Release 1.5.304

Should help with latency issues some people have and not put too much extra
burden on the mixer, hopefully

Alc/ALc.c

@@ -33,6 +33,7 @@
 #include "AL/alc.h"
 #include "alThunk.h"
 #include "alSource.h"
+#include "alBuffer.h"
 #include "alExtension.h"
 #include "alAuxEffectSlot.h"
 #include "bs2b.h"
@@ -182,6 +183,8 @@ static ALCint alcEFXMinorVersion = 0;
 static ALCdevice *g_pDeviceList = NULL;
 static ALCuint    g_ulDeviceCount = 0;
 
+static CRITICAL_SECTION g_csMutex;
+
 // Context List
 static ALCcontext *g_pContextList = NULL;
 static ALCuint     g_ulContextCount = 0;
@@ -189,23 +192,69 @@ static ALCuint     g_ulContextCount = 0;
 // Context Error
 static ALCenum g_eLastContextError = ALC_NO_ERROR;
 
+static ALboolean init_done = AL_FALSE;
+
 ///////////////////////////////////////////////////////
 
 
 ///////////////////////////////////////////////////////
 // ALC Related helper functions
+#ifdef _WIN32
+BOOL APIENTRY DllMain(HANDLE hModule,DWORD ul_reason_for_call,LPVOID lpReserved)
+{
+    (void)lpReserved;
+
+    // Perform actions based on the reason for calling.
+    switch(ul_reason_for_call)
+    {
+        case DLL_PROCESS_ATTACH:
+            DisableThreadLibraryCalls(hModule);
+            break;
+
+        case DLL_PROCESS_DETACH:
+            if(!init_done)
+                break;
+            ReleaseALC();
+            ReleaseALBuffers();
+            ReleaseALEffects();
+            ReleaseALFilters();
+            FreeALConfig();
+            ALTHUNK_EXIT();
+            DeleteCriticalSection(&g_csMutex);
+            break;
+    }
+    return TRUE;
+}
+#else
+#ifdef HAVE_GCC_DESTRUCTOR
+static void my_deinit() __attribute__((destructor));
+static void my_deinit()
+{
+    static ALenum once = AL_FALSE;
+    if(once || !init_done) return;
+    once = AL_TRUE;
+
+    ReleaseALC();
+    ReleaseALBuffers();
+    ReleaseALEffects();
+    ReleaseALFilters();
+    FreeALConfig();
+    ALTHUNK_EXIT();
+    DeleteCriticalSection(&g_csMutex);
+}
+#endif
+#endif
 
 static void InitAL(void)
 {
-    static int done = 0;
-    if(!done)
+    if(!init_done)
     {
         int i;
         const char *devs, *str;
 
-        done = 1;
+        init_done = AL_TRUE;
 
-        InitializeCriticalSection(&_alMutex);
+        InitializeCriticalSection(&g_csMutex);
         ALTHUNK_INIT();
         ReadALConfig();
 
@@ -335,7 +384,7 @@ ALCvoid SetALCError(ALenum errorCode)
 ALCvoid SuspendContext(ALCcontext *pContext)
 {
     (void)pContext;
-    EnterCriticalSection(&_alMutex);
+    EnterCriticalSection(&g_csMutex);
 }
 
 
@@ -347,7 +396,7 @@ ALCvoid SuspendContext(ALCcontext *pContext)
 ALCvoid ProcessContext(ALCcontext *pContext)
 {
     (void)pContext;
-    LeaveCriticalSection(&_alMutex);
+    LeaveCriticalSection(&g_csMutex);
 }
 
 
@@ -392,7 +441,7 @@ static ALvoid InitContext(ALCcontext *pContext)
     pContext->lNumStereoSources = 1;
     pContext->lNumMonoSources = pContext->Device->MaxNoOfSources - pContext->lNumStereoSources;
 
-    strcpy(pContext->ExtensionList, "AL_EXT_EXPONENT_DISTANCE AL_EXT_FLOAT32 AL_EXT_IMA4 AL_EXT_LINEAR_DISTANCE AL_EXT_MCFORMATS AL_EXT_OFFSET AL_LOKI_quadriphonic");
+    pContext->ExtensionList = "AL_EXT_EXPONENT_DISTANCE AL_EXT_FLOAT32 AL_EXT_IMA4 AL_EXT_LINEAR_DISTANCE AL_EXT_MCFORMATS AL_EXT_OFFSET AL_LOKI_quadriphonic";
 
     level = GetConfigValueInt(NULL, "cf_level", 0);
     if(level > 0 && level <= 6)
@@ -1169,9 +1218,9 @@ ALCAPI ALCdevice* ALCAPIENTRY alcOpenDevice(const ALCchar *deviceName)
         if(!aluChannelsFromFormat(device->Format))
             device->Format = AL_FORMAT_STEREO16;
 
-        device->UpdateSize = GetConfigValueInt(NULL, "refresh", 8192);
+        device->UpdateSize = GetConfigValueInt(NULL, "refresh", 4096);
         if((ALint)device->UpdateSize <= 0)
-            device->UpdateSize = 8192;
+            device->UpdateSize = 4096;
 
         device->MaxNoOfSources = GetConfigValueInt(NULL, "sources", 256);
         if((ALint)device->MaxNoOfSources <= 0)
@@ -1230,7 +1279,12 @@ ALCAPI ALCboolean ALCAPIENTRY alcCloseDevice(ALCdevice *pDevice)
         ProcessContext(NULL);
 
         if(pDevice->Context)
+        {
+#ifdef _DEBUG
+            AL_PRINT("alcCloseDevice(): destroying 1 Context\n");
+#endif
             alcDestroyContext(pDevice->Context);
+        }
         ALCdevice_ClosePlayback(pDevice);
 
         //Release device structure
@@ -1248,11 +1302,9 @@ ALCAPI ALCboolean ALCAPIENTRY alcCloseDevice(ALCdevice *pDevice)
 
 ALCvoid ReleaseALC(ALCvoid)
 {
-    InitAL();
-
 #ifdef _DEBUG
-    if(g_ulContextCount > 0)
-        AL_PRINT("exit() %u device(s) and %u context(s) NOT deleted\n", g_ulDeviceCount, g_ulContextCount);
+    if(g_ulDeviceCount > 0)
+        AL_PRINT("exit(): closing %u Device%s\n", g_ulDeviceCount, (g_ulDeviceCount>1)?"s":"");
 #endif
 
     while(g_pDeviceList)

Alc/ALu.c

@@ -64,7 +64,7 @@ typedef long long ALint64;
 #define __min min
 #endif
 
-#define BUFFERSIZE 48000
+#define BUFFERSIZE 24000
 #define FRACTIONBITS 14
 #define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
 #define MAX_PITCH 4
@@ -163,6 +163,42 @@ __inline ALuint aluChannelsFromFormat(ALenum format)
 }
 
 
+static __inline ALfloat lpFilter(FILTER *iir, ALfloat input)
+{
+    unsigned int i;
+    float *hist1_ptr,*hist2_ptr,*coef_ptr;
+    ALfloat output,new_hist,history1,history2;
+
+    coef_ptr = iir->coef;                /* coefficient pointer */
+
+    hist1_ptr = iir->history;            /* first history */
+    hist2_ptr = hist1_ptr + 1;           /* next history */
+
+    /* 1st number of coefficients array is overall input scale factor,
+     * or filter gain */
+    output = input * (*coef_ptr++);
+
+    for(i = 0;i < FILTER_SECTIONS;i++)
+    {
+        history1 = *hist1_ptr;           /* history values */
+        history2 = *hist2_ptr;
+
+        output = output - history1 * (*coef_ptr++);
+        new_hist = output - history2 * (*coef_ptr++);    /* poles */
+
+        output = new_hist + history1 * (*coef_ptr++);
+        output = output + history2 * (*coef_ptr++);      /* zeros */
+
+        *hist2_ptr++ = *hist1_ptr;
+        *hist1_ptr++ = new_hist;
+        hist1_ptr++;
+        hist2_ptr++;
+    }
+
+    return output;
+}
+
+
 static __inline ALshort aluF2S(ALfloat Value)
 {
     ALint i;
@@ -210,20 +246,9 @@ static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat matrix[3][3])
     memcpy(vector, result, sizeof(result));
 }
 
-static __inline ALfloat aluComputeSample(ALfloat GainHF, ALfloat sample, ALfloat LastSample)
+static __inline ALfloat aluComputeSample(ALfloat GainHF, ALfloat sample, ALfloat LowSample)
 {
-    if(GainHF < 1.0f)
-    {
-        if(GainHF > 0.0f)
-        {
-            sample *= GainHF;
-            sample += LastSample * (1.0f-GainHF);
-        }
-        else
-            sample = 0.0f;
-    }
-
-    return sample;
+    return LowSample + ((sample - LowSample) * GainHF);
 }
 
 static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
@@ -247,7 +272,7 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
     ALfloat WetGainHF = 1.0f;
 
     //Get context properties
-    DopplerFactor   = ALContext->DopplerFactor;
+    DopplerFactor   = ALContext->DopplerFactor * ALSource->DopplerFactor;
     DopplerVelocity = ALContext->DopplerVelocity;
     flSpeedOfSound  = ALContext->flSpeedOfSound;
 
@@ -273,6 +298,10 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
     if(isMono != AL_FALSE)
     {
         //1. Translate Listener to origin (convert to head relative)
+        // Note that Direction and SourceToListener are *not* transformed.
+        // SourceToListener is used with the source and listener velocities,
+        // which are untransformed, and Direction is used with SourceToListener
+        // for the sound cone
         if(ALSource->bHeadRelative==AL_FALSE)
         {
             // Build transform matrix
@@ -290,10 +319,21 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
             Position[0] -= ALContext->Listener.Position[0];
             Position[1] -= ALContext->Listener.Position[1];
             Position[2] -= ALContext->Listener.Position[2];
+
+            SourceToListener[0] = -Position[0];
+            SourceToListener[1] = -Position[1];
+            SourceToListener[2] = -Position[2];
+
             // Transform source position and direction into listener space
             aluMatrixVector(Position, Matrix);
-            aluMatrixVector(Direction, Matrix);
         }
+        else
+        {
+            SourceToListener[0] = -Position[0];
+            SourceToListener[1] = -Position[1];
+            SourceToListener[2] = -Position[2];
+        }
+        aluNormalize(SourceToListener);
         aluNormalize(Direction);
 
         //2. Calculate distance attenuation
@@ -302,7 +342,7 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
         if(ALSource->Send[0].Slot && !ALSource->Send[0].Slot->AuxSendAuto)
         {
             if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
-                RoomRolloff = ALSource->Send[0].Slot->effect.Reverb.RoomRolloffFactor;
+                RoomRolloff += ALSource->Send[0].Slot->effect.Reverb.RoomRolloffFactor;
         }
 
         flAttenuation = 1.0f;
@@ -372,11 +412,6 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
         WetMix = __max(WetMix,MinVolume);
 
