VWR-2662: OpenAL support (patch by Tofu Linden)

1 files changed, 91 insertions, 180 deletions

diff --git a/linden/indra/llaudio/audioengine_fmod.cpp b/linden/indra/llaudio/audioengine_fmod.cpp
index 16d820c..354ef95 100644
--- a/linden/indra/llaudio/audioengine_fmod.cpp
+++ b/linden/indra/llaudio/audioengine_fmod.cpp
@@ -1,7 +1,6 @@
 /** 
  * @file audioengine_fmod.cpp
- * @brief Implementation of LLAudioEngine class abstracting the audio
- * support as a FMOD 3D implementation
+ * @brief Implementation of LLAudioEngine class abstracting the audio support as a FMOD 3D implementation
  *
  * $LicenseInfo:firstyear=2002&license=viewergpl$
  * 
@@ -46,27 +45,12 @@
 
 #include "sound_ids.h"
 
+extern "C" {
+        void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int length, void* userdata);
+}
 
-void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int length, void* userdata);
 FSOUND_DSPUNIT *gWindDSP = NULL;
 
-// These globals for the wind filter.  Blech!
-F64 gbuf0 = 0.0;
-F64 gbuf1 = 0.0;
-F64 gbuf2 = 0.0;
-F64 gbuf3 = 0.0;
-F64 gbuf4 = 0.0;
-F64 gbuf5 = 0.0;
-F64 gY0 = 0.0;
-F64 gY1 = 0.0;
-
-F32 gTargetGain = 0.f;
-F32 gCurrentGain = 0.f;
-F32 gTargetFreq = 100.f;
-F32 gCurrentFreq = 100.f;
-F32 gTargetPanGainR = 0.5f;
-F32 gCurrentPanGainR = 0.5f;
-
 
 // Safe strcpy
 #if 0 //(unused)  //LL_WINDOWS || LL_LINUX
@@ -94,9 +78,10 @@ static size_t strlcpy( char* dest, const char* src, size_t dst_size )
 
 LLAudioEngine_FMOD::LLAudioEngine_FMOD()
 {
-        mInited = FALSE;
+        mInited = false;
         mCurrentInternetStreamp = NULL;
         mInternetStreamChannel = -1;
+        mWindGen = NULL;
 }
 
 
@@ -105,7 +90,7 @@ LLAudioEngine_FMOD::~LLAudioEngine_FMOD()
 }
 
 
-BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
+bool LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
 {
         mFadeIn = -10000;
 
@@ -124,7 +109,7 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
         {
                 LL_WARNS("AppInit") << "Error : You are using the wrong FMOD version (" << version
                         << ")!  You should be using FMOD " << FMOD_VERSION << LL_ENDL;
-                //return FALSE;
+                //return false;
         }
 
         U32 fmod_flags = 0x0;
@@ -139,7 +124,7 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
         {
                 LL_WARNS("AppInit") << "Error setting FMOD window: "
                         << FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
-                return FALSE;
+                return false;
         }
         // Play audio when we don't have focus.
         // (For example, IM client on top of us.)
@@ -167,7 +152,7 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
         // on top of ALSA is ironically more reliable than raw ALSA.
         // Ack, and ESD has more reliable failure modes - but has worse
         // latency - than all of them, so wins for now.
-        BOOL audio_ok = FALSE;
+        bool audio_ok = false;
 
