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Diffstat (limited to 'Alc/mixer.c')
| -rw-r--r-- | Alc/mixer.c | 638 |
1 files changed, 0 insertions, 638 deletions
diff --git a/Alc/mixer.c b/Alc/mixer.c deleted file mode 100644 index 712075f1..00000000 --- a/Alc/mixer.c +++ /dev/null @@ -1,638 +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., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - * Or go to http://www.gnu.org/copyleft/lgpl.html - */ - -#include "config.h" - -#include <math.h> -#include <stdlib.h> -#include <string.h> -#include <ctype.h> -#include <assert.h> - -#include "alMain.h" -#include "AL/al.h" -#include "AL/alc.h" -#include "alSource.h" -#include "alBuffer.h" -#include "alListener.h" -#include "alAuxEffectSlot.h" -#include "alu.h" - -#include "mixer_defs.h" - - -static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE, - "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!"); - -extern inline void InitiatePositionArrays(ALuint frac, ALuint increment, ALuint *frac_arr, ALuint *pos_arr, ALuint size); - -alignas(16) union ResamplerCoeffs ResampleCoeffs; - - -enum Resampler { - PointResampler, - LinearResampler, - FIR4Resampler, - FIR8Resampler, - BSincResampler, - - ResamplerDefault = LinearResampler -}; - -/* FIR8 requires 3 extra samples before the current position, and 4 after. */ -static_assert(MAX_PRE_SAMPLES >= 3, "MAX_PRE_SAMPLES must be at least 3!"); -static_assert(MAX_POST_SAMPLES >= 4, "MAX_POST_SAMPLES must be at least 4!"); - - -static HrtfMixerFunc MixHrtfSamples = MixHrtf_C; -static MixerFunc MixSamples = Mix_C; -static ResamplerFunc ResampleSamples = Resample_point32_C; - -static inline HrtfMixerFunc SelectHrtfMixer(void) -{ -#ifdef HAVE_SSE - if((CPUCapFlags&CPU_CAP_SSE)) - return MixHrtf_SSE; -#endif -#ifdef HAVE_NEON - if((CPUCapFlags&CPU_CAP_NEON)) - return MixHrtf_Neon; -#endif - - return MixHrtf_C; -} - -static inline MixerFunc SelectMixer(void) -{ -#ifdef HAVE_SSE - if((CPUCapFlags&CPU_CAP_SSE)) - return Mix_SSE; -#endif -#ifdef HAVE_NEON - if((CPUCapFlags&CPU_CAP_NEON)) - return Mix_Neon; -#endif - - return Mix_C; -} - -static inline ResamplerFunc SelectResampler(enum Resampler resampler) -{ - switch(resampler) - { - case PointResampler: - return Resample_point32_C; - case LinearResampler: -#ifdef HAVE_SSE4_1 - if((CPUCapFlags&CPU_CAP_SSE4_1)) - return Resample_lerp32_SSE41; -#endif -#ifdef HAVE_SSE2 - if((CPUCapFlags&CPU_CAP_SSE2)) - return Resample_lerp32_SSE2; -#endif - return Resample_lerp32_C; - case FIR4Resampler: -#ifdef HAVE_SSE4_1 - if((CPUCapFlags&CPU_CAP_SSE4_1)) - return Resample_fir4_32_SSE41; -#endif -#ifdef HAVE_SSE3 - if((CPUCapFlags&CPU_CAP_SSE3)) - return Resample_fir4_32_SSE3; -#endif - return Resample_fir4_32_C; - case FIR8Resampler: -#ifdef HAVE_SSE4_1 - if((CPUCapFlags&CPU_CAP_SSE4_1)) - return Resample_fir8_32_SSE41; -#endif -#ifdef HAVE_SSE3 - if((CPUCapFlags&CPU_CAP_SSE3)) - return Resample_fir8_32_SSE3; -#endif - return Resample_fir8_32_C; - case BSincResampler: -#ifdef HAVE_SSE - if((CPUCapFlags&CPU_CAP_SSE)) - return Resample_bsinc32_SSE; -#endif - return Resample_bsinc32_C; - } - - return Resample_point32_C; -} - - -/* The sinc resampler makes use of a Kaiser window to limit the needed sample - * points to 4 and 8, respectively. - */ - -#ifndef M_PI -#define M_PI (3.14159265358979323846) -#endif -static inline double Sinc(double x) -{ - if(x == 0.0) return 1.0; - return sin(x*M_PI) / (x*M_PI); -} - -/* The zero-order modified Bessel function of the first kind, used for the - * Kaiser window. - * - * I_0(x) = sum_{k=0}^inf (1 / k!)