         //3. Apply directional soundcones
-        SourceToListener[0] = -Position[0];
-        SourceToListener[1] = -Position[1];
-        SourceToListener[2] = -Position[2];
-        aluNormalize(SourceToListener);
-
         Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f /
                 3.141592654f;
         if(Angle >= InnerAngle && Angle <= OuterAngle)
@@ -458,6 +493,7 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
 
             if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
             {
+                WetMix *= ALSource->Send[0].Slot->effect.Reverb.Gain;
                 WetGainHF *= ALSource->Send[0].Slot->effect.Reverb.GainHF;
                 WetGainHF *= pow(ALSource->Send[0].Slot->effect.Reverb.AirAbsorptionGainHF,
                                  Distance * MetersPerUnit);
@@ -477,17 +513,20 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
         switch(aluChannelsFromFormat(OutputFormat))
         {
             case 1:
-                drysend[FRONT_LEFT]  = DryMix * aluSqrt(1.0f); //Direct
-                drysend[FRONT_RIGHT] = DryMix * aluSqrt(1.0f); //Direct
-                wetsend[FRONT_LEFT]  = WetMix * aluSqrt(1.0f); //Room
-                wetsend[FRONT_RIGHT] = WetMix * aluSqrt(1.0f); //Room
-                break;
             case 2:
                 PanningLR = 0.5f + 0.5f*Position[0];
                 drysend[FRONT_LEFT]  = DryMix * aluSqrt(1.0f-PanningLR); //L Direct
                 drysend[FRONT_RIGHT] = DryMix * aluSqrt(     PanningLR); //R Direct
+                drysend[BACK_LEFT]   = drysend[FRONT_LEFT];
+                drysend[BACK_RIGHT]  = drysend[FRONT_RIGHT];
+                drysend[SIDE_LEFT]   = drysend[FRONT_LEFT];
+                drysend[SIDE_RIGHT]  = drysend[FRONT_RIGHT];
                 wetsend[FRONT_LEFT]  = WetMix * aluSqrt(1.0f-PanningLR); //L Room
                 wetsend[FRONT_RIGHT] = WetMix * aluSqrt(     PanningLR); //R Room
+                wetsend[BACK_LEFT]   = wetsend[FRONT_LEFT];
+                wetsend[BACK_RIGHT]  = wetsend[FRONT_RIGHT];
+                wetsend[SIDE_LEFT]   = wetsend[FRONT_LEFT];
+                wetsend[SIDE_RIGHT]  = wetsend[FRONT_RIGHT];
                 break;
             case 4:
             /* TODO: Add center/lfe channel in spatial calculations? */
@@ -503,10 +542,14 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
                 drysend[FRONT_RIGHT] = DryMix * aluSqrt((     PanningLR)*(1.0f-PanningFB));
                 drysend[BACK_LEFT]   = DryMix * aluSqrt((1.0f-PanningLR)*(     PanningFB));
                 drysend[BACK_RIGHT]  = DryMix * aluSqrt((     PanningLR)*(     PanningFB));
+                drysend[SIDE_LEFT]   = (drysend[FRONT_LEFT] +drysend[BACK_LEFT])  * 0.5f;
+                drysend[SIDE_RIGHT]  = (drysend[FRONT_RIGHT]+drysend[BACK_RIGHT]) * 0.5f;
                 wetsend[FRONT_LEFT]  = WetMix * aluSqrt((1.0f-PanningLR)*(1.0f-PanningFB));
                 wetsend[FRONT_RIGHT] = WetMix * aluSqrt((     PanningLR)*(1.0f-PanningFB));
                 wetsend[BACK_LEFT]   = WetMix * aluSqrt((1.0f-PanningLR)*(     PanningFB));
                 wetsend[BACK_RIGHT]  = WetMix * aluSqrt((     PanningLR)*(     PanningFB));
+                wetsend[SIDE_LEFT]   = (wetsend[FRONT_LEFT] +wetsend[BACK_LEFT])  * 0.5f;
+                wetsend[SIDE_RIGHT]  = (wetsend[FRONT_RIGHT]+wetsend[BACK_RIGHT]) * 0.5f;
                 break;
             case 7:
             case 8:
@@ -585,7 +628,6 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
 {
     static float DryBuffer[BUFFERSIZE][OUTPUTCHANNELS];
     static float WetBuffer[BUFFERSIZE][OUTPUTCHANNELS];
-    static float ReverbBuffer[BUFFERSIZE];
     ALfloat DrySend[OUTPUTCHANNELS] = { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f };
     ALfloat WetSend[OUTPUTCHANNELS] = { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f };
     ALfloat DryGainHF = 0.0f;
@@ -593,11 +635,10 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
     ALuint BlockAlign,BufferSize;
     ALuint DataSize=0,DataPosInt=0,DataPosFrac=0;
     ALuint Channels,Frequency,ulExtraSamples;
-    ALfloat DrySample, WetSample;
-    ALboolean doReverb;
     ALfloat Pitch;
-    ALint Looping,increment,State;
-    ALuint Buffer,fraction;
+    ALint Looping,State;
+    ALint fraction,increment;
+    ALuint Buffer;
     ALuint SamplesToDo;
     ALsource *ALSource;
     ALbuffer *ALBuffer;
@@ -608,6 +649,7 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
     ALbufferlistitem *BufferListItem;
     ALuint loop;
     ALint64 DataSize64,DataPos64;
+    FILTER *Filter;
 
     SuspendContext(ALContext);
 
@@ -626,7 +668,6 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
         //Clear mixing buffer
         memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
         memset(WetBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
-        memset(ReverbBuffer, 0, SamplesToDo*sizeof(ALfloat));
 
         //Actual mixing loop
         while(ALSource)
@@ -634,10 +675,6 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
             j = 0;
             State = ALSource->state;
 
-            doReverb = ((ALSource->Send[0].Slot &&
-                            ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB) ?
-                        AL_TRUE : AL_FALSE);
-
             while(State == AL_PLAYING && j < SamplesToDo)
             {
                 DataSize = 0;
@@ -666,8 +703,7 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
                     //Get source info
                     DataPosInt = ALSource->position;
                     DataPosFrac = ALSource->position_fraction;
-                    DrySample = ALSource->LastDrySample;
-                    WetSample = ALSource->LastWetSample;
+                    Filter = &ALSource->iirFilter;
 
                     //Compute 18.14 fixed point step
                     increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS));
@@ -675,13 +711,13 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
                         increment = (MAX_PITCH<<FRACTIONBITS);
 
                     //Figure out how many samples we can mix.
-                    //Pitch must be <= 4 (the number below !)
-                    DataSize64 = DataSize+MAX_PITCH;
+                    DataSize64 = DataSize;
                     DataSize64 <<= FRACTIONBITS;
                     DataPos64 = DataPosInt;
                     DataPos64 <<= FRACTIONBITS;
                     DataPos64 += DataPosFrac;
-                    BufferSize = (ALuint)((DataSize64-DataPos64) / increment);
+                    BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment);
+
                     BufferListItem = ALSource->queue;
                     for(loop = 0; loop < ALSource->BuffersPlayed; loop++)
                     {
@@ -695,7 +731,7 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
                             ALbuffer *NextBuf = (ALbuffer*)ALTHUNK_LOOKUPENTRY(BufferListItem->next->buffer);
                             if(NextBuf && NextBuf->data)
                             {
-                                ulExtraSamples = min(NextBuf->size, (ALint)(16*Channels));
+                                ulExtraSamples = min(NextBuf->size, (ALint)(ALBuffer->padding*Channels*2));
                                 memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
                             }
                         }
@@ -704,7 +740,7 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
                             ALbuffer *NextBuf = (ALbuffer*)ALTHUNK_LOOKUPENTRY(ALSource->queue->buffer);
                             if (NextBuf && NextBuf->data)
                             {
-                                ulExtraSamples = min(NextBuf->size, (ALint)(16*Channels));
+                                ulExtraSamples = min(NextBuf->size, (ALint)(ALBuffer->padding*Channels*2));
                                 memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
                             }
                         }
@@ -717,32 +753,31 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
                     {
                         k = DataPosFrac>>FRACTIONBITS;
                         fraction = DataPosFrac&FRACTIONMASK;
+
                         if(Channels==1)
                         {
+                            ALfloat sample, lowsamp, outsamp;
                             //First order interpolator
-                            ALfloat sample = (ALfloat)((ALshort)(((Data[k]*((1L<<FRACTIONBITS)-fraction))+(Data[k+1]*(fraction)))>>FRACTIONBITS));
+                            sample = (Data[k]*((1<<FRACTIONBITS)-fraction) +
+                                      Data[k+1]*fraction) >> FRACTIONBITS;
+                            lowsamp = lpFilter(Filter, sample);
 
                             //Direct path final mix buffer and panning
-                            DrySample = aluComputeSample(DryGainHF, sample, DrySample);
-                            DryBuffer[j][FRONT_LEFT]  += DrySample*DrySend[FRONT_LEFT];
-                            DryBuffer[j][FRONT_RIGHT] += DrySample*DrySend[FRONT_RIGHT];
-                            DryBuffer[j][SIDE_LEFT]   += DrySample*DrySend[SIDE_LEFT];
-                            DryBuffer[j][SIDE_RIGHT]  += DrySample*DrySend[SIDE_RIGHT];
-                            DryBuffer[j][BACK_LEFT]   += DrySample*DrySend[BACK_LEFT];
-                            DryBuffer[j][BACK_RIGHT]  += DrySample*DrySend[BACK_RIGHT];
+                            outsamp = aluComputeSample(DryGainHF, sample, lowsamp);
+                            DryBuffer[j][FRONT_LEFT]  += outsamp*DrySend[FRONT_LEFT];
+                            DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT];
+                            DryBuffer[j][SIDE_LEFT]   += outsamp*DrySend[SIDE_LEFT];
+                            DryBuffer[j][SIDE_RIGHT]  += outsamp*DrySend[SIDE_RIGHT];
+                            DryBuffer[j][BACK_LEFT]   += outsamp*DrySend[BACK_LEFT];
+                            DryBuffer[j][BACK_RIGHT]  += outsamp*DrySend[BACK_RIGHT];
                             //Room path final mix buffer and panning
-                            WetSample = aluComputeSample(WetGainHF, sample, WetSample);
-                            if(doReverb)
-                                ReverbBuffer[j] += WetSample;
-                            else
-                            {
-                                WetBuffer[j][FRONT_LEFT]  += WetSample*WetSend[FRONT_LEFT];
-                                WetBuffer[j][FRONT_RIGHT] += WetSample*WetSend[FRONT_RIGHT];
-                                WetBuffer[j][SIDE_LEFT]   += WetSample*WetSend[SIDE_LEFT];
-                                WetBuffer[j][SIDE_RIGHT]  += WetSample*WetSend[SIDE_RIGHT];
-                                WetBuffer[j][BACK_LEFT]   += WetSample*WetSend[BACK_LEFT];
-                                WetBuffer[j][BACK_RIGHT]  += WetSample*WetSend[BACK_RIGHT];
-                            }
+                            outsamp = aluComputeSample(WetGainHF, sample, lowsamp);
+                            WetBuffer[j][FRONT_LEFT]  += outsamp*WetSend[FRONT_LEFT];
+                            WetBuffer[j][FRONT_RIGHT] += outsamp*WetSend[FRONT_RIGHT];
+                            WetBuffer[j][SIDE_LEFT]   += outsamp*WetSend[SIDE_LEFT];
+                            WetBuffer[j][SIDE_RIGHT]  += outsamp*WetSend[SIDE_RIGHT];
+                            WetBuffer[j][BACK_LEFT]   += outsamp*WetSend[BACK_LEFT];
+                            WetBuffer[j][BACK_RIGHT]  += outsamp*WetSend[BACK_RIGHT];
                         }
                         else
                         {
@@ -816,8 +851,6 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
                     //Update source info
                     ALSource->position = DataPosInt;
                     ALSource->position_fraction = DataPosFrac;
-                    ALSource->LastDrySample = DrySample;
-                    ALSource->LastWetSample = WetSample;
                 }
 
