510 lines
17 KiB
C
510 lines
17 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 "alMain.h"
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#include "AL/al.h"
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#include "AL/alc.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 "alu.h"
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#include "mixer_defs.h"
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extern inline void InitiatePositionArrays(ALuint frac, ALuint increment, ALuint *frac_arr, ALuint *pos_arr, ALuint size);
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static inline HrtfMixerFunc SelectHrtfMixer(void)
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{
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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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#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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return MixHrtf_C;
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}
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static inline MixerFunc SelectMixer(void)
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{
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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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#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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return Mix_C;
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}
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static inline ResamplerFunc SelectResampler(enum Resampler Resampler, ALuint increment)
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{
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if(increment == FRACTIONONE)
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return Resample_copy32_C;
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switch(Resampler)
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{
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case PointResampler:
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return Resample_point32_C;
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case LinearResampler:
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#ifdef HAVE_SSE4_1
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if((CPUCapFlags&CPU_CAP_SSE4_1))
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return Resample_lerp32_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_lerp32_SSE2;
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#endif
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return Resample_lerp32_C;
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case CubicResampler:
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return Resample_cubic32_C;
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case ResamplerMax:
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/* Shouldn't happen */
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break;
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}
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return Resample_point32_C;
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}
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static inline ALfloat Sample_ALbyte(ALbyte val)
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{ return val * (1.0f/127.0f); }
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static inline ALfloat Sample_ALshort(ALshort val)
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{ return val * (1.0f/32767.0f); }
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static inline ALfloat Sample_ALfloat(ALfloat val)
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{ return val; }
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#define DECL_TEMPLATE(T) \
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static void Load_##T(ALfloat *dst, const T *src, ALuint srcstep, ALuint samples)\
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{ \
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ALuint i; \
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for(i = 0;i < samples;i++) \
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dst[i] = Sample_##T(src[i*srcstep]); \
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}
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DECL_TEMPLATE(ALbyte)
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DECL_TEMPLATE(ALshort)
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DECL_TEMPLATE(ALfloat)
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#undef DECL_TEMPLATE
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static void LoadSamples(ALfloat *dst, const ALvoid *src, ALuint srcstep, enum FmtType srctype, ALuint samples)
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{
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switch(srctype)
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{
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case FmtByte:
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Load_ALbyte(dst, src, srcstep, samples);
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break;
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case FmtShort:
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Load_ALshort(dst, src, srcstep, samples);
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break;
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case FmtFloat:
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Load_ALfloat(dst, src, srcstep, samples);
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break;
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}
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}
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static void SilenceSamples(ALfloat *dst, ALuint samples)
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{
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ALuint i;
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for(i = 0;i < samples;i++)
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dst[i] = 0.0f;
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}
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static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter,
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ALfloat *restrict dst, const ALfloat *restrict src,
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ALuint numsamples, enum ActiveFilters type)
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{
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ALuint i;
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switch(type)
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{
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case AF_None:
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break;
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case AF_LowPass:
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ALfilterState_process(lpfilter, dst, src, numsamples);
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return dst;
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case AF_HighPass:
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ALfilterState_process(hpfilter, 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[64];
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ALuint todo = minu(64, numsamples-i);
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ALfilterState_process(lpfilter, temp, src+i, todo);
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ALfilterState_process(hpfilter, 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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ALvoid MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALuint SamplesToDo)
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{
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MixerFunc Mix;
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HrtfMixerFunc HrtfMix;
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ResamplerFunc Resample;
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ALbufferlistitem *BufferListItem;
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ALuint DataPosInt, DataPosFrac;
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ALboolean Looping;
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ALuint increment;
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enum Resampler Resampler;
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ALenum State;
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ALuint OutPos;
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ALuint NumChannels;
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ALuint SampleSize;
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ALint64 DataSize64;
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ALuint chan, j;
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/* Get source info */
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State = Source->state;
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BufferListItem = ATOMIC_LOAD(&Source->current_buffer);
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DataPosInt = Source->position;
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DataPosFrac = Source->position_fraction;
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Looping = Source->Looping;
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increment = voice->Step;
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Resampler = (increment==FRACTIONONE) ? PointResampler : Source->Resampler;
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NumChannels = Source->NumChannels;
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SampleSize = Source->SampleSize;
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Mix = SelectMixer();
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HrtfMix = SelectHrtfMixer();
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Resample = SelectResampler(Resampler, increment);
