mirror of
https://github.com/etlegacy/etlegacy-libs.git
synced 2024-11-14 16:41:24 +00:00
762 lines
28 KiB
C
762 lines
28 KiB
C
/**
|
|
* 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 "sample_cvt.h"
|
|
#include "alu.h"
|
|
#include "alconfig.h"
|
|
#include "ringbuffer.h"
|
|
|
|
#include "cpu_caps.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(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALsizei *restrict pos_arr, ALsizei size);
|
|
|
|
|
|
/* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */
|
|
static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!");
|
|
|
|
|
|
enum Resampler ResamplerDefault = LinearResampler;
|
|
|
|
MixerFunc MixSamples = Mix_C;
|
|
RowMixerFunc MixRowSamples = MixRow_C;
|
|
static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
|
|
static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
|
|
|
|
static MixerFunc SelectMixer(void)
|
|
{
|
|
#ifdef HAVE_NEON
|
|
if((CPUCapFlags&CPU_CAP_NEON))
|
|
return Mix_Neon;
|
|
#endif
|
|
#ifdef HAVE_SSE
|
|
if((CPUCapFlags&CPU_CAP_SSE))
|
|
return Mix_SSE;
|
|
#endif
|
|
return Mix_C;
|
|
}
|
|
|
|
static RowMixerFunc SelectRowMixer(void)
|
|
{
|
|
#ifdef HAVE_NEON
|
|
if((CPUCapFlags&CPU_CAP_NEON))
|
|
return MixRow_Neon;
|
|
#endif
|
|
#ifdef HAVE_SSE
|
|
if((CPUCapFlags&CPU_CAP_SSE))
|
|
return MixRow_SSE;
|
|
#endif
|
|
return MixRow_C;
|
|
}
|
|
|
|
static inline HrtfMixerFunc SelectHrtfMixer(void)
|
|
{
|
|
#ifdef HAVE_NEON
|
|
if((CPUCapFlags&CPU_CAP_NEON))
|
|
return MixHrtf_Neon;
|
|
#endif
|
|
#ifdef HAVE_SSE
|
|
if((CPUCapFlags&CPU_CAP_SSE))
|
|
return MixHrtf_SSE;
|
|
#endif
|
|
return MixHrtf_C;
|
|
}
|
|
|
|
static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
|
|
{
|
|
#ifdef HAVE_NEON
|
|
if((CPUCapFlags&CPU_CAP_NEON))
|
|
return MixHrtfBlend_Neon;
|
|
#endif
|
|
#ifdef HAVE_SSE
|
|
if((CPUCapFlags&CPU_CAP_SSE))
|
|
return MixHrtfBlend_SSE;
|
|
#endif
|
|
return MixHrtfBlend_C;
|
|
}
|
|
|
|
ResamplerFunc SelectResampler(enum Resampler resampler)
|
|
{
|
|
switch(resampler)
|
|
{
|
|
case PointResampler:
|
|
return Resample_point_C;
|
|
case LinearResampler:
|
|
#ifdef HAVE_NEON
|
|
if((CPUCapFlags&CPU_CAP_NEON))
|
|
return Resample_lerp_Neon;
|
|
#endif
|
|
#ifdef HAVE_SSE4_1
|
|
if((CPUCapFlags&CPU_CAP_SSE4_1))
|
|
return Resample_lerp_SSE41;
|
|
#endif
|
|
#ifdef HAVE_SSE2
|
|
if((CPUCapFlags&CPU_CAP_SSE2))
|
|
return Resample_lerp_SSE2;
|
|
#endif
|
|
return Resample_lerp_C;
|
|
case FIR4Resampler:
|
|
return Resample_cubic_C;
|
|
case BSinc12Resampler:
|
|
case BSinc24Resampler:
|
|
#ifdef HAVE_NEON
|
|
if((CPUCapFlags&CPU_CAP_NEON))
|
|
return Resample_bsinc_Neon;
|
|
#endif
|
|
#ifdef HAVE_SSE
|
|
if((CPUCapFlags&CPU_CAP_SSE))
|
|
return Resample_bsinc_SSE;
|
|
#endif
|
|
return Resample_bsinc_C;
|
|
}
|
|
|
|
return Resample_point_C;
|
|
}
|
|
|
|
|
|
void aluInitMixer(void)
|
|
{
|
|
const char *str;
|
|
|
|
if(ConfigValueStr(NULL, NULL, "resampler", &str))
|
|
{
|