                 //Handle looping sources
@@ -907,27 +940,26 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
                 ALuint Length = ALEffectSlot->ReverbLength;
                 ALfloat DecayGain = ALEffectSlot->ReverbDecayGain;
                 ALfloat DecayHFRatio = ALEffectSlot->effect.Reverb.DecayHFRatio;
-                ALfloat Gain = ALEffectSlot->effect.Reverb.Gain;
                 ALfloat ReflectGain = ALEffectSlot->effect.Reverb.ReflectionsGain;
                 ALfloat LateReverbGain = ALEffectSlot->effect.Reverb.LateReverbGain;
-                ALfloat LastDecaySample = ALEffectSlot->LastDecaySample;
-                ALfloat sample;
+                ALfloat sample, lowsample;
 
+                Filter = &ALEffectSlot->iirFilter;
                 for(i = 0;i < SamplesToDo;i++)
                 {
-                    DelayBuffer[Pos] = ReverbBuffer[i] * Gain;
+                    sample  = WetBuffer[i][FRONT_LEFT] +WetBuffer[i][SIDE_LEFT] +WetBuffer[i][BACK_LEFT];
+                    sample += WetBuffer[i][FRONT_RIGHT]+WetBuffer[i][SIDE_RIGHT]+WetBuffer[i][BACK_RIGHT];
+                    DelayBuffer[Pos] = sample / 6.0f;
 
                     sample = DelayBuffer[ReflectPos] * ReflectGain;
 
                     DelayBuffer[LatePos] *= LateReverbGain;
 
                     Pos = (Pos+1) % Length;
-                    DelayBuffer[Pos] *= DecayHFRatio;
-                    DelayBuffer[Pos] += LastDecaySample * (1.0f-DecayHFRatio);
-                    LastDecaySample = DelayBuffer[Pos];
-                    DelayBuffer[Pos] *= DecayGain;
+                    lowsample = lpFilter(Filter, DelayBuffer[Pos]);
+                    lowsample += (DelayBuffer[Pos]-lowsample) * DecayHFRatio;
 
-                    DelayBuffer[LatePos] += DelayBuffer[Pos];
+                    DelayBuffer[LatePos] += lowsample * DecayGain;
 
                     sample += DelayBuffer[LatePos];
 
@@ -945,7 +977,6 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
                 ALEffectSlot->ReverbPos = Pos;
                 ALEffectSlot->ReverbLatePos = LatePos;
                 ALEffectSlot->ReverbReflectPos = ReflectPos;
-                ALEffectSlot->LastDecaySample = LastDecaySample;
             }
 
             ALEffectSlot = ALEffectSlot->next;

Alc/alcThread.c

@@ -83,22 +83,21 @@ ALuint StopThread(ALvoid *thread)
 typedef struct {
     ALuint (*func)(ALvoid*);
     ALvoid *ptr;
+    ALuint ret;
     pthread_t thread;
 } ThreadInfo;
 
 static void *StarterFunc(void *ptr)
 {
     ThreadInfo *inf = (ThreadInfo*)ptr;
-    ALint ret;
-
-    ret = inf->func(inf->ptr);
-    return (void*)ret;
+    inf->ret = inf->func(inf->ptr);
+    return NULL;
 }
 
 ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
 {
     ThreadInfo *inf = malloc(sizeof(ThreadInfo));
-    if(!inf) return 0;
+    if(!inf) return NULL;
 
     inf->func = func;
     inf->ptr = ptr;
@@ -114,12 +113,14 @@ ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
 ALuint StopThread(ALvoid *thread)
 {
     ThreadInfo *inf = thread;
-    void *ret;
+    ALuint ret;
+
+    pthread_join(inf->thread, NULL);
+    ret = inf->ret;
 
-    pthread_join(inf->thread, &ret);
     free(inf);
 
-    return (ALuint)ret;
+    return ret;
 }
 
 #endif

Alc/alsa.c

@@ -40,7 +40,7 @@ typedef struct {
     ALvoid *buffer;
     ALsizei size;
 
-    int killNow;
+    volatile int killNow;
     ALvoid *thread;
 } alsa_data;
 

Alc/dsound.c

@@ -48,7 +48,7 @@ typedef struct {
     LPDIRECTSOUNDBUFFER    DSpbuffer;
     LPDIRECTSOUNDBUFFER    DSsbuffer;
 
-    int killNow;
+    volatile int killNow;
     ALvoid *thread;
 } DSoundData;
 
@@ -173,7 +173,7 @@ static ALCboolean DSoundOpenPlayback(ALCdevice *device, const ALCchar *deviceNam
 
     if(SUCCEEDED(hr))
     {
-        if(*(GetConfigValue(NULL, "format", "")) != 0)
+        if(*(GetConfigValue(NULL, "format", "")) == 0)
             hr = IDirectSound_GetSpeakerConfig(pData->lpDS, &speakers);
         else
         {

