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Mix B-Format sources directly to the dry buffer

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Now the only thing that utilizes FOAOut is reverb output.
This commit is contained in:
Chris Robinson
2019-02-21 17:48:08 -08:00
parent 1eea3cb2d1
commit 7a9d67934f
2 changed files with 34 additions and 21 deletions
Download Patch File Download Diff File Expand all files Collapse all files
Alc/alu.cpp
+15 -19
View File
@@ -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);
}
}
}
Alc/mixvoice.cpp
+19 -2
View File
@@ -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,