[17] | 1 | /* -*- mode: C; tab-width:8; c-basic-offset:8 -*- |
---|
| 2 | * vi:set ts=8: |
---|
| 3 | * |
---|
| 4 | * al_filter.c |
---|
| 5 | * |
---|
| 6 | * Contains filters. |
---|
| 7 | * |
---|
| 8 | * |
---|
| 9 | * Short guide to filters: |
---|
| 10 | * |
---|
| 11 | * Filters all have the prefix alf_<something>. Each filter |
---|
| 12 | * defined in software_time_filters or software_frequency_filters |
---|
| 13 | * is applied to each source that finds its way into the mixer. |
---|
| 14 | * |
---|
| 15 | * ApplyFilters takes a chunk of data from the original buffer |
---|
| 16 | * associated with the passed source. |
---|
| 17 | * |
---|
| 18 | * This chunk is understood to be that block of data samp->_orig_buffer |
---|
| 19 | * offset src->soundpos to src->soundpos + bufsiz, where src is the |
---|
| 20 | * passed AL_source, samp is the AL_buffer associated with src, and |
---|
| 21 | * bufsiz is the length of the chunk of data that we want. It is usually |
---|
| 22 | * set to _AL_DEF_BUFSIZE, unless specified by ALC_BUFFERSIZE in the |
---|
| 23 | * application. |
---|
| 24 | * |
---|
| 25 | * Applying filters to a source does not (should not) change the original |
---|
| 26 | * pcm data. ApplyFilters will split the original pcm data prior to |
---|
| 27 | * calling each filter, and filters should restrict themselves to |
---|
| 28 | * manipulating the passed data. |
---|
| 29 | * |
---|
| 30 | * time domain filters (those defines in software_time_filters) are |
---|
| 31 | * passed: |
---|
| 32 | * ALuint cid |
---|
| 33 | * identifier for the context that this source belongs to |
---|
| 34 | * AL_source *src |
---|
| 35 | * source that the filter should be applied to |
---|
| 36 | * AL_buffer *samp |
---|
| 37 | * buffer that the source is associated with |
---|
| 38 | * ALshort **buffers |
---|
| 39 | * arrays of points to PCM data, one element per |
---|
| 40 | * channel(left/right/etc) |
---|
| 41 | * ALuint nc |
---|
| 42 | * number of elements in buffers |
---|
| 43 | * ALuint len |
---|
| 44 | * byte length of each element in buffers |
---|
| 45 | * |
---|
| 46 | * Filters are expected to alter buffers[0..nc-1] in place. After |
---|
| 47 | * the ApplyFilter iteration is over, the resulting data is mixed into |
---|
| 48 | * the main mix buffer and forgotten. The data altered is cumulative, |
---|
| 49 | * that is to say, if two filters alf_f and alf_g occur in sequential |
---|
| 50 | * order, alf_g will see the pcm data after alf_f has altered it. |
---|
| 51 | * |
---|
| 52 | * FINER POINTS: |
---|
| 53 | * |
---|
| 54 | * A lot of the filters make effects by modulating amplitude and delay. |
---|
| 55 | * Because these changes are cumulative, we can reduce the application |
---|
| 56 | * of amplitude and delay changes to one operation. This is the point |
---|
| 57 | * of SourceParamApply --- filters can make changes to srcParams.gain |
---|
| 58 | * and srcParams.delay in a source and have those changes applied at |
---|
| 59 | * the end of the ApplyFilters call for the source. These values are |
