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diff --git a/libs/qm-dsp/dsp/rateconversion/Resampler.h b/libs/qm-dsp/dsp/rateconversion/Resampler.h
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+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+/*
+    QM DSP Library
+
+    Centre for Digital Music, Queen Mary, University of London.
+    This file by Chris Cannam.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#ifndef RESAMPLER_H
+#define RESAMPLER_H
+
+#include <vector>
+
+/**
+ * Resampler resamples a stream from one integer sample rate to
+ * another (arbitrary) rate, using a kaiser-windowed sinc filter.  The
+ * results and performance are pretty similar to libraries such as
+ * libsamplerate, though this implementation does not support
+ * time-varying ratios (the ratio is fixed on construction).
+ *
+ * See also Decimator, which is faster and rougher but supports only
+ * power-of-two downsampling factors.
+ */
+class Resampler
+{
+public:
+    /**
+     * Construct a Resampler to resample from sourceRate to
+     * targetRate.
+     */
+    Resampler(int sourceRate, int targetRate);
+
+    /**
+     * Construct a Resampler to resample from sourceRate to
+     * targetRate, using the given filter parameters.
+     */
+    Resampler(int sourceRate, int targetRate,
+              double snr, double bandwidth);
+
+    virtual ~Resampler();
+
+    /**
+     * Read n input samples from src and write resampled data to
+     * dst. The return value is the number of samples written, which
+     * will be no more than ceil((n * targetRate) / sourceRate). The
+     * caller must ensure the dst buffer has enough space for the
+     * samples returned.
+     */
+    int process(const double *src, double *dst, int n);
+
+    /**
+     * Read n input samples from src and return resampled data by
+     * value.
+     */
+    std::vector<double> process(const double *src, int n);
+
+    /**
+     * Return the number of samples of latency at the output due by
+     * the filter. (That is, the output will be delayed by this number
+     * of samples relative to the input.)
+     */
+    int getLatency() const { return m_latency; }
+
+    /**
+     * Carry out a one-off resample of a single block of n
+     * samples. The output is latency-compensated.
+     */
+    static std::vector<double> resample
+    (int sourceRate, int targetRate, const double *data, int n);
+
+private:
+    int m_sourceRate;
+    int m_targetRate;
+    int m_gcd;
+    int m_filterLength;
+    int m_bufferLength;
+    int m_latency;
+    double m_peakToPole;
+    
+    struct Phase {
+        int nextPhase;
+        std::vector<double> filter;
+        int drop;
+    };
+
+    Phase *m_phaseData;
+    int m_phase;
+    std::vector<double> m_buffer;
+    int m_bufferOrigin;
+
+    void initialise(double, double);
+    double reconstructOne();
+};
+
+#endif
+