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AuroraOpenALSoft/Alc/mixer.c

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Move the core mixer functions to a separate source file
2010-08-04 06:19:36 +00:00
/**
* 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 <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 "bs2b.h"
#define FRACTIONBITS 14
#define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
#define MAX_PITCH 65536
/* Minimum ramp length in milliseconds. The value below was chosen to
* adequately reduce clicks and pops from harsh gain changes. */
#define MIN_RAMP_LENGTH 16
static __inline ALfloat aluF2F(ALfloat Value)
{
return Value;
}
static __inline ALshort aluF2S(ALfloat Value)
{
ALint i;
if(Value < 0.0f)
{
i = (ALint)(Value*32768.0f);
i = max(-32768, i);
}
else
{
i = (ALint)(Value*32767.0f);
i = min( 32767, i);
}
return ((ALshort)i);
}
static __inline ALubyte aluF2UB(ALfloat Value)
{
ALshort i = aluF2S(Value);
return (i>>8)+128;
}
static __inline ALfloat point(ALfloat val1, ALfloat val2, ALint frac)
{
return val1;
(void)val2;
(void)frac;
}
static __inline ALfloat lerp(ALfloat val1, ALfloat val2, ALint frac)
{
return val1 + ((val2-val1)*(frac * (1.0f/(1<<FRACTIONBITS))));
}
static __inline ALfloat cos_lerp(ALfloat val1, ALfloat val2, ALint frac)
{
ALfloat mult = (1.0f-cos(frac * (1.0f/(1<<FRACTIONBITS)) * M_PI)) * 0.5f;
return val1 + ((val2-val1)*mult);
}
static void MixSource(ALsource *ALSource, ALCcontext *ALContext,
float (*DryBuffer)[OUTPUTCHANNELS], ALuint SamplesToDo)
{
static float DummyBuffer[BUFFERSIZE];
ALfloat *WetBuffer[MAX_SENDS];
ALfloat DrySend[OUTPUTCHANNELS];
ALfloat dryGainStep[OUTPUTCHANNELS];
ALfloat wetGainStep[MAX_SENDS];
ALuint i, j, out;
ALfloat value, outsamp;
ALbufferlistitem *BufferListItem;
ALint64 DataSize64,DataPos64;
FILTER *DryFilter, *WetFilter[MAX_SENDS];
ALfloat WetSend[MAX_SENDS];
ALuint rampLength;
ALuint DeviceFreq;
ALint increment;
ALuint DataPosInt, DataPosFrac;
resampler_t Resampler;
ALuint BuffersPlayed;
ALboolean Looping;
ALenum State;
if(ALSource->NeedsUpdate)
{
ALsource_Update(ALSource, ALContext);
ALSource->NeedsUpdate = AL_FALSE;
}
/* Get source info */
Resampler = ALSource->Resampler;
State = ALSource->state;
BuffersPlayed = ALSource->BuffersPlayed;
DataPosInt = ALSource->position;
DataPosFrac = ALSource->position_fraction;
Looping = ALSource->bLooping;
if(ALSource->FirstStart)
{
for(i = 0;i < OUTPUTCHANNELS;i++)
DrySend[i] = ALSource->Params.DryGains[i];
for(i = 0;i < MAX_SENDS;i++)
WetSend[i] = ALSource->Params.WetGains[i];
}
else
{
for(i = 0;i < OUTPUTCHANNELS;i++)
DrySend[i] = ALSource->DryGains[i];
for(i = 0;i < MAX_SENDS;i++)
WetSend[i] = ALSource->WetGains[i];
}
Calculate gain steps once during a source mix
2010-08-06 14:55:00 +00:00
/* Compute the gain steps for each output channel */
Calculate the actual stepping value in the source update method
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DeviceFreq = ALContext->Device->Frequency;
Calculate gain steps once during a source mix
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rampLength = DeviceFreq * MIN_RAMP_LENGTH / 1000;
rampLength = max(rampLength, SamplesToDo);
for(i = 0;i < OUTPUTCHANNELS;i++)
dryGainStep[i] = (ALSource->Params.DryGains[i]-DrySend[i]) /
rampLength;
for(i = 0;i < MAX_SENDS;i++)
wetGainStep[i] = (ALSource->Params.WetGains[i]-WetSend[i]) /
rampLength;
Move the core mixer functions to a separate source file
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DryFilter = &ALSource->Params.iirFilter;
for(i = 0;i < MAX_SENDS;i++)
{
WetFilter[i] = &ALSource->Params.Send[i].iirFilter;
WetBuffer[i] = (ALSource->Send[i].Slot ?