---|
| 60 | * reset to their defaults at the top of the ApplyFilters call. |
---|
| 61 | * |
---|
| 62 | */ |
---|
| 63 | #include "al_siteconfig.h" |
---|
| 64 | |
---|
| 65 | #include <AL/alext.h> |
---|
| 66 | |
---|
| 67 | #include <math.h> |
---|
| 68 | #include <stdlib.h> |
---|
| 69 | #include <string.h> |
---|
| 70 | #include <float.h> |
---|
| 71 | |
---|
| 72 | #include "al_buffer.h" |
---|
| 73 | #include "al_debug.h" |
---|
| 74 | #include "al_error.h" |
---|
| 75 | #include "al_filter.h" |
---|
| 76 | #include "al_listen.h" |
---|
| 77 | #include "al_main.h" |
---|
| 78 | #include "al_mixer.h" |
---|
| 79 | #include "al_source.h" |
---|
| 80 | #include "al_queue.h" |
---|
| 81 | #include "al_vector.h" |
---|
| 82 | |
---|
| 83 | #include "alc/alc_context.h" |
---|
| 84 | #include "alc/alc_speaker.h" |
---|
| 85 | |
---|
| 86 | #define MIN(a,b) (((a) < (b)) ? (a) : (b)) |
---|
| 87 | #define MAX(a,b) (((a) < (b)) ? (b) : (a)) |
---|
| 88 | |
---|
| 89 | #define MIN_PITCH 0.25f |
---|
| 90 | |
---|
| 91 | #define USE_TPITCH_LOOKUP 1 /* icculus change here JIV FIXME */ |
---|
| 92 | |
---|
| 93 | /* |
---|
| 94 | * _AL_CUTTOFF_ATTENUATION is the value below which, sounds are not further |
---|
| 95 | * distance attenuated. The purpose of this culling is to avoid pop-off |
---|
| 96 | * artifacts. |
---|
| 97 | * |
---|
| 98 | * Elias: This has been found to cause insufficient distance attenuation |
---|
| 99 | * and has therefore been effectively disabled by setting it to 0. If no |
---|
| 100 | * problems show up, the value should be completely removed. |
---|
| 101 | * The original was value 0.01 |
---|
| 102 | */ |
---|
| 103 | #define _AL_CUTTOFF_ATTENUATION 0.00 |
---|
| 104 | |
---|
| 105 | /* |
---|
| 106 | * TPITCH_MAX sets the number of discrete values for AL_PITCH we can have. |
---|
| 107 | * You can set AL_PITCH to anything, but integer rounding will ensure that |
---|
| 108 | * it will fall beween MIN_SCALE and 2.0. |
---|
| 109 | * |
---|
| 110 | * 2.0 is an arbitrary constant, and likely to be changed. |
---|
| 111 | */ |
---|
| 112 | #define TPITCH_MAX 256 |
---|
| 113 | |
---|
| 114 | /* |
---|
| 115 | * The default software time domain filters. |
---|
| 116 | * |
---|
| 117 | * I wish I could say that the order of these does not matter, |
---|
| 118 | * but it does. Namely, tdoppler and tpitch must occur in that |
---|
| 119 | * order, and they must occur before any other filter. listenergain must |
---|
| 120 | * occur last. |
---|
| 121 | */ |
---|
| 122 | static time_filter_set software_time_filters[] = { |
---|
| 123 | { "tdoppler", alf_tdoppler }, /* time-domain doppler filter */ |
---|
| 124 | { "tpitch", alf_tpitch }, /* time-domain pitch filter */ |
---|
| 125 | { "da", alf_da }, |
---|
| 126 | { "reverb", alf_reverb }, |
---|
| 127 | { "coning", alf_coning }, |
---|
| 128 | { "panning", alf_panning }, |
---|
| 129 | { "minmax", alf_minmax }, |
---|
| 130 | { "listenergain", alf_listenergain }, |
---|
| 131 | { { 0 }, NULL } |
---|
| 132 | }; |
---|
| 133 | |
---|
| 134 | /* |
