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-rw-r--r--doc/contributions.txt2
-rw-r--r--indra/llaudio/llaudioengine.cpp5
-rw-r--r--indra/llaudio/llaudioengine.h2
-rw-r--r--indra/llaudio/llaudioengine_fmod.cpp92
-rw-r--r--indra/llaudio/llaudioengine_fmod.h7
-rw-r--r--indra/llaudio/llaudioengine_openal.cpp10
-rw-r--r--indra/llaudio/llaudioengine_openal.h24
-rw-r--r--indra/llaudio/llwindgen.h165
-rw-r--r--indra/llmath/llmath.h15
9 files changed, 196 insertions, 126 deletions
diff --git a/doc/contributions.txt b/doc/contributions.txt
index 5667b69c28..a74cf32afa 100644
--- a/doc/contributions.txt
+++ b/doc/contributions.txt
@@ -25,6 +25,7 @@ Aimee Trescothick
VWR-3903
VWR-4083
VWR-4106
+ VWR-5308
VWR-6348
VWR-6358
VWR-6360
@@ -252,6 +253,7 @@ Gigs Taggart
VWR-2491
VWR-2502
VWR-2331
+ VWR-5308
VWR-8781
VWR-8783
Ginko Bayliss
diff --git a/indra/llaudio/llaudioengine.cpp b/indra/llaudio/llaudioengine.cpp
index b92ccd1d77..9f4c108dff 100644
--- a/indra/llaudio/llaudioengine.cpp
+++ b/indra/llaudio/llaudioengine.cpp
@@ -548,12 +548,11 @@ void LLAudioEngine::enableWind(bool enable)
{
if (enable && (!mEnableWind))
{
- initWind();
- mEnableWind = enable;
+ mEnableWind = initWind();
}
else if (mEnableWind && (!enable))
{
- mEnableWind = enable;
+ mEnableWind = false;
cleanupWind();
}
}
diff --git a/indra/llaudio/llaudioengine.h b/indra/llaudio/llaudioengine.h
index d287104204..5876cef4ea 100644
--- a/indra/llaudio/llaudioengine.h
+++ b/indra/llaudio/llaudioengine.h
@@ -195,7 +195,7 @@ protected:
virtual LLAudioBuffer *createBuffer() = 0;
virtual LLAudioChannel *createChannel() = 0;
- virtual void initWind() = 0;
+ virtual bool initWind() = 0;
virtual void cleanupWind() = 0;
virtual void setInternalGain(F32 gain) = 0;
diff --git a/indra/llaudio/llaudioengine_fmod.cpp b/indra/llaudio/llaudioengine_fmod.cpp
index d7f58defca..7a8a04afa1 100644
--- a/indra/llaudio/llaudioengine_fmod.cpp
+++ b/indra/llaudio/llaudioengine_fmod.cpp
@@ -54,13 +54,12 @@ extern "C" {
void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int length, void* userdata);
}
-FSOUND_DSPUNIT *gWindDSP = NULL;
-
LLAudioEngine_FMOD::LLAudioEngine_FMOD()
{
mInited = false;
mWindGen = NULL;
+ mWindDSP = NULL;
}
