1461 lines
53 KiB
C
1461 lines
53 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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 "alThunk.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 "bs2b.h"
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#if defined(HAVE_STDINT_H)
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#include <stdint.h>
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typedef int64_t ALint64;
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#elif defined(HAVE___INT64)
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typedef __int64 ALint64;
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#elif (SIZEOF_LONG == 8)
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typedef long ALint64;
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#elif (SIZEOF_LONG_LONG == 8)
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typedef long long ALint64;
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#endif
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#define FRACTIONBITS 14
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#define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
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#define MAX_PITCH 65536
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/* Minimum ramp length in milliseconds. The value below was chosen to
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* adequately reduce clicks and pops from harsh gain changes. */
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#define MIN_RAMP_LENGTH 16
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ALboolean DuplicateStereo = AL_FALSE;
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static __inline ALfloat aluF2F(ALfloat Value)
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{
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return Value;
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}
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static __inline ALshort aluF2S(ALfloat Value)
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{
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ALint i;
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if(Value < 0.0f)
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{
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i = (ALint)(Value*32768.0f);
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i = max(-32768, i);
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}
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else
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{
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i = (ALint)(Value*32767.0f);
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i = min( 32767, i);
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}
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return ((ALshort)i);
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}
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static __inline ALubyte aluF2UB(ALfloat Value)
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{
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ALshort i = aluF2S(Value);
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return (i>>8)+128;
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}
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static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
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{
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outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
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outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
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outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
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}
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static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
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{
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return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
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inVector1[2]*inVector2[2];
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}
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static __inline ALvoid aluNormalize(ALfloat *inVector)
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{
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ALfloat length, inverse_length;
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length = aluSqrt(aluDotproduct(inVector, inVector));
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if(length != 0.0f)
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{
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inverse_length = 1.0f/length;
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inVector[0] *= inverse_length;
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inVector[1] *= inverse_length;
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inVector[2] *= inverse_length;
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}
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}
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static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
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{
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ALfloat temp[4] = {
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vector[0], vector[1], vector[2], w
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};
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vector[0] = temp[0]*matrix[0][0] + temp[1]*matrix[1][0] + temp[2]*matrix[2][0] + temp[3]*matrix[3][0];
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vector[1] = temp[0]*matrix[0][1] + temp[1]*matrix[1][1] + temp[2]*matrix[2][1] + temp[3]*matrix[3][1];
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vector[2] = temp[0]*matrix[0][2] + temp[1]*matrix[1][2] + temp[2]*matrix[2][2] + temp[3]*matrix[3][2];
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}
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static ALvoid SetSpeakerArrangement(const char *name, ALfloat SpeakerAngle[OUTPUTCHANNELS],
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ALint Speaker2Chan[OUTPUTCHANNELS], ALint chans)
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{
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const char *confkey;
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const char *next;
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const char *sep;
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const char *end;
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int i, val;
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confkey = GetConfigValue(NULL, name, "");
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next = confkey;
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while(next && *next)
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{
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confkey = next;
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next = strchr(confkey, ',');
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if(next)
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{
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do {
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next++;
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} while(isspace(*next));
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}
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sep = strchr(confkey, '=');
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if(!sep || confkey == sep)
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continue;
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end = sep - 1;
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while(isspace(*end) && end != confkey)
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end--;
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end++;
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if(strncmp(confkey, "fl", end-confkey) == 0)
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val = FRONT_LEFT;
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else if(strncmp(confkey, "fr", end-confkey) == 0)
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val = FRONT_RIGHT;
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else if(strncmp(confkey, "fc", end-confkey) == 0)
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val = FRONT_CENTER;
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else if(strncmp(confkey, "bl", end-confkey) == 0)
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val = BACK_LEFT;
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else if(strncmp(confkey, "br", end-confkey) == 0)
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val = BACK_RIGHT;
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else if(strncmp(confkey, "bc", end-confkey) == 0)
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val = BACK_CENTER;
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else if(strncmp(confkey, "sl", end-confkey) == 0)
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val = SIDE_LEFT;
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else if(strncmp(confkey, "sr", end-confkey) == 0)
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val = SIDE_RIGHT;
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else
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{
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AL_PRINT("Unknown speaker for %s: \"%c%c\"\n", name, confkey[0], confkey[1]);
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continue;
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}
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sep++;
