1 | /* -*- mode: C; tab-width:8; c-basic-offset:8 -*- |
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2 | * vi:set ts=8: |
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3 | * |
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4 | * al_filter.c |
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5 | * |
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6 | * Contains filters. |
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7 | * |
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8 | * |
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9 | * Short guide to filters: |
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10 | * |
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11 | * Filters all have the prefix alf_<something>. Each filter |
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12 | * defined in software_time_filters or software_frequency_filters |
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13 | * is applied to each source that finds its way into the mixer. |
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14 | * |
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15 | * ApplyFilters takes a chunk of data from the original buffer |
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16 | * associated with the passed source. |
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17 | * |
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18 | * This chunk is understood to be that block of data samp->_orig_buffer |
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19 | * offset src->soundpos to src->soundpos + bufsiz, where src is the |
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20 | * passed AL_source, samp is the AL_buffer associated with src, and |
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21 | * bufsiz is the length of the chunk of data that we want. It is usually |
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22 | * set to _AL_DEF_BUFSIZE, unless specified by ALC_BUFFERSIZE in the |
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23 | * application. |
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24 | * |
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25 | * Applying filters to a source does not (should not) change the original |
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26 | * pcm data. ApplyFilters will split the original pcm data prior to |
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27 | * calling each filter, and filters should restrict themselves to |
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28 | * manipulating the passed data. |
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29 | * |
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30 | * time domain filters (those defines in software_time_filters) are |
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31 | * passed: |
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32 | * ALuint cid |
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33 | * identifier for the context that this source belongs to |
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34 | * AL_source *src |
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35 | * source that the filter should be applied to |
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36 | * AL_buffer *samp |
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37 | * buffer that the source is associated with |
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38 | * ALshort **buffers |
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39 | * arrays of points to PCM data, one element per |
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40 | * channel(left/right/etc) |
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41 | * ALuint nc |
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42 | * number of elements in buffers |
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43 | * ALuint len |
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44 | * byte length of each element in buffers |
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45 | * |
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46 | * Filters are expected to alter buffers[0..nc-1] in place. After |
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47 | * the ApplyFilter iteration is over, the resulting data is mixed into |
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48 | * the main mix buffer and forgotten. The data altered is cumulative, |
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49 | * that is to say, if two filters alf_f and alf_g occur in sequential |
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50 | * order, alf_g will see the pcm data after alf_f has altered it. |
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51 | * |
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52 | * FINER POINTS: |
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53 | * |
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54 | * A lot of the filters make effects by modulating amplitude and delay. |
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55 | * Because these changes are cumulative, we can reduce the application |
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56 | * of amplitude and delay changes to one operation. This is the point |
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57 | * of SourceParamApply --- filters can make changes to srcParams.gain |
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58 | * and srcParams.delay in a source and have those changes applied at |
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59 | * the end of the ApplyFilters call for the source. These values are |
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60 | * reset to their defaults at the top of the ApplyFilters call. |
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61 | * |
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62 | */ |
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63 | #include "al_siteconfig.h" |
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64 | |
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65 | #include <AL/alext.h> |
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66 | |
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67 | #include <math.h> |
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68 | #include <stdlib.h> |
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69 | #include <string.h> |
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70 | #include <float.h> |
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71 | |
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72 | #include "al_buffer.h" |
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73 | #include "al_debug.h" |
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74 | #include "al_error.h" |
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75 | #include "al_filter.h" |
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76 | #include "al_listen.h" |
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77 | #include "al_main.h" |
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78 | #include "al_mixer.h" |
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79 | #include "al_source.h" |
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80 | #include "al_queue.h" |
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81 | #include "al_vector.h" |
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82 | |
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83 | #include "alc/alc_context.h" |
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84 | #include "alc/alc_speaker.h" |
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85 | |
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86 | #define MIN(a,b) (((a) < (b)) ? (a) : (b)) |
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87 | #define MAX(a,b) (((a) < (b)) ? (b) : (a)) |
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88 | |
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89 | #define MIN_PITCH 0.25f |
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90 | |
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91 | #define USE_TPITCH_LOOKUP 1 /* icculus change here JIV FIXME */ |
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92 | |
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93 | /* |
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94 | * _AL_CUTTOFF_ATTENUATION is the value below which, sounds are not further |
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95 | * distance attenuated. The purpose of this culling is to avoid pop-off |
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96 | * artifacts. |
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97 | * |
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98 | * Elias: This has been found to cause insufficient distance attenuation |
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99 | * and has therefore been effectively disabled by setting it to 0. If no |
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100 | * problems show up, the value should be completely removed. |
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101 | * The original was value 0.01 |
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102 | */ |
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103 | #define _AL_CUTTOFF_ATTENUATION 0.00 |
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104 | |
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105 | /* |
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106 | * TPITCH_MAX sets the number of discrete values for AL_PITCH we can have. |
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107 | * You can set AL_PITCH to anything, but integer rounding will ensure that |
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108 | * it will fall beween MIN_SCALE and 2.0. |
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109 | * |
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110 | * 2.0 is an arbitrary constant, and likely to be changed. |
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111 | */ |
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112 | #define TPITCH_MAX 256 |
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113 | |
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114 | /* |
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115 | * The default software time domain filters. |
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116 | * |
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117 | * I wish I could say that the order of these does not matter, |
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118 | * but it does. Namely, tdoppler and tpitch must occur in that |
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119 | * order, and they must occur before any other filter. listenergain must |
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120 | * occur last. |
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121 | */ |
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122 | static time_filter_set software_time_filters[] = { |
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123 | { "tdoppler", alf_tdoppler }, /* time-domain doppler filter */ |
