|  | /* | 
|  | Simple DirectMedia Layer | 
|  | Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org> | 
|  |  | 
|  | This software is provided 'as-is', without any express or implied | 
|  | warranty.  In no event will the authors be held liable for any damages | 
|  | arising from the use of this software. | 
|  |  | 
|  | Permission is granted to anyone to use this software for any purpose, | 
|  | including commercial applications, and to alter it and redistribute it | 
|  | freely, subject to the following restrictions: | 
|  |  | 
|  | 1. The origin of this software must not be misrepresented; you must not | 
|  | claim that you wrote the original software. If you use this software | 
|  | in a product, an acknowledgment in the product documentation would be | 
|  | appreciated but is not required. | 
|  | 2. Altered source versions must be plainly marked as such, and must not be | 
|  | misrepresented as being the original software. | 
|  | 3. This notice may not be removed or altered from any source distribution. | 
|  | */ | 
|  |  | 
|  | /* DO NOT EDIT, THIS FILE WAS GENERATED BY build-scripts/gen_audio_channel_conversion.c */ | 
|  |  | 
|  | static void SDLCALL SDL_ConvertMonoToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 1) * 2))) - 2; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 1; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("mono", "stereo"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 1); i; i--, src -= 1, dst -= 2) { | 
|  | const float srcFC = src[0]; | 
|  | dst[1] /* FR */ = srcFC; | 
|  | dst[0] /* FL */ = srcFC; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt * 2; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertMonoTo21(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 1) * 3))) - 3; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 1; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("mono", "2.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 1); i; i--, src -= 1, dst -= 3) { | 
|  | const float srcFC = src[0]; | 
|  | dst[2] /* LFE */ = 0.0f; | 
|  | dst[1] /* FR */ = srcFC; | 
|  | dst[0] /* FL */ = srcFC; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt * 3; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertMonoToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 1) * 4))) - 4; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 1; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("mono", "quad"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 1); i; i--, src -= 1, dst -= 4) { | 
|  | const float srcFC = src[0]; | 
|  | dst[3] /* BR */ = 0.0f; | 
|  | dst[2] /* BL */ = 0.0f; | 
|  | dst[1] /* FR */ = srcFC; | 
|  | dst[0] /* FL */ = srcFC; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt * 4; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertMonoTo41(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 1) * 5))) - 5; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 1; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("mono", "4.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 1); i; i--, src -= 1, dst -= 5) { | 
|  | const float srcFC = src[0]; | 
|  | dst[4] /* BR */ = 0.0f; | 
|  | dst[3] /* BL */ = 0.0f; | 
|  | dst[2] /* LFE */ = 0.0f; | 
|  | dst[1] /* FR */ = srcFC; | 
|  | dst[0] /* FL */ = srcFC; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt * 5; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertMonoTo51(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 1) * 6))) - 6; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 1; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("mono", "5.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 1); i; i--, src -= 1, dst -= 6) { | 
|  | const float srcFC = src[0]; | 
|  | dst[5] /* BR */ = 0.0f; | 
|  | dst[4] /* BL */ = 0.0f; | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = srcFC; | 
|  | dst[0] /* FL */ = srcFC; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt * 6; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertMonoTo61(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 1) * 7))) - 7; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 1; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("mono", "6.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 1); i; i--, src -= 1, dst -= 7) { | 
|  | const float srcFC = src[0]; | 
|  | dst[6] /* SR */ = 0.0f; | 
|  | dst[5] /* SL */ = 0.0f; | 
|  | dst[4] /* BC */ = 0.0f; | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = srcFC; | 
