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https://github.com/CTCaer/RetroArch.git
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add a faster (approximated) version of the CC resampler using SSE
intrinsics
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aliaspider
parent
82806dcd85
commit
d1a785823d
@ -28,13 +28,8 @@
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#define RARCH_LOG(...) fprintf(stderr, __VA_ARGS__)
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#endif
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typedef struct audio_frame_int16
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{
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int16_t l;
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int16_t r;
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} audio_frame_int16_t;
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#ifdef _MIPS_ARCH_ALLEGREX1
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#ifdef _MIPS_ARCH_ALLEGREX
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static void resampler_CC_process(void *re_, struct resampler_data *data)
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{
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(void)re_;
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@ -121,7 +116,7 @@ static void resampler_CC_process(void *re_, struct resampler_data *data)
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outp++;
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}
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/* The VFPU state is assumed to remain intact
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/* The VFPU state is assumed to remain intact
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* in-between calls to resampler_CC_process. */
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done:
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@ -151,6 +146,251 @@ static void *resampler_CC_init(double bandwidth_mod)
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RARCH_LOG("\nConvoluted Cosine resampler (VFPU): \n");
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return (void*)-1;
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}
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#elif defined(__SSE__)
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/* uses a fast polynomial approximation
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* since SSE lacks native support for trigonometric functions
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* cc_int is approximated with P(X) = X - (3/4)*X^3 + (1/4)*X^5
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*/
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#include <xmmintrin.h>
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#ifndef CC_RESAMPLER_PRECISION
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#define CC_RESAMPLER_PRECISION 1
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#endif
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typedef struct rarch_CC_resampler
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{
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__m128 previous;
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__m128 current;
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float distance;
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void (*process)(void *re, struct resampler_data *data);
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} rarch_CC_resampler_t;
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static void resampler_CC_downsample(void *re_, struct resampler_data *data)
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{
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float ratio, b;
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
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audio_frame_float_t *inp_max = (audio_frame_float_t*)(inp + data->input_frames);
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audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
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ratio = 1.0 / data->ratio;
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b = data->ratio; /* cutoff frequency. */
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__m128 vec_previous = _mm_loadu_ps((float*)&re->previous);
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__m128 vec_current = _mm_loadu_ps((float*)&re->current);
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while (inp != inp_max)
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{
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__m128 vec_ratio =
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_mm_mul_ps(_mm_set_ps1(ratio), _mm_set_ps(3.0, 2.0, 1.0, 0.0));
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__m128 vec_w = _mm_sub_ps(_mm_set_ps1(re->distance), vec_ratio);
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__m128 vec_w1 = _mm_add_ps(vec_w , _mm_set_ps1(0.5));
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__m128 vec_w2 = _mm_sub_ps(vec_w , _mm_set_ps1(0.5));
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__m128 vec_b = _mm_set_ps1(b);
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vec_w1 = _mm_mul_ps(vec_w1, vec_b);
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vec_w2 = _mm_mul_ps(vec_w2, vec_b);
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#if (CC_RESAMPLER_PRECISION > 0)
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__m128 vec_ww1 = _mm_mul_ps(vec_w1, vec_w1);
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__m128 vec_ww2 = _mm_mul_ps(vec_w2, vec_w2);
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vec_ww1 = _mm_mul_ps(vec_ww1, _mm_sub_ps(_mm_set_ps1(3.0),vec_ww1));
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vec_ww2 = _mm_mul_ps(vec_ww2, _mm_sub_ps(_mm_set_ps1(3.0),vec_ww2));
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vec_ww1 = _mm_mul_ps(_mm_set_ps1(1.0/4.0), vec_ww1);
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vec_ww2 = _mm_mul_ps(_mm_set_ps1(1.0/4.0), vec_ww2);
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vec_w1 = _mm_mul_ps(vec_w1, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww1));
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vec_w2 = _mm_mul_ps(vec_w2, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww2));
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#endif
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vec_w1 = _mm_min_ps(vec_w1, _mm_set_ps1( 0.5));
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vec_w2 = _mm_min_ps(vec_w2, _mm_set_ps1( 0.5));
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vec_w1 = _mm_max_ps(vec_w1, _mm_set_ps1(-0.5));
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vec_w2 = _mm_max_ps(vec_w2, _mm_set_ps1(-0.5));
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vec_w = _mm_sub_ps(vec_w1, vec_w2);
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__m128 vec_w_previous =
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_mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(1, 1, 0, 0));
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__m128 vec_w_current =
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_mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(3, 3, 2, 2));
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__m128 vec_in = _mm_loadl_pi(_mm_setzero_ps(),(__m64*)inp);
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vec_in = _mm_shuffle_ps(vec_in,vec_in,_MM_SHUFFLE(1, 0, 1, 0));
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vec_previous =
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_mm_add_ps(vec_previous, _mm_mul_ps(vec_in, vec_w_previous));
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vec_current =
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_mm_add_ps(vec_current, _mm_mul_ps(vec_in, vec_w_current));
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re->distance++;
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inp++;
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if (re->distance > (ratio + 0.5))
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{
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_mm_storel_pi((__m64*)outp, vec_previous);
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vec_previous =
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_mm_shuffle_ps(vec_previous,vec_current,_MM_SHUFFLE(1, 0, 3, 2));
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vec_current =
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_mm_shuffle_ps(vec_current,_mm_setzero_ps(),_MM_SHUFFLE(1, 0, 3, 2));
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re->distance -= ratio;
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outp++;
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}
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}
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_mm_storeu_ps((float*)&re->previous, vec_previous);
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_mm_storeu_ps((float*)&re->current, vec_current);
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data->output_frames = outp - (audio_frame_float_t*)data->data_out;
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}
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#ifndef min
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#define min(a, b) ((a) < (b) ? (a) : (b))
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#endif
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static void resampler_CC_upsample(void *re_, struct resampler_data *data)