^2 (x / 2)^(2 k) - * = sum_{k=0}^inf ((x / 2)^k / k!)^2 - */ -static double BesselI_0(double x) -{ - double term, sum, x2, y, last_sum; - int k; - - /* Start at k=1 since k=0 is trivial. */ - term = 1.0; - sum = 1.0; - x2 = x / 2.0; - k = 1; - - /* Let the integration converge until the term of the sum is no longer - * significant. - */ - do { - y = x2 / k; - k ++; - last_sum = sum; - term *= y * y; - sum += term; - } while(sum != last_sum); - return sum; -} - -/* Calculate a Kaiser window from the given beta value and a normalized k - * [-1, 1]. - * - * w(k) = { I_0(B sqrt(1 - k^2)) / I_0(B), -1 <= k <= 1 - * { 0, elsewhere. - * - * Where k can be calculated as: - * - * k = i / l, where -l <= i <= l. - * - * or: - * - * k = 2 i / M - 1, where 0 <= i <= M. - */ -static inline double Kaiser(double b, double k) -{ - if(k <= -1.0 || k >= 1.0) return 0.0; - return BesselI_0(b * sqrt(1.0 - (k*k))) / BesselI_0(b); -} - -static inline double CalcKaiserBeta(double rejection) -{ - if(rejection > 50.0) - return 0.1102 * (rejection - 8.7); - if(rejection >= 21.0) - return (0.5842 * pow(rejection - 21.0, 0.4)) + - (0.07886 * (rejection - 21.0)); - return 0.0; -} - -static float SincKaiser(double r, double x) -{ - /* Limit rippling to -60dB. */ - return (float)(Kaiser(CalcKaiserBeta(60.0), x / r) * Sinc(x)); -} - - -void aluInitMixer(void) -{ - enum Resampler resampler = ResamplerDefault; - const char *str; - ALuint i; - - if(ConfigValueStr(NULL, NULL, "resampler", &str)) - { - if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0) - resampler = PointResampler; - else if(strcasecmp(str, "linear") == 0) - resampler = LinearResampler; - else if(strcasecmp(str, "sinc4") == 0) - resampler = FIR4Resampler; - else if(strcasecmp(str, "sinc8") == 0) - resampler = FIR8Resampler; - else if(strcasecmp(str, "bsinc") == 0) - resampler = BSincResampler; - else if(strcasecmp(str, "cubic") == 0) - { - WARN("Resampler option \"cubic\" is deprecated, using sinc4\n"); - resampler = FIR4Resampler; - } - else - { - char *end; - long n = strtol(str, &end, 0); - if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler)) - resampler = n; - else - WARN("Invalid resampler: %s\n", str); - } - } - - if(resampler == FIR8Resampler) - for(i = 0;i < FRACTIONONE;i++) - { - ALdouble mu = (ALdouble)i / FRACTIONONE; - ResampleCoeffs.FIR8[i][0] = SincKaiser(4.0, mu - -3.0); - ResampleCoeffs.FIR8[i][1] = SincKaiser(4.0, mu - -2.0); - ResampleCoeffs.FIR8[i][2] = SincKaiser(4.0, mu - -1.0); - ResampleCoeffs.FIR8[i][3] = SincKaiser(4.0, mu - 0.0); - ResampleCoeffs.FIR8[i][4] = SincKaiser(4.0, mu - 1.0); - ResampleCoeffs.FIR8[i][5] = SincKaiser(4.0, mu - 2.0); - ResampleCoeffs.FIR8[i][6] = SincKaiser(4.0, mu - 3.0); - ResampleCoeffs.FIR8[i][7] = SincKaiser(4.0, mu - 4.0); - } - else if(resampler == FIR4Resampler) - for(i = 0;i < FRACTIONONE;i++) - { - ALdouble mu = (ALdouble)i / FRACTIONONE; - ResampleCoeffs.FIR4[i][0] = SincKaiser(2.0, mu - -1.0); - ResampleCoeffs.FIR4[i][1] = SincKaiser(2.0, mu - 0.0); - ResampleCoeffs.FIR4[i][2] = SincKaiser(2.0, mu - 1.0); - ResampleCoeffs.FIR4[i][3] = SincKaiser(2.0, mu - 2.0); - } - - MixHrtfSamples = SelectHrtfMixer(); - MixSamples = SelectMixer(); - ResampleSamples = SelectResampler(resampler); -} - - -static inline ALfloat Sample_ALbyte(ALbyte val) -{ return val * (1.0f/127.0f); } - -static inline ALfloat Sample_ALshort(ALshort val) -{ return val * (1.0f/32767.0f); } - -static inline ALfloat Sample_ALfloat(ALfloat val) -{ return val; } - -#define DECL_TEMPLATE(T) \ -static inline void Load_##T(ALfloat *dst, const T *src, ALuint srcstep, ALuint samples)\ -{ \ - ALuint i; \ - for(i = 0;i < samples;i++) \ - dst[i] = Sample_##T(src[i*srcstep]); \ -} - -DECL_TEMPLATE(ALbyte) -DECL_TEMPLATE(ALshort) -DECL_TEMPLATE(ALfloat) - -#undef DECL_TEMPLATE - -static void LoadSamples(ALfloat *dst, const ALvoid *src, ALuint srcstep, enum FmtType srctype, ALuint samples) -{ - switch(srctype) - { - case FmtByte: - Load_ALbyte(dst, src, srcstep, samples); - break; - case FmtShort: - Load_ALshort(dst, src, srcstep, samples); - break; - case FmtFloat: - Load_ALfloat(dst, src, srcstep, samples); - break; - } -} - -static inline void SilenceSamples(ALfloat *dst, ALuint samples) -{ - ALuint i; - for(i = 0;i < samples;i++) - dst[i] = 0.0f; -} - - -static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter, - ALfloat *restrict dst, const ALfloat *restrict src, - ALuint numsamples, enum ActiveFilters type) -{ - ALuint i; - switch(type) - { - case AF_None: - ALfilterState_processPassthru(lpfilter, src, numsamples); - ALfilterState_processPassthru(hpfilter, src, numsamples); - break; - - case AF_LowPass: - ALfilterState_process(lpfilter, dst, src, numsamples); - ALfilterState_processPassthru(hpfilter, dst, numsamples); - return dst; - case AF_HighPass: - ALfilterState_processPassthru(lpfilter, src, numsamples); - ALfilterState_process(hpfilter, dst, src, numsamples); - return dst; - - case AF_BandPass: - for(i = 0;i < numsamples;) - { - ALfloat temp[256]; - ALuint todo = minu(256, numsamples-i); - - ALfilterState_process(lpfilter, temp, src+i, todo); - ALfilterState_process(hpfilter, dst+i, temp, todo); - i += todo; - } - return dst; - } - return src; -} - - -ALvoid MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALuint SamplesToDo) -{ - ResamplerFunc Resample; - ALbufferlistitem *BufferListItem; - ALuint DataPosInt, DataPosFrac; - ALboolean Looping; - ALuint increment; - ALenum State; - ALuint OutPos; - ALuint NumChannels; - ALuint SampleSize; - ALint64 DataSize64; - ALuint IrSize; - ALuint chan, j; - - /* Get source info */ - State = Source->state; - BufferListItem = ATOMIC_LOAD(&Source->current_buffer); - DataPosInt = Source->position; - DataPosFrac = Source->position_fraction; - Looping = Source->Looping; - NumChannels = Source->NumChannels; - SampleSize = Source->SampleSize; - increment = voice->Step; - - IrSize = (Device->Hrtf ? GetHrtfIrSize(Device->Hrtf) : 0); - - Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ? - Resample_copy32_C : ResampleSamples); - - OutPos = 0; - do { - ALuint SrcBufferSize, DstBufferSize; - - /* Figure out how many buffer samples will be