Rename ComputeBFormatGains to ComputeFirstOrderGains
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@ -663,7 +663,7 @@ ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const A
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MixGains *gains = voice->Direct.Gains[c];
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ALfloat Target[MAX_OUTPUT_CHANNELS];
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ComputeBFormatGains(Device->AmbiCoeffs, Device->NumChannels, matrix.m[c], DryGain, Target);
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ComputeFirstOrderGains(Device->AmbiCoeffs, Device->NumChannels, matrix.m[c], DryGain, Target);
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for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
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gains[i].Target = Target[i];
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}
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@ -698,8 +698,8 @@ ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const A
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ALfloat Target[MAX_OUTPUT_CHANNELS];
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ALuint j;
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ComputeBFormatGains(Slot->AmbiCoeffs, Slot->NumChannels,
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matrix.m[c], WetGain[i], Target);
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ComputeFirstOrderGains(Slot->AmbiCoeffs, Slot->NumChannels,
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matrix.m[c], WetGain[i], Target);
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for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
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gains[j].Target = Target[j];
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}
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@ -71,8 +71,8 @@ static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCdevice
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0.0f, 0.0f, 0.0f, scale
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);
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for(i = 0;i < 4;i++)
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ComputeBFormatGains(device->AmbiCoeffs, device->NumChannels,
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matrix.m[i], slot->Gain, state->Gain[i]);
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ComputeFirstOrderGains(device->AmbiCoeffs, device->NumChannels,
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matrix.m[i], slot->Gain, state->Gain[i]);
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}
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static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
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@ -111,8 +111,8 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *
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0.0f, 0.0f, 0.0f, gain
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);
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for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
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ComputeBFormatGains(device->AmbiCoeffs, device->NumChannels,
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matrix.m[i], slot->Gain, state->Gain[i]);
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ComputeFirstOrderGains(device->AmbiCoeffs, device->NumChannels,
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matrix.m[i], slot->Gain, state->Gain[i]);
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/* Calculate coefficients for the each type of filter. Note that the shelf
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* filters' gain is for the reference frequency, which is the centerpoint
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@ -130,8 +130,8 @@ static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCdevice *
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0.0f, 0.0f, 0.0f, scale
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);
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for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
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ComputeBFormatGains(Device->AmbiCoeffs, Device->NumChannels,
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matrix.m[i], Slot->Gain, state->Gain[i]);
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ComputeFirstOrderGains(Device->AmbiCoeffs, Device->NumChannels,
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matrix.m[i], Slot->Gain, state->Gain[i]);
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}
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static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
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@ -154,7 +154,7 @@ void ComputePanningGains(const ChannelConfig *chancoeffs, ALuint numchans, const
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gains[i] = 0.0f;
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}
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void ComputeBFormatGains(const ChannelConfig *chancoeffs, ALuint numchans, const ALfloat mtx[4], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS])
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void ComputeFirstOrderGains(const ChannelConfig *chancoeffs, ALuint numchans, const ALfloat mtx[4], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS])
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{
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ALuint i, j;
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@ -322,13 +322,13 @@ void ComputeAmbientGains(const ChannelConfig *chancoeffs, ALuint numchans, ALflo
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void ComputePanningGains(const ChannelConfig *chancoeffs, ALuint numchans, const ALfloat coeffs[MAX_AMBI_COEFFS], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
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/**
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* ComputeBFormatGains
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* ComputeFirstOrderGains
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*
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* Sets channel gains for a given (first-order) B-Format input channel. The
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* matrix is a 1x4 'slice' of the rotation matrix for the given channel used to
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* orient the soundfield.
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* Sets channel gains for a first-order ambisonics input channel. The matrix is
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* a 1x4 'slice' of a transform matrix for the input channel, used to scale and
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* orient the sound samples.
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*/
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void ComputeBFormatGains(const ChannelConfig *chancoeffs, ALuint numchans, const ALfloat mtx[4], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
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void ComputeFirstOrderGains(const ChannelConfig *chancoeffs, ALuint numchans, const ALfloat mtx[4], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
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ALvoid UpdateContextSources(ALCcontext *context);
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