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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
201 lines
6.2 KiB
C++
201 lines
6.2 KiB
C++
////////////////////////////////////////////////////////////////////////////////
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///
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/// Cubic interpolation routine.
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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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// $Id: InterpolateCubic.cpp 179 2014-01-06 18:41:42Z 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 <stddef.h>
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#include <math.h>
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#include "InterpolateCubic.h"
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#include "STTypes.h"
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using namespace soundtouch;
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// cubic interpolation coefficients
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static const float _coeffs[]=
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{ -0.5f, 1.0f, -0.5f, 0.0f,
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1.5f, -2.5f, 0.0f, 1.0f,
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-1.5f, 2.0f, 0.5f, 0.0f,
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0.5f, -0.5f, 0.0f, 0.0f};
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InterpolateCubic::InterpolateCubic()
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{
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fract = 0;
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}
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void InterpolateCubic::resetRegisters()
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{
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fract = 0;
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}
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/// Transpose mono audio. Returns number of produced output samples, and
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/// updates "srcSamples" to amount of consumed source samples
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int InterpolateCubic::transposeMono(SAMPLETYPE *pdest,
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const SAMPLETYPE *psrc,
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int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 4;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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float out;
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const float x3 = 1.0f;
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const float x2 = (float)fract; // x
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const float x1 = x2*x2; // x^2
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const float x0 = x1*x2; // x^3
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float y0, y1, y2, y3;
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assert(fract < 1.0);
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y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
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y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
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y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
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y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
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out = y0 * psrc[0] + y1 * psrc[1] + y2 * psrc[2] + y3 * psrc[3];
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pdest[i] = (SAMPLETYPE)out;
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i ++;
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// update position fraction
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fract += rate;
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// update whole positions
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int whole = (int)fract;
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fract -= whole;
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psrc += whole;
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srcCount += whole;
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}
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srcSamples = srcCount;
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return i;
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}
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/// Transpose stereo audio. Returns number of produced output samples, and
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/// updates "srcSamples" to amount of consumed source samples
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int InterpolateCubic::transposeStereo(SAMPLETYPE *pdest,
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const SAMPLETYPE *psrc,
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int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 4;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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const float x3 = 1.0f;
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const float x2 = (float)fract; // x
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const float x1 = x2*x2; // x^2
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const float x0 = x1*x2; // x^3
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float y0, y1, y2, y3;
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float out0, out1;
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assert(fract < 1.0);
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y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
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y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
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y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
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y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
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out0 = y0 * psrc[0] + y1 * psrc[2] + y2 * psrc[4] + y3 * psrc[6];
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out1 = y0 * psrc[1] + y1 * psrc[3] + y2 * psrc[5] + y3 * psrc[7];
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pdest[2*i] = (SAMPLETYPE)out0;
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pdest[2*i+1] = (SAMPLETYPE)out1;
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i ++;
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// update position fraction
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fract += rate;
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// update whole positions
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int whole = (int)fract;
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fract -= whole;
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psrc += 2*whole;
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srcCount += whole;
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}
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srcSamples = srcCount;
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return i;
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}
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/// Transpose multi-channel audio. Returns number of produced output samples, and
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/// updates "srcSamples" to amount of consumed source samples
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int InterpolateCubic::transposeMulti(SAMPLETYPE *pdest,
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const SAMPLETYPE *psrc,
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int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 4;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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const float x3 = 1.0f;
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const float x2 = (float)fract; // x
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const float x1 = x2*x2; // x^2
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const float x0 = x1*x2; // x^3
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float y0, y1, y2, y3;
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assert(fract < 1.0);
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y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
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y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
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y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
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y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
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for (int c = 0; c < numChannels; c ++)
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{
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float out;
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out = y0 * psrc[c] + y1 * psrc[c + numChannels] + y2 * psrc[c + 2 * numChannels] + y3 * psrc[c + 3 * numChannels];
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pdest[0] = (SAMPLETYPE)out;
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pdest ++;
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}
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i ++;
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// update position fraction
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fract += rate;
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// update whole positions
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int whole = (int)fract;
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fract -= whole;
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psrc += numChannels*whole;
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srcCount += whole;
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}
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srcSamples = srcCount;
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return i;
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}
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