---|
| 135 | * compute_sa( ALfloat *source_pos, ALfloat source_max, |
---|
| 136 | * ALfloat source_ref, ALfloat source_gain, |
---|
| 137 | * ALfloat source_rolloff, |
---|
| 138 | * ALfloat *speaker_pos, |
---|
| 139 | * ALfloat (*df)( ALfloat dist, ALfloat rolloff, |
---|
| 140 | * ALfloat ref, ALfloat max)) |
---|
| 141 | * |
---|
| 142 | * computes distance attenuation with respect to a speaker position. |
---|
| 143 | * |
---|
| 144 | * This is some normalized value which gets expotenially closer to 1.0 |
---|
| 145 | * as the source approaches the listener. The minimum attenuation is |
---|
| 146 | * AL_CUTTOFF_ATTENUATION, which approached when the source approaches |
---|
| 147 | * the max distance. |
---|
| 148 | * |
---|
| 149 | * source_pos = source position [x/y/z] |
---|
| 150 | * source_max = source specific max distance |
---|
| 151 | * speaker_pos = speaker position [x/y/z] |
---|
| 152 | * ref = source's reference distance |
---|
| 153 | * df = distance model function |
---|
| 154 | * max = maximum distance, beyond which everything is clamped at |
---|
| 155 | * some small value near, but not equal to, zero. |
---|
| 156 | */ |
---|
| 157 | static ALfloat compute_sa( ALfloat *source_pos, ALfloat source_max, |
---|
| 158 | ALfloat source_ref, ALfloat source_gain, |
---|
| 159 | ALfloat source_rolloff, |
---|
| 160 | ALfloat *speaker_pos, |
---|
| 161 | ALfloat df( ALfloat dist, ALfloat rolloff, ALfloat ref, ALfloat max )); |
---|
| 162 | |
---|
| 163 | #if USE_TPITCH_LOOKUP |
---|
| 164 | /* |
---|
| 165 | * our quick lookup table for our time domain pitch filter. |
---|
| 166 | * |
---|
| 167 | * |
---|
| 168 | * We initialize each element in offsets to be a set of offsets, |
---|
| 169 | * such that |
---|
| 170 | * |
---|
| 171 | * offset[x][y] = int portion of y * pitch |
---|
| 172 | * fractinoal[x][y] = fractional portion of y * pitch |
---|
| 173 | * |
---|
| 174 | * Where x is the discrete integer value of pitch, and y is |
---|
| 175 | * any value between 0 and the length of a buffer. |
---|
| 176 | * |
---|
| 177 | * What's the point? To save the pain of float->int conversion at |
---|
| 178 | * runtime, which is needed to map the original PCM data to a |
---|
| 179 | * "pitch modified" mapping of the same data. |
---|
| 180 | * |
---|
| 181 | */ |
---|
| 182 | static struct { |
---|
| 183 | int *offsets[TPITCH_MAX]; /* use int instead of ALint because |
---|
| 184 | * these are array indexes |
---|
| 185 | */ |
---|
| 186 | float *fractionals[TPITCH_MAX]; |
---|
| 187 | ALuint max; |
---|
| 188 | ALuint middle; /* the index which pitch == 1.0 corresponds to */ |
---|
| 189 | ALuint len; /* length of offsets[0...TPITCH_MAX] in samples */ |
---|
| 190 | } tpitch_lookup = { { NULL }, { NULL }, 0, 0, 0 }; |
---|
| 191 | |
---|
| 192 | /* func associated with our tpitch lookup */ |
---|
| 193 | static void init_tpitch_lookup(ALuint len); |
---|
| 194 | #endif |
---|
| 195 | |
---|
| 196 | static ALfloat compute_doppler_pitch(ALfloat *object1, ALfloat *o1_vel, |
---|
| 197 | ALfloat *object2, ALfloat *o2_vel, |
