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1 /**
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2 * OpenAL cross platform audio library
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3 * Copyright (C) 1999-2007 by authors.
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4 * This library is free software; you can redistribute it and/or
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5 * modify it under the terms of the GNU Library General Public
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6 * License as published by the Free Software Foundation; either
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7 * version 2 of the License, or (at your option) any later version.
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8 *
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9 * This library is distributed in the hope that it will be useful,
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10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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12 * Library General Public License for more details.
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13 *
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14 * You should have received a copy of the GNU Library General Public
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15 * License along with this library; if not, write to the
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16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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17 * Boston, MA 02111-1307, USA.
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18 * Or go to http://www.gnu.org/copyleft/lgpl.html
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19 */
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20
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21 #include "config.h"
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22
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23 #include <math.h>
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24 #include <stdlib.h>
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25 #include <string.h>
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26 #include <ctype.h>
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27 #include <assert.h>
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28
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29 #include "alMain.h"
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30 #include "AL/al.h"
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31 #include "AL/alc.h"
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32 #include "alSource.h"
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33 #include "alBuffer.h"
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34 #include "alListener.h"
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35 #include "alAuxEffectSlot.h"
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36 #include "alu.h"
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37 #include "bs2b.h"
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38
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39
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40 static __inline ALvoid aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
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41 {
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42 outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
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43 outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
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44 outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
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45 }
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46
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47 static __inline ALfloat aluDotproduct(const ALfloat *inVector1, const ALfloat *inVector2)
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48 {
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49 return inVector1[0]*inVector2[0] + inVector1[1]*inVector2[1] +
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50 inVector1[2]*inVector2[2];
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51 }
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52
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53 static __inline ALvoid aluNormalize(ALfloat *inVector)
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54 {
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55 ALfloat length, inverse_length;
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56
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57 length = aluSqrt(aluDotproduct(inVector, inVector));
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58 if(length != 0.0f)
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59 {
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60 inverse_length = 1.0f/length;
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61 inVector[0] *= inverse_length;
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62 inVector[1] *= inverse_length;
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63 inVector[2] *= inverse_length;
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64 }
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65 }
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66
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67 static __inline ALvoid aluMatrixVector(ALfloat *vector,ALfloat w,ALfloat matrix[4][4])
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68 {
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69 ALfloat temp[4] = {
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70 vector[0], vector[1], vector[2], w
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71 };
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72
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73 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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74 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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75 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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76 }
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77
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78
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79 ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
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80 {
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81 static const ALfloat angles_Mono[1] = { 0.0f };
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82 static const ALfloat angles_Stereo[2] = { -30.0f, 30.0f };
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83 static const ALfloat angles_Rear[2] = { -150.0f, 150.0f };
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84 static const ALfloat angles_Quad[4] = { -45.0f, 45.0f, -135.0f, 135.0f };
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85 static const ALfloat angles_X51[6] = { -30.0f, 30.0f, 0.0f, 0.0f,
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86 -110.0f, 110.0f };
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87 static const ALfloat angles_X61[7] = { -30.0f, 30.0f, 0.0f, 0.0f,
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88 180.0f, -90.0f, 90.0f };
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89 static const ALfloat angles_X71[8] = { -30.0f, 30.0f, 0.0f, 0.0f,
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90 -110.0f, 110.0f, -90.0f, 90.0f };
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91
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92 static const enum Channel chans_Mono[1] = { FRONT_CENTER };
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93 static const enum Channel chans_Stereo[2] = { FRONT_LEFT, FRONT_RIGHT };
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94 static const enum Channel chans_Rear[2] = { BACK_LEFT, BACK_RIGHT };
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95 static const enum Channel chans_Quad[4] = { FRONT_LEFT, FRONT_RIGHT,
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96 BACK_LEFT, BACK_RIGHT };
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97 static const enum Channel chans_X51[6] = { FRONT_LEFT, FRONT_RIGHT,
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98 FRONT_CENTER, LFE,
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99 BACK_LEFT, BACK_RIGHT };
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100 static const enum Channel chans_X61[7] = { FRONT_LEFT, FRONT_RIGHT,
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101 FRONT_CENTER, LFE, BACK_CENTER,
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102 SIDE_LEFT, SIDE_RIGHT };
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103 static const enum Channel chans_X71[8] = { FRONT_LEFT, FRONT_RIGHT,
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104 FRONT_CENTER, LFE,
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105 BACK_LEFT, BACK_RIGHT,
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106 SIDE_LEFT, SIDE_RIGHT };
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107
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108 ALCdevice *Device = ALContext->Device;
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109 ALfloat SourceVolume,ListenerGain,MinVolume,MaxVolume;
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110 ALbufferlistitem *BufferListItem;
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111 enum DevFmtChannels DevChans;
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112 enum FmtChannels Channels;
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113 ALfloat (*SrcMatrix)[MAXCHANNELS];
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114 ALfloat DryGain, DryGainHF;
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115 ALfloat WetGain[MAX_SENDS];
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116 ALfloat WetGainHF[MAX_SENDS];
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117 ALint NumSends, Frequency;