ALSource->Send[i].Slot->WetBuffer :
DummyBuffer);
}
/* Get current buffer queue item */
BufferListItem = ALSource->queue;
Remove an unnecessary check The number of buffers played should never exceed the buffer count when playing, and the buffer list is used later on regardless
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for(i = 0;i < BuffersPlayed;i++)
Move the core mixer functions to a separate source file
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BufferListItem = BufferListItem->next;
j = 0;
do {
Calculate the actual stepping value in the source update method
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const ALbuffer *ALBuffer;
Move the core mixer functions to a separate source file
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ALfloat *Data = NULL;
Calculate the actual stepping value in the source update method
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ALuint DataSize = 0;
Move the core mixer functions to a separate source file
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ALuint LoopStart = 0;
ALuint LoopEnd = 0;
Calculate the actual stepping value in the source update method
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ALuint Channels, Bytes;
Move the core mixer functions to a separate source file
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ALuint BufferSize;
/* Get buffer info */
if((ALBuffer=BufferListItem->buffer) != NULL)
{
Data = ALBuffer->data;
DataSize = ALBuffer->size;
Calculate the actual stepping value in the source update method
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DataSize /= aluFrameSizeFromFormat(ALBuffer->format);
Move the core mixer functions to a separate source file
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LoopStart = ALBuffer->LoopStart;
LoopEnd = ALBuffer->LoopEnd;
Calculate the actual stepping value in the source update method
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Channels = aluChannelsFromFormat(ALBuffer->format);
Bytes = aluBytesFromFormat(ALBuffer->format);
Move the core mixer functions to a separate source file
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}
if(Looping && ALSource->lSourceType == AL_STATIC)
{
/* If current offset is beyond the loop range, do not loop */
if(DataPosInt >= LoopEnd)
Looping = AL_FALSE;
}
if(!Looping || ALSource->lSourceType != AL_STATIC)
{
/* Non-looping and non-static sources ignore loop points */
LoopStart = 0;
LoopEnd = DataSize;
}
if(DataPosInt >= DataSize)
goto skipmix;
if(BufferListItem->next)
{
ALbuffer *NextBuf = BufferListItem->next->buffer;
if(NextBuf && NextBuf->size)
{
ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
ulExtraSamples = min(NextBuf->size, ulExtraSamples);
memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
}
}
else if(Looping)
{
ALbuffer *NextBuf = ALSource->queue->buffer;
if(NextBuf && NextBuf->size)
{
ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
ulExtraSamples = min(NextBuf->size, ulExtraSamples);
memcpy(&Data[DataSize*Channels], &NextBuf->data[LoopStart*Channels], ulExtraSamples);
}
}
else
memset(&Data[DataSize*Channels], 0, (BUFFER_PADDING*Channels*Bytes));
Calculate the actual stepping value in the source update method