         if (!audio_ok)
                 if (NULL == getenv("LL_BAD_ESD")) /*Flawfinder: ignore*/
@@ -178,7 +163,7 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
                         {
                                 LL_DEBUGS("AppInit") << "ESD audio output initialized OKAY"
                                         << LL_ENDL;
-                                audio_ok = TRUE;
+                                audio_ok = true;
                         } else {
                                 LL_WARNS("AppInit") << "ESD audio output FAILED to initialize: "
                                         << FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
@@ -195,7 +180,7 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
                            FSOUND_Init(44100, num_channels, fmod_flags))
                         {
                                 LL_DEBUGS("AppInit") << "OSS audio output initialized OKAY" << LL_ENDL;
-                                audio_ok = TRUE;
+                                audio_ok = true;
                         } else {
                                 LL_WARNS("AppInit") << "OSS audio output FAILED to initialize: "
                                         << FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
@@ -212,7 +197,7 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
                            FSOUND_Init(44100, num_channels, fmod_flags))
                         {
                                 LL_DEBUGS("AppInit") << "ALSA audio output initialized OKAY" << LL_ENDL;
-                                audio_ok = TRUE;
+                                audio_ok = true;
                         } else {
                                 LL_WARNS("AppInit") << "ALSA audio output FAILED to initialize: "
                                         << FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
@@ -224,7 +209,7 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
         if (!audio_ok)
         {
                 LL_WARNS("AppInit") << "Overall audio init failure." << LL_ENDL;
-                return FALSE;
+                return false;
         }
 
         // On Linux, FMOD causes a SIGPIPE for some netstream error
@@ -250,7 +235,7 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
         {
                 LL_WARNS("AppInit") << "Error initializing FMOD: "
                         << FMOD_ErrorString(FSOUND_GetError()) << LL_ENDL;
-                return FALSE;
+                return false;
         }
         
 #endif
@@ -259,17 +244,9 @@ BOOL LLAudioEngine_FMOD::init(const S32 num_channels, void* userdata)
 
         LL_DEBUGS("AppInit") << "LLAudioEngine_FMOD::init() FMOD initialized correctly" << LL_ENDL;
 
-        mInited = TRUE;
+        mInited = true;
 
-        return TRUE;
-}
-
-
-void LLAudioEngine_FMOD::idle(F32 max_decode_time)
-{
-        LLAudioEngine::idle(max_decode_time);
-
-        updateInternetStream();
+        return true;
 }
 
 
@@ -287,7 +264,7 @@ void LLAudioEngine_FMOD::shutdown()
 {
         if (gWindDSP)
         {
-                FSOUND_DSP_SetActive(gWindDSP,FALSE);
+                FSOUND_DSP_SetActive(gWindDSP,false);
                 FSOUND_DSP_Free(gWindDSP);
         }
 
@@ -318,13 +295,15 @@ LLAudioChannel *LLAudioEngine_FMOD::createChannel()
 
 void LLAudioEngine_FMOD::initWind()
 {
+        mWindGen = new LLWindGen<MIXBUFFERFORMAT>;
+
         if (!gWindDSP)
         {
-                gWindDSP = FSOUND_DSP_Create(&windCallback, FSOUND_DSP_DEFAULTPRIORITY_CLEARUNIT + 20, NULL);
+                gWindDSP = FSOUND_DSP_Create(&windCallback, FSOUND_DSP_DEFAULTPRIORITY_CLEARUNIT + 20, mWindGen);
         }
         if (gWindDSP)
         {
-                FSOUND_DSP_SetActive(gWindDSP, TRUE);
+                FSOUND_DSP_SetActive(gWindDSP, true);
         }
         mNextWindUpdate = 0.0;
 }
@@ -334,10 +313,13 @@ void LLAudioEngine_FMOD::cleanupWind()
 {
         if (gWindDSP)
         {
-                FSOUND_DSP_SetActive(gWindDSP, FALSE);
+                FSOUND_DSP_SetActive(gWindDSP, false);
                 FSOUND_DSP_Free(gWindDSP);
                 gWindDSP = NULL;
         }
+
+        delete mWindGen;
+        mWindGen = NULL;
 }
 
 
@@ -367,9 +349,9 @@ void LLAudioEngine_FMOD::updateWind(LLVector3 wind_vec, F32 camera_height_above_
                 pitch = 1.0 + mapWindVecToPitch(wind_vec);
                 center_freq = 80.0 * pow(pitch,2.5*(mapWindVecToGain(wind_vec)+1.0));
                 
-                gTargetFreq = (F32)center_freq;
-                gTargetGain = (F32)mapWindVecToGain(wind_vec) * mMaxWindGain;
-                gTargetPanGainR = (F32)mapWindVecToPan(wind_vec);
+                mWindGen->mTargetFreq = (F32)center_freq;
+                mWindGen->mTargetGain = (F32)mapWindVecToGain(wind_vec) * mMaxWindGain;
+                mWindGen->mTargetPanGainR = (F32)mapWindVecToPan(wind_vec);
         }
 }
 