---|
| 198 | ALfloat factor, ALfloat speed); |
---|
| 199 | |
---|
| 200 | /* |
---|
| 201 | * _alInitTimeFilters( time_filter_set *tf_ptr_ref ) |
---|
| 202 | * |
---|
| 203 | * Initializes tf_ptr_ref to the current set of time filters, and initialize |
---|
| 204 | * tpitch_lookup_max if it hasn't been initialized before. |
---|
| 205 | */ |
---|
| 206 | void _alInitTimeFilters( time_filter_set *tf_ptr_ref ) { |
---|
| 207 | ALuint i = 0; |
---|
| 208 | |
---|
| 209 | do { |
---|
| 210 | tf_ptr_ref[i] = software_time_filters[i]; |
---|
| 211 | } while(software_time_filters[i++].filter != NULL); |
---|
| 212 | |
---|
| 213 | #if USE_TPITCH_LOOKUP |
---|
| 214 | /* |
---|
| 215 | * init tpitch_loopup only if it hasn't been initialized |
---|
| 216 | * yet. |
---|
| 217 | */ |
---|
| 218 | if(tpitch_lookup.max != TPITCH_MAX) { |
---|
| 219 | tpitch_lookup.max = TPITCH_MAX; |
---|
| 220 | tpitch_lookup.middle = TPITCH_MAX / 2; |
---|
| 221 | |
---|
| 222 | for(i = 0; i < tpitch_lookup.max; i++) |
---|
| 223 | { |
---|
| 224 | free(tpitch_lookup.offsets[i]); |
---|
| 225 | free(tpitch_lookup.fractionals[i]); |
---|
| 226 | tpitch_lookup.offsets[i] = 0; |
---|
| 227 | tpitch_lookup.fractionals[i] = 0; |
---|
| 228 | } |
---|
| 229 | } |
---|
| 230 | #endif |
---|
| 231 | |
---|
| 232 | return; |
---|
| 233 | } |
---|
| 234 | |
---|
| 235 | /* |
---|
| 236 | * _alDestroyFilters( void ) |
---|
| 237 | * |
---|
| 238 | * Deallocates data structures used by the filters and helper functions. |
---|
| 239 | */ |
---|
| 240 | void _alDestroyFilters( void ) { |
---|
| 241 | #if USE_TPITCH_LOOKUP |
---|
| 242 | ALuint i; |
---|
| 243 | |
---|
| 244 | for(i = 0; i < TPITCH_MAX; i++) |
---|
| 245 | { |
---|
| 246 | free(tpitch_lookup.offsets[i]); |
---|
| 247 | free(tpitch_lookup.fractionals[i]); |
---|
| 248 | |
---|
| 249 | tpitch_lookup.offsets[i] = 0; |
---|
| 250 | tpitch_lookup.fractionals[i] = 0; |
---|
| 251 | } |
---|
| 252 | tpitch_lookup.len = 0; |
---|
| 253 | #endif |
---|
| 254 | |
---|
| 255 | return; |
---|
| 256 | } |
---|
| 257 | |
---|
| 258 | /* |
---|
| 259 | * _alApplyFilters( ALuint cid, ALuint sid ) |
---|
| 260 | * |
---|
| 261 | * _alApplyFilters is called from the mixing function, and is passed |
---|
| 262 | * a source id and the context where this sourceid has meaning. |
---|
| 263 | * |
---|
| 264 | * The filters that are applied to the source are determined by the |
---|
| 265 | * context. Each context is initialized such that it contains a table |
---|
| 266 | * of the software filters. Extensions and plugins can be later loaded |
---|
| 267 | * to override the default functionality. The point being, each context's |
---|
| 268 | * filter "signature" may be different. |
---|
| 269 | * |
---|
| 270 | * assumes locked source sid |
---|
| 271 | */ |
---|
| 272 | void _alApplyFilters( ALuint cid, ALuint sid ) { |
---|
| 273 | AL_source *src; |
---|
| 274 | AL_buffer *samp; |
---|
| 275 | time_filter_set *cc_tfilters; |
---|
| 276 | time_filter *tf; |
---|
