AudioPort.cpp revision 138f77425a0956d867f078881f91628518ae8258
1/* 2 * Copyright (C) 2015 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17#define LOG_TAG "APM::AudioPort" 18//#define LOG_NDEBUG 0 19#include <media/AudioResamplerPublic.h> 20#include "AudioPort.h" 21#include "HwModule.h" 22#include "AudioGain.h" 23#include "ConfigParsingUtils.h" 24#include "audio_policy_conf.h" 25#include <policy.h> 26 27namespace android { 28 29int32_t volatile AudioPort::mNextUniqueId = 1; 30 31// --- AudioPort class implementation 32 33AudioPort::AudioPort(const String8& name, audio_port_type_t type, 34 audio_port_role_t role) : 35 mName(name), mType(type), mRole(role), mFlags(0) 36{ 37 mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) || 38 ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK)); 39} 40 41void AudioPort::attach(const sp<HwModule>& module) 42{ 43 mModule = module; 44} 45 46audio_port_handle_t AudioPort::getNextUniqueId() 47{ 48 return static_cast<audio_port_handle_t>(android_atomic_inc(&mNextUniqueId)); 49} 50 51audio_module_handle_t AudioPort::getModuleHandle() const 52{ 53 if (mModule == 0) { 54 return 0; 55 } 56 return mModule->mHandle; 57} 58 59uint32_t AudioPort::getModuleVersion() const 60{ 61 if (mModule == 0) { 62 return 0; 63 } 64 return mModule->mHalVersion; 65} 66 67const char *AudioPort::getModuleName() const 68{ 69 if (mModule == 0) { 70 return ""; 71 } 72 return mModule->mName; 73} 74 75void AudioPort::toAudioPort(struct audio_port *port) const 76{ 77 port->role = mRole; 78 port->type = mType; 79 strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN); 80 unsigned int i; 81 for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) { 82 if (mSamplingRates[i] != 0) { 83 port->sample_rates[i] = mSamplingRates[i]; 84 } 85 } 86 port->num_sample_rates = i; 87 for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) { 88 if (mChannelMasks[i] != 0) { 89 port->channel_masks[i] = mChannelMasks[i]; 90 } 91 } 92 port->num_channel_masks = i; 93 for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) { 94 if (mFormats[i] != 0) { 95 port->formats[i] = mFormats[i]; 96 } 97 } 98 port->num_formats = i; 99 100 ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size()); 101 102 for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) { 103 port->gains[i] = mGains[i]->mGain; 104 } 105 port->num_gains = i; 106} 107 108void AudioPort::importAudioPort(const sp<AudioPort> port) { 109 for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) { 110 const uint32_t rate = port->mSamplingRates.itemAt(k); 111 if (rate != 0) { // skip "dynamic" rates 112 bool hasRate = false; 113 for (size_t l = 0 ; l < mSamplingRates.size() ; l++) { 114 if (rate == mSamplingRates.itemAt(l)) { 115 hasRate = true; 116 break; 117 } 118 } 119 if (!hasRate) { // never import a sampling rate twice 120 mSamplingRates.add(rate); 121 } 122 } 123 } 124 for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) { 125 const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k); 126 if (mask != 0) { // skip "dynamic" masks 127 bool hasMask = false; 128 for (size_t l = 0 ; l < mChannelMasks.size() ; l++) { 129 if (mask == mChannelMasks.itemAt(l)) { 130 hasMask = true; 131 break; 132 } 133 } 134 if (!hasMask) { // never import a channel mask twice 135 mChannelMasks.add(mask); 136 } 137 } 138 } 139 for (size_t k = 0 ; k < port->mFormats.size() ; k++) { 140 const audio_format_t format = port->mFormats.itemAt(k); 141 if (format != 0) { // skip "dynamic" formats 142 bool hasFormat = false; 143 for (size_t l = 0 ; l < mFormats.size() ; l++) { 144 if (format == mFormats.itemAt(l)) { 145 hasFormat = true; 146 break; 147 } 148 } 149 if (!hasFormat) { // never import a format twice 150 mFormats.add(format); 151 } 152 } 153 } 154 for (size_t k = 0 ; k < port->mGains.size() ; k++) { 155 sp<AudioGain> gain = port->mGains.itemAt(k); 156 if (gain != 0) { 157 bool hasGain = false; 158 for (size_t