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/**
* 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"
extern inline void InitiatePositionArrays(ALuint frac, ALuint increment, ALuint *frac_arr, ALuint *pos_arr, ALuint size);
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 CubicResampler:
return Resample_cubic32_C;
case ResamplerMax:
/* Shouldn't happen */
break;
}
return Resample_point32_C;
}
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 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 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:
break;
case AF_LowPass:
ALfilterState_process(lpfilter, dst, src, numsamples);
return dst;
case AF_HighPass:
ALfilterState_process(hpfilter, dst, src, numsamples);
return dst;
case AF_BandPass:
for(i = 0;i < numsamples;)
{
ALfloat temp[64];
ALuint todo = minu(64, 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)
{
MixerFunc Mix;
ResamplerFunc Resample;
ALbufferlistitem *BufferListItem;
ALuint DataPosInt, DataPosFrac;
ALboolean isbformat = AL_FALSE;
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;
BufferListItem = ATOMIC_LOAD(&Source->current_buffer);
DataPosInt = Source->position;
DataPosFrac = Source->position_fraction;
Looping = Source->Looping;
Resampler = Source->Resampler;
NumChannels = Source->NumChannels;
SampleSize = Source->SampleSize;
increment = voice->Step;
while(BufferListItem)
{
ALbuffer *buffer;
if((buffer=BufferListItem->buffer) != NULL)
{
isbformat = (buffer->FmtChannels == FmtBFormat2D ||
buffer->FmtChannels == FmtBFormat3D);
break;
}
BufferListItem = BufferListItem->next;
}
Mix = SelectMixer();
Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
Resample_copy32_C : SelectResampler(Resampler));
OutPos = 0;
do {
const ALuint BufferPrePadding = ResamplerPrePadding[Resampler];
const ALuint BufferPadding = ResamplerPadding[Resampler];
ALuint SrcBufferSize, DstBufferSize;
/* Figure out how many buffer samples will be needed */
DataSize64 = SamplesToDo-OutPos;
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 -= 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 = 0;
if(Source->SourceType == AL_STATIC)
{
const ALbuffer *ALBuffer = BufferListItem->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);
SilenceSamples(&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);
LoadSamples(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
NumChannels, ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
SilenceSamples(&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);
SilenceSamples(&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);
LoadSamples(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
NumChannels, ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
DataSize = LoopEnd-LoopStart;
while(SrcBufferSize > SrcDataSize)
{
DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
LoadSamples(&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)
{
ALbufferlistitem *prev;
if((prev=tmpiter->prev) != NULL)
tmpiter = prev;
else if(Looping)
{
while(tmpiter->next)
tmpiter = tmpiter->next;
}
else
{
ALuint DataSize = minu(SrcBufferSize - SrcDataSize, pos);
SilenceSamples(&SrcData[SrcDataSize], DataSize);
SrcDataSize += DataSize;
pos = 0;
break;
}
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);
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;
}
}
}
/* Now resample, then filter and mix to the appropriate outputs. */
ResampledData = Resample(
&SrcData[BufferPrePadding], 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
);
Mix(samples, Device->NumChannels, parms->OutBuffer, parms->Mix.Gains[chan],
parms->Counter, OutPos, DstBufferSize);
}
/* Only the first channel for B-Format buffers (W channel) goes to
* the send paths. */
if(chan > 0 && isbformat)
continue;
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
);
Mix(samples, 1, parms->OutBuffer, &parms->Gain,
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;
}
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