510 lines
25 KiB
C++
510 lines
25 KiB
C++
// Standalone tests for reasampler::PitchShifter — the S16 Preserve-engine DSP core. No VST3,
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// no REAPER, no vendor, no test framework. The compile-time proof it does NOT drag the WDL
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// <windows.h> chain is the CMake target linking only pitch_shift (+ peaks).
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//
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// Covers (PLAN.md S16 / CONTEXT.md §Pitch engine modes — Preserve):
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// 1. duration invariance — N inputs yield N outputs at every shift ratio (the load-bearing
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// Preserve property: a transposed render is the SAME frame length as the un-transposed one).
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// 2. unity pass-through fidelity — ratio 1.0 reproduces the input closely (a shifter at unity
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// must not mangle the signal).
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// 3. transpose direction — an octave-up shift raises the observed pitch (period shortens), an
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// octave-down lowers it (period lengthens), measured on a synthesized sine.
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// 4. RT discipline surrogate — after configure()+warm() (the off-thread setup), a long
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// process() run never resizes the ring (checked via window() constancy) and never returns
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// NaN/inf; pass-through (unconfigured) returns input verbatim.
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// 5. spectral purity + onset integrity (GA / GA2 regressions) — a PRIMED repitched PURE
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// SINE must come out as a SINGLE tone at the shifted frequency FROM THE VERY FIRST
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// MILLISECOND: no zero-gaps anywhere (the GA2 DAW report: silence-warmed rings made
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// every early splice jump into zeros — burst/gap/burst stutter in the first few ms),
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// and a per-block least-squares residual floor that catches harmonics, splice-cadence
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// sideband combs, and crossfade cancellation alike. Ratios cover the FULL playable
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// range the DAW report exercised: +2/-3 st, +/-1 octave, +24 st, +48 st (C8 from C4,
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// ratio 16) and -36 st (C1 from C4, ratio 1/8).
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// 6. unity + latency contract — asserted bit-exactly: a warm()ed shifter at ratio 1.0 IS a
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// clean window delay; a prime()d one has ZERO added latency (out[i] == src[i] to the
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// bit) — the GA2 immediate-onset claim.
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#include "../src/vst/pitch_shift.h"
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#include <cmath>
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#include <cstdio>
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#include <vector>
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using namespace reasampler;
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static int g_fail = 0;
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#define CHECK(cond) do { if(!(cond)) { \
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std::printf("FAIL line %d: %s\n", __LINE__, #cond); ++g_fail; } } while(0)
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static bool approx(double a, double b, double tol) { return std::fabs(a - b) <= tol; }
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constexpr double kPi = 3.14159265358979323846;
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// A sine of `cycles` periods over `frames` frames.
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static std::vector<AudioSample> sine(std::size_t frames, double cycles) {
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std::vector<AudioSample> s(frames);
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for (std::size_t i = 0; i < frames; ++i) {
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s[i] = static_cast<float>(std::sin(2.0 * kPi * cycles *
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static_cast<double>(i) / static_cast<double>(frames)));
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}
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return s;
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}
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// Average spacing between positive-going zero crossings (the observed period).
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static double observedPeriod(const std::vector<AudioSample>& out, std::size_t from) {
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std::vector<std::size_t> up;
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for (std::size_t i = from + 1; i < out.size(); ++i) {
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if (out[i - 1] <= 0.0f && out[i] > 0.0f) up.push_back(i);
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}
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if (up.size() < 2) return 0.0;
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double sum = 0.0;
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for (std::size_t i = 1; i < up.size(); ++i) sum += static_cast<double>(up[i] - up[i - 1]);
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return sum / static_cast<double>(up.size() - 1);
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}
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// --- 1. Duration invariance across shift ratios. ---
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static void testDurationInvariance() {
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// The core Preserve property: whatever the shift ratio, one input frame yields one output
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// frame. So a shifter fed N frames produces exactly N frames — a transposed render is the
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// same length as an un-transposed one (unlike Varispeed, where an octave up halves length).