@@ -457,11 +439,11 @@ LLAudioChannelFMOD::~LLAudioChannelFMOD()
 }
 
 
-BOOL LLAudioChannelFMOD::updateBuffer()
+bool LLAudioChannelFMOD::updateBuffer()
 {
         if (LLAudioChannel::updateBuffer())
         {
-                // Base class update returned TRUE, which means that we need to actually
+                // Base class update returned true, which means that we need to actually
                 // set up the channel for a different buffer.
 
                 LLAudioBufferFMOD *bufferp = (LLAudioBufferFMOD *)mCurrentSourcep->getCurrentBuffer();
@@ -473,13 +455,13 @@ BOOL LLAudioChannelFMOD::updateBuffer()
                         // This is bad, there should ALWAYS be a sample associated with a legit
                         // buffer.
                         llerrs << "No FMOD sample!" << llendl;
-                        return FALSE;
+                        return false;
                 }
 
 
                 // Actually play the sound.  Start it off paused so we can do all the necessary
                 // setup.
-                mChannelID = FSOUND_PlaySoundEx(FSOUND_FREE, samplep, FSOUND_DSP_GetSFXUnit(), TRUE);
+                mChannelID = FSOUND_PlaySoundEx(FSOUND_FREE, samplep, FSOUND_DSP_GetSFXUnit(), true);
 
                 //llinfos << "Setting up channel " << std::hex << mChannelID << std::dec << llendl;
         }
@@ -501,7 +483,7 @@ BOOL LLAudioChannelFMOD::updateBuffer()
                 }
         }
 
-        return TRUE;
+        return true;
 }
 
 
@@ -524,12 +506,12 @@ void LLAudioChannelFMOD::update3DPosition()
         if (mCurrentSourcep->isAmbient())
         {
                 // Ambient sound, don't need to do any positional updates.
-                bufferp->set3DMode(FALSE);
+                bufferp->set3DMode(false);
         }
         else
         {
                 // Localized sound.  Update the position and velocity of the sound.
-                bufferp->set3DMode(TRUE);
+                bufferp->set3DMode(true);
 
                 LLVector3 float_pos;
                 float_pos.setVec(mCurrentSourcep->getPositionGlobal());
@@ -556,7 +538,7 @@ void LLAudioChannelFMOD::updateLoop()
         U32 cur_pos = FSOUND_GetCurrentPosition(mChannelID);
         if (cur_pos < (U32)mLastSamplePos)
         {
-                mLoopedThisFrame = TRUE;
+                mLoopedThisFrame = true;
         }
         mLastSamplePos = cur_pos;
 }
@@ -589,11 +571,11 @@ void LLAudioChannelFMOD::play()
                 return;
         }
 
-        if (!FSOUND_SetPaused(mChannelID, FALSE))
+        if (!FSOUND_SetPaused(mChannelID, false))
         {
                 llwarns << "LLAudioChannelFMOD::play error: " << FMOD_ErrorString(FSOUND_GetError()) << llendl;
         }
-        getSource()->setPlayedOnce(TRUE);
+        getSource()->setPlayedOnce(true);
 }
 
 
@@ -618,11 +600,11 @@ void LLAudioChannelFMOD::playSynced(LLAudioChannel *channelp)
 }
 
 
-BOOL LLAudioChannelFMOD::isPlaying()
+bool LLAudioChannelFMOD::isPlaying()
 {
         if (!mChannelID)
         {
-                return FALSE;
+                return false;
         }
 
         return FSOUND_IsPlaying(mChannelID) && (!FSOUND_GetPaused(mChannelID));
@@ -652,14 +634,14 @@ LLAudioBufferFMOD::~LLAudioBufferFMOD()
 }
 
 
-BOOL LLAudioBufferFMOD::loadWAV(const std::string& filename)
+bool LLAudioBufferFMOD::loadWAV(const std::string& filename)
 {
         // Try to open a wav file from disk.  This will eventually go away, as we don't
         // really want to block doing this.
         if (filename.empty())
         {
                 // invalid filename, abort.
-                return FALSE;
+                return false;
         }
 