|  | dst[0] /* FL */ = srcFC; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt * 7; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertMonoTo71(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 1) * 8))) - 8; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 1; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("mono", "7.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 1); i; i--, src -= 1, dst -= 8) { | 
|  | const float srcFC = src[0]; | 
|  | dst[7] /* SR */ = 0.0f; | 
|  | dst[6] /* SL */ = 0.0f; | 
|  | dst[5] /* BR */ = 0.0f; | 
|  | dst[4] /* BL */ = 0.0f; | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = srcFC; | 
|  | dst[0] /* FL */ = srcFC; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt * 8; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertStereoToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("stereo", "mono"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 2); i; i--, src += 2, dst += 1) { | 
|  | dst[0] /* FC */ = (src[0] * 0.500000000f) + (src[1] * 0.500000000f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt / 2; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertStereoTo21(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 2) * 3))) - 3; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 2; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("stereo", "2.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 2); i; i--, src -= 2, dst -= 3) { | 
|  | dst[2] /* LFE */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 2) * 3; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertStereoToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 2) * 4))) - 4; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 2; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("stereo", "quad"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 2); i; i--, src -= 2, dst -= 4) { | 
|  | dst[3] /* BR */ = 0.0f; | 
|  | dst[2] /* BL */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 2) * 4; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertStereoTo41(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 2) * 5))) - 5; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 2; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("stereo", "4.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 2); i; i--, src -= 2, dst -= 5) { | 
|  | dst[4] /* BR */ = 0.0f; | 
|  | dst[3] /* BL */ = 0.0f; | 
|  | dst[2] /* LFE */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 2) * 5; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertStereoTo51(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 2) * 6))) - 6; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 2; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("stereo", "5.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 2); i; i--, src -= 2, dst -= 6) { | 
|  | dst[5] /* BR */ = 0.0f; | 
|  | dst[4] /* BL */ = 0.0f; | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 2) * 6; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertStereoTo61(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 2) * 7))) - 7; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 2; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("stereo", "6.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 2); i; i--, src -= 2, dst -= 7) { | 
|  | dst[6] /* SR */ = 0.0f; | 
|  | dst[5] /* SL */ = 0.0f; | 
|  | dst[4] /* BC */ = 0.0f; | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 2) * 7; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertStereoTo71(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 2) * 8))) - 8; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 2; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("stereo", "7.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 2); i; i--, src -= 2, dst -= 8) { | 
|  | dst[7] /* SR */ = 0.0f; | 
|  | dst[6] /* SL */ = 0.0f; | 
|  | dst[5] /* BR */ = 0.0f; | 
|  | dst[4] /* BL */ = 0.0f; | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 2) * 8; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert21ToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("2.1", "mono"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 3); i; i--, src += 3, dst += 1) { | 
|  | dst[0] /* FC */ = (src[0] * 0.333333343f) + (src[1] * 0.333333343f) + (src[2] * 0.333333343f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt / 3; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert21ToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("2.1", "stereo"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 3); i; i--, src += 3, dst += 2) { | 
|  | const float srcLFE = src[2]; | 
|  | dst[0] /* FL */ = (src[0] * 0.800000012f) + (srcLFE * 0.200000003f); | 