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const std::size_t n = 4000;
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const std::vector<AudioSample> in = sine(n, 40.0);
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const double ratios[] = {0.5, 1.0, 2.0, std::pow(2.0, 7.0 / 12.0)};
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for (double r : ratios) {
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PitchShifter ps;
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ps.configure(2205); // ~50 ms @ 44.1k
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ps.warm();
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ps.setShiftRatio(r);
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std::size_t produced = 0;
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for (std::size_t i = 0; i < n; ++i) {
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const AudioSample o = ps.process(in[i]);
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(void)o;
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++produced; // exactly one output per input, unconditionally.
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}
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CHECK(produced == n); // duration held at every ratio.
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}
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}
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// --- 2. Unity pass-through fidelity. ---
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static void testUnityRoughlyReproduces() {
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// At ratio 1.0 the shifter should reproduce the input's PITCH faithfully (the OLA taps run
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// in lockstep with the writer). Amplitude/phase warble is allowed (basic OLA), but the
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// observed period must match the source period within a small tolerance past the warm-up.
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const std::size_t n = 8000;
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const double cycles = 40.0;
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const double nativePeriod = static_cast<double>(n) / cycles; // 200
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const std::vector<AudioSample> in = sine(n, cycles);
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PitchShifter ps;
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ps.configure(2205);
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ps.warm();
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ps.setShiftRatio(1.0);
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std::vector<AudioSample> out(n);
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for (std::size_t i = 0; i < n; ++i) out[i] = ps.process(in[i]);
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// Measure past the initial half-window latency region.
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const double p = observedPeriod(out, 3000);
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CHECK(p > 0.0);
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CHECK(approx(p, nativePeriod, nativePeriod * 0.10)); // within 10% of source period
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}
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// --- 3. Transpose direction: up shortens the period, down lengthens it. ---
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static void testTransposeDirection() {
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const std::size_t n = 12000;
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const double cycles = 60.0;
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const double nativePeriod = static_cast<double>(n) / cycles; // 200
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const std::vector<AudioSample> in = sine(n, cycles);
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// Octave up: output period ~ half the source period (higher pitch).
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{
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PitchShifter ps;
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ps.configure(2205);
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ps.warm();
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ps.setShiftRatio(2.0);
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std::vector<AudioSample> out(n);
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for (std::size_t i = 0; i < n; ++i) out[i] = ps.process(in[i]);
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const double p = observedPeriod(out, 4000);
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CHECK(p > 0.0);
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CHECK(approx(p, nativePeriod / 2.0, nativePeriod * 0.15)); // period halves
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}
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// Octave down: output period ~ double the source period (lower pitch).
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{
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PitchShifter ps;
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ps.configure(2205);
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ps.warm();
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ps.setShiftRatio(0.5);
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std::vector<AudioSample> out(n);
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for (std::size_t i = 0; i < n; ++i) out[i] = ps.process(in[i]);
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const double p = observedPeriod(out, 4000);
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CHECK(p > 0.0);
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CHECK(approx(p, nativePeriod * 2.0, nativePeriod * 0.30)); // period doubles
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}
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}
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// --- 4. RT discipline surrogate + pass-through. ---
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static void testRtDisciplineAndPassthrough() {
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// Unconfigured shifter passes input through verbatim (a Varispeed voice never allocates one).
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{
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PitchShifter ps;
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CHECK(!ps.configured());
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CHECK(ps.process(0.37f) == 0.37f); // exact pass-through
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CHECK(ps.process(-0.9f) == -0.9f);
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}
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// Configured: the window is fixed at configure() and never changes across a long run (no
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// per-frame Resize), and no output is NaN/inf (numerically well-behaved OLA).