         S32 file_size = 0;
@@ -667,7 +649,7 @@ BOOL LLAudioBufferFMOD::loadWAV(const std::string& filename)
         if (!apr_file)
         {
                 // File not found, abort.
-                return FALSE;
+                return false;
         }
         apr_file_close(apr_file);
 
@@ -717,11 +699,11 @@ BOOL LLAudioBufferFMOD::loadWAV(const std::string& filename)
                 //
                 // file is probably corrupt - remove it.
                 LLFile::remove(filename);
-                return FALSE;
+                return false;
         }
 
-        // Everything went well, return TRUE
-        return TRUE;
+        // Everything went well, return true
+        return true;
 }
 
 
@@ -736,7 +718,7 @@ U32 LLAudioBufferFMOD::getLength()
 }
 
 
-void LLAudioBufferFMOD::set3DMode(BOOL use3d)
+void LLAudioBufferFMOD::set3DMode(bool use3d)
 {
         U16 current_mode = FSOUND_Sample_GetMode(mSamplep);
         
@@ -765,7 +747,7 @@ void LLAudioEngine_FMOD::initInternetStream()
 {
         // Number of milliseconds of audio to buffer for the audio card.
         // Must be larger than the usual Second Life frame stutter time.
-    FSOUND_Stream_SetBufferSize(200);
+        FSOUND_Stream_SetBufferSize(200);
 
         // Here's where we set the size of the network buffer and some buffering 
         // parameters.  In this case we want a network buffer of 16k, we want it 
@@ -810,19 +792,19 @@ signed char F_CALLBACKAPI LLAudioEngine_FMOD::callbackMetaData(char *name, char
     if (!strcmp("ARTIST", name))
     {
         strlcpy(self->mInternetStreamArtist, value, 256);
-        self->mInternetStreamNewMetaData = TRUE;
-        return TRUE;
+        self->mInternetStreamNewMetaData = true;
+        return true;
     }
 
     if (!strcmp("TITLE", name))
     {
         strlcpy(self->mInternetStreamTitle, value, 256);
-        self->mInternetStreamNewMetaData = TRUE;
-        return TRUE;
+        self->mInternetStreamNewMetaData = true;
+        return true;
     }
         */
 
-    return TRUE;
+    return true;
 }
 
 
@@ -867,7 +849,7 @@ void LLAudioEngine_FMOD::updateInternetStream()
                         {
                                 // Reset volume to previously set volume
                                 setInternetStreamGain(mInternetStreamGain);
-                                FSOUND_SetPaused(mInternetStreamChannel, FALSE);
+                                FSOUND_SetPaused(mInternetStreamChannel, false);
                                 //FSOUND_Stream_Net_SetMetadataCallback(mInternetStream, callbackMetaData, this);
                         }
                 }
@@ -909,7 +891,7 @@ void LLAudioEngine_FMOD::stopInternetStream()
 {
         if (mInternetStreamChannel != -1)
         {
-                FSOUND_SetPaused(mInternetStreamChannel, TRUE);
+                FSOUND_SetPaused(mInternetStreamChannel, true);
                 FSOUND_SetPriority(mInternetStreamChannel, 0);
                 mInternetStreamChannel = -1;
         }
@@ -971,16 +953,10 @@ int LLAudioEngine_FMOD::isInternetStreamPlaying()
 }
 
 
-void LLAudioEngine_FMOD::getInternetStreamInfo(char* artist_out, char* title_out)
-{
-        //strlcpy(artist_out, mInternetStreamArtist, 256);
-        //strlcpy(title_out, mInternetStreamTitle, 256);
-}
-
-
 void LLAudioEngine_FMOD::setInternetStreamGain(F32 vol)
 {
-        LLAudioEngine::setInternetStreamGain(vol);
+        mInternetStreamGain = vol;
+
         if (mInternetStreamChannel != -1)
         {
                 vol = llclamp(vol, 0.f, 1.f);
@@ -990,15 +966,9 @@ void LLAudioEngine_FMOD::setInternetStreamGain(F32 vol)
 }
 
