Merge branch 'v1.19' of git://repo.or.cz/openal-soft into v1.19v1.19

1 files changed, 0 insertions, 638 deletions

diff --git a/Alc/mixer.c b/Alc/mixer.c
deleted file mode 100644
index 712075f1..00000000
--- a/Alc/mixer.c
+++ /dev/null
@@ -1,638 +0,0 @@
-/**
- * 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          = Source->state;
-    BufferListItem = ATOMIC_LOAD(&Source->current_buffer);
-    DataPosInt     = Source->position;
-    DataPosFrac    = Source->position_fraction;
-    Looping        = Source->Looping;
-    NumChannels    = Source->NumChannels;
-    SampleSize     = Source->SampleSize;
-    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;
-
-        /* 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)
-                    MixSamples(samples, parms->OutChannels, parms->OutBuffer, parms->Gains[chan],
-                               parms->Counter, OutPos, DstBufferSize);
-                else
-                    MixHrtfSamples(parms->OutBuffer, samples, parms->Counter, voice->Offset,
-                                   OutPos, IrSize, &parms->Hrtf[chan].Params,
-                                   &parms->Hrtf[chan].State, DstBufferSize);
-            }
-
-            for(j = 0;j < Device->NumAuxSends;j++)
-            {
-                SendParams *parms = &voice->Send[j];
-                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
-                );
-                MixSamples(samples, 1, parms->OutBuffer, &parms->Gains[chan],
-                           parms->Counter, OutPos, DstBufferSize);
-            }
-        }
-        /* Update positions */
-        DataPosFrac += increment*DstBufferSize;
-        DataPosInt  += DataPosFrac>>FRACTIONBITS;
-        DataPosFrac &= FRACTIONMASK;
-
-        OutPos += DstBufferSize;
-        voice->Offset += DstBufferSize;
-        voice->Direct.Counter = maxu(voice->Direct.Counter, DstBufferSize) - DstBufferSize;
-        for(j = 0;j < Device->NumAuxSends;j++)
-            voice->Send[j].Counter = maxu(voice->Send[j].Counter, DstBufferSize) - 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);
-
-    /* Update source info */
-    Source->state             = State;
-    ATOMIC_STORE(&Source->current_buffer, BufferListItem);
-    Source->position          = DataPosInt;
-    Source->position_fraction = DataPosFrac;
-}