 * sample3) + last_sample3) >> 2); dst[6] = (Sint32) (((3 * sample2) + last_sample2) >> 2); dst[5] = (Sint32) (((3 * sample1) + last_sample1) >> 2); dst[4] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); src += 16; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src -= 6; dst[11] = (Sint32) ((sample5 + last_sample5) >> 1); dst[10] = (Sint32) ((sample4 + last_sample4) >> 1); dst[9] = (Sint32) ((sample3 + last_sample3) >> 1); dst[8] = (Sint32) ((sample2 + last_sample2) >> 1); dst[7] = (Sint32) ((sample1 + last_sample1) >> 1); dst[6] = (Sint32) ((sample0 + last_sample0) >> 1); dst[5] = (Sint32) sample5; dst[4] = (Sint32) sample4; dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); src += 12; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); dst[4] = (Sint32) ((sample4 + last_sample4) >> 1); dst[5] = (Sint32) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src -= 6; dst[23] = (Sint32) ((sample5 + (3 * last_sample5)) >> 2); dst[22] = (Sint32) ((sample4 + (3 * last_sample4)) >> 2); dst[21] = (Sint32) ((sample3 + (3 * last_sample3)) >> 2); dst[20] = (Sint32) ((sample2 + (3 * last_sample2)) >> 2); dst[19] = (Sint32) ((sample1 + (3 * last_sample1)) >> 2); dst[18] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[17] = (Sint32) ((sample5 + last_sample5) >> 1); dst[16] = (Sint32) ((sample4 + last_sample4) >> 1); dst[15] = (Sint32) ((sample3 + last_sample3) >> 1); dst[14] = (Sint32) ((sample2 + last_sample2) >> 1); dst[13] = (Sint32) ((sample1 + last_sample1) >> 1); dst[12] = (Sint32) ((sample0 + last_sample0) >> 1); dst[11] = (Sint32) (((3 * sample5) + last_sample5) >> 2); dst[10] = (Sint32) (((3 * sample4) + last_sample4) >> 2); dst[9] = (Sint32) (((3 * sample3) + last_sample3) >> 2); dst[8] = (Sint32) (((3 * sample2) + last_sample2) >> 2); dst[7] = (Sint32) (((3 * sample1) + last_sample1) >> 2); dst[6] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[5] = (Sint32) sample5; dst[4] = (Sint32) sample4; dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); src += 24; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); dst[4] = (Sint32) ((sample4 + last_sample4) >> 1); dst[5] = (Sint32) ((sample5 + last_sample5) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_S32MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8 * 2; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample7 = (Sint64) ((Sint32) SDL_SwapBE32(src[7])); Sint64 last_sample6 = (Sint64) ((Sint32) SDL_SwapBE32(src[6])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample7 = (Sint64) ((Sint32) SDL_SwapBE32(src[7])); const Sint64 sample6 = (Sint64) ((Sint32) SDL_SwapBE32(src[6])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src -= 8; dst[15] = (Sint32) ((sample7 + last_sample7) >> 1); dst[14] = (Sint32) ((sample6 + last_sample6) >> 1); dst[13] = (Sint32) ((sample5 + last_sample5) >> 1); dst[12] = (Sint32) ((sample4 + last_sample4) >> 1); dst[11] = (Sint32) ((sample3 + last_sample3) >> 1); dst[10] = (Sint32) ((sample2 + last_sample2) >> 1); dst[9] = (Sint32) ((sample1 + last_sample1) >> 1); dst[8] = (Sint32) ((sample0 + last_sample0) >> 1); dst[7] = (Sint32) sample7; dst[6] = (Sint32) sample6; dst[5] = (Sint32) sample5; dst[4] = (Sint32) sample4; dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_S32MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); Sint64 last_sample6 = (Sint64) ((Sint32) SDL_SwapBE32(src[6])); Sint64 last_sample7 = (Sint64) ((Sint32) SDL_SwapBE32(src[7])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); const Sint64 sample6 = (Sint64) ((Sint32) SDL_SwapBE32(src[6])); const Sint64 sample7 = (Sint64) ((Sint32) SDL_SwapBE32(src[7])); src += 16; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); dst[4] = (Sint32) ((sample4 + last_sample4) >> 1); dst[5] = (Sint32) ((sample5 + last_sample5) >> 1); dst[6] = (Sint32) ((sample6 + last_sample6) >> 1); dst[7] = (Sint32) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_S32MSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_S32MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8 * 4; const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8; const Sint32 *target = ((const Sint32 *) cvt->buf); Sint64 last_sample7 = (Sint64) ((Sint32) SDL_SwapBE32(src[7])); Sint64 last_sample6 = (Sint64) ((Sint32) SDL_SwapBE32(src[6])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); while (dst >= target) { const Sint64 sample7 = (Sint64) ((Sint32) SDL_SwapBE32(src[7])); const Sint64 sample6 = (Sint64) ((Sint32) SDL_SwapBE32(src[6])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); src -= 8; dst[31] = (Sint32) ((sample7 + (3 * last_sample7)) >> 2); dst[30] = (Sint32) ((sample6 + (3 * last_sample6)) >> 2); dst[29] = (Sint32) ((sample5 + (3 * last_sample5)) >> 2); dst[28] = (Sint32) ((sample4 + (3 * last_sample4)) >> 2); dst[27] = (Sint32) ((sample3 + (3 * last_sample3)) >> 2); dst[26] = (Sint32) ((sample2 + (3 * last_sample2)) >> 2); dst[25] = (Sint32) ((sample1 + (3 * last_sample1)) >> 2); dst[24] = (Sint32) ((sample0 + (3 * last_sample0)) >> 2); dst[23] = (Sint32) ((sample7 + last_sample7) >> 1); dst[22] = (Sint32) ((sample6 + last_sample6) >> 1); dst[21] = (Sint32) ((sample5 + last_sample5) >> 1); dst[20] = (Sint32) ((sample4 + last_sample4) >> 1); dst[19] = (Sint32) ((sample3 + last_sample3) >> 1); dst[18] = (Sint32) ((sample2 + last_sample2) >> 1); dst[17] = (Sint32) ((sample1 + last_sample1) >> 1); dst[16] = (Sint32) ((sample0 + last_sample0) >> 1); dst[15] = (Sint32) (((3 * sample7) + last_sample7) >> 2); dst[14] = (Sint32) (((3 * sample6) + last_sample6) >> 2); dst[13] = (Sint32) (((3 * sample5) + last_sample5) >> 2); dst[12] = (Sint32) (((3 * sample4) + last_sample4) >> 2); dst[11] = (Sint32) (((3 * sample3) + last_sample3) >> 2); dst[10] = (Sint32) (((3 * sample2) + last_sample2) >> 2); dst[9] = (Sint32) (((3 * sample1) + last_sample1) >> 2); dst[8] = (Sint32) (((3 * sample0) + last_sample0) >> 2); dst[7] = (Sint32) sample7; dst[6] = (Sint32) sample6; dst[5] = (Sint32) sample5; dst[4] = (Sint32) sample4; dst[3] = (Sint32) sample3; dst[2] = (Sint32) sample2; dst[1] = (Sint32) sample1; dst[0] = (Sint32) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_S32MSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_S32MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; Sint32 *dst = (Sint32 *) cvt->buf; const Sint32 *src = (Sint32 *) cvt->buf; const Sint32 *target = (const Sint32 *) (cvt->buf + dstsize); Sint64 last_sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); Sint64 