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{
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PitchShifter ps;
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ps.configure(1024);
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ps.warm();
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const std::int64_t w = ps.window();
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CHECK(w == 1024);
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ps.setShiftRatio(std::pow(2.0, 5.0 / 12.0));
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const std::vector<AudioSample> in = sine(20000, 100.0);
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for (std::size_t i = 0; i < in.size(); ++i) {
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const AudioSample o = ps.process(in[i]);
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CHECK(std::isfinite(o));
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}
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CHECK(ps.window() == w); // window unchanged -> ring never resized mid-run
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}
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// A non-positive shift ratio is ignored (keeps the last valid ratio) — never stalls/reverses.
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{
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PitchShifter ps;
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ps.configure(512);
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ps.warm();
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ps.setShiftRatio(1.0);
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ps.setShiftRatio(-2.0); // ignored
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ps.setShiftRatio(0.0); // ignored
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for (int i = 0; i < 2000; ++i) CHECK(std::isfinite(ps.process(0.5f)));
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}
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// Degenerate window (<= 1) stays pass-through even after configure.
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{
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PitchShifter ps;
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ps.configure(1);
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CHECK(!ps.configured());
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CHECK(ps.process(0.25f) == 0.25f);
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}
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}
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// --- 5. Spectral purity + onset integrity: a PRIMED repitched pure sine is a SINGLE shifted
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// tone from the very first millisecond. ---
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static void testRepitchSpectralPurityAndOnset() {
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// Frequencies are in cycles/sample (rate-free). The source tone is chosen ADVERSARIALLY
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// on TWO axes simultaneously:
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// (a) f0*(w/2) ≈ (2205/2)/196.37 ≈ 5.609 cycles (frac ≈ 0.609) — content half a window
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// apart in the ring is near ANTI-PHASE. The old dual-tap design cancelled almost
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// completely at every crossfade midpoint for such tones — the DAW "severe beating /
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// multiple partials from a pure sine" bug.
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// (b) ringLen_*f0 ≈ 4410/196.37 ≈ 22.46 — near the half-integer alignment that makes the
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// pre-fix fade-headroom artifact visible at ratio 4 (+24 st). Additionally, period
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// 196.37 is NON-INTEGER, so the correlation peak is NOT on the integer lag grid; the
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// sub-sample parabolic refinement is LOAD-BEARING to stay at the -84 dB floor — the
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// old integer f0=1/196 put the optimum on the grid and the parabola contributed
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// nothing, making the -30 dB floor reachable without it.
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//
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// The shifter is driven exactly as the Voice drives it since GA2: prime() with the first
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// window of the source, then stream the CONTINUATION — so the measurements start at
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// output frame 0 and the onset regime (early splices near the primed boundary, the DAW
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// "zero-sample gaps in the first few ms" report) is inside the assertions, not skipped.
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const std::int64_t w = 2205; // ~50 ms @ 44.1k (the product window)
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const double f0 = 1.0 / 196.37; // NON-INTEGER period: the sub-sample correlation peak
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// is NOT on the integer grid, so the parabolic
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// refinement MUST contribute to achieve a clean
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// aligned splice — reverting it now FAILS this test.
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// (Old integer 1/196 put the optimum on the grid,
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// letting the parabola contribute nothing; the −30 dB
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// residual floor was then reachable without it.)