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118 const ALfloat *SpeakerGain;
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119 const ALfloat *angles = NULL;
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120 const enum Channel *chans = NULL;
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121 enum Resampler Resampler;
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122 ALint num_channels = 0;
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123 ALboolean VirtualChannels;
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124 ALfloat Pitch;
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125 ALfloat cw;
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126 ALuint pos;
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127 ALint i, c;
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128
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129 /* Get device properties */
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130 DevChans = ALContext->Device->FmtChans;
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131 NumSends = ALContext->Device->NumAuxSends;
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132 Frequency = ALContext->Device->Frequency;
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133
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134 /* Get listener properties */
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135 ListenerGain = ALContext->Listener.Gain;
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136
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137 /* Get source properties */
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138 SourceVolume = ALSource->flGain;
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139 MinVolume = ALSource->flMinGain;
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140 MaxVolume = ALSource->flMaxGain;
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141 Pitch = ALSource->flPitch;
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142 Resampler = ALSource->Resampler;
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143 VirtualChannels = ALSource->VirtualChannels;
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144
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145 /* Calculate the stepping value */
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146 Channels = FmtMono;
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147 BufferListItem = ALSource->queue;
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148 while(BufferListItem != NULL)
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149 {
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150 ALbuffer *ALBuffer;
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151 if((ALBuffer=BufferListItem->buffer) != NULL)
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152 {
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153 ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
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154 ALSource->SampleSize;
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155 maxstep -= ResamplerPadding[Resampler] +
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156 ResamplerPrePadding[Resampler] + 1;
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157 maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
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158
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159 Pitch = Pitch * ALBuffer->Frequency / Frequency;
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160 if(Pitch > (ALfloat)maxstep)
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161 ALSource->Params.Step = maxstep<<FRACTIONBITS;
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162 else
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163 {
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164 ALSource->Params.Step = Pitch*FRACTIONONE;
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165 if(ALSource->Params.Step == 0)
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166 ALSource->Params.Step = 1;
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167 }
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168
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169 Channels = ALBuffer->FmtChannels;
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170
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171 if(ALSource->VirtualChannels && (Device->Flags&DEVICE_USE_HRTF))
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172 ALSource->Params.DoMix = SelectHrtfMixer(ALBuffer,
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173 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
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174 Resampler);
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175 else
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176 ALSource->Params.DoMix = SelectMixer(ALBuffer,
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177 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
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178 Resampler);
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179 break;
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180 }
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181 BufferListItem = BufferListItem->next;
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182 }
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183
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184 /* Calculate gains */
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185 DryGain = clampf(SourceVolume, MinVolume, MaxVolume);
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186 DryGainHF = 1.0f;
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187 switch(ALSource->DirectFilter.type)
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188 {
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189 case AL_FILTER_LOWPASS:
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190 DryGain *= ALSource->DirectFilter.Gain;
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191 DryGainHF *= ALSource->DirectFilter.GainHF;
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192 break;
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193 }
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194 for(i = 0;i < NumSends;i++)
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195 {
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196 WetGain[i] = clampf(SourceVolume, MinVolume, MaxVolume);
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197 WetGainHF[i] = 1.0f;
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198 switch(ALSource->Send[i].WetFilter.type)
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199 {
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200 case AL_FILTER_LOWPASS:
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201 WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
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202 WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
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203 break;
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204 }
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205 }
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206
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207 SrcMatrix = ALSource->Params.DryGains;
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208 for(i = 0;i < MAXCHANNELS;i++)
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209 {
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210 for(c = 0;c < MAXCHANNELS;c++)
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211 SrcMatrix[i][c] = 0.0f;
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212 }
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213 switch(Channels)
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214 {
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215 case FmtMono:
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216 angles = angles_Mono;
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217 chans = chans_Mono;
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218 num_channels = 1;
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219 break;
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220 case FmtStereo:
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221 if(VirtualChannels && (ALContext->Device->Flags&DEVICE_DUPLICATE_STEREO))
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222 {
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223 DryGain *= aluSqrt(2.0f/4.0f);
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224 for(c = 0;c < 2;c++)
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225 {
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226 pos = aluCart2LUTpos(cos(angles_Rear[c] * (M_PI/180.0)),
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227 sin(angles_Rear[c] * (M_PI/180.0)));
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228 SpeakerGain = Device->PanningLUT[pos];
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229
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230 for(i = 0;i < (ALint)Device->NumChan;i++)
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231 {
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232 enum Channel chan = Device->Speaker2Chan[i];
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233 SrcMatrix[c][chan] += DryGain * ListenerGain *
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234 SpeakerGain[chan];
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235 }
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236 }
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237 }
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238 angles = angles_Stereo;
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239 chans = chans_Stereo;
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240 num_channels = 2;
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241 break;
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242
