Add a specific output for first-order sources

9 files changed, 62 insertions, 5 deletions

diff --git a/Alc/ALc.c b/Alc/ALc.c
index 091a5473..8ca33592 100644
--- a/Alc/ALc.c
+++ b/Alc/ALc.c
@@ -2133,7 +2133,9 @@ static ALCenum UpdateDeviceParams(ALCdevice *device, const ALCint *attrList)
 
     /* Allocate extra channels for any post-filter output. */
     size = device->Dry.NumChannels * sizeof(device->Dry.Buffer[0]);
-    if(device->Hrtf || device->Uhj_Encoder || device->AmbiDecoder)
+    if(device->AmbiDecoder && bformatdec_getOrder(device->AmbiDecoder) >= 2)
+        size += (ChannelsFromDevFmt(device->FmtChans)+4) * sizeof(device->Dry.Buffer[0]);
+    else if(device->Hrtf || device->Uhj_Encoder || device->AmbiDecoder)
         size += ChannelsFromDevFmt(device->FmtChans) * sizeof(device->Dry.Buffer[0]);
     device->Dry.Buffer = al_calloc(16, size);
     if(!device->Dry.Buffer)
@@ -2157,6 +2159,22 @@ static ALCenum UpdateDeviceParams(ALCdevice *device, const ALCint *attrList)
         device->RealOut.NumChannels = device->Dry.NumChannels;
     }
 
+    if(device->AmbiDecoder && bformatdec_getOrder(device->AmbiDecoder) >= 2)
+    {
+        /* Higher-order high quality decoding requires upsampling first-order
+         * content, so make sure to mix it separately.
+         */
+        device->FOAOut.Buffer = device->RealOut.Buffer + device->RealOut.NumChannels;
+        device->FOAOut.NumChannels = 4;
+    }
+    else
+    {
+        device->FOAOut.Buffer = device->Dry.Buffer;
+        device->FOAOut.NumChannels = device->Dry.NumChannels;
+        memcpy(device->FOAOut.AmbiCoeffs, device->Dry.AmbiCoeffs,
+               sizeof(device->FOAOut.AmbiCoeffs));
+    }
+
     SetMixerFPUMode(&oldMode);
     V0(device->Backend,lock)();
     context = ATOMIC_LOAD(&device->ContextList);
diff --git a/Alc/ALu.c b/Alc/ALu.c
index 55ed84dc..b5b7de9c 100644
--- a/Alc/ALu.c
+++ b/Alc/ALu.c
@@ -513,8 +513,10 @@ ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const A
                     0.0f, -V[0]*scale,  V[1]*scale, -V[2]*scale
         );
 
+        voice->Direct.OutBuffer = Device->FOAOut.Buffer;
+        voice->Direct.OutChannels = Device->FOAOut.NumChannels;
         for(c = 0;c < num_channels;c++)
-            ComputeFirstOrderGains(Device->Dry.AmbiCoeffs, Device->Dry.NumChannels, matrix.m[c],
+            ComputeFirstOrderGains(Device->FOAOut.AmbiCoeffs, Device->FOAOut.NumChannels, matrix.m[c],
                                    DryGain, voice->Direct.Gains[c].Target);
 
         /* Rebuild the matrix, without the second- or third-order output
diff --git a/Alc/bformatdec.c b/Alc/bformatdec.c
index f5a8e8c4..37be67e4 100644
--- a/Alc/bformatdec.c
+++ b/Alc/bformatdec.c
@@ -142,6 +142,14 @@ void bformatdec_free(BFormatDec *dec)
     }
 }
 
+int bformatdec_getOrder(const struct BFormatDec *dec)
+{
+    if(dec->NumChannels > 9) return 3;
+    if(dec->NumChannels > 4) return 2;
+    if(dec->NumChannels > 1) return 1;
+    return 0;
+}
+
 void bformatdec_reset(BFormatDec *dec, const AmbDecConf *conf, ALuint chancount, ALuint srate, const ALuint chanmap[MAX_OUTPUT_CHANNELS])
 {
     const ALfloat *coeff_scale = UnitScale;
diff --git a/Alc/bformatdec.h b/Alc/bformatdec.h
index a322f41d..5f17d711 100644
--- a/Alc/bformatdec.h
+++ b/Alc/bformatdec.h
@@ -8,6 +8,7 @@ struct BFormatDec;
 
 struct BFormatDec *bformatdec_alloc();
 void bformatdec_free(struct BFormatDec *dec);
+int bformatdec_getOrder(const struct BFormatDec *dec);
 void bformatdec_reset(struct BFormatDec *dec, const struct AmbDecConf *conf, ALuint chancount, ALuint srate, const ALuint chanmap[MAX_OUTPUT_CHANNELS]);
 void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALuint OutChannels, ALfloat (*restrict InSamples)[BUFFERSIZE], ALuint SamplesToDo);
 
