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authorChris Robinson <[email protected]>2011-07-16 03:35:51 -0700
committerChris Robinson <[email protected]>2011-07-16 03:35:51 -0700
commitce6db53e5f63c5445e5884ac91d4ce4030be4198 (patch)
tree223e23c82c50bc8ea194e20b471a149e920ee105
parent292ea0607ddd3df4edced130cc041ba2de13cf27 (diff)
Apply the slot gain during the effect update method
-rw-r--r--Alc/alcDedicated.c17
-rw-r--r--Alc/alcEcho.c29
-rw-r--r--Alc/alcModulator.c46
-rw-r--r--Alc/alcReverb.c60
4 files changed, 80 insertions, 72 deletions
diff --git a/Alc/alcDedicated.c b/Alc/alcDedicated.c
index d12130f8..c812877a 100644
--- a/Alc/alcDedicated.c
+++ b/Alc/alcDedicated.c
@@ -55,38 +55,45 @@ static ALvoid DedicatedDLGUpdate(ALeffectState *effect, ALCcontext *Context, con
ALdedicatedState *state = (ALdedicatedState*)effect;
ALCdevice *device = Context->Device;
const ALfloat *SpeakerGain;
+ ALfloat Gain;
ALint pos;
ALsizei s;
pos = aluCart2LUTpos(1.0f, 0.0f);
SpeakerGain = device->PanningLUT[pos];
+ Gain = Slot->Gain * Slot->effect.Params.Dedicated.Gain;
for(s = 0;s < MAXCHANNELS;s++)
- state->gains[s] = SpeakerGain[s] * Slot->effect.Params.Dedicated.Gain;
+ state->gains[s] = SpeakerGain[s] * Gain;
}
static ALvoid DedicatedLFEUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffectslot *Slot)
{
ALdedicatedState *state = (ALdedicatedState*)effect;
+ ALfloat Gain;
ALsizei s;
(void)Context;
+ Gain = Slot->Gain * Slot->effect.Params.Dedicated.Gain;
for(s = 0;s < MAXCHANNELS;s++)
state->gains[s] = 0.0f;
- state->gains[LFE] = Slot->effect.Params.Dedicated.Gain;
+ state->gains[LFE] = Gain;
}
static ALvoid DedicatedProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS])
{
ALdedicatedState *state = (ALdedicatedState*)effect;
const ALfloat *gains = state->gains;
- ALuint i;
+ ALuint i, s;
+ (void)Slot;
for(i = 0;i < SamplesToDo;i++)
{
- ALsizei s;
+ ALfloat sample;
+
+ sample = SamplesIn[i];
for(s = 0;s < MAXCHANNELS;s++)
- SamplesOut[i][s] = SamplesIn[i] * gains[s] * Slot->Gain;
+ SamplesOut[i][s] = sample * gains[s];
}
}
diff --git a/Alc/alcEcho.c b/Alc/alcEcho.c
index f60290f9..b62c124a 100644
--- a/Alc/alcEcho.c
+++ b/Alc/alcEcho.c
@@ -90,22 +90,16 @@ static ALboolean EchoDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
for(i = 0;i < state->BufferLength;i++)
state->SampleBuffer[i] = 0.0f;
- for(i = 0;i < MAXCHANNELS;i++)
- state->Gain[i] = 0.0f;
- for(i = 0;i < Device->NumChan;i++)
- {
- enum Channel chan = Device->Speaker2Chan[i];
- state->Gain[chan] = 1.0f;
- }
-
return AL_TRUE;
}
static ALvoid EchoUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffectslot *Slot)
{
ALechoState *state = (ALechoState*)effect;
- ALuint frequency = Context->Device->Frequency;
- ALfloat lrpan, cw, g;
+ ALCdevice *Device = Context->Device;
+ ALuint frequency = Device->Frequency;
+ ALfloat lrpan, cw, g, gain;
+ ALuint i;
state->Tap[0].delay = (ALuint)(Slot->effect.Params.Echo.Delay * frequency) + 1;
state->Tap[1].delay = (ALuint)(Slot->effect.Params.Echo.LRDelay * frequency);
@@ -120,6 +114,15 @@ static ALvoid EchoUpdate(ALeffectState *effect, ALCcontext *Context, const ALeff
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / frequency);
g = 1.0f - Slot->effect.Params.Echo.Damping;
state->iirFilter.coeff = lpCoeffCalc(g, cw);
+
+ gain = Slot->Gain;
+ for(i = 0;i < MAXCHANNELS;i++)
+ state->Gain[i] = 0.0f;
+ for(i = 0;i < Device->NumChan;i++)
+ {
+ enum Channel chan = Device->Speaker2Chan[i];
+ state->Gain[chan] = gain;
+ }
}
static ALvoid EchoProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS])