|  | dst[1] /* FR */ = (src[1] * 0.800000012f) + (srcLFE * 0.200000003f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 3) * 2; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert21ToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 3) * 4))) - 4; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 3; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("2.1", "quad"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 3); i; i--, src -= 3, dst -= 4) { | 
|  | const float srcLFE = src[2]; | 
|  | dst[3] /* BR */ = (srcLFE * 0.111111112f); | 
|  | dst[2] /* BL */ = (srcLFE * 0.111111112f); | 
|  | dst[1] /* FR */ = (srcLFE * 0.111111112f) + (src[1] * 0.888888896f); | 
|  | dst[0] /* FL */ = (srcLFE * 0.111111112f) + (src[0] * 0.888888896f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 3) * 4; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert21To41(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 3) * 5))) - 5; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 3; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("2.1", "4.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 3); i; i--, src -= 3, dst -= 5) { | 
|  | dst[4] /* BR */ = 0.0f; | 
|  | dst[3] /* BL */ = 0.0f; | 
|  | dst[2] /* LFE */ = src[2]; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 3) * 5; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert21To51(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 3) * 6))) - 6; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 3; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("2.1", "5.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 3); i; i--, src -= 3, dst -= 6) { | 
|  | dst[5] /* BR */ = 0.0f; | 
|  | dst[4] /* BL */ = 0.0f; | 
|  | dst[3] /* LFE */ = src[2]; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 3) * 6; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert21To61(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 3) * 7))) - 7; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 3; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("2.1", "6.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 3); i; i--, src -= 3, dst -= 7) { | 
|  | dst[6] /* SR */ = 0.0f; | 
|  | dst[5] /* SL */ = 0.0f; | 
|  | dst[4] /* BC */ = 0.0f; | 
|  | dst[3] /* LFE */ = src[2]; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 3) * 7; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert21To71(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 3) * 8))) - 8; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 3; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("2.1", "7.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 3); i; i--, src -= 3, dst -= 8) { | 
|  | dst[7] /* SR */ = 0.0f; | 
|  | dst[6] /* SL */ = 0.0f; | 
|  | dst[5] /* BR */ = 0.0f; | 
|  | dst[4] /* BL */ = 0.0f; | 
|  | dst[3] /* LFE */ = src[2]; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 3) * 8; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertQuadToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("quad", "mono"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 4); i; i--, src += 4, dst += 1) { | 
|  | dst[0] /* FC */ = (src[0] * 0.250000000f) + (src[1] * 0.250000000f) + (src[2] * 0.250000000f) + (src[3] * 0.250000000f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt / 4; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertQuadToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("quad", "stereo"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 4); i; i--, src += 4, dst += 2) { | 
|  | const float srcBL = src[2]; | 
|  | const float srcBR = src[3]; | 
|  | dst[0] /* FL */ = (src[0] * 0.421000004f) + (srcBL * 0.358999997f) + (srcBR * 0.219999999f); | 
|  | dst[1] /* FR */ = (src[1] * 0.421000004f) + (srcBL * 0.219999999f) + (srcBR * 0.358999997f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 4) * 2; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertQuadTo21(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("quad", "2.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 4); i; i--, src += 4, dst += 3) { | 
|  | const float srcBL = src[2]; | 
|  | const float srcBR = src[3]; | 
|  | dst[0] /* FL */ = (src[0] * 0.421000004f) + (srcBL * 0.358999997f) + (srcBR * 0.219999999f); | 
|  | dst[1] /* FR */ = (src[1] * 0.421000004f) + (srcBL * 0.219999999f) + (srcBR * 0.358999997f); | 
|  | dst[2] /* LFE */ = 0.0f; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 4) * 3; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertQuadTo41(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 4) * 5))) - 5; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 4; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("quad", "4.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 4); i; i--, src -= 4, dst -= 5) { | 
