Mix B-Format sources directly to the dry buffer

Now the only thing that utilizes FOAOut is reverb output.

2 files changed, 34 insertions, 21 deletions

diff --git a/Alc/alu.cpp b/Alc/alu.cpp
index 0eac5547..95889626 100644
--- a/Alc/alu.cpp
+++ b/Alc/alu.cpp
@@ -630,11 +630,8 @@ void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev
                 voice->Flags |= VOICE_HAS_NFC;
             }
 
-            /* Always render B-Format sources to the FOA output, to ensure
-             * smooth changes if it switches between panned and unpanned.
-             */
-            voice->Direct.Buffer = Device->FOAOut.Buffer;
-            voice->Direct.Channels = Device->FOAOut.NumChannels;
+            voice->Direct.Buffer = Device->Dry.Buffer;
+            voice->Direct.Channels = Device->Dry.NumChannels;
 
             /* A scalar of 1.5 for plain stereo results in +/-60 degrees being
              * moved to +/-90 degrees for direct right and left speaker
@@ -646,7 +643,7 @@ void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev
 
             /* NOTE: W needs to be scaled due to FuMa normalization. */
             const ALfloat &scale0 = AmbiScale::FromFuMa[0];
-            ComputePanGains(&Device->FOAOut, coeffs, DryGain*scale0,
+            ComputePanGains(&Device->Dry, coeffs, DryGain*scale0,
                 voice->Direct.Params[0].Gains.Target);
             for(ALsizei i{0};i < NumSends;i++)
             {
@@ -697,26 +694,25 @@ void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev
             const ALfloat &yscale = AmbiScale::FromFuMa[1];
             const ALfloat &zscale = AmbiScale::FromFuMa[2];
             const ALfloat &xscale = AmbiScale::FromFuMa[3];
-            const alu::Matrix matrix{
-            //    ACN0          ACN1          ACN2          ACN3
-                wscale,         0.0f,         0.0f,         0.0f, // FuMa W
-                  0.0f, -N[0]*xscale,  N[1]*xscale, -N[2]*xscale, // FuMa X
-                  0.0f,  U[0]*yscale, -U[1]*yscale,  U[2]*yscale, // FuMa Y
-                  0.0f, -V[0]*zscale,  V[1]*zscale, -V[2]*zscale  // FuMa Z
+            const ALfloat matrix[4][MAX_AMBI_CHANNELS]{
+            //      ACN0          ACN1          ACN2          ACN3
+                { wscale,         0.0f,         0.0f,         0.0f }, // FuMa W
+                {   0.0f, -N[0]*xscale,  N[1]*xscale, -N[2]*xscale }, // FuMa X
+                {   0.0f,  U[0]*yscale, -U[1]*yscale,  U[2]*yscale }, // FuMa Y
+                {   0.0f, -V[0]*zscale,  V[1]*zscale, -V[2]*zscale }  // FuMa Z
             };
 
-            voice->Direct.Buffer = Device->FOAOut.Buffer;
-            voice->Direct.Channels = Device->FOAOut.NumChannels;
+            voice->Direct.Buffer = Device->Dry.Buffer;
+            voice->Direct.Channels = Device->Dry.NumChannels;
             for(ALsizei c{0};c < num_channels;c++)
-                ComputePanGains(&Device->FOAOut, matrix[c].data(), DryGain,
-                                voice->Direct.Params[c].Gains.Target);
+                ComputePanGains(&Device->Dry, matrix[c], DryGain,
+                    voice->Direct.Params[c].Gains.Target);
             for(ALsizei i{0};i < NumSends;i++)
             {
                 if(const ALeffectslot *Slot{SendSlots[i]})
                     for(ALsizei c{0};c < num_channels;c++)
-                        ComputePanningGainsBF(Slot->ChanMap, Slot->WetBuffer.size(), matrix[c].data(),
-                            WetGain[i], voice->Send[i].Params[c].Gains.Target
-                        );
+                        ComputePanningGainsBF(Slot->ChanMap, Slot->WetBuffer.size(), matrix[c],
+                            WetGain[i], voice->Send[i].Params[c].Gains.Target);
             }
         }
     }
diff --git a/Alc/mixvoice.cpp b/Alc/mixvoice.cpp
index a0518ed5..78ceea26 100644
--- a/Alc/mixvoice.cpp
+++ b/Alc/mixvoice.cpp
@@ -539,13 +539,30 @@ ALboolean MixSource(ALvoice *voice, const ALuint SourceID, ALCcontext *Context,
 
             /* Store the last source samples used for next time. */
             std::copy_n(&SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
-                        voice->PrevSamples[chan].size(), std::begin(voice->PrevSamples[chan]));
+                voice->PrevSamples[chan].size(), std::begin(voice->PrevSamples[chan]));
 
-            /* Now resample, then filter and mix to the appropriate outputs. */
+            /* Resample, then apply ambisonic upsampling as needed. */
             const ALfloat *ResampledData{Resample(&voice->ResampleState,
                 &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
                 Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
             )};
+            if((voice->Flags&VOICE_IS_AMBISONIC))
+            {
+                /* TODO: Does not properly handle HOA sources. Currently only
+                 * first-order sources are possible, but in the future it would
+                 * be desirable.
+                 */
+                const ALfloat hfscale{(chan==0) ? voice->AmbiScales[0] : voice->AmbiScales[1]};
+                ALfloat (&hfbuf)[BUFFERSIZE] = Device->TempBuffer[SOURCE_DATA_BUF];
+                ALfloat (&lfbuf)[BUFFERSIZE] = Device->TempBuffer[RESAMPLED_BUF];
+
+                voice->AmbiSplitter[chan].process(hfbuf, lfbuf, ResampledData, DstBufferSize);
+                MixRowSamples(lfbuf, &hfscale, &hfbuf, 1, 0, DstBufferSize);
+
+                ResampledData = lfbuf;
+            }
+
+            /* Now filter and mix to the appropriate outputs. */
             {
                 DirectParams &parms = voice->Direct.Params[chan];
                 const ALfloat *samples{DoFilters(&parms.LowPass, &parms.HighPass,