mirror of
https://github.com/libretro/RetroArch.git
synced 2024-12-04 22:46:46 +00:00
290 lines
9.0 KiB
C
290 lines
9.0 KiB
C
/* RetroArch - A frontend for libretro.
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* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
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* Copyright (C) 2014 - Ali Bouhlel ( aliaspider@gmail.com )
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*
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* RetroArch is free software: you can redistribute it and/or modify it under the terms
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* of the GNU General Public License as published by the Free Software Found-
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* ation, either version 3 of the License, or (at your option) any later version.
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*
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* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
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* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
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* PURPOSE. See the GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along with RetroArch.
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* If not, see <http://www.gnu.org/licenses/>.
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*/
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#include "../boolean.h"
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#include "utils.h"
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#include "../general.h"
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#include "../performance.h"
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#if defined(__SSE2__)
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#include <emmintrin.h>
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#elif defined(__ALTIVEC__)
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#include <altivec.h>
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#endif
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void audio_convert_s16_to_float_C(float *out,
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const int16_t *in, size_t samples, float gain)
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{
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size_t i;
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gain = gain / 0x8000;
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for (i = 0; i < samples; i++)
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out[i] = (float)in[i] * gain;
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}
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void audio_convert_float_to_s16_C(int16_t *out,
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const float *in, size_t samples)
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{
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size_t i;
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for (i = 0; i < samples; i++)
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{
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int32_t val = (int32_t)(in[i] * 0x8000);
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out[i] = (val > 0x7FFF) ? 0x7FFF :
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(val < -0x8000 ? -0x8000 : (int16_t)val);
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}
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}
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#if defined(__SSE2__)
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void audio_convert_s16_to_float_SSE2(float *out,
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const int16_t *in, size_t samples, float gain)
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{
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float fgain = gain / UINT32_C(0x80000000);
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__m128 factor = _mm_set1_ps(fgain);
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size_t i;
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for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
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{
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__m128i input = _mm_loadu_si128((const __m128i *)in);
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__m128i regs[2] = {
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_mm_unpacklo_epi16(_mm_setzero_si128(), input),
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_mm_unpackhi_epi16(_mm_setzero_si128(), input),
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};
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__m128 output[2] = {
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_mm_mul_ps(_mm_cvtepi32_ps(regs[0]), factor),
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_mm_mul_ps(_mm_cvtepi32_ps(regs[1]), factor),
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};
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_mm_storeu_ps(out + 0, output[0]);
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_mm_storeu_ps(out + 4, output[1]);
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}
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audio_convert_s16_to_float_C(out, in, samples - i, gain);
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}
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void audio_convert_float_to_s16_SSE2(int16_t *out,
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const float *in, size_t samples)
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{
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__m128 factor = _mm_set1_ps((float)0x8000);
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size_t i;
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for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
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{
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__m128 input[2] = { _mm_loadu_ps(in + 0), _mm_loadu_ps(in + 4) };
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__m128 res[2] = { _mm_mul_ps(input[0], factor),
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_mm_mul_ps(input[1], factor) };
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__m128i ints[2] = { _mm_cvtps_epi32(res[0]), _mm_cvtps_epi32(res[1]) };
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__m128i packed = _mm_packs_epi32(ints[0], ints[1]);
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_mm_storeu_si128((__m128i *)out, packed);
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}
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audio_convert_float_to_s16_C(out, in, samples - i);
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}
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#elif defined(__ALTIVEC__)
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void audio_convert_s16_to_float_altivec(float *out,
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const int16_t *in, size_t samples, float gain)
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{
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size_t samples_in = samples;
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const vector float gain_vec = { gain, gain , gain, gain };
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const vector float zero_vec = { 0.0f, 0.0f, 0.0f, 0.0f};
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/* Unaligned loads/store is a bit expensive, so we
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* optimize for the good path (very likely). */
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if (((uintptr_t)out & 15) + ((uintptr_t)in & 15) == 0)
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{
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size_t i;
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for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
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{
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vector signed short input = vec_ld(0, in);
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vector signed int hi = vec_unpackh(input);
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vector signed int lo = vec_unpackl(input);
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vector float out_hi = vec_madd(vec_ctf(hi, 15), gain_vec, zero_vec);
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vector float out_lo = vec_madd(vec_ctf(lo, 15), gain_vec, zero_vec);
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vec_st(out_hi, 0, out);
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vec_st(out_lo, 16, out);
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}
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samples_in -= i;
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}
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audio_convert_s16_to_float_C(out, in, samples_in, gain);
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}
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void audio_convert_float_to_s16_altivec(int16_t *out,
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const float *in, size_t samples)
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{
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int samples_in = samples;
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/* Unaligned loads/store is a bit expensive,
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* so we optimize for the good path (very likely). */
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if (((uintptr_t)out & 15) + ((uintptr_t)in & 15) == 0)
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{
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size_t i;
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for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
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{
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vector float input0 = vec_ld( 0, in);
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vector float input1 = vec_ld(16, in);