|
if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
|
|
ResamplerDefault = PointResampler;
|
|
else if(strcasecmp(str, "linear") == 0)
|
|
ResamplerDefault = LinearResampler;
|
|
else if(strcasecmp(str, "cubic") == 0)
|
|
ResamplerDefault = FIR4Resampler;
|
|
else if(strcasecmp(str, "bsinc12") == 0)
|
|
ResamplerDefault = BSinc12Resampler;
|
|
else if(strcasecmp(str, "bsinc24") == 0)
|
|
ResamplerDefault = BSinc24Resampler;
|
|
else if(strcasecmp(str, "bsinc") == 0)
|
|
{
|
|
WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
|
|
ResamplerDefault = BSinc12Resampler;
|
|
}
|
|
else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0)
|
|
{
|
|
WARN("Resampler option \"%s\" is deprecated, using cubic\n", str);
|
|
ResamplerDefault = FIR4Resampler;
|
|
}
|
|
else
|
|
{
|
|
char *end;
|
|
long n = strtol(str, &end, 0);
|
|
if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
|
|
ResamplerDefault = n;
|
|
else
|
|
WARN("Invalid resampler: %s\n", str);
|
|
}
|
|
}
|
|
|
|
MixHrtfBlendSamples = SelectHrtfBlendMixer();
|
|
MixHrtfSamples = SelectHrtfMixer();
|
|
MixSamples = SelectMixer();
|
|
MixRowSamples = SelectRowMixer();
|
|
}
|
|
|
|
|
|
static void SendAsyncEvent(ALCcontext *context, ALuint enumtype, ALenum type,
|
|
ALuint objid, ALuint param, const char *msg)
|
|
{
|
|
AsyncEvent evt = ASYNC_EVENT(enumtype);
|
|
evt.u.user.type = type;
|
|
evt.u.user.id = objid;
|
|
evt.u.user.param = param;
|
|
strcpy(evt.u.user.msg, msg);
|
|
if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1)
|
|
alsem_post(&context->EventSem);
|
|
}
|
|
|
|
|
|
static inline ALfloat Sample_ALubyte(ALubyte val)
|
|
{ return (val-128) * (1.0f/128.0f); }
|
|
|
|
static inline ALfloat Sample_ALshort(ALshort val)
|
|
{ return val * (1.0f/32768.0f); }
|
|
|
|
static inline ALfloat Sample_ALfloat(ALfloat val)
|
|
{ return val; }
|
|
|
|
static inline ALfloat Sample_ALdouble(ALdouble val)
|
|
{ return (ALfloat)val; }
|
|
|
|
typedef ALubyte ALmulaw;
|
|
static inline ALfloat Sample_ALmulaw(ALmulaw val)
|
|
{ return muLawDecompressionTable[val] * (1.0f/32768.0f); }
|
|
|
|
typedef ALubyte ALalaw;
|
|
static inline ALfloat Sample_ALalaw(ALalaw val)
|
|
{ return aLawDecompressionTable[val] * (1.0f/32768.0f); }
|
|
|
|
#define DECL_TEMPLATE(T) \
|
|
static inline void Load_##T(ALfloat *restrict dst, const T *restrict src, \
|
|
ALint srcstep, ALsizei samples) \
|
|
{ \
|
|
ALsizei i; \
|
|
for(i = 0;i < samples;i++) \
|
|
dst[i] += Sample_##T(src[i*srcstep]); \
|
|
}
|
|
|
|
DECL_TEMPLATE(ALubyte)
|
|
DECL_TEMPLATE(ALshort)
|
|
DECL_TEMPLATE(ALfloat)
|
|
DECL_TEMPLATE(ALdouble)
|
|
DECL_TEMPLATE(ALmulaw)
|
|
DECL_TEMPLATE(ALalaw)
|
|
|
|
#undef DECL_TEMPLATE
|
|
|
|
static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep,
|
|
enum FmtType srctype, ALsizei samples)
|
|
{
|
|
#define HANDLE_FMT(ET, ST) case ET: Load_##ST(dst, src, srcstep, samples); break
|
|
switch(srctype)
|
|
{
|
|
HANDLE_FMT(FmtUByte, ALubyte);
