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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
165 lines
6.2 KiB
C++
165 lines
6.2 KiB
C++
////////////////////////////////////////////////////////////////////////////////
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///
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/// Beats-per-minute (BPM) detection routine.
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///
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/// The beat detection algorithm works as follows:
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/// - Use function 'inputSamples' to input a chunks of samples to the class for
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/// analysis. It's a good idea to enter a large sound file or stream in smallish
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/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
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/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
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/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
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/// Simple averaging is used for anti-alias filtering because the resulting signal
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/// quality isn't of that high importance.
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/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
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/// taking absolute value that's smoothed by sliding average. Signal levels that
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/// are below a couple of times the general RMS amplitude level are cut away to
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/// leave only notable peaks there.
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/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
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/// autocorrelation function of the enveloped signal.
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/// - After whole sound data file has been analyzed as above, the bpm level is
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/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
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/// function, calculates it's precise location and converts this reading to bpm's.
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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: 2012-08-30 22:53:44 +0300 (Thu, 30 Aug 2012) $
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// File revision : $Revision: 4 $
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//
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// $Id: BPMDetect.h 150 2012-08-30 19:53:44Z 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 _BPMDetect_H_
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#define _BPMDetect_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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/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
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#define MIN_BPM 29
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/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
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#define MAX_BPM 200
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/// Class for calculating BPM rate for audio data.
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class BPMDetect
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{
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protected:
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/// Auto-correlation accumulator bins.
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float *xcorr;
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/// Amplitude envelope sliding average approximation level accumulator
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double envelopeAccu;
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/// RMS volume sliding average approximation level accumulator
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double RMSVolumeAccu;
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/// Sample average counter.
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int decimateCount;
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/// Sample average accumulator for FIFO-like decimation.
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soundtouch::LONG_SAMPLETYPE decimateSum;
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/// Decimate sound by this coefficient to reach approx. 500 Hz.
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int decimateBy;
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/// Auto-correlation window length
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int windowLen;
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/// Number of channels (1 = mono, 2 = stereo)
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int channels;
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/// sample rate
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int sampleRate;
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/// Beginning of auto-correlation window: Autocorrelation isn't being updated for
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/// the first these many correlation bins.
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int windowStart;
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/// FIFO-buffer for decimated processing samples.
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soundtouch::FIFOSampleBuffer *buffer;
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/// Updates auto-correlation function for given number of decimated samples that
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/// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
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/// though).
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void updateXCorr(int process_samples /// How many samples are processed.
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);
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/// Decimates samples to approx. 500 Hz.
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///
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/// \return Number of output samples.
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int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
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const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
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int numsamples ///< Number of source samples.
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);
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/// Calculates amplitude envelope for the buffer of samples.
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/// Result is output to 'samples'.
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void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
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int numsamples ///< Number of samples in buffer
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);
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/// remove constant bias from xcorr data
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void removeBias();
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public:
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/// Constructor.
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BPMDetect(int numChannels, ///< Number of channels in sample data.
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int sampleRate ///< Sample rate in Hz.
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);
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/// Destructor.
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virtual ~BPMDetect();
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/// Inputs a block of samples for analyzing: Envelopes the samples and then
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/// updates the autocorrelation estimation. When whole song data has been input
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/// in smaller blocks using this function, read the resulting bpm with 'getBpm'
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/// function.
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///
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/// Notice that data in 'samples' array can be disrupted in processing.
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void inputSamples(const soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
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int numSamples ///< Number of samples in buffer
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);
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/// Analyzes the results and returns the BPM rate. Use this function to read result
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/// after whole song data has been input to the class by consecutive calls of
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/// 'inputSamples' function.
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///
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/// \return Beats-per-minute rate, or zero if detection failed.
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float getBpm();
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};
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}
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#endif // _BPMDetect_H_
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