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124 | { "tpitch", alf_tpitch }, /* time-domain pitch filter */ |
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125 | { "da", alf_da }, |
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126 | { "reverb", alf_reverb }, |
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127 | { "coning", alf_coning }, |
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128 | { "panning", alf_panning }, |
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129 | { "minmax", alf_minmax }, |
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130 | { "listenergain", alf_listenergain }, |
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131 | { { 0 }, NULL } |
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132 | }; |
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133 | |
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134 | /* |
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135 | * compute_sa( ALfloat *source_pos, ALfloat source_max, |
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136 | * ALfloat source_ref, ALfloat source_gain, |
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137 | * ALfloat source_rolloff, |
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138 | * ALfloat *speaker_pos, |
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139 | * ALfloat (*df)( ALfloat dist, ALfloat rolloff, |
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140 | * ALfloat ref, ALfloat max)) |
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141 | * |
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142 | * computes distance attenuation with respect to a speaker position. |
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143 | * |
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144 | * This is some normalized value which gets expotenially closer to 1.0 |
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145 | * as the source approaches the listener. The minimum attenuation is |
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146 | * AL_CUTTOFF_ATTENUATION, which approached when the source approaches |
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147 | * the max distance. |
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148 | * |
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149 | * source_pos = source position [x/y/z] |
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150 | * source_max = source specific max distance |
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151 | * speaker_pos = speaker position [x/y/z] |
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152 | * ref = source's reference distance |
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153 | * df = distance model function |
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154 | * max = maximum distance, beyond which everything is clamped at |
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155 | * some small value near, but not equal to, zero. |
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156 | */ |
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157 | static ALfloat compute_sa( ALfloat *source_pos, ALfloat source_max, |
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158 | ALfloat source_ref, ALfloat source_gain, |
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159 | ALfloat source_rolloff, |
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160 | ALfloat *speaker_pos, |
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161 | ALfloat df( ALfloat dist, ALfloat rolloff, ALfloat ref, ALfloat max )); |
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162 | |
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163 | #if USE_TPITCH_LOOKUP |
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164 | /* |
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165 | * our quick lookup table for our time domain pitch filter. |
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166 | * |
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167 | * |
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168 | * We initialize each element in offsets to be a set of offsets, |
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169 | * such that |
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170 | * |
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171 | * offset[x][y] = int portion of y * pitch |
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172 | * fractinoal[x][y] = fractional portion of y * pitch |
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173 | * |
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174 | * Where x is the discrete integer value of pitch, and y is |
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175 | * any value between 0 and the length of a buffer. |
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176 | * |
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177 | * What's the point? To save the pain of float->int conversion at |
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178 | * runtime, which is needed to map the original PCM data to a |
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179 | * "pitch modified" mapping of the same data. |
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180 | * |
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181 | */ |
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182 | static struct { |
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183 | int *offsets[TPITCH_MAX]; /* use int instead of ALint because |
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184 | * these are array indexes |
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185 | */ |
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186 | float *fractionals[TPITCH_MAX]; |
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187 | ALuint max; |
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188 | ALuint middle; /* the index which pitch == 1.0 corresponds to */ |
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189 | ALuint len; /* length of offsets[0...TPITCH_MAX] in samples */ |
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190 | } tpitch_lookup = { { NULL }, { NULL }, 0, 0, 0 }; |
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191 | |
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192 | /* func associated with our tpitch lookup */ |
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193 | static void init_tpitch_lookup(ALuint len); |
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194 | #endif |
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195 | |
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196 | static ALfloat compute_doppler_pitch(ALfloat *object1, ALfloat *o1_vel, |
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197 | ALfloat *object2, ALfloat *o2_vel, |
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198 | ALfloat factor, ALfloat speed); |
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199 | |
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200 | /* |
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201 | * _alInitTimeFilters( time_filter_set *tf_ptr_ref ) |
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202 | * |
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203 | * Initializes tf_ptr_ref to the current set of time filters, and initialize |
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204 | * tpitch_lookup_max if it hasn't been initialized before. |
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205 | */ |
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206 | void _alInitTimeFilters( time_filter_set *tf_ptr_ref ) { |
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207 | ALuint i = 0; |
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208 | |
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209 | do { |
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210 | tf_ptr_ref[i] = software_time_filters[i]; |
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211 | } while(software_time_filters[i++].filter != NULL); |
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212 | |
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213 | #if USE_TPITCH_LOOKUP |
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214 | /* |
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215 | * init tpitch_loopup only if it hasn't been initialized |
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216 | * yet. |
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217 | */ |
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218 | if(tpitch_lookup.max != TPITCH_MAX) { |
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219 | tpitch_lookup.max = TPITCH_MAX; |
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220 | tpitch_lookup.middle = TPITCH_MAX / 2; |
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221 | |
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222 | for(i = 0; i < tpitch_lookup.max; i++) |
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223 | { |
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224 | free(tpitch_lookup.offsets[i]); |
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225 | free(tpitch_lookup.fractionals[i]); |
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226 | tpitch_lookup.offsets[i] = 0; |
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227 | tpitch_lookup.fractionals[i] = 0; |
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228 | } |
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229 | } |
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230 | #endif |
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231 | |
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232 | return; |
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233 | } |
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234 | |
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235 | /* |
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236 | * _alDestroyFilters( void ) |
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237 | * |
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238 | * Deallocates data structures used by the filters and helper functions. |
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239 | */ |
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240 | void _alDestroyFilters( void ) { |
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241 | #if USE_TPITCH_LOOKUP |
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242 | ALuint i; |
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243 | |
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244 | for(i = 0; i < TPITCH_MAX; i++) |
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245 | { |
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246 | free(tpitch_lookup.offsets[i]); |
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247 | free(tpitch_lookup.fractionals[i]); |
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248 | |
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249 | tpitch_lookup.offsets[i] = 0; |
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250 | tpitch_lookup.fractionals[i] = 0; |