@@ -258,10 +257,10 @@ void LLAudioEngine_FMOD::allocateListener(void)
void LLAudioEngine_FMOD::shutdown()
{
- if (gWindDSP)
+ if (mWindDSP)
{
- FSOUND_DSP_SetActive(gWindDSP,false);
- FSOUND_DSP_Free(gWindDSP);
+ FSOUND_DSP_SetActive(mWindDSP,false);
+ FSOUND_DSP_Free(mWindDSP);
}
stopInternetStream();
@@ -289,29 +288,66 @@ LLAudioChannel * LLAudioEngine_FMOD::createChannel()
}
-void LLAudioEngine_FMOD::initWind()
+bool LLAudioEngine_FMOD::initWind()
{
- mWindGen = new LLWindGen<MIXBUFFERFORMAT>;
+ if (!mWindGen)
+ {
+ bool enable;
+
+ switch (FSOUND_GetMixer())
+ {
+ case FSOUND_MIXER_MMXP5:
+ case FSOUND_MIXER_MMXP6:
+ case FSOUND_MIXER_QUALITY_MMXP5:
+ case FSOUND_MIXER_QUALITY_MMXP6:
+ enable = (typeid(MIXBUFFERFORMAT) == typeid(S16));
+ break;
+ case FSOUND_MIXER_BLENDMODE:
+ enable = (typeid(MIXBUFFERFORMAT) == typeid(S32));
+ break;
+ case FSOUND_MIXER_QUALITY_FPU:
+ enable = (typeid(MIXBUFFERFORMAT) == typeid(F32));
+ break;
+ default:
+ // FSOUND_GetMixer() does not return a valid mixer type on Darwin
+ LL_INFOS("AppInit") << "Unknown FMOD mixer type, assuming default" << LL_ENDL;
+ enable = true;
+ break;
+ }
+
+ if (enable)
+ {
+ mWindGen = new LLWindGen<MIXBUFFERFORMAT>(FSOUND_GetOutputRate());
+ }
+ else
+ {
+ LL_WARNS("AppInit") << "Incompatible FMOD mixer type, wind noise disabled" << LL_ENDL;
+ }
+ }
+
+ mNextWindUpdate = 0.0;
- if (!gWindDSP)
+ if (mWindGen && !mWindDSP)
{
- gWindDSP = FSOUND_DSP_Create(&windCallback, FSOUND_DSP_DEFAULTPRIORITY_CLEARUNIT + 20, mWindGen);
+ mWindDSP = FSOUND_DSP_Create(&windCallback, FSOUND_DSP_DEFAULTPRIORITY_CLEARUNIT + 20, mWindGen);
}
- if (gWindDSP)
+ if (mWindDSP)
{
- FSOUND_DSP_SetActive(gWindDSP, true);
+ FSOUND_DSP_SetActive(mWindDSP, true);
+ return true;
}
- mNextWindUpdate = 0.0;
+
+ return false;
}
void LLAudioEngine_FMOD::cleanupWind()
{
- if (gWindDSP)
+ if (mWindDSP)
{
- FSOUND_DSP_SetActive(gWindDSP, false);
- FSOUND_DSP_Free(gWindDSP);
- gWindDSP = NULL;
+ FSOUND_DSP_SetActive(mWindDSP, false);
+ FSOUND_DSP_Free(mWindDSP);
+ mWindDSP = NULL;
}
delete mWindGen;
@@ -740,30 +776,12 @@ void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int len
// 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
+ // userdata = user parameter passed through in FSOUND_DSP_Create.