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while(isspace(*sep))
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sep++;
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for(i = 0;i < chans;i++)
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{
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if(Speaker2Chan[i] == val)
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{
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val = strtol(sep, NULL, 10);
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if(val >= -180 && val <= 180)
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SpeakerAngle[i] = val * M_PI/180.0f;
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else
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AL_PRINT("Invalid angle for speaker \"%c%c\": %d\n", confkey[0], confkey[1], val);
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break;
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}
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}
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}
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for(i = 1;i < chans;i++)
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{
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if(SpeakerAngle[i] <= SpeakerAngle[i-1])
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{
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AL_PRINT("Speaker %d of %d does not follow previous: %f > %f\n", i, chans,
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SpeakerAngle[i-1] * 180.0f/M_PI, SpeakerAngle[i] * 180.0f/M_PI);
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SpeakerAngle[i] = SpeakerAngle[i-1] + 1 * 180.0f/M_PI;
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}
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}
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}
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static __inline ALfloat aluLUTpos2Angle(ALint pos)
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{
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if(pos < QUADRANT_NUM)
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return aluAtan((ALfloat)pos / (ALfloat)(QUADRANT_NUM - pos));
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if(pos < 2 * QUADRANT_NUM)
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return M_PI_2 + aluAtan((ALfloat)(pos - QUADRANT_NUM) / (ALfloat)(2 * QUADRANT_NUM - pos));
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if(pos < 3 * QUADRANT_NUM)
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return aluAtan((ALfloat)(pos - 2 * QUADRANT_NUM) / (ALfloat)(3 * QUADRANT_NUM - pos)) - M_PI;
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return aluAtan((ALfloat)(pos - 3 * QUADRANT_NUM) / (ALfloat)(4 * QUADRANT_NUM - pos)) - M_PI_2;
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}
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ALvoid aluInitPanning(ALCcontext *Context)
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{
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ALint pos, offset, s;
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ALfloat Alpha, Theta;
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ALfloat SpeakerAngle[OUTPUTCHANNELS];
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ALint Speaker2Chan[OUTPUTCHANNELS];
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for(s = 0;s < OUTPUTCHANNELS;s++)
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{
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int s2;
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for(s2 = 0;s2 < OUTPUTCHANNELS;s2++)
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Context->ChannelMatrix[s][s2] = ((s==s2) ? 1.0f : 0.0f);
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}
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switch(Context->Device->Format)
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{
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/* Mono is rendered as stereo, then downmixed during post-process */
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case AL_FORMAT_MONO8:
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case AL_FORMAT_MONO16:
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case AL_FORMAT_MONO_FLOAT32:
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Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = 1.0f;
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Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = 1.0f;
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Context->ChannelMatrix[BACK_LEFT][FRONT_LEFT] = 1.0f;
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Context->ChannelMatrix[BACK_RIGHT][FRONT_RIGHT] = 1.0f;
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Context->ChannelMatrix[BACK_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 2;
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Speaker2Chan[0] = FRONT_LEFT;
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Speaker2Chan[1] = FRONT_RIGHT;
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SpeakerAngle[0] = -90.0f * M_PI/180.0f;
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SpeakerAngle[1] = 90.0f * M_PI/180.0f;
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break;
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case AL_FORMAT_STEREO8:
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case AL_FORMAT_STEREO16:
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case AL_FORMAT_STEREO_FLOAT32:
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Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = 1.0f;
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Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = 1.0f;
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Context->ChannelMatrix[BACK_LEFT][FRONT_LEFT] = 1.0f;
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Context->ChannelMatrix[BACK_RIGHT][FRONT_RIGHT] = 1.0f;
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Context->ChannelMatrix[BACK_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 2;
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Speaker2Chan[0] = FRONT_LEFT;
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Speaker2Chan[1] = FRONT_RIGHT;
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SpeakerAngle[0] = -90.0f * M_PI/180.0f;
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SpeakerAngle[1] = 90.0f * M_PI/180.0f;
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SetSpeakerArrangement("layout_STEREO", SpeakerAngle, Speaker2Chan, Context->NumChan);
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break;
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case AL_FORMAT_QUAD8:
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case AL_FORMAT_QUAD16:
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case AL_FORMAT_QUAD32:
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Context->ChannelMatrix[FRONT_CENTER][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[FRONT_CENTER][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 4;
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Speaker2Chan[0] = BACK_LEFT;
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Speaker2Chan[1] = FRONT_LEFT;
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Speaker2Chan[2] = FRONT_RIGHT;
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Speaker2Chan[3] = BACK_RIGHT;
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SpeakerAngle[0] = -135.0f * M_PI/180.0f;
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SpeakerAngle[1] = -45.0f * M_PI/180.0f;
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SpeakerAngle[2] = 45.0f * M_PI/180.0f;
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SpeakerAngle[3] = 135.0f * M_PI/180.0f;
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SetSpeakerArrangement("layout_QUAD", SpeakerAngle, Speaker2Chan, Context->NumChan);
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break;
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case AL_FORMAT_51CHN8:
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case AL_FORMAT_51CHN16:
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case AL_FORMAT_51CHN32:
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Context->ChannelMatrix[SIDE_LEFT][FRONT_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_LEFT][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_RIGHT][FRONT_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[SIDE_RIGHT][BACK_RIGHT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 5;
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Speaker2Chan[0] = BACK_LEFT;
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Speaker2Chan[1] = FRONT_LEFT;
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Speaker2Chan[2] = FRONT_CENTER;
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Speaker2Chan[3] = FRONT_RIGHT;
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Speaker2Chan[4] = BACK_RIGHT;
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SpeakerAngle[0] = -110.0f * M_PI/180.0f;
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SpeakerAngle[1] = -30.0f * M_PI/180.0f;
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SpeakerAngle[2] = 0.0f * M_PI/180.0f;
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SpeakerAngle[3] = 30.0f * M_PI/180.0f;
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SpeakerAngle[4] = 110.0f * M_PI/180.0f;
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SetSpeakerArrangement("layout_51CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
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break;
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case AL_FORMAT_61CHN8:
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case AL_FORMAT_61CHN16:
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case AL_FORMAT_61CHN32:
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Context->ChannelMatrix[BACK_LEFT][BACK_CENTER] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_LEFT][SIDE_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_RIGHT][BACK_CENTER] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_RIGHT][SIDE_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 6;
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Speaker2Chan[0] = SIDE_LEFT;
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Speaker2Chan[1] = FRONT_LEFT;
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Speaker2Chan[2] = FRONT_CENTER;
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Speaker2Chan[3] = FRONT_RIGHT;
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Speaker2Chan[4] = SIDE_RIGHT;
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Speaker2Chan[5] = BACK_CENTER;
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SpeakerAngle[0] = -90.0f * M_PI/180.0f;
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SpeakerAngle[1] = -30.0f * M_PI/180.0f;
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SpeakerAngle[2] = 0.0f * M_PI/180.0f;
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SpeakerAngle[3] = 30.0f * M_PI/180.0f;
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SpeakerAngle[4] = 90.0f * M_PI/180.0f;
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SpeakerAngle[5] = 180.0f * M_PI/180.0f;
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SetSpeakerArrangement("layout_61CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
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break;
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case AL_FORMAT_71CHN8:
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case AL_FORMAT_71CHN16:
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case AL_FORMAT_71CHN32:
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Context->ChannelMatrix[BACK_CENTER][BACK_LEFT] = aluSqrt(0.5);
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Context->ChannelMatrix[BACK_CENTER][BACK_RIGHT] = aluSqrt(0.5);
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Context->NumChan = 7;
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Speaker2Chan[0] = BACK_LEFT;
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Speaker2Chan[1] = SIDE_LEFT;
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Speaker2Chan[2] = FRONT_LEFT;
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Speaker2Chan[3] = FRONT_CENTER;
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Speaker2Chan[4] = FRONT_RIGHT;
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Speaker2Chan[5] = SIDE_RIGHT;
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Speaker2Chan[6] = BACK_RIGHT;
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SpeakerAngle[0] = -150.0f * M_PI/180.0f;
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SpeakerAngle[1] = -90.0f * M_PI/180.0f;
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SpeakerAngle[2] = -30.0f * M_PI/180.0f;
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SpeakerAngle[3] = 0.0f * M_PI/180.0f;
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SpeakerAngle[4] = 30.0f * M_PI/180.0f;
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SpeakerAngle[5] = 90.0f * M_PI/180.0f;
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SpeakerAngle[6] = 150.0f * M_PI/180.0f;
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SetSpeakerArrangement("layout_71CHN", SpeakerAngle, Speaker2Chan, Context->NumChan);
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break;
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default:
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assert(0);
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}
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for(pos = 0; pos < LUT_NUM; pos++)
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{
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/* source angle */
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Theta = aluLUTpos2Angle(pos);
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/* clear all values */
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offset = OUTPUTCHANNELS * pos;
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for(s = 0; s < OUTPUTCHANNELS; s++)
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Context->PanningLUT[offset+s] = 0.0f;
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/* set panning values */
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for(s = 0; s < Context->NumChan - 1; s++)
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{
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if(Theta >= SpeakerAngle[s] && Theta < SpeakerAngle[s+1])
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{
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/* source between speaker s and speaker s+1 */
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Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
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(SpeakerAngle[s+1]-SpeakerAngle[s]);
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Context->PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
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Context->PanningLUT[offset + Speaker2Chan[s+1]] = sin(Alpha);
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break;
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}
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}
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if(s == Context->NumChan - 1)
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{
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/* source between last and first speaker */
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if(Theta < SpeakerAngle[0])
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Theta += 2.0f * M_PI;
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Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
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(2.0f * M_PI + SpeakerAngle[0]-SpeakerAngle[s]);
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Context->PanningLUT[offset + Speaker2Chan[s]] = cos(Alpha);
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Context->PanningLUT[offset + Speaker2Chan[0]] = sin(Alpha);
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}
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}
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}
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|
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static ALvoid CalcNonAttnSourceParams(const ALCcontext *ALContext, ALsource *ALSource)
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{
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ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
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ALfloat DryGain, DryGainHF;
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ALfloat WetGain[MAX_SENDS];
|
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ALfloat WetGainHF[MAX_SENDS];
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ALint NumSends, Frequency;
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ALint i;
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ALfloat cw, a, g;
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//Get context properties
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NumSends = ALContext->Device->NumAuxSends;
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Frequency = ALContext->Device->NumAuxSends;
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//Get listener properties
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ListenerGain = ALContext->Listener.Gain;
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//Get source properties
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SourceVolume = ALSource->flGain;
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MinVolume = ALSource->flMinGain;
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MaxVolume = ALSource->flMaxGain;
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|
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//1. Multi-channel buffers always play "normal"
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ALSource->Params.Pitch = ALSource->flPitch;
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DryGain = SourceVolume;
|
|
DryGain = __min(DryGain,MaxVolume);
|
|
DryGain = __max(DryGain,MinVolume);
|
|
DryGainHF = 1.0f;
|
|
|
|
switch(ALSource->DirectFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
DryGain *= ALSource->DirectFilter.Gain;
|
|
DryGainHF *= ALSource->DirectFilter.GainHF;
|
|
break;
|
|
}
|
|
|
|
ALSource->Params.DryGains[FRONT_LEFT] = DryGain * ListenerGain;
|
|
ALSource->Params.DryGains[FRONT_RIGHT] = DryGain * ListenerGain;
|
|
ALSource->Params.DryGains[SIDE_LEFT] = DryGain * ListenerGain;
|
|
ALSource->Params.DryGains[SIDE_RIGHT] = DryGain * ListenerGain;
|
|
ALSource->Params.DryGains[BACK_LEFT] = DryGain * ListenerGain;
|
|
ALSource->Params.DryGains[BACK_RIGHT] = DryGain * ListenerGain;
|
|
ALSource->Params.DryGains[FRONT_CENTER] = DryGain * ListenerGain;
|
|
ALSource->Params.DryGains[BACK_CENTER] = DryGain * ListenerGain;
|
|
ALSource->Params.DryGains[LFE] = DryGain * ListenerGain;
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
WetGain[i] = SourceVolume;
|
|
WetGain[i] = __min(WetGain[i],MaxVolume);
|
|
WetGain[i] = __max(WetGain[i],MinVolume);
|
|
WetGainHF[i] = 1.0f;
|
|
|
|
switch(ALSource->Send[i].WetFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
|
|
WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
|
|
break;
|
|
}
|
|
|
|
ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain;
|
|
}
|
|
for(i = NumSends;i < MAX_SENDS;i++)
|
|
{
|
|
ALSource->Params.WetGains[i] = 0.0f;