 
-const std::string& LLAudioEngine_FMOD::getInternetStreamURL()
-{
-        return mInternetStreamURL;
-}
-
-
 LLAudioStreamFMOD::LLAudioStreamFMOD(const std::string& url) :
         mInternetStream(NULL),
-        mReady(FALSE)
+        mReady(false)
 {
         mInternetStreamURL = url;
         mInternetStream = FSOUND_Stream_Open(url.c_str(), FSOUND_NORMAL | FSOUND_NONBLOCKING, 0, 0);
@@ -1007,11 +977,11 @@ LLAudioStreamFMOD::LLAudioStreamFMOD(const std::string& url) :
                 llwarns << "Couldn't open fmod stream, error "
                         << FMOD_ErrorString(FSOUND_GetError())
                         << llendl;
-                mReady = FALSE;
+                mReady = false;
                 return;
         }
 
-        mReady = TRUE;
+        mReady = true;
 }
 
 int LLAudioStreamFMOD::startStream()
@@ -1026,10 +996,10 @@ int LLAudioStreamFMOD::startStream()
         // Make sure the stream is set to 2D mode.
         FSOUND_Stream_SetMode(mInternetStream, FSOUND_2D);
 
-        return FSOUND_Stream_PlayEx(FSOUND_FREE, mInternetStream, NULL, TRUE);
+        return FSOUND_Stream_PlayEx(FSOUND_FREE, mInternetStream, NULL, true);
 }
 
-BOOL LLAudioStreamFMOD::stopStream()
+bool LLAudioStreamFMOD::stopStream()
 {
         if (mInternetStream)
         {
@@ -1039,34 +1009,34 @@ BOOL LLAudioStreamFMOD::stopStream()
                 unsigned int flags = 0x0;
                 FSOUND_Stream_Net_GetStatus(mInternetStream, &status, &read_percent, &bitrate, &flags);
 
-                BOOL close = TRUE;
+                bool close = true;
                 switch (status)
                 {
                 case FSOUND_STREAM_NET_CONNECTING:
-                        close = FALSE;
+                        close = false;
                         break;
                 case FSOUND_STREAM_NET_NOTCONNECTED:
                 case FSOUND_STREAM_NET_BUFFERING:
                 case FSOUND_STREAM_NET_READY:
                 case FSOUND_STREAM_NET_ERROR:
                 default:
-                        close = TRUE;
+                        close = true;
                 }
 
                 if (close)
                 {
                         FSOUND_Stream_Close(mInternetStream);
                         mInternetStream = NULL;
-                        return TRUE;
+                        return true;
                 }
                 else
                 {
-                        return FALSE;
+                        return false;
                 }
         }
         else
         {
-                return TRUE;
+                return true;
         }
 }
 
@@ -1076,94 +1046,35 @@ int LLAudioStreamFMOD::getOpenState()
         return open_state;
 }
 
-/* This determines the format of the mixbuffer being passed in. change if you want to support int32 or float32 */
-#if LL_DARWIN
-        #define MIXBUFFERFORMAT S32
-#else
-        #define MIXBUFFERFORMAT S16
-#endif
-
-inline MIXBUFFERFORMAT clipSample(MIXBUFFERFORMAT sample, MIXBUFFERFORMAT min, MIXBUFFERFORMAT max)
-{
-        if (sample > max)
-                sample = max;
-        else if (sample < min)
-                sample = min;
-        
-        return sample;
-}
-
-void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int length, void*)
+void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int length, void* userdata)
 {
-// originalbuffer = fsounds original mixbuffer.
-// newbuffer = the buffer passed from the previous DSP unit.
-// length = length in samples at this mix time.
-// param = user parameter passed through in FSOUND_DSP_Create.
-//
-// modify the buffer in some fashion
+        // originalbuffer = fmod's original mixbuffer.
+        // newbuffer = the buffer passed from the previous DSP unit.
+        // length = length in samples at this mix time.
+        // param = user parameter passed through in FSOUND_DSP_Create.
+        //
+        // modify the buffer in some fashion
 
