diff options
author | Chris Robinson <[email protected]> | 2016-09-05 05:30:27 -0700 |
---|---|---|
committer | Chris Robinson <[email protected]> | 2016-09-05 06:08:01 -0700 |
commit | a20576bbd747bdcfef2a7e6b09e9ba3039185ae5 (patch) | |
tree | 17f4bf43a5d5f4584d587c0e85f4ecf8db151621 | |
parent | 8a64f071213f387c4675012ec1df51b9295564b3 (diff) |
Do multiple samples at once for reverb modulation
-rw-r--r-- | Alc/effects/reverb.c | 87 |
1 files changed, 49 insertions, 38 deletions
diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c index 0ffff585..3bdf9b91 100644 --- a/Alc/effects/reverb.c +++ b/Alc/effects/reverb.c @@ -1080,43 +1080,52 @@ static inline ALvoid DelayLineIn(DelayLine *Delay, ALuint offset, ALfloat in) Delay->Line[offset&Delay->Mask] = in; } -// Given an input sample, this function produces modulation for the late +// Given some input samples, this function produces modulation for the late // reverb. -static inline ALfloat EAXModulation(ALreverbState *State, ALuint offset, ALfloat in) +static void EAXModulation(ALreverbState *State, ALuint offset, ALfloat*restrict dst, const ALfloat*restrict src, ALuint todo) { ALfloat sinus, frac, fdelay; ALfloat out0, out1; - ALuint delay; + ALuint delay, i; - // Calculate the sinus rythm (dependent on modulation time and the - // sampling rate). The center of the sinus is moved to reduce the delay - // of the effect when the time or depth are low. - sinus = 1.0f - cosf(F_TAU * State->Mod.Index / State->Mod.Range); + for(i = 0;i < todo;i++) + { + /* Calculate the sinus rythm (dependent on modulation time and the + * sampling rate). The center of the sinus is moved to reduce the + * delay of the effect when the time or depth are low. + */ + sinus = 1.0f - cosf(F_TAU * State->Mod.Index / State->Mod.Range); - // Step the modulation index forward, keeping it bound to its range. - State->Mod.Index = (State->Mod.Index + 1) % State->Mod.Range; + /* Step the modulation index forward, keeping it bound to its range. */ + State->Mod.Index = (State->Mod.Index + 1) % State->Mod.Range; - // The depth determines the range over which to read the input samples - // from, so it must be filtered to reduce the distortion caused by even - // small parameter changes. - State->Mod.Filter = lerp(State->Mod.Filter, State->Mod.Depth, - State->Mod.Coeff); + /* The depth determines the range over which to read the input samples + * from, so it must be filtered to reduce the distortion caused by even + * small parameter changes. + */ + State->Mod.Filter = lerp(State->Mod.Filter, State->Mod.Depth, + State->Mod.Coeff); - // Calculate the read offset and fraction between it and the next sample. - frac = modff(State->Mod.Filter*sinus, &fdelay); - delay = fastf2u(fdelay); + /* Calculate the read offset and fraction between it and the next + * sample. + */ + frac = modff(State->Mod.Filter*sinus, &fdelay); + delay = fastf2u(fdelay); - /* Add the incoming sample to the delay line first, so a 0 delay gets the - * incoming sample. - */ - DelayLineIn(&State->Mod.Delay, offset, in); - /* Get the two samples crossed by the offset delay */ - out0 = DelayLineOut(&State->Mod.Delay, offset - delay); - out1 = DelayLineOut(&State->Mod.Delay, offset - delay - 1); - - // The output is obtained by linearly interpolating the two samples that - // were acquired above. - return lerp(out0, out1, frac); + /* Add the incoming sample to the delay line first, so a 0 delay gets + * the incoming sample. + */ + DelayLineIn(&State->Mod.Delay, offset, src[i]); + /* Get the two samples crossed by the offset delay */ + out0 = DelayLineOut(&State->Mod.Delay, offset - delay); + out1 = DelayLineOut(&State->Mod.Delay, offset - delay - 1); + offset++; + + /* The output is obtained by linearly interpolating the two samples + * that were acquired above. + */ + dst[i] = lerp(out0, out1, frac); + } } // Given some input sample, this function produces four-channel outputs for the @@ -1371,18 +1380,20 @@ static inline ALvoid EAXVerbPass(ALreverbState *State, ALuint todo, const ALfloa // Band-pass and modulate the incoming samples (use the early buffer as temp storage). ALfilterState_process(&State->LpFilter, &early[0][0], input, todo); - ALfilterState_process(&State->HpFilter, &early[MAX_UPDATE_SAMPLES/4][0], &early[0][0], todo); + ALfilterState_process(&State->HpFilter, + &early[MAX_UPDATE_SAMPLES/4][0], &early[0][0], todo + ); + // Perform any modulation on the input. + EAXModulation(State, State->Offset, + &early[MAX_UPDATE_SAMPLES*2/4][0], &early[MAX_UPDATE_SAMPLES/4][0], + todo + ); + // Feed the initial delay line. for(i = 0;i < todo;i++) - { - ALfloat sample = early[(MAX_UPDATE_SAMPLES/4)+(i>>2)][i&3]; - - // Perform any modulation on the input. - sample = EAXModulation(State, State->Offset+i, sample); - - // Feed the initial delay line. - DelayLineIn(&State->Delay, State->Offset+i, sample); - } + DelayLineIn(&State->Delay, State->Offset+i, + early[(MAX_UPDATE_SAMPLES*2/4)+(i>>2)][i&3] + ); // Calculate the early reflection from the first delay tap. EarlyReflection(State, todo, early); |