@@ -129,9 +132,9 @@ static ALvoid EchoProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuin
const ALuint tap1 = state->Tap[0].delay;
const ALuint tap2 = state->Tap[1].delay;
ALuint offset = state->Offset;
- const ALfloat gain = Slot->Gain;
ALfloat samp[2], smp;
ALuint i;
+ (void)Slot;
for(i = 0;i < SamplesToDo;i++,offset++)
{
@@ -149,10 +152,6 @@ static ALvoid EchoProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuin
smp = lpFilter2P(&state->iirFilter, 0, smp+SamplesIn[i]);
state->SampleBuffer[offset&mask] = smp * state->FeedGain;
- // Apply slot gain
- samp[0] *= gain;
- samp[1] *= gain;
-
SamplesOut[i][FRONT_LEFT] += state->Gain[FRONT_LEFT] * samp[0];
SamplesOut[i][FRONT_RIGHT] += state->Gain[FRONT_RIGHT] * samp[1];
SamplesOut[i][SIDE_LEFT] += state->Gain[SIDE_LEFT] * samp[0];
diff --git a/Alc/alcModulator.c b/Alc/alcModulator.c
index 1908007c..c4085fa3 100644
--- a/Alc/alcModulator.c
+++ b/Alc/alcModulator.c
@@ -82,11 +82,9 @@ static __inline ALfloat hpFilter1P(FILTER *iir, ALuint offset, ALfloat input)
#define DECL_TEMPLATE(func) \
-static void Process##func(ALmodulatorState *state, const ALeffectslot *Slot, \
- ALuint SamplesToDo, const ALfloat *SamplesIn, \
- ALfloat (*SamplesOut)[MAXCHANNELS]) \
+static void Process##func(ALmodulatorState *state, ALuint SamplesToDo, \
+ const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS]) \
{ \
- const ALfloat gain = Slot->Gain; \
const ALuint step = state->step; \
ALuint index = state->index; \
ALfloat samp; \
@@ -102,9 +100,6 @@ static void Process##func(ALmodulatorState *state, const ALeffectslot *Slot, \
\
samp = hpFilter1P(&state->iirFilter, 0, samp); \
\
- /* Apply slot gain */ \
- samp *= gain; \
- \
SamplesOut[i][FRONT_LEFT] += state->Gain[FRONT_LEFT] * samp; \
SamplesOut[i][FRONT_RIGHT] += state->Gain[FRONT_RIGHT] * samp; \
SamplesOut[i][FRONT_CENTER] += state->Gain[FRONT_CENTER] * samp; \
@@ -132,24 +127,17 @@ static ALvoid ModulatorDestroy(ALeffectState *effect)
static ALboolean ModulatorDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
{
- ALmodulatorState *state = (ALmodulatorState*)effect;
- ALuint index;
-
- for(index = 0;index < MAXCHANNELS;index++)
- state->Gain[index] = 0.0f;
- for(index = 0;index < Device->NumChan;index++)
- {
- enum Channel chan = Device->Speaker2Chan[index];
- state->Gain[chan] = 1.0f;
- }
-
return AL_TRUE;
+ (void)effect;
+ (void)Device;
}
static ALvoid ModulatorUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffectslot *Slot)
{
ALmodulatorState *state = (ALmodulatorState*)effect;
- ALfloat cw, a = 0.0f;
+ ALCdevice *Device = Context->Device;
+ ALfloat gain, cw, a = 0.0f;
+ ALuint index;
if(Slot->effect.Params.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
state->Waveform = SINUSOID;
@@ -159,32 +147,42 @@ static ALvoid ModulatorUpdate(ALeffectState *effect, ALCcontext *Context, const
state->Waveform = SQUARE;
state->step = Slot->effect.Params.Modulator.Frequency*(1<<WAVEFORM_FRACBITS) /
- Context->Device->Frequency;
+ Device->Frequency;
if(!state->step)
state->step = 1;
cw = cos(2.0*M_PI * Slot->effect.Params.Modulator.HighPassCutoff /
- Context->Device->Frequency);
+ Device->Frequency);
a = (2.0f-cw) - aluSqrt(aluPow(2.0f-cw, 2.0f) - 1.0f);
state->iirFilter.coeff = a;
+
+ gain = Slot->Gain;
+ for(index = 0;index < MAXCHANNELS;index++)
+ state->Gain[index] = 0.0f;
+ for(index = 0;index < Device->NumChan;index++)
+ {
+ enum Channel chan = Device->Speaker2Chan[index];
+ state->Gain[chan] = gain;
+ }
}
