756 lines
27 KiB
C
756 lines
27 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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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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extern inline void InitiatePositionArrays(ALuint frac, ALint increment, ALuint *restrict frac_arr, ALint *restrict pos_arr, ALsizei size);
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alignas(16) ALfloat ResampleCoeffs_FIR4[FRACTIONONE][4];
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enum Resampler {
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PointResampler,
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LinearResampler,
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FIR4Resampler,
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BSincResampler,
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ResamplerDefault = LinearResampler
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};
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/* BSinc requires up to 11 extra samples before the current position, and 12 after. */
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static_assert(MAX_PRE_SAMPLES >= 11, "MAX_PRE_SAMPLES must be at least 11!");
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static_assert(MAX_POST_SAMPLES >= 12, "MAX_POST_SAMPLES must be at least 12!");
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static MixerFunc MixSamples = Mix_C;
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static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
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static ResamplerFunc ResampleSamples = Resample_point32_C;
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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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RowMixerFunc SelectRowMixer(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 MixRow_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 MixRow_Neon;
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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 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_point32_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_lerp32_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_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 FIR4Resampler:
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Resample_fir4_32_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_fir4_32_SSE41;
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#endif
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#ifdef HAVE_SSE3
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if((CPUCapFlags&CPU_CAP_SSE3))
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return Resample_fir4_32_SSE3;
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#endif
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return Resample_fir4_32_C;
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case BSincResampler:
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Resample_bsinc32_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_bsinc32_SSE;
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#endif
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return Resample_bsinc32_C;
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}
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return Resample_point32_C;
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}
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/* The sinc resampler makes use of a Kaiser window to limit the needed sample
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* points to 4 and 8, respectively.
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*/
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#ifndef M_PI
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#define M_PI (3.14159265358979323846)
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#endif
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static inline double Sinc(double x)
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{
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if(x == 0.0) return 1.0;
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return sin(x*M_PI) / (x*M_PI);
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}
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/* The zero-order modified Bessel function of the first kind, used for the
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* Kaiser window.
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*
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* I_0(x) = sum_{k=0}^inf (1 / k!)^2 (x / 2)^(2 k)
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* = sum_{k=0}^inf ((x / 2)^k / k!)^2
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*/
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static double BesselI_0(double x)
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{
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double term, sum, x2, y, last_sum;
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int k;
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/* Start at k=1 since k=0 is trivial. */
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term = 1.0;
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sum = 1.0;
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x2 = x / 2.0;
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k = 1;
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/* Let the integration converge until the term of the sum is no longer
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* significant.
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*/
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do {
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y = x2 / k;
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k ++;
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last_sum = sum;
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term *= y * y;
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sum += term;
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} while(sum != last_sum);
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return sum;
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}
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/* Calculate a Kaiser window from the given beta value and a normalized k
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* [-1, 1].
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*
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* w(k) = { I_0(B sqrt(1 - k^2)) / I_0(B), -1 <= k <= 1
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* { 0, elsewhere.
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*
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* Where k can be calculated as:
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*
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* k = i / l, where -l <= i <= l.
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*
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* or:
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*
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* k = 2 i / M - 1, where 0 <= i <= M.