|  | dst[4] /* BR */ = src[3]; | 
|  | dst[3] /* BL */ = src[2]; | 
|  | dst[2] /* LFE */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 4) * 5; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertQuadTo51(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 4) * 6))) - 6; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 4; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("quad", "5.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 4); i; i--, src -= 4, dst -= 6) { | 
|  | dst[5] /* BR */ = src[3]; | 
|  | dst[4] /* BL */ = src[2]; | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 4) * 6; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertQuadTo61(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 4) * 7))) - 7; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 4; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("quad", "6.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 4); i; i--, src -= 4, dst -= 7) { | 
|  | const float srcBL = src[2]; | 
|  | const float srcBR = src[3]; | 
|  | dst[6] /* SR */ = (srcBR * 0.796000004f); | 
|  | dst[5] /* SL */ = (srcBL * 0.796000004f); | 
|  | dst[4] /* BC */ = (srcBR * 0.500000000f) + (srcBL * 0.500000000f); | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = (src[1] * 0.939999998f); | 
|  | dst[0] /* FL */ = (src[0] * 0.939999998f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 4) * 7; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_ConvertQuadTo71(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 4) * 8))) - 8; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 4; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("quad", "7.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 4); i; i--, src -= 4, dst -= 8) { | 
|  | dst[7] /* SR */ = 0.0f; | 
|  | dst[6] /* SL */ = 0.0f; | 
|  | dst[5] /* BR */ = src[3]; | 
|  | dst[4] /* BL */ = src[2]; | 
|  | dst[3] /* LFE */ = 0.0f; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 4) * 8; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert41ToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("4.1", "mono"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 5); i; i--, src += 5, dst += 1) { | 
|  | dst[0] /* FC */ = (src[0] * 0.200000003f) + (src[1] * 0.200000003f) + (src[2] * 0.200000003f) + (src[3] * 0.200000003f) + (src[4] * 0.200000003f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt / 5; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert41ToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("4.1", "stereo"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 5); i; i--, src += 5, dst += 2) { | 
|  | const float srcLFE = src[2]; | 
|  | const float srcBL = src[3]; | 
|  | const float srcBR = src[4]; | 
|  | dst[0] /* FL */ = (src[0] * 0.374222219f) + (srcLFE * 0.111111112f) + (srcBL * 0.319111109f) + (srcBR * 0.195555553f); | 
|  | dst[1] /* FR */ = (src[1] * 0.374222219f) + (srcLFE * 0.111111112f) + (srcBL * 0.195555553f) + (srcBR * 0.319111109f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 5) * 2; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert41To21(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("4.1", "2.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 5); i; i--, src += 5, dst += 3) { | 
|  | const float srcBL = src[3]; | 
|  | const float srcBR = src[4]; | 
|  | dst[0] /* FL */ = (src[0] * 0.421000004f) + (srcBL * 0.358999997f) + (srcBR * 0.219999999f); | 
|  | dst[1] /* FR */ = (src[1] * 0.421000004f) + (srcBL * 0.219999999f) + (srcBR * 0.358999997f); | 
|  | dst[2] /* LFE */ = src[2]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 5) * 3; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert41ToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("4.1", "quad"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 5); i; i--, src += 5, dst += 4) { | 
|  | const float srcLFE = src[2]; | 
|  | dst[0] /* FL */ = (src[0] * 0.941176474f) + (srcLFE * 0.058823530f); | 
|  | dst[1] /* FR */ = (src[1] * 0.941176474f) + (srcLFE * 0.058823530f); | 
|  | dst[2] /* BL */ = (srcLFE * 0.058823530f) + (src[3] * 0.941176474f); | 
|  | dst[3] /* BR */ = (srcLFE * 0.058823530f) + (src[4] * 0.941176474f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 5) * 4; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert41To51(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 5) * 6))) - 6; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 5; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("4.1", "5.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 5); i; i--, src -= 5, dst -= 6) { | 
|  | dst[5] /* BR */ = src[4]; | 