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const double ratios[] = {std::pow(2.0, 2.0 / 12.0), // +2 semitones (D from C)
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std::pow(2.0, -3.0 / 12.0), // -3 semitones (down-shift path)
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2.0, // octave up (the GA2 report: C5)
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std::pow(2.0, 24.0 / 12.0), // +24 st: ratio 4 — the ratio-scaled-
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// fade target (unscaled fade would
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// read stale data at ~75% gain)
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std::pow(2.0, 48.0 / 12.0), // +48 st: ratio 16 — C8 from C4 (the
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// GA2 "awful at C8" report; fast
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// splice cadence, short fades)
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std::pow(2.0, -12.0 / 12.0), // octave down (full down-shift path)
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std::pow(2.0, -36.0 / 12.0)}; // -36 st: ratio 1/8 — C1 from C4
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// (the GA2 down-shift report)
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for (double r : ratios) {
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PitchShifter ps;
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ps.configure(w);
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const std::size_t n = 120000;
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std::vector<AudioSample> src(n + static_cast<std::size_t>(w));
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for (std::size_t i = 0; i < src.size(); ++i) {
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src[i] = static_cast<AudioSample>(
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std::sin(2.0 * kPi * f0 * static_cast<double>(i)));
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}
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ps.prime(src.data(), w); // the Voice's note-on path: real content, not warm zeros
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ps.setShiftRatio(r);
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std::vector<double> out(n);
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for (std::size_t i = 0; i < n; ++i) {
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out[i] = static_cast<double>(ps.process(src[i + static_cast<std::size_t>(w)]));
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}
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// (a) ONSET/GAP integrity over the ENTIRE run, frame 0 included: no near-zero run
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// longer than 32 frames (~0.7 ms). A unit-amplitude shifted sine dwells below 1e-3
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// for well under one frame per zero crossing even at the lowest ratio here, while the
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// pre-fix onset gaps were hundreds to thousands of frames of literal silence.
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std::size_t worstGap = 0, run = 0;
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for (std::size_t i = 0; i < n; ++i) {
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if (std::fabs(out[i]) < 1e-3) {
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++run;
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if (run > worstGap) worstGap = run;
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} else {
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run = 0;
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}
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}
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CHECK(worstGap < 32);
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// (b) PER-BLOCK least-squares fit of a*sin + b*cos at the SHIFTED frequency, from the
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// FIRST block. Fitting phase per block deliberately tolerates the slow (pitch-true,
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// inaudible) SOLA phase wander across seconds while catching everything audible:
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// harmonics ("square-ish"), splice-cadence sideband combs (the spectrogram alias
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// lines), crossfade cancellation, and onset gaps all land in the residual or collapse
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// the in-block fit amplitude. Solve the exact 2x2 normal equations per block.
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const double f1 = r * f0;
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const std::size_t block = 4096;
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for (std::size_t b0 = 0; b0 + block <= n; b0 += block) {
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double sss = 0.0, scc = 0.0, ssc = 0.0, sys = 0.0, syc = 0.0;
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for (std::size_t i = b0; i < b0 + block; ++i) {
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const double ph = 2.0 * kPi * f1 * static_cast<double>(i);
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const double s = std::sin(ph), c = std::cos(ph);
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sss += s * s; scc += c * c; ssc += s * c;
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sys += out[i] * s; syc += out[i] * c;
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}
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const double det = sss * scc - ssc * ssc;
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CHECK(det > 0.0);
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const double a = (sys * scc - syc * ssc) / det;
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const double b = (syc * sss - sys * ssc) / det;
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double residSq = 0.0, fitSq = 0.0;
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for (std::size_t i = b0; i < b0 + block; ++i) {
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const double ph = 2.0 * kPi * f1 * static_cast<double>(i);
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const double fit = a * std::sin(ph) + b * std::cos(ph);
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residSq += (out[i] - fit) * (out[i] - fit);
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fitSq += fit * fit;
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}
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const double fitRms = std::sqrt(fitSq / static_cast<double>(block));
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const double residRms = std::sqrt(residSq / static_cast<double>(block));
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// The shifted tone is there at full amplitude (unit sine RMS ~0.707) in EVERY
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// block — a gapped or beating block collapses this...
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CHECK(fitRms > 0.6);
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CHECK(fitRms < 0.8);
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// ...and it is the ONLY thing there: residual at least 84 dB under the tone.
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// With the non-integer f0=1/196.37, the sub-sample parabolic refinement is
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// LOAD-BEARING: reverting it raises the floor to ~-50 dB (ratio 1/8), failing here.
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// With integer f0=1/196 the optimum was on the integer grid and the parabola
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// contributed nothing — the old floor of -30 dB was reachable without it.
