/**
* 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"
static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
"MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
extern inline void InitiatePositionArrays(ALuint frac, ALuint increment, ALuint *frac_arr, ALuint *pos_arr, ALuint size);
alignas(16) union ResamplerCoeffs ResampleCoeffs;
enum Resampler {
PointResampler,
LinearResampler,
FIR4Resampler,
FIR8Resampler,
BSincResampler,
ResamplerDefault = LinearResampler
};
/* FIR8 requires 3 extra samples before the current position, and 4 after. */
static_assert(MAX_PRE_SAMPLES >= 3, "MAX_PRE_SAMPLES must be at least 3!");
static_assert(MAX_POST_SAMPLES >= 4, "MAX_POST_SAMPLES must be at least 4!");
static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
static MixerFunc MixSamples = Mix_C;
static ResamplerFunc ResampleSamples = Resample_point32_C;
static inline HrtfMixerFunc SelectHrtfMixer(void)
{
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixHrtf_SSE;
#endif
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return MixHrtf_Neon;
#endif
return MixHrtf_C;
}
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 FIR4Resampler:
#ifdef HAVE_SSE4_1
if((CPUCapFlags&CPU_CAP_SSE4_1))
return Resample_fir4_32_SSE41;
#endif
#ifdef HAVE_SSE3
if((CPUCapFlags&CPU_CAP_SSE3))
return Resample_fir4_32_SSE3;
#endif
return Resample_fir4_32_C;
case FIR8Resampler:
#ifdef HAVE_SSE4_1
if((CPUCapFlags&CPU_CAP_SSE4_1))
return Resample_fir8_32_SSE41;
#endif
#ifdef HAVE_SSE3
if((CPUCapFlags&CPU_CAP_SSE3))
return Resample_fir8_32_SSE3;
#endif
return Resample_fir8_32_C;
case BSincResampler:
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return Resample_bsinc32_SSE;
#endif
return Resample_bsinc32_C;
}
return Resample_point32_C;
}
/* The sinc resampler makes use of a Kaiser window to limit the needed sample
* points to 4 and 8, respectively.
*/
#ifndef M_PI
#define M_PI (3.14159265358979323846)
#endif
static inline double Sinc(double x)
{
if(x == 0.0) return 1.0;
return sin(x*M_PI) / (x*M_PI);
}
/* The zero-order modified Bessel function of the first kind, used for the
* Kaiser window.
*
* I_0(x) = sum_{k=0}^inf (1 / k!)^2 (x / 2)^(2 k)
* = sum_{k=0}^inf ((x / 2)^k / k!)^2
*/
static double BesselI_0(double x)
{
double term, sum, x2, y, last_sum;
int k;
/* Start at k=1 since k=0 is trivial. */
term = 1.0;
sum = 1.0;
x2 = x / 2.0;
k = 1;
/* Let the integration converge until the term of the sum is no longer
* significant.
*/
do {
y = x2 / k;
k ++;
last_sum = sum;
term *= y * y;
sum += term;
} while(sum != last_sum);
return sum;
}
/* Calculate a Kaiser window from the given beta value and a normalized k
* [-1, 1].
*
* w(k) = { I_0(B sqrt(1 - k^2)) / I_0(B), -1 <= k <= 1
* { 0, elsewhere.
*
* Where k can be calculated as:
*
* k = i / l, where -l <= i <= l.
*
* or:
*
* k = 2 i / M - 1, where 0 <= i <= M.