Alc/lpfilter.c

@@ -0,0 +1,458 @@
+/* ----------------- file filterIIR00.c begin ----------------- */
+/* 
+Resonant low pass filter source code.
+By baltrax@hotmail.com (Zxform)
+*/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <math.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+
+
+static void szxform(
+    double *a0, double *a1, double *a2,   /* numerator coefficients */
+    double *b0, double *b1, double *b2,   /* denominator coefficients */
+    double fc,           /* Filter cutoff frequency */
+    double fs,           /* sampling rate */
+    double *k,           /* overall gain factor */
+    float *coef);        /* pointer to 4 iir coefficients */
+
+/*
+ * --------------------------------------------------------------------
+ *
+ * lpFilter - Perform IIR filtering sample by sample on floats
+ *
+ * Implements cascaded direct form II second order sections.
+ * Requires FILTER structure for history and coefficients.
+ * The size of the history array is 2*FILTER_SECTIONS.
+ * The size of the coefficient array is 4*FILTER_SECTIONS + 1 because
+ * the first coefficient is the overall scale factor for the filter.
+ * Returns one output sample for each input sample.
+ *
+ * float lpFilter(FILTER *iir,float input)
+ *
+ *     FILTER *iir        pointer to FILTER structure
+ *     float input        new float input sample
+ *
+ * Returns float value giving the current output.
+ * --------------------------------------------------------------------
+ */
+
+/*** moved to ALu.c ***/
+
+/*
+ * --------------------------------------------------------------------
+ *
+ * InitLowPassFilter()
+ *
+ * Initialize filter coefficients.
+ * We create a 4th order filter (24 db/oct rolloff), consisting
+ * of two second order sections.
+ * --------------------------------------------------------------------
+ */
+int InitLowPassFilter(ALCcontext *Context, FILTER *iir)
+{
+    float    *coef;
+    double   fs, fc; /* Sampling frequency, cutoff frequency */
+    double   Q;      /* Resonance > 1.0 < 1000 */
+    unsigned nInd;
+    double   a0, a1, a2, b0, b1, b2;
+    double   k;      /* overall gain factor */
+    struct {
+        double a0, a1, a2;       /* numerator coefficients */
+        double b0, b1, b2;       /* denominator coefficients */
+    } ProtoCoef[FILTER_SECTIONS];      /* Filter prototype coefficients,
+                                          1 for each filter section */
+
+
+    /*
+     * Setup filter s-domain coefficients
+     */
+    /* Section 1 */
+    ProtoCoef[0].a0 = 1.0;
+    ProtoCoef[0].a1 = 0;
+    ProtoCoef[0].a2 = 0;
+    ProtoCoef[0].b0 = 1.0;
+    ProtoCoef[0].b1 = 0.765367;
+    ProtoCoef[0].b2 = 1.0;
+
+    /* Section 2 */
+    ProtoCoef[1].a0 = 1.0;
+    ProtoCoef[1].a1 = 0;
+    ProtoCoef[1].a2 = 0;
+    ProtoCoef[1].b0 = 1.0;
+    ProtoCoef[1].b1 = 1.847759;
+    ProtoCoef[1].b2 = 1.0;
+
+    /* Clear the coefficient and history arrays */
+    memset(iir->coef, 0, sizeof(iir->coef));
+    memset(iir->history, 0, sizeof(iir->history));
+
+    k = 1.0;              /* Set overall filter gain */
+    coef = iir->coef + 1; /* Skip k, or gain */
+
+    Q = 1;                   /* Resonance */
+    fc = LOWPASSFREQCUTOFF;  /* Filter cutoff (Hz) */
+    fs = Context->Frequency; /* Sampling frequency (Hz) */
+
+    /*
+     * Compute z-domain coefficients for each biquad section
+     * for new Cutoff Frequency and Resonance
+     */
+    for (nInd = 0; nInd < FILTER_SECTIONS; nInd++)
+    {
+        a0 = ProtoCoef[nInd].a0;
+        a1 = ProtoCoef[nInd].a1;
+        a2 = ProtoCoef[nInd].a2;
+
+        b0 = ProtoCoef[nInd].b0;
+        b1 = ProtoCoef[nInd].b1 / Q;      /* Divide by resonance or Q */
+        b2 = ProtoCoef[nInd].b2;
+        szxform(&a0, &a1, &a2, &b0, &b1, &b2, fc, fs, &k, coef);
+        coef += 4;                       /* Point to next filter section */
+    }
+
+    /* Update overall filter gain in coef array */
+    iir->coef[0] = k;
+
+    return 0;
+}
+/* ----------------- file filterIIR00.c end ----------------- */
+
+
+/* ----------------- file bilinear.c begin ----------------- */
+/*
+ * ----------------------------------------------------------
+ *      bilinear.c
+ *
+ *      Perform bilinear transformation on s-domain coefficients
+ *      of 2nd order biquad section.
+ *      First design an analog filter and use s-domain coefficients
+ *      as input to szxform() to convert them to z-domain.
+ *
+ * Here's the butterworth polinomials for 2nd, 4th and 6th order sections.
+ *      When we construct a 24 db/oct filter, we take to 2nd order
+ *      sections and compute the coefficients separately for each section.
+ *
+ *      n       Polinomials
+ * --------------------------------------------------------------------
+ *      2       s^2 + 1.4142s +1
+ *      4       (s^2 + 0.765367s + 1) (s^2 + 1.847759s + 1)
+ *      6       (s^2 + 0.5176387s + 1) (s^2 + 1.414214 + 1) (s^2 + 1.931852s + 1)
+ *
+ *      Where n is a filter order.
+ *      For n=4, or two second order sections, we have following equasions for each
+ *      2nd order stage:
+ *
+ *      (1 / (s^2 + (1/Q) * 0.765367s + 1)) * (1 / (s^2 + (1/Q) * 1.847759s + 1))
+ *
+ *      Where Q is filter quality factor in the range of
+ *      1 to 1000. The overall filter Q is a product of all
+ *      2nd order stages. For example, the 6th order filter
+ *      (3 stages, or biquads) with individual Q of 2 will
+ *      have filter Q = 2 * 2 * 2 = 8.
+ *
+ *      The nominator part is just 1.
+ *      The denominator coefficients for stage 1 of filter are:
+ *      b2 = 1; b1 = 0.765367; b0 = 1;
+ *      numerator is
+ *      a2 = 0; a1 = 0; a0 = 1;
+ *
+ *      The denominator coefficients for stage 1 of filter are:
+ *      b2 = 1; b1 = 1.847759; b0 = 1;
+ *      numerator is
+ *      a2 = 0; a1 = 0; a0 = 1;
+ *
+ *      These coefficients are used directly by the szxform()
+ *      and bilinear() functions. For all stages the numerator
+ *      is the same and the only thing that is different between
+ *      different stages is 1st order coefficient. The rest of
+ *      coefficients are the same for any stage and equal to 1.
+ *
+ *      Any filter could be constructed using this approach.
+ *
+ *      References:
+ *             Van Valkenburg, "Analog Filter Design"
+ *             Oxford University Press 1982
+ *             ISBN 0-19-510734-9
+ *
+ *             C Language Algorithms for Digital Signal Processing
+ *             Paul Embree, Bruce Kimble
+ *             Prentice Hall, 1991
+ *             ISBN 0-13-133406-9
+ *
+ *             Digital Filter Designer's Handbook
+ *             With C++ Algorithms
+ *             Britton Rorabaugh
+ *             McGraw Hill, 1997
+ *             ISBN 0-07-053806-9
+ * ----------------------------------------------------------
+ */
+
+static void prewarp(double *a0, double *a1, double *a2, double fc, double fs);
+static void bilinear(
+    double a0, double a1, double a2,    /* numerator coefficients */
+    double b0, double b1, double b2,    /* denominator coefficients */
+    double *k,                          /* overall gain factor */
+    double fs,                          /* sampling rate */
+    float *coef);                       /* pointer to 4 iir coefficients */
+
+
+/*
+ * ----------------------------------------------------------
+ *      Pre-warp the coefficients of a numerator or denominator.
+ *      Note that a0 is assumed to be 1, so there is no wrapping
+ *      of it.
+ * ----------------------------------------------------------
+ */
+static void prewarp(
+    double *a0, double *a1, double *a2,
+    double fc, double fs)
+{
+    double wp, pi;
+
+    pi = 4.0 * atan(1.0);
+    wp = 2.0 * fs * tan(pi * fc / fs);
+
+    *a2 = (*a2) / (wp * wp);
+    *a1 = (*a1) / wp;
+    (void)a0;
+}
+
+
+/*
+ * ----------------------------------------------------------
+ * bilinear()
+ *
+ * Transform the numerator and denominator coefficients
+ * of s-domain biquad section into corresponding
+ * z-domain coefficients.
+ *
+ *      Store the 4 IIR coefficients in array pointed by coef
+ *      in following order:
+ *             beta1, beta2    (denominator)
+ *             alpha1, alpha2  (numerator)
+ *
+ * Arguments:
+ *             a0-a2   - s-domain numerator coefficients
+ *             b0-b2   - s-domain denominator coefficients
+ *             k       - filter gain factor. initially set to 1
+ *                       and modified by each biquad section in such
+ *                       a way, as to make it the coefficient by
+ *                       which to multiply the overall filter gain
+ *                       in order to achieve a desired overall filter gain,
+ *                       specified in initial value of k.
+ *             fs      - sampling rate (Hz)
+ *             coef    - array of z-domain coefficients to be filled in.
+ *
+ * Return:
+ *             On return, set coef z-domain coefficients
+ * ----------------------------------------------------------
+ */
+static void bilinear(
+    double a0, double a1, double a2,    /* numerator coefficients */
+    double b0, double b1, double b2,    /* denominator coefficients */
+    double *k,                          /* overall gain factor */
+    double fs,                          /* sampling rate */
+    float *coef                         /* pointer to 4 iir coefficients */
+)
+{
+    double ad, bd;
+
+    /* alpha (Numerator in s-domain) */
+    ad = 4. * a2 * fs * fs + 2. * a1 * fs + a0;
+    /* beta (Denominator in s-domain) */
+    bd = 4. * b2 * fs * fs + 2. * b1* fs + b0;
+
+    /* update gain constant for this section */
+    *k *= ad/bd;
+
+    /* Denominator */
+    *coef++ = (2.*b0 - 8.*b2*fs*fs) / bd;         /* beta1 */
+    *coef++ = (4.*b2*fs*fs - 2.*b1*fs + b0) / bd; /* beta2 */
+
+    /* Nominator */
+    *coef++ = (2.*a0 - 8.*a2*fs*fs) / ad;         /* alpha1 */
+    *coef = (4.*a2*fs*fs - 2.*a1*fs + a0) / ad;   /* alpha2 */
+}
+
+
+/*
+ * ----------------------------------------------------------
+ * Transform from s to z domain using bilinear transform
+ * with prewarp.
+ *
+ * Arguments:
+ *      For argument description look at bilinear()
+ *
+ *      coef - pointer to array of floating point coefficients,
+ *                     corresponding to output of bilinear transofrm
+ *                     (z domain).
+ *
+ * Note: frequencies are in Hz.
+ * ----------------------------------------------------------
+ */
+static void szxform(
+    double *a0, double *a1, double *a2, /* numerator coefficients */
+    double *b0, double *b1, double *b2, /* denominator coefficients */
+    double fc,                          /* Filter cutoff frequency */
+    double fs,                          /* sampling rate */
+    double *k,                          /* overall gain factor */
+    float *coef)                        /* pointer to 4 iir coefficients */
+{
+    /* Calculate a1 and a2 and overwrite the original values */
+    prewarp(a0, a1, a2, fc, fs);
+    prewarp(b0, b1, b2, fc, fs);
+    bilinear(*a0, *a1, *a2, *b0, *b1, *b2, k, fs, coef);
+}
+
+
+/* ----------------- file bilinear.c end ----------------- */
+
+/* ----------------- file filter.txt begin -----------------
+How to construct a kewl low pass resonant filter?
+
+Lets assume we want to create a filter for analog synth.
+The filter rolloff is 24 db/oct, which corresponds to 4th
+order filter. Filter of first order is equivalent to RC circuit
+and has max rolloff of 6 db/oct.
+
+We will use classical Butterworth IIR filter design, as it
+exactly corresponds to our requirements.
+
+A common practice is to chain several 2nd order sections,
+or biquads, as they commonly called, in order to achive a higher
+order filter. Each 2nd order section is a 2nd order filter, which
+has 12 db/oct roloff. So, we need 2 of those sections in series.
+
+To compute those sections, we use standard Butterworth polinomials,
+or so called s-domain representation and convert it into z-domain,
+or digital domain. The reason we need to do this is because
+the filter theory exists for analog filters for a long time
+and there exist no theory of working in digital domain directly.
+So the common practice is to take standard analog filter design
+and use so called bilinear transform to convert the butterworth
+equasion coefficients into z-domain.
+
+Once we compute the z-domain coefficients, we can use them in
+a very simple transfer function, such as iir_filter() in our
+C source code, in order to perform the filtering function.
+The filter itself is the simpliest thing in the world.
+The most complicated thing is computing the coefficients
+for z-domain.
+
+Ok, lets look at butterworth polynomials, arranged as a series
+of 2nd order sections:
+
+ * Note: n is filter order.
+ *
+ *      n       Polynomials
+ *      --------------------------------------------------------------------
+ *      2       s^2 + 1.4142s +1
+ *      4       (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1)
+ *      6       (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1)
+ *
+ * For n=4 we have following equasion for the filter transfer function:
+ *
+ *                     1                              1
+ * T(s) = --------------------------- * ----------------------------
+ *        s^2 + (1/Q) * 0.765367s + 1   s^2 + (1/Q) * 1.847759s + 1
+ *
+
+The filter consists of two 2nd order secions since highest s power is 2.
+Now we can take the coefficients, or the numbers by which s is multiplied
+and plug them into a standard formula to be used by bilinear transform.
+
+Our standard form for each 2nd order secion is:
+
+       a2 * s^2 + a1 * s + a0
+H(s) = ----------------------
+       b2 * s^2 + b1 * s + b0
+
+Note that butterworth nominator is 1 for all filter sections,
+which means s^2 = 0 and s^1 = 0
+
+Lets convert standard butterworth polinomials into this form:
+
+       0 + 0 + 1                  0 + 0 + 1
+-------------------------- * --------------------------
+1 + ((1/Q) * 0.765367) + 1   1 + ((1/Q) * 1.847759) + 1
+
+Section 1:
+a2 = 0; a1 = 0; a0 = 1;
+b2 = 1; b1 = 0.5176387; b0 = 1;
+
+Section 2:
+a2 = 0; a1 = 0; a0 = 1;
+b2 = 1; b1 = 1.847759; b0 = 1;
+
+That Q is filter quality factor or resonance, in the range of
+1 to 1000. The overall filter Q is a product of all 2nd order stages.
+For example, the 6th order filter (3 stages, or biquads)
+with individual Q of 2 will have filter Q = 2 * 2 * 2 = 8.
+
+These a and b coefficients are used directly by the szxform()
+and bilinear() functions.
+
+The transfer function for z-domain is:
+
+       1 + alpha1 * z^(-1) + alpha2 * z^(-2)
+H(z) = -------------------------------------
+       1 + beta1 * z^(-1) + beta2 * z^(-2)
+
+When you need to change the filter frequency cutoff or resonance,
+or Q, you call the szxform() function with proper a and b
+coefficients and the new filter cutoff frequency or resonance.
+You also need to supply the sampling rate and filter gain you want
+to achive. For our purposes the gain = 1.
+
+We call szxform() function 2 times becase we have 2 filter sections.
+Each call provides different coefficients.
+
+The gain argument to szxform() is a pointer to desired filter
+gain variable.
+
+double k = 1.0;      // overall gain factor
+
+Upon return from each call, the k argument will be set to a value,
+by which to multiply our actual signal in order for the gain
+to be one. On second call to szxform() we provide k that was
+changed by the previous section. During actual audio filtering
+function iir_filter() will use this k
+
+Summary:
+
+Our filter is pretty close to ideal in terms of all relevant
+parameters and filter stability even with extremely large values
+of resonance. This filter design has been verified under all
+variations of parameters and it all appears to work as advertized.
+
+Good luck with it.
+If you ever make a directX wrapper for it, post it to comp.dsp.
+
+
+ *
+ * ----------------------------------------------------------
+ *References:
+ *Van Valkenburg, "Analog Filter Design"
+ *Oxford University Press 1982
+ *ISBN 0-19-510734-9
+ *
+ *C Language Algorithms for Digital Signal Processing
+ *Paul Embree, Bruce Kimble
+ *Prentice Hall, 1991
+ *ISBN 0-13-133406-9
+ *
+ *Digital Filter Designer's Handbook
+ *With C++ Algorithms
+ *Britton Rorabaugh
+ *McGraw Hill, 1997
+ *ISBN 0-07-053806-9
+ * ----------------------------------------------------------
+
+
+
+// ----------------- file filter.txt end ----------------- */

