diff options
author | Chris Robinson <[email protected]> | 2015-11-01 04:43:55 -0800 |
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committer | Chris Robinson <[email protected]> | 2015-11-01 05:41:06 -0800 |
commit | c57f57192067e2d68cfe4ab0fc9479d2453bfbda (patch) | |
tree | fb9dabd6296c87ff72f4271774ae9951898d9227 /Alc | |
parent | f094d94608e00b1b08bd8c607d16651072323bb5 (diff) |
Pass in the Q parameter for setting the filter parameters
Also better handle the peaking filter gain.
Diffstat (limited to 'Alc')
-rw-r--r-- | Alc/ALu.c | 40 | ||||
-rw-r--r-- | Alc/effects/distortion.c | 6 | ||||
-rw-r--r-- | Alc/effects/echo.c | 10 | ||||
-rw-r--r-- | Alc/effects/equalizer.c | 34 | ||||
-rw-r--r-- | Alc/effects/reverb.c | 7 |
5 files changed, 60 insertions, 37 deletions
@@ -667,18 +667,20 @@ ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const A { ALfloat hfscale = ALSource->Direct.HFReference / Frequency; ALfloat lfscale = ALSource->Direct.LFReference / Frequency; + DryGainHF = maxf(DryGainHF, 0.0001f); + DryGainLF = maxf(DryGainLF, 0.0001f); for(c = 0;c < num_channels;c++) { voice->Direct.Filters[c].ActiveType = AF_None; if(DryGainHF != 1.0f) voice->Direct.Filters[c].ActiveType |= AF_LowPass; if(DryGainLF != 1.0f) voice->Direct.Filters[c].ActiveType |= AF_HighPass; ALfilterState_setParams( - &voice->Direct.Filters[c].LowPass, ALfilterType_HighShelf, DryGainHF, - hfscale, 0.0f + &voice->Direct.Filters[c].LowPass, ALfilterType_HighShelf, + DryGainHF, hfscale, calc_rcpQ_from_slope(DryGainHF, 0.75f) ); ALfilterState_setParams( - &voice->Direct.Filters[c].HighPass, ALfilterType_LowShelf, DryGainLF, - lfscale, 0.0f + &voice->Direct.Filters[c].HighPass, ALfilterType_LowShelf, + DryGainLF, lfscale, calc_rcpQ_from_slope(DryGainLF, 0.75f) ); } } @@ -686,18 +688,20 @@ ALvoid CalcNonAttnSourceParams(ALvoice *voice, const ALsource *ALSource, const A { ALfloat hfscale = ALSource->Send[i].HFReference / Frequency; ALfloat lfscale = ALSource->Send[i].LFReference / Frequency; + WetGainHF[i] = maxf(WetGainHF[i], 0.0001f); + WetGainLF[i] = maxf(WetGainLF[i], 0.0001f); for(c = 0;c < num_channels;c++) { voice->Send[i].Filters[c].ActiveType = AF_None; if(WetGainHF[i] != 1.0f) voice->Send[i].Filters[c].ActiveType |= AF_LowPass; if(WetGainLF[i] != 1.0f) voice->Send[i].Filters[c].ActiveType |= AF_HighPass; ALfilterState_setParams( - &voice->Send[i].Filters[c].LowPass, ALfilterType_HighShelf, WetGainHF[i], - hfscale, 0.0f + &voice->Send[i].Filters[c].LowPass, ALfilterType_HighShelf, + WetGainHF[i], hfscale, calc_rcpQ_from_slope(WetGainHF[i], 0.75f) ); ALfilterState_setParams( - &voice->Send[i].Filters[c].HighPass, ALfilterType_LowShelf, WetGainLF[i], - lfscale, 0.0f + &voice->Send[i].Filters[c].HighPass, ALfilterType_LowShelf, + WetGainLF[i], lfscale, calc_rcpQ_from_slope(WetGainLF[i], 0.75f) ); } } @@ -1139,32 +1143,36 @@ ALvoid CalcSourceParams(ALvoice *voice, const ALsource *ALSource, const ALCconte { ALfloat hfscale = ALSource->Direct.HFReference / Frequency; ALfloat lfscale = ALSource->Direct.LFReference / Frequency; + DryGainHF = maxf(DryGainHF, 0.0001f); + DryGainLF = maxf(DryGainLF, 0.0001f); voice->Direct.Filters[0].ActiveType = AF_None; if(DryGainHF != 1.0f) voice->Direct.Filters[0].ActiveType |= AF_LowPass; if(DryGainLF != 1.0f) voice->Direct.Filters[0].ActiveType |= AF_HighPass; ALfilterState_setParams( - &voice->Direct.Filters[0].LowPass, ALfilterType_HighShelf, DryGainHF, - hfscale, 