LLWindGen<LLAudioEngine_FMOD::MIXBUFFERFORMAT> *windgen =
(LLWindGen<LLAudioEngine_FMOD::MIXBUFFERFORMAT> *)userdata;
- U8 stride;
-
-#if LL_DARWIN
- stride = sizeof(LLAudioEngine_FMOD::MIXBUFFERFORMAT);
-#else
- int mixertype = FSOUND_GetMixer();
- if (mixertype == FSOUND_MIXER_BLENDMODE ||
- mixertype == FSOUND_MIXER_QUALITY_FPU)
- {
- stride = 4;
- }
- else
- {
- stride = 2;
- }
-#endif
-
- newbuffer = windgen->windGenerate((LLAudioEngine_FMOD::MIXBUFFERFORMAT *)newbuffer, length, stride);
+
+ newbuffer = windgen->windGenerate((LLAudioEngine_FMOD::MIXBUFFERFORMAT *)newbuffer, length);
return newbuffer;
}
diff --git a/indra/llaudio/llaudioengine_fmod.h b/indra/llaudio/llaudioengine_fmod.h
index 3968657cba..0e386a3884 100644
--- a/indra/llaudio/llaudioengine_fmod.h
+++ b/indra/llaudio/llaudioengine_fmod.h
@@ -55,15 +55,15 @@ public:
virtual void shutdown();
- /*virtual*/ void initWind();
+ /*virtual*/ bool initWind();
/*virtual*/ void cleanupWind();
/*virtual*/void updateWind(LLVector3 direction, F32 camera_height_above_water);
#if LL_DARWIN
- typedef S32 MIXBUFFERFORMAT;
+ typedef S32 MIXBUFFERFORMAT;
#else
- typedef S16 MIXBUFFERFORMAT;
+ typedef S16 MIXBUFFERFORMAT;
#endif
protected:
@@ -83,6 +83,7 @@ protected:
void* mUserData;
LLWindGen<MIXBUFFERFORMAT> *mWindGen;
+ FSOUND_DSPUNIT *mWindDSP;
};
diff --git a/indra/llaudio/llaudioengine_openal.cpp b/indra/llaudio/llaudioengine_openal.cpp
index a5982ccbd6..887c791790 100644
--- a/indra/llaudio/llaudioengine_openal.cpp
+++ b/indra/llaudio/llaudioengine_openal.cpp
@@ -370,7 +370,7 @@ U32 LLAudioBufferOpenAL::getLength()
// ------------
-void LLAudioEngine_OpenAL::initWind()
+bool LLAudioEngine_OpenAL::initWind()
{
ALenum error;
llinfos << "LLAudioEngine_OpenAL::initWind() start" << llendl;
@@ -397,10 +397,12 @@ void LLAudioEngine_OpenAL::initWind()
if(mWindBuf==NULL)
{
llerrs << "LLAudioEngine_OpenAL::initWind() Error creating wind memory buffer" << llendl;
- mEnableWind=false;
+ return false;
}
llinfos << "LLAudioEngine_OpenAL::initWind() done" << llendl;
+
+ return true;
}
void LLAudioEngine_OpenAL::cleanupWind()
@@ -508,14 +510,14 @@ void LLAudioEngine_OpenAL::updateWind(LLVector3 wind_vec, F32 camera_altitude)
alGenBuffers(1,&buffer);
if((error=alGetError()) != AL_NO_ERROR)
{
- llwarns << "LLAudioEngine_OpenAL::initWind() Error creating wind buffer: " << error << llendl;
+ llwarns << "LLAudioEngine_OpenAL::updateWind() Error creating wind buffer: " << error << llendl;
break;
}
alBufferData(buffer,
AL_FORMAT_STEREO16,
mWindGen->windGenerate(mWindBuf,
- mWindBufSamples, 2),
+ mWindBufSamples),
mWindBufBytes,
mWindBufFreq);
error = alGetError();
diff --git a/indra/llaudio/llaudioengine_openal.h b/indra/llaudio/llaudioengine_openal.h
index 5aca03e195..16125b2476 100644
--- a/indra/llaudio/llaudioengine_openal.h
+++ b/indra/llaudio/llaudioengine_openal.h
@@ -57,23 +57,23 @@ class LLAudioEngine_OpenAL : public LLAudioEngine
LLAudioBuffer* createBuffer();
LLAudioChannel* createChannel();
- /*virtual*/ void initWind();
+ /*virtual*/ bool initWind();
/*virtual*/ void cleanupWind();
/*virtual*/ void updateWind(LLVector3 direction, F32 camera_altitude);
private:
void * windDSP(void *newbuffer, int length);
- typedef S16 WIND_SAMPLE_T;
- LLWindGen<WIND_SAMPLE_T> *mWindGen;
- S16 *mWindBuf;
- U32 mWindBufFreq;
- U32 mWindBufSamples;
- U32 mWindBufBytes;