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251 | } |
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252 | tpitch_lookup.len = 0; |
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253 | #endif |
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254 | |
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255 | return; |
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256 | } |
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257 | |
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258 | /* |
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259 | * _alApplyFilters( ALuint cid, ALuint sid ) |
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260 | * |
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261 | * _alApplyFilters is called from the mixing function, and is passed |
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262 | * a source id and the context where this sourceid has meaning. |
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263 | * |
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264 | * The filters that are applied to the source are determined by the |
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265 | * context. Each context is initialized such that it contains a table |
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266 | * of the software filters. Extensions and plugins can be later loaded |
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267 | * to override the default functionality. The point being, each context's |
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268 | * filter "signature" may be different. |
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269 | * |
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270 | * assumes locked source sid |
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271 | */ |
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272 | void _alApplyFilters( ALuint cid, ALuint sid ) { |
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273 | AL_source *src; |
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274 | AL_buffer *samp; |
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275 | time_filter_set *cc_tfilters; |
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276 | time_filter *tf; |
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277 | ALuint mixbuflen; /* byte size of total data to compose (all channels) */ |
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278 | ALint len; /* byte size of one channel's worth of data to compose */ |
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279 | ALint filterlen; /* filterlen is adjusted below to take into account looping, etc */ |
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280 | int ic; /* internal (canon) chans */ |
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281 | int mc; /* mixer chans (==speakers) */ |
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282 | ALboolean *boolp; /* for determining bool flags */ |
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283 | int i; |
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284 | |
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285 | /* initialize */ |
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286 | ic = _alGetChannelsFromFormat( _ALC_CANON_FMT ); |
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287 | |
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288 | _alcLockContext( cid ); |
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289 | |
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290 | mc = _alcGetNumSpeakers( cid ); |
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291 | mixbuflen = _alcGetWriteBufsiz( cid ); |
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292 | |
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293 | samp = _alGetBufferFromSid( cid, sid ); |
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294 | if(samp == NULL) { |
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295 | _alDebug(ALD_MAXIMUS, __FILE__, __LINE__, |
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296 | "_alFilter: null samp, sid == %d", sid); |
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297 | |
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298 | _alcUnlockContext( cid ); |
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299 | return; |
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300 | } |
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301 | |
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302 | _alcUnlockContext( cid ); |
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303 | |
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304 | len = mixbuflen * ((float) ic / mc); |
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305 | filterlen = len; |
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306 | |
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307 | /* |
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308 | * Allocate scratch space to hold enough data for the source |
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309 | * about to be split. We allocate more space in case of a |
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310 | * multichannel source. |
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311 | */ |
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312 | if(f_buffers.len < len / sizeof (ALshort)) |
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313 | { |
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314 | void *temp; |
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315 | ALuint newlen = len * _alGetChannelsFromFormat(samp->format); |
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316 | |
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317 | for(i = 0; i < mc; i++) |
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318 | { |
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319 | temp = realloc(f_buffers.data[i], newlen); |
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320 | if(temp == NULL) { |
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321 | /* FIXME: do something */ |
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322 | } |
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323 | |
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324 | f_buffers.data[i] = temp; |
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325 | } |
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326 | |
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327 | f_buffers.len = newlen; |
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328 | } |
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329 | |
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330 | #if USE_TPITCH_LOOKUP |
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331 | if(tpitch_lookup.len < (ALuint) len) |
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332 | { |
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333 | init_tpitch_lookup(len); |
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334 | } |
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335 | #endif |
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336 | |
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337 | src = _alGetSource(cid, sid); |
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338 | if(src == NULL) { |
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339 | _alDebug(ALD_MAXIMUS, __FILE__, __LINE__, |
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340 | "_alFilter: null src, sid == %d", sid); |
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341 | return; |
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342 | } |
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343 | |
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344 | /* streaming buffer? set soundpos */ |
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345 | if(samp->flags & ALB_STREAMING) { |
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346 | src->srcParams.soundpos = samp->streampos; |
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347 | |
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348 | if(samp->streampos > samp->size) { |
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349 | memset(src->srcParams.outbuf, 0, len); |
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350 | |
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351 | #ifdef DEBUG_MAXIMUS |
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352 | fprintf(stderr, "underflow!!!!!!!!!!!!!!!!\n"); |
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353 | #endif |
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354 | return; /* underflow */ |
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355 | } |
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356 | } |
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357 | |
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358 | _alSourceParamReset(src); /* reset srcParam settings */ |
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359 | |
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360 | _alSplitSources(cid, sid, mc, len, samp, (ALshort **) f_buffers.data); |
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361 | |
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362 | /* |
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363 | * translate head relative sources |
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364 | */ |
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365 | boolp = _alGetSourceParam(src, AL_SOURCE_RELATIVE); |
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366 | |
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367 | if(boolp != NULL) { |
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368 | _alDebug(ALD_SOURCE, __FILE__, __LINE__, |
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369 | "_alApplyFilters: sid %d relative boolp = %d", sid, *boolp ); |
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370 | |
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371 | if(*boolp == AL_TRUE) { |
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372 | /* This is a RELATIVE source, which means we must |
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373 | * translate it before applying any sort of positional |
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374 | * filter to it. |
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375 | */ |
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376 | AL_context *cc; |
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377 | |
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378 | _alcLockContext( cid ); |
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379 | |
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380 | cc = _alcGetContext(cid); |
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381 | if(cc != NULL) { |
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382 | _alSourceTranslate(src, cc->listener.position ); |
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383 | } |
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384 | |
