From e3a70e50216850aff2721ce3cc4a3e1c2cee311f Mon Sep 17 00:00:00 2001
From: Chris Robinson <chris.kcat@gmail.com>
Date: Wed, 12 May 2010 04:56:03 -0700
Subject: [PATCH] Scale output of effects to compensate for device down-mixing

---
 Alc/alcEcho.c      |  9 ++++++++-
 Alc/alcModulator.c | 12 +++++++++---
 Alc/alcReverb.c    | 13 +++++++++++--
 3 files changed, 28 insertions(+), 6 deletions(-)

diff --git a/Alc/alcEcho.c b/Alc/alcEcho.c
index d9700527..d6f7fbb3 100644
--- a/Alc/alcEcho.c
+++ b/Alc/alcEcho.c
@@ -49,6 +49,8 @@ typedef struct ALechoState {
 
     ALfloat FeedGain;
 
+    ALfloat Scale;
+
     FILTER iirFilter;
     ALfloat history[2];
 } ALechoState;
@@ -88,6 +90,9 @@ static ALboolean EchoDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
     for(i = 0;i < state->BufferLength;i++)
         state->SampleBuffer[i] = 0.0f;
 
+    state->Scale = aluSqrt(Device->NumChan / 6.0f);
+    state->Scale = __min(state->Scale, 1.0f);
+
     return AL_TRUE;
 }
 
@@ -122,7 +127,7 @@ 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;
+    const ALfloat gain = Slot->Gain * state->Scale;
     ALfloat samp[2], smp;
     ALuint i;
 
@@ -178,6 +183,8 @@ ALeffectState *EchoCreate(void)
     state->GainL = 0.0f;
     state->GainR = 0.0f;
 
+    state->Scale = 1.0f;
+
     state->iirFilter.coeff = 0.0f;
     state->iirFilter.history[0] = 0.0f;
     state->iirFilter.history[1] = 0.0f;
diff --git a/Alc/alcModulator.c b/Alc/alcModulator.c
index cabbaeb6..0578ee1c 100644
--- a/Alc/alcModulator.c
+++ b/Alc/alcModulator.c
@@ -43,6 +43,8 @@ typedef struct ALmodulatorState {
     ALuint index;
     ALuint step;
 
+    ALfloat Scale;
+
     FILTER iirFilter;
     ALfloat history[1];
 } ALmodulatorState;
@@ -87,9 +89,11 @@ static ALvoid ModulatorDestroy(ALeffectState *effect)
 
 static ALboolean ModulatorDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
 {
+    ALmodulatorState *state = (ALmodulatorState*)effect;
+
+    state->Scale = aluSqrt(Device->NumChan / 8.0f);
+
     return AL_TRUE;
-    (void)effect;
-    (void)Device;
 }
 
 static ALvoid ModulatorUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffect *Effect)
@@ -117,7 +121,7 @@ static ALvoid ModulatorUpdate(ALeffectState *effect, ALCcontext *Context, const
 static ALvoid ModulatorProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[OUTPUTCHANNELS])
 {
     ALmodulatorState *state = (ALmodulatorState*)effect;
-    const ALfloat gain = Slot->Gain;
+    const ALfloat gain = Slot->Gain * state->Scale;
     const ALuint step = state->step;
     ALuint index = state->index;
     ALfloat samp;
@@ -187,6 +191,8 @@ ALeffectState *ModulatorCreate(void)
     state->index = 0.0f;
     state->step = 1.0f;
 
+    state->Scale = 1.0f;
+
     state->iirFilter.coeff = 0.0f;
     state->iirFilter.history[0] = 0.0f;
 
diff --git a/Alc/alcReverb.c b/Alc/alcReverb.c
index 3529c08d..6ca47d2d 100644
--- a/Alc/alcReverb.c
+++ b/Alc/alcReverb.c
@@ -129,6 +129,9 @@ typedef struct ALverbState {
     } Echo;
     // The current read offset for all delay lines.
     ALuint Offset;
+
+    // Gain scale to account for device down-mixing
+    ALfloat Scale;
 } ALverbState;
 
 /* This coefficient is used to define the maximum frequency range controlled
@@ -987,6 +990,8 @@ static ALboolean VerbDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
     if(!AllocLines(AL_FALSE, frequency, State))
         return AL_FALSE;
 
+    State->Scale = aluSqrt(Device->NumChan / 8.0f);
+
     // The early reflection and late all-pass filter line lengths are static,
     // so their offsets only need to be calculated once.
     for(index = 0;index < 4;index++)
@@ -1012,6 +1017,8 @@ static ALboolean EAXVerbDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
     if(!AllocLines(AL_TRUE, frequency, State))
         return AL_FALSE;
 
+    State->Scale = aluSqrt(Device->NumChan / 8.0f);
+
     // Calculate the modulation filter coefficient.  Notice that the exponent
     // is calculated given the current sample rate.  This ensures that the
     // resulting filter response over time is consistent across all sample
@@ -1141,7 +1148,7 @@ 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;
+    ALfloat gain = Slot->Gain * State->Scale;
 
     for(index = 0;index < SamplesToDo;index++)
     {
@@ -1173,7 +1180,7 @@ static ALvoid EAXVerbProcess(ALeffectState *effect, const ALeffectslot *Slot, AL
     ALverbState *State = (ALverbState*)effect;
     ALuint index;
     ALfloat early[4], late[4];
-    ALfloat gain = Slot->Gain;
+    ALfloat gain = Slot->Gain * State->Scale;
 
     for(index = 0;index < SamplesToDo;index++)
     {
@@ -1304,6 +1311,8 @@ ALeffectState *VerbCreate(void)
 
     State->Offset = 0;
 
+    State->Scale = 1.0f;
+
     return &State->state;
 }
 
-- 
2.11.4.GIT