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* DSP: Implement Pipe 2 Pipe 2 is a DSP pipe that is used to initialize both the DSP hardware (the application signals to the DSP to initialize) and the application (the DSP provides the memory location of structures in the shared memory region). * AudioCore: Implement codecs (DecodeADPCM, DecodePCM8, DecodePCM16) * DSP Pipes: Implement as FIFO * AudioCore: File structure * AudioCore: More structure * AudioCore: Buffer management * DSP/Source: Reorganise Source's AdvanceFrame. * Audio Output * lolidk * huh? * interp * More interp stuff * oops * Zero State * Don't mix Source frame if it's not enabled * DSP: Forgot to zero a buffer, adjusted thread synchronisation, adjusted format spec for buffers * asdf * Get it to compile and tweak stretching a bit. * revert stretch test * deleted accidental partial catch submodule commit * new audio stretching algorithm * update .gitmodule * fix OS X build * remove getopt from rubberband * #include <stddef> to audio_core.h * typo * -framework Accelerate * OptionTransientsSmooth -> OptionTransientsCrisp * tweak stretch tempo smoothing coefficient. also switch back to smooth. * tweak mroe * remove printf * sola * #include <cmath> * VERY QUICK MERGE TO GET IT WORKING DOESN'T ACTIVATE AUDIO FILTERS * Reminder to self * fix comparison * common/thread: Correct code style * Thread: Make Barrier reusable * fix threading synchonisation code * add profiling code * print error to console when audio clips * fix metallic sound * reduce logspam
373 lines
13 KiB
C++
373 lines
13 KiB
C++
////////////////////////////////////////////////////////////////////////////////
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///
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/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE
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/// optimized functions have been gathered into this single source
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/// code file, regardless to their class or original source code file, in order
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/// to ease porting the library to other compiler and processor platforms.
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///
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/// The SSE-optimizations are programmed using SSE compiler intrinsics that
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/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
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/// should compile with both toolsets.
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///
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/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
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/// 6.0 processor pack" update to support SSE instruction set. The update is
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/// available for download at Microsoft Developers Network, see here:
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/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
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///
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/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
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/// perform a search with keywords "processor pack".
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///
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/// Author : Copyright (c) Olli Parviainen
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/// Author e-mail : oparviai 'at' iki.fi
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/// SoundTouch WWW: http://www.surina.net/soundtouch
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///
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////////////////////////////////////////////////////////////////////////////////
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//
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// Last changed : $Date: 2015-08-09 00:00:15 +0300 (Sun, 09 Aug 2015) $
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// File revision : $Revision: 4 $
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//
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// $Id: sse_optimized.cpp 226 2015-08-08 21:00:15Z oparviai $
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//
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////////////////////////////////////////////////////////////////////////////////
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//
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// License :
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//
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// SoundTouch audio processing library
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// Copyright (c) Olli Parviainen
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//
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// This library is free software; you can redistribute it and/or
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// modify it under the terms of the GNU Lesser General Public
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// License as published by the Free Software Foundation; either
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// version 2.1 of the License, or (at your option) any later version.
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//
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// This library is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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// Lesser General Public License for more details.
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//
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// You should have received a copy of the GNU Lesser General Public
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// License along with this library; if not, write to the Free Software
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// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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//
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////////////////////////////////////////////////////////////////////////////////
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#include "cpu_detect.h"
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#include "STTypes.h"
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using namespace soundtouch;
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#ifdef SOUNDTOUCH_ALLOW_SSE
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// SSE routines available only with float sample type
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//////////////////////////////////////////////////////////////////////////////
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//
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// implementation of SSE optimized functions of class 'TDStretchSSE'
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//
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//////////////////////////////////////////////////////////////////////////////
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#include "TDStretch.h"
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#include <xmmintrin.h>
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#include <math.h>
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// Calculates cross correlation of two buffers
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double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm)
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{
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int i;
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const float *pVec1;
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const __m128 *pVec2;
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__m128 vSum, vNorm;
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// Note. It means a major slow-down if the routine needs to tolerate
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// unaligned __m128 memory accesses. It's way faster if we can skip
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// unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.