-        U8 *cursamplep = (U8*)newbuffer;
-        U8   wordsize = 2;
+        LLWindGen<LLAudioEngine_FMOD::MIXBUFFERFORMAT> *windgen =
+                (LLWindGen<LLAudioEngine_FMOD::MIXBUFFERFORMAT> *)userdata;
+        U8 stride;
 
 #if LL_DARWIN
-        wordsize = sizeof(MIXBUFFERFORMAT);
+        stride = sizeof(LLAudioEngine_FMOD::MIXBUFFERFORMAT);
 #else
-    int         mixertype = FSOUND_GetMixer();
-    if (mixertype == FSOUND_MIXER_BLENDMODE || mixertype == FSOUND_MIXER_QUALITY_FPU)
-    {
-                wordsize = 4;
-    }
-#endif
-
-    double bandwidth = 50;
-    double inputSamplingRate = 44100;
-        double a0,b1,b2;
- 
-        // calculate resonant filter coeffs
-    b2 = exp(-(F_TWO_PI) * (bandwidth / inputSamplingRate));
-
-        while (length--)
-        {
-                gCurrentFreq = (float)((0.999 * gCurrentFreq) + (0.001 * gTargetFreq));
-                gCurrentGain = (float)((0.999 * gCurrentGain) + (0.001 * gTargetGain));
-                gCurrentPanGainR = (float)((0.999 * gCurrentPanGainR) + (0.001 * gTargetPanGainR));
-                b1 = (-4.0 * b2) / (1.0 + b2) * cos(F_TWO_PI * (gCurrentFreq / inputSamplingRate));
-            a0 = (1.0 - b2) * sqrt(1.0 - (b1 * b1) / (4.0 * b2));
-                double nextSample;
-
-            // start with white noise
-                nextSample = ll_frand(2.0f) - 1.0f;
-                                                                         
-#if 1 // LLAE_WIND_PINK apply pinking filter
-                gbuf0 = 0.997f * gbuf0 + 0.0126502f * nextSample; 
-        gbuf1 = 0.985f * gbuf1 + 0.0139083f * nextSample;
-        gbuf2 = 0.950f * gbuf2 + 0.0205439f * nextSample;
-        gbuf3 = 0.850f * gbuf3 + 0.0387225f * nextSample;
-        gbuf4 = 0.620f * gbuf4 + 0.0465932f * nextSample;
-        gbuf5 = 0.250f * gbuf5 + 0.1093477f * nextSample;
-                          
-        nextSample = gbuf0 + gbuf1 + gbuf2 + gbuf3 + gbuf4 + gbuf5;
-#endif
-                        
-#if 1 //LLAE_WIND_RESONANT // do a resonant filter on the noise
-        nextSample = (double)( a0 * nextSample - b1 * gY0 - b2 * gY1 );
-
-        gY1 = gY0;
-        gY0 = nextSample;
+        int mixertype = FSOUND_GetMixer();
+        if (mixertype == FSOUND_MIXER_BLENDMODE ||
+            mixertype == FSOUND_MIXER_QUALITY_FPU)
+        {
+                stride = 4;
+        }
+        else
+        {
+                stride = 2;
+        }
 #endif
 
-            nextSample *= gCurrentGain;
-                
-                MIXBUFFERFORMAT sample;
-
-                sample = llfloor(((F32)nextSample*32768.f*(1.0f - gCurrentPanGainR))+0.5f);
-                *(MIXBUFFERFORMAT*)cursamplep = clipSample((*(MIXBUFFERFORMAT*)cursamplep) + sample, -32768, 32767);
-                cursamplep += wordsize;
-
-                sample = llfloor(((F32)nextSample*32768.f*gCurrentPanGainR)+0.5f);
-                *(MIXBUFFERFORMAT*)cursamplep = clipSample((*(MIXBUFFERFORMAT*)cursamplep) + sample, -32768, 32767);
-                cursamplep += wordsize;
-        }
+        newbuffer = windgen->windGenerate((LLAudioEngine_FMOD::MIXBUFFERFORMAT *)newbuffer, length, stride);
 
         return newbuffer;
 }