last_sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); Sint64 last_sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); Sint64 last_sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); Sint64 last_sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); Sint64 last_sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); Sint64 last_sample6 = (Sint64) ((Sint32) SDL_SwapBE32(src[6])); Sint64 last_sample7 = (Sint64) ((Sint32) SDL_SwapBE32(src[7])); while (dst < target) { const Sint64 sample0 = (Sint64) ((Sint32) SDL_SwapBE32(src[0])); const Sint64 sample1 = (Sint64) ((Sint32) SDL_SwapBE32(src[1])); const Sint64 sample2 = (Sint64) ((Sint32) SDL_SwapBE32(src[2])); const Sint64 sample3 = (Sint64) ((Sint32) SDL_SwapBE32(src[3])); const Sint64 sample4 = (Sint64) ((Sint32) SDL_SwapBE32(src[4])); const Sint64 sample5 = (Sint64) ((Sint32) SDL_SwapBE32(src[5])); const Sint64 sample6 = (Sint64) ((Sint32) SDL_SwapBE32(src[6])); const Sint64 sample7 = (Sint64) ((Sint32) SDL_SwapBE32(src[7])); src += 32; dst[0] = (Sint32) ((sample0 + last_sample0) >> 1); dst[1] = (Sint32) ((sample1 + last_sample1) >> 1); dst[2] = (Sint32) ((sample2 + last_sample2) >> 1); dst[3] = (Sint32) ((sample3 + last_sample3) >> 1); dst[4] = (Sint32) ((sample4 + last_sample4) >> 1); dst[5] = (Sint32) ((sample5 + last_sample5) >> 1); dst[6] = (Sint32) ((sample6 + last_sample6) >> 1); dst[7] = (Sint32) ((sample7 + last_sample7) >> 1); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 1 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1; const float *target = ((const float *) cvt->buf); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); src--; dst[1] = (float) ((sample0 + last_sample0) * 0.5); dst[0] = (float) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); src += 2; dst[0] = (float) ((sample0 + last_sample0) * 0.5); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 1 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1; const float *target = ((const float *) cvt->buf); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); src--; dst[3] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[2] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[0] = (float) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32LSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); src += 4; dst[0] = (float) ((sample0 + last_sample0) * 0.5); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 2 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2; const float *target = ((const float *) cvt->buf); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample0 = (double) SDL_SwapFloatLE(src[0]); src -= 2; dst[3] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); src += 4; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 2 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2; const float *target = ((const float *) cvt->buf); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample0 = (double) SDL_SwapFloatLE(src[0]); src -= 2; dst[7] = (float) ((sample1 + (3.0 * last_sample1)) * 0.25); dst[6] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[5] = (float) ((sample1 + last_sample1) * 0.5); dst[4] = (float) ((sample0 + last_sample0) * 0.5); dst[3] = (float) (((3.0 * sample1) + last_sample1) * 0.25); dst[2] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32LSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); src += 8; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 4 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4; const float *target = ((const float *) cvt->buf); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample0 = (double) SDL_SwapFloatLE(src[0]); src -= 4; dst[7] = (float) ((sample3 + last_sample3) * 0.5); dst[6] = (float) ((sample2 + last_sample2) * 0.5); dst[5] = (float) ((sample1 + last_sample1) * 0.5); dst[4] = (float) ((sample0 + last_sample0) * 0.5); dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); src += 8; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 4 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4; const float *target = ((const float *) cvt->buf); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample0 = (double) SDL_SwapFloatLE(src[0]); src -= 4; dst[15] = (float) ((sample3 + (3.0 * last_sample3)) * 0.25); dst[14] = (float) ((sample2 + (3.0 * last_sample2)) * 0.25); dst[13] = (float) ((sample1 + (3.0 * last_sample1)) * 0.25); dst[12] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[11] = (float) ((sample3 + last_sample3) * 0.5); dst[10] = (float) ((sample2 + last_sample2) * 0.5); dst[9] = (float) ((sample1 + last_sample1) * 0.5); dst[8] = (float) ((sample0 + last_sample0) * 0.5); dst[7] = (float) (((3.0 * sample3) + last_sample3) * 0.25); dst[6] = (float) (((3.0 * sample2) + last_sample2) * 0.25); dst[5] = (float) (((3.0 * sample1) + last_sample1) * 0.25); dst[4] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32LSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); src += 16; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 6 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6; const float *target = ((const float *) cvt->buf); double last_sample5 = (double) SDL_SwapFloatLE(src[5]); double last_sample4 = (double) SDL_SwapFloatLE(src[4]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample5 = (double) SDL_SwapFloatLE(src[5]); const double sample4 = (double) SDL_SwapFloatLE(src[4]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample0 = (double) SDL_SwapFloatLE(src[0]); src -= 6; dst[11] = (float) ((sample5 + last_sample5) * 0.5); dst[10] = (float) ((sample4 + last_sample4) * 0.5); dst[9] = (float) ((sample3 + last_sample3) * 0.5); dst[8] = (float) ((sample2 + last_sample2) * 0.5); dst[7] = (float) ((sample1 + last_sample1) * 0.5); dst[6] = (float) ((sample0 + last_sample0) * 0.5); dst[5] = (float) sample5; dst[4] = (float) sample4; dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample4 = (double) SDL_SwapFloatLE(src[4]); double last_sample5 = (double) SDL_SwapFloatLE(src[5]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample4 = (double) SDL_SwapFloatLE(src[4]); const double sample5 = (double) SDL_SwapFloatLE(src[5]); src += 12; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); dst[4] = (float) ((sample4 + last_sample4) * 0.5); dst[5] = (float) ((sample5 + last_sample5) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 6 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6; const float *target = ((const float *) cvt->buf); double last_sample5 = (double) SDL_SwapFloatLE(src[5]); double last_sample4 = (double) SDL_SwapFloatLE(src[4]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample5 = (double) SDL_SwapFloatLE(src[5]); const double sample4 = (double) SDL_SwapFloatLE(src[4]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample0 = (double) SDL_SwapFloatLE(src[0]); src -= 6; dst[23] = (float) ((sample5 + (3.0 * last_sample5)) * 0.25); dst[22] = (float) ((sample4 + (3.0 * last_sample4)) * 0.25); dst[21] = (float) ((sample3 + (3.0 * last_sample3)) * 0.25); dst[20] = (float) ((sample2 + (3.0 * last_sample2)) * 0.25); dst[19] = (float) ((sample1 + (3.0 * last_sample1)) * 0.25); dst[18] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[17] = (float) ((sample5 + last_sample5) * 0.5); dst[16] = (float) ((sample4 + last_sample4) * 0.5); dst[15] = (float) ((sample3 + last_sample3) * 0.5); dst[14] = (float) ((sample2 + last_sample2) * 0.5); dst[13] = (float) ((sample1 + last_sample1) * 0.5); dst[12] = (float) ((sample0 + last_sample0) * 0.5); dst[11] = (float) (((3.0 * sample5) + last_sample5) * 0.25); dst[10] = (float) (((3.0 * sample4) + last_sample4) * 0.25); dst[9] = (float) (((3.0 * sample3) + last_sample3) * 0.25); dst[8] = (float) (((3.0 * sample2) + last_sample2) * 0.25); dst[7] = (float) (((3.0 * sample1) + last_sample1) * 0.25); dst[6] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[5] = (float) sample5; dst[4] = (float) sample4; dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32LSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample4 = (double) SDL_SwapFloatLE(src[4]); double last_sample5 = (double) SDL_SwapFloatLE(src[5]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample4 = (double) SDL_SwapFloatLE(src[4]); const double sample5 = (double) SDL_SwapFloatLE(src[5]); src += 24; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); dst[4] = (float) ((sample4 + last_sample4) * 0.5); dst[5] = (float) ((sample5 + last_sample5) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 8 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8; const float *target = ((const float *) cvt->buf); double last_sample7 = (double) SDL_SwapFloatLE(src[7]); double last_sample6 = (double) SDL_SwapFloatLE(src[6]); double last_sample5 = (double) SDL_SwapFloatLE(src[5]); double last_sample4 = (double) SDL_SwapFloatLE(src[4]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample7 = (double) SDL_SwapFloatLE(src[7]); const double sample6 = (double) SDL_SwapFloatLE(src[6]); const double sample5 = (double) SDL_SwapFloatLE(src[5]); const double sample4 = (double) SDL_SwapFloatLE(src[4]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample0 = (double) SDL_SwapFloatLE(src[0]); src -= 8; dst[15] = (float) ((sample7 + last_sample7) * 0.5); dst[14] = (float) ((sample6 + last_sample6) * 0.5); dst[13] = (float) ((sample5 + last_sample5) * 0.5); dst[12] = (float) ((sample4 + last_sample4) * 0.5); dst[11] = (float) ((sample3 + last_sample3) * 0.5); dst[10] = (float) ((sample2 + last_sample2) * 0.5); dst[9] = (float) ((sample1 + last_sample1) * 0.5); dst[8] = (float) ((sample0 + last_sample0) * 0.5); dst[7] = (float) sample7; dst[6] = (float) sample6; dst[5] = (float) sample5; dst[4] = (float) sample4; dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample4 = (double) SDL_SwapFloatLE(src[4]); double last_sample5 = (double) SDL_SwapFloatLE(src[5]); double last_sample6 = (double) SDL_SwapFloatLE(src[6]); double last_sample7 = (double) SDL_SwapFloatLE(src[7]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample4 = (double) SDL_SwapFloatLE(src[4]); const double sample5 = (double) SDL_SwapFloatLE(src[5]); const double sample6 = (double) SDL_SwapFloatLE(src[6]); const double sample7 = (double) SDL_SwapFloatLE(src[7]); src += 16; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); dst[4] = (float) ((sample4 + last_sample4) * 0.5); dst[5] = (float) ((sample5 + last_sample5) * 0.5); dst[6] = (float) ((sample6 + last_sample6) * 0.5); dst[7] = (float) ((sample7 + last_sample7) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32LSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 8 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8; const float *target = ((const float *) cvt->buf); double last_sample7 = (double) SDL_SwapFloatLE(src[7]); double last_sample6 = (double) SDL_SwapFloatLE(src[6]); double last_sample5 = (double) SDL_SwapFloatLE(src[5]); double last_sample4 = (double) SDL_SwapFloatLE(src[4]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); while (dst >= target) { const double sample7 = (double) SDL_SwapFloatLE(src[7]); const double sample6 = (double) SDL_SwapFloatLE(src[6]); const double sample5 = (double) SDL_SwapFloatLE(src[5]); const double sample4 = (double) SDL_SwapFloatLE(src[4]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample0 = (double) SDL_SwapFloatLE(src[0]); src -= 8; dst[31] = (float) ((sample7 + (3.0 * last_sample7)) * 0.25); dst[30] = (float) ((sample6 + (3.0 * last_sample6)) * 0.25); dst[29] = (float) ((sample5 + (3.0 * last_sample5)) * 0.25); dst[28] = (float) ((sample4 + (3.0 * last_sample4)) * 0.25); dst[27] = (float) ((sample3 + (3.0 * last_sample3)) * 0.25); dst[26] = (float) ((sample2 + (3.0 * last_sample2)) * 0.25); dst[25] = (float) ((sample1 + (3.0 * last_sample1)) * 0.25); dst[24] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[23] = (float) ((sample7 + last_sample7) * 0.5); dst[22] = (float) ((sample6 + last_sample6) * 0.5); dst[21] = (float) ((sample5 + last_sample5) * 0.5); dst[20] = (float) ((sample4 + last_sample4) * 0.5); dst[19] = (float) ((sample3 + last_sample3) * 0.5); dst[18] = (float) ((sample2 + last_sample2) * 0.5); dst[17] = (float) ((sample1 + last_sample1) * 0.5); dst[16] = (float) ((sample0 + last_sample0) * 0.5); dst[15] = (float) (((3.0 * sample7) + last_sample7) * 0.25); dst[14] = (float) (((3.0 * sample6) + last_sample6) * 0.25); dst[13] = (float) (((3.0 * sample5) + last_sample5) * 0.25); dst[12] = (float) (((3.0 * sample4) + last_sample4) * 0.25); dst[11] = (float) (((3.0 * sample3) + last_sample3) * 0.25); dst[10] = (float) (((3.0 * sample2) + last_sample2) * 0.25); dst[9] = (float) (((3.0 * sample1) + last_sample1) * 0.25); dst[8] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[7] = (float) sample7; dst[6] = (float) sample6; dst[5] = (float) sample5; dst[4] = (float) sample4; dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32LSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32LSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatLE(src[0]); double