|
|
HANDLE_FMT(FmtShort, ALshort);
|
|
HANDLE_FMT(FmtFloat, ALfloat);
|
|
HANDLE_FMT(FmtDouble, ALdouble);
|
|
HANDLE_FMT(FmtMulaw, ALmulaw);
|
|
HANDLE_FMT(FmtAlaw, ALalaw);
|
|
}
|
|
#undef HANDLE_FMT
|
|
}
|
|
|
|
|
|
static const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter,
|
|
ALfloat *restrict dst, const ALfloat *restrict src,
|
|
ALsizei numsamples, enum ActiveFilters type)
|
|
{
|
|
ALsizei i;
|
|
switch(type)
|
|
{
|
|
case AF_None:
|
|
BiquadFilter_passthru(lpfilter, numsamples);
|
|
BiquadFilter_passthru(hpfilter, numsamples);
|
|
break;
|
|
|
|
case AF_LowPass:
|
|
BiquadFilter_process(lpfilter, dst, src, numsamples);
|
|
BiquadFilter_passthru(hpfilter, numsamples);
|
|
return dst;
|
|
case AF_HighPass:
|
|
BiquadFilter_passthru(lpfilter, numsamples);
|
|
BiquadFilter_process(hpfilter, dst, src, numsamples);
|
|
return dst;
|
|
|
|
case AF_BandPass:
|
|
for(i = 0;i < numsamples;)
|
|
{
|
|
ALfloat temp[256];
|
|
ALsizei todo = mini(256, numsamples-i);
|
|
|
|
BiquadFilter_process(lpfilter, temp, src+i, todo);
|
|
BiquadFilter_process(hpfilter, dst+i, temp, todo);
|
|
i += todo;
|
|
}
|
|
return dst;
|
|
}
|
|
return src;
|
|
}
|
|
|
|
|
|
/* This function uses these device temp buffers. */
|
|
#define SOURCE_DATA_BUF 0
|
|
#define RESAMPLED_BUF 1
|
|
#define FILTERED_BUF 2
|
|
#define NFC_DATA_BUF 3
|
|
ALboolean MixSource(ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo)
|
|
{
|
|
ALCdevice *Device = Context->Device;
|
|
ALbufferlistitem *BufferListItem;
|
|
ALbufferlistitem *BufferLoopItem;
|
|
ALsizei NumChannels, SampleSize;
|
|
ALbitfieldSOFT enabledevt;
|
|
ALsizei buffers_done = 0;
|
|
ResamplerFunc Resample;
|
|
ALsizei DataPosInt;
|
|
ALsizei DataPosFrac;
|
|
ALint64 DataSize64;
|
|
ALint increment;
|
|
ALsizei Counter;
|
|
ALsizei OutPos;
|
|
ALsizei IrSize;
|
|
bool isplaying;
|
|
bool firstpass;
|
|
bool isstatic;
|
|
ALsizei chan;
|
|
ALsizei send;
|
|
|
|
/* Get source info */
|
|
isplaying = true; /* Will only be called while playing. */
|
|
isstatic = !!(voice->Flags&VOICE_IS_STATIC);
|
|
DataPosInt = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
|
|
DataPosFrac = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
|
|
BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
|
|
BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
|
|
NumChannels = voice->NumChannels;
|
|
SampleSize = voice->SampleSize;
|
|
increment = voice->Step;
|
|
|
|
IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
|
|
|
|
Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
|
|
Resample_copy_C : voice->Resampler);
|
|
|
|
Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
|
|
firstpass = true;
|
|
OutPos = 0;
|
|
|
|
do {
|
|
ALsizei SrcBufferSize, DstBufferSize;
|
|
|
|
/* Figure out how many buffer samples will be needed */
|
|
DataSize64 = SamplesToDo-OutPos;
|
|
DataSize64 *= increment;
|
|
DataSize64 += DataPosFrac+FRACTIONMASK;
|
|
DataSize64 >>= FRACTIONBITS;
