1 /* 2 ** 3 ** Copyright 2007, The Android Open Source Project 4 ** 5 ** Licensed under the Apache License, Version 2.0 (the "License"); 6 ** you may not use this file except in compliance with the License. 7 ** You may obtain a copy of the License at 8 ** 9 ** http://www.apache.org/licenses/LICENSE-2.0 10 ** 11 ** Unless required by applicable law or agreed to in writing, software 12 ** distributed under the License is distributed on an "AS IS" BASIS, 13 ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14 ** See the License for the specific language governing permissions and 15 ** limitations under the License. 16 */ 17 18 19 #define LOG_TAG "AudioFlinger" 20 //#define LOG_NDEBUG 0 21 22 #include "Configuration.h" 23 #include <dirent.h> 24 #include <math.h> 25 #include <signal.h> 26 #include <sys/time.h> 27 #include <sys/resource.h> 28 29 #include <binder/IPCThreadState.h> 30 #include <binder/IServiceManager.h> 31 #include <utils/Log.h> 32 #include <utils/Trace.h> 33 #include <binder/Parcel.h> 34 #include <media/audiohal/DeviceHalInterface.h> 35 #include <media/audiohal/DevicesFactoryHalInterface.h> 36 #include <media/audiohal/EffectsFactoryHalInterface.h> 37 #include <media/AudioParameter.h> 38 #include <media/TypeConverter.h> 39 #include <memunreachable/memunreachable.h> 40 #include <utils/String16.h> 41 #include <utils/threads.h> 42 #include <utils/Atomic.h> 43 44 #include <cutils/properties.h> 45 46 #include <system/audio.h> 47 48 #include "AudioFlinger.h" 49 #include "ServiceUtilities.h" 50 51 #include <media/AudioResamplerPublic.h> 52 53 #include <system/audio_effects/effect_visualizer.h> 54 #include <system/audio_effects/effect_ns.h> 55 #include <system/audio_effects/effect_aec.h> 56 57 #include <audio_utils/primitives.h> 58 59 #include <powermanager/PowerManager.h> 60 61 #include <media/IMediaLogService.h> 62 #include <media/MemoryLeakTrackUtil.h> 63 #include <media/nbaio/Pipe.h> 64 #include <media/nbaio/PipeReader.h> 65 #include <media/AudioParameter.h> 66 #include <mediautils/BatteryNotifier.h> 67 #include <private/android_filesystem_config.h> 68 69 //#define BUFLOG_NDEBUG 0 70 #include <BufLog.h> 71 72 #include "TypedLogger.h" 73 74 // ---------------------------------------------------------------------------- 75 76 // Note: the following macro is used for extremely verbose logging message. In 77 // order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78 // 0; but one side effect of this is to turn all LOGV's as well. Some messages 79 // are so verbose that we want to suppress them even when we have ALOG_ASSERT 80 // turned on. Do not uncomment the #def below unless you really know what you 81 // are doing and want to see all of the extremely verbose messages. 82 //#define VERY_VERY_VERBOSE_LOGGING 83 #ifdef VERY_VERY_VERBOSE_LOGGING 84 #define ALOGVV ALOGV 85 #else 86 #define ALOGVV(a...) do { } while(0) 87 #endif 88 89 namespace android { 90 91 static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 92 static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 93 static const char kClientLockedString[] = "Client lock is taken\n"; 94 static const char kNoEffectsFactory[] = "Effects Factory is absent\n"; 95 96 97 nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 98 99 uint32_t AudioFlinger::mScreenState; 100 101 102 #ifdef TEE_SINK 103 bool AudioFlinger::mTeeSinkInputEnabled = false; 104 bool AudioFlinger::mTeeSinkOutputEnabled = false; 105 bool AudioFlinger::mTeeSinkTrackEnabled = false; 106 107 size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 108 size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 109 size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 110 #endif 111 112 // In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 113 // we define a minimum time during which a global effect is considered enabled. 114 static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 115 116 Mutex gLock; 117 wp<AudioFlinger> gAudioFlinger; 118 119 // Keep a strong reference to media.log service around forever. 120 // The service is within our parent process so it can never die in a way that we could observe. 121 // These two variables are const after initialization. 122 static sp<IBinder> sMediaLogServiceAsBinder; 123 static sp<IMediaLogService> sMediaLogService; 124 125 static pthread_once_t sMediaLogOnce = PTHREAD_ONCE_INIT; 126 127 static void sMediaLogInit() 128 { 129 sMediaLogServiceAsBinder = defaultServiceManager()->getService(String16("media.log")); 130 if (sMediaLogServiceAsBinder != 0) { 131 sMediaLogService = interface_cast<IMediaLogService>(sMediaLogServiceAsBinder); 132 } 133 } 134 135 // ---------------------------------------------------------------------------- 136 137 std::string formatToString(audio_format_t format) { 138 std::string result; 139 FormatConverter::toString(format, result); 140 return result; 141 } 142 143 // ---------------------------------------------------------------------------- 144 145 AudioFlinger::AudioFlinger() 146 : BnAudioFlinger(), 147 mMediaLogNotifier(new AudioFlinger::MediaLogNotifier()), 148 mPrimaryHardwareDev(NULL), 149 mAudioHwDevs(NULL), 150 mHardwareStatus(AUDIO_HW_IDLE), 151 mMasterVolume(1.0f), 152 mMasterMute(false), 153 // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), 154 mMode(AUDIO_MODE_INVALID), 155 mBtNrecIsOff(false), 156 mIsLowRamDevice(true), 157 mIsDeviceTypeKnown(false), 158 mGlobalEffectEnableTime(0), 159 mSystemReady(false) 160 { 161 // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum 162 for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) { 163 // zero ID has a special meaning, so unavailable 164 mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX; 165 } 166 167 getpid_cached = getpid(); 168 const bool doLog = property_get_bool("ro.test_harness", false); 169 if (doLog) { 170 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 171 MemoryHeapBase::READ_ONLY); 172 (void) pthread_once(&sMediaLogOnce, sMediaLogInit); 173 } 174 175 // reset battery stats. 176 // if the audio service has crashed, battery stats could be left 177 // in bad state, reset the state upon service start. 178 BatteryNotifier::getInstance().noteResetAudio(); 179 180 mDevicesFactoryHal = DevicesFactoryHalInterface::create(); 181 mEffectsFactoryHal = EffectsFactoryHalInterface::create(); 182 183 mMediaLogNotifier->run("MediaLogNotifier"); 184 185 #ifdef TEE_SINK 186 char value[PROPERTY_VALUE_MAX]; 187 (void) property_get("ro.debuggable", value, "0"); 188 int debuggable = atoi(value); 189 int teeEnabled = 0; 190 if (debuggable) { 191 (void) property_get("af.tee", value, "0"); 192 teeEnabled = atoi(value); 193 } 194 // FIXME symbolic constants here 195 if (teeEnabled & 1) { 196 mTeeSinkInputEnabled = true; 197 } 198 if (teeEnabled & 2) { 199 mTeeSinkOutputEnabled = true; 200 } 201 if (teeEnabled & 4) { 202 mTeeSinkTrackEnabled = true; 203 } 204 #endif 205 } 206 207 void AudioFlinger::onFirstRef() 208 { 209 Mutex::Autolock _l(mLock); 210 211 /* TODO: move all this work into an Init() function */ 212 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 213 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 214 uint32_t int_val; 215 if (1 == sscanf(val_str, "%u", &int_val)) { 216 mStandbyTimeInNsecs = milliseconds(int_val); 217 ALOGI("Using %u mSec as standby time.", int_val); 218 } else { 219 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 220 ALOGI("Using default %u mSec as standby time.", 221 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 222 } 223 } 224 225 mPatchPanel = new PatchPanel(this); 226 227 mMode = AUDIO_MODE_NORMAL; 228 229 gAudioFlinger = this; 230 } 231 232 AudioFlinger::~AudioFlinger() 233 { 234 while (!mRecordThreads.isEmpty()) { 235 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 236 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 237 } 238 while (!mPlaybackThreads.isEmpty()) { 239 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 240 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 241 } 242 243 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 244 // no mHardwareLock needed, as there are no other references to this 245 delete mAudioHwDevs.valueAt(i); 246 } 247 248 // Tell media.log service about any old writers that still need to be unregistered 249 if (sMediaLogService != 0) { 250 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 251 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 252 mUnregisteredWriters.pop(); 253 sMediaLogService->unregisterWriter(iMemory); 254 } 255 } 256 } 257 258 //static 259 __attribute__ ((visibility ("default"))) 260 status_t MmapStreamInterface::openMmapStream(MmapStreamInterface::stream_direction_t direction, 261 const audio_attributes_t *attr, 262 audio_config_base_t *config, 263 const AudioClient& client, 264 audio_port_handle_t *deviceId, 265 const sp<MmapStreamCallback>& callback, 266 sp<MmapStreamInterface>& interface, 267 audio_port_handle_t *handle) 268 { 269 sp<AudioFlinger> af; 270 { 271 Mutex::Autolock _l(gLock); 272 af = gAudioFlinger.promote(); 273 } 274 status_t ret = NO_INIT; 275 if (af != 0) { 276 ret = af->openMmapStream( 277 direction, attr, config, client, deviceId, callback, interface, handle); 278 } 279 return ret; 280 } 281 282 status_t AudioFlinger::openMmapStream(MmapStreamInterface::stream_direction_t direction, 283 const audio_attributes_t *attr, 284 audio_config_base_t *config, 285 const AudioClient& client, 286 audio_port_handle_t *deviceId, 287 const sp<MmapStreamCallback>& callback, 288 sp<MmapStreamInterface>& interface, 289 audio_port_handle_t *handle) 290 { 291 status_t ret = initCheck(); 292 if (ret != NO_ERROR) { 293 return ret; 294 } 295 296 audio_session_t sessionId = (audio_session_t) newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 297 audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT; 298 audio_io_handle_t io = AUDIO_IO_HANDLE_NONE; 299 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE; 300 if (direction == MmapStreamInterface::DIRECTION_OUTPUT) { 301 audio_config_t fullConfig = AUDIO_CONFIG_INITIALIZER; 302 fullConfig.sample_rate = config->sample_rate; 303 fullConfig.channel_mask = config->channel_mask; 304 fullConfig.format = config->format; 305 ret = AudioSystem::getOutputForAttr(attr, &io, 306 sessionId, 307 &streamType, client.clientUid, 308 &fullConfig, 