|
|
WetGainHF[i] = 1.0f;
|
|
}
|
|
|
|
/* Update filter coefficients. Calculations based on the I3DL2
|
|
* spec. */
|
|
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
|
|
/* We use two chained one-pole filters, so we need to take the
|
|
* square root of the squared gain, which is the same as the base
|
|
* gain. */
|
|
g = __max(DryGainHF, 0.01f);
|
|
a = 0.0f;
|
|
/* Be careful with gains < 0.0001, as that causes the coefficient
|
|
* head towards 1, which will flatten the signal */
|
|
if(g < 0.9999f) /* 1-epsilon */
|
|
a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) /
|
|
(1 - g);
|
|
ALSource->Params.iirFilter.coeff = a;
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
/* We use a one-pole filter, so we need to take the squared gain */
|
|
g = __max(WetGainHF[i], 0.1f);
|
|
g *= g;
|
|
a = 0.0f;
|
|
if(g < 0.9999f) /* 1-epsilon */
|
|
a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) /
|
|
(1 - g);
|
|
ALSource->Params.Send[i].iirFilter.coeff = a;
|
|
}
|
|
}
|
|
|
|
static ALvoid CalcSourceParams(const ALCcontext *ALContext, ALsource *ALSource)
|
|
{
|
|
ALfloat InnerAngle,OuterAngle,Angle,Distance,DryMix,OrigDist;
|
|
ALfloat Direction[3],Position[3],SourceToListener[3];
|
|
ALfloat Velocity[3],ListenerVel[3];
|
|
ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff,OuterGainHF;
|
|
ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
|
|
ALfloat DopplerFactor, DopplerVelocity, flSpeedOfSound;
|
|
ALfloat Matrix[4][4];
|
|
ALfloat flAttenuation;
|
|
ALfloat RoomAttenuation[MAX_SENDS];
|
|
ALfloat MetersPerUnit;
|
|
ALfloat RoomRolloff[MAX_SENDS];
|
|
ALfloat DryGainHF = 1.0f;
|
|
ALfloat WetGain[MAX_SENDS];
|
|
ALfloat WetGainHF[MAX_SENDS];
|
|
ALfloat DirGain, AmbientGain;
|
|
ALfloat length;
|
|
const ALfloat *SpeakerGain;
|
|
ALuint Frequency;
|
|
ALint NumSends;
|
|
ALint pos, s, i;
|
|
ALfloat cw, a, g;
|
|
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
WetGainHF[i] = 1.0f;
|
|
|
|
//Get context properties
|
|
DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
|
|
DopplerVelocity = ALContext->DopplerVelocity;
|
|
flSpeedOfSound = ALContext->flSpeedOfSound;
|
|
NumSends = ALContext->Device->NumAuxSends;
|
|
Frequency = ALContext->Device->Frequency;
|
|
|
|
//Get listener properties
|
|
ListenerGain = ALContext->Listener.Gain;
|
|
MetersPerUnit = ALContext->Listener.MetersPerUnit;
|
|
memcpy(ListenerVel, ALContext->Listener.Velocity, sizeof(ALContext->Listener.Velocity));
|
|
|
|
//Get source properties
|
|
SourceVolume = ALSource->flGain;
|
|
memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition));
|
|
memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
|
|
memcpy(Velocity, ALSource->vVelocity, sizeof(ALSource->vVelocity));
|
|
MinVolume = ALSource->flMinGain;
|
|
MaxVolume = ALSource->flMaxGain;
|
|
MinDist = ALSource->flRefDistance;
|
|
MaxDist = ALSource->flMaxDistance;
|
|
Rolloff = ALSource->flRollOffFactor;
|
|
InnerAngle = ALSource->flInnerAngle;
|
|
OuterAngle = ALSource->flOuterAngle;
|
|
OuterGainHF = ALSource->OuterGainHF;
|
|
|
|
//1. Translate Listener to origin (convert to head relative)
|
|
if(ALSource->bHeadRelative==AL_FALSE)
|
|
{
|
|
ALfloat U[3],V[3],N[3],P[3];
|
|
|
|
// Build transform matrix
|
|
memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector
|
|
aluNormalize(N); // Normalized At-vector
|
|
memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector
|
|
aluNormalize(V); // Normalized Up-vector
|
|
aluCrossproduct(N, V, U); // Right-vector
|
|
aluNormalize(U); // Normalized Right-vector
|
|
P[0] = -(ALContext->Listener.Position[0]*U[0] + // Translation
|
|
ALContext->Listener.Position[1]*U[1] +
|
|
ALContext->Listener.Position[2]*U[2]);
|
|
P[1] = -(ALContext->Listener.Position[0]*V[0] +
|
|
ALContext->Listener.Position[1]*V[1] +
|
|
ALContext->Listener.Position[2]*V[2]);
|
|
P[2] = -(ALContext->Listener.Position[0]*-N[0] +
|
|
ALContext->Listener.Position[1]*-N[1] +
|
|
ALContext->Listener.Position[2]*-N[2]);
|
|
Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f;
|
|
Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f;
|
|
Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f;
|
|
Matrix[3][0] = P[0]; Matrix[3][1] = P[1]; Matrix[3][2] = P[2]; Matrix[3][3] = 1.0f;
|
|
|
|
// Transform source position and direction into listener space
|
|
aluMatrixVector(Position, 1.0f, Matrix);
|
|
aluMatrixVector(Direction, 0.0f, Matrix);
|
|
// Transform source and listener velocity into listener space
|
|
aluMatrixVector(Velocity, 0.0f, Matrix);
|
|
aluMatrixVector(ListenerVel, 0.0f, Matrix);
|
|
}
|
|
else
|
|
ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f;
|
|
|
|
SourceToListener[0] = -Position[0];
|
|
SourceToListener[1] = -Position[1];
|
|
SourceToListener[2] = -Position[2];
|
|
aluNormalize(SourceToListener);
|
|
aluNormalize(Direction);
|
|
|
|
//2. Calculate distance attenuation
|
|
Distance = aluSqrt(aluDotproduct(Position, Position));
|
|
OrigDist = Distance;
|
|
|
|
flAttenuation = 1.0f;
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
RoomAttenuation[i] = 1.0f;
|
|
|
|
RoomRolloff[i] = ALSource->RoomRolloffFactor;
|
|
if(ALSource->Send[i].Slot &&
|
|
(ALSource->Send[i].Slot->effect.type == AL_EFFECT_REVERB ||
|
|
ALSource->Send[i].Slot->effect.type == AL_EFFECT_EAXREVERB))
|
|
RoomRolloff[i] += ALSource->Send[i].Slot->effect.Reverb.RoomRolloffFactor;
|
|
}
|
|
|
|
switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
|
|
ALContext->DistanceModel)
|
|
{
|
|
case AL_INVERSE_DISTANCE_CLAMPED:
|
|
Distance=__max(Distance,MinDist);
|
|
Distance=__min(Distance,MaxDist);
|
|
if(MaxDist < MinDist)
|
|
break;
|
|
//fall-through
|
|
case AL_INVERSE_DISTANCE:
|
|
if(MinDist > 0.0f)
|
|
{
|
|
if((MinDist + (Rolloff * (Distance - MinDist))) > 0.0f)
|
|
flAttenuation = MinDist / (MinDist + (Rolloff * (Distance - MinDist)));
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if((MinDist + (RoomRolloff[i] * (Distance - MinDist))) > 0.0f)
|
|
RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (Distance - MinDist)));
|
|
}
|
|
}
|
|
break;
|
|
|
|
case AL_LINEAR_DISTANCE_CLAMPED:
|
|
Distance=__max(Distance,MinDist);
|
|
Distance=__min(Distance,MaxDist);
|
|
if(MaxDist < MinDist)
|
|
break;
|
|
//fall-through
|
|
case AL_LINEAR_DISTANCE:
|
|
Distance=__min(Distance,MaxDist);
|
|
if(MaxDist != MinDist)
|
|
{
|
|
flAttenuation = 1.0f - (Rolloff*(Distance-MinDist)/(MaxDist - MinDist));
|
|
for(i = 0;i < NumSends;i++)
|
|
RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(Distance-MinDist)/(MaxDist - MinDist));
|
|
}
|
|
break;
|
|
|
|
case AL_EXPONENT_DISTANCE_CLAMPED:
|
|
Distance=__max(Distance,MinDist);
|
|
Distance=__min(Distance,MaxDist);
|
|
if(MaxDist < MinDist)
|
|
break;
|
|
//fall-through
|
|
case AL_EXPONENT_DISTANCE:
|
|
if(Distance > 0.0f && MinDist > 0.0f)
|
|
{
|
|
flAttenuation = (ALfloat)pow(Distance/MinDist, -Rolloff);
|
|
for(i = 0;i < NumSends;i++)
|
|
RoomAttenuation[i] = (ALfloat)pow(Distance/MinDist, -RoomRolloff[i]);
|
|
}
|
|
break;
|
|
|
|
case AL_NONE:
|
|
break;
|
|
}
|
|
|
|
// Source Gain + Attenuation
|
|
DryMix = SourceVolume * flAttenuation;
|
|
for(i = 0;i < NumSends;i++)
|
|
WetGain[i] = SourceVolume * RoomAttenuation[i];
|
|
|
|
// Distance-based air absorption
|
|
if(ALSource->AirAbsorptionFactor > 0.0f && flAttenuation < 1.0f)
|
|
{
|
|
ALfloat absorb = 0.0f;
|
|
|
|
// Absorption calculation is done in dB
|
|
if(flAttenuation > 0.0f)
|
|
{
|
|
absorb = (MinDist/flAttenuation - MinDist)*MetersPerUnit *
|
|
(ALSource->AirAbsorptionFactor*AIRABSORBGAINDBHF);
|
|
// Convert dB to linear gain before applying
|
|
absorb = pow(10.0, absorb/20.0);
|
|
}
|
|
DryGainHF *= absorb;
|
|
for(i = 0;i < NumSends;i++)
|
|
WetGainHF[i] *= absorb;
|
|
}
|
|
|
|
//3. Apply directional soundcones
|
|
Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * 180.0f/M_PI;
|
|
if(Angle >= InnerAngle && Angle <= OuterAngle)
|
|
{
|
|
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
|
|
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f)*scale);
|
|
ConeHF = (1.0f+(OuterGainHF-1.0f)*scale);
|
|
}
|
|
else if(Angle > OuterAngle)
|
|
{
|
|
ConeVolume = (1.0f+(ALSource->flOuterGain-1.0f));
|
|
ConeHF = (1.0f+(OuterGainHF-1.0f));
|
|
}
|
|
else
|
|
{
|
|
ConeVolume = 1.0f;
|
|
ConeHF = 1.0f;
|
|
}
|
|
|
|
// Apply some high-frequency attenuation for sources behind the listener
|
|
// NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however
|
|
// that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is
|
|
// the same as SourceToListener[2]
|
|
Angle = aluAcos(SourceToListener[2]) * 180.0f/M_PI;
|
|
// Sources within the minimum distance attenuate less
|
|
if(OrigDist < MinDist)
|
|
Angle *= OrigDist/MinDist;
|
|
if(Angle > 90.0f)
|
|
{
|
|
ALfloat scale = (Angle-90.0f) / (180.1f-90.0f); // .1 to account for fp errors
|
|
ConeHF *= 1.0f - (ALContext->Device->HeadDampen*scale);
|
|
}
|
|
|
|
DryMix *= ConeVolume;
|
|
if(ALSource->DryGainHFAuto)
|
|
DryGainHF *= ConeHF;
|
|
|
|
// Clamp to Min/Max Gain
|
|
DryMix = __min(DryMix,MaxVolume);
|
|
DryMix = __max(DryMix,MinVolume);
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
if(ALSource->Send[i].Slot &&
|
|
ALSource->Send[i].Slot->effect.type != AL_EFFECT_NULL)
|
|
{
|
|
if(ALSource->Send[i].Slot->AuxSendAuto)
|
|
{
|
|
if(ALSource->WetGainAuto)
|
|
WetGain[i] *= ConeVolume;
|
|
if(ALSource->WetGainHFAuto)
|
|
WetGainHF[i] *= ConeHF;
|
|
|
|
// Clamp to Min/Max Gain
|
|
WetGain[i] = __min(WetGain[i],MaxVolume);
|
|
WetGain[i] = __max(WetGain[i],MinVolume);
|
|
|
|
if(ALSource->Send[i].Slot->effect.type == AL_EFFECT_REVERB ||
|
|
ALSource->Send[i].Slot->effect.type == AL_EFFECT_EAXREVERB)
|
|
{
|
|
/* Apply a decay-time transformation to the wet path,
|
|
* based on the attenuation of the dry path. This should
|
|
* better approximate the statistical attenuation model
|
|
* for the reverb effect.