| 277 | ALuint mixbuflen; /* byte size of total data to compose (all channels) */ |
---|
| 278 | ALint len; /* byte size of one channel's worth of data to compose */ |
---|
| 279 | ALint filterlen; /* filterlen is adjusted below to take into account looping, etc */ |
---|
| 280 | int ic; /* internal (canon) chans */ |
---|
| 281 | int mc; /* mixer chans (==speakers) */ |
---|
| 282 | ALboolean *boolp; /* for determining bool flags */ |
---|
| 283 | int i; |
---|
| 284 | |
---|
| 285 | /* initialize */ |
---|
| 286 | ic = _alGetChannelsFromFormat( _ALC_CANON_FMT ); |
---|
| 287 | |
---|
| 288 | _alcLockContext( cid ); |
---|
| 289 | |
---|
| 290 | mc = _alcGetNumSpeakers( cid ); |
---|
| 291 | mixbuflen = _alcGetWriteBufsiz( cid ); |
---|
| 292 | |
---|
| 293 | samp = _alGetBufferFromSid( cid, sid ); |
---|
| 294 | if(samp == NULL) { |
---|
| 295 | _alDebug(ALD_MAXIMUS, __FILE__, __LINE__, |
---|
| 296 | "_alFilter: null samp, sid == %d", sid); |
---|
| 297 | |
---|
| 298 | _alcUnlockContext( cid ); |
---|
| 299 | return; |
---|
| 300 | } |
---|
| 301 | |
---|
| 302 | _alcUnlockContext( cid ); |
---|
| 303 | |
---|
| 304 | len = mixbuflen * ((float) ic / mc); |
---|
| 305 | filterlen = len; |
---|
| 306 | |
---|
| 307 | /* |
---|
| 308 | * Allocate scratch space to hold enough data for the source |
---|
| 309 | * about to be split. We allocate more space in case of a |
---|
| 310 | * multichannel source. |
---|
| 311 | */ |
---|
| 312 | if(f_buffers.len < len / sizeof (ALshort)) |
---|
| 313 | { |
---|
| 314 | void *temp; |
---|
| 315 | ALuint newlen = len * _alGetChannelsFromFormat(samp->format); |
---|
| 316 | |
---|
| 317 | for(i = 0; i < mc; i++) |
---|
| 318 | { |
---|
| 319 | temp = realloc(f_buffers.data[i], newlen); |
---|
| 320 | if(temp == NULL) { |
---|
| 321 | /* FIXME: do something */ |
---|
| 322 | } |
---|
| 323 | |
---|
| 324 | f_buffers.data[i] = temp; |
---|
| 325 | } |
---|
| 326 | |
---|
| 327 | f_buffers.len = newlen; |
---|
| 328 | } |
---|
| 329 | |
---|
| 330 | #if USE_TPITCH_LOOKUP |
---|
| 331 | if(tpitch_lookup.len < (ALuint) len) |
---|
| 332 | { |
---|
| 333 | init_tpitch_lookup(len); |
---|
| 334 | } |
---|
| 335 | #endif |
---|
| 336 | |
---|
| 337 | src = _alGetSource(cid, sid); |
---|
| 338 | if(src == NULL) { |
---|
| 339 | _alDebug(ALD_MAXIMUS, __FILE__, __LINE__, |
---|
| 340 | "_alFilter: null src, sid == %d", sid); |
---|
| 341 | return; |
---|
| 342 | } |
---|
| 343 | |
---|
| 344 | /* streaming buffer? set soundpos */ |
---|
| 345 | if(samp->flags & ALB_STREAMING) { |
---|
| 346 | src->srcParams.soundpos = samp->streampos; |
---|
| 347 | |
---|
| 348 | if(samp->streampos > samp->size) { |
---|
| 349 | memset(src->srcParams.outbuf, 0, len); |
---|
| 350 | |
---|
| 351 | #ifdef DEBUG_MAXIMUS |
---|
| 352 | fprintf(stderr, "underflow!!!!!!!!!!!!!!!!\n"); |
---|
| 353 | #endif |
---|
| 354 | return; /* underflow */ |
---|
| 355 | } |
---|
| 356 | } |
---|
| 357 | |
---|
| 358 | _alSourceParamReset(src); /* reset srcParam settings */ |
---|
| 359 | |
---|
| 360 | _alSplitSources(cid, sid, mc, len, samp, (ALshort **) f_buffers.data); |
