From b9e192b78a384ff13d87c606502373725042509c Mon Sep 17 00:00:00 2001
From: Chris Robinson <chris.kcat@gmail.com>
Date: Thu, 5 Nov 2015 09:42:08 -0800
Subject: Implement a band-limited sinc resampler

This is essentially a 12-point sinc resampler, unless it's resampling to a rate
higher than the output, at which point it will vary between 12 and 24 points
and do anti-aliasing to avoid/reduce frequencies going over nyquist.

Code provided by Christopher Fitzgerald.
---
 Alc/ALu.c | 74 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 74 insertions(+)

(limited to 'Alc/ALu.c')

diff --git a/Alc/ALu.c b/Alc/ALu.c
index ba0a8bd4..b566dc48 100644
--- a/Alc/ALu.c
+++ b/Alc/ALu.c
@@ -173,6 +173,78 @@ static inline aluVector aluMatrixVector(const aluMatrix *mtx, const aluVector *v
 }
 
 
+/* Prepares the interpolator for a given rate (determined by increment).  A
+ * result of AL_FALSE indicates that the filter output will completely cut
+ * the input signal.
+ *
+ * With a bit of work, and a trade of memory for CPU cost, this could be
+ * modified for use with an interpolated increment for buttery-smooth pitch
+ * changes.
+ */
+static ALboolean BsincPrepare(const ALuint increment, BsincState *state)
+{
+    static const ALfloat scaleBase = 0.1510579f, scaleRange = 1.177937f;
+    static const ALuint m[BSINC_SCALE_COUNT] = { 24, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 12 };
+    static const ALuint to[4][BSINC_SCALE_COUNT] = {
+        { 0, 24, 408, 792, 1176, 1560, 1944, 2328, 2648, 2968, 3288, 3544, 3800, 4056, 4248, 4440 },
+        { 4632, 5016, 5400, 5784, 6168, 6552, 6936, 7320, 7640, 7960, 8280, 8536, 8792, 9048, 9240, 0 },
+        { 0, 9432, 9816, 10200, 10584, 10968, 11352, 11736, 12056, 12376, 12696, 12952, 13208, 13464, 13656, 13848 },
+        { 14040, 14424, 14808, 15192, 15576, 15960, 16344, 16728, 17048, 17368, 17688, 17944, 18200, 18456, 18648, 0 }
+    };
+    static const ALuint tm[2][BSINC_SCALE_COUNT] = {
+        { 0, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 12 },
+        { 24, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 0 }
+    };
+    ALfloat sf;
+    ALuint si, pi;
+    ALboolean uncut = AL_TRUE;
+
+    if(increment > FRACTIONONE)
+    {
+        sf = (ALfloat)FRACTIONONE / increment;
+        if(sf < scaleBase)
+        {
+            /* Signal has been completely cut.  The return result can be used
+             * to skip the filter (and output zeros) as an optimization.
+             */
+            sf = 0.0f;
+            si = 0;
+            uncut = AL_FALSE;
+        }
+        else
+        {
+            sf = (BSINC_SCALE_COUNT - 1) * (sf - scaleBase) * scaleRange;
+            si = fastf2u(sf);
+            /* The interpolation factor is fit to this diagonally-symmetric
+             * curve to reduce the transition ripple caused by interpolating
+             * different scales of the sinc function.
+             */
+            sf = 1.0f - cosf(asinf(sf - si));
+        }
+    }
+    else
+    {
+        sf = 0.0f;
+        si = BSINC_SCALE_COUNT - 1;
+    }
+
+    state->sf = sf;
+    state->m = m[si];
+    state->l = -(ALint)((m[si] / 2) - 1);
+    /* The CPU cost of this table re-mapping could be traded for the memory
+     * cost of a complete table map (1024 elements large).
+     */
+    for(pi = 0;pi < BSINC_PHASE_COUNT;pi++)
+    {
+        state->coeffs[pi].filter  = &bsincTab[to[0][si] + tm[0][si]*pi];
+        state->coeffs[pi].scDelta = &bsincTab[to[1][si] + tm[1][si]*pi];
+        state->coeffs[pi].phDelta = &bsincTab[to[2][si] + tm[0][si]*pi];
+        state->coeffs[pi].spDelta = &bsincTab[to[3][si] + tm[1][si]*pi];
+    }
+    return uncut;
+}
+
+
 /* Calculates the fade time from the changes in gain and listener to source
  * angle between updates. The result is a the time, in seconds, for the
  * transition to complete.
@@ -413,6 +485,7 @@ ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const A
                 voice->Step = MAX_PITCH<<FRACTIONBITS;
             else
                 voice->Step = maxi(fastf2i(Pitch*FRACTIONONE + 0.5f), 1);
+            BsincPrepare(voice->Step, &voice->SincState);
 
             Channels = ALBuffer->FmtChannels;
             break;
@@ -1023,6 +1096,7 @@ ALvoid CalcSourceParams(ALvoice *voice, const ALsource *ALSource, const ALCconte
                 voice->Step = MAX_PITCH<<FRACTIONBITS;
             else
                 voice->Step = maxi(fastf2i(Pitch*FRACTIONONE + 0.5f), 1);
+            BsincPrepare(voice->Step, &voice->SincState);
 
             break;
         }
-- 
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