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*/
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static inline double Kaiser(double b, double k)
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{
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if(k <= -1.0 || k >= 1.0) return 0.0;
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return BesselI_0(b * sqrt(1.0 - (k*k))) / BesselI_0(b);
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}
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static inline double CalcKaiserBeta(double rejection)
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{
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if(rejection > 50.0)
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return 0.1102 * (rejection - 8.7);
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if(rejection >= 21.0)
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return (0.5842 * pow(rejection - 21.0, 0.4)) +
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(0.07886 * (rejection - 21.0));
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return 0.0;
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}
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static float SincKaiser(double r, double x)
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{
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/* Limit rippling to -60dB. */
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return (float)(Kaiser(CalcKaiserBeta(60.0), x / r) * Sinc(x));
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}
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void aluInitMixer(void)
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{
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enum Resampler resampler = ResamplerDefault;
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const char *str;
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ALuint i;
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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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resampler = PointResampler;
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else if(strcasecmp(str, "linear") == 0)
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resampler = LinearResampler;
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else if(strcasecmp(str, "sinc4") == 0)
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resampler = FIR4Resampler;
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else if(strcasecmp(str, "bsinc") == 0)
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resampler = BSincResampler;
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else if(strcasecmp(str, "cubic") == 0 || strcasecmp(str, "sinc8") == 0)
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{
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WARN("Resampler option \"%s\" is deprecated, using sinc4\n", str);
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resampler = 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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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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for(i = 0;i < FRACTIONONE;i++)
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{
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ALdouble mu = (ALdouble)i / FRACTIONONE;
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ResampleCoeffs_FIR4[i][0] = SincKaiser(2.0, mu - -1.0);
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ResampleCoeffs_FIR4[i][1] = SincKaiser(2.0, mu - 0.0);
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ResampleCoeffs_FIR4[i][2] = SincKaiser(2.0, mu - 1.0);
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ResampleCoeffs_FIR4[i][3] = SincKaiser(2.0, mu - 2.0);
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}
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MixHrtfSamples = SelectHrtfMixer();
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MixSamples = SelectMixer();
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ResampleSamples = SelectResampler(resampler);
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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 inline void Load_##T(ALfloat *dst, const T *src, ALint srcstep, ALsizei samples)\
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{ \
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ALsizei 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, ALint srcstep, enum FmtType srctype, ALsizei 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 inline void SilenceSamples(ALfloat *dst, ALsizei samples)
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{
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ALsizei 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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ALsizei numsamples, enum ActiveFilters 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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ALfilterState_processPassthru(lpfilter, src, numsamples);
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ALfilterState_processPassthru(hpfilter, src, numsamples);
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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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ALfilterState_processPassthru(hpfilter, dst, numsamples);
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return dst;
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case AF_HighPass:
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ALfilterState_processPassthru(lpfilter, src, numsamples);
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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[256];
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ALsizei todo = mini(256, 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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ALboolean MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALsizei SamplesToDo)
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{
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ResamplerFunc Resample;
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ALbufferlistitem *BufferListItem;
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ALuint DataPosInt, DataPosFrac;
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bool isplaying = true;
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bool islooping;
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ALint increment;
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ALsizei OutPos;
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ALsizei NumChannels;
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ALsizei SampleSize;
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ALint64 DataSize64;
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ALsizei Counter;
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ALsizei IrSize;
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ALsizei chan;
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ALsizei send;
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/* Get source info */
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DataPosInt = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
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DataPosFrac = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
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BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