309 (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | 310 AUDIO_OUTPUT_FLAG_DIRECT), 311 deviceId, &portId); 312 } else { 313 ret = AudioSystem::getInputForAttr(attr, &io, 314 sessionId, 315 client.clientPid, 316 client.clientUid, 317 config, 318 AUDIO_INPUT_FLAG_MMAP_NOIRQ, deviceId, &portId); 319 } 320 if (ret != NO_ERROR) { 321 return ret; 322 } 323 324 // at this stage, a MmapThread was created when openOutput() or openInput() was called by 325 // audio policy manager and we can retrieve it 326 sp<MmapThread> thread = mMmapThreads.valueFor(io); 327 if (thread != 0) { 328 interface = new MmapThreadHandle(thread); 329 thread->configure(attr, streamType, sessionId, callback, *deviceId, portId); 330 *handle = portId; 331 } else { 332 if (direction == MmapStreamInterface::DIRECTION_OUTPUT) { 333 AudioSystem::releaseOutput(io, streamType, sessionId); 334 } else { 335 AudioSystem::releaseInput(io, sessionId); 336 } 337 ret = NO_INIT; 338 } 339 340 ALOGV("%s done status %d portId %d", __FUNCTION__, ret, portId); 341 342 return ret; 343 } 344 345 static const char * const audio_interfaces[] = { 346 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 347 AUDIO_HARDWARE_MODULE_ID_A2DP, 348 AUDIO_HARDWARE_MODULE_ID_USB, 349 }; 350 351 AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 352 audio_module_handle_t module, 353 audio_devices_t devices) 354 { 355 // if module is 0, the request comes from an old policy manager and we should load 356 // well known modules 357 if (module == 0) { 358 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 359 for (size_t i = 0; i < arraysize(audio_interfaces); i++) { 360 loadHwModule_l(audio_interfaces[i]); 361 } 362 // then try to find a module supporting the requested device. 363 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 364 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 365 sp<DeviceHalInterface> dev = audioHwDevice->hwDevice(); 366 uint32_t supportedDevices; 367 if (dev->getSupportedDevices(&supportedDevices) == OK && 368 (supportedDevices & devices) == devices) { 369 return audioHwDevice; 370 } 371 } 372 } else { 373 // check a match for the requested module handle 374 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 375 if (audioHwDevice != NULL) { 376 return audioHwDevice; 377 } 378 } 379 380 return NULL; 381 } 382 383 void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 384 { 385 const size_t SIZE = 256; 386 char buffer[SIZE]; 387 String8 result; 388 389 result.append("Clients:\n"); 390 for (size_t i = 0; i < mClients.size(); ++i) { 391 sp<Client> client = mClients.valueAt(i).promote(); 392 if (client != 0) { 393 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 394 result.append(buffer); 395 } 396 } 397 398 result.append("Notification Clients:\n"); 399 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 400 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 401 result.append(buffer); 402 } 403 404 result.append("Global session refs:\n"); 405 result.append(" session pid count\n"); 406 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 407 AudioSessionRef *r = mAudioSessionRefs[i]; 408 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 409 result.append(buffer); 410 } 411 write(fd, result.string(), result.size()); 412 } 413 414 415 void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 416 { 417 const size_t SIZE = 256; 418 char buffer[SIZE]; 419 String8 result; 420 hardware_call_state hardwareStatus = mHardwareStatus; 421 422 snprintf(buffer, SIZE, "Hardware status: %d\n" 423 "Standby Time mSec: %u\n", 424 hardwareStatus, 425 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 426 result.append(buffer); 427 write(fd, result.string(), result.size()); 428 } 429 430 void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 431 { 432 const size_t SIZE = 256; 433 char buffer[SIZE]; 434 String8 result; 435 snprintf(buffer, SIZE, "Permission Denial: " 436 "can't dump AudioFlinger from pid=%d, uid=%d\n", 437 IPCThreadState::self()->getCallingPid(), 438 IPCThreadState::self()->getCallingUid()); 439 result.append(buffer); 440 write(fd, result.string(), result.size()); 441 } 442 443 bool AudioFlinger::dumpTryLock(Mutex& mutex) 444 { 445 bool locked = false; 446 for (int i = 0; i < kDumpLockRetries; ++i) { 447 if (mutex.tryLock() == NO_ERROR) { 448 locked = true; 449 break; 450 } 451 usleep(kDumpLockSleepUs); 452 } 453 return locked; 454 } 455 456 status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 457 { 458 if (!dumpAllowed()) { 459 dumpPermissionDenial(fd, args); 460 } else { 461 // get state of hardware lock 462 bool hardwareLocked = dumpTryLock(mHardwareLock); 463 if (!hardwareLocked) { 464 String8 result(kHardwareLockedString); 465 write(fd, result.string(), result.size()); 466 } else { 467 mHardwareLock.unlock(); 468 } 469 470 bool locked = dumpTryLock(mLock); 471 472 // failed to lock - AudioFlinger is probably deadlocked 473 if (!locked) { 474 String8 result(kDeadlockedString); 475 write(fd, result.string(), result.size()); 476 } 477 478 bool clientLocked = dumpTryLock(mClientLock); 479 if (!clientLocked) { 480 String8 result(kClientLockedString); 481 write(fd, result.string(), result.size()); 482 } 483 484 if (mEffectsFactoryHal != 0) { 485 mEffectsFactoryHal->dumpEffects(fd); 486 } else { 487 String8 result(kNoEffectsFactory); 488 write(fd, result.string(), result.size()); 489 } 490 491 dumpClients(fd, args); 492 if (clientLocked) { 493 mClientLock.unlock(); 494 } 495 496 dumpInternals(fd, args); 497 498 // dump playback threads 499 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 500 mPlaybackThreads.valueAt(i)->dump(fd, args); 501 } 502 503 // dump record threads 504 for (size_t i = 0; i < mRecordThreads.size(); i++) { 505 mRecordThreads.valueAt(i)->dump(fd, args); 506 } 507 508 // dump mmap threads 509 for (size_t i = 0; i < mMmapThreads.size(); i++) { 510 mMmapThreads.valueAt(i)->dump(fd, args); 511 } 512 513 // dump orphan effect chains 514 if (mOrphanEffectChains.size() != 0) { 515 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 516 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 517 mOrphanEffectChains.valueAt(i)->dump(fd, args); 518 } 519 } 520 // dump all hardware devs 521 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 522 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 523 dev->dump(fd); 524 } 525 526 #ifdef TEE_SINK 527 // dump the serially shared record tee sink 528 if (mRecordTeeSource != 0) { 529 dumpTee(fd, mRecordTeeSource, AUDIO_IO_HANDLE_NONE, 'C'); 530 } 531 #endif 532 533 BUFLOG_RESET; 534 535 if (locked) { 536 mLock.unlock(); 537 } 538 539 // append a copy of media.log here by forwarding fd to it, but don't attempt 540 // to lookup the service if it's not running, as it will block for a second 541 if (sMediaLogServiceAsBinder != 0) { 542 dprintf(fd, "\nmedia.log:\n"); 543 Vector<String16> args; 544 sMediaLogServiceAsBinder->dump(fd, args); 545 } 546 547 // check for optional arguments 548 bool dumpMem = false; 549 bool unreachableMemory = false; 550 for (const auto &arg : args) { 551 if (arg == String16("-m")) { 552 dumpMem = true; 553 } else if (arg == String16("--unreachable")) { 554 unreachableMemory = true; 555 } 556 } 557 558 if (dumpMem) { 559 dprintf(fd, "\nDumping memory:\n"); 560 std::string s = dumpMemoryAddresses(100 /* limit */); 561 write(fd, s.c_str(), s.size()); 562 } 563 if (unreachableMemory) { 564 dprintf(fd, "\nDumping unreachable memory:\n"); 565 // TODO - should limit be an argument parameter? 566 std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */); 567 write(fd, s.c_str(), s.size()); 568 } 569 } 570 return NO_ERROR; 571 } 572 573 sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 574 { 575 Mutex::Autolock _cl(mClientLock); 576 // If pid is already in the mClients wp<> map, then use that entry 577 // (for which promote() is always != 0), otherwise create a new entry and Client. 578 sp<Client> client = mClients.valueFor(pid).promote(); 579 if (client == 0) { 580 client = new Client(this, pid); 581 mClients.add(pid, client); 582 } 583 584 return client; 585 } 586 587 sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 588 { 589 // If there is no memory allocated for logs, return a dummy writer that does nothing. 590 // Similarly if we can't contact the media.log service, also return a dummy writer. 591 if (mLogMemoryDealer == 0 || sMediaLogService == 0) { 592 return new NBLog::Writer(); 593 } 594 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 595 // If allocation fails, consult the vector of previously unregistered writers 596 // and garbage-collect one or more them until an allocation succeeds 597 if (shared == 0) { 598 Mutex::Autolock _l(mUnregisteredWritersLock); 599 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 600 { 601 // Pick the oldest stale writer to garbage-collect 602 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 603 mUnregisteredWriters.removeAt(0); 604 sMediaLogService->unregisterWriter(iMemory); 605 // Now the media.log remote reference to IMemory is gone. When our last local 606 // reference to IMemory also drops to zero at end of this block, 607 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 608 } 609 // Re-attempt the allocation 610 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 611 if (shared != 0) { 612 goto success; 613 } 614 } 615 // Even after garbage-collecting all old writers, there is still not enough memory, 616 // so return a dummy writer 617 return new NBLog::Writer(); 618 } 619 success: 620 NBLog::Shared *sharedRawPtr = (NBLog::Shared *) shared->pointer(); 621 new((void *) sharedRawPtr) NBLog::Shared(); // placement new here, but the corresponding 622 // explicit destructor not needed since it is POD 623 sMediaLogService->registerWriter(shared, size, name); 624 return new NBLog::Writer(shared, size); 625 } 626 627 void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 628 { 629 if (writer == 0) { 630 return; 631 } 632 sp<IMemory> iMemory(writer->getIMemory()); 633 if (iMemory == 0) { 634 return; 635 } 636 // Rather than removing the writer immediately, append it to a queue of old writers to 637 // be garbage-collected later. This allows us to continue to view old logs for a while. 