last_sample1 = (double) SDL_SwapFloatLE(src[1]); double last_sample2 = (double) SDL_SwapFloatLE(src[2]); double last_sample3 = (double) SDL_SwapFloatLE(src[3]); double last_sample4 = (double) SDL_SwapFloatLE(src[4]); double last_sample5 = (double) SDL_SwapFloatLE(src[5]); double last_sample6 = (double) SDL_SwapFloatLE(src[6]); double last_sample7 = (double) SDL_SwapFloatLE(src[7]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatLE(src[0]); const double sample1 = (double) SDL_SwapFloatLE(src[1]); const double sample2 = (double) SDL_SwapFloatLE(src[2]); const double sample3 = (double) SDL_SwapFloatLE(src[3]); const double sample4 = (double) SDL_SwapFloatLE(src[4]); const double sample5 = (double) SDL_SwapFloatLE(src[5]); const double sample6 = (double) SDL_SwapFloatLE(src[6]); const double sample7 = (double) SDL_SwapFloatLE(src[7]); src += 32; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); dst[4] = (float) ((sample4 + last_sample4) * 0.5); dst[5] = (float) ((sample5 + last_sample5) * 0.5); dst[6] = (float) ((sample6 + last_sample6) * 0.5); dst[7] = (float) ((sample7 + last_sample7) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 1 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1; const float *target = ((const float *) cvt->buf); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); src--; dst[1] = (float) ((sample0 + last_sample0) * 0.5); dst[0] = (float) sample0; last_sample0 = sample0; dst -= 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_1c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); src += 2; dst[0] = (float) ((sample0 + last_sample0) * 0.5); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 1 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1; const float *target = ((const float *) cvt->buf); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); src--; dst[3] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[2] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[0] = (float) sample0; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_1c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32MSB, 1 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); src += 4; dst[0] = (float) ((sample0 + last_sample0) * 0.5); last_sample0 = sample0; dst++; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 2 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2; const float *target = ((const float *) cvt->buf); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample0 = (double) SDL_SwapFloatBE(src[0]); src -= 2; dst[3] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_2c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); src += 4; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 2 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2; const float *target = ((const float *) cvt->buf); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample0 = (double) SDL_SwapFloatBE(src[0]); src -= 2; dst[7] = (float) ((sample1 + (3.0 * last_sample1)) * 0.25); dst[6] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[5] = (float) ((sample1 + last_sample1) * 0.5); dst[4] = (float) ((sample0 + last_sample0) * 0.5); dst[3] = (float) (((3.0 * sample1) + last_sample1) * 0.25); dst[2] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_2c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32MSB, 2 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); src += 8; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); last_sample0 = sample0; last_sample1 = sample1; dst += 2; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 4 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4; const float *target = ((const float *) cvt->buf); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample0 = (double) SDL_SwapFloatBE(src[0]); src -= 4; dst[7] = (float) ((sample3 + last_sample3) * 0.5); dst[6] = (float) ((sample2 + last_sample2) * 0.5); dst[5] = (float) ((sample1 + last_sample1) * 0.5); dst[4] = (float) ((sample0 + last_sample0) * 0.5); dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_4c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); src += 8; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 4 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4; const float *target = ((const float *) cvt->buf); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample0 = (double) SDL_SwapFloatBE(src[0]); src -= 4; dst[15] = (float) ((sample3 + (3.0 * last_sample3)) * 0.25); dst[14] = (float) ((sample2 + (3.0 * last_sample2)) * 0.25); dst[13] = (float) ((sample1 + (3.0 * last_sample1)) * 0.25); dst[12] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[11] = (float) ((sample3 + last_sample3) * 0.5); dst[10] = (float) ((sample2 + last_sample2) * 0.5); dst[9] = (float) ((sample1 + last_sample1) * 0.5); dst[8] = (float) ((sample0 + last_sample0) * 0.5); dst[7] = (float) (((3.0 * sample3) + last_sample3) * 0.25); dst[6] = (float) (((3.0 * sample2) + last_sample2) * 0.25); dst[5] = (float) (((3.0 * sample1) + last_sample1) * 0.25); dst[4] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_4c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32MSB, 4 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); src += 16; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; dst += 4; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 6 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6; const float *target = ((const float *) cvt->buf); double last_sample5 = (double) SDL_SwapFloatBE(src[5]); double last_sample4 = (double) SDL_SwapFloatBE(src[4]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample5 = (double) SDL_SwapFloatBE(src[5]); const double sample4 = (double) SDL_SwapFloatBE(src[4]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample0 = (double) SDL_SwapFloatBE(src[0]); src -= 6; dst[11] = (float) ((sample5 + last_sample5) * 0.5); dst[10] = (float) ((sample4 + last_sample4) * 0.5); dst[9] = (float) ((sample3 + last_sample3) * 0.5); dst[8] = (float) ((sample2 + last_sample2) * 0.5); dst[7] = (float) ((sample1 + last_sample1) * 0.5); dst[6] = (float) ((sample0 + last_sample0) * 0.5); dst[5] = (float) sample5; dst[4] = (float) sample4; dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 12; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_6c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample4 = (double) SDL_SwapFloatBE(src[4]); double last_sample5 = (double) SDL_SwapFloatBE(src[5]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample4 = (double) SDL_SwapFloatBE(src[4]); const double sample5 = (double) SDL_SwapFloatBE(src[5]); src += 12; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); dst[4] = (float) ((sample4 + last_sample4) * 0.5); dst[5] = (float) ((sample5 + last_sample5) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 6 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6; const float *target = ((const float *) cvt->buf); double last_sample5 = (double) SDL_SwapFloatBE(src[5]); double last_sample4 = (double) SDL_SwapFloatBE(src[4]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample5 = (double) SDL_SwapFloatBE(src[5]); const double sample4 = (double) SDL_SwapFloatBE(src[4]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample0 = (double) SDL_SwapFloatBE(src[0]); src -= 6; dst[23] = (float) ((sample5 + (3.0 * last_sample5)) * 0.25); dst[22] = (float) ((sample4 + (3.0 * last_sample4)) * 0.25); dst[21] = (float) ((sample3 + (3.0 * last_sample3)) * 0.25); dst[20] = (float) ((sample2 + (3.0 * last_sample2)) * 0.25); dst[19] = (float) ((sample1 + (3.0 * last_sample1)) * 0.25); dst[18] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[17] = (float) ((sample5 + last_sample5) * 0.5); dst[16] = (float) ((sample4 + last_sample4) * 0.5); dst[15] = (float) ((sample3 + last_sample3) * 0.5); dst[14] = (float) ((sample2 + last_sample2) * 0.5); dst[13] = (float) ((sample1 + last_sample1) * 0.5); dst[12] = (float) ((sample0 + last_sample0) * 0.5); dst[11] = (float) (((3.0 * sample5) + last_sample5) * 0.25); dst[10] = (float) (((3.0 * sample4) + last_sample4) * 0.25); dst[9] = (float) (((3.0 * sample3) + last_sample3) * 0.25); dst[8] = (float) (((3.0 * sample2) + last_sample2) * 0.25); dst[7] = (float) (((3.0 * sample1) + last_sample1) * 0.25); dst[6] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[5] = (float) sample5; dst[4] = (float) sample4; dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 24; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_6c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32MSB, 6 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample4 = (double) SDL_SwapFloatBE(src[4]); double last_sample5 = (double) SDL_SwapFloatBE(src[5]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample4 = (double) SDL_SwapFloatBE(src[4]); const double sample5 = (double) SDL_SwapFloatBE(src[5]); src += 24; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); dst[4] = (float) ((sample4 + last_sample4) * 0.5); dst[5] = (float) ((sample5 + last_sample5) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; dst += 6; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x2) AUDIO_F32MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 2; float *dst = ((float *) (cvt->buf + dstsize)) - 8 * 2; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8; const float *target = ((const float *) cvt->buf); double last_sample7 = (double) SDL_SwapFloatBE(src[7]); double last_sample6 = (double) SDL_SwapFloatBE(src[6]); double last_sample5 = (double) SDL_SwapFloatBE(src[5]); double last_sample4 = (double) SDL_SwapFloatBE(src[4]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample7 = (double) SDL_SwapFloatBE(src[7]); const double sample6 = (double) SDL_SwapFloatBE(src[6]); const double sample5 = (double) SDL_SwapFloatBE(src[5]); const double sample4 = (double) SDL_SwapFloatBE(src[4]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample0 = (double) SDL_SwapFloatBE(src[0]); src -= 8; dst[15] = (float) ((sample7 + last_sample7) * 0.5); dst[14] = (float) ((sample6 + last_sample6) * 0.5); dst[13] = (float) ((sample5 + last_sample5) * 0.5); dst[12] = (float) ((sample4 + last_sample4) * 0.5); dst[11] = (float) ((sample3 + last_sample3) * 0.5); dst[10] = (float) ((sample2 + last_sample2) * 0.5); dst[9] = (float) ((sample1 + last_sample1) * 0.5); dst[8] = (float) ((sample0 + last_sample0) * 0.5); dst[7] = (float) sample7; dst[6] = (float) sample6; dst[5] = (float) sample5; dst[4] = (float) sample4; dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 16; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_8c_x2(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x2) AUDIO_F32MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 2; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample4 = (double) SDL_SwapFloatBE(src[4]); double last_sample5 = (double) SDL_SwapFloatBE(src[5]); double last_sample6 = (double) SDL_SwapFloatBE(src[6]); double last_sample7 = (double) SDL_SwapFloatBE(src[7]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample4 = (double) SDL_SwapFloatBE(src[4]); const double sample5 = (double) SDL_SwapFloatBE(src[5]); const double sample6 = (double) SDL_SwapFloatBE(src[6]); const double sample7 = (double) SDL_SwapFloatBE(src[7]); src += 16; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); dst[4] = (float) ((sample4 + last_sample4) * 0.5); dst[5] = (float) ((sample5 + last_sample5) * 0.5); dst[6] = (float) ((sample6 + last_sample6) * 0.5); dst[7] = (float) ((sample7 + last_sample7) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Upsample_F32MSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Upsample (x4) AUDIO_F32MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt * 4; float *dst = ((float *) (cvt->buf + dstsize)) - 8 * 4; const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8; const float *target = ((const float *) cvt->buf); double last_sample7 = (double) SDL_SwapFloatBE(src[7]); double last_sample6 = (double) SDL_SwapFloatBE(src[6]); double last_sample5 = (double) SDL_SwapFloatBE(src[5]); double last_sample4 = (double) SDL_SwapFloatBE(src[4]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); while (dst >= target) { const double sample7 = (double) SDL_SwapFloatBE(src[7]); const double sample6 = (double) SDL_SwapFloatBE(src[6]); const double sample5 = (double) SDL_SwapFloatBE(src[5]); const double sample4 = (double) SDL_SwapFloatBE(src[4]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample0 = (double) SDL_SwapFloatBE(src[0]); src -= 8; dst[31] = (float) ((sample7 + (3.0 * last_sample7)) * 0.25); dst[30] = (float) ((sample6 + (3.0 * last_sample6)) * 0.25); dst[29] = (float) ((sample5 + (3.0 * last_sample5)) * 0.25); dst[28] = (float) ((sample4 + (3.0 * last_sample4)) * 0.25); dst[27] = (float) ((sample3 + (3.0 * last_sample3)) * 0.25); dst[26] = (float) ((sample2 + (3.0 * last_sample2)) * 0.25); dst[25] = (float) ((sample1 + (3.0 * last_sample1)) * 0.25); dst[24] = (float) ((sample0 + (3.0 * last_sample0)) * 