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// Engine steady-state measures -84 dB and better across all 7 tested ratios;
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// moderate ratios (+/-2 st, octaves) sit at -88 dB typical.
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CHECK(residRms < 0.000063 * fitRms); // -84 dB floor
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}
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}
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}
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// --- 6. Unity + latency contract: warm = bit-exact window delay; primed = bit-exact ZERO
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// latency. ---
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static void testUnityBitExactAndLatency() {
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// A configured shifter at ratio 1.0 parks the tap mid-band (no splice ever fires) at an
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// integral delay (no interpolation error). After warm() that delay is exactly one window
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// of declared silence, so out[i] == in[i - w] to the bit. After prime() with the first
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// window of source the tap sits ON src[0] — out[i] == src[i] to the bit from the very
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// first frame: the GA2 zero-structural-latency (immediate onset) claim.
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const std::int64_t w = 2205; // the product window
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const std::size_t n = 6000;
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const std::vector<AudioSample> in = sine(n + static_cast<std::size_t>(w), 37.0);
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// warm(): a clean, bit-exact one-window delay of the streamed input.
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{
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PitchShifter ps;
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ps.configure(w);
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ps.warm();
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ps.setShiftRatio(1.0);
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std::vector<AudioSample> out(n);
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for (std::size_t i = 0; i < n; ++i) out[i] = ps.process(in[i]);
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std::size_t badSilence = 0, badDelay = 0;
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for (std::size_t i = 0; i < static_cast<std::size_t>(w); ++i) {
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if (out[i] != 0.0f) ++badSilence; // pre-latency region: declared silence, exact
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}
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for (std::size_t i = static_cast<std::size_t>(w); i < n; ++i) {
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if (out[i] != in[i - static_cast<std::size_t>(w)]) ++badDelay; // bit-exact delay
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}
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CHECK(badSilence == 0);
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CHECK(badDelay == 0);
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}
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// prime(): zero added latency — the output IS the source from frame 0, bit-exact.
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{
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PitchShifter ps;
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ps.configure(w);
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ps.prime(in.data(), w);
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ps.setShiftRatio(1.0);
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std::size_t badZeroLat = 0;
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for (std::size_t i = 0; i < n; ++i) {
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if (ps.process(in[i + static_cast<std::size_t>(w)]) != in[i]) ++badZeroLat;
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}
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CHECK(badZeroLat == 0);
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}
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}
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// --- 7. Tail wind-down (GA3): freezeTail() at source exhaustion keeps the output a
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// continuous, full-amplitude tone at the shifted frequency — the splice machinery
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// recycles the ring's frozen ALL-REAL tail instead of chopping against held-DC
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// padding (the DAW "ring modulation" troughs growing toward the note end). ---
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static void testFreezeTailContinuousTone() {
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const std::int64_t w = 2205;
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const double f0 = 1.0 / 196.37; // non-integer period (the test-5 adversarial tone)
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const std::size_t stream = 20000; // frames fed before exhaustion (several splice cycles)
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const double ratios[] = {std::pow(2.0, 7.0 / 12.0), // +7 st (the DAW report regime)
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2.0, // octave up
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std::pow(2.0, 24.0 / 12.0), // +24 st: fast frozen drain
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std::pow(2.0, -5.0 / 12.0), // -5 st (down-shift tail)
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1.0}; // unity: frozen delay drains at 1 —
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// splices NOW fire even at unity
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for (double r : ratios) {
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PitchShifter ps;
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ps.configure(w);
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std::vector<AudioSample> src(stream + static_cast<std::size_t>(w));
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for (std::size_t i = 0; i < src.size(); ++i) {
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src[i] = static_cast<AudioSample>(
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std::sin(2.0 * kPi * f0 * static_cast<double>(i)));
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}
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ps.prime(src.data(), w);
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ps.setShiftRatio(r);
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for (std::size_t i = 0; i < stream; ++i) {
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(void)ps.process(src[i + static_cast<std::size_t>(w)]);
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}
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// Source exhausted: freeze (idempotent) and keep producing for one full window — the
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// longest a Voice runs frozen (its own note end lands within a window of exhaustion).