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385 | _alcUnlockContext( cid ); |
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386 | } |
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387 | } |
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388 | |
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389 | /* |
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390 | * adjust len to account for end of sample, looping, etc |
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391 | */ |
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392 | if(filterlen > _alSourceBytesLeft(src, samp)) |
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393 | { |
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394 | /* John Quigley's patch, check it out -- jiv */ |
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395 | if((_alSourceIsLooping( src ) == AL_FALSE) |
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396 | && (src->srcParams.new_readindex == -1)) |
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397 | { |
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398 | /* Non looping source */ |
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399 | filterlen = _alSourceBytesLeft(src, samp); |
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400 | } |
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401 | } |
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402 | |
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403 | if(filterlen > 0) |
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404 | { |
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405 | cc_tfilters = _alcGetTimeFilters(cid); |
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406 | |
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407 | /* apply time domain filters */ |
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408 | for(i = 0; cc_tfilters[i].filter != NULL; i++) |
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409 | { |
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410 | tf = cc_tfilters[i].filter; |
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411 | |
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412 | tf(cid, src, samp, (ALshort **) f_buffers.data, |
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413 | mc, filterlen); |
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414 | } |
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415 | |
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416 | /* |
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417 | * Apply gain and delay for filters that don't actually touch |
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418 | * the data ( alf_da). |
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419 | */ |
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420 | _alSourceParamApply(src, mc, filterlen, |
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421 | (ALshort **) f_buffers.data); |
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422 | } |
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423 | |
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424 | |
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425 | /* |
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426 | * Take the resulting pcm data in f_buffers, and mix these into |
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427 | * the source's temporary output buffer. |
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428 | */ |
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429 | _alCollapseSource(cid, sid, mc, mixbuflen, (ALshort **) f_buffers.data); |
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430 | |
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431 | /* |
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432 | * head RELATIVE sources need to be untranslated, lest their |
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433 | * position become weird. |
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434 | */ |
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435 | if((boolp != NULL) && (*boolp == AL_TRUE)) { |
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436 | AL_context *cc; |
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437 | ALfloat ipos[3]; /* inverse listener position */ |
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438 | |
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439 | _alcLockContext( cid ); |
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440 | |
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441 | cc = _alcGetContext(cid); |
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442 | |
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443 | if(cc != NULL) { |
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444 | _alVectorInverse(ipos, cc->listener.position); |
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445 | _alSourceTranslate(src, ipos); |
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446 | } |
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447 | |
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448 | _alcUnlockContext( cid ); |
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449 | } |
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450 | |
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451 | return; |
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452 | } |
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453 | |
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454 | /* |
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455 | * alf_coning |
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456 | * |
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457 | * Implements the coning filter, which is used when CONE_INNER_ANGLE |
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458 | * or CONE_OUTER_ANGLE is set. This is used for directional sounds. |
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459 | * |
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460 | * The spec is vague as to the actual requirements of directional sounds, |
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461 | * and Carlo has suggested that we maintain the DirectSound meaning for |
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462 | * directional sounds, namely (in my interpretation): |
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463 | * |
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464 | * The inner, outer cone define three zones: inside inner cone |
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465 | * (INSIDE), between inner and outer cone (BETWEEN, outside outer cone, |
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466 | * (OUTSIDE). |
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467 | * |
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468 | * In INSIDE, the gain of the sound is attenuated as a normal |
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469 | * positional source. |
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470 | * |
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471 | * In OUTSIDE, the gain is set to some value specified by the user. |
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472 | * |
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473 | * In BETWEEN, the gain is transitionally set to some value between |
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474 | * what it would be in INSIDE and OUTSIDE. |
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475 | * |
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476 | * This requires an additional source paramter, like CONE_OUTSIDE_ATTENUATION, |
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477 | * and quite frankly seems goofy. This implementation implements the |
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478 | * following convention: |
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479 | * |
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480 | * In INSIDE, the gain of the sound is attenuated as a normal |
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481 | * positional source. |
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482 | * |
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483 | * In OUTSIDE, the gain is set to _AL_CUTTOFF_ATTENUATION |
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484 | * |
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485 | * In BETWEEN, the gain is transitionally set to some value between |
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486 | * what it would be in INSIDE and OUTSIDE. |
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487 | * |
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488 | * Well, okay that's still pretty goofy. Folks who want to set a |
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489 | * minimum attenuation can stil do so using AL_SOURCE_ATTENUATION_MIN. |
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490 | * |
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491 | * IMPLEMENTATION: |
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492 | * okay, we check the angle between the speaker position and |
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493 | * the source's direction vector, using the source's position |
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494 | * as the origin. This angle we call theta. |
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495 | * |
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496 | * Then, we compare theta with the outer cone angle. If it's greater, |
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497 | * we use the min attenuation. If it's less, we compare theta with |
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498 | * the inner cone angle. If it's greater, we attenuate as normal. |
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499 | * Otherwise, we don't attenuate at all (full volume, pitch etc). |
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500 | * |
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501 | * assumes locked source |
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502 | * |
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503 | * FIXME: please check my math. |
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504 | * - with an AL_NONE distance model, should this do anything |
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505 | * at all? |
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506 | */ |
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507 | void alf_coning( ALuint cid, |
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508 | AL_source *src, |
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509 | UNUSED(AL_buffer *samp), |
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510 | UNUSED(ALshort **buffers), |
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511 | ALuint nc, |
---|
512 | UNUSED(ALuint len)) { |
---|
513 | AL_context *cc; |
---|
514 | ALfloat sa; /* speaker attenuation */ |
---|
515 | ALfloat *sp; /* source position */ |
---|
516 | ALfloat *sd; /* source direction */ |
---|
517 | ALfloat lp[3]; /* listener position */ |
