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Add OpenAL-Soft 1.15.1.
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rude
@@ -0,0 +1,421 @@
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/**
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* OpenAL cross platform audio library
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* Copyright (C) 1999-2007 by authors.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include <math.h>
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#include <stdlib.h>
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#include <string.h>
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#include <ctype.h>
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#include <assert.h>
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#include "alMain.h"
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#include "AL/al.h"
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#include "AL/alc.h"
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#include "alSource.h"
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#include "alBuffer.h"
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#include "alListener.h"
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#include "alAuxEffectSlot.h"
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#include "alu.h"
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#include "bs2b.h"
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static __inline ALfloat Sample_ALbyte(ALbyte val)
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{ return val * (1.0f/127.0f); }
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static __inline ALfloat Sample_ALshort(ALshort val)
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{ return val * (1.0f/32767.0f); }
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static __inline ALfloat Sample_ALfloat(ALfloat val)
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{ return val; }
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#define DECL_TEMPLATE(T) \
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static void Load_##T(ALfloat *dst, const T *src, ALuint srcstep, ALuint samples)\
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{ \
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ALuint i; \
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for(i = 0;i < samples;i++) \
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dst[i] = Sample_##T(src[i*srcstep]); \
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}
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DECL_TEMPLATE(ALbyte)
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DECL_TEMPLATE(ALshort)
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DECL_TEMPLATE(ALfloat)
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#undef DECL_TEMPLATE
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static void LoadData(ALfloat *dst, const ALvoid *src, ALuint srcstep, enum FmtType srctype, ALuint samples)
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{
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switch(srctype)
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{
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case FmtByte:
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Load_ALbyte(dst, src, srcstep, samples);
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break;
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case FmtShort:
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Load_ALshort(dst, src, srcstep, samples);
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break;
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case FmtFloat:
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Load_ALfloat(dst, src, srcstep, samples);
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break;
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}
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}
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static void SilenceData(ALfloat *dst, ALuint samples)
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{
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ALuint i;
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for(i = 0;i < samples;i++)
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dst[i] = 0.0f;
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}
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static void Filter2P(FILTER *filter, ALuint chan, ALfloat *RESTRICT dst,
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const ALfloat *RESTRICT src, ALuint numsamples)
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{
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ALuint i;
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for(i = 0;i < numsamples;i++)
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dst[i] = lpFilter2P(filter, chan, src[i]);
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dst[i] = lpFilter2PC(filter, chan, src[i]);
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}
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ALvoid MixSource(ALsource *Source, ALCdevice *Device, ALuint SamplesToDo)
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{
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ALbufferlistitem *BufferListItem;
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ALuint DataPosInt, DataPosFrac;
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ALuint BuffersPlayed;
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ALboolean Looping;
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ALuint increment;
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enum Resampler Resampler;
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ALenum State;
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ALuint OutPos;
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ALuint NumChannels;
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ALuint SampleSize;
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ALint64 DataSize64;
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ALuint chan, j;
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/* Get source info */
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State = Source->state;
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BuffersPlayed = Source->BuffersPlayed;
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DataPosInt = Source->position;
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DataPosFrac = Source->position_fraction;
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Looping = Source->Looping;
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increment = Source->Params.Step;
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Resampler = (increment==FRACTIONONE) ? PointResampler : Source->Resampler;
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NumChannels = Source->NumChannels;
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SampleSize = Source->SampleSize;
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/* Get current buffer queue item */
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BufferListItem = Source->queue;
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for(j = 0;j < BuffersPlayed;j++)
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BufferListItem = BufferListItem->next;
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OutPos = 0;
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do {
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const ALuint BufferPrePadding = ResamplerPrePadding[Resampler];
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const ALuint BufferPadding = ResamplerPadding[Resampler];
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ALuint SrcBufferSize, DstBufferSize;
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/* Figure out how many buffer bytes will be needed */
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DataSize64 = SamplesToDo-OutPos+1;
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DataSize64 *= increment;
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DataSize64 += DataPosFrac+FRACTIONMASK;
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DataSize64 >>= FRACTIONBITS;
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DataSize64 += BufferPadding+BufferPrePadding;
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SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE);
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/* Figure out how many samples we can actually mix from this. */
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DataSize64 = SrcBufferSize;
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DataSize64 -= BufferPadding+BufferPrePadding;
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DataSize64 <<= FRACTIONBITS;
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DataSize64 -= increment;
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DataSize64 -= DataPosFrac;