|
|
DataSize64 += MAX_RESAMPLE_PADDING*2;
|
|
SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
|
|
|
|
/* Figure out how many samples we can actually mix from this. */
|
|
DataSize64 = SrcBufferSize;
|
|
DataSize64 -= MAX_RESAMPLE_PADDING*2;
|
|
DataSize64 <<= FRACTIONBITS;
|
|
DataSize64 -= DataPosFrac;
|
|
DstBufferSize = (ALsizei)mini64((DataSize64+(increment-1)) / increment,
|
|
SamplesToDo - OutPos);
|
|
|
|
/* Some mixers like having a multiple of 4, so try to give that unless
|
|
* this is the last update. */
|
|
if(DstBufferSize < SamplesToDo-OutPos)
|
|
DstBufferSize &= ~3;
|
|
|
|
/* It's impossible to have a buffer list item with no entries. */
|
|
assert(BufferListItem->num_buffers > 0);
|
|
|
|
for(chan = 0;chan < NumChannels;chan++)
|
|
{
|
|
const ALfloat *ResampledData;
|
|
ALfloat *SrcData = Device->TempBuffer[SOURCE_DATA_BUF];
|
|
ALsizei FilledAmt;
|
|
|
|
/* Load the previous samples into the source data first, and clear the rest. */
|
|
memcpy(SrcData, voice->PrevSamples[chan], MAX_RESAMPLE_PADDING*sizeof(ALfloat));
|
|
memset(SrcData+MAX_RESAMPLE_PADDING, 0, (BUFFERSIZE-MAX_RESAMPLE_PADDING)*
|
|
sizeof(ALfloat));
|
|
FilledAmt = MAX_RESAMPLE_PADDING;
|
|
|
|
if(isstatic)
|
|
{
|
|
/* TODO: For static sources, loop points are taken from the
|
|
* first buffer (should be adjusted by any buffer offset, to
|
|
* possibly be added later).
|
|
*/
|
|
const ALbuffer *Buffer0 = BufferListItem->buffers[0];
|
|
const ALsizei LoopStart = Buffer0->LoopStart;
|
|
const ALsizei LoopEnd = Buffer0->LoopEnd;
|
|
const ALsizei LoopSize = LoopEnd - LoopStart;
|
|
|
|
/* If current pos is beyond the loop range, do not loop */
|
|
if(!BufferLoopItem || DataPosInt >= LoopEnd)
|
|
{
|
|
ALsizei SizeToDo = SrcBufferSize - FilledAmt;
|
|
ALsizei CompLen = 0;
|
|
ALsizei i;
|
|
|
|
BufferLoopItem = NULL;
|
|
|
|
for(i = 0;i < BufferListItem->num_buffers;i++)
|
|
{
|
|
const ALbuffer *buffer = BufferListItem->buffers[i];
|
|
const ALubyte *Data = buffer->data;
|
|
ALsizei DataSize;
|
|
|
|
if(DataPosInt >= buffer->SampleLen)
|
|
continue;
|
|
|
|
/* Load what's left to play from the buffer */
|
|
DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
|
|
CompLen = maxi(CompLen, DataSize);
|
|
|
|
LoadSamples(&SrcData[FilledAmt],
|
|
&Data[(DataPosInt*NumChannels + chan)*SampleSize],
|
|
NumChannels, buffer->FmtType, DataSize
|
|
);
|
|
}
|
|
FilledAmt += CompLen;
|
|
}
|
|
else
|
|
{
|
|
ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt);
|
|
ALsizei CompLen = 0;
|
|
ALsizei i;
|
|
|
|
for(i = 0;i < BufferListItem->num_buffers;i++)
|
|
{
|
|
const ALbuffer *buffer = BufferListItem->buffers[i];
|
|
const ALubyte *Data = buffer->data;
|
|
ALsizei DataSize;
|
|
|
|
if(DataPosInt >= buffer->SampleLen)
|
|
continue;
|
|
|
|
/* Load what's left of this loop iteration */
|
|
DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
|
|
CompLen = maxi(CompLen, DataSize);
|
|
|
|
LoadSamples(&SrcData[FilledAmt],