Alc/oss.c

@@ -52,7 +52,7 @@ static char *oss_device_capture;
 
 typedef struct {
     int fd;
-    int killNow;
+    volatile int killNow;
     ALvoid *thread;
 
     ALubyte *mix_data;

Alc/wave.c

@@ -35,7 +35,7 @@ typedef struct {
     ALvoid *buffer;
     ALuint size;
 
-    int killNow;
+    volatile int killNow;
     ALvoid *thread;
 } wave_data;
 

CMakeLists.txt

@@ -1,5 +1,11 @@
 # CMake build file list for OpenAL
 
+CMAKE_MINIMUM_REQUIRED(VERSION 2.4)
+
+IF(COMMAND CMAKE_POLICY)
+  CMAKE_POLICY(SET CMP0003 NEW)
+ENDIF(COMMAND CMAKE_POLICY)
+
 INCLUDE(CheckFunctionExists)
 INCLUDE(CheckLibraryExists)
 INCLUDE(CheckIncludeFile)
@@ -9,7 +15,7 @@ INCLUDE(CheckCSourceCompiles)
 INCLUDE(CheckTypeSize)
 
 
-PROJECT(OpenAL)
+PROJECT(OpenAL C)
 
 
 SET(CMAKE_ALLOW_LOOSE_LOOP_CONSTRUCTS TRUE)
@@ -27,10 +33,19 @@ OPTION(WERROR  "Treat compile warnings as errors"      OFF)
 OPTION(EXAMPLES  "Build example programs"              ON)
 
 
+IF(WIN32)
+    SET(LIBNAME openal32)
+ELSE()
+    SET(LIBNAME openal)
+ENDIF()
+
 SET(LIB_MAJOR_VERSION "1")
-SET(LIB_MINOR_VERSION "4")
-SET(LIB_BUILD_VERSION "270")
+SET(LIB_MINOR_VERSION "5")
+SET(LIB_BUILD_VERSION "304")
 SET(LIB_VERSION "${LIB_MAJOR_VERSION}.${LIB_MINOR_VERSION}.${LIB_BUILD_VERSION}")
+IF(NOT DEFINED LIB_INSTALL_DIR)
+  SET(LIB_INSTALL_DIR "lib")
+ENDIF(NOT DEFINED LIB_INSTALL_DIR)
 
 
 CHECK_TYPE_SIZE("long" SIZEOF_LONG)
@@ -203,7 +218,6 @@ SET(OPENAL_OBJS  OpenAL32/alAuxEffectSlot.c
                  OpenAL32/alSource.c
                  OpenAL32/alState.c
                  OpenAL32/alThunk.c
-                 OpenAL32/OpenAL32.c
 )
 SET(ALC_OBJS  Alc/ALc.c
               Alc/ALu.c
@@ -211,6 +225,7 @@ SET(ALC_OBJS  Alc/ALc.c
               Alc/alcRing.c
               Alc/alcThread.c
               Alc/bs2b.c
+              Alc/lpfilter.c
               Alc/wave.c
 )
 
@@ -283,20 +298,26 @@ ENDIF()
 # This is always available
 SET(BACKENDS  "${BACKENDS} WaveFile")
 
+# Needed for openal.pc.in
+SET(prefix ${CMAKE_INSTALL_PREFIX})
+SET(exec_prefix "\${prefix}")
+SET(libdir "\${exec_prefix}/${LIB_INSTALL_DIR}")
+SET(bindir "\${exec_prefix}/bin")
+SET(includedir "\${prefix}/include")
+SET(PACKAGE_VERSION "${LIB_VERSION}")
+
 # End configuration
 CONFIGURE_FILE(
     "${OpenAL_SOURCE_DIR}/config.h.in"
     "${OpenAL_BINARY_DIR}/config.h")
+CONFIGURE_FILE(
+    "${OpenAL_SOURCE_DIR}/admin/pkgconfig/openal.pc.in"
+    "${OpenAL_BINARY_DIR}/admin/pkgconfig/openal.pc"
+    @ONLY)
 
 ADD_DEFINITIONS(-DAL_BUILD_LIBRARY)
 
 # Build a shared library
-IF(WIN32)
-    SET(LIBNAME openal32)
-ELSE()
-    SET(LIBNAME openal)
-ENDIF()
-
 ADD_LIBRARY(${LIBNAME} SHARED ${OPENAL_OBJS} ${ALC_OBJS})
 SET_TARGET_PROPERTIES(${LIBNAME} PROPERTIES VERSION ${LIB_VERSION}
                                             SOVERSION ${LIB_MAJOR_VERSION})
@@ -309,22 +330,24 @@ TARGET_LINK_LIBRARIES(${LIBNAME} ${EXTRA_LIBS})
 # Add an install target here
 INSTALL(TARGETS ${LIBNAME}
         RUNTIME DESTINATION bin
-        LIBRARY DESTINATION lib
-        ARCHIVE DESTINATION lib
+        LIBRARY DESTINATION ${LIB_INSTALL_DIR}
+        ARCHIVE DESTINATION ${LIB_INSTALL_DIR}
 )
 INSTALL(FILES include/AL/al.h
               include/AL/alc.h
               include/AL/alext.h
         DESTINATION include/AL
 )
+INSTALL(FILES "${OpenAL_BINARY_DIR}/admin/pkgconfig/openal.pc"
+        DESTINATION lib/pkgconfig)
 
 IF(EXAMPLES)
     ADD_EXECUTABLE(openal-info examples/openal-info.c)
     TARGET_LINK_LIBRARIES(openal-info ${LIBNAME})
     INSTALL(TARGETS openal-info
             RUNTIME DESTINATION bin
-            LIBRARY DESTINATION lib
-            ARCHIVE DESTINATION lib
+            LIBRARY DESTINATION ${LIB_INSTALL_DIR}
+            ARCHIVE DESTINATION ${LIB_INSTALL_DIR}
     )
 ENDIF()
 
@@ -332,3 +355,9 @@ MESSAGE(STATUS "")
 MESSAGE(STATUS "Building OpenAL with support for the following backends:")
 MESSAGE(STATUS "    ${BACKENDS}")
 MESSAGE(STATUS "")
+
+IF(WIN32 AND NOT HAVE_DSOUND)
+    MESSAGE(STATUS "WARNING: Building the Windows version without DirectSound output")
+    MESSAGE(STATUS "         This is probably NOT what you want!")
+    MESSAGE(STATUS "")
+ENDIF()

OpenAL32/Include/alAuxEffectSlot.h

@@ -1,8 +1,9 @@
 #ifndef _AL_AUXEFFECTSLOT_H_
 #define _AL_AUXEFFECTSLOT_H_
 
-#include "alEffect.h"
 #include "AL/al.h"
+#include "alEffect.h"
+#include "alFilter.h"
 
 #ifdef __cplusplus
 extern "C" {
@@ -28,7 +29,8 @@ typedef struct ALeffectslot
     ALuint ReverbReflectPos;
     ALuint ReverbLatePos;
     ALfloat ReverbDecayGain;
-    ALfloat LastDecaySample;
+
+    FILTER iirFilter;
 
     ALuint refcount;
 

OpenAL32/Include/alBuffer.h

@@ -18,6 +18,7 @@ typedef struct ALbuffer_struct
     ALshort *data;
     ALsizei  size;
     ALsizei  frequency;
+    ALsizei  padding;
     ALenum   state;
     ALuint   refcount; // Number of sources using this buffer (deletion can only occur when this is 0)
     struct ALbuffer_struct *next;

OpenAL32/Include/alFilter.h

@@ -7,6 +7,13 @@
 extern "C" {
 #endif
 
+#define FILTER_SECTIONS 2   /* 2 filter sections for 24 db/oct filter */
+
+typedef struct {
+    float history[2*FILTER_SECTIONS];  /* history in filter */
+    float coef[4*FILTER_SECTIONS + 1]; /* coefficients of filter */
+} FILTER;
+
 #define AL_FILTER_TYPE                                     0x8001
 
 #define AL_FILTER_NULL                                     0x0000
@@ -48,6 +55,8 @@ AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *pf
 
 ALvoid ReleaseALFilters(ALvoid);
 
+int InitLowPassFilter(ALCcontext *Context, FILTER *iir);
+
 #ifdef __cplusplus
 }
 #endif

OpenAL32/Include/alMain.h

@@ -103,8 +103,6 @@ static inline void Sleep(ALuint t)
 extern "C" {
 #endif
 
-extern CRITICAL_SECTION _alMutex;
-
 extern char _alDebug[256];
 
 #define AL_PRINT(...) do {                             \
@@ -125,6 +123,9 @@ extern char _alDebug[256];
 #define SPEEDOFSOUNDMETRESPERSEC   (343.3f)
 #define AIRABSORBGAINHF            (0.994f)
 
+#define LOWPASSFREQCUTOFF          (5000)
+
+
 typedef struct {
     ALCboolean (*OpenPlayback)(ALCdevice*, const ALCchar*);
     void (*ClosePlayback)(ALCdevice*);
@@ -200,7 +201,7 @@ struct ALCcontext_struct
     ALint       lNumStereoSources;
 
     ALCdevice  *Device;
-    ALCchar     ExtensionList[1024];
+    const ALCchar *ExtensionList;
 
     struct bs2b *bs2b;
 

OpenAL32/Include/alSource.h

@@ -70,18 +70,19 @@ typedef struct ALsource
         ALfilter WetFilter;
     } Send[MAX_SENDS];
 
-    ALfloat LastDrySample;
-    ALfloat LastWetSample;
-
     ALboolean DryGainHFAuto;
     ALboolean WetGainAuto;
     ALboolean WetGainHFAuto;
     ALfloat   OuterGainHF;
 
+    FILTER iirFilter;
+
     ALfloat AirAbsorptionFactor;
 
     ALfloat RoomRolloffFactor;
 
+    ALfloat DopplerFactor;
+
     // Index to itself
     ALuint source;
 

OpenAL32/OpenAL32.c

@@ -1,74 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- *  Boston, MA  02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include "alMain.h"
-#include "alBuffer.h"
-#include "alFilter.h"
-#include "alEffect.h"
-#include "alAuxEffectSlot.h"
-#include "alThunk.h"
-
-CRITICAL_SECTION _alMutex;
-
-#ifdef _WIN32
-BOOL APIENTRY DllMain(HANDLE hModule,DWORD ul_reason_for_call,LPVOID lpReserved)
-{
-    (void)lpReserved;
-
-    // Perform actions based on the reason for calling.
-    switch(ul_reason_for_call)
-    {
-        case DLL_PROCESS_ATTACH:
-            DisableThreadLibraryCalls(hModule);
-            break;
-
-        case DLL_PROCESS_DETACH:
-            ReleaseALC();
-            ReleaseALBuffers();
-            ReleaseALEffects();
-            ReleaseALFilters();
-            FreeALConfig();
-            ALTHUNK_EXIT();
-            DeleteCriticalSection(&_alMutex);
-            break;
-    }
-    return TRUE;
-}
-#else
-#ifdef HAVE_GCC_DESTRUCTOR
-static void my_deinit() __attribute__((destructor));
-static void my_deinit()
-{
-    static ALenum once = AL_FALSE;
-    if(once) return;
-    once = AL_TRUE;
-
-    ReleaseALC();
-    ReleaseALBuffers();
-    ReleaseALEffects();
-    ReleaseALFilters();
-    FreeALConfig();
-    ALTHUNK_EXIT();
-    DeleteCriticalSection(&_alMutex);
-}
-#endif
-#endif