638 Mutex::Autolock _l(mUnregisteredWritersLock); 639 mUnregisteredWriters.push(writer); 640 } 641 642 // IAudioFlinger interface 643 644 645 sp<IAudioTrack> AudioFlinger::createTrack( 646 audio_stream_type_t streamType, 647 uint32_t sampleRate, 648 audio_format_t format, 649 audio_channel_mask_t channelMask, 650 size_t *frameCount, 651 audio_output_flags_t *flags, 652 const sp<IMemory>& sharedBuffer, 653 audio_io_handle_t output, 654 pid_t pid, 655 pid_t tid, 656 audio_session_t *sessionId, 657 int clientUid, 658 status_t *status, 659 audio_port_handle_t portId) 660 { 661 sp<PlaybackThread::Track> track; 662 sp<TrackHandle> trackHandle; 663 sp<Client> client; 664 status_t lStatus; 665 audio_session_t lSessionId; 666 667 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 668 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 669 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 670 ALOGW_IF(pid != -1 && pid != callingPid, 671 "%s uid %d pid %d tried to pass itself off as pid %d", 672 __func__, callingUid, callingPid, pid); 673 pid = callingPid; 674 } 675 676 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 677 // but if someone uses binder directly they could bypass that and cause us to crash 678 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 679 ALOGE("createTrack() invalid stream type %d", streamType); 680 lStatus = BAD_VALUE; 681 goto Exit; 682 } 683 684 // further sample rate checks are performed by createTrack_l() depending on the thread type 685 if (sampleRate == 0) { 686 ALOGE("createTrack() invalid sample rate %u", sampleRate); 687 lStatus = BAD_VALUE; 688 goto Exit; 689 } 690 691 // further channel mask checks are performed by createTrack_l() depending on the thread type 692 if (!audio_is_output_channel(channelMask)) { 693 ALOGE("createTrack() invalid channel mask %#x", channelMask); 694 lStatus = BAD_VALUE; 695 goto Exit; 696 } 697 698 // further format checks are performed by createTrack_l() depending on the thread type 699 if (!audio_is_valid_format(format)) { 700 ALOGE("createTrack() invalid format %#x", format); 701 lStatus = BAD_VALUE; 702 goto Exit; 703 } 704 705 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 706 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 707 lStatus = BAD_VALUE; 708 goto Exit; 709 } 710 711 { 712 Mutex::Autolock _l(mLock); 713 PlaybackThread *thread = checkPlaybackThread_l(output); 714 if (thread == NULL) { 715 ALOGE("no playback thread found for output handle %d", output); 716 lStatus = BAD_VALUE; 717 goto Exit; 718 } 719 720 client = registerPid(pid); 721 722 PlaybackThread *effectThread = NULL; 723 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 724 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 725 ALOGE("createTrack() invalid session ID %d", *sessionId); 726 lStatus = BAD_VALUE; 727 goto Exit; 728 } 729 lSessionId = *sessionId; 730 // check if an effect chain with the same session ID is present on another 731 // output thread and move it here. 732 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 733 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 734 if (mPlaybackThreads.keyAt(i) != output) { 735 uint32_t sessions = t->hasAudioSession(lSessionId); 736 if (sessions & ThreadBase::EFFECT_SESSION) { 737 effectThread = t.get(); 738 break; 739 } 740 } 741 } 742 } else { 743 // if no audio session id is provided, create one here 744 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 745 if (sessionId != NULL) { 746 *sessionId = lSessionId; 747 } 748 } 749 ALOGV("createTrack() lSessionId: %d", lSessionId); 750 751 track = thread->createTrack_l(client, streamType, sampleRate, format, 752 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, 753 clientUid, &lStatus, portId); 754 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 755 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 756 757 // move effect chain to this output thread if an effect on same session was waiting 758 // for a track to be created 759 if (lStatus == NO_ERROR && effectThread != NULL) { 760 // no risk of deadlock because AudioFlinger::mLock is held 761 Mutex::Autolock _dl(thread->mLock); 762 Mutex::Autolock _sl(effectThread->mLock); 763 moveEffectChain_l(lSessionId, effectThread, thread, true); 764 } 765 766 // Look for sync events awaiting for a session to be used. 767 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 768 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 769 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 770 if (lStatus == NO_ERROR) { 771 (void) track->setSyncEvent(mPendingSyncEvents[i]); 772 } else { 773 mPendingSyncEvents[i]->cancel(); 774 } 775 mPendingSyncEvents.removeAt(i); 776 i--; 777 } 778 } 779 } 780 781 setAudioHwSyncForSession_l(thread, lSessionId); 782 } 783 784 if (lStatus != NO_ERROR) { 785 // remove local strong reference to Client before deleting the Track so that the 786 // Client destructor is called by the TrackBase destructor with mClientLock held 787 // Don't hold mClientLock when releasing the reference on the track as the 788 // destructor will acquire it. 789 { 790 Mutex::Autolock _cl(mClientLock); 791 client.clear(); 792 } 793 track.clear(); 794 goto Exit; 795 } 796 797 // return handle to client 798 trackHandle = new TrackHandle(track); 799 800 Exit: 801 *status = lStatus; 802 return trackHandle; 803 } 804 805 uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 806 { 807 Mutex::Autolock _l(mLock); 808 ThreadBase *thread = checkThread_l(ioHandle); 809 if (thread == NULL) { 810 ALOGW("sampleRate() unknown thread %d", ioHandle); 811 return 0; 812 } 813 return thread->sampleRate(); 814 } 815 816 audio_format_t AudioFlinger::format(audio_io_handle_t output) const 817 { 818 Mutex::Autolock _l(mLock); 819 PlaybackThread *thread = checkPlaybackThread_l(output); 820 if (thread == NULL) { 821 ALOGW("format() unknown thread %d", output); 822 return AUDIO_FORMAT_INVALID; 823 } 824 return thread->format(); 825 } 826 827 size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 828 { 829 Mutex::Autolock _l(mLock); 830 ThreadBase *thread = checkThread_l(ioHandle); 831 if (thread == NULL) { 832 ALOGW("frameCount() unknown thread %d", ioHandle); 833 return 0; 834 } 835 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 836 // should examine all callers and fix them to handle smaller counts 837 return thread->frameCount(); 838 } 839 840 size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const 841 { 842 Mutex::Autolock _l(mLock); 843 ThreadBase *thread = checkThread_l(ioHandle); 844 if (thread == NULL) { 845 ALOGW("frameCountHAL() unknown thread %d", ioHandle); 846 return 0; 847 } 848 return thread->frameCountHAL(); 849 } 850 851 uint32_t AudioFlinger::latency(audio_io_handle_t output) const 852 { 853 Mutex::Autolock _l(mLock); 854 PlaybackThread *thread = checkPlaybackThread_l(output); 855 if (thread == NULL) { 856 ALOGW("latency(): no playback thread found for output handle %d", output); 857 return 0; 858 } 859 return thread->latency(); 860 } 861 862 status_t AudioFlinger::setMasterVolume(float value) 863 { 864 status_t ret = initCheck(); 865 if (ret != NO_ERROR) { 866 return ret; 867 } 868 869 // check calling permissions 870 if (!settingsAllowed()) { 871 return PERMISSION_DENIED; 872 } 873 874 Mutex::Autolock _l(mLock); 875 mMasterVolume = value; 876 877 // Set master volume in the HALs which support it. 878 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 879 AutoMutex lock(mHardwareLock); 880 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 881 882 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 883 if (dev->canSetMasterVolume()) { 884 dev->hwDevice()->setMasterVolume(value); 885 } 886 mHardwareStatus = AUDIO_HW_IDLE; 887 } 888 889 // Now set the master volume in each playback thread. Playback threads 890 // assigned to HALs which do not have master volume support will apply 891 // master volume during the mix operation. Threads with HALs which do 892 // support master volume will simply ignore the setting. 893 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 894 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 895 continue; 896 } 897 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 898 } 899 900 return NO_ERROR; 901 } 902 903 status_t AudioFlinger::setMode(audio_mode_t mode) 904 { 905 status_t ret = initCheck(); 906 if (ret != NO_ERROR) { 907 return ret; 908 } 909 910 // check calling permissions 911 if (!settingsAllowed()) { 912 return PERMISSION_DENIED; 913 } 914 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 915 ALOGW("Illegal value: setMode(%d)", mode); 916 return BAD_VALUE; 917 } 918 919 { // scope for the lock 920 AutoMutex lock(mHardwareLock); 921 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 922 mHardwareStatus = AUDIO_HW_SET_MODE; 923 ret = dev->setMode(mode); 924 mHardwareStatus = AUDIO_HW_IDLE; 925 } 926 927 if (NO_ERROR == ret) { 928 Mutex::Autolock _l(mLock); 929 mMode = mode; 930 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 931 mPlaybackThreads.valueAt(i)->setMode(mode); 932 } 933 934 return ret; 935 } 936 937 status_t AudioFlinger::setMicMute(bool state) 938 { 939 status_t ret = initCheck(); 940 if (ret != NO_ERROR) { 941 return ret; 942 } 943 944 // check calling permissions 945 if (!settingsAllowed()) { 946 return PERMISSION_DENIED; 947 } 948 949 AutoMutex lock(mHardwareLock); 950 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 951 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 952 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 953 status_t result = dev->setMicMute(state); 954 if (result != NO_ERROR) { 955 ret = result; 956 } 957 } 958 mHardwareStatus = AUDIO_HW_IDLE; 959 return ret; 960 } 961 962 bool AudioFlinger::getMicMute() const 963 { 964 status_t ret = initCheck(); 965 if (ret != NO_ERROR) { 966 return false; 967 } 968 bool mute = true; 969 bool state = AUDIO_MODE_INVALID; 970 AutoMutex lock(mHardwareLock); 971 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 972 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 973 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 974 status_t result = dev->getMicMute(&state); 975 if (result == NO_ERROR) { 976 mute = mute && state; 977 } 978 } 979 mHardwareStatus = AUDIO_HW_IDLE; 980 981 return mute; 982 } 983 984 status_t AudioFlinger::setMasterMute(bool muted) 985 { 986 status_t ret = initCheck(); 987 if (ret != NO_ERROR) { 988 return ret; 989 } 990 991 // check calling permissions 992 if (!settingsAllowed()) { 993 return PERMISSION_DENIED; 994 } 995 996 Mutex::Autolock _l(mLock); 997 mMasterMute = muted; 998 999 // Set master mute in the HALs which support it. 1000 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1001 AutoMutex lock(mHardwareLock); 1002 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 1003 1004 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1005 if (dev->canSetMasterMute()) { 1006 dev->hwDevice()->setMasterMute(muted); 1007 } 1008 mHardwareStatus = AUDIO_HW_IDLE; 1009 } 1010 1011 // Now set the master mute in each playback thread. Playback threads 1012 // assigned to HALs which do not have master mute support will apply master 1013 // mute during the mix operation. Threads with HALs which do support master 1014 // mute will simply ignore the setting. 