*/
static inline double Kaiser(double b, double k)
{
if(k <= -1.0 || k >= 1.0) return 0.0;
return BesselI_0(b * sqrt(1.0 - (k*k))) / BesselI_0(b);
}
static inline double CalcKaiserBeta(double rejection)
{
if(rejection > 50.0)
return 0.1102 * (rejection - 8.7);
if(rejection >= 21.0)
return (0.5842 * pow(rejection - 21.0, 0.4)) +
(0.07886 * (rejection - 21.0));
return 0.0;
}
static float SincKaiser(double r, double x)
{
/* Limit rippling to -60dB. */
return (float)(Kaiser(CalcKaiserBeta(60.0), x / r) * Sinc(x));
}
void aluInitMixer(void)
{
enum Resampler resampler = ResamplerDefault;
const char *str;
ALuint i;
if(ConfigValueStr(NULL, NULL, "resampler", &str))
{
if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
resampler = PointResampler;
else if(strcasecmp(str, "linear") == 0)
resampler = LinearResampler;
else if(strcasecmp(str, "sinc4") == 0)
resampler = FIR4Resampler;
else if(strcasecmp(str, "sinc8") == 0)
resampler = FIR8Resampler;
else if(strcasecmp(str, "bsinc") == 0)
resampler = BSincResampler;
else if(strcasecmp(str, "cubic") == 0)
{
WARN("Resampler option \"cubic\" is deprecated, using sinc4\n");
resampler = FIR4Resampler;
}
else
{
char *end;
long n = strtol(str, &end, 0);
if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
resampler = n;
else
WARN("Invalid resampler: %s\n", str);
}
}
if(resampler == FIR8Resampler)
for(i = 0;i < FRACTIONONE;i++)
{
ALdouble mu = (ALdouble)i / FRACTIONONE;
ResampleCoeffs.FIR8[i][0] = SincKaiser(4.0, mu - -3.0);
ResampleCoeffs.FIR8[i][1] = SincKaiser(4.0, mu - -2.0);
ResampleCoeffs.FIR8[i][2] = SincKaiser(4.0, mu - -1.0);
ResampleCoeffs.FIR8[i][3] = SincKaiser(4.0, mu - 0.0);
ResampleCoeffs.FIR8[i][4] = SincKaiser(4.0, mu - 1.0);
ResampleCoeffs.FIR8[i][5] = SincKaiser(4.0, mu - 2.0);
ResampleCoeffs.FIR8[i][6] = SincKaiser(4.0, mu - 3.0);
ResampleCoeffs.FIR8[i][7] = SincKaiser(4.0, mu - 4.0);
}
else if(resampler == FIR4Resampler)
for(i = 0;i < FRACTIONONE;i++)
{
ALdouble mu = (ALdouble)i / FRACTIONONE;
ResampleCoeffs.FIR4[i][0] = SincKaiser(2.0, mu - -1.0);
ResampleCoeffs.FIR4[i][1] = SincKaiser(2.0, mu - 0.0);
ResampleCoeffs.FIR4[i][2] = SincKaiser(2.0, mu - 1.0);
ResampleCoeffs.FIR4[i][3] = SincKaiser(2.0, mu - 2.0);
}
MixHrtfSamples = SelectHrtfMixer();
MixSamples = SelectMixer();
ResampleSamples = SelectResampler(resampler);
}
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 inline 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 inline 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:
ALfilterState_processPassthru(lpfilter, src, numsamples);
ALfilterState_processPassthru(hpfilter, src, numsamples);
break;
case AF_LowPass:
ALfilterState_process(lpfilter, dst, src, numsamples);
ALfilterState_processPassthru(hpfilter, dst, numsamples);
return dst;
case AF_HighPass:
ALfilterState_processPassthru(lpfilter, src, numsamples);
ALfilterState_process(hpfilter, dst, src, numsamples);
return dst;
case AF_BandPass:
for(i = 0;i < numsamples;)
{
ALfloat temp[256];
ALuint todo = minu(256, 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)
{
ResamplerFunc Resample;
ALbufferlistitem *BufferListItem;
ALuint DataPosInt, DataPosFrac;
ALboolean Looping;
ALuint increment;
ALenum State;
ALuint OutPos;
ALuint NumChannels;
ALuint SampleSize;
ALint64 DataSize64;
ALuint IrSize;
ALuint chan, j;
/* Get source info */
State = AL_PLAYING; /* Only called while playing. */