0.25); dst[23] = (float) ((sample7 + last_sample7) * 0.5); dst[22] = (float) ((sample6 + last_sample6) * 0.5); dst[21] = (float) ((sample5 + last_sample5) * 0.5); dst[20] = (float) ((sample4 + last_sample4) * 0.5); dst[19] = (float) ((sample3 + last_sample3) * 0.5); dst[18] = (float) ((sample2 + last_sample2) * 0.5); dst[17] = (float) ((sample1 + last_sample1) * 0.5); dst[16] = (float) ((sample0 + last_sample0) * 0.5); dst[15] = (float) (((3.0 * sample7) + last_sample7) * 0.25); dst[14] = (float) (((3.0 * sample6) + last_sample6) * 0.25); dst[13] = (float) (((3.0 * sample5) + last_sample5) * 0.25); dst[12] = (float) (((3.0 * sample4) + last_sample4) * 0.25); dst[11] = (float) (((3.0 * sample3) + last_sample3) * 0.25); dst[10] = (float) (((3.0 * sample2) + last_sample2) * 0.25); dst[9] = (float) (((3.0 * sample1) + last_sample1) * 0.25); dst[8] = (float) (((3.0 * sample0) + last_sample0) * 0.25); dst[7] = (float) sample7; dst[6] = (float) sample6; dst[5] = (float) sample5; dst[4] = (float) sample4; dst[3] = (float) sample3; dst[2] = (float) sample2; dst[1] = (float) sample1; dst[0] = (float) sample0; last_sample7 = sample7; last_sample6 = sample6; last_sample5 = sample5; last_sample4 = sample4; last_sample3 = sample3; last_sample2 = sample2; last_sample1 = sample1; last_sample0 = sample0; dst -= 32; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } static void SDLCALL SDL_Downsample_F32MSB_8c_x4(SDL_AudioCVT * cvt, SDL_AudioFormat format) { #if DEBUG_CONVERT fprintf(stderr, "Downsample (x4) AUDIO_F32MSB, 8 channels.\n"); #endif const int dstsize = cvt->len_cvt / 4; float *dst = (float *) cvt->buf; const float *src = (float *) cvt->buf; const float *target = (const float *) (cvt->buf + dstsize); double last_sample0 = (double) SDL_SwapFloatBE(src[0]); double last_sample1 = (double) SDL_SwapFloatBE(src[1]); double last_sample2 = (double) SDL_SwapFloatBE(src[2]); double last_sample3 = (double) SDL_SwapFloatBE(src[3]); double last_sample4 = (double) SDL_SwapFloatBE(src[4]); double last_sample5 = (double) SDL_SwapFloatBE(src[5]); double last_sample6 = (double) SDL_SwapFloatBE(src[6]); double last_sample7 = (double) SDL_SwapFloatBE(src[7]); while (dst < target) { const double sample0 = (double) SDL_SwapFloatBE(src[0]); const double sample1 = (double) SDL_SwapFloatBE(src[1]); const double sample2 = (double) SDL_SwapFloatBE(src[2]); const double sample3 = (double) SDL_SwapFloatBE(src[3]); const double sample4 = (double) SDL_SwapFloatBE(src[4]); const double sample5 = (double) SDL_SwapFloatBE(src[5]); const double sample6 = (double) SDL_SwapFloatBE(src[6]); const double sample7 = (double) SDL_SwapFloatBE(src[7]); src += 32; dst[0] = (float) ((sample0 + last_sample0) * 0.5); dst[1] = (float) ((sample1 + last_sample1) * 0.5); dst[2] = (float) ((sample2 + last_sample2) * 0.5); dst[3] = (float) ((sample3 + last_sample3) * 0.5); dst[4] = (float) ((sample4 + last_sample4) * 0.5); dst[5] = (float) ((sample5 + last_sample5) * 0.5); dst[6] = (float) ((sample6 + last_sample6) * 0.5); dst[7] = (float) ((sample7 + last_sample7) * 0.5); last_sample0 = sample0; last_sample1 = sample1; last_sample2 = sample2; last_sample3 = sample3; last_sample4 = sample4; last_sample5 = sample5; last_sample6 = sample6; last_sample7 = sample7; dst += 8; } cvt->len_cvt = dstsize; if (cvt->filters[++cvt->filter_index]) { cvt->filters[cvt->filter_index] (cvt, format); } } #endif /* !LESS_RESAMPLERS */ #endif /* !NO_RESAMPLERS */ const SDL_AudioRateFilters sdl_audio_rate_filters[] = { #if !NO_RESAMPLERS { AUDIO_U8, 1, 0, 0, SDL_Downsample_U8_1c }, { AUDIO_U8, 1, 1, 0, SDL_Upsample_U8_1c }, { AUDIO_U8, 2, 0, 0, SDL_Downsample_U8_2c }, { AUDIO_U8, 2, 1, 0, SDL_Upsample_U8_2c }, { AUDIO_U8, 4, 0, 0, SDL_Downsample_U8_4c }, { AUDIO_U8, 4, 1, 0, SDL_Upsample_U8_4c }, { AUDIO_U8, 6, 0, 0, SDL_Downsample_U8_6c }, { AUDIO_U8, 6, 1, 0, SDL_Upsample_U8_6c }, { AUDIO_U8, 8, 0, 0, SDL_Downsample_U8_8c }, { AUDIO_U8, 8, 1, 0, SDL_Upsample_U8_8c }, { AUDIO_S8, 1, 0, 0, SDL_Downsample_S8_1c }, { AUDIO_S8, 1, 1, 0, SDL_Upsample_S8_1c }, { AUDIO_S8, 2, 0, 0, SDL_Downsample_S8_2c }, { AUDIO_S8, 2, 1, 0, SDL_Upsample_S8_2c }, { AUDIO_S8, 4, 0, 0, SDL_Downsample_S8_4c }, { AUDIO_S8, 4, 1, 0, SDL_Upsample_S8_4c }, { AUDIO_S8, 6, 0, 0, SDL_Downsample_S8_6c }, { AUDIO_S8, 6, 1, 0, SDL_Upsample_S8_6c }, { AUDIO_S8, 8, 0, 0, SDL_Downsample_S8_8c }, { AUDIO_S8, 8, 1, 0, SDL_Upsample_S8_8c }, { AUDIO_U16LSB, 1, 0, 0, SDL_Downsample_U16LSB_1c }, { AUDIO_U16LSB, 1, 1, 0, SDL_Upsample_U16LSB_1c }, { AUDIO_U16LSB, 2, 0, 0, SDL_Downsample_U16LSB_2c }, { AUDIO_U16LSB, 2, 1, 0, SDL_Upsample_U16LSB_2c }, { AUDIO_U16LSB, 4, 0, 0, SDL_Downsample_U16LSB_4c }, { AUDIO_U16LSB, 4, 1, 0, SDL_Upsample_U16LSB_4c }, { AUDIO_U16LSB, 6, 0, 0, SDL_Downsample_U16LSB_6c }, { AUDIO_U16LSB, 6, 1, 0, SDL_Upsample_U16LSB_6c }, { AUDIO_U16LSB, 8, 0, 0, SDL_Downsample_U16LSB_8c }, { AUDIO_U16LSB, 8, 1, 0, SDL_Upsample_U16LSB_8c }, { AUDIO_S16LSB, 1, 0, 0, SDL_Downsample_S16LSB_1c }, { AUDIO_S16LSB, 1, 1, 0, SDL_Upsample_S16LSB_1c }, { AUDIO_S16LSB, 2, 0, 0, SDL_Downsample_S16LSB_2c }, { AUDIO_S16LSB, 2, 1, 0, SDL_Upsample_S16LSB_2c }, { AUDIO_S16LSB, 4, 0, 0, SDL_Downsample_S16LSB_4c }, { AUDIO_S16LSB, 4, 1, 0, SDL_Upsample_S16LSB_4c }, { AUDIO_S16LSB, 6, 0, 0, SDL_Downsample_S16LSB_6c }, { AUDIO_S16LSB, 6, 1, 0, SDL_Upsample_S16LSB_6c }, { AUDIO_S16LSB, 8, 0, 0, SDL_Downsample_S16LSB_8c }, { AUDIO_S16LSB, 8, 1, 0, SDL_Upsample_S16LSB_8c }, { AUDIO_U16MSB, 1, 0, 0, SDL_Downsample_U16MSB_1c }, { AUDIO_U16MSB, 1, 1, 0, SDL_Upsample_U16MSB_1c }, { AUDIO_U16MSB, 2, 0, 0, SDL_Downsample_U16MSB_2c }, { AUDIO_U16MSB, 2, 1, 0, SDL_Upsample_U16MSB_2c }, { AUDIO_U16MSB, 4, 0, 0, SDL_Downsample_U16MSB_4c }, { AUDIO_U16MSB, 4, 1, 0, SDL_Upsample_U16MSB_4c }, { AUDIO_U16MSB, 6, 0, 0, SDL_Downsample_U16MSB_6c }, { AUDIO_U16MSB, 6, 1, 0, SDL_Upsample_U16MSB_6c }, { AUDIO_U16MSB, 8, 0, 0, SDL_Downsample_U16MSB_8c }, { AUDIO_U16MSB, 8, 1, 0, SDL_Upsample_U16MSB_8c }, { AUDIO_S16MSB, 1, 0, 0, SDL_Downsample_S16MSB_1c }, { AUDIO_S16MSB, 1, 1, 0, SDL_Upsample_S16MSB_1c }, { AUDIO_S16MSB, 2, 0, 0, SDL_Downsample_S16MSB_2c }, { AUDIO_S16MSB, 2, 1, 0, SDL_Upsample_S16MSB_2c }, { AUDIO_S16MSB, 4, 0, 0, SDL_Downsample_S16MSB_4c }, { AUDIO_S16MSB, 4, 1, 0, SDL_Upsample_S16MSB_4c }, { AUDIO_S16MSB, 6, 0, 0, SDL_Downsample_S16MSB_6c }, { AUDIO_S16MSB, 6, 1, 0, SDL_Upsample_S16MSB_6c }, { AUDIO_S16MSB, 8, 0, 0, SDL_Downsample_S16MSB_8c }, { AUDIO_S16MSB, 8, 1, 0, SDL_Upsample_S16MSB_8c }, { AUDIO_S32LSB, 1, 0, 0, SDL_Downsample_S32LSB_1c }, { AUDIO_S32LSB, 1, 1, 0, SDL_Upsample_S32LSB_1c }, { AUDIO_S32LSB, 2, 0, 0, SDL_Downsample_S32LSB_2c }, { AUDIO_S32LSB, 2, 1, 0, SDL_Upsample_S32LSB_2c }, { AUDIO_S32LSB, 4, 0, 0, SDL_Downsample_S32LSB_4c }, { AUDIO_S32LSB, 4, 1, 0, SDL_Upsample_S32LSB_4c }, { AUDIO_S32LSB, 6, 0, 0, SDL_Downsample_S32LSB_6c }, { AUDIO_S32LSB, 6, 1, 0, SDL_Upsample_S32LSB_6c }, { AUDIO_S32LSB, 8, 0, 0, SDL_Downsample_S32LSB_8c }, { AUDIO_S32LSB, 8, 1, 0, SDL_Upsample_S32LSB_8c }, { AUDIO_S32MSB, 1, 0, 0, SDL_Downsample_S32MSB_1c }, { AUDIO_S32MSB, 1, 1, 0, SDL_Upsample_S32MSB_1c }, { AUDIO_S32MSB, 2, 0, 0, SDL_Downsample_S32MSB_2c }, { AUDIO_S32MSB, 2, 1, 0, SDL_Upsample_S32MSB_2c }, { AUDIO_S32MSB, 4, 0, 0, SDL_Downsample_S32MSB_4c }, { AUDIO_S32MSB, 4, 1, 0, SDL_Upsample_S32MSB_4c }, { AUDIO_S32MSB, 6, 0, 0, SDL_Downsample_S32MSB_6c }, { AUDIO_S32MSB, 6, 1, 0, SDL_Upsample_S32MSB_6c }, { AUDIO_S32MSB, 8, 0, 0, SDL_Downsample_S32MSB_8c }, { AUDIO_S32MSB, 8, 1, 0, SDL_Upsample_S32MSB_8c }, { AUDIO_F32LSB, 1, 0, 0, SDL_Downsample_F32LSB_1c }, { AUDIO_F32LSB, 1, 1, 0, SDL_Upsample_F32LSB_1c }, { AUDIO_F32LSB, 2, 0, 0, SDL_Downsample_F32LSB_2c }, { AUDIO_F32LSB, 2, 1, 0, SDL_Upsample_F32LSB_2c }, { AUDIO_F32LSB, 4, 0, 0, SDL_Downsample_F32LSB_4c }, { AUDIO_F32LSB, 4, 1, 0, SDL_Upsample_F32LSB_4c }, { AUDIO_F32LSB, 6, 0, 0, SDL_Downsample_F32LSB_6c }, { AUDIO_F32LSB, 6, 1, 0, SDL_Upsample_F32LSB_6c }, { AUDIO_F32LSB, 8, 0, 0, SDL_Downsample_F32LSB_8c }, { AUDIO_F32LSB, 8, 1, 0, SDL_Upsample_F32LSB_8c }, { AUDIO_F32MSB, 1, 0, 0, SDL_Downsample_F32MSB_1c }, { AUDIO_F32MSB, 1, 1, 0, SDL_Upsample_F32MSB_1c }, { AUDIO_F32MSB, 2, 0, 0, SDL_Downsample_F32MSB_2c }, { AUDIO_F32MSB, 2, 1, 0, SDL_Upsample_F32MSB_2c }, { AUDIO_F32MSB, 4, 0, 0, SDL_Downsample_F32MSB_4c }, { AUDIO_F32MSB, 4, 1, 0, SDL_Upsample_F32MSB_4c }, { AUDIO_F32MSB, 6, 0, 0, SDL_Downsample_F32MSB_6c }, { AUDIO_F32MSB, 6, 1, 0, SDL_Upsample_F32MSB_6c }, { AUDIO_F32MSB, 8, 0, 0, SDL_Downsample_F32MSB_8c }, { AUDIO_F32MSB, 8, 1, 0, SDL_Upsample_F32MSB_8c }, #if !LESS_RESAMPLERS { AUDIO_U8, 1, 0, 2, SDL_Downsample_U8_1c_x2 }, { AUDIO_U8, 1, 1, 2, SDL_Upsample_U8_1c_x2 }, { AUDIO_U8, 1, 0, 4, SDL_Downsample_U8_1c_x4 }, { AUDIO_U8, 1, 1, 4, SDL_Upsample_U8_1c_x4 }, { AUDIO_U8, 2, 0, 2, SDL_Downsample_U8_2c_x2 }, { AUDIO_U8, 2, 1, 2, SDL_Upsample_U8_2c_x2 }, { AUDIO_U8, 2, 0, 4, SDL_Downsample_U8_2c_x4 }, { AUDIO_U8, 2, 1, 4, SDL_Upsample_U8_2c_x4 }, { AUDIO_U8, 4, 0, 2, SDL_Downsample_U8_4c_x2 }, { AUDIO_U8, 4, 1, 2, SDL_Upsample_U8_4c_x2 }, { AUDIO_U8, 4, 0, 4, SDL_Downsample_U8_4c_x4 }, { AUDIO_U8, 4, 1, 4, SDL_Upsample_U8_4c_x4 }, { AUDIO_U8, 6, 0, 2, SDL_Downsample_U8_6c_x2 }, { AUDIO_U8, 6, 1, 2, SDL_Upsample_U8_6c_x2 }, { AUDIO_U8, 6, 0, 4, SDL_Downsample_U8_6c_x4 }, { AUDIO_U8, 6, 1, 4, SDL_Upsample_U8_6c_x4 }, { AUDIO_U8, 8, 0, 2, SDL_Downsample_U8_8c_x2 }, { AUDIO_U8, 8, 1, 2, SDL_Upsample_U8_8c_x2 }, { AUDIO_U8, 8, 0, 4, SDL_Downsample_U8_8c_x4 }, { AUDIO_U8, 8, 1, 4, SDL_Upsample_U8_8c_x4 }, { AUDIO_S8, 1, 0, 2, SDL_Downsample_S8_1c_x2 }, { AUDIO_S8, 1, 1, 2, SDL_Upsample_S8_1c_x2 }, { AUDIO_S8, 1, 0, 4, SDL_Downsample_S8_1c_x4 }, { AUDIO_S8, 1, 1, 4, SDL_Upsample_S8_1c_x4 }, { AUDIO_S8, 2, 0, 2, SDL_Downsample_S8_2c_x2 }, { AUDIO_S8, 2, 1, 2, SDL_Upsample_S8_2c_x2 }, { AUDIO_S8, 2, 0, 4, SDL_Downsample_S8_2c_x4 }, { AUDIO_S8, 2, 1, 4, SDL_Upsample_S8_2c_x4 }, { AUDIO_S8, 4, 0, 2, SDL_Downsample_S8_4c_x2 }, { AUDIO_S8, 4, 1, 2, SDL_Upsample_S8_4c_x2 }, { AUDIO_S8, 4, 0, 4, SDL_Downsample_S8_4c_x4 }, { AUDIO_S8, 4, 1, 4, SDL_Upsample_S8_4c_x4 }, { AUDIO_S8, 6, 0, 2, SDL_Downsample_S8_6c_x2 }, { AUDIO_S8, 6, 1, 2, SDL_Upsample_S8_6c_x2 }, { AUDIO_S8, 6, 0, 4, SDL_Downsample_S8_6c_x4 }, { AUDIO_S8, 6, 1, 4, SDL_Upsample_S8_6c_x4 }, { AUDIO_S8, 8, 0, 2, SDL_Downsample_S8_8c_x2 }, { AUDIO_S8, 8, 1, 2, SDL_Upsample_S8_8c_x2 }, { AUDIO_S8, 8, 0, 4, SDL_Downsample_S8_8c_x4 }, { AUDIO_S8, 8, 1, 4, SDL_Upsample_S8_8c_x4 }, { AUDIO_U16LSB, 1, 0, 2, SDL_Downsample_U16LSB_1c_x2 }, { AUDIO_U16LSB, 1, 1, 2, SDL_Upsample_U16LSB_1c_x2 }, { AUDIO_U16LSB, 1, 0, 4, SDL_Downsample_U16LSB_1c_x4 }, { AUDIO_U16LSB, 1, 1, 4, SDL_Upsample_U16LSB_1c_x4 }, { AUDIO_U16LSB, 2, 0, 2, SDL_Downsample_U16LSB_2c_x2 }, { AUDIO_U16LSB, 2, 1, 2, SDL_Upsample_U16LSB_2c_x2 }, { AUDIO_U16LSB, 2, 0, 4, SDL_Downsample_U16LSB_2c_x4 }, { AUDIO_U16LSB, 2, 1, 4, SDL_Upsample_U16LSB_2c_x4 }, { AUDIO_U16LSB, 4, 0, 2, SDL_Downsample_U16LSB_4c_x2 }, { AUDIO_U16LSB, 4, 1, 2, SDL_Upsample_U16LSB_4c_x2 }, { AUDIO_U16LSB, 4, 0, 4, SDL_Downsample_U16LSB_4c_x4 }, { AUDIO_U16LSB, 4, 1, 4, SDL_Upsample_U16LSB_4c_x4 }, { AUDIO_U16LSB, 6, 0, 2, SDL_Downsample_U16LSB_6c_x2 }, { AUDIO_U16LSB, 6, 1, 2, SDL_Upsample_U16LSB_6c_x2 }, { AUDIO_U16LSB, 6, 0, 4, SDL_Downsample_U16LSB_6c_x4 }, { AUDIO_U16LSB, 6, 1, 4, SDL_Upsample_U16LSB_6c_x4 }, { AUDIO_U16LSB, 8, 0, 2, SDL_Downsample_U16LSB_8c_x2 }, { AUDIO_U16LSB, 8, 1, 2, SDL_Upsample_U16LSB_8c_x2 }, { AUDIO_U16LSB, 8, 0, 4, SDL_Downsample_U16LSB_8c_x4 }, { AUDIO_U16LSB, 8, 1, 4, SDL_Upsample_U16LSB_8c_x4 }, { AUDIO_S16LSB, 1, 0, 2, SDL_Downsample_S16LSB_1c_x2 }, { AUDIO_S16LSB, 1, 1, 2, SDL_Upsample_S16LSB_1c_x2 }, { AUDIO_S16LSB, 1, 0, 4, SDL_Downsample_S16LSB_1c_x4 }, { AUDIO_S16LSB, 1, 1, 4, SDL_Upsample_S16LSB_1c_x4 }, { AUDIO_S16LSB, 2, 0, 2, SDL_Downsample_S16LSB_2c_x2 }, { AUDIO_S16LSB, 2, 1, 2, SDL_Upsample_S16LSB_2c_x2 }, { AUDIO_S16LSB, 2, 0, 4, SDL_Downsample_S16LSB_2c_x4 }, { AUDIO_S16LSB, 2, 1, 4, SDL_Upsample_S16LSB_2c_x4 }, { AUDIO_S16LSB, 4, 0, 2, SDL_Downsample_S16LSB_4c_x2 }, { AUDIO_S16LSB, 4, 1, 2, SDL_Upsample_S16LSB_4c_x2 }, { AUDIO_S16LSB, 4, 0, 4, SDL_Downsample_S16LSB_4c_x4 }, { AUDIO_S16LSB, 4, 1, 4, SDL_Upsample_S16LSB_4c_x4 }, { AUDIO_S16LSB, 6, 0, 2, SDL_Downsample_S16LSB_6c_x2 }, { AUDIO_S16LSB, 6, 1, 2, SDL_Upsample_S16LSB_6c_x2 }, { AUDIO_S16LSB, 6, 0, 4, SDL_Downsample_S16LSB_6c_x4 }, { AUDIO_S16LSB, 6, 1, 4, SDL_Upsample_S16LSB_6c_x4 }, { AUDIO_S16LSB, 8, 0, 2, SDL_Downsample_S16LSB_8c_x2 }, { AUDIO_S16LSB, 8, 1, 2, SDL_Upsample_S16LSB_8c_x2 }, { AUDIO_S16LSB, 8, 0, 4, SDL_Downsample_S16LSB_8c_x4 }, { AUDIO_S16LSB, 8, 1, 4, SDL_Upsample_S16LSB_8c_x4 }, { AUDIO_U16MSB, 1, 0, 2, SDL_Downsample_U16MSB_1c_x2 }, { AUDIO_U16MSB, 1, 1, 2, SDL_Upsample_U16MSB_1c_x2 }, { AUDIO_U16MSB, 1, 0, 4, SDL_Downsample_U16MSB_1c_x4 }, { AUDIO_U16MSB, 1, 1, 4, SDL_Upsample_U16MSB_1c_x4 }, { AUDIO_U16MSB, 2, 0, 2, SDL_Downsample_U16MSB_2c_x2 }, { AUDIO_U16MSB, 