1015 Vector<VolumeInterface *> volumeInterfaces = getAllVolumeInterfaces_l(); 1016 for (size_t i = 0; i < volumeInterfaces.size(); i++) { 1017 volumeInterfaces[i]->setMasterMute(muted); 1018 } 1019 1020 return NO_ERROR; 1021 } 1022 1023 float AudioFlinger::masterVolume() const 1024 { 1025 Mutex::Autolock _l(mLock); 1026 return masterVolume_l(); 1027 } 1028 1029 bool AudioFlinger::masterMute() const 1030 { 1031 Mutex::Autolock _l(mLock); 1032 return masterMute_l(); 1033 } 1034 1035 float AudioFlinger::masterVolume_l() const 1036 { 1037 return mMasterVolume; 1038 } 1039 1040 bool AudioFlinger::masterMute_l() const 1041 { 1042 return mMasterMute; 1043 } 1044 1045 status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 1046 { 1047 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 1048 ALOGW("checkStreamType() invalid stream %d", stream); 1049 return BAD_VALUE; 1050 } 1051 pid_t caller = IPCThreadState::self()->getCallingPid(); 1052 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 1053 ALOGW("checkStreamType() pid %d cannot use internal stream type %d", caller, stream); 1054 return PERMISSION_DENIED; 1055 } 1056 1057 return NO_ERROR; 1058 } 1059 1060 status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 1061 audio_io_handle_t output) 1062 { 1063 // check calling permissions 1064 if (!settingsAllowed()) { 1065 return PERMISSION_DENIED; 1066 } 1067 1068 status_t status = checkStreamType(stream); 1069 if (status != NO_ERROR) { 1070 return status; 1071 } 1072 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 1073 1074 AutoMutex lock(mLock); 1075 Vector<VolumeInterface *> volumeInterfaces; 1076 if (output != AUDIO_IO_HANDLE_NONE) { 1077 VolumeInterface *volumeInterface = getVolumeInterface_l(output); 1078 if (volumeInterface == NULL) { 1079 return BAD_VALUE; 1080 } 1081 volumeInterfaces.add(volumeInterface); 1082 } 1083 1084 mStreamTypes[stream].volume = value; 1085 1086 if (volumeInterfaces.size() == 0) { 1087 volumeInterfaces = getAllVolumeInterfaces_l(); 1088 } 1089 for (size_t i = 0; i < volumeInterfaces.size(); i++) { 1090 volumeInterfaces[i]->setStreamVolume(stream, value); 1091 } 1092 1093 return NO_ERROR; 1094 } 1095 1096 status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 1097 { 1098 // check calling permissions 1099 if (!settingsAllowed()) { 1100 return PERMISSION_DENIED; 1101 } 1102 1103 status_t status = checkStreamType(stream); 1104 if (status != NO_ERROR) { 1105 return status; 1106 } 1107 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1108 1109 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1110 ALOGE("setStreamMute() invalid stream %d", stream); 1111 return BAD_VALUE; 1112 } 1113 1114 AutoMutex lock(mLock); 1115 mStreamTypes[stream].mute = muted; 1116 Vector<VolumeInterface *> volumeInterfaces = getAllVolumeInterfaces_l(); 1117 for (size_t i = 0; i < volumeInterfaces.size(); i++) { 1118 volumeInterfaces[i]->setStreamMute(stream, muted); 1119 } 1120 1121 return NO_ERROR; 1122 } 1123 1124 float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1125 { 1126 status_t status = checkStreamType(stream); 1127 if (status != NO_ERROR) { 1128 return 0.0f; 1129 } 1130 1131 AutoMutex lock(mLock); 1132 float volume; 1133 if (output != AUDIO_IO_HANDLE_NONE) { 1134 VolumeInterface *volumeInterface = getVolumeInterface_l(output); 1135 if (volumeInterface != NULL) { 1136 volume = volumeInterface->streamVolume(stream); 1137 } else { 1138 volume = 0.0f; 1139 } 1140 } else { 1141 volume = streamVolume_l(stream); 1142 } 1143 1144 return volume; 1145 } 1146 1147 bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1148 { 1149 status_t status = checkStreamType(stream); 1150 if (status != NO_ERROR) { 1151 return true; 1152 } 1153 1154 AutoMutex lock(mLock); 1155 return streamMute_l(stream); 1156 } 1157 1158 1159 void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1160 { 1161 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1162 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1163 } 1164 } 1165 1166 status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1167 { 1168 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1169 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1170 1171 // check calling permissions 1172 if (!settingsAllowed()) { 1173 return PERMISSION_DENIED; 1174 } 1175 1176 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1177 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1178 Mutex::Autolock _l(mLock); 1179 // result will remain NO_INIT if no audio device is present 1180 status_t final_result = NO_INIT; 1181 { 1182 AutoMutex lock(mHardwareLock); 1183 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1184 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1185 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1186 status_t result = dev->setParameters(keyValuePairs); 1187 // return success if at least one audio device accepts the parameters as not all 1188 // HALs are requested to support all parameters. If no audio device supports the 1189 // requested parameters, the last error is reported. 1190 if (final_result != NO_ERROR) { 1191 final_result = result; 1192 } 1193 } 1194 mHardwareStatus = AUDIO_HW_IDLE; 1195 } 1196 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1197 AudioParameter param = AudioParameter(keyValuePairs); 1198 String8 value; 1199 if (param.get(String8(AudioParameter::keyBtNrec), value) == NO_ERROR) { 1200 bool btNrecIsOff = (value == AudioParameter::valueOff); 1201 if (mBtNrecIsOff.exchange(btNrecIsOff) != btNrecIsOff) { 1202 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1203 mRecordThreads.valueAt(i)->checkBtNrec(); 1204 } 1205 } 1206 } 1207 String8 screenState; 1208 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1209 bool isOff = (screenState == AudioParameter::valueOff); 1210 if (isOff != (AudioFlinger::mScreenState & 1)) { 1211 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1212 } 1213 } 1214 return final_result; 1215 } 1216 1217 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1218 // and the thread is exited once the lock is released 1219 sp<ThreadBase> thread; 1220 { 1221 Mutex::Autolock _l(mLock); 1222 thread = checkPlaybackThread_l(ioHandle); 1223 if (thread == 0) { 1224 thread = checkRecordThread_l(ioHandle); 1225 if (thread == 0) { 1226 thread = checkMmapThread_l(ioHandle); 1227 } 1228 } else if (thread == primaryPlaybackThread_l()) { 1229 // indicate output device change to all input threads for pre processing 1230 AudioParameter param = AudioParameter(keyValuePairs); 1231 int value; 1232 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1233 (value != 0)) { 1234 broacastParametersToRecordThreads_l(keyValuePairs); 1235 } 1236 } 1237 } 1238 if (thread != 0) { 1239 return thread->setParameters(keyValuePairs); 1240 } 1241 return BAD_VALUE; 1242 } 1243 1244 String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1245 { 1246 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1247 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1248 1249 Mutex::Autolock _l(mLock); 1250 1251 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1252 String8 out_s8; 1253 1254 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1255 String8 s; 1256 status_t result; 1257 { 1258 AutoMutex lock(mHardwareLock); 1259 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1260 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1261 result = dev->getParameters(keys, &s); 1262 mHardwareStatus = AUDIO_HW_IDLE; 1263 } 1264 if (result == OK) out_s8 += s; 1265 } 1266 return out_s8; 1267 } 1268 1269 ThreadBase *thread = (ThreadBase *)checkPlaybackThread_l(ioHandle); 1270 if (thread == NULL) { 1271 thread = (ThreadBase *)checkRecordThread_l(ioHandle); 1272 if (thread == NULL) { 1273 thread = (ThreadBase *)checkMmapThread_l(ioHandle); 1274 if (thread == NULL) { 1275 return String8(""); 1276 } 1277 } 1278 } 1279 return thread->getParameters(keys); 1280 } 1281 1282 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1283 audio_channel_mask_t channelMask) const 1284 { 1285 status_t ret = initCheck(); 1286 if (ret != NO_ERROR) { 1287 return 0; 1288 } 1289 if ((sampleRate == 0) || 1290 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1291 !audio_is_input_channel(channelMask)) { 1292 return 0; 1293 } 1294 1295 AutoMutex lock(mHardwareLock); 1296 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1297 audio_config_t config, proposed; 1298 memset(&proposed, 0, sizeof(proposed)); 1299 proposed.sample_rate = sampleRate; 1300 proposed.channel_mask = channelMask; 1301 proposed.format = format; 1302 1303 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1304 size_t frames; 1305 for (;;) { 1306 // Note: config is currently a const parameter for get_input_buffer_size() 1307 // but we use a copy from proposed in case config changes from the call. 1308 config = proposed; 1309 status_t result = dev->getInputBufferSize(&config, &frames); 1310 if (result == OK && frames != 0) { 1311 break; // hal success, config is the result 1312 } 1313 // change one parameter of the configuration each iteration to a more "common" value 1314 // to see if the device will support it. 1315 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1316 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1317 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1318 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1319 } else { 1320 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1321 "format %#x, channelMask 0x%X", 1322 sampleRate, format, channelMask); 1323 break; // retries failed, break out of loop with frames == 0. 