BufferListItem = ATOMIC_LOAD(&Source->current_buffer);
DataPosInt = ATOMIC_LOAD(&Source->position, almemory_order_relaxed);
DataPosFrac = ATOMIC_LOAD(&Source->position_fraction, almemory_order_relaxed);
NumChannels = Source->NumChannels;
SampleSize = Source->SampleSize;
Looping = voice->Looping;
increment = voice->Step;
IrSize = (Device->Hrtf ? GetHrtfIrSize(Device->Hrtf) : 0);
Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
Resample_copy32_C : ResampleSamples);
OutPos = 0;
do {
ALuint SrcBufferSize, DstBufferSize;
ALuint Counter;
ALfloat Delta;
if(!voice->Moving)
{
Counter = 0;
Delta = 0.0f;
}
else
{
Counter = SamplesToDo - OutPos;
Delta = 1.0f / (ALfloat)Counter;
}
/* Figure out how many buffer samples will be needed */
DataSize64 = SamplesToDo-OutPos;
DataSize64 *= increment;
DataSize64 += DataPosFrac+FRACTIONMASK;
DataSize64 >>= FRACTIONBITS;
DataSize64 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE);
/* Figure out how many samples we can actually mix from this. */
DataSize64 = SrcBufferSize;
DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
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;
/* Load the previous samples into the source data first. */
memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat));
SrcDataSize = MAX_PRE_SAMPLES;
if(Source->SourceType == AL_STATIC)
{
const ALbuffer *ALBuffer = BufferListItem->buffer;
const ALubyte *Data = ALBuffer->data;
ALuint DataSize;
ALuint pos;
/* Offset buffer data to current channel */
Data += chan*SampleSize;
/* If current pos is beyond the loop range, do not loop */
if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd)
{
Looping = AL_FALSE;
/* Load what's left to play from the source buffer, and
* clear the rest of the temp buffer */
pos = DataPosInt;
DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos);
LoadSamples(&SrcData[SrcDataSize], &Data[pos * NumChannels*SampleSize],
NumChannels, ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
SrcDataSize += SrcBufferSize - SrcDataSize;
}
else
{
ALuint LoopStart = ALBuffer->LoopStart;
ALuint LoopEnd = ALBuffer->LoopEnd;
/* Load what's left of this loop iteration, then load
* repeats of the loop section */
pos = DataPosInt;
DataSize = LoopEnd - pos;
DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
LoadSamples(&SrcData[SrcDataSize], &Data[pos * NumChannels*SampleSize],
NumChannels, ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
DataSize = LoopEnd-LoopStart;
while(SrcBufferSize > SrcDataSize)
{
DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*SampleSize],
NumChannels, ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
}
}
}
else
{
/* Crawl the buffer queue to fill in the temp buffer */
ALbufferlistitem *tmpiter = BufferListItem;
ALuint pos = DataPosInt;
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;
}
}
}
/* Store the last source samples used for next time. */
memcpy(voice->PrevSamples[chan],
&SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
MAX_PRE_SAMPLES*sizeof(ALfloat)
);
/* Now resample, then filter and mix to the appropriate outputs. */
ResampledData = Resample(&voice->SincState,
&SrcData[MAX_PRE_SAMPLES], 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
);
if(!voice->IsHrtf)
{
ALfloat *restrict currents = parms->Gains[chan].Current;
const ALfloat *targets = parms->Gains[chan].Target;
MixGains gains[MAX_OUTPUT_CHANNELS];
if(!Counter)
{
