734 lines
28 KiB
C++
734 lines
28 KiB
C++
/**
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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.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 "AL/al.h"
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#include "AL/alc.h"
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#include "alMain.h"
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#include "alcontext.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 "sample_cvt.h"
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#include "alu.h"
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#include "alconfig.h"
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#include "ringbuffer.h"
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#include "cpu_caps.h"
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#include "mixer/defs.h"
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static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
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"MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
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/* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */
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static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!");
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enum Resampler ResamplerDefault = LinearResampler;
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MixerFunc MixSamples = Mix_C;
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RowMixerFunc MixRowSamples = MixRow_C;
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static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
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static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
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static MixerFunc SelectMixer(void)
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{
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Mix_Neon;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return Mix_SSE;
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#endif
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return Mix_C;
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}
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static RowMixerFunc SelectRowMixer(void)
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{
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return MixRow_Neon;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return MixRow_SSE;
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#endif
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return MixRow_C;
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}
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static inline HrtfMixerFunc SelectHrtfMixer(void)
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{
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return MixHrtf_Neon;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return MixHrtf_SSE;
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#endif
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return MixHrtf_C;
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}
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static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
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{
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return MixHrtfBlend_Neon;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return MixHrtfBlend_SSE;
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#endif
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return MixHrtfBlend_C;
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}
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ResamplerFunc SelectResampler(enum Resampler resampler)
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{
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switch(resampler)
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{
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case PointResampler:
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return Resample_point_C;
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case LinearResampler:
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Resample_lerp_Neon;
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#endif
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#ifdef HAVE_SSE4_1
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if((CPUCapFlags&CPU_CAP_SSE4_1))
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return Resample_lerp_SSE41;
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#endif
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#ifdef HAVE_SSE2
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if((CPUCapFlags&CPU_CAP_SSE2))
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return Resample_lerp_SSE2;
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#endif
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return Resample_lerp_C;
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case FIR4Resampler:
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return Resample_cubic_C;
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case BSinc12Resampler:
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case BSinc24Resampler:
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Resample_bsinc_Neon;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return Resample_bsinc_SSE;
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#endif
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return Resample_bsinc_C;
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}
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return Resample_point_C;
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}
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void aluInitMixer(void)
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{
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const char *str;
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if(ConfigValueStr(NULL, NULL, "resampler", &str))
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{
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if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
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ResamplerDefault = PointResampler;
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else if(strcasecmp(str, "linear") == 0)
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ResamplerDefault = LinearResampler;
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else if(strcasecmp(str, "cubic") == 0)
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ResamplerDefault = FIR4Resampler;
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else if(strcasecmp(str, "bsinc12") == 0)
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ResamplerDefault = BSinc12Resampler;
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else if(strcasecmp(str, "bsinc24") == 0)
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ResamplerDefault = BSinc24Resampler;
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else if(strcasecmp(str, "bsinc") == 0)
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{
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WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
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ResamplerDefault = BSinc12Resampler;
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}
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else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0)
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{
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WARN("Resampler option \"%s\" is deprecated, using cubic\n", str);
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ResamplerDefault = FIR4Resampler;
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}
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else
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{
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char *end;
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long n = strtol(str, &end, 0);
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if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
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ResamplerDefault = static_cast<enum Resampler>(n);
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else
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WARN("Invalid resampler: %s\n", str);
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}
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}
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MixHrtfBlendSamples = SelectHrtfBlendMixer();
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MixHrtfSamples = SelectHrtfMixer();
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MixSamples = SelectMixer();
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MixRowSamples = SelectRowMixer();
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}
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namespace {
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/* Base template left undefined. Should be marked =delete, but Clang 3.8.1
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* chokes on that given the inline specializations.