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/* Compute 18.14 fixed point step */
if(ALSource->Params.Pitch > (float)MAX_PITCH)
increment = MAX_PITCH << FRACTIONBITS;
else if(!(ALSource->Params.Pitch > 0.f))
increment = (1<<FRACTIONBITS);
else
{
increment = (ALint)(ALSource->Params.Pitch*(1L<<FRACTIONBITS));
if(increment == 0) increment = 1;
}
Move the core mixer functions to a separate source file
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/* Figure out how many samples we can mix. */
DataSize64 = LoopEnd;
DataSize64 <<= FRACTIONBITS;
DataPos64 = DataPosInt;
DataPos64 <<= FRACTIONBITS;
DataPos64 += DataPosFrac;
BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment);
BufferSize = min(BufferSize, (SamplesToDo-j));
/* Actual sample mixing loops */
if(Channels == 1) /* Mono */
{
#define DO_MIX(resampler) do { \
while(BufferSize--) \
{ \
for(i = 0;i < OUTPUTCHANNELS;i++) \
DrySend[i] += dryGainStep[i]; \
for(i = 0;i < MAX_SENDS;i++) \
WetSend[i] += wetGainStep[i]; \
\
/* First order interpolator */ \
value = (resampler)(Data[DataPosInt], Data[DataPosInt+1], \
DataPosFrac); \
\
/* Direct path final mix buffer and panning */ \
outsamp = lpFilter4P(DryFilter, 0, value); \
DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT]; \
DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT]; \
DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT]; \
DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT]; \
DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT]; \
DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT]; \
DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER]; \
DryBuffer[j][BACK_CENTER] += outsamp*DrySend[BACK_CENTER]; \
\
/* Room path final mix buffer and panning */ \
for(i = 0;i < MAX_SENDS;i++) \
{ \
outsamp = lpFilter2P(WetFilter[i], 0, value); \
WetBuffer[i][j] += outsamp*WetSend[i]; \
} \
\
DataPosFrac += increment; \
DataPosInt += DataPosFrac>>FRACTIONBITS; \
DataPosFrac &= FRACTIONMASK; \
j++; \
} \
} while(0)
switch(Resampler)
{
case POINT_RESAMPLER:
DO_MIX(point); break;
case LINEAR_RESAMPLER:
DO_MIX(lerp); break;
case COSINE_RESAMPLER:
DO_MIX(cos_lerp); break;
case RESAMPLER_MIN:
case RESAMPLER_MAX:
break;
}
#undef DO_MIX
}
Combine stereo and stereo-duplicate mixers Use a separate update method for stereo sources to the set appropriate channel gains
2010-08-05 08:07:20 +00:00
else if(Channels == 2) /* Stereo */
Move the core mixer functions to a separate source file
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{
const int chans[] = {
FRONT_LEFT, FRONT_RIGHT
};
const int chans2[] = {
BACK_LEFT, SIDE_LEFT, BACK_RIGHT, SIDE_RIGHT
};
#define DO_MIX(resampler) do { \
const ALfloat scaler = 1.0f/Channels; \
while(BufferSize--) \
{ \
for(i = 0;i < OUTPUTCHANNELS;i++) \
DrySend[i] += dryGainStep[i]; \
for(i = 0;i < MAX_SENDS;i++) \
WetSend[i] += wetGainStep[i]; \
\
for(i = 0;i < Channels;i++) \
{ \
value = (resampler)(Data[DataPosInt*Channels + i], \
Data[(DataPosInt+1)*Channels + i], \
DataPosFrac); \
\
Combine stereo and stereo-duplicate mixers Use a separate update method for stereo sources to the set appropriate channel gains
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outsamp = lpFilter2P(DryFilter, chans[i]*2, value); \
