RetroArch/audio/utils.c
2012-12-05 23:17:07 +01:00

161 lines
5.3 KiB
C

/* RetroArch - A frontend for libretro.
* Copyright (C) 2010-2012 - Hans-Kristian Arntzen
*
* RetroArch is free software: you can redistribute it and/or modify it under the terms
* of the GNU General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
* PURPOSE. See the GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along with RetroArch.
* If not, see <http://www.gnu.org/licenses/>.
*/
#include "utils.h"
#if defined(__SSE2__)
#include <emmintrin.h>
#elif defined(__ALTIVEC__)
#include <altivec.h>
#endif
void audio_convert_s16_to_float_C(float *out,
const int16_t *in, size_t samples, float gain)
{
gain = gain / 0x8000;
for (size_t i = 0; i < samples; i++)
out[i] = (float)in[i] * gain;
}
void audio_convert_float_to_s16_C(int16_t *out,
const float *in, size_t samples)
{
for (size_t i = 0; i < samples; i++)
{
int32_t val = (int32_t)(in[i] * 0x8000);
out[i] = (val > 0x7FFF) ? 0x7FFF : (val < -0x8000 ? -0x8000 : (int16_t)val);
}
}
#if defined(__SSE2__)
void audio_convert_s16_to_float_SSE2(float *out,
const int16_t *in, size_t samples, float gain)
{
float fgain = gain / (0x7fff * 0x10000);
__m128 factor = _mm_set1_ps(fgain);
size_t i;
for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
{
__m128i input = _mm_loadu_si128((const __m128i *)in);
__m128i regs[2] = {
_mm_unpacklo_epi16(_mm_setzero_si128(), input),
_mm_unpackhi_epi16(_mm_setzero_si128(), input),
};
__m128 output[2] = {
_mm_mul_ps(_mm_cvtepi32_ps(regs[0]), factor),
_mm_mul_ps(_mm_cvtepi32_ps(regs[1]), factor),
};
_mm_storeu_ps(out + 0, output[0]);
_mm_storeu_ps(out + 4, output[1]);
}
audio_convert_s16_to_float_C(out, in, samples - i, gain);
}
void audio_convert_float_to_s16_SSE2(int16_t *out,
const float *in, size_t samples)
{
__m128 factor = _mm_set1_ps((float)0x7fff);
size_t i;
for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
{
__m128 input[2] = { _mm_loadu_ps(in + 0), _mm_loadu_ps(in + 4) };
__m128 res[2] = { _mm_mul_ps(input[0], factor), _mm_mul_ps(input[1], factor) };
__m128i ints[2] = { _mm_cvtps_epi32(res[0]), _mm_cvtps_epi32(res[1]) };
__m128i packed = _mm_packs_epi32(ints[0], ints[1]);
_mm_storeu_si128((__m128i *)out, packed);
}
audio_convert_float_to_s16_C(out, in, samples - i);
}
#elif defined(__ALTIVEC__)
void audio_convert_s16_to_float_altivec(float *out,
const int16_t *in, size_t samples, float gain)
{
const vector float gain_vec = vec_splats(gain);
const vector float zero_vec = vec_splats(0.0f);
// Unaligned loads/store is a bit expensive, so we optimize for the good path (very likely).
if (((uintptr_t)out & 15) + ((uintptr_t)in & 15) == 0)
{
size_t i;
for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
{
vector signed short input = vec_ld(0, in);
vector signed int hi = vec_unpackh(input);
vector signed int lo = vec_unpackl(input);
vector float out_hi = vec_madd(vec_ctf(hi, 15), gain_vec, zero_vec);
vector float out_lo = vec_madd(vec_ctf(lo, 15), gain_vec, zero_vec);
vec_st(out_hi, 0, out);
vec_st(out_lo, 16, out);
}
audio_convert_s16_to_float_C(out, in, samples - i, gain);
}
else
audio_convert_s16_to_float_C(out, in, samples, gain);
}
void audio_convert_float_to_s16_altivec(int16_t *out,
const float *in, size_t samples)
{
// Unaligned loads/store is a bit expensive, so we optimize for the good path (very likely).
if (((uintptr_t)out & 15) + ((uintptr_t)in & 15) == 0)
{
size_t i;
for (i = 0; i + 8 <= samples; i += 8, in += 8, out += 8)
{
vector float input0 = vec_ld( 0, in);
vector float input1 = vec_ld(16, in);
vector signed int result0 = vec_cts(input0, 15);
vector signed int result1 = vec_cts(input1, 15);
vec_st(vec_packs(result0, result1), 0, out);
}
audio_convert_float_to_s16_C(out, in, samples - i);
}
else
audio_convert_float_to_s16_C(out, in, samples);
}
#elif defined(HAVE_NEON)
void audio_convert_s16_float_asm(float *out, const int16_t *in, size_t samples);
void audio_convert_s16_to_float_neon(float *out, const int16_t *in, size_t samples,
float gain)
{
(void)gain; // gain is ignored for now.
size_t aligned_samples = samples & ~7;
audio_convert_s16_float_asm(out, in, aligned_samples);
// Could do all conversion in ASM, but keep it simple for now.
audio_convert_s16_to_float_C(out + aligned_samples, in + aligned_samples,
samples - aligned_samples, 1.0f);
}
void audio_convert_float_s16_asm(int16_t *out, const float *in, size_t samples);
void audio_convert_float_to_s16_neon(int16_t *out, const float *in, size_t samples)
{
size_t aligned_samples = samples & ~7;
audio_convert_float_s16_asm(out, in, aligned_samples);
audio_convert_float_to_s16_C(out + aligned_samples, in + aligned_samples,
samples - aligned_samples);
}
#endif