static ALvoid ModulatorProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS])
{
ALmodulatorState *state = (ALmodulatorState*)effect;
+ (void)Slot;
switch(state->Waveform)
{
case SINUSOID:
- ProcessSin(state, Slot, SamplesToDo, SamplesIn, SamplesOut);
+ ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut);
break;
case SAWTOOTH:
- ProcessSaw(state, Slot, SamplesToDo, SamplesIn, SamplesOut);
+ ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut);
break;
case SQUARE:
- ProcessSquare(state, Slot, SamplesToDo, SamplesIn, SamplesOut);
+ ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut);
break;
}
}
diff --git a/Alc/alcReverb.c b/Alc/alcReverb.c
index 4282e9ee..43197912 100644
--- a/Alc/alcReverb.c
+++ b/Alc/alcReverb.c
@@ -589,7 +589,7 @@ static ALvoid UpdateEchoLine(ALfloat reverbGain, ALfloat lateGain, ALfloat echoT
}
// Update the early and late 3D panning gains.
-static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALverbState *State)
+static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALverbState *State)
{
ALfloat earlyPan[3] = { ReflectionsPan[0], ReflectionsPan[1],
ReflectionsPan[2] };
@@ -635,7 +635,7 @@ static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *Reflection
for(index = 0;index < Device->NumChan;index++)
{
enum Channel chan = Device->Speaker2Chan[index];
- State->Early.PanGain[chan] = lerp(1.0, speakerGain[chan], dirGain);
+ State->Early.PanGain[chan] = lerp(1.0, speakerGain[chan], dirGain) * Gain;
}
@@ -648,7 +648,7 @@ static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *Reflection
for(index = 0;index < Device->NumChan;index++)
{
enum Channel chan = Device->Speaker2Chan[index];
- State->Late.PanGain[chan] = lerp(1.0, speakerGain[chan], dirGain);
+ State->Late.PanGain[chan] = lerp(1.0, speakerGain[chan], dirGain) * Gain;
}
}
@@ -1014,14 +1014,6 @@ static ALboolean VerbDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
frequency);
}
- for(index = 0;index < MAXCHANNELS;index++)
- State->Gain[index] = 0.0f;
- for(index = 0;index < Device->NumChan;index++)
- {
- enum Channel chan = Device->Speaker2Chan[index];
- State->Gain[chan] = 1.0f;
- }
-
return AL_TRUE;
}
@@ -1066,8 +1058,10 @@ static ALboolean EAXVerbDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
static ALvoid VerbUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffectslot *Slot)
{
ALverbState *State = (ALverbState*)effect;
- ALuint frequency = Context->Device->Frequency;
- ALfloat cw, x, y, hfRatio;
+ ALCdevice *Device = Context->Device;
+ ALuint frequency = Device->Frequency;
+ ALfloat cw, x, y, hfRatio, gain;
+ ALuint index;
// Calculate the master low-pass filter (from the master effect HF gain).
cw = CalcI3DL2HFreq(Slot->effect.Params.Reverb.HFReference, frequency);
@@ -1105,6 +1099,16 @@ static ALvoid VerbUpdate(ALeffectState *effect, ALCcontext *Context, const ALeff
UpdateLateLines(Slot->effect.Params.Reverb.Gain, Slot->effect.Params.Reverb.LateReverbGain,
x, Slot->effect.Params.Reverb.Density, Slot->effect.Params.Reverb.DecayTime,
Slot->effect.Params.Reverb.Diffusion, hfRatio, cw, frequency, State);
+
+ // Update channel gains
+ gain = Slot->Gain;
+ for(index = 0;index < MAXCHANNELS;index++)
+ State->Gain[index] = 0.0f;
+ for(index = 0;index < Device->NumChan;index++)
+ {
+ enum Channel chan = Device->Speaker2Chan[index];
+ State->Gain[chan] = gain;
+ }
}
// This updates the EAX reverb state. This is called any time the EAX reverb
@@ -1165,7 +1169,7 @@ static ALvoid EAXVerbUpdate(ALeffectState *effect, ALCcontext *Context, const AL
// Update early and late 3D panning.