---|
518 | ALfloat theta; /* angle between listener and source's direction |
---|
519 | * vector, with the source's position as origin. |
---|
520 | */ |
---|
521 | ALfloat srcDir[3]; |
---|
522 | ALfloat icone; /* inner cone angle. */ |
---|
523 | ALfloat ocone; /* outer cone angle. */ |
---|
524 | ALfloat (*df)( ALfloat dist, ALfloat rolloff, ALfloat ref, ALfloat max ); /* distance model func */ |
---|
525 | ALfloat smax; /* source specific max distance */ |
---|
526 | ALfloat ref; /* source specific reference distance */ |
---|
527 | ALfloat gain; /* source specific gain */ |
---|
528 | ALfloat outergain; /* source specific outer gain */ |
---|
529 | ALfloat rolloff; /* source specific rolloff factor */ |
---|
530 | void *temp; |
---|
531 | ALuint i; |
---|
532 | |
---|
533 | _alcLockContext( cid ); |
---|
534 | cc = _alcGetContext( cid ); |
---|
535 | if(cc == NULL) { |
---|
536 | /* ugh. bad context id */ |
---|
537 | |
---|
538 | _alcUnlockContext( cid ); |
---|
539 | return; |
---|
540 | } |
---|
541 | |
---|
542 | /* |
---|
543 | * The source specific max is set to max at this point, but may be |
---|
544 | * altered below of the application has set it. |
---|
545 | */ |
---|
546 | smax = FLT_MAX; |
---|
547 | df = cc->distance_func; |
---|
548 | |
---|
549 | _alcUnlockContext( cid ); |
---|
550 | |
---|
551 | alGetListenerfv(AL_POSITION, lp); |
---|
552 | |
---|
553 | /* If no direction set, return */ |
---|
554 | sd = _alGetSourceParam( src, AL_DIRECTION ); |
---|
555 | if(sd == NULL) { |
---|
556 | /* |
---|
557 | * source has no direction (normal). leave it for alf_da |
---|
558 | */ |
---|
559 | return; |
---|
560 | } |
---|
561 | |
---|
562 | sp = _alGetSourceParam( src, AL_POSITION ); |
---|
563 | if(sp == NULL) { |
---|
564 | /* If no position set, return */ |
---|
565 | |
---|
566 | return; |
---|
567 | } |
---|
568 | |
---|
569 | /* get source specific ref distance */ |
---|
570 | temp = _alGetSourceParam( src, AL_REFERENCE_DISTANCE ); |
---|
571 | if( temp != NULL ) { |
---|
572 | ref = * (ALfloat *) temp; |
---|
573 | } else { |
---|
574 | _alSourceGetParamDefault( AL_REFERENCE_DISTANCE, &ref ); |
---|
575 | } |
---|
576 | |
---|
577 | /* get source specific gain */ |
---|
578 | temp = _alGetSourceParam( src, AL_GAIN ); |
---|
579 | if( temp != NULL ) { |
---|
580 | gain = * (ALfloat *) temp; |
---|
581 | } else { |
---|
582 | _alSourceGetParamDefault( AL_GAIN, &gain ); |
---|
583 | } |
---|
584 | |
---|
585 | /* get source specific max distance */ |
---|
586 | temp = _alGetSourceParam( src, AL_MAX_DISTANCE ); |
---|
587 | if( temp != NULL ) { |
---|
588 | smax = * (ALfloat *) temp; |
---|
589 | } else { |
---|
590 | _alSourceGetParamDefault( AL_MAX_DISTANCE, &smax ); |
---|
591 | } |
---|
592 | |
---|
593 | /* get source specific rolloff factor */ |
---|
594 | temp = _alGetSourceParam( src, AL_ROLLOFF_FACTOR ); |
---|
595 | if( temp != NULL ) { |
---|
596 | rolloff = * (ALfloat *) temp; |
---|
597 | } else { |
---|
598 | _alSourceGetParamDefault( AL_ROLLOFF_FACTOR, &rolloff ); |
---|
599 | } |
---|
600 | |
---|
601 | srcDir[0] = sp[0] + sd[0]; |
---|
602 | srcDir[1] = sp[1] + sd[1]; |
---|
603 | srcDir[2] = sp[2] + sd[2]; |
---|
604 | |
---|
605 | /* |
---|
606 | * Get CONE settings. |
---|
607 | * |
---|
608 | * If unset, use 360.0 degrees |
---|
609 | */ |
---|
610 | temp = _alGetSourceParam( src, AL_CONE_INNER_ANGLE ); |
---|
611 | if(temp != NULL) { |
---|
612 | icone = _alDegreeToRadian(* (ALfloat *) temp); |
---|
613 | } else { |
---|
614 | _alSourceGetParamDefault( AL_CONE_INNER_ANGLE, &icone ); |
---|
615 | } |
---|
616 | |
---|
617 | temp = _alGetSourceParam( src, AL_CONE_OUTER_ANGLE ); |
---|
618 | if(temp != NULL) { |
---|
619 | ocone = _alDegreeToRadian(* (ALfloat *) temp); |
---|
620 | } else { |
---|
621 | _alSourceGetParamDefault( AL_CONE_OUTER_ANGLE, &ocone ); |
---|
622 | } |
---|
623 | |
---|
624 | temp = _alGetSourceParam( src, AL_CONE_OUTER_GAIN ); |
---|
625 | if(temp != NULL) { |
---|
626 | outergain = * (ALfloat *) temp; |
---|
627 | } else { |
---|
628 | _alSourceGetParamDefault( AL_CONE_OUTER_GAIN, &outergain ); |
---|
629 | } |
---|
630 | |
---|
631 | _alDebug(ALD_SOURCE, __FILE__, __LINE__, |
---|
632 | "alf_coning: sid %d icone %f ocone %f", src->sid, icone, ocone ); |
---|
633 | |
---|
634 | theta = _alVectorAngleBetween(sp, lp, srcDir); |
---|
635 | |
---|
636 | if( theta <= (icone / 2.0f) ) { |
---|
637 | /* |
---|
638 | * INSIDE: |
---|
639 | * |
---|
640 | * attenuate normally |
---|
641 | */ |
---|
642 | _alDebug(ALD_SOURCE, __FILE__, __LINE__, |
---|
643 | "alf_coning: sid %d theta %f INSIDE", |
---|
644 | src->sid, theta ); |
---|
645 | |
---|
646 | /* |
---|
647 | * speaker[i] is in inner cone, don't do |
---|
648 | * anything. |
---|
649 | */ |
---|
650 | sa = compute_sa( sp, smax, ref, gain, rolloff, lp, df ); |
---|
651 | } else if( theta <= ( ocone / 2.0f) ) { |
---|
652 | /* |
---|
653 | * BETWEEN: |
---|
654 | * |
---|
655 | * kind of cheesy, but we average the INSIDE |
---|
656 | * and OUTSIDE attenuation. |
---|
657 | */ |
---|
658 | _alDebug(ALD_SOURCE, __FILE__, __LINE__, |
---|
659 | "alf_coning: sid %d theta %f BETWEEN", |
---|
660 | src->sid, theta); |
---|
661 | |
---|
662 | sa = compute_sa( sp, smax, ref, gain, rolloff, lp, df ); |
---|
663 | |
---|
664 | sa += _AL_CUTTOFF_ATTENUATION; |
---|
665 | sa /= 2; |
---|
666 | } else { |
---|
667 | /* |
---|
668 | * OUTSIDE: |
---|
669 | * |
---|
670 | * Set to attenuation_min |
---|
671 | */ |
---|
672 | _alDebug(ALD_SOURCE, __FILE__, __LINE__, |
---|
673 | "alf_coning: sid %d theta %f OUTSIDE", |
---|
674 | src->sid, theta ); |
---|
675 | |
---|
676 | if( outergain < _AL_CUTTOFF_ATTENUATION ) { |
---|
677 | sa = _AL_CUTTOFF_ATTENUATION; |
---|
678 | } else { |
---|
679 | sa = outergain; |
---|
680 | } |
---|
681 | } |
---|
682 | |
---|
683 | for(i = 0; i < nc; i++) { |
---|
684 | /* set gain, to be applied in SourceParamApply */ |
---|
685 | src->srcParams.gain[i] *= sa; |
---|
686 | } |
---|
687 | |
---|
688 | return; |
---|
689 | } |
---|
690 | |
---|
691 | /* |
---|
692 | * alf_reverb |
---|
693 | * |
---|
694 | * As far as reverb implementations go, this sucks. Should be |
---|
695 | * frequency based? |
---|
696 | * |
---|
697 | * Should be able to be applied in sequence for second order |
---|
698 | * approximations. |
---|
699 | * |
---|
700 | * FIXME: this is so ugly! And consumes a ton of memory. |
---|
701 | */ |
---|
702 | void alf_reverb( UNUSED(ALuint cid), |
---|
703 | AL_source *src, |
---|
704 | AL_buffer *samp, |
---|
705 | ALshort **buffers, |
---|
706 | ALuint nc, |
---|
707 | ALuint len ) { |
---|
708 | ALshort *bpt; /* pointer to passed buffers */ |
---|
709 | ALshort *rpt; /* pointer to reverb buffers */ |
---|
710 | ALuint i; |
---|
711 | ALfloat scale = src->reverb_scale; |
---|
712 | ALuint delay = src->reverb_delay; |
---|
713 | ALuint k; |
---|
714 | int sample; |
---|
715 | |
---|
716 | /* with a delay of 0.0, no reverb possible or needed */ |
---|
717 | if(!(src->flags & ALS_REVERB)) { |
---|
718 | return; |
---|
719 | } |
---|
720 | |
---|
721 | /* |
---|
722 | * initialize persistent reverb buffers if they haven't been |
---|
723 | * done before |
---|
724 | */ |
---|
725 | for(i = 0; i < nc; i++) { |
---|
726 | if(src->reverb_buf[i] == NULL) { |
---|
727 | src->reverb_buf[i] = malloc(samp->size); |
---|
728 | memset(src->reverb_buf[i], 0, samp->size); |
---|
729 | } |
---|
730 | } |
---|
731 | |
---|
732 | if(src->srcParams.soundpos > delay) { |
---|
733 | int revoffset = ((src->srcParams.soundpos - delay) / sizeof(ALshort)); |
---|
734 | |
---|
735 | for(i = 0; i < nc; i++) { |
---|
736 | bpt = buffers[i]; |
---|
737 | rpt = src->reverb_buf[i]; |
---|
738 | rpt += revoffset; |
---|
739 | |
---|
740 | for(k = 0; k < len / sizeof(ALshort); k++) { |
---|
741 | sample = bpt[k] + rpt[k] * scale; |
---|
742 | |
---|
743 | if(sample > canon_max) { |
---|
744 | sample = canon_max; |
---|
745 | } else if (sample < canon_min) { |
---|
746 | sample = canon_min; |
---|
747 | } |
---|
748 | |
---|
749 | bpt[k] = sample; |
---|
750 | } |
---|
751 | } |
---|
752 | } |
---|
753 | |
---|
754 | _alBuffersAppend(src->reverb_buf, |
---|
755 | (void **) buffers, len, src->reverbpos, nc); |
---|
756 | |
---|
757 | src->reverbpos += len; |
---|
758 | |
---|
759 | return; |
---|
760 | } |
---|
761 | |
---|
762 | /* |
---|
763 | * alf_da |
---|
764 | * |
---|
765 | * alf_da implements distance attenuation. We call compute_sa to get the |
---|
766 | * per-speaker attenuation for each channel, and manipulate the srcParam gain |
---|
767 | * settings to effect that computation. |
---|
768 | * |
---|
769 | * alf_da returns early if we discover that the source has |
---|
770 | * either CONE_INNER_ANGLE or CONE_OUTER_ANGLE set (ie, is a |
---|
771 | * directional source). In those cases, alf_coning should do |
---|
772 | * the distance attenuation. |
---|
773 | * |
---|
774 | * assumes locked source |
---|
775 | * |
---|
776 | * FIXME: |
---|
777 | * Remind me to clean this up. |
---|
778 | */ |
---|
779 | void alf_da( ALuint cid, |
---|
780 | AL_source *src, |
---|
781 | UNUSED(AL_buffer *samp), |
---|
782 | UNUSED(ALshort **buffers), |
---|
783 | ALuint nc, |
---|
784 | UNUSED(ALuint len)) { |
---|
785 | AL_context *cc; |
---|
786 | ALfloat *sp; /* source position */ |
---|
787 | ALfloat sa; /* speaker attenuation */ |
---|
788 | ALfloat listener_position[3]; |
---|
789 | ALfloat *temp; |
---|
790 | ALuint i; |
---|
791 | ALfloat (*df)( ALfloat dist, ALfloat rolloff, ALfloat ref, ALfloat max ); /* distance model func */ |
---|
792 | ALfloat gain; /* source specific gain */ |
---|
793 | ALfloat ref; /* source specific ref distance */ |
---|
794 | ALfloat smax; /* source specific max distance */ |
---|
795 | ALfloat rolloff; /* source specific rolloff factor */ |
---|
796 | |
---|
797 | /* get distance scale */ |
---|
798 | _alcLockContext( cid ); |
---|
799 | cc = _alcGetContext(cid); |
---|
800 | if(cc == NULL) { |
---|
801 | _alcUnlockContext( cid ); |
---|
802 | |
---|
803 | /* ugh. bad context id */ |
---|
804 | return; |
---|
805 | } |
---|
806 | |
---|
807 | df = cc->distance_func; |
---|
808 | |
---|
809 | _alcUnlockContext( cid ); |
---|
810 | |
---|
811 | /* |
---|
812 | * |
---|
813 | * The source specific max is set to max at this point, but may be |
---|
814 | * altered below of the application has set it. |
---|
815 | */ |
---|
816 | smax = FLT_MAX; |
---|
817 | |
---|
818 | /* |
---|
819 | * if coning is enabled for this source, then we want to |
---|
820 | * let the coning filter take care of attenuating since |
---|
821 | * it has more information then we do. |
---|
822 | * |
---|
823 | * We check the direction flag because coning may not |
---|
824 | * be set (ie, they use defaults) |
---|
825 | */ |
---|
826 | temp = _alGetSourceParam(src, AL_DIRECTION); |
---|
827 | if(temp != NULL) { |
---|
828 | /* |
---|
829 | * This sound has it's direction set, so leave it |
---|
830 | * to the coning filter. |
---|
831 | */ |
---|
832 | _alDebug( ALD_SOURCE, __FILE__, __LINE__, |
---|
833 | "Directional sound, probably not right" ); |
---|
834 | |
---|
835 | return; |
---|
836 | } |
---|
837 | |
---|
838 | /* ambient near listener */ |
---|