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vector signed int result0 = vec_cts(input0, 15);
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vector signed int result1 = vec_cts(input1, 15);
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vec_st(vec_packs(result0, result1), 0, out);
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}
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samples_in -= i;
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}
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audio_convert_float_to_s16_C(out, in, samples_in);
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}
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#elif defined(__ARM_NEON__)
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/* Avoid potential hard-float/soft-float ABI issues. */
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void audio_convert_s16_float_asm(float *out, const int16_t *in,
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size_t samples, const float *gain);
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static void audio_convert_s16_to_float_neon(float *out,
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const int16_t *in, size_t samples, float gain)
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{
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size_t aligned_samples = samples & ~7;
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if (aligned_samples)
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audio_convert_s16_float_asm(out, in, aligned_samples, &gain);
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/* Could do all conversion in ASM, but keep it simple for now. */
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audio_convert_s16_to_float_C(out + aligned_samples, in + aligned_samples,
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samples - aligned_samples, gain);
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}
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void audio_convert_float_s16_asm(int16_t *out, const float *in, size_t samples);
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static void audio_convert_float_to_s16_neon(int16_t *out,
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const float *in, size_t samples)
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{
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size_t aligned_samples = samples & ~7;
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if (aligned_samples)
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audio_convert_float_s16_asm(out, in, aligned_samples);
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audio_convert_float_to_s16_C(out + aligned_samples, in + aligned_samples,
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samples - aligned_samples);
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}
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#elif defined(_MIPS_ARCH_ALLEGREX)
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void audio_convert_s16_to_float_ALLEGREX(float *out,
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const int16_t *in, size_t samples, float gain)
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{
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#ifdef DEBUG
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/* Make sure the buffer is 16 byte aligned, this should be the
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* default behaviour of malloc in the PSPSDK.
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* Only the output buffer can be assumed to be 16-byte aligned. */
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rarch_assert(((uintptr_t)out & 0xf) == 0);
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#endif
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size_t i;
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gain = gain / 0x8000;
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__asm__ (
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".set push \n"
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".set noreorder \n"
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"mtv %0, s200 \n"
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".set pop \n"
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::"r"(gain));
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for (i = 0; i + 16 <= samples; i += 16)
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{
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__asm__ (
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".set push \n"
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".set noreorder \n"
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"lv.s s100, 0(%0) \n"
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"lv.s s101, 4(%0) \n"
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"lv.s s110, 8(%0) \n"
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"lv.s s111, 12(%0) \n"
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"lv.s s120, 16(%0) \n"
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"lv.s s121, 20(%0) \n"
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"lv.s s130, 24(%0) \n"
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"lv.s s131, 28(%0) \n"
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"vs2i.p c100, c100 \n"
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"vs2i.p c110, c110 \n"
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"vs2i.p c120, c120 \n"
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"vs2i.p c130, c130 \n"
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"vi2f.q c100, c100, 16 \n"
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"vi2f.q c110, c110, 16 \n"
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"vi2f.q c120, c120, 16 \n"
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"vi2f.q c130, c130, 16 \n"
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"vmscl.q e100, e100, s200 \n"
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"sv.q c100, 0(%1) \n"
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"sv.q c110, 16(%1) \n"
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"sv.q c120, 32(%1) \n"
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"sv.q c130, 48(%1) \n"
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".set pop \n"
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:: "r"(in + i), "r"(out + i));
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}
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for (; i < samples; i++)
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out[i] = (float)in[i] * gain;
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}
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void audio_convert_float_to_s16_ALLEGREX(int16_t *out,
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const float *in, size_t samples)
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{
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#ifdef DEBUG
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/* Make sure the buffers are 16 byte aligned, this should be
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* the default behaviour of malloc in the PSPSDK.
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* Both buffers are allocated by RetroArch, so can assume alignment. */
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rarch_assert(((uintptr_t)in & 0xf) == 0);
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rarch_assert(((uintptr_t)out & 0xf) == 0);
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#endif
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size_t i;
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for (i = 0; i + 8 <= samples; i += 8)
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{
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__asm__ (
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".set push \n"
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".set noreorder \n"
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"lv.q c100, 0(%0) \n"
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"lv.q c110, 16(%0) \n"
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"vf2in.q c100, c100, 31 \n"
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"vf2in.q c110, c110, 31 \n"
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"vi2s.q c100, c100 \n"
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"vi2s.q c102, c110 \n"
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"sv.q c100, 0(%1) \n"
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".set pop \n"
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:: "r"(in + i), "r"(out + i));
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}
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for (; i < samples; i++)
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{
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int32_t val = (int32_t)(in[i] * 0x8000);
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out[i] = (val > 0x7FFF) ? 0x7FFF :
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(val < -0x8000 ? -0x8000 : (int16_t)val);
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}
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}
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#endif
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void audio_convert_init_simd(void)
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{
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#if defined(__ARM_NEON__)
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unsigned cpu = rarch_get_cpu_features();
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audio_convert_s16_to_float_arm = cpu & RETRO_SIMD_NEON ?
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audio_convert_s16_to_float_neon : audio_convert_s16_to_float_C;
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audio_convert_float_to_s16_arm = cpu & RETRO_SIMD_NEON ?
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audio_convert_float_to_s16_neon : audio_convert_float_to_s16_C;
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#endif
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}
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