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// This can mean up to ~ 10-fold difference (incl. part of which is
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// due to skipping every second round for stereo sound though).
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//
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// Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
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// for choosing if this little cheating is allowed.
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#ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
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// Little cheating allowed, return valid correlation only for
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// aligned locations, meaning every second round for stereo sound.
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#define _MM_LOAD _mm_load_ps
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if (((ulongptr)pV1) & 15) return -1e50; // skip unaligned locations
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#else
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// No cheating allowed, use unaligned load & take the resulting
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// performance hit.
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#define _MM_LOAD _mm_loadu_ps
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#endif
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// ensure overlapLength is divisible by 8
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assert((overlapLength % 8) == 0);
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// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
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// Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
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pVec1 = (const float*)pV1;
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pVec2 = (const __m128*)pV2;
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vSum = vNorm = _mm_setzero_ps();
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// Unroll the loop by factor of 4 * 4 operations. Use same routine for
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// stereo & mono, for mono it just means twice the amount of unrolling.
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for (i = 0; i < channels * overlapLength / 16; i ++)
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{
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__m128 vTemp;
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// vSum += pV1[0..3] * pV2[0..3]
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vTemp = _MM_LOAD(pVec1);
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vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp ,pVec2[0]));
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vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
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// vSum += pV1[4..7] * pV2[4..7]
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vTemp = _MM_LOAD(pVec1 + 4);
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vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[1]));
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vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
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// vSum += pV1[8..11] * pV2[8..11]
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vTemp = _MM_LOAD(pVec1 + 8);
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vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[2]));
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vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
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// vSum += pV1[12..15] * pV2[12..15]
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vTemp = _MM_LOAD(pVec1 + 12);
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vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[3]));
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vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
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pVec1 += 16;
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pVec2 += 4;
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}
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// return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
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float *pvNorm = (float*)&vNorm;
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float norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
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anorm = norm;
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float *pvSum = (float*)&vSum;
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return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / sqrt(norm < 1e-9 ? 1.0 : norm);
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/* This is approximately corresponding routine in C-language yet without normalization:
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double corr, norm;
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uint i;
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// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
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corr = norm = 0.0;
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for (i = 0; i < channels * overlapLength / 16; i ++)
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{
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corr += pV1[0] * pV2[0] +
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pV1[1] * pV2[1] +
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pV1[2] * pV2[2] +
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pV1[3] * pV2[3] +
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pV1[4] * pV2[4] +
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pV1[5] * pV2[5] +
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pV1[6] * pV2[6] +
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pV1[7] * pV2[7] +
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pV1[8] * pV2[8] +
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pV1[9] * pV2[9] +
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pV1[10] * pV2[10] +
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pV1[11] * pV2[11] +
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pV1[12] * pV2[12] +
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pV1[13] * pV2[13] +
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pV1[14] * pV2[14] +
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pV1[15] * pV2[15];
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for (j = 0; j < 15; j ++) norm += pV1[j] * pV1[j];
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pV1 += 16;
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pV2 += 16;
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}
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return corr / sqrt(norm);
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*/
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}
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double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm)
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{
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// call usual calcCrossCorr function because SSE does not show big benefit of
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// accumulating "norm" value, and also the "norm" rolling algorithm would get
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// complicated due to SSE-specific alignment-vs-nonexact correlation rules.