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243 case FmtRear:
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244 angles = angles_Rear;
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245 chans = chans_Rear;
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246 num_channels = 2;
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247 break;
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248
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249 case FmtQuad:
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250 angles = angles_Quad;
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251 chans = chans_Quad;
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252 num_channels = 4;
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253 break;
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254
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255 case FmtX51:
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256 angles = angles_X51;
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257 chans = chans_X51;
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258 num_channels = 6;
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259 break;
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260
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261 case FmtX61:
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262 angles = angles_X61;
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263 chans = chans_X61;
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264 num_channels = 7;
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265 break;
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266
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267 case FmtX71:
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268 angles = angles_X71;
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269 chans = chans_X71;
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270 num_channels = 8;
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271 break;
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272 }
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273
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274 if(VirtualChannels == AL_FALSE)
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275 {
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276 for(c = 0;c < num_channels;c++)
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277 SrcMatrix[c][chans[c]] += DryGain * ListenerGain;
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278 }
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279 else if((Device->Flags&DEVICE_USE_HRTF))
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280 {
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281 for(c = 0;c < num_channels;c++)
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282 {
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283 if(chans[c] == LFE)
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284 {
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285 /* Skip LFE */
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286 ALSource->Params.HrtfDelay[c][0] = 0;
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287 ALSource->Params.HrtfDelay[c][1] = 0;
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288 for(i = 0;i < HRIR_LENGTH;i++)
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289 {
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290 ALSource->Params.HrtfCoeffs[c][i][0] = 0.0f;
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291 ALSource->Params.HrtfCoeffs[c][i][1] = 0.0f;
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292 }
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293 }
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294 else
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295 {
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296 /* Get the static HRIR coefficients and delays for this
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297 * channel. */
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298 GetLerpedHrtfCoeffs(0.0, angles[c] * (M_PI/180.0),
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299 DryGain*ListenerGain,
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300 ALSource->Params.HrtfCoeffs[c],
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301 ALSource->Params.HrtfDelay[c]);
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302 }
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303 ALSource->HrtfCounter = 0;
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304 }
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305 }
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306 else
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307 {
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308 for(c = 0;c < num_channels;c++)
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309 {
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310 if(chans[c] == LFE) /* Special-case LFE */
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311 {
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312 SrcMatrix[c][LFE] += DryGain * ListenerGain;
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313 continue;
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314 }
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315 pos = aluCart2LUTpos(cos(angles[c] * (M_PI/180.0)),
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316 sin(angles[c] * (M_PI/180.0)));
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317 SpeakerGain = Device->PanningLUT[pos];
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318
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319 for(i = 0;i < (ALint)Device->NumChan;i++)
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320 {
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321 enum Channel chan = Device->Speaker2Chan[i];
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322 SrcMatrix[c][chan] += DryGain * ListenerGain *
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323 SpeakerGain[chan];
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rlm@0
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324 }
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325 }
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326 }
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327 for(i = 0;i < NumSends;i++)
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328 {
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329 ALSource->Params.Send[i].Slot = ALSource->Send[i].Slot;
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330 ALSource->Params.Send[i].WetGain = WetGain[i] * ListenerGain;
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331 }
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rlm@0
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332
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rlm@0
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333 /* Update filter coefficients. Calculations based on the I3DL2
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334 * spec. */
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|
335 cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
|
rlm@0
|
336
|
rlm@0
|
337 /* We use two chained one-pole filters, so we need to take the
|
rlm@0
|
338 * square root of the squared gain, which is the same as the base
|
rlm@0
|
339 * gain. */
|
rlm@0
|
340 ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
|
rlm@0
|
341 for(i = 0;i < NumSends;i++)
|
rlm@0
|
342 {
|
rlm@0
|
343 /* We use a one-pole filter, so we need to take the squared gain */
|
rlm@0
|
344 ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
|
rlm@0
|
345 ALSource->Params.Send[i].iirFilter.coeff = a;
|
rlm@0
|
346 }
|
rlm@0
|
347 }
|
rlm@0
|
348
|
rlm@0
|
349 ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
|
rlm@0
|
350 {
|
rlm@0
|
351 const ALCdevice *Device = ALContext->Device;
|
rlm@0
|
352 ALfloat InnerAngle,OuterAngle,Angle,Distance,ClampedDist;
|
rlm@0
|
353 ALfloat Direction[3],Position[3],SourceToListener[3];
|
rlm@0
|
354 ALfloat Velocity[3],ListenerVel[3];
|
rlm@0
|
355 ALfloat MinVolume,MaxVolume,MinDist,MaxDist,Rolloff;
|
rlm@0
|
356 ALfloat ConeVolume,ConeHF,SourceVolume,ListenerGain;
|
rlm@0
|
357 ALfloat DopplerFactor, DopplerVelocity, SpeedOfSound;
|
rlm@0
|
358 ALfloat AirAbsorptionFactor;
|
rlm@0
|
359 ALfloat RoomAirAbsorption[MAX_SENDS];
|
rlm@0
|
360 ALbufferlistitem *BufferListItem;
|
rlm@0
|
361 ALfloat Attenuation, EffectiveDist;
|
rlm@0
|
362 ALfloat RoomAttenuation[MAX_SENDS];
|
rlm@0
|
363 ALfloat MetersPerUnit;
|
rlm@0
|
364 ALfloat RoomRolloffBase;
|
rlm@0
|
365 ALfloat RoomRolloff[MAX_SENDS];
|
rlm@0
|
366 ALfloat DecayDistance[MAX_SENDS];
|
rlm@0
|
367 ALfloat DryGain;
|
rlm@0
|
368 ALfloat DryGainHF;
|
rlm@0
|
369 ALboolean DryGainHFAuto;
|
rlm@0
|
370 ALfloat WetGain[MAX_SENDS];
|
rlm@0
|
371 ALfloat WetGainHF[MAX_SENDS];
|
rlm@0
|
372 ALboolean WetGainAuto;
|
rlm@0
|
373 ALboolean WetGainHFAuto;
|
rlm@0
|
374 enum Resampler Resampler;
|
rlm@0
|
375 ALfloat Pitch;
|
rlm@0
|
376 ALuint Frequency;
|
rlm@0
|
377 ALint NumSends;
|
rlm@0
|
378 ALfloat cw;
|
rlm@0
|
379 ALint i;
|
rlm@0
|
380
|
rlm@0
|
381 DryGainHF = 1.0f;
|
rlm@0
|
382 for(i = 0;i < MAX_SENDS;i++)
|
rlm@0
|
383 WetGainHF[i] = 1.0f;
|
rlm@0
|
384
|
rlm@0
|
385 //Get context properties
|
rlm@0
|
386 DopplerFactor = ALContext->DopplerFactor * ALSource->DopplerFactor;
|
rlm@0
|
387 DopplerVelocity = ALContext->DopplerVelocity;
|
rlm@0
|
388 SpeedOfSound = ALContext->flSpeedOfSound;
|
rlm@0
|
389 NumSends = Device->NumAuxSends;
|
rlm@0
|
390 Frequency = Device->Frequency;
|
rlm@0
|
391
|
rlm@0
|
392 //Get listener properties
|
rlm@0
|
393 ListenerGain = ALContext->Listener.Gain;
|
rlm@0
|
394 MetersPerUnit = ALContext->Listener.MetersPerUnit;
|
rlm@0
|
395 memcpy(ListenerVel, ALContext->Listener.Velocity, sizeof(ALContext->Listener.Velocity));
|
rlm@0
|
396
|
rlm@0
|
397 //Get source properties
|
rlm@0
|
398 SourceVolume = ALSource->flGain;
|
rlm@0
|
399 MinVolume = ALSource->flMinGain;
|
rlm@0
|