1324 } 1325 } 1326 mHardwareStatus = AUDIO_HW_IDLE; 1327 if (frames > 0 && config.sample_rate != sampleRate) { 1328 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1329 } 1330 return frames; // may be converted to bytes at the Java level. 1331 } 1332 1333 uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1334 { 1335 Mutex::Autolock _l(mLock); 1336 1337 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1338 if (recordThread != NULL) { 1339 return recordThread->getInputFramesLost(); 1340 } 1341 return 0; 1342 } 1343 1344 status_t AudioFlinger::setVoiceVolume(float value) 1345 { 1346 status_t ret = initCheck(); 1347 if (ret != NO_ERROR) { 1348 return ret; 1349 } 1350 1351 // check calling permissions 1352 if (!settingsAllowed()) { 1353 return PERMISSION_DENIED; 1354 } 1355 1356 AutoMutex lock(mHardwareLock); 1357 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1358 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1359 ret = dev->setVoiceVolume(value); 1360 mHardwareStatus = AUDIO_HW_IDLE; 1361 1362 return ret; 1363 } 1364 1365 status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1366 audio_io_handle_t output) const 1367 { 1368 Mutex::Autolock _l(mLock); 1369 1370 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1371 if (playbackThread != NULL) { 1372 return playbackThread->getRenderPosition(halFrames, dspFrames); 1373 } 1374 1375 return BAD_VALUE; 1376 } 1377 1378 void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1379 { 1380 Mutex::Autolock _l(mLock); 1381 if (client == 0) { 1382 return; 1383 } 1384 pid_t pid = IPCThreadState::self()->getCallingPid(); 1385 { 1386 Mutex::Autolock _cl(mClientLock); 1387 if (mNotificationClients.indexOfKey(pid) < 0) { 1388 sp<NotificationClient> notificationClient = new NotificationClient(this, 1389 client, 1390 pid); 1391 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1392 1393 mNotificationClients.add(pid, notificationClient); 1394 1395 sp<IBinder> binder = IInterface::asBinder(client); 1396 binder->linkToDeath(notificationClient); 1397 } 1398 } 1399 1400 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1401 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1402 // the config change is always sent from playback or record threads to avoid deadlock 1403 // with AudioSystem::gLock 1404 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1405 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_REGISTERED, pid); 1406 } 1407 1408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1409 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_REGISTERED, pid); 1410 } 1411 } 1412 1413 void AudioFlinger::removeNotificationClient(pid_t pid) 1414 { 1415 Mutex::Autolock _l(mLock); 1416 { 1417 Mutex::Autolock _cl(mClientLock); 1418 mNotificationClients.removeItem(pid); 1419 } 1420 1421 ALOGV("%d died, releasing its sessions", pid); 1422 size_t num = mAudioSessionRefs.size(); 1423 bool removed = false; 1424 for (size_t i = 0; i < num; ) { 1425 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1426 ALOGV(" pid %d @ %zu", ref->mPid, i); 1427 if (ref->mPid == pid) { 1428 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1429 mAudioSessionRefs.removeAt(i); 1430 delete ref; 1431 removed = true; 1432 num--; 1433 } else { 1434 i++; 1435 } 1436 } 1437 if (removed) { 1438 purgeStaleEffects_l(); 1439 } 1440 } 1441 1442 void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1443 const sp<AudioIoDescriptor>& ioDesc, 1444 pid_t pid) 1445 { 1446 Mutex::Autolock _l(mClientLock); 1447 size_t size = mNotificationClients.size(); 1448 for (size_t i = 0; i < size; i++) { 1449 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1450 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1451 } 1452 } 1453 } 1454 1455 // removeClient_l() must be called with AudioFlinger::mClientLock held 1456 void AudioFlinger::removeClient_l(pid_t pid) 1457 { 1458 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1459 IPCThreadState::self()->getCallingPid()); 1460 mClients.removeItem(pid); 1461 } 1462 1463 // getEffectThread_l() must be called with AudioFlinger::mLock held 1464 sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1465 int EffectId) 1466 { 1467 sp<PlaybackThread> thread; 1468 1469 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1470 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1471 ALOG_ASSERT(thread == 0); 1472 thread = mPlaybackThreads.valueAt(i); 1473 } 1474 } 1475 1476 return thread; 1477 } 1478 1479 1480 1481 // ---------------------------------------------------------------------------- 1482 1483 AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1484 : RefBase(), 1485 mAudioFlinger(audioFlinger), 1486 mPid(pid) 1487 { 1488 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1489 heapSize *= 1024; 1490 if (!heapSize) { 1491 heapSize = kClientSharedHeapSizeBytes; 1492 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1493 // invalidated tracks 1494 if (!audioFlinger->isLowRamDevice()) { 1495 heapSize *= kClientSharedHeapSizeMultiplier; 1496 } 1497 } 1498 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1499 } 1500 1501 // Client destructor must be called with AudioFlinger::mClientLock held 1502 AudioFlinger::Client::~Client() 1503 { 1504 mAudioFlinger->removeClient_l(mPid); 1505 } 1506 1507 sp<MemoryDealer> AudioFlinger::Client::heap() const 1508 { 1509 return mMemoryDealer; 1510 } 1511 1512 // ---------------------------------------------------------------------------- 1513 1514 AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1515 const sp<IAudioFlingerClient>& client, 1516 pid_t pid) 1517 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1518 { 1519 } 1520 1521 AudioFlinger::NotificationClient::~NotificationClient() 1522 { 1523 } 1524 1525 void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1526 { 1527 sp<NotificationClient> keep(this); 1528 mAudioFlinger->removeNotificationClient(mPid); 1529 } 1530 1531 // ---------------------------------------------------------------------------- 1532 AudioFlinger::MediaLogNotifier::MediaLogNotifier() 1533 : mPendingRequests(false) {} 1534 1535 1536 void AudioFlinger::MediaLogNotifier::requestMerge() { 1537 AutoMutex _l(mMutex); 1538 mPendingRequests = true; 1539 mCond.signal(); 1540 } 1541 1542 bool AudioFlinger::MediaLogNotifier::threadLoop() { 1543 // Should already have been checked, but just in case 1544 if (sMediaLogService == 0) { 1545 return false; 1546 } 1547 // Wait until there are pending requests 1548 { 1549 AutoMutex _l(mMutex); 1550 mPendingRequests = false; // to ignore past requests 1551 while (!mPendingRequests) { 1552 mCond.wait(mMutex); 1553 // TODO may also need an exitPending check 1554 } 1555 mPendingRequests = false; 1556 } 1557 // Execute the actual MediaLogService binder call and ignore extra requests for a while 1558 sMediaLogService->requestMergeWakeup(); 1559 usleep(kPostTriggerSleepPeriod); 1560 return true; 1561 } 1562 1563 void AudioFlinger::requestLogMerge() { 1564 mMediaLogNotifier->requestMerge(); 1565 } 1566 1567 // ---------------------------------------------------------------------------- 1568 1569 sp<IAudioRecord> AudioFlinger::openRecord( 1570 audio_io_handle_t input, 1571 uint32_t sampleRate, 1572 audio_format_t format, 1573 audio_channel_mask_t channelMask, 1574 const String16& opPackageName, 1575 size_t *frameCount, 1576 audio_input_flags_t *flags, 1577 pid_t pid, 1578 pid_t tid, 1579 int clientUid, 1580 audio_session_t *sessionId, 1581 size_t *notificationFrames, 1582 sp<IMemory>& cblk, 1583 sp<IMemory>& buffers, 1584 status_t *status, 1585 audio_port_handle_t portId) 1586 { 1587 sp<RecordThread::RecordTrack> recordTrack; 1588 sp<RecordHandle> recordHandle; 1589 sp<Client> client; 1590 status_t lStatus; 1591 audio_session_t lSessionId; 1592 1593 cblk.clear(); 1594 buffers.clear(); 1595 1596 bool updatePid = (pid == -1); 1597 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1598 if (!isTrustedCallingUid(callingUid)) { 1599 ALOGW_IF((uid_t)clientUid != callingUid, 1600 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1601 clientUid = callingUid; 1602 updatePid = true; 1603 } 1604 1605 if (updatePid) { 1606 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1607 ALOGW_IF(pid != -1 && pid != callingPid, 1608 "%s uid %d pid %d tried to pass itself off as pid %d", 1609 __func__, callingUid, callingPid, pid); 1610 pid = callingPid; 1611 } 1612 1613 // check calling permissions 1614 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1615 ALOGE("openRecord() permission denied: recording not allowed"); 1616 lStatus = PERMISSION_DENIED; 1617 goto Exit; 1618 } 1619 1620 // further sample rate checks are performed by createRecordTrack_l() 1621 if (sampleRate == 0) { 1622 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1623 lStatus = BAD_VALUE; 1624 goto Exit; 1625 } 1626 1627 // we don't yet support anything other than linear PCM 1628 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1629 ALOGE("openRecord() invalid format %#x", format); 1630 lStatus = BAD_VALUE; 1631 goto Exit; 1632 } 1633 1634 // further channel mask checks are performed by createRecordTrack_l() 1635 if (!audio_is_input_channel(channelMask)) { 1636 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1637 lStatus = BAD_VALUE; 1638 goto Exit; 1639 } 1640 1641 { 1642 Mutex::Autolock _l(mLock); 1643 RecordThread *thread = checkRecordThread_l(input); 1644 if (thread == NULL) { 1645 ALOGE("openRecord() checkRecordThread_l failed"); 1646 lStatus = BAD_VALUE; 1647 goto Exit; 1648 } 1649 1650 client = registerPid(pid); 1651 1652 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1653 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1654 lStatus = BAD_VALUE; 1655 goto Exit; 1656 } 1657 lSessionId = *sessionId; 1658 } else { 1659 // if no audio session id is provided, create one here 1660 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1661 if (sessionId != NULL) { 1662 *sessionId = lSessionId; 1663 } 1664 } 1665 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1666 1667 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1668 frameCount, lSessionId, notificationFrames, 1669 clientUid, flags, tid, &lStatus, portId); 1670 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1671 1672 if (lStatus == NO_ERROR) { 1673 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1674 // session and move it to this thread. 