|  | dst[4] /* BL */ = src[3]; | 
|  | dst[3] /* LFE */ = src[2]; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 5) * 6; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert41To61(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 5) * 7))) - 7; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 5; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("4.1", "6.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 5); i; i--, src -= 5, dst -= 7) { | 
|  | const float srcBL = src[3]; | 
|  | const float srcBR = src[4]; | 
|  | dst[6] /* SR */ = (srcBR * 0.796000004f); | 
|  | dst[5] /* SL */ = (srcBL * 0.796000004f); | 
|  | dst[4] /* BC */ = (srcBR * 0.500000000f) + (srcBL * 0.500000000f); | 
|  | dst[3] /* LFE */ = src[2]; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = (src[1] * 0.939999998f); | 
|  | dst[0] /* FL */ = (src[0] * 0.939999998f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 5) * 7; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert41To71(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 5) * 8))) - 8; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 5; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("4.1", "7.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 5); i; i--, src -= 5, dst -= 8) { | 
|  | dst[7] /* SR */ = 0.0f; | 
|  | dst[6] /* SL */ = 0.0f; | 
|  | dst[5] /* BR */ = src[4]; | 
|  | dst[4] /* BL */ = src[3]; | 
|  | dst[3] /* LFE */ = src[2]; | 
|  | dst[2] /* FC */ = 0.0f; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 5) * 8; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert51ToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("5.1", "mono"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 6); i; i--, src += 6, dst += 1) { | 
|  | dst[0] /* FC */ = (src[0] * 0.166666672f) + (src[1] * 0.166666672f) + (src[2] * 0.166666672f) + (src[3] * 0.166666672f) + (src[4] * 0.166666672f) + (src[5] * 0.166666672f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt / 6; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert51ToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("5.1", "stereo"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 6); i; i--, src += 6, dst += 2) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcLFE = src[3]; | 
|  | const float srcBL = src[4]; | 
|  | const float srcBR = src[5]; | 
|  | dst[0] /* FL */ = (src[0] * 0.294545442f) + (srcFC * 0.208181813f) + (srcLFE * 0.090909094f) + (srcBL * 0.251818180f) + (srcBR * 0.154545456f); | 
|  | dst[1] /* FR */ = (src[1] * 0.294545442f) + (srcFC * 0.208181813f) + (srcLFE * 0.090909094f) + (srcBL * 0.154545456f) + (srcBR * 0.251818180f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 6) * 2; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert51To21(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("5.1", "2.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 6); i; i--, src += 6, dst += 3) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcBL = src[4]; | 
|  | const float srcBR = src[5]; | 
|  | dst[0] /* FL */ = (src[0] * 0.324000001f) + (srcFC * 0.229000002f) + (srcBL * 0.277000010f) + (srcBR * 0.170000002f); | 
|  | dst[1] /* FR */ = (src[1] * 0.324000001f) + (srcFC * 0.229000002f) + (srcBL * 0.170000002f) + (srcBR * 0.277000010f); | 
|  | dst[2] /* LFE */ = src[3]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 6) * 3; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert51ToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("5.1", "quad"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 6); i; i--, src += 6, dst += 4) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcLFE = src[3]; | 
|  | dst[0] /* FL */ = (src[0] * 0.558095276f) + (srcFC * 0.394285709f) + (srcLFE * 0.047619049f); | 
|  | dst[1] /* FR */ = (src[1] * 0.558095276f) + (srcFC * 0.394285709f) + (srcLFE * 0.047619049f); | 
|  | dst[2] /* BL */ = (srcLFE * 0.047619049f) + (src[4] * 0.558095276f); | 
|  | dst[3] /* BR */ = (srcLFE * 0.047619049f) + (src[5] * 0.558095276f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 6) * 4; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert51To41(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("5.1", "4.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 6); i; i--, src += 6, dst += 5) { | 
|  | const float srcFC = src[2]; | 
|  | dst[0] /* FL */ = (src[0] * 0.586000025f) + (srcFC * 0.414000005f); | 
|  | dst[1] /* FR */ = (src[1] * 0.586000025f) + (srcFC * 0.414000005f); | 
|  | dst[2] /* LFE */ = src[3]; | 
|  | dst[3] /* BL */ = (src[4] * 0.586000025f); | 
