AudioPort.cpp revision 18aa27016a94d0fee243637a80fd0741f89e08f2
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) { // if not found, check the channel index table 276 channelMask = (audio_channel_mask_t) 277 ConfigParsingUtils::stringToEnum(sIndexChannelsNameToEnumTable, 278 ARRAY_SIZE(sIndexChannelsNameToEnumTable), 279 str); 280 } 281 if (channelMask != 0) { 282 mChannelMasks.add(channelMask); 283 } 284 str = strtok(NULL, "|"); 285 } 286 return; 287} 288 289audio_gain_mode_t AudioPort::loadGainMode(char *name) 290{ 291 const char *str = strtok(name, "|"); 292 293 ALOGV("loadGainMode() %s", name); 294 audio_gain_mode_t mode = 0; 295 while (str != NULL) { 296 mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable, 297 ARRAY_SIZE(sGainModeNameToEnumTable), 298 str); 299 str = strtok(NULL, "|"); 300 } 301 return mode; 302} 303 304void AudioPort::loadGain(cnode *root, int index) 305{ 306 cnode *node = root->first_child; 307 308 sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask); 309 310 while (node) { 311 if (strcmp(node->name, GAIN_MODE) == 0) { 312 gain->mGain.mode = loadGainMode((char *)node->value); 313 } else if (strcmp(node->name, GAIN_CHANNELS) == 0) { 314 if (mUseInChannelMask) { 315 gain->mGain.channel_mask = 316 (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable, 317 ARRAY_SIZE(sInChannelsNameToEnumTable), 318 (char *)node->value); 319 } else { 320 gain->mGain.channel_mask = 321 (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable, 322 ARRAY_SIZE(sOutChannelsNameToEnumTable), 323 (char *)node->value); 324 } 325 } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) { 326 gain->mGain.min_value = atoi((char *)node->value); 327 } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) { 328 gain->mGain.max_value = atoi((char *)node->value); 329 } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) { 330 gain->mGain.default_value = atoi((char *)node->value); 331 } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) { 332 gain->mGain.step_value = atoi((char *)node->value); 333 } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) { 334 gain->mGain.min_ramp_ms = atoi((char *)node->value); 335 } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) { 336 gain->mGain.max_ramp_ms = atoi((char *)node->value); 337 } 338 node = node->next; 339 } 340 341 ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d", 342 gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value); 343 344 if (gain->mGain.mode == 0) { 345 return; 346 } 347 mGains.add(gain); 348} 349 350void AudioPort::loadGains(cnode *root) 351{ 352 cnode *node = root->first_child; 353 int index = 0; 354 while (node) { 355 ALOGV("loadGains() loading gain %s", node->name); 356 loadGain(node, index++); 357 node = node->next; 358 } 359} 360 361status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const 362{ 363 if (mSamplingRates.isEmpty()) { 364 return NO_ERROR; 365 } 366 367 for (size_t i = 0; i < mSamplingRates.size(); i ++) { 368 if (mSamplingRates[i] == samplingRate) { 369 return NO_ERROR; 370 } 371 } 372 return BAD_VALUE; 373} 374 375status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate, 376 uint32_t *updatedSamplingRate) const 377{ 378 if (mSamplingRates.isEmpty()) { 379 if (updatedSamplingRate != NULL) { 380 *updatedSamplingRate = samplingRate; 381 } 382 return NO_ERROR; 383 } 384 385 // Search for the closest supported sampling rate that is above (preferred) 386 // or below (acceptable) the desired sampling rate, within a permitted ratio. 387 // The sampling rates do not need to be sorted in ascending order. 