diff --git a/Alc/effects/compressor.c b/Alc/effects/compressor.c
index ae859793..6358f672 100644
--- a/Alc/effects/compressor.c
+++ b/Alc/effects/compressor.c
@@ -70,8 +70,11 @@ static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCdevice
         0.0f,  0.0f, scale,  0.0f,
         0.0f,  0.0f,  0.0f, scale
     );
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
     for(i = 0;i < 4;i++)
-        ComputeFirstOrderGains(device->Dry.AmbiCoeffs, device->Dry.NumChannels,
+        ComputeFirstOrderGains(device->FOAOut.AmbiCoeffs, device->FOAOut.NumChannels,
                                matrix.m[i], slot->Gain, state->Gain[i]);
 }
 
diff --git a/Alc/effects/equalizer.c b/Alc/effects/equalizer.c
index 479a8536..ffc9830f 100644
--- a/Alc/effects/equalizer.c
+++ b/Alc/effects/equalizer.c
@@ -110,8 +110,11 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *
         0.0f, 0.0f, gain, 0.0f,
         0.0f, 0.0f, 0.0f, gain
     );
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
     for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
-        ComputeFirstOrderGains(device->Dry.AmbiCoeffs, device->Dry.NumChannels,
+        ComputeFirstOrderGains(device->FOAOut.AmbiCoeffs, device->FOAOut.NumChannels,
                                matrix.m[i], slot->Gain, state->Gain[i]);
 
     /* Calculate coefficients for the each type of filter. Note that the shelf
diff --git a/Alc/effects/modulator.c b/Alc/effects/modulator.c
index 4183b908..c283d4b0 100644
--- a/Alc/effects/modulator.c
+++ b/Alc/effects/modulator.c
@@ -129,8 +129,11 @@ static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCdevice *
         0.0f,  0.0f, scale,  0.0f,
         0.0f,  0.0f,  0.0f, scale
     );
+
+    STATIC_CAST(ALeffectState,state)->OutBuffer = Device->FOAOut.Buffer;
+    STATIC_CAST(ALeffectState,state)->OutChannels = Device->FOAOut.NumChannels;
     for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
-        ComputeFirstOrderGains(Device->Dry.AmbiCoeffs, Device->Dry.NumChannels,
+        ComputeFirstOrderGains(Device->FOAOut.AmbiCoeffs, Device->FOAOut.NumChannels,
                                matrix.m[i], Slot->Gain, state->Gain[i]);
 }
 
diff --git a/Alc/panning.c b/Alc/panning.c
index 16c84b6d..401d29d6 100644
--- a/Alc/panning.c
+++ b/Alc/panning.c
@@ -646,6 +646,16 @@ ALvoid aluInitPanning(ALCdevice *device)
         TRACE("Enabling %s-band ambisonic decoder\n", (conf.FreqBands==1)?"single":"dual");
         bformatdec_reset(device->AmbiDecoder, &conf, count, device->Frequency, speakermap);
         ambdec_deinit(&conf);
+
+        if(bformatdec_getOrder(device->AmbiDecoder) >= 2)
+        {
+            memset(device->FOAOut.AmbiCoeffs, 0, sizeof(device->FOAOut.AmbiCoeffs));
+            device->FOAOut.AmbiCoeffs[0][0] = 1.0f;
+            device->FOAOut.AmbiCoeffs[1][1] = 1.0f;
+            device->FOAOut.AmbiCoeffs[2][2] = 1.0f;
+            device->FOAOut.AmbiCoeffs[3][3] = 1.0f;
+        }
+
         return;
 
     ambi_fail:
diff --git a/OpenAL32/Include/alMain.h b/OpenAL32/Include/alMain.h
index c71e844f..b870059b 100644
--- a/OpenAL32/Include/alMain.h
+++ b/OpenAL32/Include/alMain.h
@@ -507,6 +507,15 @@ struct ALCdevice_struct
         ALuint NumChannels;
     } Dry;
 
+    /* First-order ambisonics output, to be upsampled to the dry buffer if different. */
+    struct {
+        /* Ambisonic coefficients for mixing. */
+        ChannelConfig AmbiCoeffs[MAX_OUTPUT_CHANNELS];
+
+        ALfloat (*Buffer)[BUFFERSIZE];
+        ALuint NumChannels;
+    } FOAOut;
+
     /* Virtual output, to be post-processed to the real output. */
     struct {
         ALfloat (*Buffer)[BUFFERSIZE];