+/*
+ Copyright (C) 2012 Paul Davis
+
+ 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.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+*/
+
#include <stdint.h>
+#include <cstdio>
+
#include "ardour/interpolation.h"
-nframes_t
-LinearInterpolation::interpolate (nframes_t nframes, Sample *input, Sample *output)
+using namespace ARDOUR;
+
+
+framecnt_t
+LinearInterpolation::interpolate (int channel, framecnt_t nframes, Sample *input, Sample *output)
{
- // the idea behind phase is that when the speed is not 1.0, we have to
- // interpolate between samples and then we have to store where we thought we were.
- // rather than being at sample N or N+1, we were at N+0.8792922
- // so the "phase" element, if you want to think about this way,
- // varies from 0 to 1, representing the "offset" between samples
- uint64_t phase = last_phase;
-
- // acceleration
- int64_t phi_delta;
-
- // phi = fixed point speed
- if (phi != target_phi) {
- phi_delta = ((int64_t)(target_phi - phi)) / nframes;
- } else {
- phi_delta = 0;
- }
-
// index in the input buffers
- nframes_t i = 0;
-
- for (nframes_t outsample = 0; outsample < nframes; ++outsample) {
- i = phase >> 24;
- Sample fractional_phase_part = (phase & fractional_part_mask) / binary_scaling_factor;
-
- // Linearly interpolate into the output buffer
- // using fixed point math
- output[outsample] =
- input[i] * (1.0f - fractional_phase_part) +
- input[i+1] * fractional_phase_part;
- phase += phi + phi_delta;
+ framecnt_t i = 0;
+
+ double acceleration = 0;
+
+ if (_speed != _target_speed) {
+ acceleration = _target_speed - _speed;
}
- last_phase = (phase & fractional_part_mask);
-
- // playback distance
+ for (framecnt_t outsample = 0; outsample < nframes; ++outsample) {
+ double const d = phase[channel] + outsample * (_speed + acceleration);
+ i = floor(d);
+ Sample fractional_phase_part = d - i;
+ if (fractional_phase_part >= 1.0) {
+ fractional_phase_part -= 1.0;
+ i++;
+ }
+
+ if (input && output) {
+ // Linearly interpolate into the output buffer
+ output[outsample] =
+ input[i] * (1.0f - fractional_phase_part) +
+ input[i+1] * fractional_phase_part;
+ }
+ }
+
+ double const distance = phase[channel] + nframes * (_speed + acceleration);
+ i = floor(distance);
+ phase[channel] = distance - i;
return i;
}
+
+framecnt_t
+CubicInterpolation::interpolate (int channel, framecnt_t nframes, Sample *input, Sample *output)
+{
+ // index in the input buffers
+ framecnt_t i = 0;
+
+ double acceleration;
+ double distance = 0.0;
+
+ if (_speed != _target_speed) {
+ acceleration = _target_speed - _speed;
+ } else {
+ acceleration = 0.0;
+ }
+
+ distance = phase[channel];
+
+ 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 (framecnt_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;
+ inm1 = input[i];
+ }
+
+ i = floor(distance);
+ phase[channel] = distance - floor(distance);
+
+ } else {
+ /* used to calculate play-distance with acceleration (silent roll)
+ * (use same algorithm as real playback for identical rounding/floor'ing)
+ */
+ for (framecnt_t outsample = 0; outsample < nframes; ++outsample) {
+ distance += _speed + acceleration;
+ }
+ i = floor(distance);
+ }
+
+ return i;
+}