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*/
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template<FmtType T>
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inline ALfloat LoadSample(typename FmtTypeTraits<T>::Type val);
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template<> inline ALfloat LoadSample<FmtUByte>(FmtTypeTraits<FmtUByte>::Type val)
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{ return (val-128) * (1.0f/128.0f); }
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template<> inline ALfloat LoadSample<FmtShort>(FmtTypeTraits<FmtShort>::Type val)
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{ return val * (1.0f/32768.0f); }
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template<> inline ALfloat LoadSample<FmtFloat>(FmtTypeTraits<FmtFloat>::Type val)
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{ return val; }
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template<> inline ALfloat LoadSample<FmtDouble>(FmtTypeTraits<FmtDouble>::Type val)
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{ return (ALfloat)val; }
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template<> inline ALfloat LoadSample<FmtMulaw>(FmtTypeTraits<FmtMulaw>::Type val)
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{ return muLawDecompressionTable[val] * (1.0f/32768.0f); }
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template<> inline ALfloat LoadSample<FmtAlaw>(FmtTypeTraits<FmtAlaw>::Type val)
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{ return aLawDecompressionTable[val] * (1.0f/32768.0f); }
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template<FmtType T>
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inline void LoadSampleArray(ALfloat *RESTRICT dst, const void *src, ALint srcstep, ALsizei samples)
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{
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using SampleType = typename FmtTypeTraits<T>::Type;
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const SampleType *ssrc = static_cast<const SampleType*>(src);
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for(ALsizei i{0};i < samples;i++)
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dst[i] += LoadSample<T>(ssrc[i*srcstep]);
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}
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void LoadSamples(ALfloat *RESTRICT dst, const ALvoid *RESTRICT src, ALint srcstep, FmtType srctype,
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ALsizei samples)
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{
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#define HANDLE_FMT(T) \
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case T: LoadSampleArray<T>(dst, src, srcstep, samples); break
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switch(srctype)
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{
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HANDLE_FMT(FmtUByte);
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HANDLE_FMT(FmtShort);
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HANDLE_FMT(FmtFloat);
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HANDLE_FMT(FmtDouble);
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HANDLE_FMT(FmtMulaw);
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HANDLE_FMT(FmtAlaw);
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}
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#undef HANDLE_FMT
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}
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const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter,
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ALfloat *RESTRICT dst, const ALfloat *RESTRICT src,
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ALsizei numsamples, int type)
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{
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ALsizei i;
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switch(type)
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{
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case AF_None:
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lpfilter->passthru(numsamples);
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hpfilter->passthru(numsamples);
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break;
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case AF_LowPass:
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lpfilter->process(dst, src, numsamples);
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hpfilter->passthru(numsamples);
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return dst;
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case AF_HighPass:
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lpfilter->passthru(numsamples);
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hpfilter->process(dst, src, numsamples);
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return dst;
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case AF_BandPass:
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for(i = 0;i < numsamples;)
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{
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ALfloat temp[256];
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ALsizei todo = mini(256, numsamples-i);
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lpfilter->process(temp, src+i, todo);
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hpfilter->process(dst+i, temp, todo);
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i += todo;
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}
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return dst;
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}
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return src;
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}
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} // namespace
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/* This function uses these device temp buffers. */
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#define SOURCE_DATA_BUF 0
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#define RESAMPLED_BUF 1
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#define FILTERED_BUF 2
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#define NFC_DATA_BUF 3
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ALboolean MixSource(ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo)
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{
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ASSUME(SamplesToDo > 0);
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/* Get source info */
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bool isplaying{true}; /* Will only be called while playing. */
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bool isstatic{(voice->Flags&VOICE_IS_STATIC) != 0};
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ALsizei DataPosInt{(ALsizei)voice->position.load(std::memory_order_acquire)};
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ALsizei DataPosFrac{voice->position_fraction.load(std::memory_order_relaxed)};
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ALbufferlistitem *BufferListItem{voice->current_buffer.load(std::memory_order_relaxed)};
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ALbufferlistitem *BufferLoopItem{voice->loop_buffer.load(std::memory_order_relaxed)};
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ALsizei NumChannels{voice->NumChannels};
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ALsizei SampleSize{voice->SampleSize};
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ALint increment{voice->Step};
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ASSUME(DataPosInt >= 0);
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ASSUME(DataPosFrac >= 0);
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ASSUME(NumChannels > 0);
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ASSUME(SampleSize > 0);
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ASSUME(increment > 0);
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ALCdevice *Device{Context->Device};
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ALsizei IrSize{Device->HrtfHandle ? Device->HrtfHandle->irSize : 0};
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ResamplerFunc Resample{(increment == FRACTIONONE && DataPosFrac == 0) ?
