X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=libs%2Fardour%2Finterpolation.cc;h=20ab584885e7f19dd6bd348c2648b5d429c08c1b;hb=86d927b4ddbcedd1d6c120b1176aaef7352773cd;hp=7aece6453cf2623261d70b9f945e7bf99cd64190;hpb=f284d28d5306114e9badc9077835683e541420e0;p=ardour.git

diff --git a/libs/ardour/interpolation.cc b/libs/ardour/interpolation.cc
index 7aece6453c..20ab584885 100644
--- a/libs/ardour/interpolation.cc
+++ b/libs/ardour/interpolation.cc
@@ -1,43 +1,136 @@
 #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;
+	// index in the input buffers
+	framecnt_t   i = 0;
+
+	double acceleration;
+	double distance = 0.0;
+
+	if (_speed != _target_speed) {
+		acceleration = _target_speed - _speed;
 	} else {
-		phi_delta = 0;
+		acceleration = 0.0;
 	}
-	
-	// 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;
+
+	distance = phase[channel];
+	for (framecnt_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) {
+		// Linearly interpolate into the output buffer
+			output[outsample] =
+				input[i] * (1.0f - fractional_phase_part) +
+				input[i+1] * fractional_phase_part;
+		}
+		distance += _speed + acceleration;
 	}
 
-	last_phase = (phase & fractional_part_mask);
-	
-	// playback distance
+	i = floor(distance);
+	phase[channel] = distance - floor(distance);
+
 	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];
+	    }
+
+    } else {
+
+	    /* not sure that this is ever utilized - it implies that one of the input/output buffers is missing */
+
+	    for (framecnt_t outsample = 0; outsample < nframes; ++outsample) {
+		    distance += _speed + acceleration;
+	    }
+    }
+
+    i = floor(distance);
+    phase[channel] = distance - floor(distance);
+
+    return i;
+}