diff options
| -rw-r--r-- | Alc/alu.cpp | 34 | ||||
| -rw-r--r-- | Alc/mixvoice.cpp | 21 |
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, |