|
|
*
|
|
* This simple equation converts the distance attenuation
|
|
* into the time it would take to reach -60 dB. From
|
|
* there it establishes an origin (0.333s; the decay time
|
|
* that will produce equal attenuation) and applies the
|
|
* current decay time. Finally, it converts the result
|
|
* back to an attenuation for the reverb path.
|
|
*/
|
|
WetGain[i] *= pow(10.0f, log10(flAttenuation) * 0.333f /
|
|
ALSource->Send[i].Slot->effect.Reverb.DecayTime);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// If the slot's auxiliary send auto is off, the data sent to
|
|
// the effect slot is the same as the dry path, sans filter
|
|
// effects
|
|
WetGain[i] = DryMix;
|
|
WetGainHF[i] = DryGainHF;
|
|
}
|
|
|
|
switch(ALSource->Send[i].WetFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
|
|
WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
|
|
break;
|
|
}
|
|
ALSource->Params.WetGains[i] = WetGain[i] * ListenerGain;
|
|
}
|
|
else
|
|
{
|
|
ALSource->Params.WetGains[i] = 0.0f;
|
|
WetGainHF[i] = 1.0f;
|
|
}
|
|
}
|
|
for(i = NumSends;i < MAX_SENDS;i++)
|
|
{
|
|
ALSource->Params.WetGains[i] = 0.0f;
|
|
WetGainHF[i] = 1.0f;
|
|
}
|
|
|
|
// Apply filter gains and filters
|
|
switch(ALSource->DirectFilter.type)
|
|
{
|
|
case AL_FILTER_LOWPASS:
|
|
DryMix *= ALSource->DirectFilter.Gain;
|
|
DryGainHF *= ALSource->DirectFilter.GainHF;
|
|
break;
|
|
}
|
|
DryMix *= ListenerGain;
|
|
|
|
// Calculate Velocity
|
|
if(DopplerFactor != 0.0f)
|
|
{
|
|
ALfloat flVSS, flVLS;
|
|
ALfloat flMaxVelocity = (DopplerVelocity * flSpeedOfSound) /
|
|
DopplerFactor;
|
|
|
|
flVSS = aluDotproduct(Velocity, SourceToListener);
|
|
if(flVSS >= flMaxVelocity)
|
|
flVSS = (flMaxVelocity - 1.0f);
|
|
else if(flVSS <= -flMaxVelocity)
|
|
flVSS = -flMaxVelocity + 1.0f;
|
|
|
|
flVLS = aluDotproduct(ListenerVel, SourceToListener);
|
|
if(flVLS >= flMaxVelocity)
|
|
flVLS = (flMaxVelocity - 1.0f);
|
|
else if(flVLS <= -flMaxVelocity)
|
|
flVLS = -flMaxVelocity + 1.0f;
|
|
|
|
ALSource->Params.Pitch = ALSource->flPitch *
|
|
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVLS)) /
|
|
((flSpeedOfSound * DopplerVelocity) - (DopplerFactor * flVSS));
|
|
}
|
|
else
|
|
ALSource->Params.Pitch = ALSource->flPitch;
|
|
|
|
// Use energy-preserving panning algorithm for multi-speaker playback
|
|
length = __max(OrigDist, MinDist);
|
|
if(length > 0.0f)
|
|
{
|
|
ALfloat invlen = 1.0f/length;
|
|
Position[0] *= invlen;
|
|
Position[1] *= invlen;
|
|
Position[2] *= invlen;
|
|
}
|
|
|
|
pos = aluCart2LUTpos(-Position[2], Position[0]);
|
|
SpeakerGain = &ALContext->PanningLUT[OUTPUTCHANNELS * pos];
|
|
|
|
DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
|
|
// elevation adjustment for directional gain. this sucks, but
|
|
// has low complexity
|
|
AmbientGain = 1.0/aluSqrt(ALContext->NumChan) * (1.0-DirGain);
|
|
for(s = 0; s < OUTPUTCHANNELS; s++)
|
|
{
|
|
ALfloat gain = SpeakerGain[s]*DirGain + AmbientGain;
|
|
ALSource->Params.DryGains[s] = DryMix * gain;
|
|
}
|
|
|
|
/* Update filter coefficients. */
|
|
cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
|
|
/* Spatialized sources use four chained one-pole filters, so we need to
|
|
* take the fourth root of the squared gain, which is the same as the
|
|
* square root of the base gain. */
|
|
g = aluSqrt(__max(DryGainHF, 0.0001f));
|
|
a = 0.0f;
|
|
if(g < 0.9999f) /* 1-epsilon */
|
|
a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) /
|
|
(1 - g);
|
|
ALSource->Params.iirFilter.coeff = a;
|
|
|
|
for(i = 0;i < NumSends;i++)
|
|
{
|
|
/* The wet path uses two chained one-pole filters, so take the
|
|
* base gain (square root of the squared gain) */
|
|
g = __max(WetGainHF[i], 0.01f);
|
|
a = 0.0f;
|
|
if(g < 0.9999f) /* 1-epsilon */
|
|
a = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) /
|
|
(1 - g);
|
|
ALSource->Params.Send[i].iirFilter.coeff = a;
|
|
}
|
|
}
|
|
|
|
static __inline ALfloat lerp(ALfloat val1, ALfloat val2, ALint frac)
|
|
{
|
|
return val1 + ((val2-val1)*(frac * (1.0f/(1<<FRACTIONBITS))));
|
|
}
|
|
|
|
static void MixSomeSources(ALCcontext *ALContext, float (*DryBuffer)[OUTPUTCHANNELS], ALuint SamplesToDo)
|
|
{
|
|
static float DummyBuffer[BUFFERSIZE];
|
|
ALfloat *WetBuffer[MAX_SENDS];
|
|
ALfloat (*Matrix)[OUTPUTCHANNELS] = ALContext->ChannelMatrix;
|
|
ALfloat DrySend[OUTPUTCHANNELS];
|
|
ALfloat dryGainStep[OUTPUTCHANNELS];
|
|
ALfloat wetGainStep[MAX_SENDS];
|
|
ALuint i, j, k, out;
|
|
ALsource *ALSource;
|
|
ALfloat value, outsamp;
|
|
ALbufferlistitem *BufferListItem;
|
|
ALint64 DataSize64,DataPos64;
|
|
FILTER *DryFilter, *WetFilter[MAX_SENDS];
|
|
ALfloat WetSend[MAX_SENDS];
|
|
ALuint rampLength;
|
|
ALuint DeviceFreq;
|
|
ALint increment;
|
|
ALuint DataPosInt, DataPosFrac;
|
|
ALuint Channels, Bytes;
|
|
ALuint Frequency;
|
|
ALuint BuffersPlayed;
|
|
ALfloat Pitch;
|
|
ALenum State;
|
|
|
|
if(!(ALSource=ALContext->Source))
|
|
return;
|
|
|
|