|  | dst[4] /* BR */ = (src[5] * 0.586000025f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 6) * 5; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert51To61(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 6) * 7))) - 7; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 6; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("5.1", "6.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 6); i; i--, src -= 6, dst -= 7) { | 
|  | const float srcBL = src[4]; | 
|  | const float srcBR = src[5]; | 
|  | dst[6] /* SR */ = (srcBR * 0.796000004f); | 
|  | dst[5] /* SL */ = (srcBL * 0.796000004f); | 
|  | dst[4] /* BC */ = (srcBR * 0.500000000f) + (srcBL * 0.500000000f); | 
|  | dst[3] /* LFE */ = src[3]; | 
|  | dst[2] /* FC */ = (src[2] * 0.939999998f); | 
|  | dst[1] /* FR */ = (src[1] * 0.939999998f); | 
|  | dst[0] /* FL */ = (src[0] * 0.939999998f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 6) * 7; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert51To71(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 6) * 8))) - 8; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 6; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("5.1", "7.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 6); i; i--, src -= 6, dst -= 8) { | 
|  | dst[7] /* SR */ = 0.0f; | 
|  | dst[6] /* SL */ = 0.0f; | 
|  | dst[5] /* BR */ = src[5]; | 
|  | dst[4] /* BL */ = src[4]; | 
|  | dst[3] /* LFE */ = src[3]; | 
|  | dst[2] /* FC */ = src[2]; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 6) * 8; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert61ToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("6.1", "mono"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 7); i; i--, src += 7, dst += 1) { | 
|  | dst[0] /* FC */ = (src[0] * 0.143142849f) + (src[1] * 0.143142849f) + (src[2] * 0.143142849f) + (src[3] * 0.142857149f) + (src[4] * 0.143142849f) + (src[5] * 0.143142849f) + (src[6] * 0.143142849f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt / 7; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert61ToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("6.1", "stereo"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 7); i; i--, src += 7, dst += 2) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcLFE = src[3]; | 
|  | const float srcBC = src[4]; | 
|  | const float srcSL = src[5]; | 
|  | const float srcSR = src[6]; | 
|  | dst[0] /* FL */ = (src[0] * 0.247384623f) + (srcFC * 0.174461529f) + (srcLFE * 0.076923080f) + (srcBC * 0.174461529f) + (srcSL * 0.226153851f) + (srcSR * 0.100615382f); | 
|  | dst[1] /* FR */ = (src[1] * 0.247384623f) + (srcFC * 0.174461529f) + (srcLFE * 0.076923080f) + (srcBC * 0.174461529f) + (srcSL * 0.100615382f) + (srcSR * 0.226153851f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 7) * 2; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert61To21(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("6.1", "2.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 7); i; i--, src += 7, dst += 3) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcBC = src[4]; | 
|  | const float srcSL = src[5]; | 
|  | const float srcSR = src[6]; | 
|  | dst[0] /* FL */ = (src[0] * 0.268000007f) + (srcFC * 0.188999996f) + (srcBC * 0.188999996f) + (srcSL * 0.245000005f) + (srcSR * 0.108999997f); | 
|  | dst[1] /* FR */ = (src[1] * 0.268000007f) + (srcFC * 0.188999996f) + (srcBC * 0.188999996f) + (srcSL * 0.108999997f) + (srcSR * 0.245000005f); | 
|  | dst[2] /* LFE */ = src[3]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 7) * 3; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert61ToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("6.1", "quad"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 7); i; i--, src += 7, dst += 4) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcLFE = src[3]; | 
|  | const float srcBC = src[4]; | 
|  | const float srcSL = src[5]; | 
|  | const float srcSR = src[6]; | 
|  | dst[0] /* FL */ = (src[0] * 0.463679999f) + (srcFC * 0.327360004f) + (srcLFE * 0.040000003f) + (srcSL * 0.168960005f); | 
|  | dst[1] /* FR */ = (src[1] * 0.463679999f) + (srcFC * 0.327360004f) + (srcLFE * 0.040000003f) + (srcSR * 0.168960005f); | 
|  | dst[2] /* BL */ = (srcLFE * 0.040000003f) + (srcBC * 0.327360004f) + (srcSL * 0.431039989f); | 
|  | dst[3] /* BR */ = (srcLFE * 0.040000003f) + (srcBC * 0.327360004f) + (srcSR * 0.431039989f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 7) * 4; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert61To41(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("6.1", "4.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 7); i; i--, src += 7, dst += 5) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcBC = src[4]; | 
