--- linden/doc/contributions.txt.orig 2008-03-06 17:02:22.000000000 +0000 +++ linden/doc/contributions.txt 2008-03-06 17:00:34.000000000 +0000 @@ -11,6 +11,7 @@ Aimee Trescothick VWR-3903 VWR-4803 + VWR-5308 Alejandro Rosenthal VWR-1184 Alissa Sabre @@ -126,6 +127,7 @@ VWR-1987 VWR-2065 VWR-2502 + VWR-5308 Ginko Bayliss VWR-4 Grazer Kline --- linden/indra/llmath/llmath.h.orig 2008-03-06 17:12:26.000000000 +0000 +++ linden/indra/llmath/llmath.h 2008-03-06 15:00:47.000000000 +0000 @@ -48,12 +48,12 @@ #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))) #endif +#ifndef fsqrtf +#define fsqrtf(x) sqrtf(x) +#endif #ifndef cosf #define cosf(x) ((F32)cos((F64)(x))) @@ -71,6 +71,9 @@ #ifndef powf #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; --- linden/indra/llaudio/audioengine_fmod.cpp.orig 2008-03-05 21:58:38.000000000 +0000 +++ linden/indra/llaudio/audioengine_fmod.cpp 2008-03-08 04:56:23.000000000 +0000 @@ -54,21 +54,9 @@ 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 @@ -348,8 +336,8 @@ void LLAudioEngine_FMOD::updateWind(LLVector3 wind_vec, F32 camera_height_above_water) { LLVector3 wind_pos; - F64 pitch; - F64 center_freq; + F32 pitch; + F32 center_freq; if (!mEnableWind) { @@ -367,10 +355,10 @@ // cerr << "Wind update" << endl; - pitch = 1.0 + mapWindVecToPitch(wind_vec); - center_freq = 80.0 * pow(pitch,2.5*(mapWindVecToGain(wind_vec)+1.0)); + pitch = 1.0f + mapWindVecToPitch(wind_vec); + center_freq = 80.0f * powf(pitch,2.5f*((F32)mapWindVecToGain(wind_vec)+1.0f)); - gTargetFreq = (F32)center_freq; + gTargetFreq = center_freq; gTargetGain = (F32)mapWindVecToGain(wind_vec) * mMaxWindGain; gTargetPanGainR = (F32)mapWindVecToPan(wind_vec); } @@ -1087,7 +1075,7 @@ #define MIXBUFFERFORMAT S16 #endif -inline MIXBUFFERFORMAT clipSample(MIXBUFFERFORMAT sample, MIXBUFFERFORMAT min, MIXBUFFERFORMAT max) +inline MIXBUFFERFORMAT clipSample(F32 sample, MIXBUFFERFORMAT min, MIXBUFFERFORMAT max) { if (sample > max) sample = max; @@ -1097,6 +1085,10 @@ return sample; } + +//NOTE: This function gets called a *lot*. +//The inner loop runs 44,100 times per second. +//Keep performance in mind if you mess with this. void * F_CALLBACKAPI windCallback(void *originalbuffer, void *newbuffer, int length, void*) { // originalbuffer = fsounds original mixbuffer. @@ -1106,67 +1098,121 @@ // // modify the buffer in some fashion + //this turns off wind synth if it is muted or very very low volume + if (gTargetGain < 0.0005f) + return newbuffer; + + static F32 gbuf0 = 0.0f; + static F32 gbuf1 = 0.0f; + static F32 gbuf2 = 0.0f; + static F32 gY0 = 0.0f; + static F32 gY1 = 0.0f; + static F32 last_sample = 0.0f; + static F32 current_freq; + static F32 current_gain; + static F32 current_pan_gain_r; + U8 *cursamplep = (U8*)newbuffer; - U8 wordsize = 2; + U8 wordsize = 2; + bool clip = TRUE; #if LL_DARWIN wordsize = sizeof(MIXBUFFERFORMAT); + clip = (wordsize == 2); + #else - int mixertype = FSOUND_GetMixer(); - if (mixertype == FSOUND_MIXER_BLENDMODE || mixertype == FSOUND_MIXER_QUALITY_FPU) - { + int mixertype = FSOUND_GetMixer(); + if (mixertype == FSOUND_MIXER_BLENDMODE || mixertype == FSOUND_MIXER_QUALITY_FPU) + { wordsize = 4; + clip = FALSE; } #endif - double bandwidth = 50; - double inputSamplingRate = 44100; - double a0,b1,b2; - + U8 subsamples = 2; + F32 filter_sample_period = (F32)subsamples / 44100.0f; + F32 bandwidth = 50.0f; + F32 a0 = 0.0f, b1 = 0.0f, b2 = 0.0f; + + bool interp_freq = false; + // calculate resonant filter coeffs - b2 = exp(-(F_TWO_PI) * (bandwidth / inputSamplingRate)); + b2 = expf(-(F_TWO_PI) * (bandwidth * filter_sample_period)); - while (length--) + //if the frequency isn't changing much, we don't need to interpolate in the inner loop + if (llabs(gTargetFreq - current_freq) > 200.0f) { - 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; + interp_freq = true; + } + else + { + current_freq = gTargetFreq; + b1 = (-4.0f * b2) / (1.0f + b2) * cosf(F_TWO_PI * (current_freq * filter_sample_period)); + a0 = (1.0f - b2) * sqrtf(1.0f - (b1 * b1) / (4.0f * b2)); + } + + + while (length) + { + current_gain = (0.999f * current_gain) + (0.001f * gTargetGain); + current_pan_gain_r = (0.999f * current_pan_gain_r) + (0.001f * gTargetPanGainR); + + if (interp_freq) + { + current_freq = (0.999f * current_freq) + (0.001f * gTargetFreq); + b1 = (-4.0f * b2) / (1.0f + b2) * cosf(F_TWO_PI * (current_freq * filter_sample_period)); + a0 = (1.0f - b2) * sqrtf(1.0f - (b1 * b1) / (4.0f * b2)); + } - // start with white noise - nextSample = ll_frand(2.0f) - 1.0f; + F32 next_sample; + + // start with white noise [-0.125, 0.125] + // next_sample = ll_frand(0.25f) - 0.125f; + next_sample = (F32)rand() * 0.25f / ((F32)RAND_MAX + 1.0f) - 0.125f; -#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 + // Pinking filter + gbuf0 = 0.99765f * gbuf0 + next_sample * 0.0990460f; + gbuf1 = 0.96300f * gbuf1 + next_sample * 0.2965164f; + gbuf2 = 0.57000f * gbuf2 + next_sample * 1.0526913f; + + next_sample = gbuf0 + gbuf1 + gbuf2 + next_sample * 0.1848f; -#if 1 //LLAE_WIND_RESONANT // do a resonant filter on the noise - nextSample = (double)( a0 * nextSample - b1 * gY0 - b2 * gY1 ); + // do a resonant filter on the noise + next_sample = ( a0 * next_sample - b1 * gY0 - b2 * gY1 ); gY1 = gY0; - gY0 = nextSample; -#endif + gY0 = next_sample; - nextSample *= gCurrentGain; + next_sample *= current_gain * 32768.f; + F32 delta = (next_sample - last_sample) / (F32)subsamples; - MIXBUFFERFORMAT sample; + F32 sample_left; + F32 sample_right; + + for (U8 i=0; i < subsamples && length; i++, length--) + { + last_sample = last_sample + delta; + + sample_right = llfloor(last_sample * current_pan_gain_r + 0.5f); + sample_left = last_sample - sample_right; + + if (clip) + { + *(MIXBUFFERFORMAT*)cursamplep = clipSample((*(MIXBUFFERFORMAT*)cursamplep) + sample_left, -32768, 32767); + cursamplep += wordsize; + + *(MIXBUFFERFORMAT*)cursamplep = clipSample((*(MIXBUFFERFORMAT*)cursamplep) + sample_right, -32768, 32767); + cursamplep += wordsize; + } + else + { + *(MIXBUFFERFORMAT*)cursamplep = (*(MIXBUFFERFORMAT*)cursamplep) + sample_left; + cursamplep += wordsize; - 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; + *(MIXBUFFERFORMAT*)cursamplep = (*(MIXBUFFERFORMAT*)cursamplep) + sample_right; + cursamplep += wordsize; + } + } } return newbuffer;