- ALuint mWindSource;
- int mNumEmptyWindALBuffers;
-
- static const int MAX_NUM_WIND_BUFFERS = 80;
- static const float WIND_BUFFER_SIZE_SEC = 0.05f; // 1/20th sec
+ typedef S16 WIND_SAMPLE_T;
+ LLWindGen<WIND_SAMPLE_T> *mWindGen;
+ S16 *mWindBuf;
+ U32 mWindBufFreq;
+ U32 mWindBufSamples;
+ U32 mWindBufBytes;
+ ALuint mWindSource;
+ int mNumEmptyWindALBuffers;
+
+ static const int MAX_NUM_WIND_BUFFERS = 80;
+ static const float WIND_BUFFER_SIZE_SEC = 0.05f; // 1/20th sec
};
class LLAudioChannelOpenAL : public LLAudioChannel
diff --git a/indra/llaudio/llwindgen.h b/indra/llaudio/llwindgen.h
index 847bfa6e9d..1908b2545f 100644
--- a/indra/llaudio/llwindgen.h
+++ b/indra/llaudio/llwindgen.h
@@ -33,104 +33,149 @@
#define WINDGEN_H
#include "llcommon.h"
-#include "llrand.h"
template <class MIXBUFFERFORMAT_T>
class LLWindGen
{
public:
- LLWindGen() :
+ LLWindGen(const U32 sample_rate = 44100) :
mTargetGain(0.f),
mTargetFreq(100.f),
mTargetPanGainR(0.5f),
- mbuf0(0.0),
- mbuf1(0.0),
- mbuf2(0.0),
- mbuf3(0.0),
- mbuf4(0.0),
- mbuf5(0.0),
- mY0(0.0),
- mY1(0.0),
+ mInputSamplingRate(sample_rate),
+ mSubSamples(2),
+ mFilterBandWidth(50.f),
+ mBuf0(0.0f),
+ mBuf1(0.0f),
+ mBuf2(0.0f),
+ mY0(0.0f),
+ mY1(0.0f),
mCurrentGain(0.f),
mCurrentFreq(100.f),
- mCurrentPanGainR(0.5f) {};
-
- static const U32 getInputSamplingRate() {return mInputSamplingRate;}
+ mCurrentPanGainR(0.5f)
+ {
+ mSamplePeriod = (F32)mSubSamples / (F32)mInputSamplingRate;
+ mB2 = expf(-F_TWO_PI * mFilterBandWidth * mSamplePeriod);
+ }
+ const U32 getInputSamplingRate() { return mInputSamplingRate; }
+
// newbuffer = the buffer passed from the previous DSP unit.
// numsamples = length in samples-per-channel at this mix time.
- // stride = number of bytes between start of each sample.
// NOTE: generates L/R interleaved stereo
- MIXBUFFERFORMAT_T* windGenerate(MIXBUFFERFORMAT_T *newbuffer, int numsamples, int stride)
+ MIXBUFFERFORMAT_T* windGenerate(MIXBUFFERFORMAT_T *newbuffer, int numsamples)
{
- U8 *cursamplep = (U8*)newbuffer;
+ MIXBUFFERFORMAT_T *cursamplep = newbuffer;
+
+ // Filter coefficients
+ F32 a0 = 0.0f, b1 = 0.0f;
- double bandwidth = 50.0F;
- double a0,b1,b2;
+ // No need to clip at normal volumes
+ bool clip = mCurrentGain > 2.0f;
- // calculate resonant filter coeffs
- b2 = exp(-(F_TWO_PI) * (bandwidth / mInputSamplingRate));
+ bool interp_freq = false;
- while (numsamples--)
+ //if the frequency isn't changing much, we don't need to interpolate in the inner loop
+ if (llabs(mTargetFreq - mCurrentFreq) < (mCurrentFreq * 0.112))
{
- mCurrentFreq = (float)((0.999 * mCurrentFreq) + (0.001 * mTargetFreq));
- mCurrentGain = (float)((0.999 * mCurrentGain) + (0.001 * mTargetGain));
- mCurrentPanGainR = (float)((0.999 * mCurrentPanGainR) + (0.001 * mTargetPanGainR));
- b1 = (-4.0 * b2) / (1.0 + b2) * cos(F_TWO_PI * (mCurrentFreq / mInputSamplingRate));
- a0 = (1.0 - b2) * sqrt(1.0 - (b1 * b1) / (4.0 * b2));
- double nextSample;
+ // calculate resonant filter coefficients
+ mCurrentFreq = mTargetFreq;
+ b1 = (-4.0f * mB2) / (1.0f + mB2) * cosf(F_TWO_PI * (mCurrentFreq * mSamplePeriod));
+ a0 = (1.0f - mB2) * sqrtf(1.0f - (b1 * b1) / (4.0f * mB2));
+ }
+ else
+ {
+ interp_freq = true;
+ }
+
+ while (numsamples)
+ {
+ F32 next_sample;
+
+ // Start with white noise
+ // This expression is fragile, rearrange it and it will break!