DeviceFreq = ALContext->Device->Frequency;
|
|
|
|
rampLength = DeviceFreq * MIN_RAMP_LENGTH / 1000;
|
|
rampLength = max(rampLength, SamplesToDo);
|
|
|
|
another_source:
|
|
State = ALSource->state;
|
|
if(State != AL_PLAYING)
|
|
{
|
|
if((ALSource=ALSource->next) != NULL)
|
|
goto another_source;
|
|
return;
|
|
}
|
|
j = 0;
|
|
|
|
/* Find buffer format */
|
|
Frequency = 0;
|
|
Channels = 0;
|
|
Bytes = 0;
|
|
BufferListItem = ALSource->queue;
|
|
while(BufferListItem != NULL)
|
|
{
|
|
ALbuffer *ALBuffer;
|
|
if((ALBuffer=BufferListItem->buffer) != NULL)
|
|
{
|
|
Channels = aluChannelsFromFormat(ALBuffer->format);
|
|
Bytes = aluBytesFromFormat(ALBuffer->format);
|
|
Frequency = ALBuffer->frequency;
|
|
break;
|
|
}
|
|
BufferListItem = BufferListItem->next;
|
|
}
|
|
|
|
/* Get source info */
|
|
BuffersPlayed = ALSource->BuffersPlayed;
|
|
DataPosInt = ALSource->position;
|
|
DataPosFrac = ALSource->position_fraction;
|
|
|
|
if(ALSource->NeedsUpdate)
|
|
{
|
|
//Only apply 3D calculations for mono buffers
|
|
if(Channels == 1)
|
|
CalcSourceParams(ALContext, ALSource);
|
|
else
|
|
CalcNonAttnSourceParams(ALContext, ALSource);
|
|
ALSource->NeedsUpdate = AL_FALSE;
|
|
}
|
|
|
|
/* Compute 18.14 fixed point step */
|
|
Pitch = (ALSource->Params.Pitch*Frequency) / DeviceFreq;
|
|
if(Pitch > (float)MAX_PITCH) Pitch = (float)MAX_PITCH;
|
|
increment = (ALint)(Pitch*(ALfloat)(1L<<FRACTIONBITS));
|
|
if(increment <= 0) increment = (1<<FRACTIONBITS);
|
|
|
|
/* Compute the gain steps for each output channel */
|
|
if(ALSource->FirstStart)
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] = ALSource->Params.DryGains[i];
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
WetSend[i] = ALSource->Params.WetGains[i];
|
|
}
|
|
else
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] = ALSource->DryGains[i];
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
WetSend[i] = ALSource->WetGains[i];
|
|
}
|
|
|
|
DryFilter = &ALSource->Params.iirFilter;
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
{
|
|
WetFilter[i] = &ALSource->Params.Send[i].iirFilter;
|
|
WetBuffer[i] = (ALSource->Send[i].Slot ?
|
|
ALSource->Send[i].Slot->WetBuffer :
|
|
DummyBuffer);
|
|
}
|
|
|
|
if(DuplicateStereo && Channels == 2)
|
|
{
|
|
Matrix[FRONT_LEFT][SIDE_LEFT] = 1.0f;
|
|
Matrix[FRONT_RIGHT][SIDE_RIGHT] = 1.0f;
|
|
Matrix[FRONT_LEFT][BACK_LEFT] = 1.0f;
|
|
Matrix[FRONT_RIGHT][BACK_RIGHT] = 1.0f;
|
|
}
|
|
else if(DuplicateStereo)
|
|
{
|
|
Matrix[FRONT_LEFT][SIDE_LEFT] = 0.0f;
|
|
Matrix[FRONT_RIGHT][SIDE_RIGHT] = 0.0f;
|
|
Matrix[FRONT_LEFT][BACK_LEFT] = 0.0f;
|
|
Matrix[FRONT_RIGHT][BACK_RIGHT] = 0.0f;
|
|
}
|
|
|
|
/* Get current buffer queue item */
|
|
BufferListItem = ALSource->queue;
|
|
for(i = 0;i < BuffersPlayed && BufferListItem;i++)
|
|
BufferListItem = BufferListItem->next;
|
|
|
|
while(State == AL_PLAYING && j < SamplesToDo)
|
|
{
|
|
ALuint DataSize = 0;
|
|
ALbuffer *ALBuffer;
|
|
ALfloat *Data;
|
|
ALuint BufferSize;
|
|
|
|
/* Get buffer info */
|
|
if((ALBuffer=BufferListItem->buffer) != NULL)
|
|
{
|
|
Data = ALBuffer->data;
|
|
DataSize = ALBuffer->size;
|
|
DataSize /= Channels * Bytes;
|
|
}
|
|
if(DataPosInt >= DataSize)
|
|
goto skipmix;
|
|
|
|
if(BufferListItem->next)
|
|
{
|
|
ALbuffer *NextBuf = BufferListItem->next->buffer;
|
|
if(NextBuf && NextBuf->data)
|
|
{
|
|
ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
|
|
ulExtraSamples = min(NextBuf->size, ulExtraSamples);
|
|
memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
|
|
}
|
|
}
|
|
else if(ALSource->bLooping)
|
|
{
|
|
ALbuffer *NextBuf = ALSource->queue->buffer;
|
|
if(NextBuf && NextBuf->data)
|
|
{
|
|
ALint ulExtraSamples = BUFFER_PADDING*Channels*Bytes;
|
|
ulExtraSamples = min(NextBuf->size, ulExtraSamples);
|
|
memcpy(&Data[DataSize*Channels], NextBuf->data, ulExtraSamples);
|
|
}
|
|
}
|
|
else
|
|
memset(&Data[DataSize*Channels], 0, (BUFFER_PADDING*Channels*Bytes));
|
|
|
|
/* Compute the gain steps for each output channel */
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
dryGainStep[i] = (ALSource->Params.DryGains[i]-
|
|
DrySend[i]) / rampLength;
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
wetGainStep[i] = (ALSource->Params.WetGains[i]-
|
|
WetSend[i]) / rampLength;
|
|
|
|
/* Figure out how many samples we can mix. */
|
|
DataSize64 = DataSize;
|
|
DataSize64 <<= FRACTIONBITS;
|
|
DataPos64 = DataPosInt;
|
|
DataPos64 <<= FRACTIONBITS;
|
|
DataPos64 += DataPosFrac;
|
|
BufferSize = (ALuint)((DataSize64-DataPos64+(increment-1)) / increment);
|
|
|
|
BufferSize = min(BufferSize, (SamplesToDo-j));
|
|
|
|
/* Actual sample mixing loop */
|
|
k = 0;
|
|
Data += DataPosInt*Channels;
|
|
|
|
if(Channels == 1) /* Mono */
|
|
{
|
|
while(BufferSize--)
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] += dryGainStep[i];
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
WetSend[i] += wetGainStep[i];
|
|
|
|
/* First order interpolator */
|