needed */ - DataSize64 = SamplesToDo-OutPos; - DataSize64 *= increment; - DataSize64 += DataPosFrac+FRACTIONMASK; - DataSize64 >>= FRACTIONBITS; - DataSize64 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES; - - SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE); - - /* Figure out how many samples we can actually mix from this. */ - DataSize64 = SrcBufferSize; - DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES; - DataSize64 <<= FRACTIONBITS; - DataSize64 -= DataPosFrac; - - DstBufferSize = (ALuint)((DataSize64+(increment-1)) / increment); - DstBufferSize = minu(DstBufferSize, (SamplesToDo-OutPos)); - - /* Some mixers like having a multiple of 4, so try to give that unless - * this is the last update. */ - if(OutPos+DstBufferSize < SamplesToDo) - DstBufferSize &= ~3; - - for(chan = 0;chan < NumChannels;chan++) - { - const ALfloat *ResampledData; - ALfloat *SrcData = Device->SourceData; - ALuint SrcDataSize; - - /* Load the previous samples into the source data first. */ - memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat)); - SrcDataSize = MAX_PRE_SAMPLES; - - if(Source->SourceType == AL_STATIC) - { - const ALbuffer *ALBuffer = BufferListItem->buffer; - const ALubyte *Data = ALBuffer->data; - ALuint DataSize; - ALuint pos; - - /* Offset buffer data to current channel */ - Data += chan*SampleSize; - - /* If current pos is beyond the loop range, do not loop */ - if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd) - { - Looping = AL_FALSE; - - /* Load what's left to play from the source buffer, and - * clear the rest of the temp buffer */ - pos = DataPosInt; - DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos); - - LoadSamples(&SrcData[SrcDataSize], &Data[pos * NumChannels*SampleSize], - NumChannels, ALBuffer->FmtType, DataSize); - SrcDataSize += DataSize; - - SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize); - SrcDataSize += SrcBufferSize - SrcDataSize; - } - else - { - ALuint LoopStart = ALBuffer->LoopStart; - ALuint LoopEnd = ALBuffer->LoopEnd; - - /* Load what's left of this loop iteration, then load - * repeats of the loop section */ - pos = DataPosInt; - DataSize = LoopEnd - pos; - DataSize = minu(SrcBufferSize - SrcDataSize, DataSize); - - LoadSamples(&SrcData[SrcDataSize], &Data[pos * NumChannels*SampleSize], - NumChannels, ALBuffer->FmtType, DataSize); - SrcDataSize += DataSize; - - DataSize = LoopEnd-LoopStart; - while(SrcBufferSize > SrcDataSize) - { - DataSize = minu(SrcBufferSize - SrcDataSize, DataSize); - - LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*SampleSize], - NumChannels, ALBuffer->FmtType, DataSize); - SrcDataSize += DataSize; - } - } - } - else - { - /* Crawl the buffer queue to fill in the temp buffer */ - ALbufferlistitem *tmpiter = BufferListItem; - ALuint pos = DataPosInt; - - while(tmpiter && SrcBufferSize > SrcDataSize) - { - const ALbuffer *ALBuffer; - if((ALBuffer=tmpiter->buffer) != NULL) - { - const ALubyte *Data = ALBuffer->data; - ALuint DataSize = ALBuffer->SampleLen; - - /* Skip the data already played */ - if(DataSize <= pos) - pos -= DataSize; - else - { - Data += (pos*NumChannels + chan)*SampleSize; - DataSize -= pos; - pos -= pos; - - DataSize = minu(SrcBufferSize - SrcDataSize, DataSize); - LoadSamples(&SrcData[SrcDataSize], Data, NumChannels, - ALBuffer->FmtType, DataSize); - SrcDataSize += DataSize; - } - } - tmpiter = tmpiter->next; - if(!tmpiter && Looping) - tmpiter = ATOMIC_LOAD(&Source->queue); - else if(!tmpiter) - { - SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize); - SrcDataSize += SrcBufferSize - SrcDataSize; - } - } - } - - /* Store the last source samples used for next time. */ - memcpy(voice->PrevSamples[chan], - &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS], - MAX_PRE_SAMPLES*sizeof(ALfloat) - ); - - /* Now resample, then filter and mix to the appropriate outputs. */ - ResampledData = Resample(&voice->SincState, - &SrcData[MAX_PRE_SAMPLES], DataPosFrac, increment, - Device->ResampledData, DstBufferSize - ); - { - DirectParams *parms = &voice->Direct; - const ALfloat *samples; - - samples = DoFilters( - &parms->Filters[chan].LowPass, &parms->Filters[chan].HighPass, - Device->FilteredData, ResampledData, DstBufferSize, - parms->Filters[chan].ActiveType - ); - if(!voice->IsHrtf) - MixSamples(samples, parms->OutChannels, parms->OutBuffer, parms->Gains[chan], - parms->Counter, OutPos, DstBufferSize); - else - MixHrtfSamples(parms->OutBuffer, samples, parms->Counter, voice->Offset, - OutPos, IrSize, &parms->Hrtf[chan].Params, - &parms->Hrtf[chan].State, DstBufferSize); - } - - for(j = 0;j < Device->NumAuxSends;j++) - { - SendParams *parms = &voice->Send[j]; - const ALfloat *samples; - - if(!parms->OutBuffer) - continue; - - samples = DoFilters( - &parms->Filters[chan].LowPass, &parms->Filters[chan].HighPass, - Device->FilteredData, ResampledData, DstBufferSize, - parms->Filters[chan].ActiveType - ); - MixSamples(samples, 1, parms->OutBuffer, &parms->Gains[chan], - parms->Counter, OutPos, DstBufferSize); - } - } - /* Update positions */ - DataPosFrac += increment*DstBufferSize; - DataPosInt += DataPosFrac>>FRACTIONBITS; - DataPosFrac &= FRACTIONMASK; - - OutPos += DstBufferSize; - voice->Offset += DstBufferSize; - voice->Direct.Counter = maxu(voice->Direct.Counter, DstBufferSize) - DstBufferSize; - for(j = 0;j < Device->NumAuxSends;j++) - voice->Send[j].Counter = maxu(voice->Send[j].Counter, DstBufferSize) - DstBufferSize; - - /* Handle looping sources */ - while(1) - { - const ALbuffer *ALBuffer; - ALuint DataSize = 0; - ALuint LoopStart = 0; - ALuint LoopEnd = 0; - - if((ALBuffer=BufferListItem->buffer) != NULL) - { - DataSize = ALBuffer->SampleLen; - LoopStart = ALBuffer->LoopStart; - LoopEnd = ALBuffer->LoopEnd; - if(LoopEnd > DataPosInt) - break; - } - - if(Looping && Source->SourceType == AL_STATIC) - { - assert(LoopEnd > LoopStart); - DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart; - break; - } - - if(DataSize > DataPosInt) - break; - - if(!(BufferListItem=BufferListItem->next)) - { - if(Looping) - BufferListItem = ATOMIC_LOAD(&Source->queue); - else - { - State = AL_STOPPED; - BufferListItem = NULL; - DataPosInt = 0; - DataPosFrac = 0; - break; - } - } - - DataPosInt -= DataSize; - } - } while(State == AL_PLAYING && OutPos < SamplesToDo); - - /* Update source info */ - Source->state = State; - ATOMIC_STORE(&Source->current_buffer, BufferListItem); - Source->position = DataPosInt; - Source->position_fraction = DataPosFrac; -} |