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return calcCrossCorr(pV1, pV2, norm);
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}
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//////////////////////////////////////////////////////////////////////////////
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//
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// implementation of SSE optimized functions of class 'FIRFilter'
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//
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//////////////////////////////////////////////////////////////////////////////
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#include "FIRFilter.h"
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FIRFilterSSE::FIRFilterSSE() : FIRFilter()
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{
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filterCoeffsAlign = NULL;
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filterCoeffsUnalign = NULL;
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}
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FIRFilterSSE::~FIRFilterSSE()
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{
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delete[] filterCoeffsUnalign;
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filterCoeffsAlign = NULL;
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filterCoeffsUnalign = NULL;
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}
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// (overloaded) Calculates filter coefficients for SSE routine
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void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
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{
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uint i;
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float fDivider;
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FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
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// Scale the filter coefficients so that it won't be necessary to scale the filtering result
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// also rearrange coefficients suitably for SSE
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// Ensure that filter coeffs array is aligned to 16-byte boundary
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delete[] filterCoeffsUnalign;
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filterCoeffsUnalign = new float[2 * newLength + 4];
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filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
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fDivider = (float)resultDivider;
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// rearrange the filter coefficients for mmx routines
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for (i = 0; i < newLength; i ++)
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{
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filterCoeffsAlign[2 * i + 0] =
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filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
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}
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}
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// SSE-optimized version of the filter routine for stereo sound
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uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
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{
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int count = (int)((numSamples - length) & (uint)-2);
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int j;
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assert(count % 2 == 0);
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if (count < 2) return 0;
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assert(source != NULL);
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assert(dest != NULL);
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assert((length % 8) == 0);
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assert(filterCoeffsAlign != NULL);
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assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
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// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
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#pragma omp parallel for
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for (j = 0; j < count; j += 2)
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{
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const float *pSrc;
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float *pDest;
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const __m128 *pFil;
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__m128 sum1, sum2;
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uint i;
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pSrc = (const float*)source + j * 2; // source audio data
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pDest = dest + j * 2; // destination audio data
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pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
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// are aligned to 16-byte boundary
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sum1 = sum2 = _mm_setzero_ps();
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for (i = 0; i < length / 8; i ++)
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{
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// Unroll loop for efficiency & calculate filter for 2*2 stereo samples
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// at each pass
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// sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
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// sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
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sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc) , pFil[0]));
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sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
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sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
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sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
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sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) , pFil[2]));
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sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
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sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
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sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
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pSrc += 16;
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pFil += 4;
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}
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// Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
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// to sum the two hi- and lo-floats of these registers together.
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// post-shuffle & add the filtered values and store to dest.
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_mm_storeu_ps(pDest, _mm_add_ps(
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_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
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_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
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));
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}
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// Ideas for further improvement:
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// 1. If it could be guaranteed that 'source' were always aligned to 16-byte
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// boundary, a faster aligned '_mm_load_ps' instruction could be used.
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// 2. If it could be guaranteed that 'dest' were always aligned to 16-byte
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// boundary, a faster '_mm_store_ps' instruction could be used.
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return (uint)count;
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/* original routine in C-language. please notice the C-version has differently
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organized coefficients though.
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double suml1, suml2;
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double sumr1, sumr2;
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uint i, j;
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for (j = 0; j < count; j += 2)
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{
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const float *ptr;
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const float *pFil;
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suml1 = sumr1 = 0.0;
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suml2 = sumr2 = 0.0;
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ptr = src;
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pFil = filterCoeffs;
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for (i = 0; i < lengthLocal; i ++)
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{
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// unroll loop for efficiency.
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suml1 += ptr[0] * pFil[0] +
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ptr[2] * pFil[2] +
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ptr[4] * pFil[4] +
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ptr[6] * pFil[6];
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sumr1 += ptr[1] * pFil[1] +
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ptr[3] * pFil[3] +
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ptr[5] * pFil[5] +
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ptr[7] * pFil[7];
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suml2 += ptr[8] * pFil[0] +
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ptr[10] * pFil[2] +
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ptr[12] * pFil[4] +
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ptr[14] * pFil[6];
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sumr2 += ptr[9] * pFil[1] +
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ptr[11] * pFil[3] +
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ptr[13] * pFil[5] +
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ptr[15] * pFil[7];
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ptr += 16;
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pFil += 8;
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}
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dest[0] = (float)suml1;
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dest[1] = (float)sumr1;
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dest[2] = (float)suml2;
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dest[3] = (float)sumr2;
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src += 4;
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dest += 4;
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}
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*/
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}
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#endif // SOUNDTOUCH_ALLOW_SSE
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