+ next_sample = (F32)rand() * (1.0f / (F32)(RAND_MAX / (U16_MAX / 8))) + (F32)(S16_MIN / 8);
- // start with white noise
- nextSample = ll_frand(2.0f) - 1.0f;
+ // Apply a pinking filter
+ // Magic numbers taken from PKE method at http://www.firstpr.com.au/dsp/pink-noise/
+ mBuf0 = mBuf0 * 0.99765f + next_sample * 0.0990460f;
+ mBuf1 = mBuf1 * 0.96300f + next_sample * 0.2965164f;
+ mBuf2 = mBuf2 * 0.57000f + next_sample * 1.0526913f;
- // apply pinking filter
- mbuf0 = 0.997f * mbuf0 + 0.0126502f * nextSample;
- mbuf1 = 0.985f * mbuf1 + 0.0139083f * nextSample;
- mbuf2 = 0.950f * mbuf2 + 0.0205439f * nextSample;
- mbuf3 = 0.850f * mbuf3 + 0.0387225f * nextSample;
- mbuf4 = 0.620f * mbuf4 + 0.0465932f * nextSample;
- mbuf5 = 0.250f * mbuf5 + 0.1093477f * nextSample;
+ next_sample = mBuf0 + mBuf1 + mBuf2 + next_sample * 0.1848f;
- nextSample = mbuf0 + mbuf1 + mbuf2 + mbuf3 + mbuf4 + mbuf5;
+ if (interp_freq)
+ {
+ // calculate and interpolate resonant filter coefficients
+ mCurrentFreq = (0.999f * mCurrentFreq) + (0.001f * mTargetFreq);
+ b1 = (-4.0f * mB2) / (1.0f + mB2) * cosf(F_TWO_PI * (mCurrentFreq * mSamplePeriod));
+ a0 = (1.0f - mB2) * sqrtf(1.0f - (b1 * b1) / (4.0f * mB2));
+ }
- // do a resonant filter on the noise
- nextSample = (double)( a0 * nextSample - b1 * mY0 - b2 * mY1 );
+ // Apply a resonant low-pass filter on the pink noise
+ next_sample = a0 * next_sample - b1 * mY0 - mB2 * mY1;
mY1 = mY0;
- mY0 = nextSample;
+ mY0 = next_sample;
- nextSample *= mCurrentGain;
+ mCurrentGain = (0.999f * mCurrentGain) + (0.001f * mTargetGain);
+ mCurrentPanGainR = (0.999f * mCurrentPanGainR) + (0.001f * mTargetPanGainR);
- MIXBUFFERFORMAT_T sample;
+ // For a 3dB pan law use:
+ // next_sample *= mCurrentGain * ((mCurrentPanGainR*(mCurrentPanGainR-1)*1.652+1.413);
+ next_sample *= mCurrentGain;
- sample = llfloor(((F32)nextSample*32768.f*(1.0f - mCurrentPanGainR))+0.5f);
- *(MIXBUFFERFORMAT_T*)cursamplep = llclamp(sample, (MIXBUFFERFORMAT_T)-32768, (MIXBUFFERFORMAT_T)32767);
- cursamplep += stride;
-
- sample = llfloor(((F32)nextSample*32768.f*mCurrentPanGainR)+0.5f);
- *(MIXBUFFERFORMAT_T*)cursamplep = llclamp(sample, (MIXBUFFERFORMAT_T)-32768, (MIXBUFFERFORMAT_T)32767);