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Resample_copy_C : voice->Resampler};
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ALsizei Counter{(voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0};
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ALsizei buffers_done{0};
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ALsizei OutPos{0};
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do {
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/* Figure out how many buffer samples will be needed */
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ALsizei DstBufferSize{SamplesToDo - OutPos};
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/* Calculate the last written dst sample pos. */
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ALint64 DataSize64{DstBufferSize - 1};
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/* Calculate the last read src sample pos. */
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DataSize64 = (DataSize64*increment + DataPosFrac) >> FRACTIONBITS;
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/* +1 to get the src sample count, include padding. */
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DataSize64 += 1 + MAX_RESAMPLE_PADDING*2;
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auto SrcBufferSize = static_cast<ALsizei>(mini64(DataSize64, BUFFERSIZE+1));
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if(SrcBufferSize > BUFFERSIZE)
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{
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SrcBufferSize = BUFFERSIZE;
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/* If the source buffer got saturated, we can't fill the desired
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* dst size. Figure out how many samples we can actually mix from
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* this.
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*/
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DataSize64 = SrcBufferSize - MAX_RESAMPLE_PADDING*2;
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DataSize64 = ((DataSize64<<FRACTIONBITS) - DataPosFrac + increment-1) / increment;
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DstBufferSize = static_cast<ALsizei>(mini64(DataSize64, DstBufferSize));
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/* Some mixers like having a multiple of 4, so try to give that
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* unless this is the last update.
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*/
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if(DstBufferSize < SamplesToDo-OutPos)
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DstBufferSize &= ~3;
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}
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/* It's impossible to have a buffer list item with no entries. */
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assert(BufferListItem->num_buffers > 0);
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for(ALsizei chan{0};chan < NumChannels;chan++)
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{
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ALfloat (&SrcData)[BUFFERSIZE] = Device->TempBuffer[SOURCE_DATA_BUF];
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/* Load the previous samples into the source data first, and clear the rest. */
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auto srciter = std::copy(std::begin(voice->PrevSamples[chan]),
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std::end(voice->PrevSamples[chan]), std::begin(SrcData));
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std::fill(srciter, std::end(SrcData), 0.0f);
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auto FilledAmt = static_cast<ALsizei>(voice->PrevSamples[chan].size());
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if(isstatic)
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{
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/* TODO: For static sources, loop points are taken from the
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* first buffer (should be adjusted by any buffer offset, to
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* possibly be added later).
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*/
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const ALbuffer *Buffer0{BufferListItem->buffers[0]};
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const ALsizei LoopStart{Buffer0->LoopStart};
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const ALsizei LoopEnd{Buffer0->LoopEnd};
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ASSUME(LoopStart >= 0);
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ASSUME(LoopEnd > LoopStart);
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/* If current pos is beyond the loop range, do not loop */
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if(!BufferLoopItem || DataPosInt >= LoopEnd)
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{
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ALsizei SizeToDo = SrcBufferSize - FilledAmt;
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BufferLoopItem = nullptr;
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ALsizei CompLen{0};
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auto load_buffer = [DataPosInt,&SrcData,NumChannels,SampleSize,chan,FilledAmt,SizeToDo,&CompLen](const ALbuffer *buffer) -> void
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{
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if(DataPosInt >= buffer->SampleLen)