l = 0 ; l < mGains.size() ; l++) { 159 if (gain == mGains.itemAt(l)) { 160 hasGain = true; 161 break; 162 } 163 } 164 if (!hasGain) { // never import a gain twice 165 mGains.add(gain); 166 } 167 } 168 } 169} 170 171void AudioPort::clearCapabilities() { 172 mChannelMasks.clear(); 173 mFormats.clear(); 174 mSamplingRates.clear(); 175 mGains.clear(); 176} 177 178void AudioPort::loadSamplingRates(char *name) 179{ 180 char *str = strtok(name, "|"); 181 182 // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling 183 // rates should be read from the output stream after it is opened for the first time 184 if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { 185 mSamplingRates.add(0); 186 return; 187 } 188 189 while (str != NULL) { 190 uint32_t rate = atoi(str); 191 if (rate != 0) { 192 ALOGV("loadSamplingRates() adding rate %d", rate); 193 mSamplingRates.add(rate); 194 } 195 str = strtok(NULL, "|"); 196 } 197} 198 199void AudioPort::loadFormats(char *name) 200{ 201 char *str = strtok(name, "|"); 202 203 // by convention, "0' in the first entry in mFormats indicates the supported formats 204 // should be read from the output stream after it is opened for the first time 205 if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { 206 mFormats.add(AUDIO_FORMAT_DEFAULT); 207 return; 208 } 209 210 while (str != NULL) { 211 audio_format_t format = (audio_format_t)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable, 212 ARRAY_SIZE(sFormatNameToEnumTable), 213 str); 214 if (format != AUDIO_FORMAT_DEFAULT) { 215 mFormats.add(format); 216 } 217 str = strtok(NULL, "|"); 218 } 219 // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry. 220 // TODO: compareFormats could be a lambda to convert between pointer-to-format to format: 221 // [](const audio_format_t *format1, const audio_format_t *format2) { 222 // return compareFormats(*format1, *format2); 223 // } 224 mFormats.sort(compareFormats); 225} 226 227void AudioPort::loadInChannels(char *name) 228{ 229 const char *str = strtok(name, "|"); 230 231 ALOGV("loadInChannels() %s", name); 232 233 if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { 234 mChannelMasks.add(0); 235 return; 236 } 237 238 while (str != NULL) { 239 audio_channel_mask_t channelMask = 240 (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable, 241 ARRAY_SIZE(sInChannelsNameToEnumTable), 242 str); 243 if (channelMask == 0) { // if not found, check the channel index table 244 channelMask = (audio_channel_mask_t) 245 ConfigParsingUtils::stringToEnum(sIndexChannelsNameToEnumTable, 246 ARRAY_SIZE(sIndexChannelsNameToEnumTable), 247 str); 248 } 249 if (channelMask != 0) { 250 ALOGV("loadInChannels() adding channelMask %#x", channelMask); 251 mChannelMasks.add(channelMask); 252 } 253 str = strtok(NULL, "|"); 254 } 255} 256 257void AudioPort::loadOutChannels(char *name) 258{ 259 const char *str = strtok(name, "|"); 260 261 ALOGV("loadOutChannels() %s", name); 262 263 // by convention, "0' in the first entry in mChannelMasks indicates the supported channel 264 // masks should be read from the output stream after it is opened for the first time 265 if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) { 266 mChannelMasks.add(0); 267 return; 268 } 269 270 while (str != NULL) { 271 audio_channel_mask_t channelMask = 272 (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable, 273 ARRAY_SIZE(sOutChannelsNameToEnumTable), 274 str); 275 if (channelMask != 0) { 276 mChannelMasks.add(channelMask); 277 } 278 str = strtok(NULL, "|"); 279 } 280 return; 281} 282 283audio_gain_mode_t AudioPort::loadGainMode(char *name) 284{ 285 const char *str = strtok(name, "|"); 286 287 ALOGV("loadGainMode() %s", name); 288 audio_gain_mode_t mode = 0; 289 while (str != NULL) { 290 mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable, 291 ARRAY_SIZE(sGainModeNameToEnumTable), 292 str); 293 str = strtok(NULL, "|"); 294 } 295 return mode; 296} 297 298void