839 | alGetListenerfv(AL_POSITION, listener_position); |
---|
840 | |
---|
841 | sp = _alGetSourceParam( src, AL_POSITION ); |
---|
842 | if(sp == NULL) { |
---|
843 | /* |
---|
844 | * As an optimization, don't do any attenuation if |
---|
845 | * the source is relative and there's no position. |
---|
846 | */ |
---|
847 | ALboolean *isrel; |
---|
848 | |
---|
849 | isrel = _alGetSourceParam( src, AL_SOURCE_RELATIVE ); |
---|
850 | if ( isrel && *isrel ) { |
---|
851 | ALfloat *gp = _alGetSourceParam(src, AL_GAIN); |
---|
852 | |
---|
853 | if(gp) |
---|
854 | { |
---|
855 | for(i = 0; i < _ALC_MAX_CHANNELS; i++) |
---|
856 | { |
---|
857 | src->srcParams.gain[i] *= *gp; |
---|
858 | } |
---|
859 | } |
---|
860 | |
---|
861 | return; |
---|
862 | } |
---|
863 | |
---|
864 | /* |
---|
865 | * no position set, so set it to the listener |
---|
866 | * postition. We should probably set it to |
---|
867 | * 0.0, 0.0, 0.0 instead. |
---|
868 | * |
---|
869 | * We fall through to get the MIN/MAX |
---|
870 | */ |
---|
871 | sp = listener_position; |
---|
872 | |
---|
873 | _alDebug(ALD_SOURCE, __FILE__, __LINE__, |
---|
874 | "alf_da: setting to listener pos, probably not right"); |
---|
875 | } |
---|
876 | |
---|
877 | /* set reference distance */ |
---|
878 | temp = _alGetSourceParam( src, AL_REFERENCE_DISTANCE ); |
---|
879 | if( temp != NULL ) { |
---|
880 | ref = * (ALfloat *) temp; |
---|
881 | } else { |
---|
882 | _alSourceGetParamDefault( AL_REFERENCE_DISTANCE, &ref ); |
---|
883 | } |
---|
884 | |
---|
885 | /* set source specific gain */ |
---|
886 | temp = _alGetSourceParam( src, AL_GAIN ); |
---|
887 | if( temp != NULL ) { |
---|
888 | gain = * (ALfloat *) temp; |
---|
889 | } else { |
---|
890 | _alSourceGetParamDefault( AL_GAIN, &gain ); |
---|
891 | } |
---|
892 | |
---|
893 | /* set source specific max distance */ |
---|
894 | temp = _alGetSourceParam( src, AL_MAX_DISTANCE ); |
---|
895 | if( temp != NULL ) { |
---|
896 | smax = * (ALfloat *) temp; |
---|
897 | } else { |
---|
898 | _alSourceGetParamDefault( AL_MAX_DISTANCE, &smax ); |
---|
899 | } |
---|
900 | |
---|
901 | /* get source specific rolloff factor */ |
---|
902 | temp = _alGetSourceParam( src, AL_ROLLOFF_FACTOR ); |
---|
903 | if( temp != NULL ) { |
---|
904 | rolloff = * (ALfloat *) temp; |
---|
905 | } else { |
---|
906 | _alSourceGetParamDefault( AL_ROLLOFF_FACTOR, &rolloff ); |
---|
907 | } |
---|
908 | |
---|
909 | sa = compute_sa( sp, smax, ref, gain, rolloff, |
---|
910 | listener_position, df ); |
---|
911 | |
---|
912 | for(i = 0; i < nc; i++) { |
---|
913 | src->srcParams.gain[i] *= sa; |
---|
914 | } |
---|
915 | |
---|
916 | return; |
---|
917 | } |
---|
918 | |
---|
919 | #if USE_TPITCH_LOOKUP |
---|
920 | /* |
---|
921 | * init_tpitch_lookup( ALuint len ) |
---|
922 | * |
---|
923 | * Initializes the tpitch lookup table. See declaration of tpitch_lookup for |
---|
924 | * information on the layout and meaning of tpitch_lookup_init. |
---|
925 | */ |
---|
926 | static void init_tpitch_lookup( ALuint len ) { |
---|
927 | ALfloat scale; |
---|
928 | ALuint i; |
---|
929 | |
---|
930 | if(tpitch_lookup.len >= len) { |
---|
931 | /* We only go through the main loop if we |
---|
932 | * haven't been initialized, or have been |
---|
933 | * initialized with less memory than needed. |
---|
934 | */ |
---|
935 | return; |
---|
936 | } |
---|
937 | tpitch_lookup.len = len; |
---|
938 | |
---|
939 | /* |
---|
940 | * initialize time domain pitch filter lookup table |
---|
941 | */ |
---|
942 | |
---|
943 | /* |
---|
944 | * For pitch < 1.0, we lower the frequency such that a pitch of |
---|
945 | * 0.5 corresponds to 1 octave drop. Is this just a linear |
---|
946 | * application of the step? |
---|
947 | */ |
---|
948 | for(i = 0; i < tpitch_lookup.max; i++) { |
---|
949 | ALfloat pitch; |
---|
950 | ALuint j; |
---|
951 | |
---|
952 | /* set offset part */ |
---|
953 | free(tpitch_lookup.offsets[i]); |
---|
954 | tpitch_lookup.offsets[i] = malloc(sizeof *tpitch_lookup.offsets * len); |
---|
955 | /* set fractional part */ |
---|
956 | free(tpitch_lookup.fractionals[i]); |
---|
957 | tpitch_lookup.fractionals[i] = malloc(sizeof *tpitch_lookup.fractionals * len); |
---|
958 | |
---|
959 | /* set iterate pitch */ |
---|
960 | pitch = 2.0 * ((float)i / (float)tpitch_lookup.max); |
---|
961 | |
---|
962 | /* initialize offset table */ |
---|
963 | scale = 0.0f; |
---|
964 | |
---|
965 | for(j = 0; j < len; j++) |
---|
966 | { |
---|
967 | float foffset = j * pitch; |
---|
968 | ALuint offset = (int) foffset; |
---|
969 | float frac = foffset - offset; |
---|
970 | |
---|
971 | tpitch_lookup.offsets[i][j] = offset; |
---|
972 | tpitch_lookup.fractionals[i][j] = frac; |
---|
973 | } |
---|
974 | } |
---|
975 | |
---|
976 | return; |
---|
977 | } |
---|
978 | #endif |
---|
979 | |
---|
980 | /* |
---|
981 | * alf_tdoppler |
---|
982 | * |
---|
983 | * This filter acts out the doppler effects, in the time domain as |
---|
984 | * opposed to frequency domain. This filter works by computing the pitch |
---|
985 | * required to represent the doppler shift, and setting the AL_PITCH attribute |
---|
986 | * of the source directly. |
---|
987 | * |
---|
988 | * FIXME: |
---|
989 | * It's not a good idea to mess with src's pitch. Some method of |
---|
990 | * expressing this computation without changing the source's attributes |
---|
991 | * should be used. |
---|
992 | * |
---|
993 | */ |
---|
994 | void alf_tdoppler( ALuint cid, |
---|
995 | AL_source *src, |
---|
996 | UNUSED(AL_buffer *samp), |
---|
997 | UNUSED(ALshort **buffers), |
---|
998 | UNUSED(ALuint nc), |
---|
999 | UNUSED(ALuint len) ) { |
---|
1000 | AL_context *cc; |
---|
1001 | ALfloat *sv; /* source velocity */ |
---|
1002 | ALfloat *sp; /* source position */ |
---|
1003 | ALfloat lv[3]; /* listener velocity */ |
---|
1004 | ALfloat lp[3]; /* listener position */ |
---|
1005 | ALfloat relative_velocity; /* speed of source wrt listener */ |
---|
1006 | #if 0 |
---|
1007 | ALfloat zeros[] = { 0.0, 0.0, 0.0 }; |
---|
1008 | #endif |
---|
1009 | AL_sourcestate *srcstate; |
---|
1010 | ALfloat doppler_factor; |
---|
1011 | ALfloat doppler_velocity; |
---|
1012 | ALfloat doppler_pitch; |
---|
1013 | |
---|
1014 | /* initialize the mixrate */ |
---|
1015 | if(src->pitch.isset == AL_TRUE) |
---|
1016 | { |
---|
1017 | src->mixrate = src->pitch.data; |
---|
1018 | } |
---|
1019 | else |
---|
1020 | { |
---|
1021 | src->mixrate = 1.0; |
---|
1022 | } |
---|
1023 | |
---|
1024 | /* lock context, get context specific stuff */ |
---|
1025 | _alcLockContext( cid ); |
---|
1026 | |
---|
1027 | cc = _alcGetContext(cid); |
---|
1028 | if( cc == NULL ) { |
---|
1029 | /* cid is an invalid context id. */ |
---|
1030 | _alcUnlockContext( cid ); |
---|
1031 | |
---|
1032 | return; |
---|
1033 | } |
---|
1034 | |
---|
1035 | doppler_factor = cc->doppler_factor; |
---|
1036 | doppler_velocity = cc->doppler_velocity; |
---|
1037 | |
---|
1038 | _alcUnlockContext( cid ); |
---|
1039 | |
---|
1040 | alGetListenerfv(AL_VELOCITY, lv); |
---|
1041 | alGetListenerfv(AL_POSITION, lp); |
---|
1042 | |
---|
1043 | sp = _alGetSourceParam(src, AL_POSITION ); |
---|
1044 | sv = _alGetSourceParam(src, AL_VELOCITY ); |
---|
1045 | |
---|
1046 | if(sp == NULL) { |
---|
1047 | return; |
---|
1048 | } |
---|
1049 | |
---|
1050 | if(sv == NULL) { |
---|
1051 | /* no velocity set, no doppler effect */ |
---|
1052 | return; |
---|
1053 | } |
---|
1054 | |
---|
1055 | if (fabs(doppler_factor) < 1.0E-6f) { |
---|
1056 | /* doppler factor set to zero, no doppler effect */ |
---|
1057 | return; |
---|
1058 | } |
---|
1059 | |
---|
1060 | #if 0 |
---|
1061 | /* ToDo: duplicate test */ |
---|
1062 | if(sv == NULL) { |
---|
1063 | /* |
---|
1064 | * if unset, set to the velocity to the |
---|
1065 | * zero vector. |
---|
1066 | */ |
---|
1067 | sv = zeros; |
---|
1068 | } |
---|
1069 | #endif |
---|
1070 | |
---|
1071 | relative_velocity = _alVectorMagnitude(sv, lv); |
---|
1072 | if(relative_velocity == 0.0) { |
---|
1073 | /* |
---|
1074 | * no relative velocity, no doppler |
---|
1075 | * |
---|
1076 | * FIXME: use epsilon |
---|
1077 | */ |
---|
1078 | |
---|
1079 | return; |
---|
1080 | } |
---|
1081 | |
---|
1082 | |
---|
1083 | srcstate = _alSourceQueueGetCurrentState(src); |
---|
1084 | if(srcstate == NULL) { |
---|
1085 | fprintf(stderr, "weird\n"); |
---|
1086 | } |
---|
1087 | |
---|
1088 | doppler_pitch = compute_doppler_pitch(lp, lv, sp, sv, |
---|
1089 | doppler_factor, doppler_velocity); |
---|
1090 | |
---|
1091 | src->mixrate *= doppler_pitch; |
---|
1092 | |
---|
1093 | #ifdef DEBUG |
---|
1094 | if(src->mixrate < MIN_PITCH) |
---|
1095 | { |
---|
1096 | _alDebug(ALD_FILTER, __FILE__, __LINE__, |
---|
1097 | "Clamping src->mixrate %f\n", |
---|
1098 | src->mixrate); |
---|
1099 | } |
---|
1100 | #endif |
---|
1101 | |
---|
1102 | src->mixrate = MAX(src->mixrate, MIN_PITCH); |
---|
1103 | src->mixrate = MIN(src->mixrate, 2.0f); |
---|
1104 | |
---|
1105 | return; |
---|
1106 | } |
---|
1107 | |
---|
1108 | /* |
---|
1109 | * alf_minmax |
---|
1110 | * |
---|
1111 | * Implements min/max gain. First min is applied, then max. |
---|
1112 | */ |
---|
1113 | void alf_minmax( UNUSED(ALuint cid), |
---|
1114 | AL_source *src, |
---|
1115 | UNUSED(AL_buffer *samp), |
---|
1116 | UNUSED(ALshort **buffers), |
---|
1117 | ALuint nc, |
---|
1118 | UNUSED(ALuint len) ) { |
---|
1119 | ALfloat *amaxp = _alGetSourceParam( src, AL_MAX_GAIN ); |
---|
1120 | ALfloat *aminp = _alGetSourceParam( src, AL_MIN_GAIN ); |
---|
1121 | ALfloat attenuation_min; |
---|
1122 | ALfloat attenuation_max; |
---|
1123 | ALuint i; |
---|
1124 | |
---|
1125 | /* |
---|
1126 | * if min or max are set, use them. Otherwise, keep defaults |
---|
1127 | */ |
---|
1128 | if(aminp != NULL) { |
---|
1129 | attenuation_min = *aminp; |
---|
1130 | } else { |
---|
1131 | _alSourceGetParamDefault( AL_MIN_GAIN, &attenuation_min ); |
---|
1132 | } |
---|
1133 | |
---|
1134 | if(amaxp != NULL) { |
---|
1135 | attenuation_max = *amaxp; |
---|
1136 | } else { |
---|
1137 | _alSourceGetParamDefault( AL_MAX_GAIN, &attenuation_max ); |
---|
1138 | } |
---|
1139 | |
---|
1140 | for(i = 0; i < nc; i++) { |
---|
1141 | if( src->srcParams.gain[i] > attenuation_max ) { |
---|
1142 | src->srcParams.gain[i] = attenuation_max; |
---|
1143 | } else if( src->srcParams.gain[i] < attenuation_min ) { |
---|
1144 | src->srcParams.gain[i] = attenuation_min; |
---|
1145 | } |
---|
1146 | } |
---|
1147 | |
---|
1148 | return; |
---|
1149 | } |
---|
1150 | |
---|
1151 | /* |
---|
1152 | * alf_listenergain |
---|
1153 | * |
---|
1154 | * Implements listener gain. |
---|
1155 | */ |
---|
1156 | void |
---|
1157 | alf_listenergain( UNUSED(ALuint cid), |
---|
1158 | AL_source *src, |
---|
1159 | UNUSED(AL_buffer *samp), |
---|
1160 | UNUSED(ALshort **buffers), |
---|
1161 | ALuint nc, |
---|
1162 | UNUSED(ALuint len) ) |
---|
1163 | { |
---|
1164 | ALfloat gain; |
---|
1165 | ALuint i; |
---|
1166 | alGetListenerfv(AL_GAIN, &gain); |
---|
1167 | for(i = 0; i < nc; i++) { |
---|
1168 | src->srcParams.gain[i] *= gain; |
---|
1169 | } |
---|
1170 | } |
---|
1171 | |
---|
1172 | /* |
---|
1173 | * compute_doppler_pitch( ALfloat *object1, ALfloat *o1_vel, |