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islooping = ATOMIC_LOAD(&Source->looping, almemory_order_relaxed);
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NumChannels = voice->NumChannels;
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SampleSize = voice->SampleSize;
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increment = voice->Step;
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IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
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Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
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Resample_copy32_C : ResampleSamples);
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Counter = (voice->Flags&VOICE_IS_MOVING) ? SamplesToDo : 0;
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OutPos = 0;
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do {
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ALsizei 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 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
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SrcBufferSize = (ALsizei)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 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
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DataSize64 <<= FRACTIONBITS;
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DataSize64 -= DataPosFrac;
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DstBufferSize = (ALsizei)((DataSize64+(increment-1)) / increment);
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DstBufferSize = mini(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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ALsizei SrcDataSize;
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/* Load the previous samples into the source data first. */
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memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat));
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SrcDataSize = MAX_PRE_SAMPLES;
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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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ALsizei DataSize;
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/* Offset buffer data to current channel */
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Data += chan*SampleSize;
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/* If current pos is beyond the loop range, do not loop */
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if(!islooping || DataPosInt >= (ALuint)ALBuffer->LoopEnd)
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{
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islooping = false;
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/* Load what's left to play from the source buffer, and
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* clear the rest of the temp buffer */
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DataSize = minu(SrcBufferSize - SrcDataSize,
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ALBuffer->SampleLen - DataPosInt);
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LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*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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ALsizei LoopStart = ALBuffer->LoopStart;
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ALsizei LoopEnd = ALBuffer->LoopEnd;
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/* Load what's left of this loop iteration, then load
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* repeats of the loop section */
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DataSize = minu(SrcBufferSize - SrcDataSize, LoopEnd - DataPosInt);
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LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*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 = mini(SrcBufferSize - SrcDataSize, DataSize);
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LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*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 = DataPosInt;
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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);
|
|
SrcDataSize += DataSize;
|
|
}
|
|
}
|
|
tmpiter = tmpiter->next;
|
|
if(!tmpiter && islooping)
|
|
tmpiter = ATOMIC_LOAD(&Source->queue, almemory_order_acquire);
|
|
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->ResampleState,
|
|
&SrcData[MAX_PRE_SAMPLES], DataPosFrac, increment,
|
|
Device->ResampledData, DstBufferSize
|
|
);
|
|
{
|
|
DirectParams *parms = &voice->Direct.Params[chan];
|
|
const ALfloat *samples;
|
|
|
|
samples = DoFilters(
|
|
&parms->LowPass, &parms->HighPass, Device->FilteredData,
|
|
ResampledData, DstBufferSize, parms->FilterType
|
|
);
|
|
if(!(voice->Flags&VOICE_IS_HRTF))
|
|
{
|
|
if(!Counter)
|
|
memcpy(parms->Gains.Current, parms->Gains.Target,
|
|
sizeof(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
|
|
{
|
|
ALfloat *nfcsamples = Device->NFCtrlData;
|
|
ALsizei chanoffset = 0;
|
|
MixSamples(samples,
|
|
voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
|
|
parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
|
|
DstBufferSize
|
|
);
|
|
chanoffset += voice->Direct.ChannelsPerOrder[0];
|
|
if(voice->Direct.ChannelsPerOrder[1] > 0)
|
|
{
|
|
NfcFilterUpdate1(&parms->NFCtrlFilter[0], nfcsamples, samples,
|
|
DstBufferSize);
|
|
MixSamples(nfcsamples,
|
|
voice->Direct.ChannelsPerOrder[1], voice->Direct.Buffer+chanoffset,
|
|
parms->Gains.Current+chanoffset, parms->Gains.Target+chanoffset,
|
|
Counter, OutPos, DstBufferSize
|
|
);
|
|
chanoffset += voice->Direct.ChannelsPerOrder[1];
|
|
}
|
|
if(voice->Direct.ChannelsPerOrder[2] > 0)
|
|
{
|
|
NfcFilterUpdate2(&parms->NFCtrlFilter[1], nfcsamples, samples,
|
|
DstBufferSize);
|
|
MixSamples(nfcsamples,
|
|
voice->Direct.ChannelsPerOrder[2], voice->Direct.Buffer+chanoffset,
|
|
parms->Gains.Current+chanoffset, parms->Gains.Target+chanoffset,
|
|
Counter, OutPos, DstBufferSize
|
|
);
|
|
chanoffset += voice->Direct.ChannelsPerOrder[2];
|
|
}
|
|
if(voice->Direct.ChannelsPerOrder[3] > 0)
|
|
{
|
|
NfcFilterUpdate3(&parms->NFCtrlFilter[2], nfcsamples, samples,
|
|
DstBufferSize);
|
|
MixSamples(nfcsamples,
|
|
voice->Direct.ChannelsPerOrder[3], voice->Direct.Buffer+chanoffset,
|
|
parms->Gains.Current+chanoffset, parms->Gains.Target+chanoffset,
|
|
Counter, OutPos, DstBufferSize
|
|
);
|
|
chanoffset += voice->Direct.ChannelsPerOrder[3];
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
MixHrtfParams hrtfparams;
|
|
int lidx, ridx;
|
|
|
|
lidx = GetChannelIdxByName(Device->RealOut, FrontLeft);
|
|
ridx = GetChannelIdxByName(Device->RealOut, FrontRight);
|
|
assert(lidx != -1 && ridx != -1);
|
|
|
|
if(!Counter)
|
|
{
|
|
parms->Hrtf.Old = parms->Hrtf.Target;
|
|
hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
|
|
hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
|
|
hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
|
|
hrtfparams.Gain = parms->Hrtf.Target.Gain;
|
|
hrtfparams.GainStep = 0.0f;
|
|
MixHrtfSamples(
|
|
voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
|
|
samples, voice->Offset, OutPos, IrSize, &hrtfparams,
|
|
&parms->Hrtf.State, DstBufferSize
|
|
);
|
|
}
|
|
else
|
|
{
|
|
HrtfState backupstate = parms->Hrtf.State;
|
|
ALfloat gain;
|
|
|
|
/* The old coefficients need to fade to silence
|
|
* completely since they'll be replaced after the mix.
|
|
* So it needs to fade out over DstBufferSize instead
|
|
* of Counter.