|
value = lerp(Data[k], Data[k+1], DataPosFrac);
|
|
|
|
/* Direct path final mix buffer and panning */
|
|
outsamp = lpFilter4P(DryFilter, 0, value);
|
|
DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT];
|
|
DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT];
|
|
DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT];
|
|
DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT];
|
|
DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT];
|
|
DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT];
|
|
DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER];
|
|
DryBuffer[j][BACK_CENTER] += outsamp*DrySend[BACK_CENTER];
|
|
|
|
/* Room path final mix buffer and panning */
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
{
|
|
outsamp = lpFilter2P(WetFilter[i], 0, value);
|
|
WetBuffer[i][j] += outsamp*WetSend[i];
|
|
}
|
|
|
|
DataPosFrac += increment;
|
|
k += DataPosFrac>>FRACTIONBITS;
|
|
DataPosFrac &= FRACTIONMASK;
|
|
j++;
|
|
}
|
|
}
|
|
else if(Channels == 2) /* Stereo */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT
|
|
};
|
|
const ALfloat scaler = aluSqrt(1.0f/Channels);
|
|
|
|
#define DO_MIX() do { \
|
|
while(BufferSize--) \
|
|
{ \
|
|
for(i = 0;i < OUTPUTCHANNELS;i++) \
|
|
DrySend[i] += dryGainStep[i]; \
|
|
for(i = 0;i < MAX_SENDS;i++) \
|
|
WetSend[i] += wetGainStep[i]; \
|
|
\
|
|
for(i = 0;i < Channels;i++) \
|
|
{ \
|
|
value = lerp(Data[k*Channels + i], Data[(k+1)*Channels + i], DataPosFrac); \
|
|
outsamp = lpFilter2P(DryFilter, chans[i]*2, value)*DrySend[chans[i]]; \
|
|
for(out = 0;out < OUTPUTCHANNELS;out++) \
|
|
DryBuffer[j][out] += outsamp*Matrix[chans[i]][out]; \
|
|
for(out = 0;out < MAX_SENDS;out++) \
|
|
{ \
|
|
outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
|
|
WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
|
|
} \
|
|
} \
|
|
\
|
|
DataPosFrac += increment; \
|
|
k += DataPosFrac>>FRACTIONBITS; \
|
|
DataPosFrac &= FRACTIONMASK; \
|
|
j++; \
|
|
} \
|
|
} while(0)
|
|
|
|
DO_MIX();
|
|
}
|
|
else if(Channels == 4) /* Quad */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
BACK_LEFT, BACK_RIGHT
|
|
};
|
|
const ALfloat scaler = aluSqrt(1.0f/Channels);
|
|
|
|
DO_MIX();
|
|
}
|
|
else if(Channels == 6) /* 5.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_LEFT, BACK_RIGHT
|
|
};
|
|
const ALfloat scaler = aluSqrt(1.0f/Channels);
|
|
|
|
DO_MIX();
|
|
}
|
|
else if(Channels == 7) /* 6.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_CENTER,
|
|
SIDE_LEFT, SIDE_RIGHT
|
|
};
|
|
const ALfloat scaler = aluSqrt(1.0f/Channels);
|
|
|
|
DO_MIX();
|
|
}
|
|
else if(Channels == 8) /* 7.1 */
|
|
{
|
|
const int chans[] = {
|
|
FRONT_LEFT, FRONT_RIGHT,
|
|
FRONT_CENTER, LFE,
|
|
BACK_LEFT, BACK_RIGHT,
|
|
SIDE_LEFT, SIDE_RIGHT
|
|
};
|
|
const ALfloat scaler = aluSqrt(1.0f/Channels);
|
|
|
|
DO_MIX();
|
|
#undef DO_MIX
|
|
}
|
|
else /* Unknown? */
|
|
{
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
DrySend[i] += dryGainStep[i]*BufferSize;
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
WetSend[i] += wetGainStep[i]*BufferSize;
|
|
while(BufferSize--)
|
|
{
|
|
DataPosFrac += increment;
|
|
k += DataPosFrac>>FRACTIONBITS;
|
|
DataPosFrac &= FRACTIONMASK;
|
|
j++;
|
|
}
|
|
}
|
|
DataPosInt += k;
|
|
|
|
skipmix:
|
|
/* Handle looping sources */
|
|
if(DataPosInt >= DataSize)
|
|
{
|
|
if(BuffersPlayed < (ALSource->BuffersInQueue-1))
|
|
{
|
|
BufferListItem = BufferListItem->next;
|
|
BuffersPlayed++;
|
|
DataPosInt -= DataSize;
|
|
}
|
|
else
|
|
{
|
|
if(!ALSource->bLooping)
|
|
{
|
|
State = AL_STOPPED;
|
|
BufferListItem = ALSource->queue;
|
|
BuffersPlayed = ALSource->BuffersInQueue;
|
|
DataPosInt = 0;
|
|
DataPosFrac = 0;
|
|
}
|
|
else
|
|
{
|
|
BufferListItem = ALSource->queue;
|
|
BuffersPlayed = 0;
|
|
if(ALSource->BuffersInQueue == 1)
|
|
DataPosInt %= DataSize;
|
|
else
|
|
DataPosInt -= DataSize;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Update source info */
|
|
ALSource->state = State;
|
|
ALSource->BuffersPlayed = BuffersPlayed;
|
|
ALSource->position = DataPosInt;
|
|
ALSource->position_fraction = DataPosFrac;
|
|
ALSource->Buffer = BufferListItem->buffer;
|
|
|
|
for(i = 0;i < OUTPUTCHANNELS;i++)
|
|
ALSource->DryGains[i] = DrySend[i];
|
|
for(i = 0;i < MAX_SENDS;i++)
|
|
ALSource->WetGains[i] = WetSend[i];
|
|
|
|
ALSource->FirstStart = AL_FALSE;
|
|
|
|
if((ALSource=ALSource->next) != NULL)
|
|
goto another_source;
|
|
}
|
|
|
|
ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
|
|
{
|
|
float (*DryBuffer)[OUTPUTCHANNELS];
|
|
const Channel *ChanMap;
|
|
ALuint SamplesToDo;
|
|
ALeffectslot *ALEffectSlot;
|
|
ALCcontext *ALContext;
|
|
int fpuState;
|
|
ALuint i, c;
|
|
|
|
SuspendContext(NULL);
|
|
|
|
#if defined(HAVE_FESETROUND)
|
|
fpuState = fegetround();
|
|
fesetround(FE_TOWARDZERO);
|
|
#elif defined(HAVE__CONTROLFP)
|
|
fpuState = _controlfp(0, 0);
|
|
_controlfp(_RC_CHOP, _MCW_RC);
|
|
#else
|
|
(void)fpuState;