AudioPort::loadGain(cnode *root, int index) 299{ 300 cnode *node = root->first_child; 301 302 sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask); 303 304 while (node) { 305 if (strcmp(node->name, GAIN_MODE) == 0) { 306 gain->mGain.mode = loadGainMode((char *)node->value); 307 } else if (strcmp(node->name, GAIN_CHANNELS) == 0) { 308 if (mUseInChannelMask) { 309 gain->mGain.channel_mask = 310 (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable, 311 ARRAY_SIZE(sInChannelsNameToEnumTable), 312 (char *)node->value); 313 } else { 314 gain->mGain.channel_mask = 315 (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable, 316 ARRAY_SIZE(sOutChannelsNameToEnumTable), 317 (char *)node->value); 318 } 319 } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) { 320 gain->mGain.min_value = atoi((char *)node->value); 321 } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) { 322 gain->mGain.max_value = atoi((char *)node->value); 323 } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) { 324 gain->mGain.default_value = atoi((char *)node->value); 325 } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) { 326 gain->mGain.step_value = atoi((char *)node->value); 327 } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) { 328 gain->mGain.min_ramp_ms = atoi((char *)node->value); 329 } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) { 330 gain->mGain.max_ramp_ms = atoi((char *)node->value); 331 } 332 node = node->next; 333 } 334 335 ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d", 336 gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value); 337 338 if (gain->mGain.mode == 0) { 339 return; 340 } 341 mGains.add(gain); 342} 343 344void AudioPort::loadGains(cnode *root) 345{ 346 cnode *node = root->first_child; 347 int index = 0; 348 while (node) { 349 ALOGV("loadGains() loading gain %s", node->name); 350 loadGain(node, index++); 351 node = node->next; 352 } 353} 354 355status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const 356{ 357 if (mSamplingRates.isEmpty()) { 358 return NO_ERROR; 359 } 360 361 for (size_t i = 0; i < mSamplingRates.size(); i ++) { 362 if (mSamplingRates[i] == samplingRate) { 363 return NO_ERROR; 364 } 365 } 366 return BAD_VALUE; 367} 368 369status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate, 370 uint32_t *updatedSamplingRate) const 371{ 372 if (mSamplingRates.isEmpty()) { 373 if (updatedSamplingRate != NULL) { 374 *updatedSamplingRate = samplingRate; 375 } 376 return NO_ERROR; 377 } 378 379 // Search for the closest supported sampling rate that is above (preferred) 380 // or below (acceptable) the desired sampling rate, within a permitted ratio. 381 // The sampling rates do not need to be sorted in ascending order. 382 ssize_t maxBelow = -1; 383 ssize_t minAbove = -1; 384 uint32_t candidate; 385 for (size_t i = 0; i < mSamplingRates.size(); i++) { 386 candidate = mSamplingRates[i]; 387 if (candidate == samplingRate) { 388 if (updatedSamplingRate != NULL) { 389 *updatedSamplingRate = candidate; 390 } 391 return NO_ERROR; 392 } 393 // candidate < desired 394 if (candidate < samplingRate) { 395 if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) { 396 maxBelow = i; 397 } 398 // candidate > desired 399 } else { 400 if (minAbove < 0 || candidate < mSamplingRates[minAbove]) { 401 minAbove = i; 402 } 403 } 404 } 405 406 // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum. 407 if (minAbove >= 0) { 408 candidate = mSamplingRates[minAbove]; 409 if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) { 410 if (updatedSamplingRate != NULL) { 411 *updatedSamplingRate = candidate; 412 } 413 return NO_ERROR; 414 } 415 } 416 // But if we have to up-sample from a lower sampling rate, that's OK. 417 if (maxBelow >= 0) { 418 candidate = mSamplingRates[maxBelow]; 419 if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) { 420 if (updatedSamplingRate != NULL) { 421 *updatedSamplingRate = candidate; 422 } 423 return NO_ERROR; 424 } 425 } 426 // leave updatedSamplingRate unmodified 427 return BAD_VALUE; 428} 429 430status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const 431{ 432 if (mChannelMasks.isEmpty()) { 433 return NO_ERROR; 434 } 435 436 for (size_t i = 0; i < mChannelMasks.size(); i++) { 437 if (mChannelMasks[i] == channelMask) { 438 return NO_ERROR; 439 } 440 } 441 return BAD_VALUE; 442} 443 444status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask, 445 audio_channel_mask_t *updatedChannelMask) const 446{ 447 if (mChannelMasks.isEmpty()) { 448 if (updatedChannelMask != NULL) { 449 *updatedChannelMask = channelMask; 450 } 451 return NO_ERROR; 452 } 453 454 const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK; 455 const bool isIndex = audio_channel_mask_get_representation(channelMask) 456 == AUDIO_CHANNEL_REPRESENTATION_INDEX; 457 int bestMatch = 0; 458 for (size_t i = 0; i < mChannelMasks.size(); i ++) { 459 audio_channel_mask_t supported = mChannelMasks[i]; 460 if (supported == channelMask) { 461 // Exact matches always taken. 462 if (updatedChannelMask != NULL) { 463 *updatedChannelMask = channelMask; 464 } 465 return NO_ERROR; 466 } 467 468 // AUDIO_CHANNEL_NONE (value: 0) is used for dynamic channel support 469 if (isRecordThread && supported != AUDIO_CHANNEL_NONE) { 470 // Approximate (best) match: 471 // The match score measures how well the supported channel mask matches the 472 // desired mask, where increasing-is-better. 473 // 474 // TODO: Some tweaks may be needed. 475 // Should be a static function of the data processing library. 476 // 477 // In priority: 478 // match score = 1000 if legacy channel conversion equivalent (always prefer this) 479 // OR 480 // match score += 100 if the channel mask representations match 481 // match score += number of channels matched. 482 // 483 // If there are no matched channels, the mask may still be accepted 484 // but the playback or record will be silent. 485 const bool isSupportedIndex = (audio_channel_mask_get_representation(supported) 486 == AUDIO_CHANNEL_REPRESENTATION_INDEX); 487 int match; 488 if (isIndex && isSupportedIndex) { 489 // index equivalence 490 match = 100 + __builtin_popcount( 491 audio_channel_mask_get_bits(channelMask) 492 & audio_channel_mask_get_bits(supported)); 493 } else if (isIndex && !isSupportedIndex) { 494 const uint32_t equivalentBits = 495 (1 << audio_channel_count_from_in_mask(supported)) - 1 ; 496 match = __builtin_popcount( 497 audio_channel_mask_get_bits(channelMask) & equivalentBits); 498 } else if (!isIndex && isSupportedIndex) { 499 const uint32_t equivalentBits = 500 (1 << audio_channel_count_from_in_mask(channelMask)) - 1; 501 match = __builtin_popcount( 502 equivalentBits & audio_channel_mask_get_bits(supported)); 503 } else { 504 // positional equivalence 505 match = 100 + __builtin_popcount( 506 audio_channel_mask_get_bits(channelMask) 507 & audio_channel_mask_get_bits(supported)); 508 switch (supported) { 509 case AUDIO_CHANNEL_IN_FRONT_BACK: 510 case AUDIO_CHANNEL_IN_STEREO: 511 if (channelMask == AUDIO_CHANNEL_IN_MONO) { 512 match = 1000; 513 } 514 break; 515 case AUDIO_CHANNEL_IN_MONO: 516 if (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK 517 || channelMask == AUDIO_CHANNEL_IN_STEREO) { 518 match = 1000; 519 } 520 break; 521 default: 522 break; 523 } 524 } 525 if (match > bestMatch) { 526 bestMatch = match; 527 if (updatedChannelMask != NULL) { 528 *updatedChannelMask = supported; 529 } else { 530 return NO_ERROR; // any match will do in this case. 531 } 532 } 533 } 534 } 535 return bestMatch > 0 ? NO_ERROR : BAD_VALUE; 536} 537 538status_t AudioPort::checkExactFormat(audio_format_t format) const 539{ 540 if (mFormats.isEmpty()) { 541 return NO_ERROR; 542 } 543 544 for (size_t i = 0; i < mFormats.size(); i ++) { 545 if (mFormats[i] == format) { 546 return NO_ERROR; 547 } 548 } 549 return BAD_VALUE; 550} 551 552status_t