|
|
&Data[(DataPosInt*NumChannels + chan)*SampleSize],
|
|
NumChannels, buffer->FmtType, DataSize
|
|
);
|
|
}
|
|
FilledAmt += CompLen;
|
|
|
|
while(SrcBufferSize > FilledAmt)
|
|
{
|
|
const ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopSize);
|
|
|
|
CompLen = 0;
|
|
for(i = 0;i < BufferListItem->num_buffers;i++)
|
|
{
|
|
const ALbuffer *buffer = BufferListItem->buffers[i];
|
|
const ALubyte *Data = buffer->data;
|
|
ALsizei DataSize;
|
|
|
|
if(LoopStart >= buffer->SampleLen)
|
|
continue;
|
|
|
|
DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart);
|
|
CompLen = maxi(CompLen, DataSize);
|
|
|
|
LoadSamples(&SrcData[FilledAmt],
|
|
&Data[(LoopStart*NumChannels + chan)*SampleSize],
|
|
NumChannels, buffer->FmtType, DataSize
|
|
);
|
|
}
|
|
FilledAmt += CompLen;
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* Crawl the buffer queue to fill in the temp buffer */
|
|
ALbufferlistitem *tmpiter = BufferListItem;
|
|
ALsizei pos = DataPosInt;
|
|
|
|
while(tmpiter && SrcBufferSize > FilledAmt)
|
|
{
|
|
ALsizei SizeToDo = SrcBufferSize - FilledAmt;
|
|
ALsizei CompLen = 0;
|
|
ALsizei i;
|
|
|
|
for(i = 0;i < tmpiter->num_buffers;i++)
|
|
{
|
|
const ALbuffer *ALBuffer = tmpiter->buffers[i];
|
|
ALsizei DataSize = ALBuffer ? ALBuffer->SampleLen : 0;
|
|
|
|
if(DataSize > pos)
|
|
{
|
|
const ALubyte *Data = ALBuffer->data;
|
|
Data += (pos*NumChannels + chan)*SampleSize;
|
|
|
|
DataSize = mini(SizeToDo, DataSize - pos);
|
|
CompLen = maxi(CompLen, DataSize);
|
|
|
|
LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
|
|
ALBuffer->FmtType, DataSize);
|
|
}
|
|
}
|
|
if(UNLIKELY(!CompLen))
|
|
pos -= tmpiter->max_samples;
|
|
else
|
|
{
|
|
FilledAmt += CompLen;
|
|
if(SrcBufferSize <= FilledAmt)
|
|
break;
|
|
pos = 0;
|
|
}
|
|
tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
|
|
if(!tmpiter) tmpiter = BufferLoopItem;
|
|
}
|
|
}
|
|
|
|
/* Store the last source samples used for next time. */
|
|
memcpy(voice->PrevSamples[chan],
|
|
&SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
|
|
MAX_RESAMPLE_PADDING*sizeof(ALfloat)
|
|
);
|
|
|
|
/* Now resample, then filter and mix to the appropriate outputs. */
|
|
ResampledData = Resample(&voice->ResampleState,
|
|
&SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
|
|
Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
|
|
);
|
|
{
|
|
DirectParams *parms = &voice->Direct.Params[chan];
|
|
const ALfloat *samples;
|
|
|
|
samples = DoFilters(
|
|
&parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
|
|
ResampledData, DstBufferSize, voice->Direct.FilterType
|
|
);
|
|
if(!(voice->Flags&VOICE_HAS_HRTF))
|
|
{
|
|
if(!Counter)
|
|
memcpy(parms->Gains.Current, parms->Gains.Target,
|
|
sizeof(parms->Gains.Current));
|
|
if(!(voice->Flags&VOICE_HAS_NFC))
|
|
MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
|
|
parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
|
|
DstBufferSize
|
|
);
|
|
else
|
|