Move the core mixer functions to a separate source file
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DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]]; \
DryBuffer[j][chans2[i*2+0]] += outsamp*DrySend[chans2[i*2+0]]; \
DryBuffer[j][chans2[i*2+1]] += outsamp*DrySend[chans2[i*2+1]]; \
\
for(out = 0;out < MAX_SENDS;out++) \
{ \
outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
} \
} \
\
DataPosFrac += increment; \
DataPosInt += DataPosFrac>>FRACTIONBITS; \
DataPosFrac &= FRACTIONMASK; \
j++; \
} \
} while(0)
switch(Resampler)
{
case POINT_RESAMPLER:
DO_MIX(point); break;
case LINEAR_RESAMPLER:
DO_MIX(lerp); break;
case COSINE_RESAMPLER:
DO_MIX(cos_lerp); break;
case RESAMPLER_MIN:
case RESAMPLER_MAX:
break;
}
#undef DO_MIX
}
Combine stereo and stereo-duplicate mixers Use a separate update method for stereo sources to the set appropriate channel gains
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else if(Channels == 4) /* Quad */
Move the core mixer functions to a separate source file
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{
const int chans[] = {
Combine stereo and stereo-duplicate mixers Use a separate update method for stereo sources to the set appropriate channel gains
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FRONT_LEFT, FRONT_RIGHT,
BACK_LEFT, BACK_RIGHT
Move the core mixer functions to a separate source file
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};
#define DO_MIX(resampler) do { \
const ALfloat scaler = 1.0f/Channels; \
while(BufferSize--) \
{ \
for(i = 0;i < OUTPUTCHANNELS;i++) \
DrySend[i] += dryGainStep[i]; \
for(i = 0;i < MAX_SENDS;i++) \
WetSend[i] += wetGainStep[i]; \
\
for(i = 0;i < Channels;i++) \
{ \
value = (resampler)(Data[DataPosInt*Channels + i], \
Data[(DataPosInt+1)*Channels + i], \
DataPosFrac); \
\
outsamp = lpFilter2P(DryFilter, chans[i]*2, value); \
DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]]; \
\
for(out = 0;out < MAX_SENDS;out++) \
{ \
outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
} \
} \
\
DataPosFrac += increment; \
DataPosInt += DataPosFrac>>FRACTIONBITS; \
DataPosFrac &= FRACTIONMASK; \
j++; \
} \
} while(0)
switch(Resampler)
{
case POINT_RESAMPLER:
DO_MIX(point); break;
case LINEAR_RESAMPLER:
DO_MIX(lerp); break;
case COSINE_RESAMPLER:
DO_MIX(cos_lerp); break;
case RESAMPLER_MIN:
case RESAMPLER_MAX:
break;
}
}
else if(Channels == 6) /* 5.1 */
{
const int chans[] = {
FRONT_LEFT, FRONT_RIGHT,
FRONT_CENTER, LFE,
BACK_LEFT, BACK_RIGHT
};
switch(Resampler)
{
case POINT_RESAMPLER:
DO_MIX(point); break;
case LINEAR_RESAMPLER:
DO_MIX(lerp); break;
case COSINE_RESAMPLER:
DO_MIX(cos_lerp); break;
case RESAMPLER_MIN:
case RESAMPLER_MAX:
break;
}
}
else if(Channels == 7) /* 6.1 */
{
const int chans[] = {
FRONT_LEFT, FRONT_RIGHT,
FRONT_CENTER, LFE,
BACK_CENTER,
SIDE_LEFT, SIDE_RIGHT
};
switch(Resampler)
{
case POINT_RESAMPLER:
DO_MIX(point); break;
case LINEAR_RESAMPLER:
DO_MIX(lerp); break;
case COSINE_RESAMPLER:
DO_MIX(cos_lerp); break;
case RESAMPLER_MIN:
case RESAMPLER_MAX:
break;
}
}
else if(Channels == 8) /* 7.1 */
{
const int chans[] = {
FRONT_LEFT, FRONT_RIGHT,
FRONT_CENTER, LFE,
BACK_LEFT, BACK_RIGHT,