AudioPort::checkCompatibleFormat(audio_format_t format, audio_format_t *updatedFormat) 553 const 554{ 555 if (mFormats.isEmpty()) { 556 if (updatedFormat != NULL) { 557 *updatedFormat = format; 558 } 559 return NO_ERROR; 560 } 561 562 const bool checkInexact = // when port is input and format is linear pcm 563 mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK 564 && audio_is_linear_pcm(format); 565 566 // iterate from best format to worst format (reverse order) 567 for (ssize_t i = mFormats.size() - 1; i >= 0 ; --i) { 568 if (mFormats[i] == format || 569 (checkInexact 570 && mFormats[i] != AUDIO_FORMAT_DEFAULT 571 && audio_is_linear_pcm(mFormats[i]))) { 572 // for inexact checks we take the first linear pcm format due to sorting. 573 if (updatedFormat != NULL) { 574 *updatedFormat = mFormats[i]; 575 } 576 return NO_ERROR; 577 } 578 } 579 return BAD_VALUE; 580} 581 582uint32_t AudioPort::pickSamplingRate() const 583{ 584 // special case for uninitialized dynamic profile 585 if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) { 586 return 0; 587 } 588 589 // For direct outputs, pick minimum sampling rate: this helps ensuring that the 590 // channel count / sampling rate combination chosen will be supported by the connected 591 // sink 592 if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && 593 (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { 594 uint32_t samplingRate = UINT_MAX; 595 for (size_t i = 0; i < mSamplingRates.size(); i ++) { 596 if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) { 597 samplingRate = mSamplingRates[i]; 598 } 599 } 600 return (samplingRate == UINT_MAX) ? 0 : samplingRate; 601 } 602 603 uint32_t samplingRate = 0; 604 uint32_t maxRate = MAX_MIXER_SAMPLING_RATE; 605 606 // For mixed output and inputs, use max mixer sampling rates. Do not 607 // limit sampling rate otherwise 608 if (mType != AUDIO_PORT_TYPE_MIX) { 609 maxRate = UINT_MAX; 610 } 611 for (size_t i = 0; i < mSamplingRates.size(); i ++) { 612 if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) { 613 samplingRate = mSamplingRates[i]; 614 } 615 } 616 return samplingRate; 617} 618 619audio_channel_mask_t AudioPort::pickChannelMask() const 620{ 621 // special case for uninitialized dynamic profile 622 if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) { 623 return AUDIO_CHANNEL_NONE; 624 } 625 audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE; 626 627 // For direct outputs, pick minimum channel count: this helps ensuring that the 628 // channel count / sampling rate combination chosen will be supported by the connected 629 // sink 630 if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && 631 (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { 632 uint32_t channelCount = UINT_MAX; 633 for (size_t i = 0; i < mChannelMasks.size(); i ++) { 634 uint32_t cnlCount; 635 if (mUseInChannelMask) { 636 cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); 637 } else { 638 cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); 639 } 640 if ((cnlCount < channelCount) && (cnlCount > 0)) { 641 channelMask = mChannelMasks[i]; 642 channelCount = cnlCount; 643 } 644 } 645 return channelMask; 646 } 647 648 uint32_t channelCount = 0; 649 uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT; 650 651 // For mixed output and inputs, use max mixer channel count. Do not 652 // limit channel count otherwise 653 if (mType != AUDIO_PORT_TYPE_MIX) { 654 maxCount = UINT_MAX; 655 } 656 for (size_t i = 0; i < mChannelMasks.size(); i ++) { 657 uint32_t cnlCount; 658 if (mUseInChannelMask) { 659 cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); 660 } else { 661 cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); 662 } 663 if ((cnlCount > channelCount) && (cnlCount <= maxCount)) { 664 channelMask = mChannelMasks[i]; 665 channelCount = cnlCount; 666 } 667 } 668 return channelMask; 669} 670 671/* format in order of increasing preference */ 672const