diff options
Diffstat (limited to 'indra/llaudio')
-rw-r--r-- | indra/llaudio/CMakeLists.txt | 12 | ||||
-rw-r--r-- | indra/llaudio/llaudiodecodemgr.cpp | 24 | ||||
-rw-r--r-- | indra/llaudio/llaudioengine.cpp | 5 | ||||
-rw-r--r-- | indra/llaudio/llaudioengine.h | 2 | ||||
-rw-r--r-- | indra/llaudio/llaudioengine_fmod.cpp | 92 | ||||
-rw-r--r-- | indra/llaudio/llaudioengine_fmod.h | 7 | ||||
-rw-r--r-- | indra/llaudio/llaudioengine_openal.cpp | 10 | ||||
-rw-r--r-- | indra/llaudio/llaudioengine_openal.h | 24 | ||||
-rw-r--r-- | indra/llaudio/llstreamingaudio_fmod.cpp | 2 | ||||
-rw-r--r-- | indra/llaudio/llwindgen.h | 166 |
10 files changed, 211 insertions, 133 deletions
diff --git a/indra/llaudio/CMakeLists.txt b/indra/llaudio/CMakeLists.txt index bfa2c34c12..e869b9717c 100644 --- a/indra/llaudio/CMakeLists.txt +++ b/indra/llaudio/CMakeLists.txt @@ -57,13 +57,11 @@ if (FMOD) llstreamingaudio_fmod.h ) - if (LINUX) - if (${CXX_VERSION_NUMBER} GREATER 419) - set_source_files_properties(llaudioengine_fmod.cpp - llstreamingaudio_fmod.cpp - COMPILE_FLAGS -Wno-write-strings) - endif (${CXX_VERSION_NUMBER} GREATER 419) - endif (LINUX) + if (LINUX OR DARWIN) + set_source_files_properties(llaudioengine_fmod.cpp + llstreamingaudio_fmod.cpp + COMPILE_FLAGS -Wno-write-strings) + endif (LINUX OR DARWIN) endif (FMOD) if (OPENAL) diff --git a/indra/llaudio/llaudiodecodemgr.cpp b/indra/llaudio/llaudiodecodemgr.cpp index 290206ee22..fc2190707a 100644 --- a/indra/llaudio/llaudiodecodemgr.cpp +++ b/indra/llaudio/llaudiodecodemgr.cpp @@ -230,19 +230,29 @@ BOOL LLVorbisDecodeState::initDecode() bool abort_decode = false; - if( vi->channels < 1 || vi->channels > LLVORBIS_CLIP_MAX_CHANNELS ) + if (vi) + { + if( vi->channels < 1 || vi->channels > LLVORBIS_CLIP_MAX_CHANNELS ) + { + abort_decode = true; + llwarns << "Bad channel count: " << vi->channels << llendl; + } + } + else // !vi { abort_decode = true; - llwarns << "Bad channel count: " << vi->channels << llendl; + llwarns << "No default bitstream found" << llendl; } - if( (size_t)sample_count > LLVORBIS_CLIP_REJECT_SAMPLES ) + if( (size_t)sample_count > LLVORBIS_CLIP_REJECT_SAMPLES || + (size_t)sample_count <= 0) { abort_decode = true; llwarns << "Illegal sample count: " << sample_count << llendl; } - if( size_guess > LLVORBIS_CLIP_REJECT_SIZE ) + if( size_guess > LLVORBIS_CLIP_REJECT_SIZE || + size_guess < 0) { abort_decode = true; llwarns << "Illegal sample size: " << size_guess << llendl; @@ -251,7 +261,11 @@ BOOL LLVorbisDecodeState::initDecode() if( abort_decode ) { llwarns << "Canceling initDecode. Bad asset: " << mUUID << llendl; - llwarns << "Bad asset encoded by: " << ov_comment(&mVF,-1)->vendor << llendl; + vorbis_comment* comment = ov_comment(&mVF,-1); + if (comment && comment->vendor) + { + llwarns << "Bad asset encoded by: " << comment->vendor << llendl; + } delete mInFilep; mInFilep = NULL; return FALSE; 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/llstreamingaudio_fmod.cpp b/indra/llaudio/llstreamingaudio_fmod.cpp index a4620fa13c..fe94688565 100644 --- a/indra/llaudio/llstreamingaudio_fmod.cpp +++ b/indra/llaudio/llstreamingaudio_fmod.cpp @@ -271,7 +271,7 @@ void LLStreamingAudio_FMOD::setGain(F32 vol) if (mFMODInternetStreamChannel != -1) { - vol = llclamp(vol, 0.f, 1.f); + vol = llclamp(vol * vol, 0.f, 1.f); int vol_int = llround(vol * 255.f); FSOUND_SetVolumeAbsolute(mFMODInternetStreamChannel, vol_int); } diff --git a/indra/llaudio/llwindgen.h b/indra/llaudio/llwindgen.h index 847bfa6e9d..0e6d0aa2ca 100644 --- a/indra/llaudio/llwindgen.h +++ b/indra/llaudio/llwindgen.h @@ -33,104 +33,150 @@ #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), + mLastSample(0.f) + { + 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 |