---|
| 361 | |
---|
| 362 | /* |
---|
| 363 | * translate head relative sources |
---|
| 364 | */ |
---|
| 365 | boolp = _alGetSourceParam(src, AL_SOURCE_RELATIVE); |
---|
| 366 | |
---|
| 367 | if(boolp != NULL) { |
---|
| 368 | _alDebug(ALD_SOURCE, __FILE__, __LINE__, |
---|
| 369 | "_alApplyFilters: sid %d relative boolp = %d", sid, *boolp ); |
---|
| 370 | |
---|
| 371 | if(*boolp == AL_TRUE) { |
---|
| 372 | /* This is a RELATIVE source, which means we must |
---|
| 373 | * translate it before applying any sort of positional |
---|
| 374 | * filter to it. |
---|
| 375 | */ |
---|
| 376 | AL_context *cc; |
---|
| 377 | |
---|
| 378 | _alcLockContext( cid ); |
---|
| 379 | |
---|
| 380 | cc = _alcGetContext(cid); |
---|
| 381 | if(cc != NULL) { |
---|
| 382 | _alSourceTranslate(src, cc->listener.position ); |
---|
| 383 | } |
---|
| 384 | |
---|
| 385 | _alcUnlockContext( cid ); |
---|
| 386 | } |
---|
| 387 | } |
---|
| 388 | |
---|
| 389 | /* |
---|
| 390 | * adjust len to account for end of sample, looping, etc |
---|
| 391 | */ |
---|
| 392 | if(filterlen > _alSourceBytesLeft(src, samp)) |
---|
| 393 | { |
---|
| 394 | /* John Quigley's patch, check it out -- jiv */ |
---|
| 395 | if((_alSourceIsLooping( src ) == AL_FALSE) |
---|
| 396 | && (src->srcParams.new_readindex == -1)) |
---|
| 397 | { |
---|
| 398 | /* Non looping source */ |
---|
| 399 | filterlen = _alSourceBytesLeft(src, samp); |
---|
| 400 | } |
---|
| 401 | } |
---|
| 402 | |
---|
| 403 | if(filterlen > 0) |
---|
| 404 | { |
---|
| 405 | cc_tfilters = _alcGetTimeFilters(cid); |
---|
| 406 | |
---|
| 407 | /* apply time domain filters */ |
---|
| 408 | for(i = 0; cc_tfilters[i].filter != NULL; i++) |
---|
| 409 | { |
---|
| 410 | tf = cc_tfilters[i].filter; |
---|
| 411 | |
---|
| 412 | tf(cid, src, samp, (ALshort **) f_buffers.data, |
---|
| 413 | mc, filterlen); |
---|
| 414 | } |
---|
| 415 | |
---|
| 416 | /* |
---|
| 417 | * Apply gain and delay for filters that don't actually touch |
---|
| 418 | * the data ( alf_da). |
---|
| 419 | */ |
---|
| 420 | _alSourceParamApply(src, mc, filterlen, |
---|
| 421 | (ALshort **) f_buffers.data); |
---|
| 422 | } |
---|
| 423 | |
---|
| 424 | |
---|
| 425 | /* |
---|
| 426 | * Take the resulting pcm data in f_buffers, and mix these into |
---|
| 427 | * the source's temporary output buffer. |
---|
| 428 | */ |
---|
| 429 | _alCollapseSource(cid, sid, mc, mixbuflen, (ALshort **) f_buffers.data); |
---|
| 430 | |
---|
| 431 | /* |
---|
| 432 | * head RELATIVE sources need to be untranslated, lest their |
---|
| 433 | * position become weird. |
---|
| 434 | */ |
---|
| 435 | if((boolp != NULL) && (*boolp == AL_TRUE)) { |
---|
| 436 | AL_context *cc; |
---|
| 437 | ALfloat ipos[3]; /* inverse listener position */ |
---|
| 438 | |
---|
| 439 | _alcLockContext( cid ); |
---|
| 440 | |
---|
| 441 | cc = _alcGetContext(cid); |
---|
| 442 | |
---|
| 443 | if(cc != NULL) { |
---|
| 444 | _alVectorInverse(ipos, cc->listener.position); |
---|
| 445 | _alSourceTranslate(src, ipos); |
---|
| 446 | } |
