| /* |
| 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. |
| */ |
| #include "../SDL_internal.h" |
| |
| #include "SDL_audio.h" |
| #include "SDL_audio_c.h" |
| #include "SDL_cpuinfo.h" |
| |
| #ifdef __ARM_NEON |
| #define HAVE_NEON_INTRINSICS 1 |
| #endif |
| |
| #ifdef __SSE2__ |
| #define HAVE_SSE2_INTRINSICS 1 |
| #endif |
| |
| #if defined(__x86_64__) && HAVE_SSE2_INTRINSICS |
| #define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* x86_64 guarantees SSE2. */ |
| #elif __MACOSX__ && HAVE_SSE2_INTRINSICS |
| #define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* Mac OS X/Intel guarantees SSE2. */ |
| #elif defined(__ARM_ARCH) && (__ARM_ARCH >= 8) && HAVE_NEON_INTRINSICS |
| #define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* ARMv8+ promise NEON. */ |
| #elif defined(__APPLE__) && defined(__ARM_ARCH) && (__ARM_ARCH >= 7) && HAVE_NEON_INTRINSICS |
| #define NEED_SCALAR_CONVERTER_FALLBACKS 0 /* All Apple ARMv7 chips promise NEON support. */ |
| #endif |
| |
| /* Set to zero if platform is guaranteed to use a SIMD codepath here. */ |
| #ifndef NEED_SCALAR_CONVERTER_FALLBACKS |
| #define NEED_SCALAR_CONVERTER_FALLBACKS 1 |
| #endif |
| |
| /* Function pointers set to a CPU-specific implementation. */ |
| SDL_AudioFilter SDL_Convert_S8_to_F32 = NULL; |
| SDL_AudioFilter SDL_Convert_U8_to_F32 = NULL; |
| SDL_AudioFilter SDL_Convert_S16_to_F32 = NULL; |
| SDL_AudioFilter SDL_Convert_U16_to_F32 = NULL; |
| SDL_AudioFilter SDL_Convert_S32_to_F32 = NULL; |
| SDL_AudioFilter SDL_Convert_F32_to_S8 = NULL; |
| SDL_AudioFilter SDL_Convert_F32_to_U8 = NULL; |
| SDL_AudioFilter SDL_Convert_F32_to_S16 = NULL; |
| SDL_AudioFilter SDL_Convert_F32_to_U16 = NULL; |
| SDL_AudioFilter SDL_Convert_F32_to_S32 = NULL; |
| |
| #define DIVBY128 0.0078125f |
| #define DIVBY32768 0.000030517578125f |
| #define DIVBY8388607 0.00000011920930376163766f |
| |
| #if NEED_SCALAR_CONVERTER_FALLBACKS |
| static void SDLCALL SDL_Convert_S8_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint8 *src = ((const Sint8 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 4)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S8", "AUDIO_F32"); |
| |
| for (i = cvt->len_cvt; i; --i, --src, --dst) { |
| *dst = ((float)*src) * DIVBY128; |
| } |
| |
| cvt->len_cvt *= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_U8_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Uint8 *src = ((const Uint8 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 4)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_U8", "AUDIO_F32"); |
| |
| for (i = cvt->len_cvt; i; --i, --src, --dst) { |
| *dst = (((float)*src) * DIVBY128) - 1.0f; |
| } |
| |
| cvt->len_cvt *= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_S16_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint16 *src = ((const Sint16 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 2)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S16", "AUDIO_F32"); |
| |
| for (i = cvt->len_cvt / sizeof(Sint16); i; --i, --src, --dst) { |
| *dst = ((float)*src) * DIVBY32768; |
| } |
| |
| cvt->len_cvt *= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_U16_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Uint16 *src = ((const Uint16 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 2)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_U16", "AUDIO_F32"); |
| |
| for (i = cvt->len_cvt / sizeof(Uint16); i; --i, --src, --dst) { |
| *dst = (((float)*src) * DIVBY32768) - 1.0f; |
| } |
| |
| cvt->len_cvt *= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_S32_to_F32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint32 *src = (const Sint32 *)cvt->buf; |
| float *dst = (float *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S32", "AUDIO_F32"); |
| |
| for (i = cvt->len_cvt / sizeof(Sint32); i; --i, ++src, ++dst) { |
| *dst = ((float)(*src >> 8)) * DIVBY8388607; |
| } |
| |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint8 *dst = (Sint8 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S8"); |
| |
| for (i = cvt->len_cvt / sizeof(float); i; --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 127; |
| } else if (sample <= -1.0f) { |
| *dst = -128; |
| } else { |
| *dst = (Sint8)(sample * 127.0f); |
| } |
| } |
| |
| cvt->len_cvt /= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S8); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_U8_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Uint8 *dst = (Uint8 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U8"); |
| |
| for (i = cvt->len_cvt / sizeof(float); i; --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 255; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint8)((sample + 1.0f) * 127.0f); |
| } |
| } |
| |
| cvt->len_cvt /= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_U8); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint16 *dst = (Sint16 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S16"); |
| |