for(j = 0;j < parms->OutChannels;j++)
{
gains[j].Target = targets[j];
gains[j].Current = gains[j].Target;
gains[j].Step = 0.0f;
}
}
else
{
for(j = 0;j < parms->OutChannels;j++)
{
ALfloat diff;
gains[j].Target = targets[j];
gains[j].Current = currents[j];
diff = gains[j].Target - gains[j].Current;
if(fabsf(diff) >= GAIN_SILENCE_THRESHOLD)
gains[j].Step = diff * Delta;
else
{
gains[j].Current = gains[j].Target;
gains[j].Step = 0.0f;
}
}
}
MixSamples(samples, parms->OutChannels, parms->OutBuffer, gains,
Counter, OutPos, DstBufferSize);
for(j = 0;j < parms->OutChannels;j++)
currents[j] = gains[j].Current;
}
else
{
MixHrtfParams hrtfparams;
int lidx, ridx;
if(!Counter)
{
parms->Hrtf[chan].Current = parms->Hrtf[chan].Target;
for(j = 0;j < HRIR_LENGTH;j++)
{
hrtfparams.Steps.Coeffs[j][0] = 0.0f;
hrtfparams.Steps.Coeffs[j][1] = 0.0f;
}
hrtfparams.Steps.Delay[0] = 0;
hrtfparams.Steps.Delay[1] = 0;
}
else
{
ALfloat coeffdiff;
ALint delaydiff;
for(j = 0;j < IrSize;j++)
{
coeffdiff = parms->Hrtf[chan].Target.Coeffs[j][0] - parms->Hrtf[chan].Current.Coeffs[j][0];
hrtfparams.Steps.Coeffs[j][0] = coeffdiff * Delta;
coeffdiff = parms->Hrtf[chan].Target.Coeffs[j][1] - parms->Hrtf[chan].Current.Coeffs[j][1];
hrtfparams.Steps.Coeffs[j][1] = coeffdiff * Delta;
}
delaydiff = (ALint)(parms->Hrtf[chan].Target.Delay[0] - parms->Hrtf[chan].Current.Delay[0]);
hrtfparams.Steps.Delay[0] = fastf2i((ALfloat)delaydiff * Delta);
delaydiff = (ALint)(parms->Hrtf[chan].Target.Delay[1] - parms->Hrtf[chan].Current.Delay[1]);
hrtfparams.Steps.Delay[1] = fastf2i((ALfloat)delaydiff * Delta);
}
hrtfparams.Target = &parms->Hrtf[chan].Target;
hrtfparams.Current = &parms->Hrtf[chan].Current;
lidx = GetChannelIdxByName(Device->RealOut, FrontLeft);
ridx = GetChannelIdxByName(Device->RealOut, FrontRight);
assert(lidx != -1 && ridx != -1);
MixHrtfSamples(parms->OutBuffer, lidx, ridx, samples, Counter, voice->Offset,
OutPos, IrSize, &hrtfparams, &parms->Hrtf[chan].State,
DstBufferSize);
}
}
for(j = 0;j < Device->NumAuxSends;j++)
{
SendParams *parms = &voice->Send[j];
ALfloat *restrict currents = parms->Gains[chan].Current;
const ALfloat *targets = parms->Gains[chan].Target;
MixGains gains[MAX_OUTPUT_CHANNELS];
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
);
if(!Counter)
{
for(j = 0;j < parms->OutChannels;j++)
{
gains[j].Target = targets[j];
gains[j].Current = gains[j].Target;
gains[j].Step = 0.0f;
}
}
else
{
for(j = 0;j < parms->OutChannels;j++)
{
ALfloat diff;
gains[j].Target = targets[j];
gains[j].Current = currents[j];
diff = gains[j].Target - gains[j].Current;
if(fabsf(diff) >= GAIN_SILENCE_THRESHOLD)
gains[j].Step = diff * Delta;
else
{
gains[j].Current = gains[j].Target;
gains[j].Step = 0.0f;
}
}
}
MixSamples(samples, parms->OutChannels, parms->OutBuffer, gains,
Counter, OutPos, DstBufferSize);
for(j = 0;j < parms->OutChannels;j++)
currents[j] = gains[j].Current;
}
}
/* Update positions */
DataPosFrac += increment*DstBufferSize;
DataPosInt += DataPosFrac>>FRACTIONBITS;
DataPosFrac &= FRACTIONMASK;
OutPos += DstBufferSize;
voice->Offset += 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);
voice->Moving = AL_TRUE;
/* Update source info */
Source->state = State;
ATOMIC_STORE(&Source->current_buffer, BufferListItem, almemory_order_relaxed);
ATOMIC_STORE(&Source->position, DataPosInt, almemory_order_relaxed);
ATOMIC_STORE(&Source->position_fraction, DataPosFrac);
}