audio_format_t AudioPort::sPcmFormatCompareTable[] = { 673 AUDIO_FORMAT_DEFAULT, 674 AUDIO_FORMAT_PCM_16_BIT, 675 AUDIO_FORMAT_PCM_8_24_BIT, 676 AUDIO_FORMAT_PCM_24_BIT_PACKED, 677 AUDIO_FORMAT_PCM_32_BIT, 678 AUDIO_FORMAT_PCM_FLOAT, 679}; 680 681int AudioPort::compareFormats(audio_format_t format1, 682 audio_format_t format2) 683{ 684 // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any 685 // compressed format and better than any PCM format. This is by design of pickFormat() 686 if (!audio_is_linear_pcm(format1)) { 687 if (!audio_is_linear_pcm(format2)) { 688 return 0; 689 } 690 return 1; 691 } 692 if (!audio_is_linear_pcm(format2)) { 693 return -1; 694 } 695 696 int index1 = -1, index2 = -1; 697 for (size_t i = 0; 698 (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1)); 699 i ++) { 700 if (sPcmFormatCompareTable[i] == format1) { 701 index1 = i; 702 } 703 if (sPcmFormatCompareTable[i] == format2) { 704 index2 = i; 705 } 706 } 707 // format1 not found => index1 < 0 => format2 > format1 708 // format2 not found => index2 < 0 => format2 < format1 709 return index1 - index2; 710} 711 712audio_format_t AudioPort::pickFormat() const 713{ 714 // special case for uninitialized dynamic profile 715 if (mFormats.size() == 1 && mFormats[0] == 0) { 716 return AUDIO_FORMAT_DEFAULT; 717 } 718 719 audio_format_t format = AUDIO_FORMAT_DEFAULT; 720 audio_format_t bestFormat = 721 AudioPort::sPcmFormatCompareTable[ 722 ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1]; 723 // For mixed output and inputs, use best mixer output format. Do not 724 // limit format otherwise 725 if ((mType != AUDIO_PORT_TYPE_MIX) || 726 ((mRole == AUDIO_PORT_ROLE_SOURCE) && 727 (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) { 728 bestFormat = AUDIO_FORMAT_INVALID; 729 } 730 731 for (size_t i = 0; i < mFormats.size(); i ++) { 732 if ((compareFormats(mFormats[i], format) > 0) && 733 (compareFormats(mFormats[i], bestFormat) <= 0)) { 734 format = mFormats[i]; 735 } 736 } 737 return format; 738} 739 740status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig, 741 int index) const 742{ 743 if (index < 0 || (size_t)index >= mGains.size()) { 744 return BAD_VALUE; 745 } 746 return mGains[index]->checkConfig(gainConfig); 747} 748 749void AudioPort::dump(int fd, int spaces) const 750{ 751 const size_t SIZE = 256; 752 char buffer[SIZE]; 753 String8 result; 754 755 if (mName.length() != 0) { 756 snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string()); 757 result.append(buffer); 758 } 759 760 if (mSamplingRates.size() != 0) { 761 snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, ""); 762 result.append(buffer); 763 for (size_t i = 0; i < mSamplingRates.size(); i++) { 764 if (i == 0 && mSamplingRates[i] == 0) { 765 snprintf(buffer, SIZE, "Dynamic"); 766 } else { 767 snprintf(buffer, SIZE, "%d", mSamplingRates[i]); 768 } 769 result.append(buffer); 770 result.append(i == (mSamplingRates.size() - 1) ? "" : ", "); 771 } 772 result.append("\n"); 773 } 774 775 if (mChannelMasks.size() != 0) { 776 snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, ""); 777 result.append(buffer); 778 for (size_t i = 0; i < mChannelMasks.size(); i++) { 779 ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]); 780 781 if (i == 0 && mChannelMasks[i] == 0) { 782 snprintf(buffer, SIZE, "Dynamic"); 783 } else { 784 snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]); 785 } 786 result.append(buffer); 787 result.append(i == (mChannelMasks.size() - 1) ? "" : ", "); 788 } 789 result.append("\n"); 790 } 791 792 if (mFormats.size() != 0) { 793 snprintf(buffer, SIZE, "%*s- formats: ", spaces, ""); 794 result.append(buffer); 795 for (size_t i = 0; i < mFormats.size(); i++) { 796 const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable, 797 ARRAY_SIZE(sFormatNameToEnumTable), 798 mFormats[i]); 799 const bool isEmptyStr = formatStr[0] == 0; 800 if (i == 0 && isEmptyStr) { 801 