400 MaxVolume = ALSource->flMaxGain;
|
rlm@0
|
401 Pitch = ALSource->flPitch;
|
rlm@0
|
402 Resampler = ALSource->Resampler;
|
rlm@0
|
403 memcpy(Position, ALSource->vPosition, sizeof(ALSource->vPosition));
|
rlm@0
|
404 memcpy(Direction, ALSource->vOrientation, sizeof(ALSource->vOrientation));
|
rlm@0
|
405 memcpy(Velocity, ALSource->vVelocity, sizeof(ALSource->vVelocity));
|
rlm@0
|
406 MinDist = ALSource->flRefDistance;
|
rlm@0
|
407 MaxDist = ALSource->flMaxDistance;
|
rlm@0
|
408 Rolloff = ALSource->flRollOffFactor;
|
rlm@0
|
409 InnerAngle = ALSource->flInnerAngle * ConeScale;
|
rlm@0
|
410 OuterAngle = ALSource->flOuterAngle * ConeScale;
|
rlm@0
|
411 AirAbsorptionFactor = ALSource->AirAbsorptionFactor;
|
rlm@0
|
412 DryGainHFAuto = ALSource->DryGainHFAuto;
|
rlm@0
|
413 WetGainAuto = ALSource->WetGainAuto;
|
rlm@0
|
414 WetGainHFAuto = ALSource->WetGainHFAuto;
|
rlm@0
|
415 RoomRolloffBase = ALSource->RoomRolloffFactor;
|
rlm@0
|
416 for(i = 0;i < NumSends;i++)
|
rlm@0
|
417 {
|
rlm@0
|
418 ALeffectslot *Slot = ALSource->Send[i].Slot;
|
rlm@0
|
419
|
rlm@0
|
420 if(!Slot || Slot->effect.type == AL_EFFECT_NULL)
|
rlm@0
|
421 {
|
rlm@0
|
422 RoomRolloff[i] = 0.0f;
|
rlm@0
|
423 DecayDistance[i] = 0.0f;
|
rlm@0
|
424 RoomAirAbsorption[i] = 1.0f;
|
rlm@0
|
425 }
|
rlm@0
|
426 else if(Slot->AuxSendAuto)
|
rlm@0
|
427 {
|
rlm@0
|
428 RoomRolloff[i] = RoomRolloffBase;
|
rlm@0
|
429 if(IsReverbEffect(Slot->effect.type))
|
rlm@0
|
430 {
|
rlm@0
|
431 RoomRolloff[i] += Slot->effect.Params.Reverb.RoomRolloffFactor;
|
rlm@0
|
432 DecayDistance[i] = Slot->effect.Params.Reverb.DecayTime *
|
rlm@0
|
433 SPEEDOFSOUNDMETRESPERSEC;
|
rlm@0
|
434 RoomAirAbsorption[i] = Slot->effect.Params.Reverb.AirAbsorptionGainHF;
|
rlm@0
|
435 }
|
rlm@0
|
436 else
|
rlm@0
|
437 {
|
rlm@0
|
438 DecayDistance[i] = 0.0f;
|
rlm@0
|
439 RoomAirAbsorption[i] = 1.0f;
|
rlm@0
|
440 }
|
rlm@0
|
441 }
|
rlm@0
|
442 else
|
rlm@0
|
443 {
|
rlm@0
|
444 /* If the slot's auxiliary send auto is off, the data sent to the
|
rlm@0
|
445 * effect slot is the same as the dry path, sans filter effects */
|
rlm@0
|
446 RoomRolloff[i] = Rolloff;
|
rlm@0
|
447 DecayDistance[i] = 0.0f;
|
rlm@0
|
448 RoomAirAbsorption[i] = AIRABSORBGAINHF;
|
rlm@0
|
449 }
|
rlm@0
|
450
|
rlm@0
|
451 ALSource->Params.Send[i].Slot = Slot;
|
rlm@0
|
452 }
|
rlm@0
|
453
|
rlm@0
|
454 //1. Translate Listener to origin (convert to head relative)
|
rlm@0
|
455 if(ALSource->bHeadRelative == AL_FALSE)
|
rlm@0
|
456 {
|
rlm@0
|
457 ALfloat U[3],V[3],N[3];
|
rlm@0
|
458 ALfloat Matrix[4][4];
|
rlm@0
|
459
|
rlm@0
|
460 // Build transform matrix
|
rlm@0
|
461 memcpy(N, ALContext->Listener.Forward, sizeof(N)); // At-vector
|
rlm@0
|
462 aluNormalize(N); // Normalized At-vector
|
rlm@0
|
463 memcpy(V, ALContext->Listener.Up, sizeof(V)); // Up-vector
|
rlm@0
|
464 aluNormalize(V); // Normalized Up-vector
|
rlm@0
|
465 aluCrossproduct(N, V, U); // Right-vector
|
rlm@0
|
466 aluNormalize(U); // Normalized Right-vector
|
rlm@0
|
467 Matrix[0][0] = U[0]; Matrix[0][1] = V[0]; Matrix[0][2] = -N[0]; Matrix[0][3] = 0.0f;
|
rlm@0
|
468 Matrix[1][0] = U[1]; Matrix[1][1] = V[1]; Matrix[1][2] = -N[1]; Matrix[1][3] = 0.0f;
|
rlm@0
|
469 Matrix[2][0] = U[2]; Matrix[2][1] = V[2]; Matrix[2][2] = -N[2]; Matrix[2][3] = 0.0f;
|
rlm@0
|
470 Matrix[3][0] = 0.0f; Matrix[3][1] = 0.0f; Matrix[3][2] = 0.0f; Matrix[3][3] = 1.0f;
|
rlm@0
|
471
|
rlm@0
|
472 // Translate position
|
rlm@0
|
473 Position[0] -= ALContext->Listener.Position[0];
|
rlm@0
|
474 Position[1] -= ALContext->Listener.Position[1];
|
rlm@0
|
475 Position[2] -= ALContext->Listener.Position[2];
|
rlm@0
|
476
|
rlm@0
|
477 // Transform source position and direction into listener space
|
rlm@0
|
478 aluMatrixVector(Position, 1.0f, Matrix);
|
rlm@0
|
479 aluMatrixVector(Direction, 0.0f, Matrix);
|
rlm@0
|
480 // Transform source and listener velocity into listener space
|
rlm@0
|
481 aluMatrixVector(Velocity, 0.0f, Matrix);
|
rlm@0
|
482 aluMatrixVector(ListenerVel, 0.0f, Matrix);
|
rlm@0
|
483 }
|
rlm@0
|
484 else
|
rlm@0
|
485 ListenerVel[0] = ListenerVel[1] = ListenerVel[2] = 0.0f;
|
rlm@0
|
486
|
rlm@0
|
487 SourceToListener[0] = -Position[0];
|
rlm@0
|
488 SourceToListener[1] = -Position[1];
|
rlm@0
|
489 SourceToListener[2] = -Position[2];
|
rlm@0
|
490 aluNormalize(SourceToListener);
|
rlm@0
|
491 aluNormalize(Direction);
|
rlm@0
|
492
|
rlm@0
|
493 //2. Calculate distance attenuation
|
rlm@0
|
494 Distance = aluSqrt(aluDotproduct(Position, Position));
|
rlm@0
|
495 ClampedDist = Distance;
|
rlm@0
|
496
|
rlm@0
|
497 Attenuation = 1.0f;
|
rlm@0
|
498 for(i = 0;i < NumSends;i++)
|
rlm@0
|
499 RoomAttenuation[i] = 1.0f;
|
rlm@0
|
500 switch(ALContext->SourceDistanceModel ? ALSource->DistanceModel :
|
rlm@0
|
501 ALContext->DistanceModel)
|
rlm@0
|
502 {
|
rlm@0
|
503 case InverseDistanceClamped:
|
rlm@0
|
504 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
|
rlm@0
|
505 if(MaxDist < MinDist)
|
rlm@0
|
506 break;
|
rlm@0
|
507 //fall-through
|
rlm@0
|
508 case InverseDistance:
|
rlm@0
|
509 if(MinDist > 0.0f)
|
rlm@0
|
510 {
|
rlm@0
|
511 if((MinDist + (Rolloff * (ClampedDist - MinDist))) > 0.0f)
|
rlm@0
|
512 Attenuation = MinDist / (MinDist + (Rolloff * (ClampedDist - MinDist)));
|
rlm@0
|
513 for(i = 0;i < NumSends;i++)
|
rlm@0
|
514 {
|
rlm@0
|
515 if((MinDist + (RoomRolloff[i] * (ClampedDist - MinDist))) > 0.0f)
|
rlm@0
|
516 RoomAttenuation[i] = MinDist / (MinDist + (RoomRolloff[i] * (ClampedDist - MinDist)));
|
rlm@0
|
517 }
|
rlm@0
|
518 }
|
rlm@0
|
519 break;
|
rlm@0
|
520
|
rlm@0
|
521 case LinearDistanceClamped:
|
rlm@0
|
522 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
|
rlm@0
|
523 if(MaxDist < MinDist)
|
rlm@0
|
524 break;
|
rlm@0
|
525 //fall-through
|
rlm@0
|
526 case LinearDistance:
|
rlm@0
|
527 if(MaxDist != MinDist)
|
rlm@0
|
528 {
|
rlm@0
|
529 Attenuation = 1.0f - (Rolloff*(ClampedDist-MinDist)/(MaxDist - MinDist));
|
rlm@0
|
530 Attenuation = maxf(Attenuation, 0.0f);
|
rlm@0
|
531 for(i = 0;i < NumSends;i++)
|
rlm@0
|
532 {
|
rlm@0
|
533 RoomAttenuation[i] = 1.0f - (RoomRolloff[i]*(ClampedDist-MinDist)/(MaxDist - MinDist));
|
rlm@0
|
534 RoomAttenuation[i] = maxf(RoomAttenuation[i], 0.0f);
|
rlm@0
|
535 }
|
rlm@0
|
536 }
|
rlm@0
|
537 break;
|
rlm@0
|
538
|
rlm@0
|
539 case ExponentDistanceClamped:
|
rlm@0
|
540 ClampedDist = clampf(ClampedDist, MinDist, MaxDist);
|
rlm@0
|
541 if(MaxDist < MinDist)
|
rlm@0
|
542 break;
|
rlm@0
|
543 //fall-through
|
rlm@0
|
544 case ExponentDistance:
|
rlm@0
|
545 if(ClampedDist > 0.0f && MinDist > 0.0f)
|
rlm@0
|
546 {
|
rlm@0
|
547 Attenuation = aluPow(ClampedDist/MinDist, -Rolloff);
|
rlm@0
|
548 for(i = 0;i < NumSends;i++)
|
rlm@0
|
549 RoomAttenuation[i] = aluPow(ClampedDist/MinDist, -RoomRolloff[i]);
|
rlm@0
|
550 }
|
rlm@0
|
551 break;
|
rlm@0
|
552
|
rlm@0
|
553 case DisableDistance:
|
rlm@0
|
554 break;
|
rlm@0
|
555 }
|
rlm@0
|
556
|
rlm@0
|
557 // Source Gain + Attenuation
|
rlm@0
|
558 DryGain = SourceVolume * Attenuation;
|
rlm@0
|
559 for(i = 0;i < NumSends;i++)
|
rlm@0
|
560 WetGain[i] = SourceVolume * RoomAttenuation[i];
|
rlm@0
|
561
|
rlm@0
|
562 // Distance-based air absorption
|
rlm@0
|
563 EffectiveDist = 0.0f;
|
rlm@0
|
564 if(MinDist > 0.0f && Attenuation < 1.0f)
|
rlm@0
|
565 EffectiveDist = (MinDist/Attenuation - MinDist)*MetersPerUnit;
|
rlm@0
|
566 if(AirAbsorptionFactor > 0.0f && EffectiveDist > 0.0f)
|
rlm@0
|
567 {
|
rlm@0
|
568 DryGainHF *= aluPow(AIRABSORBGAINHF, AirAbsorptionFactor*EffectiveDist);
|
rlm@0
|
569 for(i = 0;i < NumSends;i++)
|
rlm@0
|
570 WetGainHF[i] *= aluPow(RoomAirAbsorption[i],
|
rlm@0
|
571 AirAbsorptionFactor*EffectiveDist);
|
rlm@0
|
572 }
|
rlm@0
|
573
|
rlm@0
|
574 //3. Apply directional soundcones
|
rlm@0
|
575 Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * (180.0/M_PI);
|
rlm@0
|
576 if(Angle >= InnerAngle && Angle <= OuterAngle)
|
rlm@0
|
577 {
|
rlm@0
|
578 ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
|
rlm@0
|
579 ConeVolume = lerp(1.0, ALSource->flOuterGain, scale);
|
rlm@0
|
580 ConeHF = lerp(1.0, ALSource->OuterGainHF, scale);
|
rlm@0
|
581 }
|
rlm@0
|
582 else if(Angle > OuterAngle)
|
rlm@0
|
583 {
|
rlm@0
|
584 ConeVolume = ALSource->flOuterGain;
|
rlm@0
|
585 ConeHF = ALSource->OuterGainHF;
|
rlm@0
|
586 }
|
rlm@0
|
587 else
|
rlm@0
|
588 {
|
rlm@0
|
589 ConeVolume = 1.0f;
|
rlm@0
|
590 ConeHF = 1.0f;
|
rlm@0
|
591 }
|
rlm@0
|
592
|
rlm@0
|
593 DryGain *= ConeVolume;
|
rlm@0
|
594 if(WetGainAuto)
|
rlm@0
|
595 {
|
rlm@0
|
596 for(i = 0;i < NumSends;i++)
|
rlm@0
|
597 WetGain[i] *= ConeVolume;
|
rlm@0
|
598 }
|
rlm@0
|
599 if(DryGainHFAuto)
|
rlm@0
|
600 DryGainHF *= ConeHF;
|
rlm@0
|
601 if(WetGainHFAuto)
|
rlm@0
|
602 {
|
rlm@0
|
603 for(i = 0;i < NumSends;i++)
|
rlm@0
|
604 WetGainHF[i] *= ConeHF;
|
rlm@0
|
605 }
|
rlm@0
|
606
|
rlm@0
|
607 // Clamp to Min/Max Gain
|
rlm@0
|
608 DryGain = clampf(DryGain, MinVolume, MaxVolume);
|
rlm@0
|
609 for(i = 0;i < NumSends;i++)
|
rlm@0
|
610 WetGain[i] = clampf(WetGain[i], MinVolume, MaxVolume);
|
rlm@0
|
611
|
rlm@0
|
612 // Apply filter gains and filters
|
rlm@0
|
613 switch(ALSource->DirectFilter.type)
|
rlm@0
|
614 {
|
rlm@0
|
615 case AL_FILTER_LOWPASS:
|
rlm@0
|
616 DryGain *= ALSource->DirectFilter.Gain;
|
rlm@0
|
617 DryGainHF *= ALSource->DirectFilter.GainHF;
|
rlm@0
|
618 break;
|
rlm@0
|
619 }
|
rlm@0
|
620 DryGain *= ListenerGain;
|
rlm@0
|
621 for(i = 0;i < NumSends;i++)
|
rlm@0
|
622 {
|
rlm@0
|
623 switch(ALSource->Send[i].WetFilter.type)
|
rlm@0
|
624 {
|
rlm@0
|
625 case AL_FILTER_LOWPASS:
|
rlm@0
|
626 WetGain[i] *= ALSource->Send[i].WetFilter.Gain;
|
rlm@0
|
627 WetGainHF[i] *= ALSource->Send[i].WetFilter.GainHF;
|
rlm@0
|
628 break;
|
rlm@0
|
629 }
|
rlm@0
|
630 WetGain[i] *= ListenerGain;
|
rlm@0
|
631 }
|
rlm@0
|
632
|
rlm@0
|
633 if(WetGainAuto)
|
rlm@0
|
634 {
|
rlm@0
|
635 /* Apply a decay-time transformation to the wet path, based on the
|
rlm@0
|
636 * attenuation of the dry path.
|
rlm@0
|
637 *
|
rlm@0
|
638 * Using the approximate (effective) source to listener distance, the
|
rlm@0
|
639 * initial decay of the reverb effect is calculated and applied to the
|
rlm@0
|
640 * wet path.