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CHECK(!ps.tailFrozen());
|
||
ps.freezeTail();
|
||
ps.freezeTail(); // double-freeze harmless
|
||
CHECK(ps.tailFrozen());
|
||
const std::size_t tail = static_cast<std::size_t>(w);
|
||
std::vector<double> out(tail);
|
||
for (std::size_t i = 0; i < tail; ++i) {
|
||
out[i] = static_cast<double>(ps.process(0.0f)); // input ignored while frozen
|
||
CHECK(std::isfinite(out[i]));
|
||
}
|
||
// (a) No dead stretches: a unit-amplitude tone dwells below 0.05 only a few frames
|
||
// per zero crossing; the pre-GA3 DC chop ran hundreds.
|
||
std::size_t worstGap = 0, run = 0;
|
||
for (std::size_t i = 0; i < tail; ++i) {
|
||
if (std::fabs(out[i]) < 0.05) {
|
||
++run;
|
||
if (run > worstGap) worstGap = run;
|
||
} else {
|
||
run = 0;
|
||
}
|
||
}
|
||
CHECK(worstGap < 24);
|
||
// (b) Full amplitude throughout: every 256-frame block spans > a half period at all
|
||
// tested ratios, so a continuous tone peaks near 1.0 in each.
|
||
for (std::size_t b = 0; b + 256 <= tail; b += 256) {
|
||
double peak = 0.0;
|
||
for (std::size_t i = b; i < b + 256; ++i) {
|
||
if (std::fabs(out[i]) > peak) peak = std::fabs(out[i]);
|
||
}
|
||
CHECK(peak > 0.5);
|
||
CHECK(peak < 1.1); // aligned complementary fades: no cancellation, no bulge
|
||
}
|
||
}
|
||
|
||
// Freeze landing MID-CROSSFADE: at ratio 2 from a fresh prime the tap drains from delay
|
||
// w at 1/frame, splices at w/4 (frame 3w/4), then fades for w/4 frames — so frame
|
||
// 3w/4 + w/8 is deterministically mid-fade. The frozen writer makes the outgoing tap
|
||
// close at the FULL ratio; the transition caps the live fade so it completes before
|
||
// reading lapped content — output must stay finite, gap-free, and bounded.
|
||
{
|
||
PitchShifter ps;
|
||
ps.configure(w);
|
||
std::vector<AudioSample> src(4 * static_cast<std::size_t>(w));
|
||
for (std::size_t i = 0; i < src.size(); ++i) {
|
||
src[i] = static_cast<AudioSample>(
|
||
std::sin(2.0 * kPi * f0 * static_cast<double>(i)));
|
||
}
|
||
ps.prime(src.data(), w);
|
||
ps.setShiftRatio(2.0);
|
||
const std::size_t preFreeze = static_cast<std::size_t>(3 * w / 4 + w / 8);
|
||
double lastPre = 0.0;
|
||
for (std::size_t i = 0; i < preFreeze; ++i) {
|
||
lastPre = static_cast<double>(ps.process(src[i + static_cast<std::size_t>(w)]));
|
||
}
|
||
ps.freezeTail();
|
||
std::size_t worstGap = 0, run = 0;
|
||
for (std::size_t i = 0; i < static_cast<std::size_t>(w); ++i) {
|
||
const double o = static_cast<double>(ps.process(0.0f));
|
||
CHECK(std::isfinite(o));
|
||
CHECK(std::fabs(o) < 1.1);
|
||
if (std::fabs(o) < 0.05) {
|
||
++run;
|
||
if (run > worstGap) worstGap = run;
|
||
} else {
|
||
run = 0;
|
||
}
|
||
}
|
||
CHECK(worstGap < 24);
|
||
// reset()/prime() clear the freeze: the shifter is fully reusable for the next
|
||
// note-on, and a primed unity run is STILL bit-exact zero-latency (no stale state).
