diff options
author | Taybin Rutkin <taybin@taybin.com> | 2005-05-18 16:08:37 +0000 |
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committer | Taybin Rutkin <taybin@taybin.com> | 2005-05-18 16:08:37 +0000 |
commit | fa7c141813844ce5b4c93ca126ed84ab568a2892 (patch) | |
tree | aafbf59e70135651039fffce7a2975099efcd1d6 /libs/soundtouch/FIRFilter.cpp | |
parent | 90dff30cf4b3dd1d9e752754395cdaf720eeae5f (diff) |
Updated to soundtouch-1.3 (plus modifications)
git-svn-id: svn://localhost/trunk/ardour2@13 d708f5d6-7413-0410-9779-e7cbd77b26cf
Diffstat (limited to 'libs/soundtouch/FIRFilter.cpp')
-rw-r--r-- | libs/soundtouch/FIRFilter.cpp | 254 |
1 files changed, 254 insertions, 0 deletions
diff --git a/libs/soundtouch/FIRFilter.cpp b/libs/soundtouch/FIRFilter.cpp new file mode 100644 index 0000000000..cc9c40d883 --- /dev/null +++ b/libs/soundtouch/FIRFilter.cpp @@ -0,0 +1,254 @@ +//////////////////////////////////////////////////////////////////////////////// +/// +/// General FIR digital filter routines with MMX optimization. +/// +/// Note : MMX optimized functions reside in a separate, platform-specific file, +/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp' +/// +/// Author : Copyright (c) Olli Parviainen +/// Author e-mail : oparviai @ iki.fi +/// SoundTouch WWW: http://www.iki.fi/oparviai/soundtouch +/// +//////////////////////////////////////////////////////////////////////////////// +// +// Last changed : $Date$ +// File revision : $Revision$ +// +// $Id$ +// +//////////////////////////////////////////////////////////////////////////////// +// +// License : +// +// SoundTouch audio processing library +// Copyright (c) Olli Parviainen +// +// This library is free software; you can redistribute it and/or +// modify it under the terms of the GNU Lesser General Public +// License as published by the Free Software Foundation; either +// version 2.1 of the License, or (at your option) any later version. +// +// This library 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 +// Lesser General Public License for more details. +// +// You should have received a copy of the GNU Lesser General Public +// License along with this library; if not, write to the Free Software +// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +// +//////////////////////////////////////////////////////////////////////////////// + +#include <memory.h> +#include <assert.h> +#include <math.h> +#include <stdlib.h> +#include <stdexcept> +#include "FIRFilter.h" +#include "cpu_detect.h" + +using namespace soundtouch; + +/***************************************************************************** + * + * Implementation of the class 'FIRFilter' + * + *****************************************************************************/ + +FIRFilter::FIRFilter() +{ + resultDivFactor = 0; + length = 0; + lengthDiv8 = 0; + filterCoeffs = NULL; +} + + +FIRFilter::~FIRFilter() +{ + delete[] filterCoeffs; +} + +// Usual C-version of the filter routine for stereo sound +uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const +{ + uint i, j, end; + LONG_SAMPLETYPE suml, sumr; +#ifdef FLOAT_SAMPLES + // when using floating point samples, use a scaler instead of a divider + // because division is much slower operation than multiplying. + double dScaler = 1.0 / (double)resultDivider; +#endif + + assert(length != 0); + + end = 2 * (numSamples - length); + + for (j = 0; j < end; j += 2) + { + const SAMPLETYPE *ptr; + + suml = sumr = 0; + ptr = src + j; + + for (i = 0; i < length; i += 4) + { + // loop is unrolled by factor of 4 here for efficiency + suml += ptr[2 * i + 0] * filterCoeffs[i + 0] + + ptr[2 * i + 2] * filterCoeffs[i + 1] + + ptr[2 * i + 