---|
| 447 | |
---|
| 448 | _alcUnlockContext( cid ); |
---|
| 449 | } |
---|
| 450 | |
---|
| 451 | return; |
---|
| 452 | } |
---|
| 453 | |
---|
| 454 | /* |
---|
| 455 | * alf_coning |
---|
| 456 | * |
---|
| 457 | * Implements the coning filter, which is used when CONE_INNER_ANGLE |
---|
| 458 | * or CONE_OUTER_ANGLE is set. This is used for directional sounds. |
---|
| 459 | * |
---|
| 460 | * The spec is vague as to the actual requirements of directional sounds, |
---|
| 461 | * and Carlo has suggested that we maintain the DirectSound meaning for |
---|
| 462 | * directional sounds, namely (in my interpretation): |
---|
| 463 | * |
---|
| 464 | * The inner, outer cone define three zones: inside inner cone |
---|
| 465 | * (INSIDE), between inner and outer cone (BETWEEN, outside outer cone, |
---|
| 466 | * (OUTSIDE). |
---|
| 467 | * |
---|
| 468 | * In INSIDE, the gain of the sound is attenuated as a normal |
---|
| 469 | * positional source. |
---|
| 470 | * |
---|
| 471 | * In OUTSIDE, the gain is set to some value specified by the user. |
---|
| 472 | * |
---|
| 473 | * In BETWEEN, the gain is transitionally set to some value between |
---|
| 474 | * what it would be in INSIDE and OUTSIDE. |
---|
| 475 | * |
---|
| 476 | * This requires an additional source paramter, like CONE_OUTSIDE_ATTENUATION, |
---|
| 477 | * and quite frankly seems goofy. This implementation implements the |
---|
| 478 | * following convention: |
---|
| 479 | * |
---|
| 480 | * In INSIDE, the gain of the sound is attenuated as a normal |
---|
| 481 | * positional source. |
---|
| 482 | * |
---|
| 483 | * In OUTSIDE, the gain is set to _AL_CUTTOFF_ATTENUATION |
---|
| 484 | * |
---|
| 485 | * In BETWEEN, the gain is transitionally set to some value between |
---|
| 486 | * what it would be in INSIDE and OUTSIDE. |
---|
| 487 | * |
---|
| 488 | * Well, okay that's still pretty goofy. Folks who want to set a |
---|
| 489 | * minimum attenuation can stil do so using AL_SOURCE_ATTENUATION_MIN. |
---|
| 490 | * |
---|
| 491 | * IMPLEMENTATION: |
---|
| 492 | * okay, we check the angle between the speaker position and |
---|
| 493 | * the source's direction vector, using the source's position |
---|
| 494 | * as the origin. This angle we call theta. |
---|
| 495 | * |
---|
| 496 | * Then, we compare theta with the outer cone angle. If it's greater, |
---|
| 497 | * we use the min attenuation. If it's less, we compare theta with |
---|
| 498 | * the inner cone angle. If it's greater, we attenuate as normal. |
---|
| 499 | * Otherwise, we don't attenuate at all (full volume, pitch etc). |
---|
| 500 | * |
---|
| 501 | * assumes locked source |
---|
| 502 | * |
---|
| 503 | * FIXME: please check my math. |
---|
| 504 | * - with an AL_NONE distance model, should this do anything |
---|
| 505 | * at all? |
---|
| 506 | */ |
---|
| 507 | void alf_coning( ALuint cid, |
---|
| 508 | AL_source *src, |
---|
| 509 | UNUSED(AL_buffer *samp), |
---|
| 510 | UNUSED(ALshort **buffers), |
---|
| 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 | } |
---|