OpenAL32/alAuxEffectSlot.c

@@ -71,6 +71,8 @@ AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslo
                         break;
                     }
 
+                    InitLowPassFilter(Context, &(*list)->iirFilter);
+
                     (*list)->Gain = 1.0;
                     (*list)->AuxSendAuto = AL_TRUE;
                     (*list)->refcount = 0;
@@ -522,7 +524,7 @@ ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context)
 {
 #ifdef _DEBUG
     if(Context->AuxiliaryEffectSlotCount > 0)
-        AL_PRINT("alcDestroyContext(): %d AuxiliaryEffectSlot(s) NOT deleted\n", Context->AuxiliaryEffectSlotCount);
+        AL_PRINT("alcDestroyContext(): deleting %d AuxiliaryEffectSlot(s)\n", Context->AuxiliaryEffectSlotCount);
 #endif
 
     while(Context->AuxiliaryEffectSlot)

OpenAL32/alBuffer.c

@@ -142,63 +142,59 @@ ALAPI ALvoid ALAPIENTRY alDeleteBuffers(ALsizei n, const ALuint *puiBuffers)
     // Check we are actually Deleting some Buffers
     if (n >= 0)
     {
-        if ((ALuint)n <= g_uiBufferCount)
+        // Check that all the buffers are valid and can actually be deleted
+        for (i = 0; i < n; i++)
         {
-            // Check that all the buffers are valid and can actually be deleted
-            for (i = 0; i < n; i++)
+            // Check for valid Buffer ID (can be NULL buffer)
+            if (alIsBuffer(puiBuffers[i]))
             {
-                // Check for valid Buffer ID (can be NULL buffer)
-                if (alIsBuffer(puiBuffers[i]))
+                // If not the NULL buffer, check that the reference count is 0
+                ALBuf = ((ALbuffer *)ALTHUNK_LOOKUPENTRY(puiBuffers[i]));
+                if (ALBuf)
                 {
-                    // If not the NULL buffer, check that the reference count is 0
-                    ALBuf = ((ALbuffer *)ALTHUNK_LOOKUPENTRY(puiBuffers[i]));
-                    if (ALBuf)
+                    if (ALBuf->refcount != 0)
                     {
-                        if (ALBuf->refcount != 0)
-                        {
-                            // Buffer still in use, cannot be deleted
-                            alSetError(AL_INVALID_OPERATION);
-                            bFailed = AL_TRUE;
-                        }
+                        // Buffer still in use, cannot be deleted
+                        alSetError(AL_INVALID_OPERATION);
+                        bFailed = AL_TRUE;
                     }
                 }
-                else
-                {
-                    // Invalid Buffer
-                    alSetError(AL_INVALID_NAME);
-                    bFailed = AL_TRUE;
-                }
             }
-
-            // If all the Buffers were valid (and have Reference Counts of 0), then we can delete them
-            if (!bFailed)
+            else
             {
-                for (i = 0; i < n; i++)
+                // Invalid Buffer
+                alSetError(AL_INVALID_NAME);
+                bFailed = AL_TRUE;
+            }
+        }
+
+        // If all the Buffers were valid (and have Reference Counts of 0), then we can delete them
+        if (!bFailed)
+        {
+            for (i = 0; i < n; i++)
+            {
+                if (puiBuffers[i] && alIsBuffer(puiBuffers[i]))
                 {
+                    ALbuffer **list = &g_pBuffers;
+
                     ALBuf=((ALbuffer *)ALTHUNK_LOOKUPENTRY(puiBuffers[i]));
-                    if (ALBuf)
-                    {
-                        ALbuffer **list = &g_pBuffers;
-                        while(*list && *list != ALBuf)
-                            list = &(*list)->next;
+                    while(*list && *list != ALBuf)
+                        list = &(*list)->next;
 
-                        if(*list)
-                            *list = (*list)->next;
+                    if(*list)
+                        *list = (*list)->next;
 
-                        // Release the memory used to store audio data
-                        free(ALBuf->data);
+                    // Release the memory used to store audio data
+                    free(ALBuf->data);
 
-                        // Release buffer structure
-                        ALTHUNK_REMOVEENTRY(puiBuffers[i]);
-                        memset(ALBuf, 0, sizeof(ALbuffer));
-                        g_uiBufferCount--;
-                        free(ALBuf);
-                    }
+                    // Release buffer structure
+                    ALTHUNK_REMOVEENTRY(puiBuffers[i]);
+                    memset(ALBuf, 0, sizeof(ALbuffer));
+                    g_uiBufferCount--;
+                    free(ALBuf);
                 }
             }
         }
-        else
-            alSetError(AL_INVALID_NAME);
     }
     else
         alSetError(AL_INVALID_VALUE);
@@ -266,7 +262,9 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
     ALint RightSample,RightIndex;
     ALuint LeftIMACode,RightIMACode;
     ALbuffer *ALBuf;
+    ALsizei padding;
     ALsizei i,j,k;
+    ALvoid *temp;
 
     Context = alcGetCurrentContext();
     SuspendContext(Context);
@@ -308,6 +306,9 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                         break;
                     }
 
+                    padding = freq / LOWPASSFREQCUTOFF;
+                    if(padding < 1) padding = 1;
+
                     switch(OrigBytes)
                     {
                     case 1:
@@ -315,10 +316,10 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                         size *= 2;
 
                         // 8bit Samples are converted to 16 bit here
-                        // Allocate 8 extra samples
-                        ALBuf->data = realloc(ALBuf->data, (8*NewChannels + size) * (1*sizeof(ALshort)));
-                        if (ALBuf->data)
+                        temp = realloc(ALBuf->data, (padding*NewChannels + size) * (1*sizeof(ALshort)));
+                        if (temp)
                         {
+                            ALBuf->data = temp;
                             for (i = 0;i < size;i+=4)
                             {
                                 ALBuf->data[i+0] = 0;
@@ -326,12 +327,13 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                                 ALBuf->data[i+2] = (ALshort)((((ALubyte*)data)[i/2+0]-128) << 8);
                                 ALBuf->data[i+3] = (ALshort)((((ALubyte*)data)[i/2+1]-128) << 8);
                             }
-                            memset(&(ALBuf->data[size]), 0, 16*NewChannels);
+                            memset(&(ALBuf->data[size]), 0, padding*NewChannels*2);
 
                             ALBuf->format = NewFormat;
                             ALBuf->eOriginalFormat = format;
                             ALBuf->size = size*1*sizeof(ALshort);
                             ALBuf->frequency = freq;
+                            ALBuf->padding = padding;
                         }
                         else
                             alSetError(AL_OUT_OF_MEMORY);
@@ -341,10 +343,10 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                         size /= sizeof(ALshort);
                         size *= 2;
 
-                        // Allocate 8 extra samples
-                        ALBuf->data = realloc(ALBuf->data, (8*NewChannels + size) * (1*sizeof(ALshort)));
-                        if (ALBuf->data)
+                        temp = realloc(ALBuf->data, (padding*NewChannels + size) * (1*sizeof(ALshort)));
+                        if (temp)
                         {
+                            ALBuf->data = temp;
                             for (i = 0;i < size;i+=4)
                             {
                                 ALBuf->data[i+0] = 0;
@@ -352,12 +354,13 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                                 ALBuf->data[i+2] = ((ALshort*)data)[i/2+0];
                                 ALBuf->data[i+3] = ((ALshort*)data)[i/2+1];
                             }
-                            memset(&(ALBuf->data[size]), 0, 16*NewChannels);
+                            memset(&(ALBuf->data[size]), 0, padding*NewChannels*2);
 
                             ALBuf->format = NewFormat;
                             ALBuf->eOriginalFormat = format;
                             ALBuf->size = size*1*sizeof(ALshort);
                             ALBuf->frequency = freq;
+                            ALBuf->padding = padding;
                         }
                         else
                             alSetError(AL_OUT_OF_MEMORY);
@@ -367,11 +370,11 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                         size /= sizeof(ALfloat);
                         size *= 2;
 
-                        // Allocate 8 extra samples
-                        ALBuf->data = realloc(ALBuf->data, (8*NewChannels + size) * (1*sizeof(ALshort)));
-                        if (ALBuf->data)
+                        temp = realloc(ALBuf->data, (padding*NewChannels + size) * (1*sizeof(ALshort)));
+                        if (temp)
                         {
                             ALint smp;
+                            ALBuf->data = temp;
                             for (i = 0;i < size;i+=4)
                             {
                                 ALBuf->data[i+0] = 0;
@@ -385,12 +388,13 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                                 smp = max(smp, -32768);
                                 ALBuf->data[i+3] = (ALshort)smp;
                             }
-                            memset(&(ALBuf->data[size]), 0, 16*NewChannels);
+                            memset(&(ALBuf->data[size]), 0, padding*NewChannels*2);
 
                             ALBuf->format = NewFormat;
                             ALBuf->eOriginalFormat = format;
                             ALBuf->size = size*1*sizeof(ALshort);
                             ALBuf->frequency = freq;
+                            ALBuf->padding = padding;
                         }
                         else
                             alSetError(AL_OUT_OF_MEMORY);
@@ -428,15 +432,19 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                     break;
 
                 case AL_FORMAT_MONO_IMA4:
+                    padding = freq / LOWPASSFREQCUTOFF;
+                    if(padding < 1) padding = 1;
+
                     // Here is where things vary:
                     // nVidia and Apple use 64+1 samples per block => block_size=36 bytes
                     // Most PC sound software uses 2040+1 samples per block -> block_size=1024 bytes
                     if ((size%36) == 0)
                     {
-                        // Allocate 8 extra samples (16 bytes)
-                        ALBuf->data=realloc(ALBuf->data,16+(size/36)*(65*sizeof(ALshort)));
-                        if (ALBuf->data)
+                        // Allocate extra padding samples
+                        temp=realloc(ALBuf->data,padding*2+(size/36)*(65*sizeof(ALshort)));
+                        if (temp)
                         {
+                            ALBuf->data = temp;
                             ALBuf->format = AL_FORMAT_MONO16;
                             ALBuf->eOriginalFormat = AL_FORMAT_MONO_IMA4;
                             IMAData=(ALuint *)data;
@@ -478,9 +486,10 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                                     IMAData++;
                                 }
                             }
-                            memset(&(ALBuf->data[(size/36*65)]), 0, 16);
+                            memset(&(ALBuf->data[(size/36*65)]), 0, padding*2);
                             ALBuf->size=size/36*65*sizeof(ALshort);
                             ALBuf->frequency=freq;
+                            ALBuf->padding=padding;
                         }
                         else
                             alSetError(AL_OUT_OF_MEMORY);
@@ -490,15 +499,19 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                     break;
 