Update3DPanning(Context->Device, Slot->effect.Params.Reverb.ReflectionsPan,
- Slot->effect.Params.Reverb.LateReverbPan, State);
+ Slot->effect.Params.Reverb.LateReverbPan, Slot->Gain, State);
}
// This processes the reverb state, given the input samples and an output
@@ -1175,8 +1179,8 @@ static ALvoid VerbProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuin
ALverbState *State = (ALverbState*)effect;
ALuint index;
ALfloat early[4], late[4], out[4];
- ALfloat gain = Slot->Gain;
const ALfloat *panGain = State->Gain;
+ (void)Slot;
for(index = 0;index < SamplesToDo;index++)
{
@@ -1184,10 +1188,10 @@ static ALvoid VerbProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuin
VerbPass(State, SamplesIn[index], early, late);
// Mix early reflections and late reverb.
- out[0] = (early[0] + late[0]) * gain;
- out[1] = (early[1] + late[1]) * gain;
- out[2] = (early[2] + late[2]) * gain;
- out[3] = (early[3] + late[3]) * gain;
+ out[0] = (early[0] + late[0]);
+ out[1] = (early[1] + late[1]);
+ out[2] = (early[2] + late[2]);
+ out[3] = (early[3] + late[3]);
// Output the results.
SamplesOut[index][FRONT_LEFT] += panGain[FRONT_LEFT] * out[0];
@@ -1208,7 +1212,7 @@ static ALvoid EAXVerbProcess(ALeffectState *effect, const ALeffectslot *Slot, AL
ALverbState *State = (ALverbState*)effect;
ALuint index;
ALfloat early[4], late[4];
- ALfloat gain = Slot->Gain;
+ (void)Slot;
for(index = 0;index < SamplesToDo;index++)
{
@@ -1220,28 +1224,28 @@ static ALvoid EAXVerbProcess(ALeffectState *effect, const ALeffectslot *Slot, AL
// reverb engine is not so scalable.
SamplesOut[index][FRONT_LEFT] +=
(State->Early.PanGain[FRONT_LEFT]*early[0] +
- State->Late.PanGain[FRONT_LEFT]*late[0]) * gain;
+ State->Late.PanGain[FRONT_LEFT]*late[0]);
SamplesOut[index][FRONT_RIGHT] +=
(State->Early.PanGain[FRONT_RIGHT]*early[1] +
- State->Late.PanGain[FRONT_RIGHT]*late[1]) * gain;
+ State->Late.PanGain[FRONT_RIGHT]*late[1]);
SamplesOut[index][FRONT_CENTER] +=
(State->Early.PanGain[FRONT_CENTER]*early[3] +
- State->Late.PanGain[FRONT_CENTER]*late[3]) * gain;
+ State->Late.PanGain[FRONT_CENTER]*late[3]);
SamplesOut[index][SIDE_LEFT] +=
(State->Early.PanGain[SIDE_LEFT]*early[0] +
- State->Late.PanGain[SIDE_LEFT]*late[0]) * gain;
+ State->Late.PanGain[SIDE_LEFT]*late[0]);
SamplesOut[index][SIDE_RIGHT] +=
(State->Early.PanGain[SIDE_RIGHT]*early[1] +
- State->Late.PanGain[SIDE_RIGHT]*late[1]) * gain;
+ State->Late.PanGain[SIDE_RIGHT]*late[1]);
SamplesOut[index][BACK_LEFT] +=
(State->Early.PanGain[BACK_LEFT]*early[0] +
- State->Late.PanGain[BACK_LEFT]*late[0]) * gain;
+ State->Late.PanGain[BACK_LEFT]*late[0]);
SamplesOut[index][BACK_RIGHT] +=
(State->Early.PanGain[BACK_RIGHT]*early[1] +
- State->Late.PanGain[BACK_RIGHT]*late[1]) * gain;
+ State->Late.PanGain[BACK_RIGHT]*late[1]);
SamplesOut[index][BACK_CENTER] +=
(State->Early.PanGain[BACK_CENTER]*early[2] +
- State->Late.PanGain[BACK_CENTER]*late[2]) * gain;
+ State->Late.PanGain[BACK_CENTER]*late[2]);
}
}