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authorPaul Davis <paul@linuxaudiosystems.com>2009-10-16 19:53:25 +0000
committerPaul Davis <paul@linuxaudiosystems.com>2009-10-16 19:53:25 +0000
commit69be1aa7bf84f9440c5b5757bd59193f7ff0d60d (patch)
tree1163a993fee1ed6c1ee539af8d7f772f816b6e5a /libs/ardour/interpolation.cc
parent4650d86312e7c191d208eefb03c299bfc1b4c031 (diff)
fix conceptually broken cubic interpolator that failed to handle the i=0 case, and don't rely on inlining for speed there
git-svn-id: svn://localhost/ardour2/branches/3.0@5792 d708f5d6-7413-0410-9779-e7cbd77b26cf
Diffstat (limited to 'libs/ardour/interpolation.cc')
-rw-r--r--libs/ardour/interpolation.cc78
1 files changed, 65 insertions, 13 deletions
diff --git a/libs/ardour/interpolation.cc b/libs/ardour/interpolation.cc
index 79ec82b482..47e3642bf3 100644
--- a/libs/ardour/interpolation.cc
+++ b/libs/ardour/interpolation.cc
@@ -61,19 +61,71 @@ CubicInterpolation::interpolate (int channel, nframes_t nframes, Sample *input,
}
distance = phase[channel];
- for (nframes_t outsample = 0; outsample < nframes; ++outsample) {
- i = floor(distance);
- Sample fractional_phase_part = distance - i;
- if (fractional_phase_part >= 1.0) {
- fractional_phase_part -= 1.0;
- i++;
- }
-
- if (input && output) {
- // Cubically interpolate into the output buffer
- output[outsample] = cube_interp(fractional_phase_part, input[i-1], input[i], input[i+1], input[i+2]);
- }
- distance += _speed + acceleration;
+
+ if (nframes < 3) {
+ /* no interpolation possible */
+
+ for (i = 0; i < nframes; ++i) {
+ output[i] = input[i];
+ }
+
+ return nframes;
+ }
+
+ /* keep this condition out of the inner loop */
+
+ if (input && output) {
+
+ Sample inm1;
+
+ if (floor (distance) == 0.0) {
+ /* best guess for the fake point we have to add to be able to interpolate at i == 0:
+ .... maintain slope of first actual segment ...
+ */
+ inm1 = input[i] - (input[i+1] - input[i]);
+ } else {
+ inm1 = input[i-1];
+ }
+
+ for (nframes_t outsample = 0; outsample < nframes; ++outsample) {
+
+ float f = floor (distance);
+ float fractional_phase_part = distance - f;
+
+ /* get the index into the input we should start with */
+
+ i = lrintf (f);
+
+ /* fractional_phase_part only reaches 1.0 thanks to float imprecision. In theory
+ it should always be < 1.0. If it ever >= 1.0, then bump the index we use
+ and back it off. This is the point where we "skip" an entire sample in the
+ input, because the phase part has accumulated so much error that we should
+ really be closer to the next sample. or something like that ...
+ */
+
+ if (fractional_phase_part >= 1.0) {
+ fractional_phase_part -= 1.0;
+ ++i;
+ }
+
+ // Cubically interpolate into the output buffer: keep this inlined for speed and rely on compiler
+ // optimization to take care of the rest
+ // shamelessly ripped from Steve Harris' swh-plugins (ladspa-util.h)
+
+ output[outsample] = input[i] + 0.5f * fractional_phase_part * (input[i+1] - inm1 +
+ fractional_phase_part * (4.0f * input[i+1] + 2.0f * inm1 - 5.0f * input[i] - input[i+2] +
+ fractional_phase_part * (3.0f * (input[i] - input[i+1]) - inm1 + input[i+2])));
+
+ distance += _speed + acceleration;
+ }
+
+ } else {
+
+ /* not sure that this is ever utilized - it implies that one of the input/output buffers is missing */
+
+ for (nframes_t outsample = 0; outsample < nframes; ++outsample) {
+ distance += _speed + acceleration;
+ }
}
i = floor(distance);