Pas the output device channel count to ALeffectState::process

12 files changed, 47 insertions, 50 deletions

diff --git a/Alc/ALu.c b/Alc/ALu.c
index 0e064898..4d056be7 100644
--- a/Alc/ALu.c
+++ b/Alc/ALu.c
@@ -1179,7 +1179,7 @@ ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
                     V((*slot)->EffectState,update)(device, *slot);
 
                 V((*slot)->EffectState,process)(SamplesToDo, (*slot)->WetBuffer[0],
-                                                device->DryBuffer);
+                                                device->DryBuffer, device->NumChannels);
 
                 for(i = 0;i < SamplesToDo;i++)
                     (*slot)->WetBuffer[0][i] = 0.0f;
@@ -1197,7 +1197,7 @@ ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
                 V((*slot)->EffectState,update)(device, *slot);
 
             V((*slot)->EffectState,process)(SamplesToDo, (*slot)->WetBuffer[0],
-                                            device->DryBuffer);
+                                            device->DryBuffer, device->NumChannels);
 
             for(i = 0;i < SamplesToDo;i++)
                 (*slot)->WetBuffer[0][i] = 0.0f;
diff --git a/Alc/effects/autowah.c b/Alc/effects/autowah.c
index 8caf1fcc..6f16743d 100644
--- a/Alc/effects/autowah.c
+++ b/Alc/effects/autowah.c
@@ -78,7 +78,7 @@ static ALvoid ALautowahState_update(ALautowahState *state, ALCdevice *device, co
     ComputeAmbientGains(device, slot->Gain, state->Gain);
 }
 
-static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE])
+static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     ALuint it, kt;
     ALuint base;
@@ -135,10 +135,10 @@ static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo,
         }
         state->GainCtrl = gain;
 
-        for(kt = 0;kt < MAX_OUTPUT_CHANNELS;kt++)
+        for(kt = 0;kt < NumChannels;kt++)
         {
             ALfloat gain = state->Gain[kt];
-            if(!(gain > GAIN_SILENCE_THRESHOLD))
+            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
                 continue;
 
             for(it = 0;it < td;it++)
diff --git a/Alc/effects/chorus.c b/Alc/effects/chorus.c
index 0dde134a..02887fde 100644
--- a/Alc/effects/chorus.c
+++ b/Alc/effects/chorus.c
@@ -203,7 +203,7 @@ DECL_TEMPLATE(Sinusoid)
 
 #undef DECL_TEMPLATE
 
-static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     ALuint it, kt;
     ALuint base;
@@ -223,17 +223,17 @@ static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, co
                 break;
         }
 
-        for(kt = 0;kt < MAX_OUTPUT_CHANNELS;kt++)
+        for(kt = 0;kt < NumChannels;kt++)
         {
             ALfloat gain = state->Gain[0][kt];
-            if(gain > GAIN_SILENCE_THRESHOLD)
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
             {
                 for(it = 0;it < td;it++)
                     SamplesOut[kt][it+base] += temps[it][0] * gain;
             }
 
             gain = state->Gain[1][kt];
-            if(gain > GAIN_SILENCE_THRESHOLD)
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
             {
                 for(it = 0;it < td;it++)
                     SamplesOut[kt][it+base] += temps[it][1] * gain;
diff --git a/Alc/effects/compressor.c b/Alc/effects/compressor.c
index ab88ecf0..f9964427 100644
--- a/Alc/effects/compressor.c
+++ b/Alc/effects/compressor.c
@@ -62,7 +62,7 @@ static ALvoid ALcompressorState_update(ALcompressorState *state, ALCdevice *devi
     ComputeAmbientGains(device, slot->Gain, state->Gain);
 }
 
-static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE])
+static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     ALuint it, kt;
     ALuint base;
@@ -116,10 +116,10 @@ static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint Samples
         }
 
 
-        for(kt = 0;kt < MAX_OUTPUT_CHANNELS;kt++)
+        for(kt = 0;kt < NumChannels;kt++)
         {
             ALfloat gain = state->Gain[kt];
-            if(!(gain > GAIN_SILENCE_THRESHOLD))
+            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
                 continue;
 
             for(it = 0;it < td;it++)
diff --git a/Alc/effects/dedicated.c b/Alc/effects/dedicated.c
index 2d955888..e09cc682 100644
--- a/Alc/effects/dedicated.c
+++ b/Alc/effects/dedicated.c
@@ -75,14 +75,14 @@ static ALvoid ALdedicatedState_update(ALdedicatedState *state, ALCdevice *device
     }
 }
 
-static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     const ALfloat *gains = state->gains;
     ALuint i, c;
 
-    for(c = 0;c < MAX_OUTPUT_CHANNELS;c++)
+    for(c = 0;c < NumChannels;c++)
     {
-        if(!(gains[c] > GAIN_SILENCE_THRESHOLD))
+        if(!(fabsf(gains[c]) > GAIN_SILENCE_THRESHOLD))
             continue;
 
         for(i = 0;i < SamplesToDo;i++)
diff --git a/Alc/effects/distortion.c b/Alc/effects/distortion.c
index 9474f3ae..7098cdf9 100644
--- a/Alc/effects/distortion.c
+++ b/Alc/effects/distortion.c
@@ -84,7 +84,7 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, ALCdevice *Devi
     ComputeAmbientGains(Device, Slot->Gain, state->Gain);
 }
 
