diff options
| -rw-r--r-- | Alc/effects/equalizer.c | 104 |
1 files changed, 43 insertions, 61 deletions
diff --git a/Alc/effects/equalizer.c b/Alc/effects/equalizer.c index 8be689d2..7df15380 100644 --- a/Alc/effects/equalizer.c +++ b/Alc/effects/equalizer.c @@ -72,19 +72,19 @@ * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */ -/* The maximum number of sample frames per update. */ -#define MAX_UPDATE_SAMPLES 256 - typedef struct ALequalizerState { DERIVE_FROM_TYPE(ALeffectState); - /* Effect gains for each channel */ - ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS]; + struct { + /* Effect gains for each channel */ + ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; + ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; - /* Effect parameters */ - ALfilterState filter[4][MAX_EFFECT_CHANNELS]; + /* Effect parameters */ + ALfilterState filter[4]; + } Chans[MAX_EFFECT_CHANNELS]; - ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES]; + ALfloat SampleBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE]; } ALequalizerState; static ALvoid ALequalizerState_Destruct(ALequalizerState *state); @@ -98,18 +98,8 @@ DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState); static void ALequalizerState_Construct(ALequalizerState *state) { - int it, ft; - ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALequalizerState, ALeffectState, state); - - /* Initialize sample history only on filter creation to avoid */ - /* sound clicks if filter settings were changed in runtime. */ - for(it = 0; it < 4; it++) - { - for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) - ALfilterState_clear(&state->filter[it][ft]); - } } static ALvoid ALequalizerState_Destruct(ALequalizerState *state) @@ -117,8 +107,17 @@ static ALvoid ALequalizerState_Destruct(ALequalizerState *state) ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } -static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state), ALCdevice *UNUSED(device)) +static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *UNUSED(device)) { + ALsizei i, j; + + for(i = 0; i < MAX_EFFECT_CHANNELS;i++) + { + for(j = 0;j < 4;j++) + ALfilterState_clear(&state->Chans[i].filter[j]); + for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) + state->Chans[i].CurrentGains[j] = 0.0f; + } return AL_TRUE; } @@ -133,7 +132,7 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputeFirstOrderGains(&device->FOAOut, IdentityMatrixf.m[i], - slot->Params.Gain, state->Gain[i]); + slot->Params.Gain, state->Chans[i].TargetGains); /* Calculate coefficients for the each type of filter. Note that the shelf * filters' gain is for the reference frequency, which is the centerpoint @@ -141,75 +140,58 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext */ gain = maxf(sqrtf(props->Equalizer.LowGain), 0.0625f); /* Limit -24dB */ freq_mult = props->Equalizer.LowCutoff/frequency; - ALfilterState_setParams(&state->filter[0][0], ALfilterType_LowShelf, + ALfilterState_setParams(&state->Chans[0].filter[0], ALfilterType_LowShelf, gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f) ); - /* Copy the filter coefficients for the other input channels. */ - for(i = 1;i < MAX_EFFECT_CHANNELS;i++) - ALfilterState_copyParams(&state->filter[0][i], &state->filter[0][0]); gain = maxf(props->Equalizer.Mid1Gain, 0.0625f); freq_mult = props->Equalizer.Mid1Center/frequency; - ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking, + ALfilterState_setParams(&state->Chans[0].filter[1], ALfilterType_Peaking, gain, freq_mult, calc_rcpQ_from_bandwidth( freq_mult, props->Equalizer.Mid1Width ) ); - for(i = 1;i < MAX_EFFECT_CHANNELS;i++) - ALfilterState_copyParams(&state->filter[1][i], &state->filter[1][0]); gain = maxf(props->Equalizer.Mid2Gain, 0.0625f); freq_mult = props->Equalizer.Mid2Center/frequency; - ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking, + ALfilterState_setParams(&state->Chans[0].filter[2], ALfilterType_Peaking, gain, freq_mult, calc_rcpQ_from_bandwidth( freq_mult, props->Equalizer.Mid2Width ) ); - for(i = 1;i < MAX_EFFECT_CHANNELS;i++) - ALfilterState_copyParams(&state->filter[2][i], &state->filter[2][0]); gain = maxf(sqrtf(props->Equalizer.HighGain), 0.0625f); freq_mult = props->Equalizer.HighCutoff/frequency; - ALfilterState_setParams(&state->filter[3][0], ALfilterType_HighShelf, + ALfilterState_setParams(&state->Chans[0].filter[3], ALfilterType_HighShelf, gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f) ); + + /* Copy the filter coefficients for the other input channels. */ for(i = 1;i < MAX_EFFECT_CHANNELS;i++) - ALfilterState_copyParams(&state->filter[3][i], &state->filter[3][0]); + { + ALfilterState_copyParams(&state->Chans[i].filter[0], &state->Chans[0].filter[0]); + ALfilterState_copyParams(&state->Chans[i].filter[1], &state->Chans[0].filter[1]); + ALfilterState_copyParams(&state->Chans[i].filter[2], &state->Chans[0].filter[2]); + ALfilterState_copyParams(&state->Chans[i].filter[3], &state->Chans[0].filter[3]); + } } static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { - ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer; - ALsizei it, kt, ft; - ALsizei base; + ALfloat (*restrict temps)[BUFFERSIZE] = state->SampleBuffer; + ALsizei c; - for(base = 0;base < SamplesToDo;) + for(c = 0;c < MAX_EFFECT_CHANNELS;c++) { - ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base); - - for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) - ALfilterState_process(&state->filter[0][ft], Samples[0][ft], &SamplesIn[ft][base], td); - for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) - ALfilterState_process(&state->filter[1][ft], Samples[1][ft], Samples[0][ft], td); - for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) - ALfilterState_process(&state->filter[2][ft], Samples[2][ft], Samples[1][ft], td); - for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) - ALfilterState_process(&state->filter[3][ft], Samples[3][ft], Samples[2][ft], td); - - for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) - { - for(kt = 0;kt < NumChannels;kt++) - { - ALfloat gain = state->Gain[ft][kt]; - if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) - continue; - - for(it = 0;it < td;it++) - SamplesOut[kt][base+it] += gain * Samples[3][ft][it]; - } - } - - base += td; + ALfilterState_process(&state->Chans[c].filter[0], temps[0], SamplesIn[c], SamplesToDo); + ALfilterState_process(&state->Chans[c].filter[1], temps[1], temps[0], SamplesToDo); + ALfilterState_process(&state->Chans[c].filter[2], temps[2], temps[1], SamplesToDo); + ALfilterState_process(&state->Chans[c].filter[3], temps[3], temps[2], SamplesToDo); + + MixSamples(temps[3], NumChannels, SamplesOut, + state->Chans[c].CurrentGains, state->Chans[c].TargetGains, + SamplesToDo, 0, SamplesToDo + ); } } |