|  | const float srcSL = src[5]; | 
|  | const float srcSR = src[6]; | 
|  | dst[0] /* FL */ = (src[0] * 0.483000010f) + (srcFC * 0.340999991f) + (srcSL * 0.175999999f); | 
|  | dst[1] /* FR */ = (src[1] * 0.483000010f) + (srcFC * 0.340999991f) + (srcSR * 0.175999999f); | 
|  | dst[2] /* LFE */ = src[3]; | 
|  | dst[3] /* BL */ = (srcBC * 0.340999991f) + (srcSL * 0.449000001f); | 
|  | dst[4] /* BR */ = (srcBC * 0.340999991f) + (srcSR * 0.449000001f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 7) * 5; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert61To51(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("6.1", "5.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 7); i; i--, src += 7, dst += 6) { | 
|  | const float srcBC = src[4]; | 
|  | const float srcSL = src[5]; | 
|  | const float srcSR = src[6]; | 
|  | dst[0] /* FL */ = (src[0] * 0.611000001f) + (srcSL * 0.223000005f); | 
|  | dst[1] /* FR */ = (src[1] * 0.611000001f) + (srcSR * 0.223000005f); | 
|  | dst[2] /* FC */ = (src[2] * 0.611000001f); | 
|  | dst[3] /* LFE */ = src[3]; | 
|  | dst[4] /* BL */ = (srcBC * 0.432000011f) + (srcSL * 0.568000019f); | 
|  | dst[5] /* BR */ = (srcBC * 0.432000011f) + (srcSR * 0.568000019f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 7) * 6; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert61To71(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = ((float *)(cvt->buf + ((cvt->len_cvt / 7) * 8))) - 8; | 
|  | const float *src = ((const float *)(cvt->buf + cvt->len_cvt)) - 7; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("6.1", "7.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | /* convert backwards, since output is growing in-place. */ | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 7); i; i--, src -= 7, dst -= 8) { | 
|  | const float srcBC = src[4]; | 
|  | dst[7] /* SR */ = src[6]; | 
|  | dst[6] /* SL */ = src[5]; | 
|  | dst[5] /* BR */ = (srcBC * 0.707000017f); | 
|  | dst[4] /* BL */ = (srcBC * 0.707000017f); | 
|  | dst[3] /* LFE */ = src[3]; | 
|  | dst[2] /* FC */ = src[2]; | 
|  | dst[1] /* FR */ = src[1]; | 
|  | dst[0] /* FL */ = src[0]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 7) * 8; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert71ToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("7.1", "mono"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 8); i; i--, src += 8, dst += 1) { | 
|  | dst[0] /* FC */ = (src[0] * 0.125125006f) + (src[1] * 0.125125006f) + (src[2] * 0.125125006f) + (src[3] * 0.125000000f) + (src[4] * 0.125125006f) + (src[5] * 0.125125006f) + (src[6] * 0.125125006f) + (src[7] * 0.125125006f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = cvt->len_cvt / 8; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert71ToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("7.1", "stereo"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 8); i; i--, src += 8, dst += 2) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcLFE = src[3]; | 
|  | const float srcBL = src[4]; | 
|  | const float srcBR = src[5]; | 
|  | const float srcSL = src[6]; | 
|  | const float srcSR = src[7]; | 
|  | dst[0] /* FL */ = (src[0] * 0.211866662f) + (srcFC * 0.150266662f) + (srcLFE * 0.066666670f) + (srcBL * 0.181066677f) + (srcBR * 0.111066669f) + (srcSL * 0.194133341f) + (srcSR * 0.085866667f); | 
|  | dst[1] /* FR */ = (src[1] * 0.211866662f) + (srcFC * 0.150266662f) + (srcLFE * 0.066666670f) + (srcBL * 0.111066669f) + (srcBR * 0.181066677f) + (srcSL * 0.085866667f) + (srcSR * 0.194133341f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 8) * 2; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert71To21(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("7.1", "2.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 8); i; i--, src += 8, dst += 3) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcBL = src[4]; | 
|  | const float srcBR = src[5]; | 
|  | const float srcSL = src[6]; | 
|  | const float srcSR = src[7]; | 
|  | dst[0] /* FL */ = (src[0] * 0.226999998f) + (srcFC * 0.160999998f) + (srcBL * 0.194000006f) + (srcBR * 0.119000003f) + (srcSL * 0.208000004f) + (srcSR * 0.092000000f); | 
|  | dst[1] /* FR */ = (src[1] * 0.226999998f) + (srcFC * 0.160999998f) + (srcBL * 0.119000003f) + (srcBR * 0.194000006f) + (srcSL * 0.092000000f) + (srcSR * 0.208000004f); | 
|  | dst[2] /* LFE */ = src[3]; | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 8) * 3; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert71ToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("7.1", "quad"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 8); i; i--, src += 8, dst += 4) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcLFE = src[3]; | 