4] * filterCoeffs[i + 2] + + ptr[2 * i + 6] * filterCoeffs[i + 3]; + sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] + + ptr[2 * i + 3] * filterCoeffs[i + 1] + + ptr[2 * i + 5] * filterCoeffs[i + 2] + + ptr[2 * i + 7] * filterCoeffs[i + 3]; + } + +#ifdef INTEGER_SAMPLES + suml >>= resultDivFactor; + sumr >>= resultDivFactor; + // saturate to 16 bit integer limits + suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml; + // saturate to 16 bit integer limits + sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr; +#else + suml *= dScaler; + sumr *= dScaler; +#endif // INTEGER_SAMPLES + dest[j] = (SAMPLETYPE)suml; + dest[j + 1] = (SAMPLETYPE)sumr; + } + return numSamples - length; +} + + + + +// Usual C-version of the filter routine for mono sound +uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const +{ + uint i, j, end; + LONG_SAMPLETYPE sum; +#ifdef FLOAT_SAMPLES + // when using floating point samples, use a scaler instead of a divider + // because division is much slower operation than multiplying. + double dScaler = 1.0 / (double)resultDivider; +#endif + + + assert(length != 0); + + end = numSamples - length; + for (j = 0; j < end; j ++) + { + sum = 0; + for (i = 0; i < length; i += 4) + { + // loop is unrolled by factor of 4 here for efficiency + sum += src[i + 0] * filterCoeffs[i + 0] + + src[i + 1] * filterCoeffs[i + 1] + + src[i + 2] * filterCoeffs[i + 2] + + src[i + 3] * filterCoeffs[i + 3]; + } +#ifdef INTEGER_SAMPLES + sum >>= resultDivFactor; + // saturate to 16 bit integer limits + sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum; +#else + sum *= dScaler; +#endif // INTEGER_SAMPLES + dest[j] = (SAMPLETYPE)sum; + src ++; + } + return end; +} + + +// Set filter coeffiecients and length. +// +// Throws an exception if filter length isn't divisible by 8 +void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor) +{ + assert(newLength > 0); + if (newLength % 8) throw std::runtime_error("FIR filter length not divisible by 8"); + + lengthDiv8 = newLength / 8; + length = lengthDiv8 * 8; + assert(length == newLength); + + resultDivFactor = uResultDivFactor; + resultDivider = (uint)pow(2, resultDivFactor); + + delete[] filterCoeffs; + filterCoeffs = new SAMPLETYPE[length]; + memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE)); +} + + +uint FIRFilter::getLength() const +{ + return length; +} + + + +// Applies the filter to the given sequence of samples. +// +// Note : The amount of outputted samples is by value of 'filter_length' +// smaller than the amount of input samples. +uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const +{ + assert(numChannels == 1 || numChannels == 2); + + assert(length > 0); + assert(lengthDiv8 * 8 == length); + if (numSamples < length) return 0; + assert(resultDivFactor >= 0); + if (numChannels == 2) + { + return evaluateFilterStereo(dest, src, numSamples); + } else { + return evaluateFilterMono(dest, src, numSamples); + } +} + +FIRFilter * FIRFilter::newInstance() +{ + uint uExtensions; + + uExtensions = detectCPUextensions(); + + // Check if MMX/SSE/3DNow! instruction set extensions supported by CPU + +#ifdef ALLOW_MMX + // MMX routines available only with integer sample types + if (uExtensions & SUPPORT_MMX) + { + return ::new FIRFilterMMX; + } + else +#endif // ALLOW_MMX + +#ifdef ALLOW_SSE + if (uExtensions & SUPPORT_SSE) + { + // SSE support + return ::new FIRFilterSSE; + } + else +#endif // ALLOW_SSE + +#ifdef ALLOW_3DNOW + if (uExtensions & SUPPORT_3DNOW) + { + // 3DNow! support + return ::new FIRFilter3DNow; + } + else +#endif // ALLOW_3DNOW + + { + // ISA optimizations not supported, use plain C version + return ::new FIRFilter; + } +} |