Add a method to apply an HF scale without band-splitting

4 files changed, 52 insertions, 25 deletions

diff --git a/Alc/effects/reverb.cpp b/Alc/effects/reverb.cpp
index 24107dc7..086ffed2 100644
--- a/Alc/effects/reverb.cpp
+++ b/Alc/effects/reverb.cpp
@@ -424,32 +424,23 @@ struct ReverbState final : public EffectState {
         for(ALsizei c{0};c < NUM_LINES;c++)
         {
             /* Apply scaling to the B-Format's HF response to "upsample" it to
-             * higher-order output. Use the early buffer to store the split
-             * signal since it's not needed anymore.
+             * higher-order output.
              */
             const ALfloat hfscale{(c==0) ? mOrderScales[0] : mOrderScales[1]};
-            ALfloat (&hfbuf)[BUFFERSIZE] = mEarlyBuffer[0];
-            ALfloat (&lfbuf)[BUFFERSIZE] = mEarlyBuffer[1];
+            mAmbiSplitter[0][c].applyHfScale(mTempSamples[c], hfscale, todo);
 
-            mAmbiSplitter[0][c].process(hfbuf, lfbuf, mTempSamples[c], todo);
-            MixRowSamples(lfbuf, &hfscale, &hfbuf, 1, 0, todo);
-
-            MixSamples(lfbuf, numOutput, samplesOut, mEarly.CurrentGain[c], mEarly.PanGain[c],
-                todo, 0, todo);
+            MixSamples(mTempSamples[c], numOutput, samplesOut, mEarly.CurrentGain[c],
+                mEarly.PanGain[c], todo, 0, todo);
         }
 
         ConvertA2B(mTempSamples, mLateBuffer, todo);
         for(ALsizei c{0};c < NUM_LINES;c++)
         {
             const ALfloat hfscale{(c==0) ? mOrderScales[0] : mOrderScales[1]};
-            ALfloat (&hfbuf)[BUFFERSIZE] = mLateBuffer[0];
-            ALfloat (&lfbuf)[BUFFERSIZE] = mLateBuffer[1];
+            mAmbiSplitter[1][c].applyHfScale(mTempSamples[c], hfscale, todo);
 
-            mAmbiSplitter[1][c].process(hfbuf, lfbuf, mTempSamples[c], todo);
-            MixRowSamples(lfbuf, &hfscale, &hfbuf, 1, 0, todo);
-
-            MixSamples(lfbuf, numOutput, samplesOut, mLate.CurrentGain[c], mLate.PanGain[c], todo,
-                0, todo);
+            MixSamples(mTempSamples[c], numOutput, samplesOut, mLate.CurrentGain[c],
+                mLate.PanGain[c], todo, 0, todo);
         }
     }
 
diff --git a/Alc/filters/splitter.cpp b/Alc/filters/splitter.cpp
index 004f8215..58a5029b 100644
--- a/Alc/filters/splitter.cpp
+++ b/Alc/filters/splitter.cpp
@@ -25,7 +25,7 @@ void BandSplitterR<Real>::init(Real f0norm)
 }
 
 template<typename Real>
-void BandSplitterR<Real>::process(Real *hpout, Real *lpout, const Real *input, int count)
+void BandSplitterR<Real>::process(Real *hpout, Real *lpout, const Real *input, const int count)
 {
     ASSUME(count > 0);
 
@@ -60,6 +60,40 @@ void BandSplitterR<Real>::process(Real *hpout, Real *lpout, const Real *input, i
     this->ap_z1 = ap_z1;
 }
 
+template<typename Real>
+void BandSplitterR<Real>::applyHfScale(Real *RESTRICT samples, const Real hfscale, const int count)
+{
+    ASSUME(count > 0);
+
+    const Real ap_coeff{this->coeff};
+    const Real lp_coeff{this->coeff*0.5f + 0.5f};
+    Real lp_z1{this->lp_z1};
+    Real lp_z2{this->lp_z2};
+    Real ap_z1{this->ap_z1};
+    auto proc_sample = [hfscale,ap_coeff,lp_coeff,&lp_z1,&lp_z2,&ap_z1](const Real in) noexcept -> Real
+    {
+        /* Low-pass sample processing. */
+        Real d{(in - lp_z1) * lp_coeff};
+        Real lp_y{lp_z1 + d};
+        lp_z1 = lp_y + d;
+
+        d = (lp_y - lp_z2) * lp_coeff;
+        lp_y = lp_z2 + d;
+        lp_z2 = lp_y + d;
+
+        /* All-pass sample processing. */
+        Real ap_y{in*ap_coeff + ap_z1};
+        ap_z1 = in - ap_y*ap_coeff;
+
+        /* High-pass generated from removing low-passed output. */
+        return (ap_y-lp_y)*hfscale + lp_y;
+    };
+    std::transform(samples, samples+count, samples, proc_sample);
+    this->lp_z1 = lp_z1;
+    this->lp_z2 = lp_z2;
+    this->ap_z1 = ap_z1;
+}
+
 template class BandSplitterR<float>;
 template class BandSplitterR<double>;
 
diff --git a/Alc/filters/splitter.h b/Alc/filters/splitter.h
index f7ea608b..83281366 100644
--- a/Alc/filters/splitter.h
+++ b/Alc/filters/splitter.h
@@ -16,7 +16,8 @@ class BandSplitterR {
 public:
     void init(Real f0norm);
     void clear() noexcept { lp_z1 = lp_z2 = ap_z1 = 0.0f; }
-    void process(Real *hpout, Real *lpout, const Real *input, int count);
+    void process(Real *hpout, Real *lpout, const Real *input, const int count);
+    void applyHfScale(Real *RESTRICT samples, const Real hfscale, const int count);
 };
 using BandSplitter = BandSplitterR<float>;
 
diff --git a/Alc/mixvoice.cpp b/Alc/mixvoice.cpp
index 3366aa61..756b69ee 100644
--- a/Alc/mixvoice.cpp
+++ b/Alc/mixvoice.cpp
@@ -612,13 +612,14 @@ void MixSource(ALvoice *voice, const ALuint SourceID, ALCcontext *Context, const
                  * be desirable.
                  */
                 const ALfloat hfscale{(chan==0) ? voice->AmbiScales[0] : voice->AmbiScales[1]};
-                ALfloat (&hfbuf)[BUFFERSIZE] = Device->FilteredData;
-                ALfloat (&lfbuf)[BUFFERSIZE] = Device->ResampledData;
-
-                voice->AmbiSplitter[chan].process(hfbuf, lfbuf, ResampledData, DstBufferSize);
-                MixRowSamples(lfbuf, &hfscale, &hfbuf, 1, 0, DstBufferSize);
-
-                ResampledData = lfbuf;
+                /* Beware the evil const_cast. It's safe since it's pointing to
+                 * either SrcData or Device->ResampledData (both non-const),
+                 * but the resample method takes its input as const float* and
+                 * may return it without copying to output, making it currently
+                 * unavoidable.
+                 */
+                voice->AmbiSplitter[chan].applyHfScale(const_cast<ALfloat*>(ResampledData),
+                    hfscale, DstBufferSize);
             }
 
             /* Now filter and mix to the appropriate outputs. */