1675 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1676 if (chain != 0) { 1677 Mutex::Autolock _l(thread->mLock); 1678 thread->addEffectChain_l(chain); 1679 } 1680 } 1681 } 1682 1683 if (lStatus != NO_ERROR) { 1684 // remove local strong reference to Client before deleting the RecordTrack so that the 1685 // Client destructor is called by the TrackBase destructor with mClientLock held 1686 // Don't hold mClientLock when releasing the reference on the track as the 1687 // destructor will acquire it. 1688 { 1689 Mutex::Autolock _cl(mClientLock); 1690 client.clear(); 1691 } 1692 recordTrack.clear(); 1693 goto Exit; 1694 } 1695 1696 cblk = recordTrack->getCblk(); 1697 buffers = recordTrack->getBuffers(); 1698 1699 // return handle to client 1700 recordHandle = new RecordHandle(recordTrack); 1701 1702 Exit: 1703 *status = lStatus; 1704 return recordHandle; 1705 } 1706 1707 1708 1709 // ---------------------------------------------------------------------------- 1710 1711 audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1712 { 1713 if (name == NULL) { 1714 return AUDIO_MODULE_HANDLE_NONE; 1715 } 1716 if (!settingsAllowed()) { 1717 return AUDIO_MODULE_HANDLE_NONE; 1718 } 1719 Mutex::Autolock _l(mLock); 1720 return loadHwModule_l(name); 1721 } 1722 1723 // loadHwModule_l() must be called with AudioFlinger::mLock held 1724 audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1725 { 1726 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1727 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1728 ALOGW("loadHwModule() module %s already loaded", name); 1729 return mAudioHwDevs.keyAt(i); 1730 } 1731 } 1732 1733 sp<DeviceHalInterface> dev; 1734 1735 int rc = mDevicesFactoryHal->openDevice(name, &dev); 1736 if (rc) { 1737 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1738 return AUDIO_MODULE_HANDLE_NONE; 1739 } 1740 1741 mHardwareStatus = AUDIO_HW_INIT; 1742 rc = dev->initCheck(); 1743 mHardwareStatus = AUDIO_HW_IDLE; 1744 if (rc) { 1745 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1746 return AUDIO_MODULE_HANDLE_NONE; 1747 } 1748 1749 // Check and cache this HAL's level of support for master mute and master 1750 // volume. If this is the first HAL opened, and it supports the get 1751 // methods, use the initial values provided by the HAL as the current 1752 // master mute and volume settings. 1753 1754 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1755 { // scope for auto-lock pattern 1756 AutoMutex lock(mHardwareLock); 1757 1758 if (0 == mAudioHwDevs.size()) { 1759 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1760 float mv; 1761 if (OK == dev->getMasterVolume(&mv)) { 1762 mMasterVolume = mv; 1763 } 1764 1765 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1766 bool mm; 1767 if (OK == dev->getMasterMute(&mm)) { 1768 mMasterMute = mm; 1769 } 1770 } 1771 1772 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1773 if (OK == dev->setMasterVolume(mMasterVolume)) { 1774 flags = static_cast<AudioHwDevice::Flags>(flags | 1775 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1776 } 1777 1778 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1779 if (OK == dev->setMasterMute(mMasterMute)) { 1780 flags = static_cast<AudioHwDevice::Flags>(flags | 1781 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1782 } 1783 1784 mHardwareStatus = AUDIO_HW_IDLE; 1785 } 1786 1787 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1788 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1789 1790 ALOGI("loadHwModule() Loaded %s audio interface, handle %d", name, handle); 1791 1792 return handle; 1793 1794 } 1795 1796 // ---------------------------------------------------------------------------- 1797 1798 uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1799 { 1800 Mutex::Autolock _l(mLock); 1801 PlaybackThread *thread = fastPlaybackThread_l(); 1802 return thread != NULL ? thread->sampleRate() : 0; 1803 } 1804 1805 size_t AudioFlinger::getPrimaryOutputFrameCount() 1806 { 1807 Mutex::Autolock _l(mLock); 1808 PlaybackThread *thread = fastPlaybackThread_l(); 1809 return thread != NULL ? thread->frameCountHAL() : 0; 1810 } 1811 1812 // ---------------------------------------------------------------------------- 1813 1814 status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1815 { 1816 uid_t uid = IPCThreadState::self()->getCallingUid(); 1817 if (uid != AID_SYSTEM) { 1818 return PERMISSION_DENIED; 1819 } 1820 Mutex::Autolock _l(mLock); 1821 if (mIsDeviceTypeKnown) { 1822 return INVALID_OPERATION; 1823 } 1824 mIsLowRamDevice = isLowRamDevice; 1825 mIsDeviceTypeKnown = true; 1826 return NO_ERROR; 1827 } 1828 1829 audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1830 { 1831 Mutex::Autolock _l(mLock); 1832 1833 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1834 if (index >= 0) { 1835 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1836 mHwAvSyncIds.valueAt(index), sessionId); 1837 return mHwAvSyncIds.valueAt(index); 1838 } 1839 1840 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1841 if (dev == NULL) { 1842 return AUDIO_HW_SYNC_INVALID; 1843 } 1844 String8 reply; 1845 AudioParameter param; 1846 if (dev->getParameters(String8(AudioParameter::keyHwAvSync), &reply) == OK) { 1847 param = AudioParameter(reply); 1848 } 1849 1850 int value; 1851 if (param.getInt(String8(AudioParameter::keyHwAvSync), value) != NO_ERROR) { 1852 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1853 return AUDIO_HW_SYNC_INVALID; 1854 } 1855 1856 // allow only one session for a given HW A/V sync ID. 1857 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1858 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1859 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1860 value, mHwAvSyncIds.keyAt(i)); 1861 mHwAvSyncIds.removeItemsAt(i); 1862 break; 1863 } 1864 } 1865 1866 mHwAvSyncIds.add(sessionId, value); 1867 1868 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1869 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1870 uint32_t sessions = thread->hasAudioSession(sessionId); 1871 if (sessions & ThreadBase::TRACK_SESSION) { 1872 AudioParameter param = AudioParameter(); 1873 param.addInt(String8(AudioParameter::keyStreamHwAvSync), value); 1874 thread->setParameters(param.toString()); 1875 break; 1876 } 1877 } 1878 1879 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1880 return (audio_hw_sync_t)value; 1881 } 1882 1883 status_t AudioFlinger::systemReady() 1884 { 1885 Mutex::Autolock _l(mLock); 1886 ALOGI("%s", __FUNCTION__); 1887 if (mSystemReady) { 1888 ALOGW("%s called twice", __FUNCTION__); 1889 return NO_ERROR; 1890 } 1891 mSystemReady = true; 1892 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1893 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1894 thread->systemReady(); 1895 } 1896 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1897 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1898 thread->systemReady(); 1899 } 1900 return NO_ERROR; 1901 } 1902 1903 // setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1904 void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1905 { 1906 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1907 if (index >= 0) { 1908 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1909 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1910 AudioParameter param = AudioParameter(); 1911 param.addInt(String8(AudioParameter::keyStreamHwAvSync), syncId); 1912 thread->setParameters(param.toString()); 1913 } 1914 } 1915 1916 1917 // ---------------------------------------------------------------------------- 1918 1919 1920 sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module, 1921 audio_io_handle_t *output, 1922 audio_config_t *config, 1923 audio_devices_t devices, 1924 const String8& address, 1925 audio_output_flags_t flags) 1926 { 1927 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1928 if (outHwDev == NULL) { 1929 return 0; 1930 } 1931 1932 if (*output == AUDIO_IO_HANDLE_NONE) { 1933 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1934 } else { 1935 // Audio Policy does not currently request a specific output handle. 1936 // If this is ever needed, see openInput_l() for example code. 1937 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1938 return 0; 1939 } 1940 1941 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1942 1943 // FOR TESTING ONLY: 1944 // This if statement allows overriding the audio policy settings 1945 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1946 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1947 // Check only for Normal Mixing mode 1948 if (kEnableExtendedPrecision) { 1949 // Specify format (uncomment one below to choose) 1950 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1951 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1952 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1953 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1954 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1955 } 1956 if (kEnableExtendedChannels) { 1957 // Specify channel mask (uncomment one below to choose) 1958 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1959 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1960 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1961 } 1962 } 1963 1964 AudioStreamOut *outputStream = NULL; 1965 status_t status = outHwDev->openOutputStream( 1966 &outputStream, 1967 *output, 1968 devices, 1969 flags, 1970 config, 1971 address.string()); 1972 1973 mHardwareStatus = AUDIO_HW_IDLE; 1974 1975 if (status == NO_ERROR) { 1976 if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) { 1977 sp<MmapPlaybackThread> thread = 1978 new MmapPlaybackThread(this, *output, outHwDev, outputStream, 1979 devices, AUDIO_DEVICE_NONE, mSystemReady); 1980 mMmapThreads.add(*output, thread); 1981 ALOGV("openOutput_l() created mmap playback thread: ID %d thread %p", 1982 *output, thread.get()); 1983 return thread; 1984 } else { 1985 sp<PlaybackThread> thread; 1986 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1987 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1988 ALOGV("openOutput_l() created