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DstBufferSize = (ALuint)((DataSize64+(increment-1)) / increment);
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DstBufferSize = minu(DstBufferSize, (SamplesToDo-OutPos));
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/* Some mixers like having a multiple of 4, so try to give that unless
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* this is the last update. */
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if(OutPos+DstBufferSize < SamplesToDo)
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DstBufferSize &= ~3;
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for(chan = 0;chan < NumChannels;chan++)
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{
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ALfloat *SrcData = Device->SampleData1;
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ALfloat *ResampledData = Device->SampleData2;
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ALuint SrcDataSize = 0;
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if(Source->SourceType == AL_STATIC)
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{
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const ALbuffer *ALBuffer = Source->queue->buffer;
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const ALubyte *Data = ALBuffer->data;
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ALuint DataSize;
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ALuint pos;
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/* If current pos is beyond the loop range, do not loop */
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if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd)
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{
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Looping = AL_FALSE;
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if(DataPosInt >= BufferPrePadding)
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pos = DataPosInt - BufferPrePadding;
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else
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{
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DataSize = BufferPrePadding - DataPosInt;
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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SilenceData(&SrcData[SrcDataSize], DataSize);
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SrcDataSize += DataSize;
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pos = 0;
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}
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/* Copy what's left to play in the source buffer, and clear the
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* rest of the temp buffer */
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DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos);
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LoadData(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
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NumChannels, ALBuffer->FmtType, DataSize);
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SrcDataSize += DataSize;
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SilenceData(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
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SrcDataSize += SrcBufferSize - SrcDataSize;
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}
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else
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{
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ALuint LoopStart = ALBuffer->LoopStart;
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ALuint LoopEnd = ALBuffer->LoopEnd;
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if(DataPosInt >= LoopStart)
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{
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pos = DataPosInt-LoopStart;
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while(pos < BufferPrePadding)
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pos += LoopEnd-LoopStart;
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pos -= BufferPrePadding;
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pos += LoopStart;
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}
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else if(DataPosInt >= BufferPrePadding)
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pos = DataPosInt - BufferPrePadding;
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else
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{
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DataSize = BufferPrePadding - DataPosInt;
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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SilenceData(&SrcData[SrcDataSize], DataSize);
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SrcDataSize += DataSize;
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pos = 0;
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}
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/* Copy what's left of this loop iteration, then copy repeats
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* of the loop section */
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DataSize = LoopEnd - pos;
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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LoadData(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
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NumChannels, ALBuffer->FmtType, DataSize);
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SrcDataSize += DataSize;
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DataSize = LoopEnd-LoopStart;
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while(SrcBufferSize > SrcDataSize)
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{
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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LoadData(&SrcData[SrcDataSize], &Data[(LoopStart*NumChannels + chan)*SampleSize],
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NumChannels, ALBuffer->FmtType, DataSize);
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SrcDataSize += DataSize;
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}
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}
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}
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else
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{
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/* Crawl the buffer queue to fill in the temp buffer */
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ALbufferlistitem *tmpiter = BufferListItem;
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ALuint pos;
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if(DataPosInt >= BufferPrePadding)
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pos = DataPosInt - BufferPrePadding;
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else
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{
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pos = BufferPrePadding - DataPosInt;
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while(pos > 0)
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{
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if(!tmpiter->prev && !Looping)
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{
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ALuint DataSize = minu(SrcBufferSize - SrcDataSize, pos);
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SilenceData(&SrcData[SrcDataSize], DataSize);
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SrcDataSize += DataSize;
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pos = 0;
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break;
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}
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if(tmpiter->prev)
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tmpiter = tmpiter->prev;
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else
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{
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while(tmpiter->next)
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tmpiter = tmpiter->next;
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}
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if(tmpiter->buffer)
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{
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if((ALuint)tmpiter->buffer->SampleLen > pos)
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{
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pos = tmpiter->buffer->SampleLen - pos;
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break;
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}
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pos -= tmpiter->buffer->SampleLen;
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}
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}