SIDE_LEFT, SIDE_RIGHT
};
switch(Resampler)
{
case POINT_RESAMPLER:
DO_MIX(point); break;
case LINEAR_RESAMPLER:
DO_MIX(lerp); break;
case COSINE_RESAMPLER:
DO_MIX(cos_lerp); break;
case RESAMPLER_MIN:
case RESAMPLER_MAX:
break;
}
#undef DO_MIX
}
else /* Unknown? */
{
for(i = 0;i < OUTPUTCHANNELS;i++)
DrySend[i] += dryGainStep[i]*BufferSize;
for(i = 0;i < MAX_SENDS;i++)
WetSend[i] += wetGainStep[i]*BufferSize;
while(BufferSize--)
{
DataPosFrac += increment;
DataPosInt += DataPosFrac>>FRACTIONBITS;
DataPosFrac &= FRACTIONMASK;
j++;
}
}
skipmix:
/* Handle looping sources */
if(DataPosInt >= LoopEnd)
{
if(BuffersPlayed < (ALSource->BuffersInQueue-1))
{
BufferListItem = BufferListItem->next;
BuffersPlayed++;
DataPosInt -= DataSize;
}
else if(Looping)
{
BufferListItem = ALSource->queue;
BuffersPlayed = 0;
if(ALSource->lSourceType == AL_STATIC)
DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
else
DataPosInt -= DataSize;
}
else
{
State = AL_STOPPED;
BufferListItem = ALSource->queue;
BuffersPlayed = ALSource->BuffersInQueue;
DataPosInt = 0;
DataPosFrac = 0;
}
}
} while(State == AL_PLAYING && j < SamplesToDo);
/* Update source info */
ALSource->state = State;
ALSource->BuffersPlayed = BuffersPlayed;
ALSource->position = DataPosInt;
ALSource->position_fraction = DataPosFrac;
ALSource->Buffer = BufferListItem->buffer;
for(i = 0;i < OUTPUTCHANNELS;i++)
ALSource->DryGains[i] = DrySend[i];
for(i = 0;i < MAX_SENDS;i++)
ALSource->WetGains[i] = WetSend[i];
ALSource->FirstStart = AL_FALSE;
}
ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
{
float (*DryBuffer)[OUTPUTCHANNELS];
ALfloat (*Matrix)[OUTPUTCHANNELS];
const ALuint *ChanMap;
ALuint SamplesToDo;
ALeffectslot *ALEffectSlot;
ALCcontext *ALContext;
ALfloat samp;
int fpuState;
ALuint i, j, c;
ALsizei e, s;
#if defined(HAVE_FESETROUND)
fpuState = fegetround();
fesetround(FE_TOWARDZERO);
#elif defined(HAVE__CONTROLFP)
fpuState = _controlfp(0, 0);
_controlfp(_RC_CHOP, _MCW_RC);
#else
(void)fpuState;
#endif
DryBuffer = device->DryBuffer;
while(size > 0)
{
/* Setup variables */
SamplesToDo = min(size, BUFFERSIZE);
/* Clear mixing buffer */
memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
SuspendContext(NULL);
for(c = 0;c < device->NumContexts;c++)
{
ALContext = device->Contexts[c];
SuspendContext(ALContext);
s = 0;
while(s < ALContext->ActiveSourceCount)
{
ALsource *Source = ALContext->ActiveSources[s];
if(Source->state != AL_PLAYING)
{
ALsizei end = --(ALContext->ActiveSourceCount);
ALContext->ActiveSources[s] = ALContext->ActiveSources[end];
continue;
}
MixSource(Source, ALContext, DryBuffer, SamplesToDo);
s++;
}
/* effect slot processing */
for(e = 0;e < ALContext->EffectSlotMap.size;e++)
{
ALEffectSlot = ALContext->EffectSlotMap.array[e].value;
Effectslot's Effectstate should never be NULL
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ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot, SamplesToDo, ALEffectSlot->WetBuffer, DryBuffer);
Move the core mixer functions to a separate source file
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for(i = 0;i < SamplesToDo;i++)
ALEffectSlot->WetBuffer[i] = 0.0f;
}
ProcessContext(ALContext);
}
device->SamplesPlayed += SamplesToDo;
ProcessContext(NULL);