|
||
ps.reset();
|
||
CHECK(!ps.tailFrozen());
|
||
ps.prime(src.data(), w);
|
||
ps.setShiftRatio(1.0);
|
||
std::size_t badZeroLat = 0;
|
||
for (std::size_t i = 0; i < 2000; ++i) {
|
||
if (ps.process(src[i + static_cast<std::size_t>(w)]) != src[i]) ++badZeroLat;
|
||
}
|
||
CHECK(badZeroLat == 0);
|
||
}
|
||
|
||
// STEP-DETECTOR: freeze-transition continuity using a ramp source where tapA and tapB
|
||
// read values that differ by a predictable constant (≈ A * w / N), making the crossfade
|
||
// gain step directly visible in the output. With a ramp, the per-frame natural change is
|
||
// A/(N) * ratio ≈ 0.0002 per frame; the un-fixed gain step is ~0.247 * (w/N) ≈ 0.062 —
|
||
// roughly 300x the natural rate. A threshold of 0.02 clearly separates fixed from unfixed.
|
||
//
|
||
// The ramp also defeats correlation-alignment (all lags score equally on a linear ramp),
|
||
// so the splice jump of one window guarantees tapA - tapB = −A*w/N regardless of lag.
|
||
{
|
||
PitchShifter ps;
|
||
ps.configure(w);
|
||
// Ramp from 0.0 to 1.0 over 4*w frames (same buffer size as the mid-crossfade case).
|
||
const std::size_t rampLen = 4 * static_cast<std::size_t>(w);
|
||
std::vector<AudioSample> ramp(rampLen);
|
||
for (std::size_t i = 0; i < rampLen; ++i) {
|
||
ramp[i] = static_cast<AudioSample>(static_cast<double>(i) /
|
||
static_cast<double>(rampLen - 1));
|
||
}
|
||
ps.prime(ramp.data(), w);
|
||
ps.setShiftRatio(2.0);
|
||
// Drive to the deterministic mid-fade freeze point: same preFreeze offset as above.
|
||
const std::size_t preFreeze = static_cast<std::size_t>(3 * w / 4 + w / 8);
|
||
double lastPre = 0.0;
|
||
for (std::size_t i = 0; i < preFreeze; ++i) {
|
||
lastPre = static_cast<double>(
|
||
ps.process(ramp[i + static_cast<std::size_t>(w)]));
|
||
}
|
||
ps.freezeTail();
|
||
// First frozen frame — if gNew steps at the freeze boundary the output jumps by
|
||
// ~deltaGain * (tapA - tapB) ≈ 0.247 * 0.25 = 0.062.
|
||
const double firstFrozen = static_cast<double>(ps.process(0.0f));
|
||
CHECK(std::isfinite(firstFrozen));
|
||
// Natural per-frame ramp advance at ratio 2 ≈ 2/(4*w - 1) ≈ 0.0002; the un-fixed
|
||
// step is ~0.062. Threshold 0.02 is 100x the natural rate but well below the step.
|
||
const double transitionStep = std::fabs(firstFrozen - lastPre);
|
||
CHECK(transitionStep < 0.02);
|
||
}
|
||
}
|
||
|
||
int main() {
|
||
testDurationInvariance();
|
||
testUnityRoughlyReproduces();
|
||
testTransposeDirection();
|
||
testRtDisciplineAndPassthrough();
|
||
testRepitchSpectralPurityAndOnset();
|
||
testUnityBitExactAndLatency();
|
||
testFreezeTailContinuousTone();
|
||
|
||
if (g_fail == 0) {
|
||
std::printf("all pitch_shift tests passed\n");
|
||
return 0;
|
||
}
|
||
std::printf("%d pitch_shift check(s) failed\n", g_fail);
|
||
return 1;
|
||
}
|