|
|
#endif
|
|
|
|
DryBuffer = device->DryBuffer;
|
|
while(size > 0)
|
|
{
|
|
/* Setup variables */
|
|
SamplesToDo = min(size, BUFFERSIZE);
|
|
|
|
/* Clear mixing buffer */
|
|
memset(DryBuffer, 0, SamplesToDo*OUTPUTCHANNELS*sizeof(ALfloat));
|
|
|
|
for(c = 0;c < device->NumContexts;c++)
|
|
{
|
|
ALContext = device->Contexts[c];
|
|
SuspendContext(ALContext);
|
|
|
|
MixSomeSources(ALContext, DryBuffer, SamplesToDo);
|
|
|
|
/* effect slot processing */
|
|
ALEffectSlot = ALContext->AuxiliaryEffectSlot;
|
|
while(ALEffectSlot)
|
|
{
|
|
if(ALEffectSlot->EffectState)
|
|
ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot, SamplesToDo, ALEffectSlot->WetBuffer, DryBuffer);
|
|
|
|
for(i = 0;i < SamplesToDo;i++)
|
|
ALEffectSlot->WetBuffer[i] = 0.0f;
|
|
ALEffectSlot = ALEffectSlot->next;
|
|
}
|
|
ProcessContext(ALContext);
|
|
}
|
|
|
|
//Post processing loop
|
|
ChanMap = device->DevChannels;
|
|
switch(device->Format)
|
|
{
|
|
#define CHECK_WRITE_FORMAT(bits, type, func) \
|
|
case AL_FORMAT_MONO##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]] + \
|
|
DryBuffer[i][ChanMap[1]]); \
|
|
buffer = ((type*)buffer) + 1; \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_STEREO##bits: \
|
|
if(device->Bs2b) \
|
|
{ \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
float samples[2]; \
|
|
samples[0] = DryBuffer[i][ChanMap[0]]; \
|
|
samples[1] = DryBuffer[i][ChanMap[1]]; \
|
|
bs2b_cross_feed(device->Bs2b, samples); \
|
|
((type*)buffer)[0] = (func)(samples[0]); \
|
|
((type*)buffer)[1] = (func)(samples[1]); \
|
|
buffer = ((type*)buffer) + 2; \
|
|
} \
|
|
} \
|
|
else \
|
|
{ \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
|
|
((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
|
|
buffer = ((type*)buffer) + 2; \
|
|
} \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_QUAD##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
|
|
((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
|
|
((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
|
|
((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
|
|
buffer = ((type*)buffer) + 4; \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_51CHN##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
|
|
((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
|
|
((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
|
|
((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
|
|
((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
|
|
((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
|
|
buffer = ((type*)buffer) + 6; \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_61CHN##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
|
|
((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
|
|
((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
|
|
((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
|
|
((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
|
|
((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
|
|
((type*)buffer)[6] = (func)(DryBuffer[i][ChanMap[6]]); \
|
|
buffer = ((type*)buffer) + 7; \
|
|
} \
|
|
break; \
|
|
case AL_FORMAT_71CHN##bits: \
|
|
for(i = 0;i < SamplesToDo;i++) \
|
|
{ \
|
|
((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
|
|
((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
|
|
((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
|
|
((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
|
|
((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
|
|
((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
|
|
((type*)buffer)[6] = (func)(DryBuffer[i][ChanMap[6]]); \
|
|
((type*)buffer)[7] = (func)(DryBuffer[i][ChanMap[7]]); \
|
|
buffer = ((type*)buffer) + 8; \
|
|
} \
|
|
break;
|
|
|
|
#define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
|
|
#define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
|
|
CHECK_WRITE_FORMAT(8, ALubyte, aluF2UB)
|
|
CHECK_WRITE_FORMAT(16, ALshort, aluF2S)
|
|
CHECK_WRITE_FORMAT(32, ALfloat, aluF2F)
|
|
#undef AL_FORMAT_STEREO32
|
|
#undef AL_FORMAT_MONO32
|
|
#undef CHECK_WRITE_FORMAT
|
|
|
|
default:
|
|
break;
|
|
}
|
|
|
|
size -= SamplesToDo;
|
|
}
|
|
|
|
#if defined(HAVE_FESETROUND)
|
|
fesetround(fpuState);
|
|
#elif defined(HAVE__CONTROLFP)
|
|
_controlfp(fpuState, 0xfffff);
|
|
#endif
|
|
|
|
ProcessContext(NULL);
|
|
}
|
|
|
|
ALvoid aluHandleDisconnect(ALCdevice *device)
|
|
{
|
|
ALuint i;
|
|
|
|
SuspendContext(NULL);
|
|
for(i = 0;i < device->NumContexts;i++)
|
|
{
|
|
ALsource *source;
|
|
|
|
SuspendContext(device->Contexts[i]);
|
|
|
|
source = device->Contexts[i]->Source;
|
|
while(source)
|
|
{
|
|
if(source->state == AL_PLAYING)
|
|
{
|
|
source->state = AL_STOPPED;
|
|
source->BuffersPlayed = source->BuffersInQueue;
|
|
source->position = 0;
|
|
source->position_fraction = 0;
|
|
}
|
|
source = source->next;
|
|
}
|
|
ProcessContext(device->Contexts[i]);
|
|
}
|
|
|
|
device->Connected = ALC_FALSE;
|
|
ProcessContext(NULL);
|
|
}
|