|
|
*/
|
|
hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Old.Coeffs);
|
|
hrtfparams.Delay[0] = parms->Hrtf.Old.Delay[0];
|
|
hrtfparams.Delay[1] = parms->Hrtf.Old.Delay[1];
|
|
hrtfparams.Gain = parms->Hrtf.Old.Gain;
|
|
hrtfparams.GainStep = -hrtfparams.Gain /
|
|
(ALfloat)DstBufferSize;
|
|
MixHrtfSamples(
|
|
voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
|
|
samples, voice->Offset, OutPos, IrSize, &hrtfparams,
|
|
&backupstate, DstBufferSize
|
|
);
|
|
|
|
/* The new coefficients need to fade in completely
|
|
* since they're replacing the old ones. To keep the
|
|
* source gain fading consistent, interpolate between
|
|
* the old and new target gain given how much of the
|
|
* fade time this mix handles.
|
|
*/
|
|
gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
|
|
minf(1.0f, (ALfloat)Counter/DstBufferSize));
|
|
hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,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)DstBufferSize;
|
|
MixHrtfSamples(
|
|
voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
|
|
samples, voice->Offset, OutPos, IrSize, &hrtfparams,
|
|
&parms->Hrtf.State, DstBufferSize
|
|
);
|
|
/* Update the old parameters with the result. */
|
|
parms->Hrtf.Old = parms->Hrtf.Target;
|
|
if(Counter > DstBufferSize)
|
|
parms->Hrtf.Old.Gain = hrtfparams.Gain;
|
|
}
|
|
}
|
|
}
|
|
|
|
for(send = 0;send < Device->NumAuxSends;send++)
|
|
{
|
|
SendParams *parms = &voice->Send[send].Params[chan];
|
|
const ALfloat *samples;
|
|
|
|
if(!voice->Send[send].Buffer)
|
|
continue;
|
|
|
|
samples = DoFilters(
|
|
&parms->LowPass, &parms->HighPass, Device->FilteredData,
|
|
ResampledData, DstBufferSize, parms->FilterType
|
|
);
|
|
|
|
if(!Counter)
|
|
memcpy(parms->Gains.Current, parms->Gains.Target,
|
|
sizeof(parms->Gains.Current));
|
|
MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
|
|
parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
|
|
);
|
|
}
|
|
}
|
|
/* Update positions */
|
|
DataPosFrac += increment*DstBufferSize;
|
|
DataPosInt += DataPosFrac>>FRACTIONBITS;
|
|
DataPosFrac &= FRACTIONMASK;
|
|
|
|
OutPos += DstBufferSize;
|
|
voice->Offset += DstBufferSize;
|
|
Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
|
|
|
|
/* Handle looping sources */
|
|
while(1)
|
|
{
|
|
const ALbuffer *ALBuffer;
|
|
ALsizei DataSize = 0;
|
|
ALsizei LoopStart = 0;
|
|
ALsizei LoopEnd = 0;
|
|
|
|
if((ALBuffer=BufferListItem->buffer) != NULL)
|
|
{
|
|
DataSize = ALBuffer->SampleLen;
|
|
LoopStart = ALBuffer->LoopStart;
|
|
LoopEnd = ALBuffer->LoopEnd;
|
|
if((ALuint)LoopEnd > DataPosInt)
|
|
break;
|
|
}
|
|
|
|
if(islooping && Source->SourceType == AL_STATIC)
|
|
{
|
|
assert(LoopEnd > LoopStart);
|
|
DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
|
|
break;
|
|
}
|
|
|
|
if((ALuint)DataSize > DataPosInt)
|
|
break;
|
|
|
|
if(!(BufferListItem=BufferListItem->next))
|
|
{
|
|
if(islooping)
|
|
BufferListItem = ATOMIC_LOAD(&Source->queue, almemory_order_acquire);
|
|
else
|
|
{
|
|
isplaying = false;
|
|
BufferListItem = NULL;
|
|
DataPosInt = 0;
|
|
DataPosFrac = 0;
|
|
break;
|
|
}
|
|
}
|
|
|
|
DataPosInt -= DataSize;
|
|
}
|
|
} while(isplaying && OutPos < SamplesToDo);
|
|
|
|
voice->Flags |= VOICE_IS_MOVING;
|
|
|
|
/* Update source info */
|
|
ATOMIC_STORE(&voice->position, DataPosInt, almemory_order_relaxed);
|
|
ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
|
|
ATOMIC_STORE(&voice->current_buffer, BufferListItem, almemory_order_release);
|
|
return isplaying;
|
|
}
|