|
rlm@0
|
641 */
|
rlm@0
|
642 for(i = 0;i < NumSends;i++)
|
rlm@0
|
643 {
|
rlm@0
|
644 if(DecayDistance[i] > 0.0f)
|
rlm@0
|
645 WetGain[i] *= aluPow(0.001f /* -60dB */,
|
rlm@0
|
646 EffectiveDist / DecayDistance[i]);
|
rlm@0
|
647 }
|
rlm@0
|
648 }
|
rlm@0
|
649
|
rlm@0
|
650 // Calculate Velocity
|
rlm@0
|
651 if(DopplerFactor != 0.0f)
|
rlm@0
|
652 {
|
rlm@0
|
653 ALfloat VSS, VLS;
|
rlm@0
|
654 ALfloat MaxVelocity = (SpeedOfSound*DopplerVelocity) /
|
rlm@0
|
655 DopplerFactor;
|
rlm@0
|
656
|
rlm@0
|
657 VSS = aluDotproduct(Velocity, SourceToListener);
|
rlm@0
|
658 if(VSS >= MaxVelocity)
|
rlm@0
|
659 VSS = (MaxVelocity - 1.0f);
|
rlm@0
|
660 else if(VSS <= -MaxVelocity)
|
rlm@0
|
661 VSS = -MaxVelocity + 1.0f;
|
rlm@0
|
662
|
rlm@0
|
663 VLS = aluDotproduct(ListenerVel, SourceToListener);
|
rlm@0
|
664 if(VLS >= MaxVelocity)
|
rlm@0
|
665 VLS = (MaxVelocity - 1.0f);
|
rlm@0
|
666 else if(VLS <= -MaxVelocity)
|
rlm@0
|
667 VLS = -MaxVelocity + 1.0f;
|
rlm@0
|
668
|
rlm@0
|
669 Pitch *= ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VLS)) /
|
rlm@0
|
670 ((SpeedOfSound*DopplerVelocity) - (DopplerFactor*VSS));
|
rlm@0
|
671 }
|
rlm@0
|
672
|
rlm@0
|
673 BufferListItem = ALSource->queue;
|
rlm@0
|
674 while(BufferListItem != NULL)
|
rlm@0
|
675 {
|
rlm@0
|
676 ALbuffer *ALBuffer;
|
rlm@0
|
677 if((ALBuffer=BufferListItem->buffer) != NULL)
|
rlm@0
|
678 {
|
rlm@0
|
679 ALint maxstep = STACK_DATA_SIZE / ALSource->NumChannels /
|
rlm@0
|
680 ALSource->SampleSize;
|
rlm@0
|
681 maxstep -= ResamplerPadding[Resampler] +
|
rlm@0
|
682 ResamplerPrePadding[Resampler] + 1;
|
rlm@0
|
683 maxstep = mini(maxstep, INT_MAX>>FRACTIONBITS);
|
rlm@0
|
684
|
rlm@0
|
685 Pitch = Pitch * ALBuffer->Frequency / Frequency;
|
rlm@0
|
686 if(Pitch > (ALfloat)maxstep)
|
rlm@0
|
687 ALSource->Params.Step = maxstep<<FRACTIONBITS;
|
rlm@0
|
688 else
|
rlm@0
|
689 {
|
rlm@0
|
690 ALSource->Params.Step = Pitch*FRACTIONONE;
|
rlm@0
|
691 if(ALSource->Params.Step == 0)
|
rlm@0
|
692 ALSource->Params.Step = 1;
|
rlm@0
|
693 }
|
rlm@0
|
694
|
rlm@0
|
695 if((Device->Flags&DEVICE_USE_HRTF))
|
rlm@0
|
696 ALSource->Params.DoMix = SelectHrtfMixer(ALBuffer,
|
rlm@0
|
697 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
|
rlm@0
|
698 Resampler);
|
rlm@0
|
699 else
|
rlm@0
|
700 ALSource->Params.DoMix = SelectMixer(ALBuffer,
|
rlm@0
|
701 (ALSource->Params.Step==FRACTIONONE) ? POINT_RESAMPLER :
|
rlm@0
|
702 Resampler);
|
rlm@0
|
703 break;
|
rlm@0
|
704 }
|
rlm@0
|
705 BufferListItem = BufferListItem->next;
|
rlm@0
|
706 }
|
rlm@0
|
707
|
rlm@0
|
708 if((Device->Flags&DEVICE_USE_HRTF))
|
rlm@0
|
709 {
|
rlm@0
|
710 // Use a binaural HRTF algorithm for stereo headphone playback
|
rlm@0
|
711 ALfloat delta, ev = 0.0f, az = 0.0f;
|
rlm@0
|
712
|
rlm@0
|
713 if(Distance > 0.0f)
|
rlm@0
|
714 {
|
rlm@0
|
715 ALfloat invlen = 1.0f/Distance;
|
rlm@0
|
716 Position[0] *= invlen;
|
rlm@0
|
717 Position[1] *= invlen;
|
rlm@0
|
718 Position[2] *= invlen;
|
rlm@0
|
719
|
rlm@0
|
720 // Calculate elevation and azimuth only when the source is not at
|
rlm@0
|
721 // the listener. This prevents +0 and -0 Z from producing
|
rlm@0
|
722 // inconsistent panning.
|
rlm@0
|
723 ev = asin(Position[1]);
|
rlm@0
|
724 az = atan2(Position[0], -Position[2]*ZScale);
|
rlm@0
|
725 }
|
rlm@0
|
726
|
rlm@0
|
727 // Check to see if the HRIR is already moving.
|
rlm@0
|
728 if(ALSource->HrtfMoving)
|
rlm@0
|
729 {
|
rlm@0
|
730 // Calculate the normalized HRTF transition factor (delta).
|
rlm@0
|
731 delta = CalcHrtfDelta(ALSource->Params.HrtfGain, DryGain,
|
rlm@0
|
732 ALSource->Params.HrtfDir, Position);
|
rlm@0
|
733 // If the delta is large enough, get the moving HRIR target
|
rlm@0
|
734 // coefficients, target delays, steppping values, and counter.
|
rlm@0
|
735 if(delta > 0.001f)
|
rlm@0
|
736 {
|
rlm@0
|
737 ALSource->HrtfCounter = GetMovingHrtfCoeffs(ev, az, DryGain,
|
rlm@0
|
738 delta, ALSource->HrtfCounter,
|
rlm@0
|
739 ALSource->Params.HrtfCoeffs[0],
|
rlm@0
|
740 ALSource->Params.HrtfDelay[0],
|
rlm@0
|
741 ALSource->Params.HrtfCoeffStep,
|
rlm@0
|
742 ALSource->Params.HrtfDelayStep);
|
rlm@0
|
743 ALSource->Params.HrtfGain = DryGain;
|
rlm@0
|
744 ALSource->Params.HrtfDir[0] = Position[0];
|
rlm@0
|
745 ALSource->Params.HrtfDir[1] = Position[1];
|
rlm@0
|
746 ALSource->Params.HrtfDir[2] = Position[2];
|
rlm@0
|
747 }
|
rlm@0
|
748 }
|
rlm@0
|
749 else
|
rlm@0
|
750 {
|
rlm@0
|
751 // Get the initial (static) HRIR coefficients and delays.