| for (i = cvt->len_cvt / sizeof(float); i; --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 32767; |
| } else if (sample <= -1.0f) { |
| *dst = -32768; |
| } else { |
| *dst = (Sint16)(sample * 32767.0f); |
| } |
| } |
| |
| cvt->len_cvt /= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S16SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_U16_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Uint16 *dst = (Uint16 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U16"); |
| |
| for (i = cvt->len_cvt / sizeof(float); i; --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 65535; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint16)((sample + 1.0f) * 32767.0f); |
| } |
| } |
| |
| cvt->len_cvt /= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_U16SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S32_Scalar(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint32 *dst = (Sint32 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S32"); |
| |
| for (i = cvt->len_cvt / sizeof(float); i; --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 2147483647; |
| } else if (sample <= -1.0f) { |
| *dst = (Sint32)-2147483648LL; |
| } else { |
| *dst = ((Sint32)(sample * 8388607.0f)) << 8; |
| } |
| } |
| |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S32SYS); |
| } |
| } |
| #endif |
| |
| #if HAVE_SSE2_INTRINSICS |
| static void SDLCALL SDL_Convert_S8_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint8 *src = ((const Sint8 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 4)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S8", "AUDIO_F32 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */ |
| for (i = cvt->len_cvt; i && (((size_t)(dst - 15)) & 15); --i, --src, --dst) { |
| *dst = ((float)*src) * DIVBY128; |
| } |
| |
| src -= 15; |
| dst -= 15; /* adjust to read SSE blocks from the start. */ |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128i *mmsrc = (const __m128i *)src; |
| const __m128i zero = _mm_setzero_si128(); |
| const __m128 divby128 = _mm_set1_ps(DIVBY128); |
| while (i >= 16) { /* 16 * 8-bit */ |
| const __m128i bytes = _mm_load_si128(mmsrc); /* get 16 sint8 into an XMM register. */ |
| /* treat as int16, shift left to clear every other sint16, then back right with sign-extend. Now sint16. */ |
| const __m128i shorts1 = _mm_srai_epi16(_mm_slli_epi16(bytes, 8), 8); |
| /* right-shift-sign-extend gets us sint16 with the other set of values. */ |
| const __m128i shorts2 = _mm_srai_epi16(bytes, 8); |
| /* unpack against zero to make these int32, shift to make them sign-extend, convert to float, multiply. Whew! */ |
| const __m128 floats1 = _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_slli_epi32(_mm_unpacklo_epi16(shorts1, zero), 16), 16)), divby128); |
| const __m128 floats2 = _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_slli_epi32(_mm_unpacklo_epi16(shorts2, zero), 16), 16)), divby128); |
| const __m128 floats3 = _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_slli_epi32(_mm_unpackhi_epi16(shorts1, zero), 16), 16)), divby128); |
| const __m128 floats4 = _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_slli_epi32(_mm_unpackhi_epi16(shorts2, zero), 16), 16)), divby128); |
| /* Interleave back into correct order, store. */ |
| _mm_store_ps(dst, _mm_unpacklo_ps(floats1, floats2)); |
| _mm_store_ps(dst + 4, _mm_unpackhi_ps(floats1, floats2)); |
| _mm_store_ps(dst + 8, _mm_unpacklo_ps(floats3, floats4)); |
| _mm_store_ps(dst + 12, _mm_unpackhi_ps(floats3, floats4)); |
| i -= 16; |
| mmsrc--; |
| dst -= 16; |
| } |
| |
| src = (const Sint8 *)mmsrc; |
| } |
| |
| src += 15; |
| dst += 15; /* adjust for any scalar finishing. */ |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = ((float)*src) * DIVBY128; |
| i--; |
| src--; |
| dst--; |
| } |
| |
| cvt->len_cvt *= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_U8_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Uint8 *src = ((const Uint8 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 4)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_U8", "AUDIO_F32 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */ |
| for (i = cvt->len_cvt; i && (((size_t)(dst - 15)) & 15); --i, --src, --dst) { |
| *dst = (((float)*src) * DIVBY128) - 1.0f; |
| } |
| |
| src -= 15; |
| dst -= 15; /* adjust to read SSE blocks from the start. */ |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128i *mmsrc = (const __m128i *)src; |
| const __m128i zero = _mm_setzero_si128(); |
| const __m128 divby128 = _mm_set1_ps(DIVBY128); |
| const __m128 minus1 = _mm_set1_ps(-1.0f); |
| while (i >= 16) { /* 16 * 8-bit */ |
| const __m128i bytes = _mm_load_si128(mmsrc); /* get 16 uint8 into an XMM register. */ |
| /* treat as int16, shift left to clear every other sint16, then back right with zero-extend. Now uint16. */ |
| const __m128i shorts1 = _mm_srli_epi16(_mm_slli_epi16(bytes, 8), 8); |
| /* right-shift-zero-extend gets us uint16 with the other set of values. */ |
| const __m128i shorts2 = _mm_srli_epi16(bytes, 8); |