|  | const float srcSL = src[6]; | 
|  | const float srcSR = src[7]; | 
|  | dst[0] /* FL */ = (src[0] * 0.466344833f) + (srcFC * 0.329241365f) + (srcLFE * 0.034482758f) + (srcSL * 0.169931039f); | 
|  | dst[1] /* FR */ = (src[1] * 0.466344833f) + (srcFC * 0.329241365f) + (srcLFE * 0.034482758f) + (srcSR * 0.169931039f); | 
|  | dst[2] /* BL */ = (srcLFE * 0.034482758f) + (src[4] * 0.466344833f) + (srcSL * 0.433517247f); | 
|  | dst[3] /* BR */ = (srcLFE * 0.034482758f) + (src[5] * 0.466344833f) + (srcSR * 0.433517247f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 8) * 4; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert71To41(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("7.1", "4.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 8); i; i--, src += 8, dst += 5) { | 
|  | const float srcFC = src[2]; | 
|  | const float srcSL = src[6]; | 
|  | const float srcSR = src[7]; | 
|  | dst[0] /* FL */ = (src[0] * 0.483000010f) + (srcFC * 0.340999991f) + (srcSL * 0.175999999f); | 
|  | dst[1] /* FR */ = (src[1] * 0.483000010f) + (srcFC * 0.340999991f) + (srcSR * 0.175999999f); | 
|  | dst[2] /* LFE */ = src[3]; | 
|  | dst[3] /* BL */ = (src[4] * 0.483000010f) + (srcSL * 0.449000001f); | 
|  | dst[4] /* BR */ = (src[5] * 0.483000010f) + (srcSR * 0.449000001f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 8) * 5; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert71To51(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("7.1", "5.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 8); i; i--, src += 8, dst += 6) { | 
|  | const float srcSL = src[6]; | 
|  | const float srcSR = src[7]; | 
|  | dst[0] /* FL */ = (src[0] * 0.518000007f) + (srcSL * 0.188999996f); | 
|  | dst[1] /* FR */ = (src[1] * 0.518000007f) + (srcSR * 0.188999996f); | 
|  | dst[2] /* FC */ = (src[2] * 0.518000007f); | 
|  | dst[3] /* LFE */ = src[3]; | 
|  | dst[4] /* BL */ = (src[4] * 0.518000007f) + (srcSL * 0.481999993f); | 
|  | dst[5] /* BR */ = (src[5] * 0.518000007f) + (srcSR * 0.481999993f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 8) * 6; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static void SDLCALL SDL_Convert71To61(SDL_AudioCVT *cvt, SDL_AudioFormat format) | 
|  | { | 
|  | float *dst = (float *)cvt->buf; | 
|  | const float *src = dst; | 
|  | int i; | 
|  |  | 
|  | LOG_DEBUG_CONVERT("7.1", "6.1"); | 
|  | SDL_assert(format == AUDIO_F32SYS); | 
|  |  | 
|  | for (i = cvt->len_cvt / (sizeof(float) * 8); i; i--, src += 8, dst += 7) { | 
|  | const float srcBL = src[4]; | 
|  | const float srcBR = src[5]; | 
|  | dst[0] /* FL */ = (src[0] * 0.541000009f); | 
|  | dst[1] /* FR */ = (src[1] * 0.541000009f); | 
|  | dst[2] /* FC */ = (src[2] * 0.541000009f); | 
|  | dst[3] /* LFE */ = src[3]; | 
|  | dst[4] /* BC */ = (srcBL * 0.287999988f) + (srcBR * 0.287999988f); | 
|  | dst[5] /* SL */ = (srcBL * 0.458999991f) + (src[6] * 0.541000009f); | 
|  | dst[6] /* SR */ = (srcBR * 0.458999991f) + (src[7] * 0.541000009f); | 
|  | } | 
|  |  | 
|  | cvt->len_cvt = (cvt->len_cvt / 8) * 7; | 
|  | if (cvt->filters[++cvt->filter_index]) { | 
|  | cvt->filters[cvt->filter_index](cvt, format); | 
|  | } | 
|  | } | 
|  |  | 
|  | static const SDL_AudioFilter channel_converters[8][8] = { /* [from][to] */ | 
|  | { NULL, SDL_ConvertMonoToStereo, SDL_ConvertMonoTo21, SDL_ConvertMonoToQuad, SDL_ConvertMonoTo41, SDL_ConvertMonoTo51, SDL_ConvertMonoTo61, SDL_ConvertMonoTo71 }, | 
|  | { SDL_ConvertStereoToMono, NULL, SDL_ConvertStereoTo21, SDL_ConvertStereoToQuad, SDL_ConvertStereoTo41, SDL_ConvertStereoTo51, SDL_ConvertStereoTo61, SDL_ConvertStereoTo71 }, | 
|  | { SDL_Convert21ToMono, SDL_Convert21ToStereo, NULL, SDL_Convert21ToQuad, SDL_Convert21To41, SDL_Convert21To51, SDL_Convert21To61, SDL_Convert21To71 }, | 
|  | { SDL_ConvertQuadToMono, SDL_ConvertQuadToStereo, SDL_ConvertQuadTo21, NULL, SDL_ConvertQuadTo41, SDL_ConvertQuadTo51, SDL_ConvertQuadTo61, SDL_ConvertQuadTo71 }, | 
|  | { SDL_Convert41ToMono, SDL_Convert41ToStereo, SDL_Convert41To21, SDL_Convert41ToQuad, NULL, SDL_Convert41To51, SDL_Convert41To61, SDL_Convert41To71 }, | 
|  | { SDL_Convert51ToMono, SDL_Convert51ToStereo, SDL_Convert51To21, SDL_Convert51ToQuad, SDL_Convert51To41, NULL, SDL_Convert51To61, SDL_Convert51To71 }, | 
|  | { SDL_Convert61ToMono, SDL_Convert61ToStereo, SDL_Convert61To21, SDL_Convert61ToQuad, SDL_Convert61To41, SDL_Convert61To51, NULL, SDL_Convert61To71 }, | 
|  | { SDL_Convert71ToMono, SDL_Convert71ToStereo, SDL_Convert71To21, SDL_Convert71ToQuad, SDL_Convert71To41, SDL_Convert71To51, SDL_Convert71To61, NULL } | 
|  | }; |