0.0f + &voice->Direct.Filters[0].LowPass, ALfilterType_HighShelf, + DryGainHF, hfscale, calc_rcpQ_from_slope(DryGainHF, 0.75f) ); ALfilterState_setParams( - &voice->Direct.Filters[0].HighPass, ALfilterType_LowShelf, DryGainLF, - lfscale, 0.0f + &voice->Direct.Filters[0].HighPass, ALfilterType_LowShelf, + DryGainLF, lfscale, calc_rcpQ_from_slope(DryGainLF, 0.75f) ); } for(i = 0;i < NumSends;i++) { ALfloat hfscale = ALSource->Send[i].HFReference / Frequency; ALfloat lfscale = ALSource->Send[i].LFReference / Frequency; + WetGainHF[i] = maxf(WetGainHF[i], 0.0001f); + WetGainLF[i] = maxf(WetGainLF[i], 0.0001f); voice->Send[i].Filters[0].ActiveType = AF_None; if(WetGainHF[i] != 1.0f) voice->Send[i].Filters[0].ActiveType |= AF_LowPass; if(WetGainLF[i] != 1.0f) voice->Send[i].Filters[0].ActiveType |= AF_HighPass; ALfilterState_setParams( - &voice->Send[i].Filters[0].LowPass, ALfilterType_HighShelf, WetGainHF[i], - hfscale, 0.0f + &voice->Send[i].Filters[0].LowPass, ALfilterType_HighShelf, + WetGainHF[i], hfscale, calc_rcpQ_from_slope(WetGainHF[i], 0.75f) ); ALfilterState_setParams( - &voice->Send[i].Filters[0].HighPass, ALfilterType_LowShelf, WetGainLF[i], - lfscale, 0.0f + &voice->Send[i].Filters[0].HighPass, ALfilterType_LowShelf, + WetGainLF[i], lfscale, calc_rcpQ_from_slope(WetGainLF[i], 0.75f) ); } } diff --git a/Alc/effects/distortion.c b/Alc/effects/distortion.c index 22e05c70..221cec39 100644 --- a/Alc/effects/distortion.c +++ b/Alc/effects/distortion.c @@ -72,14 +72,16 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, ALCdevice *Devi /* Bandwidth value is constant in octaves */ bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f); ALfilterState_setParams(&state->lowpass, ALfilterType_LowPass, 1.0f, - cutoff / (frequency*4.0f), bandwidth); + cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth) + ); /* Bandpass filter */ cutoff = Slot->EffectProps.Distortion.EQCenter; /* Convert bandwidth in Hz to octaves */ bandwidth = Slot->EffectProps.Distortion.EQBandwidth / (cutoff * 0.67f); ALfilterState_setParams(&state->bandpass, ALfilterType_BandPass, 1.0f, - cutoff / (frequency*4.0f), bandwidth); + cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth) + ); ComputeAmbientGains(Device, Slot->Gain, state->Gain); } diff --git a/Alc/effects/echo.c b/Alc/effects/echo.c index b852a096..f5a53c36 100644 --- a/Alc/effects/echo.c +++ b/Alc/effects/echo.c @@ -85,8 +85,7 @@ static ALvoid ALechoState_update(ALechoState *state, ALCdevice *Device, const AL { ALfloat pandir[3] = { 0.0f, 0.0f, 0.0f }; ALuint frequency = Device->Frequency; - ALfloat gain = Slot->Gain; - ALfloat lrpan; + ALfloat gain, lrpan; state->Tap[0].delay = fastf2u(Slot->EffectProps.Echo.Delay * frequency) + 1; state->Tap[1].delay = fastf2u(Slot->EffectProps.Echo.LRDelay * frequency); @@ -96,9 +95,12 @@ static ALvoid ALechoState_update(ALechoState *state, ALCdevice *Device, const AL state->FeedGain = Slot->EffectProps.Echo.Feedback; + gain = minf(1.0f - Slot->EffectProps.Echo.Damping, 0.01f); ALfilterState_setParams(&state->Filter, ALfilterType_HighShelf, - 1.0f - Slot->EffectProps.Echo.Damping, - LOWPASSFREQREF/frequency, 0.0f); + gain, LOWPASSFREQREF/frequency, + calc_rcpQ_from_slope(gain, 0.75f)); + + gain = Slot->Gain; /* First tap panning */ pandir[0] = -lrpan; diff --git a/Alc/effects/equalizer.c b/Alc/effects/equalizer.c