snprintf(buffer, SIZE, "Dynamic"); 802 } else { 803 if (isEmptyStr) { 804 snprintf(buffer, SIZE, "%#x", mFormats[i]); 805 } else { 806 snprintf(buffer, SIZE, "%s", formatStr); 807 } 808 } 809 result.append(buffer); 810 result.append(i == (mFormats.size() - 1) ? "" : ", "); 811 } 812 result.append("\n"); 813 } 814 write(fd, result.string(), result.size()); 815 if (mGains.size() != 0) { 816 snprintf(buffer, SIZE, "%*s- gains:\n", spaces, ""); 817 write(fd, buffer, strlen(buffer) + 1); 818 for (size_t i = 0; i < mGains.size(); i++) { 819 mGains[i]->dump(fd, spaces + 2, i); 820 } 821 } 822} 823 824void AudioPort::log(const char* indent) const 825{ 826 ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.string(), mType, mRole); 827} 828 829// --- AudioPortConfig class implementation 830 831AudioPortConfig::AudioPortConfig() 832{ 833 mSamplingRate = 0; 834 mChannelMask = AUDIO_CHANNEL_NONE; 835 mFormat = AUDIO_FORMAT_INVALID; 836 mGain.index = -1; 837} 838 839status_t AudioPortConfig::applyAudioPortConfig( 840 const struct audio_port_config *config, 841 struct audio_port_config *backupConfig) 842{ 843 struct audio_port_config localBackupConfig; 844 status_t status = NO_ERROR; 845 846 localBackupConfig.config_mask = config->config_mask; 847 toAudioPortConfig(&localBackupConfig); 848 849 sp<AudioPort> audioport = getAudioPort(); 850 if (audioport == 0) { 851 status = NO_INIT; 852 goto exit; 853 } 854 if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { 855 status = audioport->checkExactSamplingRate(config->sample_rate); 856 if (status != NO_ERROR) { 857 goto exit; 858 } 859 mSamplingRate = config->sample_rate; 860 } 861 if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { 862 status = audioport->checkExactChannelMask(config->channel_mask); 863 if (status != NO_ERROR) { 864 goto exit; 865 } 866 mChannelMask = config->channel_mask; 867 } 868 if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) { 869 status = audioport->checkExactFormat(config->format); 870 if (status != NO_ERROR) { 871 goto exit; 872 } 873 mFormat = config->format; 874 } 875 if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) { 876 status = audioport->checkGain(&config->gain, config->gain.index); 877 if (status != NO_ERROR) { 878 goto exit; 879 } 880 mGain = config->gain; 881 } 882 883exit: 884 if (status != NO_ERROR) { 885 applyAudioPortConfig(&localBackupConfig); 886 } 887 if (backupConfig != NULL) { 888 *backupConfig = localBackupConfig; 889 } 890 return status; 891} 892 893void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig, 894 const struct audio_port_config *srcConfig) const 895{ 896 if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { 897 dstConfig->sample_rate = mSamplingRate; 898 if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) { 899 dstConfig->sample_rate = srcConfig->sample_rate; 900 } 901 } else { 902 dstConfig->sample_rate = 0; 903 } 904 if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { 905 dstConfig->channel_mask = mChannelMask; 906 if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) { 907 dstConfig->channel_mask = srcConfig->channel_mask; 908 } 909 } else { 910 dstConfig->channel_mask = AUDIO_CHANNEL_NONE; 911 } 912 if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) { 913 dstConfig->format = mFormat; 914 if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) { 915 dstConfig->format = srcConfig->format; 916 } 917 } else { 918 dstConfig->format = AUDIO_FORMAT_INVALID; 919 } 920 if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) { 921 dstConfig->gain = mGain; 922 if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) { 923 dstConfig->gain = srcConfig->gain; 924 } 925 } else { 926 dstConfig->gain.index = -1; 927 } 928 if (dstConfig->gain.index != -1) { 929 dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN; 930 } else { 931 dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN; 932 } 933} 934 935}; // namespace android 936