---|
1174 | * ALfloat *object2, ALfloat *o2_vel, |
---|
1175 | * ALfloat factor, |
---|
1176 | * ALfloat speed ) |
---|
1177 | * |
---|
1178 | * compute_doppler_pitch is meant to return a value spanning 0.5 to 1.5, |
---|
1179 | * which is meant to simulate the frequency shift undergone by sources |
---|
1180 | * in relative movement wrt the listener. |
---|
1181 | */ |
---|
1182 | static ALfloat compute_doppler_pitch( ALfloat *object1, ALfloat *o1_vel, |
---|
1183 | ALfloat *object2, ALfloat *o2_vel, |
---|
1184 | ALfloat factor, /* doppler_factor */ |
---|
1185 | ALfloat speed ) { /* propagation_speed */ |
---|
1186 | ALfloat between[3]; /* Unit vector pointing in the direction |
---|
1187 | * from one object to the other |
---|
1188 | */ |
---|
1189 | ALfloat obj1V, obj2V; /* Relative scalar velocity components */ |
---|
1190 | ALfloat ratio; /* Ratio of relative velocities */ |
---|
1191 | ALfloat retval; /* final doppler shift */ |
---|
1192 | |
---|
1193 | /* |
---|
1194 | * Set up the "between" vector which points from one object to the |
---|
1195 | * other |
---|
1196 | */ |
---|
1197 | between[0] = object2[0] - object1[0]; |
---|
1198 | between[1] = object2[1] - object1[1]; |
---|
1199 | between[2] = object2[2] - object1[2]; |
---|
1200 | |
---|
1201 | _alVectorNormalize( between, between ); |
---|
1202 | |
---|
1203 | /* |
---|
1204 | * Compute the dot product of the velocity vector and the "between" |
---|
1205 | * vector. |
---|
1206 | * |
---|
1207 | * The _alVectorDotp function is not set up for computing dot products |
---|
1208 | * for actual vectors (it works for three points that define two |
---|
1209 | * vectors from a common origin), so I'll do it here. |
---|
1210 | */ |
---|
1211 | obj1V = o1_vel[0] * between[0]; |
---|
1212 | obj1V += o1_vel[1] * between[1]; |
---|
1213 | obj1V += o1_vel[2] * between[2]; |
---|
1214 | |
---|
1215 | /* Now compute the dot product for the second object */ |
---|
1216 | obj2V = o2_vel[0] * between[0]; |
---|
1217 | obj2V += o2_vel[1] * between[1]; |
---|
1218 | obj2V += o2_vel[2] * between[2]; |
---|
1219 | |
---|
1220 | /* |
---|
1221 | * Apply the Doppler factor by modifying the source and listener |
---|
1222 | * velocities. This will exaggerate or reduce the Doppler |
---|
1223 | * effect as expected. |
---|
1224 | */ |
---|
1225 | obj1V *= factor; |
---|
1226 | obj2V *= factor; |
---|
1227 | |
---|
1228 | /* |
---|
1229 | * Now compute the obj1/obj2 velocity ratio, taking into account |
---|
1230 | * the propagation speed. This formula is straight from the spec. |
---|
1231 | */ |
---|
1232 | obj1V = speed - obj1V; |
---|
1233 | obj2V = speed + obj2V; |
---|
1234 | ratio = obj1V / obj2V; |
---|
1235 | |
---|
1236 | /* Finally, return the ratio */ |
---|
1237 | retval = ratio; |
---|
1238 | |
---|
1239 | return retval; |
---|
1240 | } |
---|
1241 | |
---|
1242 | #if USE_TPITCH_LOOKUP |
---|
1243 | /* |
---|
1244 | * alf_tpitch |
---|
1245 | * |
---|
1246 | * this filter acts out AL_PITCH. |
---|
1247 | * |
---|
1248 | * This filter is implements AL_PITCH, but - oh-ho! - in the |
---|
1249 | * time domain. All that good fft mojo going to waste. |
---|
1250 | */ |
---|
1251 | void alf_tpitch( UNUSED(ALuint cid), |
---|
1252 | AL_source *src, |
---|
1253 | AL_buffer *samp, |
---|
1254 | ALshort **buffers, |
---|
1255 | ALuint nc, |
---|
1256 | ALuint len ) { |
---|
1257 | ALshort *obufptr = NULL; /* pointer to unmolested buffer data */ |
---|
1258 | ALshort *bufptr = NULL; /* pointer to buffers[0..nc-1] */ |
---|
1259 | ALuint l_index; /* index into lookup table */ |
---|
1260 | ALint ipos = 0; /* used to store offsets temporarily */ |
---|
1261 | ALuint i; |
---|
1262 | int *offsets; /* pointer to set of offsets in lookup table */ |
---|
1263 | float *fractionals; /* pointer to set of fractionals in lookup table */ |
---|
1264 | int bufchans; |
---|
1265 | ALfloat pitch; |
---|
1266 | |
---|
1267 | pitch = src->mixrate; |
---|
1268 | |
---|
1269 | if (pitch == 1.0 && !(src->flags & ALS_NEEDPITCH)) { |
---|
1270 | /* |
---|
1271 | * mixrate is at the default, so changing pitch is unnecessary. |
---|
1272 | */ |
---|
1273 | return; |
---|
1274 | } |
---|
1275 | |
---|
1276 | bufchans = _alGetChannelsFromFormat(samp->format); /* we need bufchans to |
---|
1277 | * scale our increment |
---|
1278 | * of the soundpos, |
---|
1279 | * because of |
---|
1280 | * multichannel format |
---|
1281 | * buffers. |
---|
1282 | */ |
---|
1283 | /* |
---|
1284 | * if pitch is out of range, return. |
---|
1285 | */ |
---|
1286 | if(pitch <= 0.0f) |
---|
1287 | { |
---|
1288 | _alDebug(ALD_FILTER, __FILE__, __LINE__, |
---|
1289 | "pitch out of range: %f, clamping", pitch); |
---|
1290 | pitch = 0.05f; |
---|
1291 | } |
---|
1292 | else if (pitch > 2.0f) |
---|
1293 | { |
---|
1294 | _alDebug(ALD_FILTER, __FILE__, __LINE__, |
---|
1295 | "pitch out of range: %f, clamping", pitch); |
---|
1296 | pitch = 2.0f; |
---|
1297 | } |
---|
1298 | |
---|
1299 | if(_alBufferIsCallback(samp) == AL_TRUE) { |
---|
1300 | /* just debugging here, remove this block */ |
---|
1301 | |
---|
1302 | _alDebug(ALD_BUFFER, __FILE__, __LINE__, |
---|
1303 | "No tpitch support for callbacks yet"); |
---|
1304 | |
---|
1305 | /* _alSetError(cid, AL_INVALID_OPERATION); */ |
---|
1306 | return; |
---|
1307 | } |
---|
1308 | |
---|
1309 | /* |
---|
1310 | * We need len in samples, not bytes. |
---|
1311 | */ |
---|
1312 | len /= sizeof(ALshort); |
---|
1313 | |
---|
1314 | /* convert pitch into index in our lookup table */ |
---|
1315 | l_index = (pitch / 2.0) * tpitch_lookup.max; |
---|
1316 | |
---|
1317 | /* |
---|
1318 | * sanity check. |
---|
1319 | */ |
---|
1320 | if(l_index >= tpitch_lookup.max) { |
---|
1321 | l_index = tpitch_lookup.max - 1; |
---|
1322 | } |
---|
1323 | |
---|
1324 | _alDebug(ALD_FILTER, __FILE__, __LINE__, |
---|
1325 | "pitch %f l_index %d", pitch, l_index); |
---|
1326 | |
---|
1327 | /* |
---|
1328 | * offsets is our set of pitch-scaled offsets, 0...pitch * len. |
---|
1329 | * |
---|
1330 | * Well, sort of. 0...pitch * len, but with len scaled such |
---|
1331 | * that we don't suffer a overrun if the buffer's original |
---|
1332 | * data is too short. |
---|
1333 | */ |
---|
1334 | offsets = tpitch_lookup.offsets[ l_index ]; |
---|
1335 | |
---|
1336 | #ifdef DEBUG_MEM |
---|
1337 | assert(l_index < TPITCH_MAX); |
---|
1338 | #endif |
---|
1339 | |
---|
1340 | /* |
---|
1341 | * Iterate over each buffers[0..nc-1] |
---|
1342 | */ |
---|
1343 | for(i = 0; i < nc; i++) { |
---|
1344 | ALint clen = len; |
---|
1345 | int j; |
---|
1346 | |
---|
1347 | /* |
---|
1348 | * Kind of breaking convention here and actually using |
---|
1349 | * the original buffer data instead of just resampling |
---|
1350 | * inside the passed buffer data. This is because we |
---|
1351 | * won't have enough data to resample pitch > 1.0. |
---|
1352 | * |
---|
1353 | * We offset our original buffer pointer by the source's |
---|
1354 | * current position, but in samples, not in bytes |
---|
1355 | * (which is what src->srcParams.soundpos is in). |
---|
1356 | */ |
---|
1357 | obufptr = samp->orig_buffers[i]; |
---|
1358 | obufptr += src->srcParams.soundpos / sizeof *obufptr; |
---|
1359 | |
---|
1360 | #ifdef DEBUG_MEM |
---|
1361 | assert(samp->orig_buffers[i]); |
---|
1362 | assert(src->srcParams.soundpos < samp->size); |
---|
1363 | #endif |
---|
1364 | |
---|
1365 | if(l_index == tpitch_lookup.middle ) { |
---|
1366 | /* when this predicate is true, the pitch is |
---|
1367 | * equal to 1, which means there is no change. |
---|
1368 | * Therefore, we short circuit. |
---|
1369 | * |
---|
1370 | * Because we're incrementing the soundpos here, |
---|
1371 | * we can't just return. |
---|
1372 | */ |
---|
1373 | |
---|
1374 | continue; |
---|
1375 | } |
---|
1376 | |
---|
1377 | /* |
---|
1378 | * set bufptr to the pcm channel that we |
---|
1379 | * are about to change in-place. |
---|
1380 | */ |
---|
1381 | bufptr = buffers[i]; |
---|
1382 | |
---|
1383 | /* |
---|
1384 | * We mess with offsets in the loop below, so reset it |
---|
1385 | * after each iteration. |
---|
1386 | */ |
---|
1387 | offsets = tpitch_lookup.offsets[ l_index ]; |
---|
1388 | fractionals = tpitch_lookup.fractionals[ l_index ]; |
---|
1389 | |
---|
1390 | /* don't run past end */ |
---|
1391 | if(((clen + 1) * pitch * sizeof(ALshort)) >= |
---|
1392 | (samp->size - src->srcParams.soundpos)) |
---|
1393 | { |
---|
1394 | clen = samp->size - src->srcParams.soundpos; |
---|
1395 | clen /= pitch; |
---|
1396 | clen /= sizeof(ALshort); |
---|
1397 | clen -= 1; |
---|
1398 | } |
---|
1399 | |
---|
1400 | /* |
---|
1401 | * this is where the "resampling" takes place. We do a |
---|
1402 | * very little bit on unrolling here, and it shouldn't |
---|
1403 | * be necessary, but seems to improve performance quite |
---|
1404 | * a bit. |
---|
1405 | */ |
---|
1406 | for(j = 0; j < clen; j++) |
---|
1407 | { |
---|
1408 | #if USE_LRINT |
---|
1409 | { |
---|
1410 | int offset = offsets[j]; |
---|
1411 | int nextoffset = offsets[j+1]; |
---|
1412 | float frac = fractionals[j]; |
---|
1413 | float firstsample = obufptr[offset]; |
---|
1414 | float nextsample = obufptr[nextoffset]; |
---|
1415 | int finalsample; |
---|
1416 | |
---|
1417 | /* do a little interpolation */ |
---|
1418 | finalsample = lrintf(firstsample + |
---|
1419 | frac * (nextsample - firstsample)); |
---|
1420 | |
---|
1421 | finalsample = MIN(finalsample, canon_max); |
---|
1422 | bufptr[j] = MAX(finalsample, canon_min); |
---|
1423 | } |
---|
1424 | #else |
---|
1425 | { |
---|
1426 | int offset = offsets[j]; |
---|
1427 | int nextoffset = offsets[j+1]; |
---|
1428 | float frac = fractionals[j]; |
---|
1429 | int firstsample = obufptr[offset]; |
---|
1430 | int nextsample = obufptr[nextoffset]; |
---|
1431 | int finalsample; |
---|
1432 | |
---|
1433 | /* do a little interpolation */ |
---|
1434 | finalsample = firstsample + |
---|
1435 | frac * (nextsample - firstsample); |
---|
1436 | |
---|
1437 | finalsample = MIN(finalsample, canon_max); |
---|
1438 | bufptr[j] = MAX(finalsample, canon_min); |
---|
1439 | } |
---|
1440 | #endif |
---|
1441 | } |
---|
1442 | |
---|
1443 | /* zero off end */ |
---|
1444 | memset(&bufptr[j], 0, (len-j)*sizeof *bufptr); |
---|
1445 | } |
---|
1446 | |
---|
1447 | /* |
---|
1448 | * Set offsets to a known good state. |
---|
1449 | */ |
---|
1450 | offsets = tpitch_lookup.offsets[l_index]; |
---|
1451 | |
---|
1452 | /* |
---|
1453 | * AL_PITCH (well, alf_tpitch actually) require that the |
---|
1454 | * main mixer func does not increment the source's soundpos, |
---|
1455 | * so we must increment it here. If we detect an overrun, we |
---|
1456 | * must reset the src's soundpos to something reasonable. |
---|
1457 | */ |
---|
1458 | ipos = (int) (len * pitch); |
---|
1459 | src->srcParams.soundpos += bufchans * ipos * sizeof(ALshort); |
---|
1460 | |
---|
1461 | if(src->srcParams.soundpos > samp->size) |
---|
1462 | { |
---|
1463 | /* |
---|
1464 | * we've reached the end of this sample. |
---|
1465 | * |
---|
1466 | * Since we're handling the soundpos incrementing for |
---|
1467 | * this source (usually done in _alMixSources), we have |
---|
1468 | * to handle all the special cases here instead of |