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return;
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/* Load what's left to play from the buffer */
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ALsizei DataSize{mini(SizeToDo, buffer->SampleLen - DataPosInt)};
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CompLen = maxi(CompLen, DataSize);
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const ALbyte *Data{buffer->mData.data()};
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LoadSamples(&SrcData[FilledAmt],
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&Data[(DataPosInt*NumChannels + chan)*SampleSize],
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NumChannels, buffer->FmtType, DataSize
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);
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};
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auto buffers_end = BufferListItem->buffers + BufferListItem->num_buffers;
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std::for_each(BufferListItem->buffers, buffers_end, load_buffer);
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FilledAmt += CompLen;
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}
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else
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{
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const ALsizei SizeToDo{mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt)};
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ALsizei CompLen{0};
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auto load_buffer = [DataPosInt,&SrcData,NumChannels,SampleSize,chan,FilledAmt,SizeToDo,&CompLen](const ALbuffer *buffer) -> void
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{
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if(DataPosInt >= buffer->SampleLen)
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return;
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/* Load what's left of this loop iteration */
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ALsizei DataSize{mini(SizeToDo, buffer->SampleLen - DataPosInt)};
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CompLen = maxi(CompLen, DataSize);
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const ALbyte *Data{buffer->mData.data()};
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LoadSamples(&SrcData[FilledAmt],
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&Data[(DataPosInt*NumChannels + chan)*SampleSize],
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NumChannels, buffer->FmtType, DataSize
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);
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};
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auto buffers_end = BufferListItem->buffers + BufferListItem->num_buffers;
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std::for_each(BufferListItem->buffers, buffers_end, load_buffer);
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FilledAmt += CompLen;
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const ALsizei LoopSize{LoopEnd - LoopStart};
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while(SrcBufferSize > FilledAmt)
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{
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const ALsizei SizeToDo{mini(SrcBufferSize - FilledAmt, LoopSize)};
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CompLen = 0;
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auto load_buffer_loop = [LoopStart,&SrcData,NumChannels,SampleSize,chan,FilledAmt,SizeToDo,&CompLen](const ALbuffer *buffer) -> void
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{
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const ALbyte *Data = buffer->mData.data();
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ALsizei DataSize;
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if(LoopStart >= buffer->SampleLen)
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return;
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DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart);
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CompLen = maxi(CompLen, DataSize);
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LoadSamples(&SrcData[FilledAmt],
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&Data[(LoopStart*NumChannels + chan)*SampleSize],
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NumChannels, buffer->FmtType, DataSize
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);
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};
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std::for_each(BufferListItem->buffers, buffers_end, load_buffer_loop);
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FilledAmt += CompLen;
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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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ALsizei pos{DataPosInt};
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while(tmpiter && SrcBufferSize > FilledAmt)
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{
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if(pos >= tmpiter->max_samples)
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{
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pos -= tmpiter->max_samples;
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tmpiter = tmpiter->next.load(std::memory_order_acquire);
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if(!tmpiter) tmpiter = BufferLoopItem;