                 case AL_FORMAT_STEREO_IMA4:
+                    padding = freq / LOWPASSFREQCUTOFF;
+                    if(padding < 1) padding = 1;
+
                     // Here is where things vary:
                     // nVidia and Apple use 64+1 samples per channel per block => block_size=72 bytes
                     // Most PC sound software uses 2040+1 samples per channel per block -> block_size=2048 bytes
                     if ((size%72) == 0)
                     {
-                        // Allocate 8 extra samples (32 bytes)
-                        ALBuf->data=realloc(ALBuf->data,32+(size/72)*(2*65*sizeof(ALshort)));
-                        if (ALBuf->data)
+                        // Allocate extra padding samples
+                        temp=realloc(ALBuf->data,padding*2*2+(size/72)*(2*65*sizeof(ALshort)));
+                        if (temp)
                         {
+                            ALBuf->data = temp;
                             ALBuf->format = AL_FORMAT_STEREO16;
                             ALBuf->eOriginalFormat = AL_FORMAT_STEREO_IMA4;
                             IMAData=(ALuint *)data;
@@ -569,9 +582,10 @@ ALAPI ALvoid ALAPIENTRY alBufferData(ALuint buffer,ALenum format,const ALvoid *d
                                     IMAData+=2;
                                 }
                             }
-                            memset(&(ALBuf->data[(size/72*2*65)]), 0, 32);
+                            memset(&(ALBuf->data[(size/72*2*65)]), 0, padding*2*2);
                             ALBuf->size=size/72*2*65*sizeof(ALshort);
                             ALBuf->frequency=freq;
+                            ALBuf->padding=padding;
                         }
                         else
                             alSetError(AL_OUT_OF_MEMORY);
@@ -997,6 +1011,8 @@ static void LoadData(ALbuffer *ALBuf, const ALubyte *data, ALsizei size, ALuint
     ALuint NewChannels = aluChannelsFromFormat(NewFormat);
     ALuint OrigBytes = aluBytesFromFormat(OrigFormat);
     ALuint OrigChannels = aluChannelsFromFormat(OrigFormat);
+    ALsizei padding = freq / LOWPASSFREQCUTOFF;
+    ALvoid *temp;
     ALsizei i;
 
     assert(aluBytesFromFormat(NewFormat) == 2);
@@ -1008,24 +1024,29 @@ static void LoadData(ALbuffer *ALBuf, const ALubyte *data, ALsizei size, ALuint
         return;
     }
 
+    /* Ensure at least one padding byte for the bilinear filter */
+    if(padding < 1)
+        padding = 1;
+
     switch(OrigBytes)
     {
     case 1:
         size /= sizeof(ALubyte);
 
         // 8bit Samples are converted to 16 bit here
-        // Allocate 8 extra samples
-        ALBuf->data = realloc(ALBuf->data, (8*NewChannels + size) * (1*sizeof(ALshort)));
-        if (ALBuf->data)
+        temp = realloc(ALBuf->data, (padding*NewChannels + size) * (1*sizeof(ALshort)));
+        if (temp)
         {
+            ALBuf->data = temp;
             for (i = 0;i < size;i++)
                 ALBuf->data[i] = (ALshort)((data[i]-128) << 8);
-            memset(&(ALBuf->data[size]), 0, 16*NewChannels);
+            memset(&(ALBuf->data[size]), 0, padding*NewChannels*2);
 
             ALBuf->format = NewFormat;
             ALBuf->eOriginalFormat = OrigFormat;
             ALBuf->size = size*1*sizeof(ALshort);
             ALBuf->frequency = freq;
+            ALBuf->padding = padding;
         }
         else
             alSetError(AL_OUT_OF_MEMORY);
@@ -1035,16 +1056,18 @@ static void LoadData(ALbuffer *ALBuf, const ALubyte *data, ALsizei size, ALuint
         size /= sizeof(ALshort);
 
         // Allocate 8 extra samples
-        ALBuf->data = realloc(ALBuf->data, (8*NewChannels + size) * (1*sizeof(ALshort)));
-        if (ALBuf->data)
+        temp = realloc(ALBuf->data, (padding*NewChannels + size) * (1*sizeof(ALshort)));
+        if (temp)
         {
+            ALBuf->data = temp;
             memcpy(ALBuf->data, data, size*1*sizeof(ALshort));
-            memset(&(ALBuf->data[size]), 0, 16*NewChannels);
+            memset(&(ALBuf->data[size]), 0, padding*NewChannels*2);
 
             ALBuf->format = NewFormat;
             ALBuf->eOriginalFormat = OrigFormat;
             ALBuf->size = size*1*sizeof(ALshort);
             ALBuf->frequency = freq;
+            ALBuf->padding = padding;
         }
         else
             alSetError(AL_OUT_OF_MEMORY);
@@ -1054,10 +1077,11 @@ static void LoadData(ALbuffer *ALBuf, const ALubyte *data, ALsizei size, ALuint
         size /= sizeof(ALfloat);
 
         // Allocate 8 extra samples
-        ALBuf->data = realloc(ALBuf->data, (8*NewChannels + size) * (1*sizeof(ALshort)));
-        if (ALBuf->data)
+        temp = realloc(ALBuf->data, (padding*NewChannels + size) * (1*sizeof(ALshort)));
+        if (temp)
         {
             ALint smp;
+            ALBuf->data = temp;
             for (i = 0;i < size;i++)
             {
                 smp = (((ALfloat*)data)[i] * 32767.5f - 0.5f);
@@ -1065,12 +1089,13 @@ static void LoadData(ALbuffer *ALBuf, const ALubyte *data, ALsizei size, ALuint
                 smp = max(smp, -32768);
                 ALBuf->data[i] = (ALshort)smp;
             }
-            memset(&(ALBuf->data[size]), 0, 16*NewChannels);
+            memset(&(ALBuf->data[size]), 0, padding*NewChannels*2);
 
             ALBuf->format = NewFormat;
             ALBuf->eOriginalFormat = OrigFormat;
             ALBuf->size = size*1*sizeof(ALshort);
             ALBuf->frequency = freq;
+            ALBuf->padding = padding;
         }
         else
             alSetError(AL_OUT_OF_MEMORY);
@@ -1094,7 +1119,7 @@ ALvoid ReleaseALBuffers(ALvoid)
 
 #ifdef _DEBUG
     if(g_uiBufferCount > 0)
-        AL_PRINT("exit() %d Buffer(s) NOT deleted\n", g_uiBufferCount);
+        AL_PRINT("exit(): deleting %d Buffer(s)\n", g_uiBufferCount);
 #endif
 
     ALBuffer = g_pBuffers;

OpenAL32/alEffect.c

@@ -590,7 +590,7 @@ ALvoid ReleaseALEffects(ALvoid)
 {
 #ifdef _DEBUG
     if(g_EffectCount > 0)
-        AL_PRINT("exit() %d Effect(s) NOT deleted\n", g_EffectCount);
+        AL_PRINT("exit(): deleting %d Effect(s)\n", g_EffectCount);
 #endif
 
     while(g_EffectList)

OpenAL32/alFilter.c

@@ -221,26 +221,29 @@ AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat flValue
     {
         ALfilter *ALFilter = (ALfilter*)ALTHUNK_LOOKUPENTRY(filter);
 
-        switch(param)
+        switch(ALFilter->type)
         {
-        case AL_LOWPASS_GAIN:
-            if(ALFilter->type == AL_FILTER_LOWPASS)
+        case AL_FILTER_LOWPASS:
+            switch(param)
             {
+            case AL_LOWPASS_GAIN:
                 if(flValue >= 0.0f && flValue <= 1.0f)
                     ALFilter->Gain = flValue;
-            }
-            else
-                alSetError(AL_INVALID_ENUM);
-            break;
+                else
+                    alSetError(AL_INVALID_VALUE);
+                break;
 
-        case AL_LOWPASS_GAINHF:
-            if(ALFilter->type == AL_FILTER_LOWPASS)
-            {
+            case AL_LOWPASS_GAINHF:
                 if(flValue >= 0.0f && flValue <= 1.0f)
                     ALFilter->GainHF = flValue;
-            }
-            else
+                else
+                    alSetError(AL_INVALID_VALUE);
+                break;
+
+            default:
                 alSetError(AL_INVALID_ENUM);
+                break;
+            }
             break;
 
         default:
@@ -265,13 +268,8 @@ AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, ALfloat *pflVa
     {
         switch(param)
         {
-        case AL_LOWPASS_GAIN:
-        case AL_LOWPASS_GAINHF:
-            alFilterf(filter, param, pflValues[0]);
-            break;
-
         default:
-            alSetError(AL_INVALID_ENUM);
+            alFilterf(filter, param, pflValues[0]);
             break;
         }
     }
@@ -285,15 +283,19 @@ AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *piVal
 {
     ALCcontext *Context;
 
-    (void)piValue;
-
     Context = alcGetCurrentContext();
     SuspendContext(Context);
 
     if (filter && alIsFilter(filter))
     {
+        ALfilter *ALFilter = (ALfilter*)ALTHUNK_LOOKUPENTRY(filter);
+
         switch(param)
         {
+        case AL_FILTER_TYPE:
+            *piValue = ALFilter->type;
+            break;
+
         default:
             alSetError(AL_INVALID_ENUM);
             break;
@@ -309,8 +311,6 @@ AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *piVa
 {
     ALCcontext *Context;
 
-    (void)piValues;
-
     Context = alcGetCurrentContext();
     SuspendContext(Context);
 
@@ -318,6 +318,10 @@ AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *piVa
     {
         switch(param)
         {
+        case AL_FILTER_TYPE:
+            alGetFilteri(filter, param, piValues);
+            break;
+
         default:
             alSetError(AL_INVALID_ENUM);
             break;
@@ -340,20 +344,23 @@ AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *pfl
     {
         ALfilter *ALFilter = (ALfilter*)ALTHUNK_LOOKUPENTRY(filter);
 
-        switch(param)
+        switch(ALFilter->type)
         {
-        case AL_LOWPASS_GAIN:
-            if(ALFilter->type == AL_FILTER_LOWPASS)
+        case AL_FILTER_LOWPASS:
+            switch(param)
+            {
+            case AL_LOWPASS_GAIN:
                 *pflValue = ALFilter->Gain;
-            else
-                alSetError(AL_INVALID_ENUM);
-            break;
+                break;
 
-        case AL_LOWPASS_GAINHF:
-            if(ALFilter->type == AL_FILTER_LOWPASS)
+            case AL_LOWPASS_GAINHF:
                 *pflValue = ALFilter->GainHF;
-            else
+                break;
+
+            default:
                 alSetError(AL_INVALID_ENUM);
+                break;
+            }
             break;
 
         default:
@@ -378,13 +385,8 @@ AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *pf
     {
         switch(param)
         {
-        case AL_LOWPASS_GAIN:
-        case AL_LOWPASS_GAINHF:
-            alGetFilterf(filter, param, pflValues);
-            break;
-
         default:
-            alSetError(AL_INVALID_ENUM);
+            alGetFilterf(filter, param, pflValues);
             break;
         }
     }
@@ -399,7 +401,7 @@ ALvoid ReleaseALFilters(ALvoid)
 {
 #ifdef _DEBUG
     if(g_FilterCount > 0)
-        AL_PRINT("exit() %d Filter(s) NOT deleted\n", g_FilterCount);
+        AL_PRINT("exit(): deleting %d Filter(s)\n", g_FilterCount);
 #endif
 
     while(g_FilterList)

OpenAL32/alSource.c

@@ -75,6 +75,8 @@ ALAPI ALvoid ALAPIENTRY alGenSources(ALsizei n,ALuint *sources)
                                 break;
                             }
 