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}
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while(tmpiter && SrcBufferSize > SrcDataSize)
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{
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const ALbuffer *ALBuffer;
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if((ALBuffer=tmpiter->buffer) != NULL)
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{
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const ALubyte *Data = ALBuffer->data;
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ALuint DataSize = ALBuffer->SampleLen;
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/* Skip the data already played */
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if(DataSize <= pos)
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pos -= DataSize;
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else
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{
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Data += (pos*NumChannels + chan)*SampleSize;
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DataSize -= pos;
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pos -= pos;
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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LoadData(&SrcData[SrcDataSize], Data, NumChannels,
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ALBuffer->FmtType, DataSize);
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SrcDataSize += DataSize;
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}
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}
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tmpiter = tmpiter->next;
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if(!tmpiter && Looping)
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tmpiter = Source->queue;
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else if(!tmpiter)
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{
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SilenceData(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
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SrcDataSize += SrcBufferSize - SrcDataSize;
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}
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}
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}
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/* Now resample, then filter and mix to the appropriate outputs. */
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Source->Params.Resample(&SrcData[BufferPrePadding], DataPosFrac,
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increment, ResampledData, DstBufferSize);
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{
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DirectParams *directparms = &Source->Params.Direct;
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Filter2P(&directparms->iirFilter, chan, SrcData, ResampledData,
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DstBufferSize);
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Source->Params.DryMix(directparms, SrcData, chan, OutPos,
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SamplesToDo, DstBufferSize);
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}
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for(j = 0;j < Device->NumAuxSends;j++)
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{
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SendParams *sendparms = &Source->Params.Send[j];
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if(!sendparms->Slot)
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continue;
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Filter2P(&sendparms->iirFilter, chan, SrcData, ResampledData,
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DstBufferSize);
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Source->Params.WetMix(sendparms, SrcData, OutPos,
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SamplesToDo, DstBufferSize);
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}
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}
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/* Update positions */
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for(j = 0;j < DstBufferSize;j++)
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{
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DataPosFrac += increment;
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DataPosInt += DataPosFrac>>FRACTIONBITS;
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DataPosFrac &= FRACTIONMASK;
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}
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OutPos += DstBufferSize;
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/* Handle looping sources */
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while(1)
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{
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const ALbuffer *ALBuffer;
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ALuint DataSize = 0;
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ALuint LoopStart = 0;
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ALuint LoopEnd = 0;
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if((ALBuffer=BufferListItem->buffer) != NULL)
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{
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DataSize = ALBuffer->SampleLen;
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LoopStart = ALBuffer->LoopStart;
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LoopEnd = ALBuffer->LoopEnd;
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if(LoopEnd > DataPosInt)
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break;
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}
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if(Looping && Source->SourceType == AL_STATIC)
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{
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DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
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break;
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}
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if(DataSize > DataPosInt)
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break;
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if(BufferListItem->next)
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{
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BufferListItem = BufferListItem->next;
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BuffersPlayed++;
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}
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else if(Looping)
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{
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BufferListItem = Source->queue;
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BuffersPlayed = 0;
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}
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else
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{
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State = AL_STOPPED;
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BufferListItem = Source->queue;
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BuffersPlayed = Source->BuffersInQueue;
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DataPosInt = 0;
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DataPosFrac = 0;
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break;
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}
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DataPosInt -= DataSize;
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}
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} while(State == AL_PLAYING && OutPos < SamplesToDo);
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/* Update source info */
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Source->state = State;
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Source->BuffersPlayed = BuffersPlayed;
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Source->position = DataPosInt;
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Source->position_fraction = DataPosFrac;
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Source->Hrtf.Offset += OutPos;
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if(State == AL_PLAYING)
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Source->Hrtf.Counter = maxu(Source->Hrtf.Counter, OutPos) - OutPos;
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else
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{
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Source->Hrtf.Counter = 0;
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Source->Hrtf.Moving = AL_FALSE;
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}
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}
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