- cursamplep += stride;
+ // delta is used to interpolate between synthesized samples
+ F32 delta = (next_sample - mLastSample) / (F32)mSubSamples;
+
+ // Fill the audio buffer, clipping if necessary
+ for (U8 i=mSubSamples; i && numsamples; --i, --numsamples)
+ {
+ mLastSample = mLastSample + delta;
+ S32 sample_right = (S32)(mLastSample * mCurrentPanGainR);
+ S32 sample_left = (S32)mLastSample - sample_right;
+
+ if (!clip)
+ {
+ *cursamplep = (MIXBUFFERFORMAT_T)sample_left;
+ ++cursamplep;
+ *cursamplep = (MIXBUFFERFORMAT_T)sample_right;
+ ++cursamplep;
+ }
+ else
+ {
+ *cursamplep = (MIXBUFFERFORMAT_T)llclamp(sample_left, (S32)S16_MIN, (S32)S16_MAX);
+ ++cursamplep;
+ *cursamplep = (MIXBUFFERFORMAT_T)llclamp(sample_right, (S32)S16_MIN, (S32)S16_MAX);
+ ++cursamplep;
+ }
+ }
}
return newbuffer;
}
-
+
+public:
F32 mTargetGain;
F32 mTargetFreq;
F32 mTargetPanGainR;
-
+
private:
- static const U32 mInputSamplingRate = 44100;
- F64 mbuf0;
- F64 mbuf1;
- F64 mbuf2;
- F64 mbuf3;
- F64 mbuf4;
- F64 mbuf5;
- F64 mY0;
- F64 mY1;
+ U32 mInputSamplingRate;
+ U8 mSubSamples;
+ F32 mSamplePeriod;
+ F32 mFilterBandWidth;
+ F32 mB2;
+
+ F32 mBuf0;
+ F32 mBuf1;
+ F32 mBuf2;
+ F32 mY0;
+ F32 mY1;
+
F32 mCurrentGain;
F32 mCurrentFreq;
F32 mCurrentPanGainR;
+ F32 mLastSample;
};
#endif
diff --git a/indra/llmath/llmath.h b/indra/llmath/llmath.h
index 209b506c30..c3c15e1374 100644
--- a/indra/llmath/llmath.h
+++ b/indra/llmath/llmath.h
@@ -61,11 +61,11 @@
#endif
// Single Precision Floating Point Routines
-#ifndef fsqrtf
-#define fsqrtf(x) ((F32)sqrt((F64)(x)))
-#endif
#ifndef sqrtf
-#define sqrtf(x) ((F32)sqrt((F64)(x)))
+#define sqrtf(x) ((F32)sqrt((F64)(x)))
+#endif
+#ifndef fsqrtf
+#define fsqrtf(x) sqrtf(x)
#endif
#ifndef cosf
@@ -78,11 +78,14 @@
#define tanf(x) ((F32)tan((F64)(x)))
#endif
#ifndef acosf
-#define acosf(x) ((F32)acos((F64)(x)))
+#define acosf(x) ((F32)acos((F64)(x)))
#endif
#ifndef powf
-#define powf(x,y) ((F32)pow((F64)(x),(F64)(y)))
+#define powf(x,y) ((F32)pow((F64)(x),(F64)(y)))
+#endif
+#ifndef expf
+#define expf(x) ((F32)exp((F64)(x)))
#endif
const F32 GRAVITY = -9.8f;
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-25 15:41:04 +0000