//Post processing loop
ChanMap = device->DevChannels;
Matrix = device->ChannelMatrix;
switch(device->Format)
{
#define CHECK_WRITE_FORMAT(bits, type, func) \
case AL_FORMAT_MONO##bits: \
for(i = 0;i < SamplesToDo;i++) \
{ \
samp = 0.0f; \
for(c = 0;c < OUTPUTCHANNELS;c++) \
samp += DryBuffer[i][c] * Matrix[c][FRONT_CENTER]; \
((type*)buffer)[ChanMap[FRONT_CENTER]] = (func)(samp); \
buffer = ((type*)buffer) + 1; \
} \
break; \
case AL_FORMAT_STEREO##bits: \
if(device->Bs2b) \
{ \
for(i = 0;i < SamplesToDo;i++) \
{ \
float samples[2] = { 0.0f, 0.0f }; \
for(c = 0;c < OUTPUTCHANNELS;c++) \
{ \
samples[0] += DryBuffer[i][c]*Matrix[c][FRONT_LEFT]; \
samples[1] += DryBuffer[i][c]*Matrix[c][FRONT_RIGHT]; \
} \
bs2b_cross_feed(device->Bs2b, samples); \
((type*)buffer)[ChanMap[FRONT_LEFT]] = (func)(samples[0]);\
((type*)buffer)[ChanMap[FRONT_RIGHT]]= (func)(samples[1]);\
buffer = ((type*)buffer) + 2; \
} \
} \
else \
{ \
for(i = 0;i < SamplesToDo;i++) \
{ \
static const Channel chans[] = { \
FRONT_LEFT, FRONT_RIGHT \
}; \
for(j = 0;j < 2;j++) \
{ \
samp = 0.0f; \
for(c = 0;c < OUTPUTCHANNELS;c++) \
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
} \
buffer = ((type*)buffer) + 2; \
} \
} \
break; \
case AL_FORMAT_QUAD##bits: \
for(i = 0;i < SamplesToDo;i++) \
{ \
static const Channel chans[] = { \
FRONT_LEFT, FRONT_RIGHT, \
BACK_LEFT, BACK_RIGHT, \
}; \
for(j = 0;j < 4;j++) \
{ \
samp = 0.0f; \
for(c = 0;c < OUTPUTCHANNELS;c++) \
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
} \
buffer = ((type*)buffer) + 4; \
} \
break; \
case AL_FORMAT_51CHN##bits: \
for(i = 0;i < SamplesToDo;i++) \
{ \
static const Channel chans[] = { \
FRONT_LEFT, FRONT_RIGHT, \
FRONT_CENTER, LFE, \
BACK_LEFT, BACK_RIGHT, \
}; \
for(j = 0;j < 6;j++) \
{ \
samp = 0.0f; \
for(c = 0;c < OUTPUTCHANNELS;c++) \
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
} \
buffer = ((type*)buffer) + 6; \
} \
break; \
case AL_FORMAT_61CHN##bits: \
for(i = 0;i < SamplesToDo;i++) \
{ \
static const Channel chans[] = { \
FRONT_LEFT, FRONT_RIGHT, \
FRONT_CENTER, LFE, BACK_CENTER, \
SIDE_LEFT, SIDE_RIGHT, \
}; \
for(j = 0;j < 7;j++) \
{ \
samp = 0.0f; \
for(c = 0;c < OUTPUTCHANNELS;c++) \
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
} \
buffer = ((type*)buffer) + 7; \
} \
break; \
case AL_FORMAT_71CHN##bits: \
for(i = 0;i < SamplesToDo;i++) \
{ \
static const Channel chans[] = { \
FRONT_LEFT, FRONT_RIGHT, \
FRONT_CENTER, LFE, \
BACK_LEFT, BACK_RIGHT, \
SIDE_LEFT, SIDE_RIGHT \
}; \
for(j = 0;j < 8;j++) \
{ \
samp = 0.0f; \
for(c = 0;c < OUTPUTCHANNELS;c++) \
samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
} \
buffer = ((type*)buffer) + 8; \
} \
break;
#define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
#define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
CHECK_WRITE_FORMAT(8, ALubyte, aluF2UB)
CHECK_WRITE_FORMAT(16, ALshort, aluF2S)
CHECK_WRITE_FORMAT(32, ALfloat, aluF2F)
#undef AL_FORMAT_STEREO32
#undef AL_FORMAT_MONO32
#undef CHECK_WRITE_FORMAT
default:
break;
}
size -= SamplesToDo;
}
#if defined(HAVE_FESETROUND)
fesetround(fpuState);
#elif defined(HAVE__CONTROLFP)
_controlfp(fpuState, 0xfffff);
#endif
}
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