offload output: ID %d thread %p", 1989 *output, thread.get()); 1990 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1991 || !isValidPcmSinkFormat(config->format) 1992 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1993 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1994 ALOGV("openOutput_l() created direct output: ID %d thread %p", 1995 *output, thread.get()); 1996 } else { 1997 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1998 ALOGV("openOutput_l() created mixer output: ID %d thread %p", 1999 *output, thread.get()); 2000 } 2001 mPlaybackThreads.add(*output, thread); 2002 return thread; 2003 } 2004 } 2005 2006 return 0; 2007 } 2008 2009 status_t AudioFlinger::openOutput(audio_module_handle_t module, 2010 audio_io_handle_t *output, 2011 audio_config_t *config, 2012 audio_devices_t *devices, 2013 const String8& address, 2014 uint32_t *latencyMs, 2015 audio_output_flags_t flags) 2016 { 2017 ALOGI("openOutput() this %p, module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, " 2018 "flags %x", 2019 this, module, 2020 (devices != NULL) ? *devices : 0, 2021 config->sample_rate, 2022 config->format, 2023 config->channel_mask, 2024 flags); 2025 2026 if (devices == NULL || *devices == AUDIO_DEVICE_NONE) { 2027 return BAD_VALUE; 2028 } 2029 2030 Mutex::Autolock _l(mLock); 2031 2032 sp<ThreadBase> thread = openOutput_l(module, output, config, *devices, address, flags); 2033 if (thread != 0) { 2034 if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) { 2035 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2036 *latencyMs = playbackThread->latency(); 2037 2038 // notify client processes of the new output creation 2039 playbackThread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 2040 2041 // the first primary output opened designates the primary hw device 2042 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 2043 ALOGI("Using module %d as the primary audio interface", module); 2044 mPrimaryHardwareDev = playbackThread->getOutput()->audioHwDev; 2045 2046 AutoMutex lock(mHardwareLock); 2047 mHardwareStatus = AUDIO_HW_SET_MODE; 2048 mPrimaryHardwareDev->hwDevice()->setMode(mMode); 2049 mHardwareStatus = AUDIO_HW_IDLE; 2050 } 2051 } else { 2052 MmapThread *mmapThread = (MmapThread *)thread.get(); 2053 mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 2054 } 2055 return NO_ERROR; 2056 } 2057 2058 return NO_INIT; 2059 } 2060 2061 audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 2062 audio_io_handle_t output2) 2063 { 2064 Mutex::Autolock _l(mLock); 2065 MixerThread *thread1 = checkMixerThread_l(output1); 2066 MixerThread *thread2 = checkMixerThread_l(output2); 2067 2068 if (thread1 == NULL || thread2 == NULL) { 2069 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 2070 output2); 2071 return AUDIO_IO_HANDLE_NONE; 2072 } 2073 2074 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 2075 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 2076 thread->addOutputTrack(thread2); 2077 mPlaybackThreads.add(id, thread); 2078 // notify client processes of the new output creation 2079 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 2080 return id; 2081 } 2082 2083 status_t AudioFlinger::closeOutput(audio_io_handle_t output) 2084 { 2085 return closeOutput_nonvirtual(output); 2086 } 2087 2088 status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 2089 { 2090 // keep strong reference on the playback thread so that 2091 // it is not destroyed while exit() is executed 2092 sp<PlaybackThread> playbackThread; 2093 sp<MmapPlaybackThread> mmapThread; 2094 { 2095 Mutex::Autolock _l(mLock); 2096 playbackThread = checkPlaybackThread_l(output); 2097 if (playbackThread != NULL) { 2098 ALOGV("closeOutput() %d", output); 2099 2100 if (playbackThread->type() == ThreadBase::MIXER) { 2101 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2102 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 2103 DuplicatingThread *dupThread = 2104 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 2105 dupThread->removeOutputTrack((MixerThread *)playbackThread.get()); 2106 } 2107 } 2108 } 2109 2110 2111 mPlaybackThreads.removeItem(output); 2112 // save all effects to the default thread 2113 if (mPlaybackThreads.size()) { 2114 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 2115 if (dstThread != NULL) { 2116 // audioflinger lock is held so order of thread lock acquisition doesn't matter 2117 Mutex::Autolock _dl(dstThread->mLock); 2118 Mutex::Autolock _sl(playbackThread->mLock); 2119 Vector< sp<EffectChain> > effectChains = playbackThread->getEffectChains_l(); 2120 for (size_t i = 0; i < effectChains.size(); i ++) { 2121 moveEffectChain_l(effectChains[i]->sessionId(), playbackThread.get(), 2122 dstThread, true); 2123 } 2124 } 2125 } 2126 } else { 2127 mmapThread = (MmapPlaybackThread *)checkMmapThread_l(output); 2128 if (mmapThread == 0) { 2129 return BAD_VALUE; 2130 } 2131 mMmapThreads.removeItem(output); 2132 ALOGD("closing mmapThread %p", mmapThread.get()); 2133 } 2134 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2135 ioDesc->mIoHandle = output; 2136 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 2137 } 2138 // The thread entity (active unit of execution) is no longer running here, 2139 // but the ThreadBase container still exists. 2140 2141 if (playbackThread != 0) { 2142 playbackThread->exit(); 2143 if (!playbackThread->isDuplicating()) { 2144 closeOutputFinish(playbackThread); 2145 } 2146 } else if (mmapThread != 0) { 2147 ALOGD("mmapThread exit()"); 2148 mmapThread->exit(); 2149 AudioStreamOut *out = mmapThread->clearOutput(); 2150 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2151 // from now on thread->mOutput is NULL 2152 delete out; 2153 } 2154 return NO_ERROR; 2155 } 2156 2157 void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread) 2158 { 2159 AudioStreamOut *out = thread->clearOutput(); 2160 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2161 // from now on thread->mOutput is NULL 2162 delete out; 2163 } 2164 2165 void AudioFlinger::closeOutputInternal_l(const sp<PlaybackThread>& thread) 2166 { 2167 mPlaybackThreads.removeItem(thread->mId); 2168 thread->exit(); 2169 closeOutputFinish(thread); 2170 } 2171 2172 status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2173 { 2174 Mutex::Autolock _l(mLock); 2175 PlaybackThread *thread = checkPlaybackThread_l(output); 2176 2177 if (thread == NULL) { 2178 return BAD_VALUE; 2179 } 2180 2181 ALOGV("suspendOutput() %d", output); 2182 thread->suspend(); 2183 2184 return NO_ERROR; 2185 } 2186 2187 status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2188 { 2189 Mutex::Autolock _l(mLock); 2190 PlaybackThread *thread = checkPlaybackThread_l(output); 2191 2192 if (thread == NULL) { 2193 return BAD_VALUE; 2194 } 2195 2196 ALOGV("restoreOutput() %d", output); 2197 2198 thread->restore(); 2199 2200 return NO_ERROR; 2201 } 2202 2203 status_t AudioFlinger::openInput(audio_module_handle_t module, 2204 audio_io_handle_t *input, 2205 audio_config_t *config, 2206 audio_devices_t *devices, 2207 const String8& address, 2208 audio_source_t source, 2209 audio_input_flags_t flags) 2210 { 2211 Mutex::Autolock _l(mLock); 2212 2213 if (*devices == AUDIO_DEVICE_NONE) { 2214 return BAD_VALUE; 2215 } 2216 2217 sp<ThreadBase> thread = openInput_l(module, input, config, *devices, address, source, flags); 2218 2219 if (thread != 0) { 2220 // notify client processes of the new input creation 2221 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2222 return NO_ERROR; 2223 } 2224 return NO_INIT; 2225 } 2226 2227 sp<AudioFlinger::ThreadBase> AudioFlinger::openInput_l(audio_module_handle_t module, 2228 audio_io_handle_t *input, 2229 audio_config_t *config, 2230 audio_devices_t devices, 2231 const String8& address, 2232 audio_source_t source, 2233 audio_input_flags_t flags) 2234 { 2235 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2236 if (inHwDev == NULL) { 2237 *input = AUDIO_IO_HANDLE_NONE; 2238 return 0; 2239 } 2240 2241 // Audio Policy can request a specific handle for hardware hotword. 2242 // The goal here is not to re-open an already opened input. 2243 // It is to use a pre-assigned I/O handle. 2244 if (*input == AUDIO_IO_HANDLE_NONE) { 2245 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2246 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2247 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2248 return 0; 2249 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2250 // This should not happen in a transient state with current design. 2251 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2252 return 0; 2253 } 2254 2255 audio_config_t halconfig = *config; 2256 sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice(); 2257 sp<StreamInHalInterface> inStream; 2258 status_t status = inHwHal->openInputStream( 2259 *input, devices, &halconfig, flags, address.string(), source, &inStream); 2260 ALOGV("openInput_l() openInputStream returned input %p, devices %x, SamplingRate %d" 2261 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2262 inStream.get(), 2263 devices, 2264 halconfig.sample_rate, 2265 halconfig.format, 2266 halconfig.channel_mask, 2267 flags, 2268 status, address.string()); 2269 2270 // If the input could not be opened with the requested parameters and we can handle the 2271 // conversion internally, try to open again with the proposed parameters. 