| /* |
| 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 } |
| }; |
| |
| /* vi: set ts=4 sw=4 expandtab: */ |