388 ssize_t maxBelow = -1; 389 ssize_t minAbove = -1; 390 uint32_t candidate; 391 for (size_t i = 0; i < mSamplingRates.size(); i++) { 392 candidate = mSamplingRates[i]; 393 if (candidate == samplingRate) { 394 if (updatedSamplingRate != NULL) { 395 *updatedSamplingRate = candidate; 396 } 397 return NO_ERROR; 398 } 399 // candidate < desired 400 if (candidate < samplingRate) { 401 if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) { 402 maxBelow = i; 403 } 404 // candidate > desired 405 } else { 406 if (minAbove < 0 || candidate < mSamplingRates[minAbove]) { 407 minAbove = i; 408 } 409 } 410 } 411 412 // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum. 413 if (minAbove >= 0) { 414 candidate = mSamplingRates[minAbove]; 415 if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) { 416 if (updatedSamplingRate != NULL) { 417 *updatedSamplingRate = candidate; 418 } 419 return NO_ERROR; 420 } 421 } 422 // But if we have to up-sample from a lower sampling rate, that's OK. 423 if (maxBelow >= 0) { 424 candidate = mSamplingRates[maxBelow]; 425 if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) { 426 if (updatedSamplingRate != NULL) { 427 *updatedSamplingRate = candidate; 428 } 429 return NO_ERROR; 430 } 431 } 432 // leave updatedSamplingRate unmodified 433 return BAD_VALUE; 434} 435 436status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const 437{ 438 if (mChannelMasks.isEmpty()) { 439 return NO_ERROR; 440 } 441 442 for (size_t i = 0; i < mChannelMasks.size(); i++) { 443 if (mChannelMasks[i] == channelMask) { 444 return NO_ERROR; 445 } 446 } 447 return BAD_VALUE; 448} 449 450status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask, 451 audio_channel_mask_t *updatedChannelMask) const 452{ 453 if (mChannelMasks.isEmpty()) { 454 if (updatedChannelMask != NULL) { 455 *updatedChannelMask = channelMask; 456 } 457 return NO_ERROR; 458 } 459 460 const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK; 461 const bool isIndex = audio_channel_mask_get_representation(channelMask) 462 == AUDIO_CHANNEL_REPRESENTATION_INDEX; 463 int bestMatch = 0; 464 for (size_t i = 0; i < mChannelMasks.size(); i ++) { 465 audio_channel_mask_t supported = mChannelMasks[i]; 466 if (supported == channelMask) { 467 // Exact matches always taken. 468 if (updatedChannelMask != NULL) { 469 *updatedChannelMask = channelMask; 470 } 471 return NO_ERROR; 472 } 473 474 // AUDIO_CHANNEL_NONE (value: 0) is used for dynamic channel support 475 if (isRecordThread && supported != AUDIO_CHANNEL_NONE) { 476 // Approximate (best) match: 477 // The match score measures how well the supported channel mask matches the 478 // desired mask, where increasing-is-better. 479 // 480 // TODO: Some tweaks may be needed. 481 // Should be a static function of the data processing library. 482 // 483 // In priority: 484 // match score = 1000 if legacy channel conversion equivalent (always prefer this) 485 // OR 486 // match score += 100 if the channel mask representations match 487 // match score += number of channels matched. 488 // 489 // If there are no matched channels, the mask may still be accepted 490 // but the playback or record will be silent. 491 const bool isSupportedIndex = (audio_channel_mask_get_representation(supported) 492 == AUDIO_CHANNEL_REPRESENTATION_INDEX); 493 int match; 494 if (isIndex && isSupportedIndex) { 495 // index equivalence 496 match = 100 + __builtin_popcount( 497 audio_channel_mask_get_bits(channelMask) 498 & audio_channel_mask_get_bits(supported)); 499 } else if (isIndex && !isSupportedIndex) { 500 const uint32_t equivalentBits = 501 (1 << audio_channel_count_from_in_mask(supported)) - 1 ; 502 match = __builtin_popcount( 503 audio_channel_mask_get_bits(channelMask) & equivalentBits); 504 } else if (!isIndex && isSupportedIndex) { 505 const uint32_t equivalentBits = 506 (1 << audio_channel_count_from_in_mask(channelMask)) - 1; 507 match = __builtin_popcount( 508 equivalentBits & audio_channel_mask_get_bits(supported)); 509 } else { 510 // positional equivalence 511 match = 100 + __builtin_popcount( 512 audio_channel_mask_get_bits(channelMask) 513 & audio_channel_mask_get_bits(supported)); 514 switch (supported) { 515 case AUDIO_CHANNEL_IN_FRONT_BACK: 516 case AUDIO_CHANNEL_IN_STEREO: 517 if (channelMask == AUDIO_CHANNEL_IN_MONO) { 518 match = 1000; 519 } 520 break; 521 case AUDIO_CHANNEL_IN_MONO: 522 if (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK 523 || channelMask == AUDIO_CHANNEL_IN_STEREO) { 524 match = 1000; 525 } 526 break; 527 default: 528 break; 529 } 530 } 531 if (match > bestMatch) { 532 bestMatch = match; 533 if (updatedChannelMask != NULL) { 534 *updatedChannelMask = supported; 535 } else { 536 return NO_ERROR; // any match will do in this case. 537 } 538 } 539 } 540 } 541 return bestMatch > 0 ? NO_ERROR : BAD_VALUE; 542} 543 544status_t AudioPort::checkExactFormat(audio_format_t format) const 545{ 546 if (mFormats.isEmpty()) { 547 return NO_ERROR; 548 } 549 550 for (size_t i = 0; i < mFormats.size(); i ++) { 551 if (mFormats[i] == format) { 552 return NO_ERROR; 553 } 554 } 555 return BAD_VALUE; 556} 557 558status_t AudioPort::checkCompatibleFormat(audio_format_t format, audio_format_t *updatedFormat) 559 const 560{ 561 if (mFormats.isEmpty()) { 562 if (updatedFormat != NULL) { 563 *updatedFormat = format; 564 } 565 return NO_ERROR; 566 } 567 568 const bool checkInexact = // when port is input and format is linear pcm 569 mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK 570 && audio_is_linear_pcm(format); 571 572 // iterate from best format to worst format (reverse order) 573 for (ssize_t i = mFormats.size() - 1; i >= 0 ; --i) { 574 if (mFormats[i] == format || 575 (checkInexact 576 && mFormats[i] != AUDIO_FORMAT_DEFAULT 577 && audio_is_linear_pcm(mFormats[i]))) { 578 // for inexact checks we take the first linear pcm format due to sorting. 579 if (updatedFormat != NULL) { 580 *updatedFormat = mFormats[i]; 581 } 582 return NO_ERROR; 583 } 584 } 585 return BAD_VALUE; 586} 587 588uint32_t AudioPort::pickSamplingRate() const 589{ 590 // special case for uninitialized dynamic profile 591 if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) { 592 return 0; 593 } 594 595 // For direct outputs, pick minimum sampling rate: this helps ensuring that the 596 // channel count / sampling rate combination chosen will be supported by the connected 597 // sink 598 if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && 599 (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { 600 uint32_t samplingRate = UINT_MAX; 601 for (size_t i = 0; i < mSamplingRates.size(); i ++) { 602 if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) { 603 samplingRate = mSamplingRates[i]; 604 } 605 } 606 return (samplingRate == UINT_MAX) ? 0 : samplingRate; 607 } 608 609 uint32_t samplingRate = 0; 610 uint32_t maxRate = MAX_MIXER_SAMPLING_RATE; 611 612 // For mixed output and inputs, use max mixer sampling rates. Do not 613 // limit sampling rate otherwise 614 if (mType != AUDIO_PORT_TYPE_MIX) { 615 maxRate = UINT_MAX; 616 } 617 for (size_t i = 0; i < mSamplingRates.size(); i ++) { 618 if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) { 619 samplingRate = mSamplingRates[i]; 620 } 621 } 622 return samplingRate; 623} 624 625audio_channel_mask_t AudioPort::pickChannelMask() const 626{ 627 // special case for uninitialized dynamic profile 628 if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) { 629 return AUDIO_CHANNEL_NONE; 630 } 631 audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE; 632 633 // For direct outputs, pick minimum channel count: this helps ensuring that the 634 // channel count / sampling rate combination chosen will be supported by the connected 635 // sink 636 if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) && 637 (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) { 638 uint32_t channelCount = UINT_MAX; 639 for (size_t i = 0; i < mChannelMasks.size(); i ++) { 640 uint32_t cnlCount; 641 if (mUseInChannelMask) { 642 cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); 643 } else { 644 cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); 645 } 646 if ((cnlCount < channelCount) && (cnlCount > 0)) { 647 channelMask = mChannelMasks[i]; 648 channelCount = cnlCount; 649 } 650 } 651 return channelMask; 652 } 653 654 uint32_t channelCount = 0; 655 uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT; 656 657 // For mixed output and inputs, use max mixer channel count. Do not 658 // limit channel count otherwise 659 if (mType != AUDIO_PORT_TYPE_MIX) { 660 maxCount = UINT_MAX; 661 } 662 for (size_t i = 0; i < mChannelMasks.size(); i ++) { 663 uint32_t cnlCount; 664 if (mUseInChannelMask) { 665 cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]); 666 } else { 667 cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]); 668 } 669 if ((cnlCount > channelCount) && (cnlCount <= maxCount)) { 670 channelMask = mChannelMasks[i]; 671 channelCount = cnlCount; 672 } 673 } 674 return channelMask; 675} 676 677/* format in order of increasing preference */ 678const audio_format_t AudioPort::sPcmFormatCompareTable[] = { 679 AUDIO_FORMAT_DEFAULT, 680 AUDIO_FORMAT_PCM_16_BIT, 681 AUDIO_FORMAT_PCM_8_24_BIT, 682 AUDIO_FORMAT_PCM_24_BIT_PACKED, 683 AUDIO_FORMAT_PCM_32_BIT, 684 AUDIO_FORMAT_PCM_FLOAT, 685}; 686 687int AudioPort::compareFormats(audio_format_t format1, 688 audio_format_t format2) 689{ 690 // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any 691 // compressed format and better than any PCM format. This is by design of pickFormat() 692 if (!audio_is_linear_pcm(format1)) { 693 if (!audio_is_linear_pcm(format2)) { 694 return 0; 695 } 696 return 1; 697 } 698 if (!audio_is_linear_pcm(format2)) { 699 return -1; 700 } 701 702 int index1 = -1, index2 = -1; 703 for (size_t i = 0; 704 (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1)); 705 i ++) { 706 if (sPcmFormatCompareTable[i] == format1) { 707 index1 = i; 708 } 709 if (sPcmFormatCompareTable[i] == format2) { 710 index2 = i; 711 } 712 } 713 // format1 not found => index1 < 0 => format2 > format1 714 // format2 not found => index2 < 0 => format2 < format1 715 return index1 - index2; 716} 717 718audio_format_t AudioPort::pickFormat() const 719{ 720 // special case for uninitialized dynamic profile 721 if (mFormats.size() == 1 && mFormats[0] == 0) { 722 return AUDIO_FORMAT_DEFAULT; 723 } 724 725 audio_format_t format = AUDIO_FORMAT_DEFAULT; 726 audio_format_t bestFormat = 727 AudioPort::sPcmFormatCompareTable[ 728 ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1]; 729 // For mixed output and inputs, use best mixer output format. Do not 730 // limit format otherwise 731 if ((mType != AUDIO_PORT_TYPE_MIX) || 732 ((mRole == AUDIO_PORT_ROLE_SOURCE) && 733 (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) { 734 bestFormat = AUDIO_FORMAT_INVALID; 735 } 736 737 for (size_t i = 0; i < mFormats.size(); i ++) { 738 if ((compareFormats(mFormats[i], format) > 0) && 739 (compareFormats(mFormats[i], bestFormat) <= 0)) { 740 format = mFormats[i]; 741 } 742 } 743 return format; 744} 745 746status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig, 747 int index) const 748{ 749 if (index < 0 || (size_t)index >= mGains.size()) { 750 return BAD_VALUE; 751 } 752 return mGains[index]->checkConfig(gainConfig); 753} 754 755void AudioPort::dump(int fd, int spaces) const 756{ 757 const size_t SIZE = 256; 758 char buffer[SIZE]; 759 String8 result; 760 761 if (mName.length() != 0) { 762 snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string()); 763 result.append(buffer); 764 } 765 766 if (mSamplingRates.size() != 0) { 767 snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, ""); 768 result.append(buffer); 769 for (size_t i = 0; i < mSamplingRates.size(); i++) { 770 if (i == 0 && mSamplingRates[i] == 0) { 771 snprintf(buffer, SIZE, "Dynamic"); 772 } else { 773 snprintf(buffer, SIZE, "%d", mSamplingRates[i]); 774 } 775 result.append(buffer); 776 result.append(i == (mSamplingRates.size() - 1) ? "" : ", "); 777 } 778 result.append("\n"); 779 } 780 781 if (mChannelMasks.size() != 0) { 782 snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, ""); 783 result.append(buffer); 784 for (size_t i = 0; i < mChannelMasks.size(); i++) { 785 ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]); 786 787 if (i == 0 && mChannelMasks[i] == 0) { 788 snprintf(buffer, SIZE, "Dynamic"); 789 } else { 790 snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]); 791 } 792 result.append(buffer); 793 result.append(i == (mChannelMasks.size() - 1) ? "" : ", "); 794 } 795 result.append("\n"); 796 } 797 798 if (mFormats.size() != 0) { 799 snprintf(buffer, SIZE, "%*s- formats: ", spaces, ""); 800 result.append(buffer); 801 for (size_t i = 0; i < mFormats.size(); i++) { 802 const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable, 