| /* unpack against zero to make these int32, convert to float, multiply, add. Whew! */ |
| /* Note that AVX2 can do floating point multiply+add in one instruction, fwiw. SSE2 cannot. */ |
| const __m128 floats1 = _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpacklo_epi16(shorts1, zero)), divby128), minus1); |
| const __m128 floats2 = _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpacklo_epi16(shorts2, zero)), divby128), minus1); |
| const __m128 floats3 = _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpackhi_epi16(shorts1, zero)), divby128), minus1); |
| const __m128 floats4 = _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpackhi_epi16(shorts2, zero)), divby128), minus1); |
| /* Interleave back into correct order, store. */ |
| _mm_store_ps(dst, _mm_unpacklo_ps(floats1, floats2)); |
| _mm_store_ps(dst + 4, _mm_unpackhi_ps(floats1, floats2)); |
| _mm_store_ps(dst + 8, _mm_unpacklo_ps(floats3, floats4)); |
| _mm_store_ps(dst + 12, _mm_unpackhi_ps(floats3, floats4)); |
| i -= 16; |
| mmsrc--; |
| dst -= 16; |
| } |
| |
| src = (const Uint8 *)mmsrc; |
| } |
| |
| src += 15; |
| dst += 15; /* adjust for any scalar finishing. */ |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = (((float)*src) * DIVBY128) - 1.0f; |
| i--; |
| src--; |
| dst--; |
| } |
| |
| cvt->len_cvt *= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_S16_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint16 *src = ((const Sint16 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 2)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S16", "AUDIO_F32 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */ |
| for (i = cvt->len_cvt / sizeof(Sint16); i && (((size_t)(dst - 7)) & 15); --i, --src, --dst) { |
| *dst = ((float)*src) * DIVBY32768; |
| } |
| |
| src -= 7; |
| dst -= 7; /* adjust to read SSE blocks from the start. */ |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128 divby32768 = _mm_set1_ps(DIVBY32768); |
| while (i >= 8) { /* 8 * 16-bit */ |
| const __m128i ints = _mm_load_si128((__m128i const *)src); /* get 8 sint16 into an XMM register. */ |
| /* treat as int32, shift left to clear every other sint16, then back right with sign-extend. Now sint32. */ |
| const __m128i a = _mm_srai_epi32(_mm_slli_epi32(ints, 16), 16); |
| /* right-shift-sign-extend gets us sint32 with the other set of values. */ |
| const __m128i b = _mm_srai_epi32(ints, 16); |
| /* Interleave these back into the right order, convert to float, multiply, store. */ |
| _mm_store_ps(dst, _mm_mul_ps(_mm_cvtepi32_ps(_mm_unpacklo_epi32(a, b)), divby32768)); |
| _mm_store_ps(dst + 4, _mm_mul_ps(_mm_cvtepi32_ps(_mm_unpackhi_epi32(a, b)), divby32768)); |
| i -= 8; |
| src -= 8; |
| dst -= 8; |
| } |
| } |
| |
| src += 7; |
| dst += 7; /* adjust for any scalar finishing. */ |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = ((float)*src) * DIVBY32768; |
| i--; |
| src--; |
| dst--; |
| } |
| |
| cvt->len_cvt *= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_U16_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Uint16 *src = ((const Uint16 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 2)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_U16", "AUDIO_F32 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */ |
| for (i = cvt->len_cvt / sizeof(Sint16); i && (((size_t)(dst - 7)) & 15); --i, --src, --dst) { |
| *dst = (((float)*src) * DIVBY32768) - 1.0f; |
| } |
| |
| src -= 7; |
| dst -= 7; /* adjust to read SSE blocks from the start. */ |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128 divby32768 = _mm_set1_ps(DIVBY32768); |
| const __m128 minus1 = _mm_set1_ps(-1.0f); |
| while (i >= 8) { /* 8 * 16-bit */ |
| const __m128i ints = _mm_load_si128((__m128i const *)src); /* get 8 sint16 into an XMM register. */ |
| /* treat as int32, shift left to clear every other sint16, then back right with zero-extend. Now sint32. */ |
| const __m128i a = _mm_srli_epi32(_mm_slli_epi32(ints, 16), 16); |
| /* right-shift-sign-extend gets us sint32 with the other set of values. */ |
| const __m128i b = _mm_srli_epi32(ints, 16); |
| /* Interleave these back into the right order, convert to float, multiply, store. */ |
| _mm_store_ps(dst, _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpacklo_epi32(a, b)), divby32768), minus1)); |
| _mm_store_ps(dst + 4, _mm_add_ps(_mm_mul_ps(_mm_cvtepi32_ps(_mm_unpackhi_epi32(a, b)), divby32768), minus1)); |
| i -= 8; |
| src -= 8; |
| dst -= 8; |
| } |
| } |
| |
| src += 7; |
| dst += 7; /* adjust for any scalar finishing. */ |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = (((float)*src) * DIVBY32768) - 1.0f; |
| i--; |
| src--; |
| dst--; |
| } |
| |
| cvt->len_cvt *= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_S32_to_F32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint32 *src = (const Sint32 *)cvt->buf; |