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OutPos = 0;
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do {
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const ALuint BufferPrePadding = ResamplerPrePadding[Resampler];
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const ALuint BufferPadding = ResamplerPadding[Resampler];
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ALuint SrcBufferSize, DstBufferSize;
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/* Figure out how many buffer samples will be needed */
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DataSize64 = SamplesToDo-OutPos;
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DataSize64 *= increment;
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DataSize64 += DataPosFrac+FRACTIONMASK;
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DataSize64 >>= FRACTIONBITS;
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DataSize64 += BufferPadding+BufferPrePadding;
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SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE);
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/* Figure out how many samples we can actually mix from this. */
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DataSize64 = SrcBufferSize;
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DataSize64 -= BufferPadding+BufferPrePadding;
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DataSize64 <<= FRACTIONBITS;
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DataSize64 -= DataPosFrac;
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DstBufferSize = (ALuint)((DataSize64+(increment-1)) / increment);
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DstBufferSize = minu(DstBufferSize, (SamplesToDo-OutPos));
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/* Some mixers like having a multiple of 4, so try to give that unless
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* this is the last update. */
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if(OutPos+DstBufferSize < SamplesToDo)
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DstBufferSize &= ~3;
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for(chan = 0;chan < NumChannels;chan++)
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{
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const ALfloat *ResampledData;
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ALfloat *SrcData = Device->SourceData;
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ALuint SrcDataSize = 0;
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if(Source->SourceType == AL_STATIC)
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{
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const ALbuffer *ALBuffer = BufferListItem->buffer;
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const ALubyte *Data = ALBuffer->data;
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ALuint DataSize;
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ALuint pos;
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/* If current pos is beyond the loop range, do not loop */
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if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd)
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{
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Looping = AL_FALSE;
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if(DataPosInt >= BufferPrePadding)
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pos = DataPosInt - BufferPrePadding;
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else
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{
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DataSize = BufferPrePadding - DataPosInt;
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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SilenceSamples(&SrcData[SrcDataSize], DataSize);
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SrcDataSize += DataSize;
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pos = 0;
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}
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/* Copy what's left to play in the source buffer, and clear the
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* rest of the temp buffer */
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DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos);
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LoadSamples(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
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NumChannels, ALBuffer->FmtType, DataSize);
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SrcDataSize += DataSize;
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SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
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SrcDataSize += SrcBufferSize - SrcDataSize;
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}
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else
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{
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ALuint LoopStart = ALBuffer->LoopStart;
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ALuint LoopEnd = ALBuffer->LoopEnd;
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if(DataPosInt >= LoopStart)
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{
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pos = DataPosInt-LoopStart;
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while(pos < BufferPrePadding)
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pos += LoopEnd-LoopStart;
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pos -= BufferPrePadding;
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pos += LoopStart;
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}
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else if(DataPosInt >= BufferPrePadding)
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pos = DataPosInt - BufferPrePadding;
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else
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{
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DataSize = BufferPrePadding - DataPosInt;
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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SilenceSamples(&SrcData[SrcDataSize], DataSize);
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SrcDataSize += DataSize;
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pos = 0;
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}
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/* Copy what's left of this loop iteration, then copy repeats
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* of the loop section */
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DataSize = LoopEnd - pos;
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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LoadSamples(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
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NumChannels, ALBuffer->FmtType, DataSize);
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SrcDataSize += DataSize;
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DataSize = LoopEnd-LoopStart;
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while(SrcBufferSize > SrcDataSize)
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{
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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LoadSamples(&SrcData[SrcDataSize], &Data[(LoopStart*NumChannels + chan)*SampleSize],
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NumChannels, ALBuffer->FmtType, DataSize);
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SrcDataSize += DataSize;
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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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ALuint pos;
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if(DataPosInt >= BufferPrePadding)
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pos = DataPosInt - BufferPrePadding;
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else
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{
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pos = BufferPrePadding - DataPosInt;
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while(pos > 0)
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{
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ALbufferlistitem *prev;
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if((prev=tmpiter->prev) != NULL)
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tmpiter = prev;
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else if(Looping)
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{
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while(tmpiter->next)
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tmpiter = tmpiter->next;
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}
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else
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{
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ALuint DataSize = minu(SrcBufferSize - SrcDataSize, pos);
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SilenceSamples(&SrcData[SrcDataSize], DataSize);
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SrcDataSize += DataSize;
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pos = 0;
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break;
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}
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if(tmpiter->buffer)
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{
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if((ALuint)tmpiter->buffer->SampleLen > pos)
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{
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pos = tmpiter->buffer->SampleLen - pos;