|
rlm@0
|
752 GetLerpedHrtfCoeffs(ev, az, DryGain,
|
rlm@0
|
753 ALSource->Params.HrtfCoeffs[0],
|
rlm@0
|
754 ALSource->Params.HrtfDelay[0]);
|
rlm@0
|
755 ALSource->HrtfCounter = 0;
|
rlm@0
|
756 ALSource->Params.HrtfGain = DryGain;
|
rlm@0
|
757 ALSource->Params.HrtfDir[0] = Position[0];
|
rlm@0
|
758 ALSource->Params.HrtfDir[1] = Position[1];
|
rlm@0
|
759 ALSource->Params.HrtfDir[2] = Position[2];
|
rlm@0
|
760 }
|
rlm@0
|
761 }
|
rlm@0
|
762 else
|
rlm@0
|
763 {
|
rlm@0
|
764 // Use energy-preserving panning algorithm for multi-speaker playback
|
rlm@0
|
765 ALfloat DirGain, AmbientGain;
|
rlm@0
|
766 const ALfloat *SpeakerGain;
|
rlm@0
|
767 ALfloat length;
|
rlm@0
|
768 ALint pos;
|
rlm@0
|
769
|
rlm@0
|
770 length = maxf(Distance, MinDist);
|
rlm@0
|
771 if(length > 0.0f)
|
rlm@0
|
772 {
|
rlm@0
|
773 ALfloat invlen = 1.0f/length;
|
rlm@0
|
774 Position[0] *= invlen;
|
rlm@0
|
775 Position[1] *= invlen;
|
rlm@0
|
776 Position[2] *= invlen;
|
rlm@0
|
777 }
|
rlm@0
|
778
|
rlm@0
|
779 pos = aluCart2LUTpos(-Position[2]*ZScale, Position[0]);
|
rlm@0
|
780 SpeakerGain = Device->PanningLUT[pos];
|
rlm@0
|
781
|
rlm@0
|
782 DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
|
rlm@0
|
783 // elevation adjustment for directional gain. this sucks, but
|
rlm@0
|
784 // has low complexity
|
rlm@0
|
785 AmbientGain = aluSqrt(1.0/Device->NumChan);
|
rlm@0
|
786 for(i = 0;i < MAXCHANNELS;i++)
|
rlm@0
|
787 {
|
rlm@0
|
788 ALuint i2;
|
rlm@0
|
789 for(i2 = 0;i2 < MAXCHANNELS;i2++)
|
rlm@0
|
790 ALSource->Params.DryGains[i][i2] = 0.0f;
|
rlm@0
|
791 }
|
rlm@0
|
792 for(i = 0;i < (ALint)Device->NumChan;i++)
|
rlm@0
|
793 {
|
rlm@0
|
794 enum Channel chan = Device->Speaker2Chan[i];
|
rlm@0
|
795 ALfloat gain = lerp(AmbientGain, SpeakerGain[chan], DirGain);
|
rlm@0
|
796 ALSource->Params.DryGains[0][chan] = DryGain * gain;
|
rlm@0
|
797 }
|
rlm@0
|
798 }
|
rlm@0
|
799 for(i = 0;i < NumSends;i++)
|
rlm@0
|
800 ALSource->Params.Send[i].WetGain = WetGain[i];
|
rlm@0
|
801
|
rlm@0
|
802 /* Update filter coefficients. */
|
rlm@0
|
803 cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
|
rlm@0
|
804
|
rlm@0
|
805 ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
|
rlm@0
|
806 for(i = 0;i < NumSends;i++)
|
rlm@0
|
807 {
|
rlm@0
|
808 ALfloat a = lpCoeffCalc(WetGainHF[i]*WetGainHF[i], cw);
|
rlm@0
|
809 ALSource->Params.Send[i].iirFilter.coeff = a;
|
rlm@0
|
810 }
|
rlm@0
|
811 }
|
rlm@0
|
812
|
rlm@0
|
813
|
rlm@0
|
814 static __inline ALfloat aluF2F(ALfloat val)
|
rlm@0
|
815 { return val; }
|
rlm@0
|
816 static __inline ALshort aluF2S(ALfloat val)
|
rlm@0
|
817 {
|
rlm@0
|
818 if(val > 1.0f) return 32767;
|
rlm@0
|
819 if(val < -1.0f) return -32768;
|
rlm@0
|
820 return (ALint)(val*32767.0f);
|
rlm@0
|
821 }
|
rlm@0
|
822 static __inline ALushort aluF2US(ALfloat val)
|
rlm@0
|
823 { return aluF2S(val)+32768; }
|
rlm@0
|
824 static __inline ALbyte aluF2B(ALfloat val)
|
rlm@0
|
825 { return aluF2S(val)>>8; }
|
rlm@0
|
826 static __inline ALubyte aluF2UB(ALfloat val)
|
rlm@0
|
827 { return aluF2US(val)>>8; }
|
rlm@0
|
828
|
rlm@0
|
829 #define DECL_TEMPLATE(T, N, func) \
|
rlm@0
|
830 static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
|
rlm@0
|
831 ALuint SamplesToDo) \
|
rlm@0
|
832 { \
|
rlm@0
|
833 ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
|
rlm@0
|
834 const enum Channel *ChanMap = device->DevChannels; \
|
rlm@0
|
835 ALuint i, j; \
|
rlm@0
|
836 \
|
rlm@0
|
837 for(i = 0;i < SamplesToDo;i++) \
|
rlm@0
|
838 { \
|
rlm@0
|
839 for(j = 0;j < N;j++) \
|
rlm@0
|
840 *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
|
rlm@0
|
841 } \
|
rlm@0
|
842 }
|
rlm@0
|
843
|
rlm@0
|
844 DECL_TEMPLATE(ALfloat, 1, aluF2F)
|
rlm@0
|
845 DECL_TEMPLATE(ALfloat, 4, aluF2F)
|
rlm@0
|
846 DECL_TEMPLATE(ALfloat, 6, aluF2F)
|
rlm@0
|
847 DECL_TEMPLATE(ALfloat, 7, aluF2F)
|
rlm@0
|
848 DECL_TEMPLATE(ALfloat, 8, aluF2F)
|
rlm@0
|
849
|
rlm@0
|
850 DECL_TEMPLATE(ALushort, 1, aluF2US)
|
rlm@0
|
851 DECL_TEMPLATE(ALushort, 4, aluF2US)
|
rlm@0
|
852 DECL_TEMPLATE(ALushort, 6, aluF2US)
|
rlm@0
|
853 DECL_TEMPLATE(ALushort, 7, aluF2US)
|
rlm@0
|
854 DECL_TEMPLATE(ALushort, 8, aluF2US)
|
rlm@0
|
855
|
rlm@0
|
856 DECL_TEMPLATE(ALshort, 1, aluF2S)
|
rlm@0
|
857 DECL_TEMPLATE(ALshort, 4, aluF2S)
|
rlm@0
|
858 DECL_TEMPLATE(ALshort, 6, aluF2S)
|
rlm@0
|
859 DECL_TEMPLATE(ALshort, 7, aluF2S)
|
rlm@0
|
860 DECL_TEMPLATE(ALshort, 8, aluF2S)
|
rlm@0
|
861
|
rlm@0
|
862 DECL_TEMPLATE(ALubyte, 1, aluF2UB)
|
rlm@0
|
863 DECL_TEMPLATE(ALubyte, 4, aluF2UB)
|
rlm@0
|
864 DECL_TEMPLATE(ALubyte, 6, aluF2UB)
|
rlm@0
|
865 DECL_TEMPLATE(ALubyte, 7, aluF2UB)
|
rlm@0
|
866 DECL_TEMPLATE(ALubyte, 8, aluF2UB)
|
rlm@0
|
867
|
rlm@0
|
868 DECL_TEMPLATE(ALbyte, 1, aluF2B)
|
rlm@0
|
869 DECL_TEMPLATE(ALbyte, 4, aluF2B)
|
rlm@0
|
870 DECL_TEMPLATE(ALbyte, 6, aluF2B)
|
rlm@0
|
871 DECL_TEMPLATE(ALbyte, 7, aluF2B)
|
rlm@0
|
872 DECL_TEMPLATE(ALbyte, 8, aluF2B)
|
rlm@0
|
873
|
rlm@0
|
874 #undef DECL_TEMPLATE
|
rlm@0
|
875
|
rlm@0
|
876 #define DECL_TEMPLATE(T, N, func) \
|
rlm@0
|
877 static void Write_##T##_##N(ALCdevice *device, T *RESTRICT buffer, \
|
rlm@0
|
878 ALuint SamplesToDo) \
|
rlm@0
|
879 { \
|
rlm@0
|