-static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     const ALfloat fc = state->edge_coeff;
     float oversample_buffer[64][4];
@@ -154,10 +154,10 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples
             temps[it] = oversample_buffer[it][0] * state->attenuation;
         }
 
-        for(kt = 0;kt < MAX_OUTPUT_CHANNELS;kt++)
+        for(kt = 0;kt < NumChannels;kt++)
         {
             ALfloat gain = state->Gain[kt];
-            if(!(gain > GAIN_SILENCE_THRESHOLD))
+            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
                 continue;
 
             for(it = 0;it < td;it++)
diff --git a/Alc/effects/echo.c b/Alc/effects/echo.c
index d6c62a30..fd85bdb0 100644
--- a/Alc/effects/echo.c
+++ b/Alc/effects/echo.c
@@ -109,7 +109,7 @@ static ALvoid ALechoState_update(ALechoState *state, ALCdevice *Device, const AL
     ComputeDirectionalGains(Device, pandir, gain, state->Gain[1]);
 }
 
-static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     const ALuint mask = state->BufferLength-1;
     const ALuint tap1 = state->Tap[0].delay;
@@ -138,17 +138,17 @@ static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const
             offset++;
         }
 
-        for(k = 0;k < MAX_OUTPUT_CHANNELS;k++)
+        for(k = 0;k < NumChannels;k++)
         {
             ALfloat gain = state->Gain[0][k];
-            if(gain > GAIN_SILENCE_THRESHOLD)
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
             {
                 for(i = 0;i < td;i++)
                     SamplesOut[k][i+base] += temps[i][0] * gain;
             }
 
             gain = state->Gain[1][k];
-            if(gain > GAIN_SILENCE_THRESHOLD)
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
             {
                 for(i = 0;i < td;i++)
                     SamplesOut[k][i+base] += temps[i][1] * gain;
diff --git a/Alc/effects/equalizer.c b/Alc/effects/equalizer.c
index dc02e9b0..7caec266 100644
--- a/Alc/effects/equalizer.c
+++ b/Alc/effects/equalizer.c
@@ -118,7 +118,7 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, ALCdevice *device
                             0.0f);
 }
 
-static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     ALuint base;
     ALuint it;
@@ -140,10 +140,10 @@ static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesTo
             temps[it] = smp;
         }
 
-        for(kt = 0;kt < MAX_OUTPUT_CHANNELS;kt++)
+        for(kt = 0;kt < NumChannels;kt++)
         {
             ALfloat gain = state->Gain[kt];
-            if(!(gain > GAIN_SILENCE_THRESHOLD))
+            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
                 continue;
 
             for(it = 0;it < td;it++)
diff --git a/Alc/effects/flanger.c b/Alc/effects/flanger.c
index e5a49a43..460bb866 100644
--- a/Alc/effects/flanger.c
+++ b/Alc/effects/flanger.c
@@ -203,7 +203,7 @@ DECL_TEMPLATE(Sinusoid)
 
 #undef DECL_TEMPLATE
 
-static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     ALuint it, kt;
     ALuint base;
@@ -223,17 +223,17 @@ static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo,
                 break;
         }
 
-        for(kt = 0;kt < MAX_OUTPUT_CHANNELS;kt++)
+        for(kt = 0;kt < NumChannels;kt++)
         {
             ALfloat gain = state->Gain[0][kt];
-            if(gain > GAIN_SILENCE_THRESHOLD)
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
             {
                 for(it = 0;it < td;it++)
                     SamplesOut[kt][it+base] += temps[it][0] * gain;
             }
 
             gain = state->Gain[1][kt];
-            if(gain > GAIN_SILENCE_THRESHOLD)
+            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
             {
                 for(it = 0;it < td;it++)
                     SamplesOut[kt][it+base] += temps[it][1] * gain;
diff --git a/Alc/effects/modulator.c b/Alc/effects/modulator.c
index c69a8a09..4d183449 100644
--- a/Alc/effects/modulator.c
+++ b/Alc/effects/modulator.c
@@ -69,7 +69,7 @@ static inline ALfloat Square(ALuint index)
 #define DECL_TEMPLATE(func)                                                   \
 static void Process##func(ALmodulatorState *state, ALuint SamplesToDo,        \
   const ALfloat *restrict SamplesIn,                                          \
-  ALfloat (*restrict SamplesOut)[BUFFERSIZE])                                 \
+  ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)             \
 {                                                                             \
     const ALuint step = state->step;                                          \
     ALuint index = state->index;                                              \
@@ -92,10 +92,10 @@ static void Process##func(ALmodulatorState *state, ALuint SamplesToDo,        \
             temps[i] = samp * func(index);                                    \
         }                                                                     \
                                                                               \
-        for(k = 0;k < MAX_OUTPUT_CHANNELS;k++)                                \
+        for(k = 0;k < NumChannels;k++)                                        \
         {                                                                     \
             ALfloat gain = state->Gain[k];                                    \
-            if(!(gain > GAIN_SILENCE_THRESHOLD))                              \
+            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))                       \
                 continue;                                                     \
                                                                               \
             for(i = 0;i < td;i++)                                             \
@@ -152,20 +152,20 @@ static ALvoid ALmodulatorState_update(ALmodulatorState *state, ALCdevice *Device
     ComputeAmbientGains(Device, Slot->Gain, state->Gain);
 }
 