{
|
|
ALfloat *nfcsamples = Device->TempBuffer[NFC_DATA_BUF];
|
|
ALsizei chanoffset = 0;
|
|
|
|
MixSamples(samples,
|
|
voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
|
|
parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
|
|
DstBufferSize
|
|
);
|
|
chanoffset += voice->Direct.ChannelsPerOrder[0];
|
|
#define APPLY_NFC_MIX(order) \
|
|
if(voice->Direct.ChannelsPerOrder[order] > 0) \
|
|
{ \
|
|
NfcFilterProcess##order(&parms->NFCtrlFilter, nfcsamples, samples, \
|
|
DstBufferSize); \
|
|
MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order], \
|
|
voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
|
|
parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize \
|
|
); \
|
|
chanoffset += voice->Direct.ChannelsPerOrder[order]; \
|
|
}
|
|
APPLY_NFC_MIX(1)
|
|
APPLY_NFC_MIX(2)
|
|
APPLY_NFC_MIX(3)
|
|
#undef APPLY_NFC_MIX
|
|
}
|
|
}
|
|
else
|
|
{
|
|
MixHrtfParams hrtfparams;
|
|
ALsizei fademix = 0;
|
|
int lidx, ridx;
|
|
|
|
lidx = GetChannelIdxByName(&Device->RealOut, FrontLeft);
|
|
ridx = GetChannelIdxByName(&Device->RealOut, FrontRight);
|
|
assert(lidx != -1 && ridx != -1);
|
|
|
|
if(!Counter)
|
|
{
|
|
/* No fading, just overwrite the old HRTF params. */
|
|
parms->Hrtf.Old = parms->Hrtf.Target;
|
|
}
|
|
else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
|
|
{
|
|
/* The old HRTF params are silent, so overwrite the old
|
|
* coefficients with the new, and reset the old gain to
|
|
* 0. The future mix will then fade from silence.
|
|
*/
|
|
parms->Hrtf.Old = parms->Hrtf.Target;
|
|
parms->Hrtf.Old.Gain = 0.0f;
|
|
}
|
|
else if(firstpass)
|
|
{
|
|
ALfloat gain;
|
|
|
|
/* Fade between the coefficients over 128 samples. */
|
|
fademix = mini(DstBufferSize, 128);
|
|
|
|
/* The new coefficients need to fade in completely
|
|
* since they're replacing the old ones. To keep the
|
|
* gain fading consistent, interpolate between the old
|
|
* and new target gains given how much of the fade time
|
|
* this mix handles.
|
|
*/
|
|
gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
|
|
minf(1.0f, (ALfloat)fademix/Counter));
|
|
hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
|
|
hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
|
|
hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
|
|
hrtfparams.Gain = 0.0f;
|
|
hrtfparams.GainStep = gain / (ALfloat)fademix;
|
|
|
|
MixHrtfBlendSamples(
|
|
voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
|
|
samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
|
|
&hrtfparams, &parms->Hrtf.State, fademix
|
|
);
|
|
/* Update the old parameters with the result. */
|
|
parms->Hrtf.Old = parms->Hrtf.Target;
|
|
if(fademix < Counter)
|
|
parms->Hrtf.Old.Gain = hrtfparams.Gain;
|
|
}
|
|
|
|
if(fademix < DstBufferSize)
|
|
{
|
|
ALsizei todo = DstBufferSize - fademix;
|
|
ALfloat gain = parms->Hrtf.Target.Gain;
|
|
|
|
/* Interpolate the target gain if the gain fading lasts
|
|
* longer than this mix.