+                            InitLowPassFilter(Context, &(*list)->iirFilter);
+
                             sources[i] = (ALuint)ALTHUNK_ADDENTRY(*list);
                             (*list)->source = sources[i];
 
@@ -137,73 +139,64 @@ ALAPI ALvoid ALAPIENTRY alDeleteSources(ALsizei n, const ALuint *sources)
 
             if (Device)
             {
-                if ((ALuint)n <= Context->SourceCount)
+                // Check that all Sources are valid (and can therefore be deleted)
+                for (i = 0; i < n; i++)
                 {
-                    // Check that all Sources are valid (and can therefore be deleted)
-                    for (i = 0; i < n; i++)
+                    if (!alIsSource(sources[i]))
                     {
-                        if (!alIsSource(sources[i]))
-                        {
-                            alSetError(AL_INVALID_NAME);
-                            bSourcesValid = AL_FALSE;
-                            break;
-                        }
-                    }
-
-                    if (bSourcesValid)
-                    {
-                        // All Sources are valid, and can be deleted
-                        for (i = 0; i < n; i++)
-                        {
-                            // Recheck that the Source is valid, because there could be duplicated Source names
-                            if (alIsSource(sources[i]))
-                            {
-                                ALSource=((ALsource *)ALTHUNK_LOOKUPENTRY(sources[i]));
-                                alSourceStop((ALuint)ALSource->source);
-
-                                // For each buffer in the source's queue, decrement its reference counter and remove it
-                                while (ALSource->queue != NULL)
-                                {
-                                    ALBufferList = ALSource->queue;
-                                    // Decrement buffer's reference counter
-                                    if (ALBufferList->buffer != 0)
-                                        ((ALbuffer*)(ALTHUNK_LOOKUPENTRY(ALBufferList->buffer)))->refcount--;
-                                    // Update queue to point to next element in list
-                                    ALSource->queue = ALBufferList->next;
-                                    // Release memory allocated for buffer list item
-                                    free(ALBufferList);
-                                }
-
-                                for(j = 0;j < MAX_SENDS;++j)
-                                {
-                                    if(ALSource->Send[j].Slot)
-                                        ALSource->Send[j].Slot->refcount--;
-                                    ALSource->Send[j].Slot = NULL;
-                                }
-
-                                // Decrement Source count
-                                Context->SourceCount--;
-
-                                // Remove Source from list of Sources
-                                list = &Context->Source;
-                                while(*list && *list != ALSource)
-                                    list = &(*list)->next;
-
-                                if(*list)
-                                    *list = (*list)->next;
-                                ALTHUNK_REMOVEENTRY(ALSource->source);
-
-                                memset(ALSource,0,sizeof(ALsource));
-                                free(ALSource);
-                            }
-                        }
-
+                        alSetError(AL_INVALID_NAME);
+                        bSourcesValid = AL_FALSE;
+                        break;
                     }
                 }
-                else
+
+                if (bSourcesValid)
                 {
-                    // Trying to delete more Sources than have been generated
-                    alSetError(AL_INVALID_NAME);
+                    // All Sources are valid, and can be deleted
+                    for (i = 0; i < n; i++)
+                    {
+                        // Recheck that the Source is valid, because there could be duplicated Source names
+                        if (alIsSource(sources[i]))
+                        {
+                            ALSource=((ALsource *)ALTHUNK_LOOKUPENTRY(sources[i]));
+                            alSourceStop((ALuint)ALSource->source);
+
+                            // For each buffer in the source's queue, decrement its reference counter and remove it
+                            while (ALSource->queue != NULL)
+                            {
+                                ALBufferList = ALSource->queue;
+                                // Decrement buffer's reference counter
+                                if (ALBufferList->buffer != 0)
+                                    ((ALbuffer*)(ALTHUNK_LOOKUPENTRY(ALBufferList->buffer)))->refcount--;
+                                // Update queue to point to next element in list
+                                ALSource->queue = ALBufferList->next;
+                                // Release memory allocated for buffer list item
+                                free(ALBufferList);
+                            }
+
+                            for(j = 0;j < MAX_SENDS;++j)
+                            {
+                                if(ALSource->Send[j].Slot)
+                                    ALSource->Send[j].Slot->refcount--;
+                                ALSource->Send[j].Slot = NULL;
+                            }
+
+                            // Decrement Source count
+                            Context->SourceCount--;
+
+                            // Remove Source from list of Sources
+                            list = &Context->Source;
+                            while(*list && *list != ALSource)
+                                list = &(*list)->next;
+
+                            if(*list)
+                                *list = (*list)->next;
+                            ALTHUNK_REMOVEENTRY(ALSource->source);
+
+                            memset(ALSource,0,sizeof(ALsource));
+                            free(ALSource);
+                        }
+                    }
                 }
             }
             else
@@ -934,6 +927,10 @@ ALAPI ALvoid ALAPIENTRY alGetSourcef(ALuint source, ALenum eParam, ALfloat *pflV
                     *pflValue = pSource->RoomRolloffFactor;
                     break;
 
+                case AL_DOPPLER_FACTOR:
+                    *pflValue = pSource->DopplerFactor;
+                    break;
+
                 default:
                     alSetError(AL_INVALID_ENUM);
                     break;
@@ -1034,6 +1031,7 @@ ALAPI ALvoid ALAPIENTRY alGetSourcefv(ALuint source, ALenum eParam, ALfloat *pfl
                 case AL_MAX_GAIN:
                 case AL_MAX_DISTANCE:
                 case AL_ROLLOFF_FACTOR:
+                case AL_DOPPLER_FACTOR:
                 case AL_CONE_OUTER_GAIN:
                 case AL_SEC_OFFSET:
                 case AL_SAMPLE_OFFSET:
@@ -1184,6 +1182,10 @@ ALAPI ALvoid ALAPIENTRY alGetSourcei(ALuint source, ALenum eParam, ALint *plValu
                     *plValue = pSource->WetGainHFAuto;
                     break;
 
+                case AL_DOPPLER_FACTOR:
+                    *plValue = (ALint)pSource->DopplerFactor;
+                    break;
+
                 default:
                     alSetError(AL_INVALID_ENUM);
                     break;
@@ -1291,6 +1293,7 @@ ALAPI void ALAPIENTRY alGetSourceiv(ALuint source, ALenum eParam, ALint* plValue
                 case AL_BYTE_OFFSET:
                 case AL_MAX_DISTANCE:
                 case AL_ROLLOFF_FACTOR:
+                case AL_DOPPLER_FACTOR:
                 case AL_REFERENCE_DISTANCE:
                 case AL_SOURCE_TYPE:
                 case AL_DIRECT_FILTER:
@@ -1980,6 +1983,7 @@ static ALvoid InitSourceParams(ALsource *pSource)
     pSource->WetGainHFAuto = AL_TRUE;
     pSource->AirAbsorptionFactor = 0.0f;
     pSource->RoomRolloffFactor = 0.0f;
+    pSource->DopplerFactor = 1.0f;
 
     pSource->state = AL_INITIAL;
     pSource->lSourceType = AL_UNDETERMINED;
@@ -2288,7 +2292,7 @@ ALvoid ReleaseALSources(ALCcontext *Context)
 {
 #ifdef _DEBUG
     if(Context->SourceCount > 0)
-        AL_PRINT("alcDestroyContext(): %d Source(s) NOT deleted\n", Context->SourceCount);
+        AL_PRINT("alcDestroyContext(): deleting %d Source(s)\n", Context->SourceCount);
 #endif
 
     while(Context->Source)

admin/pkgconfig/openal.pc.in

@@ -0,0 +1,11 @@
+prefix=@prefix@
+exec_prefix=@exec_prefix@
+libdir=@libdir@
+includedir=@includedir@
+
+Name: OpenAL
+Description: OpenAL is a cross-platform 3D audio API
+Requires: @PKG_CONFIG_REQUIRES@
+Version: @PACKAGE_VERSION@
+Libs: -L${libdir} -l@LIBNAME@ @PKG_CONFIG_LIBS@
+Cflags: -I${includedir}

alsoftrc.sample

@@ -38,7 +38,7 @@ cf_level = 0  # Sets the crossfeed level for stereo output. Valid values are:
 
 frequency = 44100  # Sets the output frequency. Default is 44100
 
-refresh = 8192  # Sets the buffer size, in frames. Default is 8192. Note that
+refresh = 4096  # Sets the buffer size, in frames. Default is 4096. Note that
                 # the actual granularity may or may not be less than this.
 
 sources = 256  # Sets the maximum number of allocatable sources. Lower values

include/AL/alc.h

@@ -166,11 +166,6 @@ typedef void ALCvoid;
 #define ALC_ATTRIBUTES_SIZE                      0x1002
 #define ALC_ALL_ATTRIBUTES                       0x1003
 
-/**
- * ALC_ENUMERATE_ALL_EXT enums
- */
-#define ALC_DEFAULT_ALL_DEVICES_SPECIFIER        0x1012
-#define ALC_ALL_DEVICES_SPECIFIER                0x1013
 
 /**
  * Capture extension

include/AL/alext.h

@@ -25,10 +25,20 @@
 extern "C" {
 #endif
 
-#ifndef AL_EXT_float32
-#define AL_EXT_float32 1
-#define AL_FORMAT_MONO_FLOAT32                   0x10010
-#define AL_FORMAT_STEREO_FLOAT32                 0x10011
+#ifndef AL_LOKI_IMA_ADPCM_format
+#define AL_LOKI_IMA_ADPCM_format 1
+#define AL_FORMAT_IMA_ADPCM_MONO16_EXT           0x10000
+#define AL_FORMAT_IMA_ADPCM_STEREO16_EXT         0x10001
+#endif
+
+#ifndef AL_LOKI_WAVE_format
+#define AL_LOKI_WAVE_format 1
+#define AL_FORMAT_WAVE_EXT                       0x10002
+#endif
+
+#ifndef AL_EXT_vorbis
+#define AL_EXT_vorbis 1
+#define AL_FORMAT_VORBIS_EXT                     0x10003
 #endif
 
 #ifndef AL_LOKI_quadriphonic
@@ -37,6 +47,25 @@ extern "C" {
 #define AL_FORMAT_QUAD16_LOKI                    0x10005
 #endif
 
+#ifndef AL_EXT_float32
+#define AL_EXT_float32 1
+#define AL_FORMAT_MONO_FLOAT32                   0x10010
+#define AL_FORMAT_STEREO_FLOAT32                 0x10011
+#endif
+
+#ifndef ALC_LOKI_audio_channel
+#define ALC_LOKI_audio_channel 1
+#define ALC_CHAN_MAIN_LOKI                       0x500001
+#define ALC_CHAN_PCM_LOKI                        0x500002
+#define ALC_CHAN_CD_LOKI                         0x500003
+#endif
+
+#ifndef ALC_ENUMERATE_ALL_EXT
+#define ALC_ENUMERATE_ALL_EXT 1
+#define ALC_DEFAULT_ALL_DEVICES_SPECIFIER        0x1012
+#define ALC_ALL_DEVICES_SPECIFIER                0x1013
+#endif
+
 #ifndef AL_EXT_MCFORMATS
 #define AL_EXT_MCFORMATS 1
 #define AL_FORMAT_QUAD8                          0x1204