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break;
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}
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pos -= tmpiter->buffer->SampleLen;
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}
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}
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}
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while(tmpiter && SrcBufferSize > SrcDataSize)
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{
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const ALbuffer *ALBuffer;
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if((ALBuffer=tmpiter->buffer) != NULL)
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{
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const ALubyte *Data = ALBuffer->data;
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ALuint DataSize = ALBuffer->SampleLen;
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/* Skip the data already played */
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if(DataSize <= pos)
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pos -= DataSize;
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else
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{
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Data += (pos*NumChannels + chan)*SampleSize;
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DataSize -= pos;
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pos -= pos;
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DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
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LoadSamples(&SrcData[SrcDataSize], Data, NumChannels,
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ALBuffer->FmtType, DataSize);
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SrcDataSize += DataSize;
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}
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}
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tmpiter = tmpiter->next;
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if(!tmpiter && Looping)
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tmpiter = ATOMIC_LOAD(&Source->queue);
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else if(!tmpiter)
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{
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SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
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SrcDataSize += SrcBufferSize - SrcDataSize;
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}
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}
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}
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/* Now resample, then filter and mix to the appropriate outputs. */
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ResampledData = Resample(
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&SrcData[BufferPrePadding], DataPosFrac, increment,
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Device->ResampledData, DstBufferSize
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);
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{
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DirectParams *parms = &voice->Direct;
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const ALfloat *samples;
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samples = DoFilters(
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&parms->Filters[chan].LowPass, &parms->Filters[chan].HighPass,
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Device->FilteredData, ResampledData, DstBufferSize,
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parms->Filters[chan].ActiveType
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);
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if(!voice->IsHrtf)
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Mix(samples, MaxChannels, parms->OutBuffer, parms->Mix.Gains[chan],
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parms->Counter, OutPos, DstBufferSize);
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else
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HrtfMix(parms->OutBuffer, samples, parms->Counter, voice->Offset,
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OutPos, parms->Mix.Hrtf.IrSize, &parms->Mix.Hrtf.Params[chan],
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&parms->Mix.Hrtf.State[chan], DstBufferSize);
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}
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for(j = 0;j < Device->NumAuxSends;j++)
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{
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SendParams *parms = &voice->Send[j];
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const ALfloat *samples;
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if(!parms->OutBuffer)
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continue;
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samples = DoFilters(
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&parms->Filters[chan].LowPass, &parms->Filters[chan].HighPass,
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Device->FilteredData, ResampledData, DstBufferSize,
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parms->Filters[chan].ActiveType
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);
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Mix(samples, 1, parms->OutBuffer, &parms->Gain,
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parms->Counter, OutPos, DstBufferSize);
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}
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}
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/* Update positions */
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DataPosFrac += increment*DstBufferSize;
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DataPosInt += DataPosFrac>>FRACTIONBITS;
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DataPosFrac &= FRACTIONMASK;
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OutPos += DstBufferSize;
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voice->Offset += DstBufferSize;
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voice->Direct.Counter = maxu(voice->Direct.Counter, DstBufferSize) - DstBufferSize;
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for(j = 0;j < Device->NumAuxSends;j++)
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voice->Send[j].Counter = maxu(voice->Send[j].Counter, DstBufferSize) - DstBufferSize;
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/* Handle looping sources */
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while(1)
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{
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const ALbuffer *ALBuffer;
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ALuint DataSize = 0;
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ALuint LoopStart = 0;
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ALuint LoopEnd = 0;
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if((ALBuffer=BufferListItem->buffer) != NULL)
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{
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DataSize = ALBuffer->SampleLen;
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LoopStart = ALBuffer->LoopStart;
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LoopEnd = ALBuffer->LoopEnd;
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if(LoopEnd > DataPosInt)
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break;
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}
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if(Looping && Source->SourceType == AL_STATIC)
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{
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assert(LoopEnd > LoopStart);
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DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
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break;
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}
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if(DataSize > DataPosInt)
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break;
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if(!(BufferListItem=BufferListItem->next))
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{
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if(Looping)
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BufferListItem = ATOMIC_LOAD(&Source->queue);
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else
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{
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State = AL_STOPPED;
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BufferListItem = NULL;
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DataPosInt = 0;
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DataPosFrac = 0;
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break;
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}
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}
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DataPosInt -= DataSize;
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}
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} while(State == AL_PLAYING && OutPos < SamplesToDo);
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/* Update source info */
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Source->state = State;
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ATOMIC_STORE(&Source->current_buffer, BufferListItem);
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Source->position = DataPosInt;
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Source->position_fraction = DataPosFrac;
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}
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