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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
101 lines
3.3 KiB
C++
101 lines
3.3 KiB
C++
////////////////////////////////////////////////////////////////////////////////
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///
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/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
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/// while maintaining the original pitch by using a time domain WSOLA-like method
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/// with several performance-increasing tweaks.
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///
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/// Anti-alias filter is used to prevent folding of high frequencies when
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/// transposing the sample rate with interpolation.
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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: 2014-01-07 21:41:23 +0200 (Tue, 07 Jan 2014) $
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// File revision : $Revision: 4 $
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//
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// $Id: AAFilter.h 187 2014-01-07 19:41:23Z 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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#ifndef AAFilter_H
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#define AAFilter_H
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#include "STTypes.h"
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#include "FIFOSampleBuffer.h"
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namespace soundtouch
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{
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class AAFilter
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{
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protected:
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class FIRFilter *pFIR;
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/// Low-pass filter cut-off frequency, negative = invalid
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double cutoffFreq;
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/// num of filter taps
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uint length;
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/// Calculate the FIR coefficients realizing the given cutoff-frequency
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void calculateCoeffs();
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public:
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AAFilter(uint length);
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~AAFilter();
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/// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
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/// frequency (nyquist frequency = 0.5). The filter will cut off the
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/// frequencies than that.
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void setCutoffFreq(double newCutoffFreq);
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/// Sets number of FIR filter taps, i.e. ~filter complexity
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void setLength(uint newLength);
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uint getLength() const;
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/// Applies the filter to the given sequence of samples.
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/// Note : The amount of outputted samples is by value of 'filter length'
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/// smaller than the amount of input samples.
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uint evaluate(SAMPLETYPE *dest,
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const SAMPLETYPE *src,
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uint numSamples,
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uint numChannels) const;
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/// Applies the filter to the given src & dest pipes, so that processed amount of
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/// samples get removed from src, and produced amount added to dest
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/// Note : The amount of outputted samples is by value of 'filter length'
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/// smaller than the amount of input samples.
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uint evaluate(FIFOSampleBuffer &dest,
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FIFOSampleBuffer &src) const;
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};
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}
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#endif
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