-static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     switch(state->Waveform)
     {
         case SINUSOID:
-            ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut);
+            ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels);
             break;
 
         case SAWTOOTH:
-            ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut);
+            ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels);
             break;
 
         case SQUARE:
-            ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut);
+            ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels);
             break;
     }
 }
diff --git a/Alc/effects/null.c b/Alc/effects/null.c
index 6dacd021..adc4ca81 100644
--- a/Alc/effects/null.c
+++ b/Alc/effects/null.c
@@ -41,11 +41,8 @@ static ALvoid ALnullState_update(ALnullState* UNUSED(state), ALCdevice* UNUSED(d
  * input to the output buffer. The result should be added to the output buffer,
  * not replace it.
  */
-static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALuint UNUSED(samplesToDo), const ALfloat *restrict UNUSED(samplesIn), ALfloat (*restrict samplesOut)[BUFFERSIZE])
+static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALuint UNUSED(samplesToDo), const ALfloat *restrict UNUSED(samplesIn), ALfloatBUFFERSIZE*restrict UNUSED(samplesOut), ALuint UNUSED(NumChannels))
 {
-    /* NOTE: Couldn't use the UNUSED macro on samplesOut due to the way GCC's
-     * __attribute__ declaration interacts with the parenthesis. */
-    (void)samplesOut;
 }
 
 /* This allocates memory to store the object, before it gets constructed.
diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c
index e1b47ec2..95957737 100644
--- a/Alc/effects/reverb.c
+++ b/Alc/effects/reverb.c
@@ -556,7 +556,7 @@ static inline ALvoid EAXVerbPass(ALreverbState *State, ALfloat in, ALfloat *rest
     State->Offset++;
 }
 
-static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     ALfloat (*restrict out)[4] = State->ReverbSamples;
     ALuint index, c;
@@ -565,10 +565,10 @@ static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint Samples
     for(index = 0;index < SamplesToDo;index++)
         VerbPass(State, SamplesIn[index], out[index]);
 
-    for(c = 0;c < MAX_OUTPUT_CHANNELS;c++)
+    for(c = 0;c < NumChannels;c++)
     {
         ALfloat gain = State->Gain[c];
-        if(!(gain > GAIN_SILENCE_THRESHOLD))
+        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
             continue;
 
         for(index = 0;index < SamplesToDo;index++)
@@ -576,7 +576,7 @@ static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint Samples
     }
 }
 
-static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     ALfloat (*restrict early)[4] = State->EarlySamples;
     ALfloat (*restrict late)[4] = State->ReverbSamples;
@@ -586,18 +586,18 @@ static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo,
     for(index = 0;index < SamplesToDo;index++)
         EAXVerbPass(State, SamplesIn[index], early[index], late[index]);
 
-    for(c = 0;c < MAX_OUTPUT_CHANNELS;c++)
+    for(c = 0;c < NumChannels;c++)
     {
         ALfloat earlyGain, lateGain;
 
         earlyGain = State->Early.PanGain[c];
-        if(earlyGain > GAIN_SILENCE_THRESHOLD)
+        if(fabsf(earlyGain) > GAIN_SILENCE_THRESHOLD)
         {
             for(index = 0;index < SamplesToDo;index++)
                 SamplesOut[c][index] += earlyGain*early[index][c&3];
         }
         lateGain = State->Late.PanGain[c];
-        if(lateGain > GAIN_SILENCE_THRESHOLD)
+        if(fabsf(lateGain) > GAIN_SILENCE_THRESHOLD)
         {
             for(index = 0;index < SamplesToDo;index++)
                 SamplesOut[c][index] += lateGain*late[index][c&3];
@@ -605,12 +605,12 @@ static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo,
     }
 }
 
-static ALvoid ALreverbState_process(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
+static ALvoid ALreverbState_process(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
 {
     if(State->IsEax)
-        ALreverbState_processEax(State, SamplesToDo, SamplesIn, SamplesOut);
+        ALreverbState_processEax(State, SamplesToDo, SamplesIn, SamplesOut, NumChannels);
     else
-        ALreverbState_processStandard(State, SamplesToDo, SamplesIn, SamplesOut);
+        ALreverbState_processStandard(State, SamplesToDo, SamplesIn, SamplesOut, NumChannels);
 }
 
 // Given the allocated sample buffer, this function updates each delay line