| float *dst = (float *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S32", "AUDIO_F32 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(Sint32); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| *dst = ((float)(*src >> 8)) * DIVBY8388607; |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128 divby8388607 = _mm_set1_ps(DIVBY8388607); |
| const __m128i *mmsrc = (const __m128i *)src; |
| while (i >= 4) { /* 4 * sint32 */ |
| /* shift out lowest bits so int fits in a float32. Small precision loss, but much faster. */ |
| _mm_store_ps(dst, _mm_mul_ps(_mm_cvtepi32_ps(_mm_srai_epi32(_mm_load_si128(mmsrc), 8)), divby8388607)); |
| i -= 4; |
| mmsrc++; |
| dst += 4; |
| } |
| src = (const Sint32 *)mmsrc; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = ((float)(*src >> 8)) * DIVBY8388607; |
| i--; |
| src++; |
| dst++; |
| } |
| |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint8 *dst = (Sint8 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S8 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 127; |
| } else if (sample <= -1.0f) { |
| *dst = -128; |
| } else { |
| *dst = (Sint8)(sample * 127.0f); |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128 one = _mm_set1_ps(1.0f); |
| const __m128 negone = _mm_set1_ps(-1.0f); |
| const __m128 mulby127 = _mm_set1_ps(127.0f); |
| __m128i *mmdst = (__m128i *)dst; |
| while (i >= 16) { /* 16 * float32 */ |
| const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src + 4)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src + 8)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src + 12)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| _mm_store_si128(mmdst, _mm_packs_epi16(_mm_packs_epi32(ints1, ints2), _mm_packs_epi32(ints3, ints4))); /* pack down, store out. */ |
| i -= 16; |
| src += 16; |
| mmdst++; |
| } |
| dst = (Sint8 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 127; |
| } else if (sample <= -1.0f) { |
| *dst = -128; |
| } else { |
| *dst = (Sint8)(sample * 127.0f); |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| cvt->len_cvt /= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S8); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_U8_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Uint8 *dst = cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U8 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 255; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint8)((sample + 1.0f) * 127.0f); |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128 one = _mm_set1_ps(1.0f); |
| const __m128 negone = _mm_set1_ps(-1.0f); |
| const __m128 mulby127 = _mm_set1_ps(127.0f); |
| __m128i *mmdst = (__m128i *)dst; |
| while (i >= 16) { /* 16 * float32 */ |
| const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src + 4)), one), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const __m128i ints3 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src + 8)), one), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const __m128i ints4 = _mm_cvtps_epi32(_mm_mul_ps(_mm_add_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src + 12)), one), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| _mm_store_si128(mmdst, _mm_packus_epi16(_mm_packs_epi32(ints1, ints2), _mm_packs_epi32(ints3, ints4))); /* pack down, store out. */ |
| i -= 16; |
| src += 16; |
| mmdst++; |
| } |
| dst = (Uint8 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 255; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint8)((sample + 1.0f) * 127.0f); |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| cvt->len_cvt /= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_U8); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint16 *dst = (Sint16 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S16 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 32767; |
| } else if (sample <= -1.0f) { |
| *dst = -32768; |
| } else { |
| *dst = (Sint16)(sample * 32767.0f); |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128 one = _mm_set1_ps(1.0f); |
| const __m128 negone = _mm_set1_ps(-1.0f); |
| const __m128 mulby32767 = _mm_set1_ps(32767.0f); |
| __m128i *mmdst = (__m128i *)dst; |
| while (i >= 8) { /* 8 * float32 */ |
| const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */ |
| const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src + 4)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */ |
| _mm_store_si128(mmdst, _mm_packs_epi32(ints1, ints2)); /* pack to sint16, store out. */ |
| i -= 8; |
| src += 8; |
| mmdst++; |
| } |
| dst = (Sint16 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 32767; |
| } else if (sample <= -1.0f) { |
| *dst = -32768; |
| } else { |
| *dst = (Sint16)(sample * 32767.0f); |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| cvt->len_cvt /= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S16SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_U16_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Uint16 *dst = (Uint16 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U16 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 65535; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint16)((sample + 1.0f) * 32767.0f); |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| /* This calculates differently than the scalar path because SSE2 can't |