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{
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float b, ratio;
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
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audio_frame_float_t *inp_max = (audio_frame_float_t*)(inp + data->input_frames);
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audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
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b = min(data->ratio, 1.00); /* cutoff frequency. */
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ratio = 1.0 / data->ratio;
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__m128 vec_previous = _mm_loadu_ps((float*)&re->previous);
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__m128 vec_current = _mm_loadu_ps((float*)&re->current);
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while (inp != inp_max)
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{
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__m128 vec_in = _mm_loadl_pi(_mm_setzero_ps(),(__m64*)inp);
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vec_previous =
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_mm_shuffle_ps(vec_previous,vec_current,_MM_SHUFFLE(1, 0, 3, 2));
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vec_current =
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_mm_shuffle_ps(vec_current,vec_in,_MM_SHUFFLE(1, 0, 3, 2));
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while (re->distance < 1.0)
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{
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__m128 vec_w =
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_mm_add_ps(_mm_set_ps1(re->distance), _mm_set_ps(-2.0, -1.0, 0.0, 1.0));
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__m128 vec_w1 = _mm_add_ps(vec_w , _mm_set_ps1(0.5));
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__m128 vec_w2 = _mm_sub_ps(vec_w , _mm_set_ps1(0.5));
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__m128 vec_b = _mm_set_ps1(b);
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vec_w1 = _mm_mul_ps(vec_w1, vec_b);
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vec_w2 = _mm_mul_ps(vec_w2, vec_b);
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#if (CC_RESAMPLER_PRECISION > 0)
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__m128 vec_ww1 = _mm_mul_ps(vec_w1, vec_w1);
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__m128 vec_ww2 = _mm_mul_ps(vec_w2, vec_w2);
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vec_ww1 = _mm_mul_ps(vec_ww1,_mm_sub_ps(_mm_set_ps1(3.0),vec_ww1));
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vec_ww2 = _mm_mul_ps(vec_ww2,_mm_sub_ps(_mm_set_ps1(3.0),vec_ww2));
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vec_ww1 = _mm_mul_ps(_mm_set_ps1(1.0 / 4.0), vec_ww1);
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vec_ww2 = _mm_mul_ps(_mm_set_ps1(1.0 / 4.0), vec_ww2);
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vec_w1 = _mm_mul_ps(vec_w1, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww1));
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vec_w2 = _mm_mul_ps(vec_w2, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww2));
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#endif
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vec_w1 = _mm_min_ps(vec_w1, _mm_set_ps1( 0.5));
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vec_w2 = _mm_min_ps(vec_w2, _mm_set_ps1( 0.5));
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vec_w1 = _mm_max_ps(vec_w1, _mm_set_ps1(-0.5));
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vec_w2 = _mm_max_ps(vec_w2, _mm_set_ps1(-0.5));
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vec_w = _mm_sub_ps(vec_w1, vec_w2);
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__m128 vec_w_previous =
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_mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(1, 1, 0, 0));
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__m128 vec_w_current =
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_mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(3, 3, 2, 2));
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__m128 vec_out = _mm_mul_ps(vec_previous, vec_w_previous);
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vec_out = _mm_add_ps(vec_out, _mm_mul_ps(vec_current, vec_w_current));
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vec_out =
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_mm_add_ps(vec_out, _mm_shuffle_ps(vec_out,vec_out,_MM_SHUFFLE(3, 2, 3, 2)));
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_mm_storel_pi((__m64*)outp,vec_out);
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re->distance += ratio;
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outp++;
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}
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re->distance -= 1.0;
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inp++;
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}
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_mm_storeu_ps((float*)&re->previous, vec_previous);
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_mm_storeu_ps((float*)&re->current, vec_current);
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data->output_frames = outp - (audio_frame_float_t*)data->data_out;
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}
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static void resampler_CC_process(void *re_, struct resampler_data *data)
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{
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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re->process(re_, data);
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}
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static void resampler_CC_free(void *re_)
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{
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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if (re)
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free(re);
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}
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static void *resampler_CC_init(double bandwidth_mod)
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{
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int i;
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)
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calloc(1, sizeof(rarch_CC_resampler_t));
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if (!re)
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return NULL;
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for (i = 0; i < 4; i++)
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{
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re->previous = _mm_setzero_ps();
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re->current = _mm_setzero_ps();
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}
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RARCH_LOG("Convoluted Cosine resampler (SSE) : ");
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/* variations of data->ratio around 0.75 are safer
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* than around 1.0 for both up/downsampler. */
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if (bandwidth_mod < 0.75)
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{
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RARCH_LOG("CC_downsample @%f \n", bandwidth_mod);
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re->process = resampler_CC_downsample;
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re->distance = 0.0;
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}
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else
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{
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RARCH_LOG("CC_upsample @%f \n", bandwidth_mod);
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re->process = resampler_CC_upsample;
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re->distance = 2.0;
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}
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return re;
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}
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#else
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/* C reference version. Not optimized. */
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@ -295,9 +535,9 @@ static void *resampler_CC_init(double bandwidth_mod)
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RARCH_LOG("Convoluted Cosine resampler (C) : ");
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/* variations of data->ratio around 0.75 are safer
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/* variations of data->ratio around 0.75 are safer
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* than around 1.0 for both up/downsampler. */
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if (bandwidth_mod < 0.75)
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if (bandwidth_mod < 0.75)
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{
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RARCH_LOG("CC_downsample @%f \n", bandwidth_mod);
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re->process = resampler_CC_downsample;
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