---|
1469 | * delegating them. |
---|
1470 | * |
---|
1471 | * These include callback, looping, and streaming |
---|
1472 | * sources. For now, we just handle looping and |
---|
1473 | * normal sources, as callback sources will probably |
---|
1474 | * require added some special case logic to _alSplitSources |
---|
1475 | * to give up a little more breathing room. |
---|
1476 | */ |
---|
1477 | if( _alSourceIsLooping( src ) == AL_TRUE ) { |
---|
1478 | /* |
---|
1479 | * looping source |
---|
1480 | * |
---|
1481 | * FIXME: |
---|
1482 | * This isn't right. soundpos should be set to |
---|
1483 | * something different, and we may need to carry |
---|
1484 | * over info so that the sound loops properly. |
---|
1485 | */ |
---|
1486 | |
---|
1487 | /* FIXME: kind of kludgy */ |
---|
1488 | src->srcParams.soundpos = 0; |
---|
1489 | } else { |
---|
1490 | /* |
---|
1491 | * let _alMixSources know it's time for this source |
---|
1492 | * to die. |
---|
1493 | */ |
---|
1494 | _alDebug(ALD_FILTER, __FILE__, __LINE__, |
---|
1495 | "tpitch: source ending"); |
---|
1496 | src->srcParams.soundpos = samp->size; |
---|
1497 | } |
---|
1498 | } |
---|
1499 | |
---|
1500 | return; |
---|
1501 | } |
---|
1502 | #else |
---|
1503 | /* |
---|
1504 | * alf_tpitch |
---|
1505 | * |
---|
1506 | * this filter acts out AL_PITCH. |
---|
1507 | * |
---|
1508 | * This filter is implements AL_PITCH, but - oh-ho! - in the |
---|
1509 | * time domain. All that good fft mojo going to waste. |
---|
1510 | */ |
---|
1511 | void alf_tpitch( UNUSED(ALuint cid), |
---|
1512 | AL_source *src, |
---|
1513 | AL_buffer *samp, |
---|
1514 | ALshort **buffers, |
---|
1515 | ALuint nc, |
---|
1516 | ALuint len ) |
---|
1517 | { |
---|
1518 | ALshort *obufptr = NULL; /* pointer to unmolested buffer data */ |
---|
1519 | ALshort *bufptr = NULL; /* pointer to buffers[0..nc-1] */ |
---|
1520 | ALint ipos = 0; /* used to store offsets temporarily */ |
---|
1521 | ALuint i; |
---|
1522 | ALuint clen; |
---|
1523 | int bufchans; |
---|
1524 | ALfloat pitch; |
---|
1525 | |
---|
1526 | pitch = src->mixrate; |
---|
1527 | |
---|
1528 | if (pitch == 1.0 && !(src->flags & ALS_NEEDPITCH)) { |
---|
1529 | /* |
---|
1530 | * mixrate is at the default, so changing pitch is unnecessary. |
---|
1531 | */ |
---|
1532 | return; |
---|
1533 | } |
---|
1534 | |
---|
1535 | bufchans = _alGetChannelsFromFormat(samp->format); /* we need bufchans to |
---|
1536 | * scale our increment |
---|
1537 | * of the soundpos, |
---|
1538 | * because of |
---|
1539 | * multichannel format |
---|
1540 | * buffers. |
---|
1541 | */ |
---|
1542 | /* |
---|
1543 | * if pitch is out of range, clamp. |
---|
1544 | */ |
---|
1545 | pitch = MIN(pitch, 2.0f); |
---|
1546 | pitch = MAX(pitch, MIN_PITCH); |
---|
1547 | |
---|
1548 | /* |
---|
1549 | * We need len in samples, not bytes. |
---|
1550 | */ |
---|
1551 | len /= sizeof(ALshort); |
---|
1552 | |
---|
1553 | _alDebug(ALD_FILTER, __FILE__, __LINE__, "pitch %f", pitch); |
---|
1554 | |
---|
1555 | /* |
---|
1556 | * Iterate over each buffers[0..nc-1] |
---|
1557 | */ |
---|
1558 | for(i = 0; i < nc; i++) { |
---|
1559 | ALuint j; |
---|
1560 | |
---|
1561 | if(pitch == 1.0f) |
---|
1562 | { |
---|
1563 | continue; |
---|
1564 | } |
---|
1565 | |
---|
1566 | /* |
---|
1567 | * Kind of breaking convention here and actually using |
---|
1568 | * the original buffer data instead of just resampling |
---|
1569 | * inside the passed buffer data. This is because we |
---|
1570 | * won't have enough data to resample pitch > 1.0. |
---|
1571 | * |
---|
1572 | * We offset our original buffer pointer by the source's |
---|
1573 | * current position, but in samples, not in bytes |
---|
1574 | * (which is what src->srcParams.soundpos is in). |
---|
1575 | */ |
---|
1576 | obufptr = samp->orig_buffers[i]; |
---|
1577 | obufptr += src->srcParams.soundpos / sizeof *obufptr; |
---|
1578 | |
---|
1579 | /* |
---|
1580 | * set bufptr to the pcm channel that we |
---|
1581 | * are about to change in-place. |
---|
1582 | */ |
---|
1583 | bufptr = buffers[i]; |
---|
1584 | |
---|
1585 | clen = len; |
---|
1586 | |
---|
1587 | /* don't run past end */ |
---|
1588 | if(((clen + 1) * pitch * sizeof(ALshort)) >= |
---|
1589 | (samp->size - src->srcParams.soundpos)) |
---|
1590 | { |
---|
1591 | clen = samp->size - src->srcParams.soundpos; |
---|
1592 | clen /= pitch; |
---|
1593 | clen /= sizeof(ALshort); |
---|
1594 | clen -= 1; |
---|
1595 | } |
---|
1596 | |
---|
1597 | /* |
---|
1598 | * this is where the "resampling" takes place. We do a |
---|
1599 | * very little bit on unrolling here, and it shouldn't |
---|
1600 | * be necessary, but seems to improve performance quite |
---|
1601 | * a bit. |
---|
1602 | */ |
---|
1603 | for(j = 0; j < clen; j++) |
---|
1604 | { |
---|
1605 | /* make sure we don't go past end of last source */ |
---|
1606 | #ifdef DEBUG_FILTER |
---|
1607 | assert(((j+1)*pitch)*2 < |
---|
1608 | samp->size - src->srcParams.soundpos); |
---|
1609 | #endif |
---|
1610 | { |
---|
1611 | float foffset = j * pitch; |
---|
1612 | int offset = (int) foffset; |
---|
1613 | float frac = foffset - offset; |
---|
1614 | int firstsample = obufptr[(int) (j * pitch)]; |
---|
1615 | int nextsample = obufptr[(int)((j+1) * pitch)]; |
---|
1616 | int finalsample; |
---|
1617 | |
---|
1618 | /* do a little interpolation */ |
---|
1619 | finalsample = firstsample + |
---|
1620 | frac * (nextsample - firstsample); |
---|
1621 | |
---|
1622 | finalsample = MIN(finalsample, canon_max); |
---|
1623 | bufptr[j] = MAX(finalsample, canon_min); |
---|
1624 | } |
---|
1625 | } |
---|
1626 | |
---|
1627 | /* JIV FIXME: use memset */ |
---|
1628 | for( ; j < len; j++) |
---|
1629 | { |
---|
1630 | bufptr[j] = 0; |
---|
1631 | } |
---|
1632 | } |
---|
1633 | |
---|
1634 | /* |
---|
1635 | * AL_PITCH (well, alf_tpitch actually) require that the |
---|
1636 | * main mixer func does not increment the source's soundpos, |
---|
1637 | * so we must increment it here. If we detect an overrun, we |
---|
1638 | * must reset the src's soundpos to something reasonable. |
---|
1639 | */ |
---|
1640 | ipos = (int) (len * pitch); |
---|
1641 | src->srcParams.soundpos += bufchans * ipos * sizeof(ALshort); |
---|
1642 | |
---|
1643 | if(src->srcParams.soundpos > samp->size) |
---|
1644 | { |
---|
1645 | /* |
---|
1646 | * we've reached the end of this sample. |
---|
1647 | * |
---|
1648 | * Since we're handling the soundpos incrementing for |
---|
1649 | * this source (usually done in _alMixSources), we have |
---|
1650 | * to handle all the special cases here instead of |
---|
1651 | * delegating them. |
---|
1652 | * |
---|
1653 | * These include callback, looping, and streaming |
---|
1654 | * sources. For now, we just handle looping and |
---|
1655 | * normal sources, as callback sources will probably |
---|
1656 | * require added some special case logic to _alSplitSources |
---|
1657 | * to give up a little more breathing room. |
---|
1658 | */ |
---|
1659 | if( _alSourceIsLooping( src ) == AL_TRUE ) { |
---|
1660 | /* |
---|
1661 | * looping source |
---|
1662 | * |
---|
1663 | * FIXME: |
---|
1664 | * This isn't right. soundpos should be set to |
---|
1665 | * something different, and we may need to carry |
---|
1666 | * over info so that the sound loops properly. |
---|
1667 | */ |
---|
1668 | |
---|
1669 | /* FIXME: kind of kludgy */ |
---|
1670 | src->srcParams.soundpos = 0; |
---|
1671 | } else { |
---|
1672 | /* |
---|
1673 | * let _alMixSources know it's time for this source |
---|
1674 | * to die. |
---|
1675 | */ |
---|
1676 | _alDebug(ALD_FILTER, __FILE__, __LINE__, |
---|
1677 | "tpitch: source ending"); |
---|
1678 | |
---|
1679 | src->srcParams.soundpos = samp->size; |
---|
1680 | } |
---|
1681 | } |
---|
1682 | |
---|
1683 | return; |
---|
1684 | } |
---|
1685 | #endif |
---|
1686 | |
---|
1687 | |
---|
1688 | /* |
---|
1689 | * compute_sa( ALfloat *source_pos, ALfloat source_max, |
---|
1690 | * ALfloat source_ref, ALfloat source_gain, |
---|
1691 | * ALfloat source_rolloff, |
---|
1692 | * ALfloat *speaker_pos, |
---|
1693 | * ALfloat (*df)( ALfloat dist, ALfloat rolloff, |
---|
1694 | * ALfloat ref, ALfloat max)) |
---|
1695 | * |
---|
1696 | * computes distance attenuation with respect to a speaker position. |
---|
1697 | * |
---|
1698 | * This is some normalized value which gets expotenially closer to 1.0 |
---|
1699 | * as the source approaches the listener. The minimum attenuation is |
---|
1700 | * AL_CUTTOFF_ATTENUATION, which approached when the source approaches |
---|
1701 | * the max distance. |
---|
1702 | * |
---|
1703 | * source_pos = source position [x/y/z] |
---|
1704 | * source_max = source specific max distance |
---|
1705 | * speaker_pos = speaker position [x/y/z] |
---|
1706 | * ref = source's reference distance |
---|
1707 | * df = distance model function |
---|
1708 | * max = maximum distance, beyond which everything is clamped at |
---|
1709 | * some small value near, but not equal to, zero. |
---|
1710 | */ |
---|
1711 | static ALfloat |
---|
1712 | compute_sa( ALfloat *source_pos, ALfloat source_max, |
---|
1713 | ALfloat source_ref, ALfloat source_gain, |
---|
1714 | ALfloat source_rolloff, |
---|
1715 | ALfloat *speaker_pos, |
---|
1716 | ALfloat (*df)( ALfloat dist, ALfloat rolloff, ALfloat ref, ALfloat max)) |
---|
1717 | { |
---|
1718 | ALfloat retval; |
---|
1719 | |
---|
1720 | /* "Optimize" for rolloff == 0.0 */ |
---|
1721 | if (source_rolloff > 0.0) { |
---|
1722 | ALfloat distance; |
---|
1723 | distance = _alVectorMagnitude( source_pos, speaker_pos ); |
---|
1724 | retval = source_gain * df( distance, source_rolloff, source_ref, source_max ); |
---|
1725 | } else { |
---|
1726 | retval = source_gain; |
---|
1727 | } |
---|
1728 | |
---|
1729 | if( retval > 1.0 ) { |
---|
1730 | return 1.0; |
---|
1731 | } |
---|
1732 | |
---|
1733 | if(retval < _AL_CUTTOFF_ATTENUATION) { |
---|
1734 | return _AL_CUTTOFF_ATTENUATION; |
---|
1735 | } |
---|
1736 | |
---|
1737 | return retval; |
---|
1738 | } |
---|
1739 | |
---|
1740 | /* |
---|
1741 | * alf_panning |
---|
1742 | * |
---|
1743 | */ |
---|
1744 | |
---|
1745 | void alf_panning( ALuint cid, |
---|
1746 | AL_source *src, |
---|
1747 | UNUSED(AL_buffer *samp), |
---|
1748 | UNUSED(ALshort **buffers), |
---|
1749 | ALuint nc, |
---|
1750 | UNUSED(ALuint len) ) { |
---|
1751 | ALfloat lp[3]; /* listener position */ |
---|
1752 | ALfloat *sp; /* source position */ |
---|
1753 | ALfloat *sd; /* speaker position */ |
---|
1754 | ALfloat m; |
---|
1755 | ALfloat sa; |
---|
1756 | ALuint i; |
---|
1757 | |
---|
1758 | alGetListenerfv(AL_POSITION, lp); |
---|
1759 | sp = _alGetSourceParam(src, AL_POSITION ); |
---|
1760 | |
---|
1761 | if ((sp == NULL) || (lp == NULL)) { |
---|
1762 | return; |
---|
1763 | } |
---|
1764 | |
---|
1765 | m = _alVectorMagnitude(lp, sp); |
---|
1766 | if (m == 0) { |
---|
1767 | /* should this use epsilon? */ |
---|
1768 | return; |
---|
1769 | } |
---|
1770 | |
---|
1771 | for (i = 0; i < nc; i++) { |
---|
1772 | sd = _alcGetSpeakerPosition(cid, i); |
---|
1773 | sa = _alVectorDotp(lp, sp, sd) / m; |
---|
1774 | sa += 1.0; |
---|
1775 | |
---|
1776 | src->srcParams.gain[i] *= sa; |
---|
1777 | } |
---|
1778 | } |
---|