| pack int32 data down to unsigned int16. _mm_packs_epi32 does signed |
| saturation, so that would corrupt our data. _mm_packus_epi32 exists, |
| but not before SSE 4.1. So we convert from float to sint16, packing |
| that down with legit signed saturation, and then xor the top bit |
| against 1. This results in the correct unsigned 16-bit value, even |
| though it looks like dark magic. */ |
| const __m128 mulby32767 = _mm_set1_ps(32767.0f); |
| const __m128i topbit = _mm_set1_epi16(-32768); |
| const __m128 one = _mm_set1_ps(1.0f); |
| const __m128 negone = _mm_set1_ps(-1.0f); |
| __m128i *mmdst = (__m128i *)dst; |
| while (i >= 8) { /* 8 * float32 */ |
| const __m128i ints1 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */ |
| const __m128i ints2 = _mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src + 4)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */ |
| _mm_store_si128(mmdst, _mm_xor_si128(_mm_packs_epi32(ints1, ints2), topbit)); /* pack to sint16, xor top bit, store out. */ |
| i -= 8; |
| src += 8; |
| mmdst++; |
| } |
| dst = (Uint16 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 65535; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint16)((sample + 1.0f) * 32767.0f); |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| cvt->len_cvt /= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_U16SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S32_SSE2(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint32 *dst = (Sint32 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S32 (using SSE2)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 2147483647; |
| } else if (sample <= -1.0f) { |
| *dst = (Sint32)-2147483648LL; |
| } else { |
| *dst = ((Sint32)(sample * 8388607.0f)) << 8; |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| SDL_assert(!i || !(((size_t)src) & 15)); |
| |
| { |
| /* Aligned! Do SSE blocks as long as we have 16 bytes available. */ |
| const __m128 one = _mm_set1_ps(1.0f); |
| const __m128 negone = _mm_set1_ps(-1.0f); |
| const __m128 mulby8388607 = _mm_set1_ps(8388607.0f); |
| __m128i *mmdst = (__m128i *)dst; |
| while (i >= 4) { /* 4 * float32 */ |
| _mm_store_si128(mmdst, _mm_slli_epi32(_mm_cvtps_epi32(_mm_mul_ps(_mm_min_ps(_mm_max_ps(negone, _mm_load_ps(src)), one), mulby8388607)), 8)); /* load 4 floats, clamp, convert to sint32 */ |
| i -= 4; |
| src += 4; |
| mmdst++; |
| } |
| dst = (Sint32 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 2147483647; |
| } else if (sample <= -1.0f) { |
| *dst = (Sint32)-2147483648LL; |
| } else { |
| *dst = ((Sint32)(sample * 8388607.0f)) << 8; |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S32SYS); |
| } |
| } |
| #endif |
| |
| #if HAVE_NEON_INTRINSICS |
| static void SDLCALL SDL_Convert_S8_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint8 *src = ((const Sint8 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 4)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S8", "AUDIO_F32 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */ |
| for (i = cvt->len_cvt; i && (((size_t)(dst - 15)) & 15); --i, --src, --dst) { |
| *dst = ((float)*src) * DIVBY128; |
| } |
| |
| src -= 15; |
| dst -= 15; /* adjust to read NEON blocks from the start. */ |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const int8_t *mmsrc = (const int8_t *)src; |
| const float32x4_t divby128 = vdupq_n_f32(DIVBY128); |
| while (i >= 16) { /* 16 * 8-bit */ |
| const int8x16_t bytes = vld1q_s8(mmsrc); /* get 16 sint8 into a NEON register. */ |
| const int16x8_t int16hi = vmovl_s8(vget_high_s8(bytes)); /* convert top 8 bytes to 8 int16 */ |
| const int16x8_t int16lo = vmovl_s8(vget_low_s8(bytes)); /* convert bottom 8 bytes to 8 int16 */ |
| /* split int16 to two int32, then convert to float, then multiply to normalize, store. */ |
| vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(int16lo))), divby128)); |
| vst1q_f32(dst + 4, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(int16lo))), divby128)); |
| vst1q_f32(dst + 8, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(int16hi))), divby128)); |
| vst1q_f32(dst + 12, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(int16hi))), divby128)); |
| i -= 16; |
| mmsrc -= 16; |
| dst -= 16; |
| } |
| |
| src = (const Sint8 *)mmsrc; |
| } |
| |
| src += 15; |
| dst += 15; /* adjust for any scalar finishing. */ |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = ((float)*src) * DIVBY128; |
| i--; |
| src--; |
| dst--; |
| } |
| |
| cvt->len_cvt *= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_U8_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Uint8 *src = ((const Uint8 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 4)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_U8", "AUDIO_F32 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */ |
| for (i = cvt->len_cvt; i && (((size_t)(dst - 15)) & 15); --i, --src, --dst) { |
| *dst = (((float)*src) * DIVBY128) - 1.0f; |
| } |
| |
| src -= 15; |
| dst -= 15; /* adjust to read NEON blocks from the start. */ |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const uint8_t *mmsrc = (const uint8_t *)src; |
| const float32x4_t divby128 = vdupq_n_f32(DIVBY128); |
| const float32x4_t negone = vdupq_n_f32(-1.0f); |
| while (i >= 16) { /* 16 * 8-bit */ |
| const uint8x16_t bytes = vld1q_u8(mmsrc); /* get 16 uint8 into a NEON register. */ |
| const uint16x8_t uint16hi = vmovl_u8(vget_high_u8(bytes)); /* convert top 8 bytes to 8 uint16 */ |
| const uint16x8_t uint16lo = vmovl_u8(vget_low_u8(bytes)); /* convert bottom 8 bytes to 8 uint16 */ |
| /* split uint16 to two uint32, then convert to float, then multiply to normalize, subtract to adjust for sign, store. */ |
| vst1q_f32(dst, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16lo))), divby128)); |
| vst1q_f32(dst + 4, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16lo))), divby128)); |
| vst1q_f32(dst + 8, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uint16hi))), divby128)); |
| vst1q_f32(dst + 12, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_high_u16(uint16hi))), divby128)); |
| i -= 16; |
| mmsrc -= 16; |
| dst -= 16; |
| } |
| |
| src = (const Uint8 *)mmsrc; |
| } |
| |
| src += 15; |
| dst += 15; /* adjust for any scalar finishing. */ |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = (((float)*src) * DIVBY128) - 1.0f; |
| i--; |
| src--; |
| dst--; |
| } |
| |
| cvt->len_cvt *= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_S16_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint16 *src = ((const Sint16 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 2)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S16", "AUDIO_F32 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */ |
| for (i = cvt->len_cvt / sizeof(Sint16); i && (((size_t)(dst - 7)) & 15); --i, --src, --dst) { |
| *dst = ((float)*src) * DIVBY32768; |
| } |
| |
| src -= 7; |
| dst -= 7; /* adjust to read NEON blocks from the start. */ |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const float32x4_t divby32768 = vdupq_n_f32(DIVBY32768); |
| while (i >= 8) { /* 8 * 16-bit */ |
| const int16x8_t ints = vld1q_s16((int16_t const *)src); /* get 8 sint16 into a NEON register. */ |
| /* split int16 to two int32, then convert to float, then multiply to normalize, store. */ |
| vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_low_s16(ints))), divby32768)); |
| vst1q_f32(dst + 4, vmulq_f32(vcvtq_f32_s32(vmovl_s16(vget_high_s16(ints))), divby32768)); |
| i -= 8; |
| src -= 8; |
| dst -= 8; |
| } |
| } |
| |
| src += 7; |
| dst += 7; /* adjust for any scalar finishing. */ |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = ((float)*src) * DIVBY32768; |
| i--; |
| src--; |
| dst--; |
| } |
| |
| cvt->len_cvt *= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_U16_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Uint16 *src = ((const Uint16 *)(cvt->buf + cvt->len_cvt)) - 1; |
| float *dst = ((float *)(cvt->buf + cvt->len_cvt * 2)) - 1; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_U16", "AUDIO_F32 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes (since buffer is growing, we don't have to worry about overreading from src) */ |
| for (i = cvt->len_cvt / sizeof(Sint16); i && (((size_t)(dst - 7)) & 15); --i, --src, --dst) { |
| *dst = (((float)*src) * DIVBY32768) - 1.0f; |
| } |
| |
| src -= 7; |
| dst -= 7; /* adjust to read NEON blocks from the start. */ |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const float32x4_t divby32768 = vdupq_n_f32(DIVBY32768); |
| const float32x4_t negone = vdupq_n_f32(-1.0f); |
| while (i >= 8) { /* 8 * 16-bit */ |
| const uint16x8_t uints = vld1q_u16((uint16_t const *)src); /* get 8 uint16 into a NEON register. */ |
| /* split uint16 to two int32, then convert to float, then multiply to normalize, subtract for sign, store. */ |
| vst1q_f32(dst, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_low_u16(uints))), divby32768)); |
| vst1q_f32(dst + 4, vmlaq_f32(negone, vcvtq_f32_u32(vmovl_u16(vget_high_u16(uints))), divby32768)); |
| i -= 8; |
| src -= 8; |
| dst -= 8; |
| } |
| } |
| |
| src += 7; |
| dst += 7; /* adjust for any scalar finishing. */ |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = (((float)*src) * DIVBY32768) - 1.0f; |
| i--; |
| src--; |
| dst--; |
| } |
| |
| cvt->len_cvt *= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_S32_to_F32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const Sint32 *src = (const Sint32 *)cvt->buf; |
| float *dst = (float *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_S32", "AUDIO_F32 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(Sint32); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| *dst = ((float)(*src >> 8)) * DIVBY8388607; |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const float32x4_t divby8388607 = vdupq_n_f32(DIVBY8388607); |