880 ALfloat (*RESTRICT DryBuffer)[MAXCHANNELS] = device->DryBuffer; \
|
rlm@0
|
881 const enum Channel *ChanMap = device->DevChannels; \
|
rlm@0
|
882 ALuint i, j; \
|
rlm@0
|
883 \
|
rlm@0
|
884 if(device->Bs2b) \
|
rlm@0
|
885 { \
|
rlm@0
|
886 for(i = 0;i < SamplesToDo;i++) \
|
rlm@0
|
887 { \
|
rlm@0
|
888 float samples[2]; \
|
rlm@0
|
889 samples[0] = DryBuffer[i][ChanMap[0]]; \
|
rlm@0
|
890 samples[1] = DryBuffer[i][ChanMap[1]]; \
|
rlm@0
|
891 bs2b_cross_feed(device->Bs2b, samples); \
|
rlm@0
|
892 *(buffer++) = func(samples[0]); \
|
rlm@0
|
893 *(buffer++) = func(samples[1]); \
|
rlm@0
|
894 } \
|
rlm@0
|
895 } \
|
rlm@0
|
896 else \
|
rlm@0
|
897 { \
|
rlm@0
|
898 for(i = 0;i < SamplesToDo;i++) \
|
rlm@0
|
899 { \
|
rlm@0
|
900 for(j = 0;j < N;j++) \
|
rlm@0
|
901 *(buffer++) = func(DryBuffer[i][ChanMap[j]]); \
|
rlm@0
|
902 } \
|
rlm@0
|
903 } \
|
rlm@0
|
904 }
|
rlm@0
|
905
|
rlm@0
|
906 DECL_TEMPLATE(ALfloat, 2, aluF2F)
|
rlm@0
|
907 DECL_TEMPLATE(ALushort, 2, aluF2US)
|
rlm@0
|
908 DECL_TEMPLATE(ALshort, 2, aluF2S)
|
rlm@0
|
909 DECL_TEMPLATE(ALubyte, 2, aluF2UB)
|
rlm@0
|
910 DECL_TEMPLATE(ALbyte, 2, aluF2B)
|
rlm@0
|
911
|
rlm@0
|
912 #undef DECL_TEMPLATE
|
rlm@0
|
913
|
rlm@0
|
914 #define DECL_TEMPLATE(T) \
|
rlm@0
|
915 static void Write_##T(ALCdevice *device, T *buffer, ALuint SamplesToDo) \
|
rlm@0
|
916 { \
|
rlm@0
|
917 switch(device->FmtChans) \
|
rlm@0
|
918 { \
|
rlm@0
|
919 case DevFmtMono: \
|
rlm@0
|
920 Write_##T##_1(device, buffer, SamplesToDo); \
|
rlm@0
|
921 break; \
|
rlm@0
|
922 case DevFmtStereo: \
|
rlm@0
|
923 Write_##T##_2(device, buffer, SamplesToDo); \
|
rlm@0
|
924 break; \
|
rlm@0
|
925 case DevFmtQuad: \
|
rlm@0
|
926 Write_##T##_4(device, buffer, SamplesToDo); \
|
rlm@0
|
927 break; \
|
rlm@0
|
928 case DevFmtX51: \
|
rlm@0
|
929 case DevFmtX51Side: \
|
rlm@0
|
930 Write_##T##_6(device, buffer, SamplesToDo); \
|
rlm@0
|
931 break; \
|
rlm@0
|
932 case DevFmtX61: \
|
rlm@0
|
933 Write_##T##_7(device, buffer, SamplesToDo); \
|
rlm@0
|
934 break; \
|
rlm@0
|
935 case DevFmtX71: \
|
rlm@0
|
936 Write_##T##_8(device, buffer, SamplesToDo); \
|
rlm@0
|
937 break; \
|
rlm@0
|
938 } \
|
rlm@0
|
939 }
|
rlm@0
|
940
|
rlm@0
|
941 DECL_TEMPLATE(ALfloat)
|
rlm@0
|
942 DECL_TEMPLATE(ALushort)
|
rlm@0
|
943 DECL_TEMPLATE(ALshort)
|
rlm@0
|
944 DECL_TEMPLATE(ALubyte)
|
rlm@0
|
945 DECL_TEMPLATE(ALbyte)
|
rlm@0
|
946
|
rlm@0
|
947 #undef DECL_TEMPLATE
|
rlm@0
|
948
|
rlm@0
|
949 ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size)
|
rlm@0
|
950 {
|
rlm@0
|
951 ALuint SamplesToDo;
|
rlm@0
|
952 ALeffectslot *ALEffectSlot;
|
rlm@0
|
953 ALCcontext **ctx, **ctx_end;
|
rlm@0
|
954 ALsource **src, **src_end;
|
rlm@0
|
955 int fpuState;
|
rlm@0
|
956 ALuint i, c;
|
rlm@0
|
957 ALsizei e;
|
rlm@0
|
958
|
rlm@0
|
959 #if defined(HAVE_FESETROUND)
|
rlm@0
|
960 fpuState = fegetround();
|
rlm@0
|
961 fesetround(FE_TOWARDZERO);
|
rlm@0
|
962 #elif defined(HAVE__CONTROLFP)
|
rlm@0
|
963 fpuState = _controlfp(0, 0);
|
rlm@0
|
964 (void)_controlfp(_RC_CHOP, _MCW_RC);
|
rlm@0
|
965 #else
|
rlm@0
|
966 (void)fpuState;
|
rlm@0
|
967 #endif
|
rlm@0
|
968
|
rlm@0
|
969 while(size > 0)
|
rlm@0
|
970 {
|
rlm@0
|
971 /* Setup variables */
|
rlm@0
|
972 SamplesToDo = minu(size, BUFFERSIZE);
|
rlm@0
|
973
|
rlm@0
|
974 /* Clear mixing buffer */
|
rlm@0
|
975 memset(device->DryBuffer, 0, SamplesToDo*MAXCHANNELS*sizeof(ALfloat));
|
rlm@0
|
976
|
rlm@0
|
977 LockDevice(device);
|
rlm@0
|
978 ctx = device->Contexts;
|
rlm@0
|
979 ctx_end = ctx + device->NumContexts;
|
rlm@0
|
980 //printf("Contexts: %d\n", device->NumContexts);
|
rlm@0
|
981 int context_number = 0;
|
rlm@0
|
982 while(ctx != ctx_end)
|
rlm@0
|
983 {
|
rlm@0
|
984 //printf("Context %d:\n", context_number++);
|
rlm@0
|
985 ALboolean DeferUpdates = (*ctx)->DeferUpdates;
|
rlm@0
|
986 ALboolean UpdateSources = AL_FALSE;
|
rlm@0
|
987
|
rlm@0
|
988 if(!DeferUpdates)
|
rlm@0
|
989 {
|
rlm@0
|
990 //printf("NOT deferring updates, whatever that means\n");
|
rlm@0
|
991 UpdateSources = (*ctx)->UpdateSources;
|
rlm@0
|
992 //printf("update sources is set to %d\n", UpdateSources);
|
rlm@0
|
993 (*ctx)->UpdateSources = AL_FALSE;
|
rlm@0
|
994 }
|
rlm@0
|
995
|
rlm@0
|
996 src = (*ctx)->ActiveSources;
|
rlm@0
|
997 src_end = src + (*ctx)->ActiveSourceCount;
|
rlm@0
|
998 //printf("number of active sources are %d\n", (*ctx)->ActiveSourceCount);
|
rlm@0
|
999 while(src != src_end)
|
rlm@0
|
1000 {
|
rlm@0
|
1001
|
rlm@0
|
1002 if((*src)->state != AL_PLAYING)
|
rlm@0
|
1003 {
|
rlm@0
|
1004 --((*ctx)->ActiveSourceCount);
|
rlm@0
|
1005 *src = *(--src_end);
|
rlm@0
|
1006 continue;
|
rlm@0
|
1007 }
|
rlm@0
|
1008
|
rlm@0
|
1009 if(!DeferUpdates && ((*src)->NeedsUpdate || UpdateSources))
|
rlm@0
|
1010 {
|
rlm@0
|
1011 (*src)->NeedsUpdate = AL_FALSE;
|
rlm@0
|
1012 ALsource_Update(*src, *ctx);
|
rlm@0
|
1013 }
|
rlm@0
|
1014 //printf("calling MixSource!\n");
|
rlm@0
|
1015 MixSource(*src, device, SamplesToDo);
|
rlm@0
|
1016 src++;
|
rlm@0
|
1017 }
|
rlm@0
|
1018
|
rlm@0
|
1019 /* effect slot processing */
|
rlm@0
|
1020 for(e = 0;e < (*ctx)->EffectSlotMap.size;e++)
|
rlm@0
|
1021 {
|
rlm@0
|
1022 ALEffectSlot = (*ctx)->EffectSlotMap.array[e].value;
|
rlm@0
|
1023
|
rlm@0
