/** * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #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 "bs2b.h" 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 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 LoadStack(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 void SilenceStack(ALfloat *dst, ALuint samples) { ALuint i; for(i = 0;i < samples;i++) dst[i] = 0.0f; } static void Filter2P(FILTER *filter, ALuint chan, ALfloat *RESTRICT dst, const ALfloat *RESTRICT src, ALuint numsamples) { ALuint i; for(i = 0;i < numsamples;i++) dst[i] = lpFilter2P(filter, chan, src[i]); dst[i] = lpFilter2PC(filter, chan, src[i]); } ALvoid MixSource(ALsource *Source, ALCdevice *Device, ALuint SamplesToDo) { ALbufferlistitem *BufferListItem; ALuint DataPosInt, DataPosFrac; ALuint BuffersPlayed; ALboolean Looping; ALuint increment; enum Resampler Resampler; ALenum State; ALuint OutPos; ALuint NumChannels; ALuint SampleSize; ALint64 DataSize64; ALuint chan, j; /* Get source info */ State = Source->state; BuffersPlayed = Source->BuffersPlayed; DataPosInt = Source->position; DataPosFrac = Source->position_fraction; Looping = Source->Looping; increment = Source->Params.Step; Resampler = (increment==FRACTIONONE) ? PointResampler : Source->Resampler; NumChannels = Source->NumChannels; SampleSize = Source->SampleSize; /* Get current buffer queue item */ BufferListItem = Source->queue; for(j = 0;j < BuffersPlayed;j++) BufferListItem = BufferListItem->next; OutPos = 0; do { const ALuint BufferPrePadding = ResamplerPrePadding[Resampler]; const ALuint BufferPadding = ResamplerPadding[Resampler]; ALuint SrcBufferSize, DstBufferSize; /* Figure out how many buffer bytes will be needed */ DataSize64 = SamplesToDo-OutPos+1; DataSize64 *= increment; DataSize64 += DataPosFrac+FRACTIONMASK; DataSize64 >>= FRACTIONBITS; DataSize64 += BufferPadding+BufferPrePadding; SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE); /* Figure out how many samples we can actually mix from this. */ DataSize64 = SrcBufferSize; DataSize64 -= BufferPadding+BufferPrePadding; DataSize64 <<= FRACTIONBITS; DataSize64 -= increment; 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++) { ALfloat *SrcData = Device->SampleData1; ALfloat *ResampledData = Device->SampleData2; ALuint SrcDataSize = 0; if(Source->SourceType == AL_STATIC) { const ALbuffer *ALBuffer = Source->queue->buffer; const ALubyte *Data = ALBuffer->data; ALuint DataSize; ALuint pos; /* If current pos is beyond the loop range, do not loop */ if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd) { Looping = AL_FALSE; if(DataPosInt >= BufferPrePadding) pos = DataPosInt - BufferPrePadding; else { DataSize = BufferPrePadding - DataPosInt; DataSize = minu(SrcBufferSize - SrcDataSize, DataSize); SilenceStack(&SrcData[SrcDataSize], DataSize); SrcDataSize += DataSize; pos = 0; } /* Copy what's left to play in the source buffer, and clear the * rest of the temp buffer */ DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos); LoadStack(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize], NumChannels, ALBuffer->FmtType, DataSize); SrcDataSize += DataSize; SilenceStack(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize); SrcDataSize += SrcBufferSize - SrcDataSize; } else { ALuint LoopStart = ALBuffer->LoopStart; ALuint LoopEnd = ALBuffer->LoopEnd; if(DataPosInt >= LoopStart) { pos = DataPosInt-LoopStart; while(pos < BufferPrePadding) pos += LoopEnd-LoopStart; pos -= BufferPrePadding; pos += LoopStart; } else if(DataPosInt >= BufferPrePadding) pos = DataPosInt - BufferPrePadding; else { DataSize = BufferPrePadding - DataPosInt; DataSize = minu(SrcBufferSize - SrcDataSize, DataSize); SilenceStack(&SrcData[SrcDataSize], DataSize); SrcDataSize += DataSize; pos = 0; } /* Copy what's left of this loop iteration, then copy repeats * of the loop section */ DataSize = LoopEnd - pos; DataSize = minu(SrcBufferSize - SrcDataSize, DataSize); LoadStack(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize], NumChannels, ALBuffer->FmtType, DataSize); SrcDataSize += DataSize; DataSize = LoopEnd-LoopStart; while(SrcBufferSize > SrcDataSize) { DataSize = minu(SrcBufferSize - SrcDataSize, DataSize); LoadStack(&SrcData[SrcDataSize], &Data[(LoopStart*NumChannels + chan)*SampleSize], NumChannels, ALBuffer->FmtType, DataSize); SrcDataSize += DataSize; } } } else { /* Crawl the buffer queue to fill in the temp buffer */ ALbufferlistitem *tmpiter = BufferListItem; ALuint pos; if(DataPosInt >= BufferPrePadding) pos = DataPosInt - BufferPrePadding; else { pos = BufferPrePadding - DataPosInt; while(pos > 0) { if(!tmpiter->prev && !Looping) { ALuint DataSize = minu(SrcBufferSize - SrcDataSize, pos); SilenceStack(&SrcData[SrcDataSize], DataSize); SrcDataSize += DataSize; pos = 0; break; } if(tmpiter->prev) tmpiter = tmpiter->prev; else { while(tmpiter->next) tmpiter = tmpiter->next; } if(tmpiter->buffer) { if((ALuint)tmpiter->buffer->SampleLen > pos) { pos = tmpiter->buffer->SampleLen - pos; break; } pos -= tmpiter->buffer->SampleLen; } } } 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); LoadStack(&SrcData[SrcDataSize], Data, NumChannels, ALBuffer->FmtType, DataSize); SrcDataSize += DataSize; } } tmpiter = tmpiter->next; if(!tmpiter && Looping) tmpiter = Source->queue; else if(!tmpiter) { SilenceStack(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize); SrcDataSize += SrcBufferSize - SrcDataSize; } } } /* Now resample, then filter and mix to the appropriate outputs. */ Source->Params.Resample(&SrcData[BufferPrePadding], DataPosFrac, increment, ResampledData, DstBufferSize); { DirectParams *directparms = &Source->Params.Direct; Filter2P(&directparms->iirFilter, chan, SrcData, ResampledData, DstBufferSize); Source->Params.DryMix(Source, Device, directparms, SrcData, chan, OutPos, SamplesToDo, DstBufferSize); } for(j = 0;j < Device->NumAuxSends;j++) { SendParams *sendparms = &Source->Params.Send[j]; if(!sendparms->Slot) continue; Filter2P(&sendparms->iirFilter, chan, SrcData, ResampledData, DstBufferSize); Source->Params.WetMix(sendparms, SrcData, OutPos, SamplesToDo, DstBufferSize); } } /* Update positions */ for(j = 0;j < DstBufferSize;j++) { DataPosFrac += increment; DataPosInt += DataPosFrac>>FRACTIONBITS; DataPosFrac &= FRACTIONMASK; } OutPos += 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) { DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart; break; } if(DataSize > DataPosInt) break; if(BufferListItem->next) { BufferListItem = BufferListItem->next; BuffersPlayed++; } else if(Looping) { BufferListItem = Source->queue; BuffersPlayed = 0; } else { State = AL_STOPPED; BufferListItem = Source->queue; BuffersPlayed = Source->BuffersInQueue; DataPosInt = 0; DataPosFrac = 0; break; } DataPosInt -= DataSize; } } while(State == AL_PLAYING && OutPos < SamplesToDo); /* Update source info */ Source->state = State; Source->BuffersPlayed = BuffersPlayed; Source->position = DataPosInt; Source->position_fraction = DataPosFrac; Source->Hrtf.Offset += OutPos; if(State == AL_PLAYING) { Source->Hrtf.Counter = maxu(Source->Hrtf.Counter, OutPos) - OutPos; Source->Hrtf.Moving = AL_TRUE; } else { Source->Hrtf.Counter = 0; Source->Hrtf.Moving = AL_FALSE; } }