| const int32_t *mmsrc = (const int32_t *)src; |
| while (i >= 4) { /* 4 * sint32 */ |
| /* shift out lowest bits so int fits in a float32. Small precision loss, but much faster. */ |
| vst1q_f32(dst, vmulq_f32(vcvtq_f32_s32(vshrq_n_s32(vld1q_s32(mmsrc), 8)), divby8388607)); |
| i -= 4; |
| mmsrc += 4; |
| dst += 4; |
| } |
| src = (const Sint32 *)mmsrc; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| *dst = ((float)(*src >> 8)) * DIVBY8388607; |
| i--; |
| src++; |
| dst++; |
| } |
| |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S8_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint8 *dst = (Sint8 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S8 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 127; |
| } else if (sample <= -1.0f) { |
| *dst = -128; |
| } else { |
| *dst = (Sint8)(sample * 127.0f); |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const float32x4_t one = vdupq_n_f32(1.0f); |
| const float32x4_t negone = vdupq_n_f32(-1.0f); |
| const float32x4_t mulby127 = vdupq_n_f32(127.0f); |
| int8_t *mmdst = (int8_t *)dst; |
| while (i >= 16) { /* 16 * float32 */ |
| const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const int32x4_t ints3 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 8)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const int32x4_t ints4 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 12)), one), mulby127)); /* load 4 floats, clamp, convert to sint32 */ |
| const int8x8_t i8lo = vmovn_s16(vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2))); /* narrow to sint16, combine, narrow to sint8 */ |
| const int8x8_t i8hi = vmovn_s16(vcombine_s16(vmovn_s32(ints3), vmovn_s32(ints4))); /* narrow to sint16, combine, narrow to sint8 */ |
| vst1q_s8(mmdst, vcombine_s8(i8lo, i8hi)); /* combine to int8x16_t, store out */ |
| i -= 16; |
| src += 16; |
| mmdst += 16; |
| } |
| dst = (Sint8 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 127; |
| } else if (sample <= -1.0f) { |
| *dst = -128; |
| } else { |
| *dst = (Sint8)(sample * 127.0f); |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| cvt->len_cvt /= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S8); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_U8_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Uint8 *dst = cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U8 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 255; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint8)((sample + 1.0f) * 127.0f); |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const float32x4_t one = vdupq_n_f32(1.0f); |
| const float32x4_t negone = vdupq_n_f32(-1.0f); |
| const float32x4_t mulby127 = vdupq_n_f32(127.0f); |
| uint8_t *mmdst = (uint8_t *)dst; |
| while (i >= 16) { /* 16 * float32 */ |
| const uint32x4_t uints1 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */ |
| const uint32x4_t uints2 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */ |
| const uint32x4_t uints3 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 8)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */ |
| const uint32x4_t uints4 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 12)), one), one), mulby127)); /* load 4 floats, clamp, convert to uint32 */ |
| const uint8x8_t ui8lo = vmovn_u16(vcombine_u16(vmovn_u32(uints1), vmovn_u32(uints2))); /* narrow to uint16, combine, narrow to uint8 */ |
| const uint8x8_t ui8hi = vmovn_u16(vcombine_u16(vmovn_u32(uints3), vmovn_u32(uints4))); /* narrow to uint16, combine, narrow to uint8 */ |
| vst1q_u8(mmdst, vcombine_u8(ui8lo, ui8hi)); /* combine to uint8x16_t, store out */ |
| i -= 16; |
| src += 16; |
| mmdst += 16; |
| } |
| |
| dst = (Uint8 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 255; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint8)((sample + 1.0f) * 127.0f); |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| cvt->len_cvt /= 4; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_U8); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S16_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint16 *dst = (Sint16 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S16 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 32767; |
| } else if (sample <= -1.0f) { |
| *dst = -32768; |
| } else { |
| *dst = (Sint16)(sample * 32767.0f); |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const float32x4_t one = vdupq_n_f32(1.0f); |
| const float32x4_t negone = vdupq_n_f32(-1.0f); |
| const float32x4_t mulby32767 = vdupq_n_f32(32767.0f); |
| int16_t *mmdst = (int16_t *)dst; |
| while (i >= 8) { /* 8 * float32 */ |
| const int32x4_t ints1 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */ |
| const int32x4_t ints2 = vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), mulby32767)); /* load 4 floats, clamp, convert to sint32 */ |
| vst1q_s16(mmdst, vcombine_s16(vmovn_s32(ints1), vmovn_s32(ints2))); /* narrow to sint16, combine, store out. */ |