|
1024 for(i = 0;i < SamplesToDo;i++)
|
rlm@0
|
1025 {
|
rlm@0
|
1026 // RLM: remove click-removal
|
rlm@0
|
1027 ALEffectSlot->WetBuffer[i] += ALEffectSlot->ClickRemoval[0];
|
rlm@0
|
1028 ALEffectSlot->ClickRemoval[0] -= ALEffectSlot->ClickRemoval[0] / 256.0f;
|
rlm@0
|
1029 }
|
rlm@0
|
1030 for(i = 0;i < 1;i++)
|
rlm@0
|
1031 {
|
rlm@0
|
1032 // RLM: remove click-removal
|
rlm@0
|
1033 ALEffectSlot->ClickRemoval[i] += ALEffectSlot->PendingClicks[i];
|
rlm@0
|
1034 ALEffectSlot->PendingClicks[i] = 0.0f;
|
rlm@0
|
1035 }
|
rlm@0
|
1036
|
rlm@0
|
1037 if(!DeferUpdates && ALEffectSlot->NeedsUpdate)
|
rlm@0
|
1038 {
|
rlm@0
|
1039 ALEffectSlot->NeedsUpdate = AL_FALSE;
|
rlm@0
|
1040 ALEffect_Update(ALEffectSlot->EffectState, *ctx, ALEffectSlot);
|
rlm@0
|
1041 }
|
rlm@0
|
1042
|
rlm@0
|
1043 ALEffect_Process(ALEffectSlot->EffectState, ALEffectSlot,
|
rlm@0
|
1044 SamplesToDo, ALEffectSlot->WetBuffer,
|
rlm@0
|
1045 device->DryBuffer);
|
rlm@0
|
1046
|
rlm@0
|
1047 for(i = 0;i < SamplesToDo;i++)
|
rlm@0
|
1048 ALEffectSlot->WetBuffer[i] = 0.0f;
|
rlm@0
|
1049 }
|
rlm@0
|
1050
|
rlm@0
|
1051 ctx++;
|
rlm@0
|
1052 }
|
rlm@0
|
1053 UnlockDevice(device);
|
rlm@0
|
1054
|
rlm@0
|
1055 //Post processing loop
|
rlm@0
|
1056 if(device->FmtChans == DevFmtMono)
|
rlm@0
|
1057 {
|
rlm@0
|
1058 for(i = 0;i < SamplesToDo;i++)
|
rlm@0
|
1059 {
|
rlm@0
|
1060 // RLM: remove click-removal
|
rlm@0
|
1061 device->DryBuffer[i][FRONT_CENTER] += device->ClickRemoval[FRONT_CENTER];
|
rlm@0
|
1062 device->ClickRemoval[FRONT_CENTER] -= device->ClickRemoval[FRONT_CENTER] / 256.0f;
|
rlm@0
|
1063 }
|
rlm@0
|
1064 // RLM: remove click-removal
|
rlm@0
|
1065 device->ClickRemoval[FRONT_CENTER] += device->PendingClicks[FRONT_CENTER];
|
rlm@0
|
1066 device->PendingClicks[FRONT_CENTER] = 0.0f;
|
rlm@0
|
1067 }
|
rlm@0
|
1068 else if(device->FmtChans == DevFmtStereo)
|
rlm@0
|
1069 {
|
rlm@0
|
1070 /* Assumes the first two channels are FRONT_LEFT and FRONT_RIGHT */
|
rlm@0
|
1071 for(i = 0;i < SamplesToDo;i++)
|
rlm@0
|
1072 {
|
rlm@0
|
1073 for(c = 0;c < 2;c++)
|
rlm@0
|
1074 {
|
rlm@0
|
1075 // RLM: remove click-removal
|
rlm@0
|
1076 device->DryBuffer[i][c] += device->ClickRemoval[c];
|
rlm@0
|
1077 device->ClickRemoval[c] -= device->ClickRemoval[c] / 256.0f;
|
rlm@0
|
1078 }
|
rlm@0
|
1079 }
|
rlm@0
|
1080 for(c = 0;c < 2;c++)
|
rlm@0
|
1081 {
|
rlm@0
|
1082 // RLM: remove click-removal
|
rlm@0
|
1083 device->ClickRemoval[c] += device->PendingClicks[c];
|
rlm@0
|
1084 device->PendingClicks[c] = 0.0f;
|
rlm@0
|
1085 }
|
rlm@0
|
1086 }
|
rlm@0
|
1087 else
|
rlm@0
|
1088 {
|
rlm@0
|
1089 for(i = 0;i < SamplesToDo;i++)
|
rlm@0
|
1090 {
|
rlm@0
|
1091 for(c = 0;c < MAXCHANNELS;c++)
|
rlm@0
|
1092 {
|
rlm@0
|
1093 // RLM: remove click-removal
|
rlm@0
|
1094 device->DryBuffer[i][c] += device->ClickRemoval[c];
|
rlm@0
|
1095 device->ClickRemoval[c] -= device->ClickRemoval[c] / 256.0f;
|
rlm@0
|
1096 }
|
rlm@0
|
1097 }
|
rlm@0
|
1098 for(c = 0;c < MAXCHANNELS;c++)
|
rlm@0
|
1099 {
|
rlm@0
|
1100 // RLM: remove click-removal
|
rlm@0
|
1101 device->ClickRemoval[c] += device->PendingClicks[c];
|
rlm@0
|
1102 device->PendingClicks[c] = 0.0f;
|
rlm@0
|
1103 }
|
rlm@0
|
1104 }
|
rlm@0
|
1105
|
rlm@0
|
1106 if(buffer)
|
rlm@0
|
1107 {
|
rlm@0
|
1108 switch(device->FmtType)
|
rlm@0
|
1109 {
|
rlm@0
|
1110 case DevFmtByte:
|
rlm@0
|
1111 Write_ALbyte(device, buffer, SamplesToDo);
|
rlm@0
|
1112 break;
|
rlm@0
|
1113 case DevFmtUByte:
|
rlm@0
|
1114 Write_ALubyte(device, buffer, SamplesToDo);
|
rlm@0
|
1115 break;
|
rlm@0
|
1116 case DevFmtShort:
|
rlm@0
|
1117 Write_ALshort(device, buffer, SamplesToDo);
|
rlm@0
|
1118 break;
|
rlm@0
|
1119 case DevFmtUShort:
|
rlm@0
|
1120 Write_ALushort(device, buffer, SamplesToDo);
|
rlm@0
|
1121 break;
|
rlm@0
|
1122 case DevFmtFloat:
|
rlm@0
|
1123 Write_ALfloat(device, buffer, SamplesToDo);
|
rlm@0
|
1124 break;
|
rlm@0
|
1125 }
|
rlm@0
|
1126 }
|
rlm@0
|
1127
|
rlm@0
|
1128 size -= SamplesToDo;
|
rlm@0
|
1129 }
|
rlm@0
|
1130
|
rlm@0
|
1131 #if defined(HAVE_FESETROUND)
|
rlm@0
|
1132 fesetround(fpuState);
|
rlm@0
|
1133 #elif defined(HAVE__CONTROLFP)
|
rlm@0
|
1134 _controlfp(fpuState, _MCW_RC);
|
rlm@0
|
1135 #endif
|
rlm@0
|
1136 }
|
rlm@0
|
1137
|
rlm@0
|
1138
|
rlm@0
|
1139
|
rlm@0
|
1140
|
rlm@0
|
1141
|
rlm@0
|
1142 ALvoid aluHandleDisconnect(ALCdevice *device)
|
rlm@0
|
1143 {
|
rlm@0
|
1144 ALuint i;
|
rlm@0
|
1145
|
rlm@0
|
1146 LockDevice(device);
|
rlm@0
|
1147 for(i = 0;i < device->NumContexts;i++)
|
rlm@0
|
1148 {
|
rlm@0
|
1149 ALCcontext *Context = device->Contexts[i];
|
rlm@0
|
1150 ALsource *source;
|
rlm@0
|
1151 ALsizei pos;
|
rlm@0
|
1152
|
rlm@0
|
1153 for(pos = 0;pos < Context->SourceMap.size;pos++)
|
rlm@0
|
1154 {
|
rlm@0
|
1155 source = Context->SourceMap.array[pos].value;
|
rlm@0
|
1156 if(source->state == AL_PLAYING)
|
rlm@0
|
1157 {
|
rlm@0
|
1158 source->state = AL_STOPPED;
|
rlm@0
|
1159 source->BuffersPlayed = source->BuffersInQueue;
|
rlm@0
|
1160 source->position = 0;
|
rlm@0
|
1161 source->position_fraction = 0;
|
rlm@0
|
1162 }
|
rlm@0
|
1163 }
|
rlm@0
|
1164 }
|
rlm@0
|
1165
|
rlm@0
|
1166 device->Connected = ALC_FALSE;
|
rlm@0
|
1167 UnlockDevice(device);
|
rlm@0
|
1168 }
|