index 6a5ed549..244667ab 100644 --- a/Alc/effects/equalizer.c +++ b/Alc/effects/equalizer.c @@ -93,29 +93,37 @@ static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state), static ALvoid ALequalizerState_update(ALequalizerState *state, ALCdevice *device, const ALeffectslot *slot) { ALfloat frequency = (ALfloat)device->Frequency; + ALfloat gain, freq_mult; ComputeAmbientGains(device, slot->Gain, state->Gain); - /* Calculate coefficients for the each type of filter */ + /* Calculate coefficients for the each type of filter. Note that the shelf + * filters' gain is for the reference frequency, which is the centerpoint + * of the transition band. + */ + gain = sqrtf(slot->EffectProps.Equalizer.LowGain); + freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency; ALfilterState_setParams(&state->filter[0], ALfilterType_LowShelf, - sqrtf(slot->EffectProps.Equalizer.LowGain), - slot->EffectProps.Equalizer.LowCutoff/frequency, - 0.0f); + gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f) + ); + gain = slot->EffectProps.Equalizer.Mid1Gain; + freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency; ALfilterState_setParams(&state->filter[1], ALfilterType_Peaking, - sqrtf(slot->EffectProps.Equalizer.Mid1Gain), - slot->EffectProps.Equalizer.Mid1Center/frequency, - slot->EffectProps.Equalizer.Mid1Width); + gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid1Width) + ); + gain = slot->EffectProps.Equalizer.Mid2Gain; + freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency; ALfilterState_setParams(&state->filter[2], ALfilterType_Peaking, - sqrtf(slot->EffectProps.Equalizer.Mid2Gain), - slot->EffectProps.Equalizer.Mid2Center/frequency, - slot->EffectProps.Equalizer.Mid2Width); + gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid2Width) + ); + gain = sqrtf(slot->EffectProps.Equalizer.HighGain); + freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency; ALfilterState_setParams(&state->filter[3], ALfilterType_HighShelf, - sqrtf(slot->EffectProps.Equalizer.HighGain), - slot->EffectProps.Equalizer.HighCutoff/frequency, - 0.0f); + gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f) + ); } static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c index 6c3f2691..71e12b33 100644 --- a/Alc/effects/reverb.c +++ b/Alc/effects/reverb.c @@ -1112,6 +1112,7 @@ static ALvoid ALreverbState_update(ALreverbState *State, ALCdevice *Device, cons const ALeffectProps *props = &Slot->EffectProps; ALuint frequency = Device->Frequency; ALfloat lfscale, hfscale, hfRatio; + ALfloat gainlf, gainhf; ALfloat cw, x, y; if(Slot->EffectType == AL_EFFECT_EAXREVERB && !EmulateEAXReverb) @@ -1121,11 +1122,13 @@ static ALvoid ALreverbState_update(ALreverbState *State, ALCdevice *Device, cons // Calculate the master filters hfscale = props->Reverb.HFReference / frequency; + gainhf = maxf(props->Reverb.GainHF, 0.0001f); ALfilterState_setParams(&State->LpFilter, ALfilterType_HighShelf, - props->Reverb.GainHF, hfscale, 0.0f); + gainhf, hfscale, calc_rcpQ_from_slope(gainhf, 0.75f)); lfscale = props->Reverb.LFReference / frequency; + gainlf = maxf(props->Reverb.GainLF, 0.0001f); ALfilterState_setParams(&State->HpFilter, ALfilterType_LowShelf, - props->Reverb.GainLF, lfscale, 0.0f); + gainlf, lfscale, calc_rcpQ_from_slope(gainlf, 0.75f)); // Update the modulator line. UpdateModulator(props->Reverb.ModulationTime, props->Reverb.ModulationDepth, |