803 ARRAY_SIZE(sFormatNameToEnumTable), 804 mFormats[i]); 805 const bool isEmptyStr = formatStr[0] == 0; 806 if (i == 0 && isEmptyStr) { 807 snprintf(buffer, SIZE, "Dynamic"); 808 } else { 809 if (isEmptyStr) { 810 snprintf(buffer, SIZE, "%#x", mFormats[i]); 811 } else { 812 snprintf(buffer, SIZE, "%s", formatStr); 813 } 814 } 815 result.append(buffer); 816 result.append(i == (mFormats.size() - 1) ? "" : ", "); 817 } 818 result.append("\n"); 819 } 820 write(fd, result.string(), result.size()); 821 if (mGains.size() != 0) { 822 snprintf(buffer, SIZE, "%*s- gains:\n", spaces, ""); 823 write(fd, buffer, strlen(buffer) + 1); 824 for (size_t i = 0; i < mGains.size(); i++) { 825 mGains[i]->dump(fd, spaces + 2, i); 826 } 827 } 828} 829 830void AudioPort::log(const char* indent) const 831{ 832 ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.string(), mType, mRole); 833} 834 835// --- AudioPortConfig class implementation 836 837AudioPortConfig::AudioPortConfig() 838{ 839 mSamplingRate = 0; 840 mChannelMask = AUDIO_CHANNEL_NONE; 841 mFormat = AUDIO_FORMAT_INVALID; 842 mGain.index = -1; 843} 844 845status_t AudioPortConfig::applyAudioPortConfig( 846 const struct audio_port_config *config, 847 struct audio_port_config *backupConfig) 848{ 849 struct audio_port_config localBackupConfig; 850 status_t status = NO_ERROR; 851 852 localBackupConfig.config_mask = config->config_mask; 853 toAudioPortConfig(&localBackupConfig); 854 855 sp<AudioPort> audioport = getAudioPort(); 856 if (audioport == 0) { 857 status = NO_INIT; 858 goto exit; 859 } 860 if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { 861 status = audioport->checkExactSamplingRate(config->sample_rate); 862 if (status != NO_ERROR) { 863 goto exit; 864 } 865 mSamplingRate = config->sample_rate; 866 } 867 if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { 868 status = audioport->checkExactChannelMask(config->channel_mask); 869 if (status != NO_ERROR) { 870 goto exit; 871 } 872 mChannelMask = config->channel_mask; 873 } 874 if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) { 875 status = audioport->checkExactFormat(config->format); 876 if (status != NO_ERROR) { 877 goto exit; 878 } 879 mFormat = config->format; 880 } 881 if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) { 882 status = audioport->checkGain(&config->gain, config->gain.index); 883 if (status != NO_ERROR) { 884 goto exit; 885 } 886 mGain = config->gain; 887 } 888 889exit: 890 if (status != NO_ERROR) { 891 applyAudioPortConfig(&localBackupConfig); 892 } 893 if (backupConfig != NULL) { 894 *backupConfig = localBackupConfig; 895 } 896 return status; 897} 898 899void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig, 900 const struct audio_port_config *srcConfig) const 901{ 902 if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) { 903 dstConfig->sample_rate = mSamplingRate; 904 if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) { 905 dstConfig->sample_rate = srcConfig->sample_rate; 906 } 907 } else { 908 dstConfig->sample_rate = 0; 909 } 910 if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) { 911 dstConfig->channel_mask = mChannelMask; 912 if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) { 913 dstConfig->channel_mask = srcConfig->channel_mask; 914 } 915 } else { 916 dstConfig->channel_mask = AUDIO_CHANNEL_NONE; 917 } 918 if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) { 919 dstConfig->format = mFormat; 920 if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) { 921 dstConfig->format = srcConfig->format; 922 } 923 } else { 924 dstConfig->format = AUDIO_FORMAT_INVALID; 925 } 926 if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) { 927 dstConfig->gain = mGain; 928 if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) { 929 dstConfig->gain = srcConfig->gain; 930 } 931 } else { 932 dstConfig->gain.index = -1; 933 } 934 if (dstConfig->gain.index != -1) { 935 dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN; 936 } else { 937 dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN; 938 } 939} 940 941}; // namespace android 942