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continue;
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}
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const ALsizei SizeToDo{SrcBufferSize - FilledAmt};
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ALsizei CompLen{0};
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auto load_buffer = [pos,&SrcData,NumChannels,SampleSize,chan,FilledAmt,SizeToDo,&CompLen](const ALbuffer *buffer) -> void
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{
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if(!buffer) return;
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ALsizei DataSize{buffer->SampleLen};
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if(pos >= DataSize) return;
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DataSize = mini(SizeToDo, DataSize - pos);
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CompLen = maxi(CompLen, DataSize);
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const ALbyte *Data{buffer->mData.data()};
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Data += (pos*NumChannels + chan)*SampleSize;
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LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
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buffer->FmtType, DataSize);
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};
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auto buffers_end = tmpiter->buffers + tmpiter->num_buffers;
|
|
std::for_each(tmpiter->buffers, buffers_end, load_buffer);
|
|
FilledAmt += CompLen;
|
|
|
|
if(SrcBufferSize <= FilledAmt)
|
|
break;
|
|
pos = 0;
|
|
tmpiter = tmpiter->next.load(std::memory_order_acquire);
|
|
if(!tmpiter) tmpiter = BufferLoopItem;
|
|
}
|
|
}
|
|
|
|
/* Store the last source samples used for next time. */
|
|
std::copy_n(&SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
|
|
voice->PrevSamples[chan].size(), std::begin(voice->PrevSamples[chan]));
|
|
|
|
/* Now resample, then filter and mix to the appropriate outputs. */
|
|
const ALfloat *ResampledData{Resample(&voice->ResampleState,
|
|
&SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
|
|
Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
|
|
)};
|
|
{
|
|
DirectParams *parms{&voice->Direct.Params[chan]};
|
|
const ALfloat *samples{DoFilters(&parms->LowPass, &parms->HighPass,
|
|
Device->TempBuffer[FILTERED_BUF], ResampledData, DstBufferSize,
|
|
voice->Direct.FilterType
|
|
)};
|
|
|
|
if(!(voice->Flags&VOICE_HAS_HRTF))
|
|
{
|
|
if(!Counter)
|
|
std::copy(std::begin(parms->Gains.Target), std::end(parms->Gains.Target),
|
|
std::begin(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
|
|
{
|
|
MixSamples(samples,
|
|
voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
|
|
parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
|
|
DstBufferSize
|
|
);
|
|
|
|
ALfloat *nfcsamples{Device->TempBuffer[NFC_DATA_BUF]};
|
|
ALsizei chanoffset{voice->Direct.ChannelsPerOrder[0]};
|
|
using FilterProc = void (NfcFilter::*)(float*,const float*,int);
|
|
auto apply_nfc = [voice,parms,samples,DstBufferSize,Counter,OutPos,&chanoffset,nfcsamples](FilterProc process, ALsizei order) -> void
|
|
{
|
|
if(voice->Direct.ChannelsPerOrder[order] < 1)
|
|
return;
|
|
(parms->NFCtrlFilter.*process)(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(&NfcFilter::process1, 1);
|
|
apply_nfc(&NfcFilter::process2, 2);
|
|
apply_nfc(&NfcFilter::process3, 3);
|
|
}
|
|
}
|
|
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(OutPos == 0)
|
|
{
|
|
/* First mixing pass, fade between the coefficients. */
|
|
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.
|
|
*/
|
|
ALfloat 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;
|
|
}
|
|
}
|
|
}
|
|
|
|
ALfloat (&FilterBuf)[BUFFERSIZE] = Device->TempBuffer[FILTERED_BUF];
|
|
auto mix_send = [Counter,OutPos,DstBufferSize,chan,ResampledData,&FilterBuf](ALvoice::SendData &send) -> void
|
|
{
|
|
if(!send.Buffer)
|
|
return;
|
|
|
|
SendParams *parms = &send.Params[chan];
|
|
const ALfloat *samples{DoFilters(&parms->LowPass, &parms->HighPass,
|
|
FilterBuf, ResampledData, DstBufferSize, send.FilterType
|
|
)};
|
|
|
|
if(!Counter)
|
|
std::copy(std::begin(parms->Gains.Target), std::end(parms->Gains.Target),
|
|
std::begin(parms->Gains.Current));
|
|
MixSamples(samples, send.Channels, send.Buffer,
|
|
parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
|
|
);
|
|
};
|
|
std::for_each(voice->Send, voice->Send+Device->NumAuxSends, mix_send);
|
|
}
|
|
/* Update positions */
|
|
DataPosFrac += increment*DstBufferSize;
|
|
DataPosInt += DataPosFrac>>FRACTIONBITS;
|
|
DataPosFrac &= FRACTIONMASK;
|
|
|
|
OutPos += DstBufferSize;
|
|
voice->Offset += DstBufferSize;
|
|
Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
|
|
|
|
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 = BufferListItem->next.load(std::memory_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 */
|
|
voice->position.store(DataPosInt, std::memory_order_relaxed);
|
|
voice->position_fraction.store(DataPosFrac, std::memory_order_relaxed);
|
|
voice->current_buffer.store(BufferListItem, std::memory_order_release);
|
|
|
|
/* Send any events now, after the position/buffer info was updated. */
|
|
ALbitfieldSOFT enabledevt{Context->EnabledEvts.load(std::memory_order_acquire)};
|
|
if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
|
|
{
|
|
AsyncEvent evt{EventType_BufferCompleted};
|
|
evt.u.bufcomp.id = SourceID;
|
|
evt.u.bufcomp.count = buffers_done;
|
|
if(ll_ringbuffer_write(Context->AsyncEvents, &evt, 1) == 1)
|
|
Context->EventSem.post();
|
|
}
|
|
|
|
return isplaying;
|
|
}
|