| i -= 8; |
| src += 8; |
| mmdst += 8; |
| } |
| dst = (Sint16 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 32767; |
| } else if (sample <= -1.0f) { |
| *dst = -32768; |
| } else { |
| *dst = (Sint16)(sample * 32767.0f); |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| cvt->len_cvt /= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S16SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_U16_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Uint16 *dst = (Uint16 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_U16 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 65535; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint16)((sample + 1.0f) * 32767.0f); |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| |
| /* Make sure src is aligned too. */ |
| if (!(((size_t)src) & 15)) { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const float32x4_t one = vdupq_n_f32(1.0f); |
| const float32x4_t negone = vdupq_n_f32(-1.0f); |
| const float32x4_t mulby32767 = vdupq_n_f32(32767.0f); |
| uint16_t *mmdst = (uint16_t *)dst; |
| while (i >= 8) { /* 8 * float32 */ |
| const uint32x4_t uints1 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), one), mulby32767)); /* load 4 floats, clamp, convert to uint32 */ |
| const uint32x4_t uints2 = vcvtq_u32_f32(vmulq_f32(vaddq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src + 4)), one), one), mulby32767)); /* load 4 floats, clamp, convert to uint32 */ |
| vst1q_u16(mmdst, vcombine_u16(vmovn_u32(uints1), vmovn_u32(uints2))); /* narrow to uint16, combine, store out. */ |
| i -= 8; |
| src += 8; |
| mmdst += 8; |
| } |
| dst = (Uint16 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 65535; |
| } else if (sample <= -1.0f) { |
| *dst = 0; |
| } else { |
| *dst = (Uint16)((sample + 1.0f) * 32767.0f); |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| cvt->len_cvt /= 2; |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_U16SYS); |
| } |
| } |
| |
| static void SDLCALL SDL_Convert_F32_to_S32_NEON(SDL_AudioCVT *cvt, SDL_AudioFormat format) |
| { |
| const float *src = (const float *)cvt->buf; |
| Sint32 *dst = (Sint32 *)cvt->buf; |
| int i; |
| |
| LOG_DEBUG_CONVERT("AUDIO_F32", "AUDIO_S32 (using NEON)"); |
| |
| /* Get dst aligned to 16 bytes */ |
| for (i = cvt->len_cvt / sizeof(float); i && (((size_t)dst) & 15); --i, ++src, ++dst) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 2147483647; |
| } else if (sample <= -1.0f) { |
| *dst = (-2147483647) - 1; |
| } else { |
| *dst = ((Sint32)(sample * 8388607.0f)) << 8; |
| } |
| } |
| |
| SDL_assert(!i || !(((size_t)dst) & 15)); |
| SDL_assert(!i || !(((size_t)src) & 15)); |
| |
| { |
| /* Aligned! Do NEON blocks as long as we have 16 bytes available. */ |
| const float32x4_t one = vdupq_n_f32(1.0f); |
| const float32x4_t negone = vdupq_n_f32(-1.0f); |
| const float32x4_t mulby8388607 = vdupq_n_f32(8388607.0f); |
| int32_t *mmdst = (int32_t *)dst; |
| while (i >= 4) { /* 4 * float32 */ |
| vst1q_s32(mmdst, vshlq_n_s32(vcvtq_s32_f32(vmulq_f32(vminq_f32(vmaxq_f32(negone, vld1q_f32(src)), one), mulby8388607)), 8)); |
| i -= 4; |
| src += 4; |
| mmdst += 4; |
| } |
| dst = (Sint32 *)mmdst; |
| } |
| |
| /* Finish off any leftovers with scalar operations. */ |
| while (i) { |
| const float sample = *src; |
| if (sample >= 1.0f) { |
| *dst = 2147483647; |
| } else if (sample <= -1.0f) { |
| *dst = (-2147483647) - 1; |
| } else { |
| *dst = ((Sint32)(sample * 8388607.0f)) << 8; |
| } |
| i--; |
| src++; |
| dst++; |
| } |
| |
| if (cvt->filters[++cvt->filter_index]) { |
| cvt->filters[cvt->filter_index](cvt, AUDIO_S32SYS); |
| } |
| } |
| #endif |
| |
| void SDL_ChooseAudioConverters(void) |
| { |
| static SDL_bool converters_chosen = SDL_FALSE; |
| |
| if (converters_chosen) { |
| return; |
| } |
| |
| #define SET_CONVERTER_FUNCS(fntype) \ |
| SDL_Convert_S8_to_F32 = SDL_Convert_S8_to_F32_##fntype; \ |
| SDL_Convert_U8_to_F32 = SDL_Convert_U8_to_F32_##fntype; \ |
| SDL_Convert_S16_to_F32 = SDL_Convert_S16_to_F32_##fntype; \ |
| SDL_Convert_U16_to_F32 = SDL_Convert_U16_to_F32_##fntype; \ |
| SDL_Convert_S32_to_F32 = SDL_Convert_S32_to_F32_##fntype; \ |
| SDL_Convert_F32_to_S8 = SDL_Convert_F32_to_S8_##fntype; \ |
| SDL_Convert_F32_to_U8 = SDL_Convert_F32_to_U8_##fntype; \ |
| SDL_Convert_F32_to_S16 = SDL_Convert_F32_to_S16_##fntype; \ |
| SDL_Convert_F32_to_U16 = SDL_Convert_F32_to_U16_##fntype; \ |
| SDL_Convert_F32_to_S32 = SDL_Convert_F32_to_S32_##fntype; \ |
| converters_chosen = SDL_TRUE |
| |
| #if HAVE_SSE2_INTRINSICS |
| if (SDL_HasSSE2()) { |
| SET_CONVERTER_FUNCS(SSE2); |
| return; |
| } |
| #endif |
| |
| #if HAVE_NEON_INTRINSICS |
| if (SDL_HasNEON()) { |
| SET_CONVERTER_FUNCS(NEON); |
| return; |
| } |
| #endif |
| |
| #if NEED_SCALAR_CONVERTER_FALLBACKS |
| SET_CONVERTER_FUNCS(Scalar); |
| #endif |
| |
| #undef SET_CONVERTER_FUNCS |
| |
| SDL_assert(converters_chosen == SDL_TRUE); |
| } |
| |
| /* vi: set ts=4 sw=4 expandtab: */ |