2, 1, 2, SDL_Upsample_U16MSB_2c_x2 }, { AUDIO_U16MSB, 2, 0, 4, SDL_Downsample_U16MSB_2c_x4 }, { AUDIO_U16MSB, 2, 1, 4, SDL_Upsample_U16MSB_2c_x4 }, { AUDIO_U16MSB, 4, 0, 2, SDL_Downsample_U16MSB_4c_x2 }, { AUDIO_U16MSB, 4, 1, 2, SDL_Upsample_U16MSB_4c_x2 }, { AUDIO_U16MSB, 4, 0, 4, SDL_Downsample_U16MSB_4c_x4 }, { AUDIO_U16MSB, 4, 1, 4, SDL_Upsample_U16MSB_4c_x4 }, { AUDIO_U16MSB, 6, 0, 2, SDL_Downsample_U16MSB_6c_x2 }, { AUDIO_U16MSB, 6, 1, 2, SDL_Upsample_U16MSB_6c_x2 }, { AUDIO_U16MSB, 6, 0, 4, SDL_Downsample_U16MSB_6c_x4 }, { AUDIO_U16MSB, 6, 1, 4, SDL_Upsample_U16MSB_6c_x4 }, { AUDIO_U16MSB, 8, 0, 2, SDL_Downsample_U16MSB_8c_x2 }, { AUDIO_U16MSB, 8, 1, 2, SDL_Upsample_U16MSB_8c_x2 }, { AUDIO_U16MSB, 8, 0, 4, SDL_Downsample_U16MSB_8c_x4 }, { AUDIO_U16MSB, 8, 1, 4, SDL_Upsample_U16MSB_8c_x4 }, { AUDIO_S16MSB, 1, 0, 2, SDL_Downsample_S16MSB_1c_x2 }, { AUDIO_S16MSB, 1, 1, 2, SDL_Upsample_S16MSB_1c_x2 }, { AUDIO_S16MSB, 1, 0, 4, SDL_Downsample_S16MSB_1c_x4 }, { AUDIO_S16MSB, 1, 1, 4, SDL_Upsample_S16MSB_1c_x4 }, { AUDIO_S16MSB, 2, 0, 2, SDL_Downsample_S16MSB_2c_x2 }, { AUDIO_S16MSB, 2, 1, 2, SDL_Upsample_S16MSB_2c_x2 }, { AUDIO_S16MSB, 2, 0, 4, SDL_Downsample_S16MSB_2c_x4 }, { AUDIO_S16MSB, 2, 1, 4, SDL_Upsample_S16MSB_2c_x4 }, { AUDIO_S16MSB, 4, 0, 2, SDL_Downsample_S16MSB_4c_x2 }, { AUDIO_S16MSB, 4, 1, 2, SDL_Upsample_S16MSB_4c_x2 }, { AUDIO_S16MSB, 4, 0, 4, SDL_Downsample_S16MSB_4c_x4 }, { AUDIO_S16MSB, 4, 1, 4, SDL_Upsample_S16MSB_4c_x4 }, { AUDIO_S16MSB, 6, 0, 2, SDL_Downsample_S16MSB_6c_x2 }, { AUDIO_S16MSB, 6, 1, 2, SDL_Upsample_S16MSB_6c_x2 }, { AUDIO_S16MSB, 6, 0, 4, SDL_Downsample_S16MSB_6c_x4 }, { AUDIO_S16MSB, 6, 1, 4, SDL_Upsample_S16MSB_6c_x4 }, { AUDIO_S16MSB, 8, 0, 2, SDL_Downsample_S16MSB_8c_x2 }, { AUDIO_S16MSB, 8, 1, 2, SDL_Upsample_S16MSB_8c_x2 }, { AUDIO_S16MSB, 8, 0, 4, SDL_Downsample_S16MSB_8c_x4 }, { AUDIO_S16MSB, 8, 1, 4, SDL_Upsample_S16MSB_8c_x4 }, { AUDIO_S32LSB, 1, 0, 2, SDL_Downsample_S32LSB_1c_x2 }, { AUDIO_S32LSB, 1, 1, 2, SDL_Upsample_S32LSB_1c_x2 }, { AUDIO_S32LSB, 1, 0, 4, SDL_Downsample_S32LSB_1c_x4 }, { AUDIO_S32LSB, 1, 1, 4, SDL_Upsample_S32LSB_1c_x4 }, { AUDIO_S32LSB, 2, 0, 2, SDL_Downsample_S32LSB_2c_x2 }, { AUDIO_S32LSB, 2, 1, 2, SDL_Upsample_S32LSB_2c_x2 }, { AUDIO_S32LSB, 2, 0, 4, SDL_Downsample_S32LSB_2c_x4 }, { AUDIO_S32LSB, 2, 1, 4, SDL_Upsample_S32LSB_2c_x4 }, { AUDIO_S32LSB, 4, 0, 2, SDL_Downsample_S32LSB_4c_x2 }, { AUDIO_S32LSB, 4, 1, 2, SDL_Upsample_S32LSB_4c_x2 }, { AUDIO_S32LSB, 4, 0, 4, SDL_Downsample_S32LSB_4c_x4 }, { AUDIO_S32LSB, 4, 1, 4, SDL_Upsample_S32LSB_4c_x4 }, { AUDIO_S32LSB, 6, 0, 2, SDL_Downsample_S32LSB_6c_x2 }, { AUDIO_S32LSB, 6, 1, 2, SDL_Upsample_S32LSB_6c_x2 }, { AUDIO_S32LSB, 6, 0, 4, SDL_Downsample_S32LSB_6c_x4 }, { AUDIO_S32LSB, 6, 1, 4, SDL_Upsample_S32LSB_6c_x4 }, { AUDIO_S32LSB, 8, 0, 2, SDL_Downsample_S32LSB_8c_x2 }, { AUDIO_S32LSB, 8, 1, 2, SDL_Upsample_S32LSB_8c_x2 }, { AUDIO_S32LSB, 8, 0, 4, SDL_Downsample_S32LSB_8c_x4 }, { AUDIO_S32LSB, 8, 1, 4, SDL_Upsample_S32LSB_8c_x4 }, { AUDIO_S32MSB, 1, 0, 2, SDL_Downsample_S32MSB_1c_x2 }, { AUDIO_S32MSB, 1, 1, 2, SDL_Upsample_S32MSB_1c_x2 }, { AUDIO_S32MSB, 1, 0, 4, SDL_Downsample_S32MSB_1c_x4 }, { AUDIO_S32MSB, 1, 1, 4, SDL_Upsample_S32MSB_1c_x4 }, { AUDIO_S32MSB, 2, 0, 2, SDL_Downsample_S32MSB_2c_x2 }, { AUDIO_S32MSB, 2, 1, 2, SDL_Upsample_S32MSB_2c_x2 }, { AUDIO_S32MSB, 2, 0, 4, SDL_Downsample_S32MSB_2c_x4 }, { AUDIO_S32MSB, 2, 1, 4, SDL_Upsample_S32MSB_2c_x4 }, { AUDIO_S32MSB, 4, 0, 2, SDL_Downsample_S32MSB_4c_x2 }, { AUDIO_S32MSB, 4, 1, 2, SDL_Upsample_S32MSB_4c_x2 }, { AUDIO_S32MSB, 4, 0, 4, SDL_Downsample_S32MSB_4c_x4 }, { AUDIO_S32MSB, 4, 1, 4, SDL_Upsample_S32MSB_4c_x4 }, { AUDIO_S32MSB, 6, 0, 2, SDL_Downsample_S32MSB_6c_x2 }, { AUDIO_S32MSB, 6, 1, 2, SDL_Upsample_S32MSB_6c_x2 }, { AUDIO_S32MSB, 6, 0, 4, SDL_Downsample_S32MSB_6c_x4 }, { AUDIO_S32MSB, 6, 1, 4, SDL_Upsample_S32MSB_6c_x4 }, { AUDIO_S32MSB, 8, 0, 2, SDL_Downsample_S32MSB_8c_x2 }, { AUDIO_S32MSB, 8, 1, 2, SDL_Upsample_S32MSB_8c_x2 }, { AUDIO_S32MSB, 8, 0, 4, SDL_Downsample_S32MSB_8c_x4 }, { AUDIO_S32MSB, 8, 1, 4, SDL_Upsample_S32MSB_8c_x4 }, { AUDIO_F32LSB, 1, 0, 2, SDL_Downsample_F32LSB_1c_x2 }, { AUDIO_F32LSB, 1, 1, 2, SDL_Upsample_F32LSB_1c_x2 }, { AUDIO_F32LSB, 1, 0, 4, SDL_Downsample_F32LSB_1c_x4 }, { AUDIO_F32LSB, 1, 1, 4, SDL_Upsample_F32LSB_1c_x4 }, { AUDIO_F32LSB, 2, 0, 2, SDL_Downsample_F32LSB_2c_x2 }, { AUDIO_F32LSB, 2, 1, 2, SDL_Upsample_F32LSB_2c_x2 }, { AUDIO_F32LSB, 2, 0, 4, SDL_Downsample_F32LSB_2c_x4 }, { AUDIO_F32LSB, 2, 1, 4, SDL_Upsample_F32LSB_2c_x4 }, { AUDIO_F32LSB, 4, 0, 2, SDL_Downsample_F32LSB_4c_x2 }, { AUDIO_F32LSB, 4, 1, 2, SDL_Upsample_F32LSB_4c_x2 }, { AUDIO_F32LSB, 4, 0, 4, SDL_Downsample_F32LSB_4c_x4 }, { AUDIO_F32LSB, 4, 1, 4, SDL_Upsample_F32LSB_4c_x4 }, { AUDIO_F32LSB, 6, 0, 2, SDL_Downsample_F32LSB_6c_x2 }, { AUDIO_F32LSB, 6, 1, 2, SDL_Upsample_F32LSB_6c_x2 }, { AUDIO_F32LSB, 6, 0, 4, SDL_Downsample_F32LSB_6c_x4 }, { AUDIO_F32LSB, 6, 1, 4, SDL_Upsample_F32LSB_6c_x4 }, { AUDIO_F32LSB, 8, 0, 2, SDL_Downsample_F32LSB_8c_x2 }, { AUDIO_F32LSB, 8, 1, 2, SDL_Upsample_F32LSB_8c_x2 }, { AUDIO_F32LSB, 8, 0, 4, SDL_Downsample_F32LSB_8c_x4 }, { AUDIO_F32LSB, 8, 1, 4, SDL_Upsample_F32LSB_8c_x4 }, { AUDIO_F32MSB, 1, 0, 2, SDL_Downsample_F32MSB_1c_x2 }, { AUDIO_F32MSB, 1, 1, 2, SDL_Upsample_F32MSB_1c_x2 }, { AUDIO_F32MSB, 1, 0, 4, SDL_Downsample_F32MSB_1c_x4 }, { AUDIO_F32MSB, 1, 1, 4, SDL_Upsample_F32MSB_1c_x4 }, { AUDIO_F32MSB, 2, 0, 2, SDL_Downsample_F32MSB_2c_x2 }, { AUDIO_F32MSB, 2, 1, 2, SDL_Upsample_F32MSB_2c_x2 }, { AUDIO_F32MSB, 2, 0, 4, SDL_Downsample_F32MSB_2c_x4 }, { AUDIO_F32MSB, 2, 1, 4, SDL_Upsample_F32MSB_2c_x4 }, { AUDIO_F32MSB, 4, 0, 2, SDL_Downsample_F32MSB_4c_x2 }, { AUDIO_F32MSB, 4, 1, 2, SDL_Upsample_F32MSB_4c_x2 }, { AUDIO_F32MSB, 4, 0, 4, SDL_Downsample_F32MSB_4c_x4 }, { AUDIO_F32MSB, 4, 1, 4, SDL_Upsample_F32MSB_4c_x4 }, { AUDIO_F32MSB, 6, 0, 2, SDL_Downsample_F32MSB_6c_x2 }, { AUDIO_F32MSB, 6, 1, 2, SDL_Upsample_F32MSB_6c_x2 }, { AUDIO_F32MSB, 6, 0, 4, SDL_Downsample_F32MSB_6c_x4 }, { AUDIO_F32MSB, 6, 1, 4, SDL_Upsample_F32MSB_6c_x4 }, { AUDIO_F32MSB, 8, 0, 2, SDL_Downsample_F32MSB_8c_x2 }, { AUDIO_F32MSB, 8, 1, 2, SDL_Upsample_F32MSB_8c_x2 }, { AUDIO_F32MSB, 8, 0, 4, SDL_Downsample_F32MSB_8c_x4 }, { AUDIO_F32MSB, 8, 1, 4, SDL_Upsample_F32MSB_8c_x4 }, #endif /* !LESS_RESAMPLERS */ #endif /* !NO_RESAMPLERS */ { 0, 0, 0, 0, NULL } }; /* 390 converters generated. */ /* *INDENT-ON* */ /* vi: set ts=4 sw=4 expandtab: */