|
|
*/
|
|
if(Counter > DstBufferSize)
|
|
gain = lerp(parms->Hrtf.Old.Gain, gain,
|
|
(ALfloat)todo/(Counter-fademix));
|
|
|
|
hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
|
|
hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
|
|
hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
|
|
hrtfparams.Gain = parms->Hrtf.Old.Gain;
|
|
hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
|
|
MixHrtfSamples(
|
|
voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
|
|
samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
|
|
&hrtfparams, &parms->Hrtf.State, todo
|
|
);
|
|
/* Store the interpolated gain or the final target gain
|
|
* depending if the fade is done.
|
|
*/
|
|
if(DstBufferSize < Counter)
|
|
parms->Hrtf.Old.Gain = gain;
|
|
else
|
|
parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
|
|
}
|
|
}
|
|
}
|
|
|
|
for(send = 0;send < Device->NumAuxSends;send++)
|
|
{
|
|
SendParams *parms = &voice->Send[send].Params[chan];
|
|
const ALfloat *samples;
|
|
|
|
if(!voice->Send[send].Buffer)
|
|
continue;
|
|
|
|
samples = DoFilters(
|
|
&parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
|
|
ResampledData, DstBufferSize, voice->Send[send].FilterType
|
|
);
|
|
|
|
if(!Counter)
|
|
memcpy(parms->Gains.Current, parms->Gains.Target,
|
|
sizeof(parms->Gains.Current));
|
|
MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
|
|
parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
|
|
);
|
|
}
|
|
}
|
|
/* Update positions */
|
|
DataPosFrac += increment*DstBufferSize;
|
|
DataPosInt += DataPosFrac>>FRACTIONBITS;
|
|
DataPosFrac &= FRACTIONMASK;
|
|
|
|
OutPos += DstBufferSize;
|
|
voice->Offset += DstBufferSize;
|
|
Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
|
|
firstpass = false;
|
|
|
|
if(isstatic)
|
|
{
|
|
if(BufferLoopItem)
|
|
{
|
|
/* Handle looping static source */
|
|
const ALbuffer *Buffer = BufferListItem->buffers[0];
|
|
ALsizei LoopStart = Buffer->LoopStart;
|
|
ALsizei LoopEnd = Buffer->LoopEnd;
|
|
if(DataPosInt >= LoopEnd)
|
|
{
|
|
assert(LoopEnd > LoopStart);
|
|
DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* Handle non-looping static source */
|
|
if(DataPosInt >= BufferListItem->max_samples)
|
|
{
|
|
isplaying = false;
|
|
BufferListItem = NULL;
|
|
DataPosInt = 0;
|
|
DataPosFrac = 0;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
else while(1)
|
|
{
|
|
/* Handle streaming source */
|
|
if(BufferListItem->max_samples > DataPosInt)
|
|
break;
|
|
|
|
DataPosInt -= BufferListItem->max_samples;
|
|
|
|
buffers_done += BufferListItem->num_buffers;
|
|
BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_relaxed);
|
|
if(!BufferListItem && !(BufferListItem=BufferLoopItem))
|
|
{
|
|
isplaying = false;
|
|
DataPosInt = 0;
|
|
DataPosFrac = 0;
|
|
break;
|
|
}
|
|
}
|
|
} while(isplaying && OutPos < SamplesToDo);
|
|
|
|
voice->Flags |= VOICE_IS_FADING;
|
|
|
|
/* Update source info */
|
|
ATOMIC_STORE(&voice->position, DataPosInt, almemory_order_relaxed);
|
|
ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
|
|
ATOMIC_STORE(&voice->current_buffer, BufferListItem, almemory_order_release);
|
|
|
|
/* Send any events now, after the position/buffer info was updated. */
|
|
enabledevt = ATOMIC_LOAD(&Context->EnabledEvts, almemory_order_acquire);
|
|
if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
|
|
SendAsyncEvent(Context, EventType_BufferCompleted,
|
|
AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, SourceID, buffers_done, "Buffer completed"
|
|
);
|
|
|
|
return isplaying;
|
|
}
|