2272 if (status == BAD_VALUE && 2273 audio_is_linear_pcm(config->format) && 2274 audio_is_linear_pcm(halconfig.format) && 2275 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2276 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2277 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2278 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2279 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2280 inStream.clear(); 2281 status = inHwHal->openInputStream( 2282 *input, devices, &halconfig, flags, address.string(), source, &inStream); 2283 // FIXME log this new status; HAL should not propose any further changes 2284 } 2285 2286 if (status == NO_ERROR && inStream != 0) { 2287 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags); 2288 if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) { 2289 sp<MmapCaptureThread> thread = 2290 new MmapCaptureThread(this, *input, 2291 inHwDev, inputStream, 2292 primaryOutputDevice_l(), devices, mSystemReady); 2293 mMmapThreads.add(*input, thread); 2294 ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input, 2295 thread.get()); 2296 return thread; 2297 } else { 2298 #ifdef TEE_SINK 2299 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2300 // or (re-)create if current Pipe is idle and does not match the new format 2301 sp<NBAIO_Sink> teeSink; 2302 enum { 2303 TEE_SINK_NO, // don't copy input 2304 TEE_SINK_NEW, // copy input using a new pipe 2305 TEE_SINK_OLD, // copy input using an existing pipe 2306 } kind; 2307 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2308 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2309 if (!mTeeSinkInputEnabled) { 2310 kind = TEE_SINK_NO; 2311 } else if (!Format_isValid(format)) { 2312 kind = TEE_SINK_NO; 2313 } else if (mRecordTeeSink == 0) { 2314 kind = TEE_SINK_NEW; 2315 } else if (mRecordTeeSink->getStrongCount() != 1) { 2316 kind = TEE_SINK_NO; 2317 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2318 kind = TEE_SINK_OLD; 2319 } else { 2320 kind = TEE_SINK_NEW; 2321 } 2322 switch (kind) { 2323 case TEE_SINK_NEW: { 2324 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2325 size_t numCounterOffers = 0; 2326 const NBAIO_Format offers[1] = {format}; 2327 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2328 ALOG_ASSERT(index == 0); 2329 PipeReader *pipeReader = new PipeReader(*pipe); 2330 numCounterOffers = 0; 2331 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2332 ALOG_ASSERT(index == 0); 2333 mRecordTeeSink = pipe; 2334 mRecordTeeSource = pipeReader; 2335 teeSink = pipe; 2336 } 2337 break; 2338 case TEE_SINK_OLD: 2339 teeSink = mRecordTeeSink; 2340 break; 2341 case TEE_SINK_NO: 2342 default: 2343 break; 2344 } 2345 #endif 2346 2347 // Start record thread 2348 // RecordThread requires both input and output device indication to forward to audio 2349 // pre processing modules 2350 sp<RecordThread> thread = new RecordThread(this, 2351 inputStream, 2352 *input, 2353 primaryOutputDevice_l(), 2354 devices, 2355 mSystemReady 2356 #ifdef TEE_SINK 2357 , teeSink 2358 #endif 2359 ); 2360 mRecordThreads.add(*input, thread); 2361 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2362 return thread; 2363 } 2364 } 2365 2366 *input = AUDIO_IO_HANDLE_NONE; 2367 return 0; 2368 } 2369 2370 status_t AudioFlinger::closeInput(audio_io_handle_t input) 2371 { 2372 return closeInput_nonvirtual(input); 2373 } 2374 2375 status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2376 { 2377 // keep strong reference on the record thread so that 2378 // it is not destroyed while exit() is executed 2379 sp<RecordThread> recordThread; 2380 sp<MmapCaptureThread> mmapThread; 2381 { 2382 Mutex::Autolock _l(mLock); 2383 recordThread = checkRecordThread_l(input); 2384 if (recordThread != 0) { 2385 ALOGV("closeInput() %d", input); 2386 2387 // If we still have effect chains, it means that a client still holds a handle 2388 // on at least one effect. We must either move the chain to an existing thread with the 2389 // same session ID or put it aside in case a new record thread is opened for a 2390 // new capture on the same session 2391 sp<EffectChain> chain; 2392 { 2393 Mutex::Autolock _sl(recordThread->mLock); 2394 Vector< sp<EffectChain> > effectChains = recordThread->getEffectChains_l(); 2395 // Note: maximum one chain per record thread 2396 if (effectChains.size() != 0) { 2397 chain = effectChains[0]; 2398 } 2399 } 2400 if (chain != 0) { 2401 // first check if a record thread is already opened with a client on same session. 2402 // This should only happen in case of overlap between one thread tear down and the 2403 // creation of its replacement 2404 size_t i; 2405 for (i = 0; i < mRecordThreads.size(); i++) { 2406 sp<RecordThread> t = mRecordThreads.valueAt(i); 2407 if (t == recordThread) { 2408 continue; 2409 } 2410 if (t->hasAudioSession(chain->sessionId()) != 0) { 2411 Mutex::Autolock _l(t->mLock); 2412 ALOGV("closeInput() found thread %d for effect session %d", 2413 t->id(), chain->sessionId()); 2414 t->addEffectChain_l(chain); 2415 break; 2416 } 2417 } 2418 // put the chain aside if we could not find a record thread with the same session id 2419 if (i == mRecordThreads.size()) { 2420 putOrphanEffectChain_l(chain); 2421 } 2422 } 2423 mRecordThreads.removeItem(input); 2424 } else { 2425 mmapThread = (MmapCaptureThread *)checkMmapThread_l(input); 2426 if (mmapThread == 0) { 2427 return BAD_VALUE; 2428 } 2429 mMmapThreads.removeItem(input); 2430 } 2431 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2432 ioDesc->mIoHandle = input; 2433 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2434 } 2435 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2436 // we have a different lock for notification client 2437 if (recordThread != 0) { 2438 closeInputFinish(recordThread); 2439 } else if (mmapThread != 0) { 2440 mmapThread->exit(); 2441 AudioStreamIn *in = mmapThread->clearInput(); 2442 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2443 // from now on thread->mInput is NULL 2444 delete in; 2445 } 2446 return NO_ERROR; 2447 } 2448 2449 void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread) 2450 { 2451 thread->exit(); 2452 AudioStreamIn *in = thread->clearInput(); 2453 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2454 // from now on thread->mInput is NULL 2455 delete in; 2456 } 2457 2458 void AudioFlinger::closeInputInternal_l(const sp<RecordThread>& thread) 2459 { 2460 mRecordThreads.removeItem(thread->mId); 2461 closeInputFinish(thread); 2462 } 2463 2464 status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2465 { 2466 Mutex::Autolock _l(mLock); 2467 ALOGV("invalidateStream() stream %d", stream); 2468 2469 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2470 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2471 thread->invalidateTracks(stream); 2472 } 2473 for (size_t i = 0; i < mMmapThreads.size(); i++) { 2474 mMmapThreads[i]->invalidateTracks(stream); 2475 } 2476 return NO_ERROR; 2477 } 2478 2479 2480 audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2481 { 2482 // This is a binder API, so a malicious client could pass in a bad parameter. 2483 // Check for that before calling the internal API nextUniqueId(). 2484 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2485 ALOGE("newAudioUniqueId invalid use %d", use); 2486 return AUDIO_UNIQUE_ID_ALLOCATE; 2487 } 2488 return nextUniqueId(use); 2489 } 2490 2491 void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2492 { 2493 Mutex::Autolock _l(mLock); 2494 pid_t caller = IPCThreadState::self()->getCallingPid(); 2495 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2496 if (pid != -1 && (caller == getpid_cached)) { 2497 caller = pid; 2498 } 2499 2500 { 2501 Mutex::Autolock _cl(mClientLock); 2502 // Ignore requests received from processes not known as notification client. The request 2503 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2504 // called from a different pid leaving a stale session reference. Also we don't know how 2505 // to clear this reference if the client process dies. 2506 if (mNotificationClients.indexOfKey(caller) < 0) { 2507 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2508 return; 2509 } 2510 } 2511 2512 size_t num = mAudioSessionRefs.size(); 2513 for (size_t i = 0; i < num; i++) { 2514 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2515 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2516 ref->mCnt++; 2517 ALOGV(" incremented refcount to %d", ref->mCnt); 2518 return; 2519 } 2520 } 2521 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2522 ALOGV(" added new entry for %d", audioSession); 2523 } 2524 2525 void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2526 { 2527 Mutex::Autolock _l(mLock); 2528 pid_t caller = IPCThreadState::self()->getCallingPid(); 2529 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2530 if (pid != -1 && (caller == getpid_cached)) { 2531 caller = pid; 2532 } 2533 size_t num = mAudioSessionRefs.size(); 2534 for (size_t i = 0; i < num; i++) { 2535 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2536 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2537 ref->mCnt--; 2538 ALOGV(" decremented refcount to %d", ref->mCnt); 2539 if (ref->mCnt == 0) { 2540 mAudioSessionRefs.removeAt(i); 2541 delete ref; 2542 purgeStaleEffects_l(); 2543 } 2544 return; 2545 } 2546 } 2547 // If the caller is mediaserver it is likely that the session being released was acquired 2548 // on behalf of a process not in notification clients and we ignore the warning. 2549 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2550 } 2551 2552 bool AudioFlinger::isSessionAcquired_l(audio_session_t audioSession) 2553 { 2554 size_t num = mAudioSessionRefs.size(); 2555 for (size_t i = 0; i < num; i++) { 2556 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2557 if (ref->mSessionid == audioSession) { 2558 return true; 2559 } 2560 } 2561 return false; 2562 } 2563 2564 void AudioFlinger::purgeStaleEffects_l() { 2565 2566 ALOGV("purging stale effects"); 2567 2568 Vector< sp<EffectChain> > chains; 2569 2570 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2571 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2572 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2573 sp<EffectChain> ec = t->mEffectChains[j]; 2574 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2575 chains.push(ec); 2576 } 2577 } 2578 } 2579 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2580 sp<RecordThread> t = mRecordThreads.valueAt(i); 2581 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2582 sp<EffectChain> ec = t->mEffectChains[j]; 2583 chains.push(ec); 2584 } 2585 } 2586 2587 for (size_t i = 0; i < chains.size(); i++) { 2588 sp<EffectChain> ec = chains[i]; 2589 int sessionid = ec->sessionId(); 2590 sp<ThreadBase> t = ec->mThread.promote(); 2591 if (t == 0) { 2592 continue; 2593 } 2594 size_t numsessionrefs = mAudioSessionRefs.size(); 2595 bool found = false; 2596 for (size_t k = 0; k < numsessionrefs; k++) { 2597 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2598 if (ref->mSessionid == sessionid) { 2599 ALOGV(" session %d still exists for %d with %d refs", 2600 sessionid, ref->mPid, ref->mCnt); 2601 found = true; 2602 break; 2603 } 2604 } 2605 if (!found) { 2606 Mutex::Autolock _l(t->mLock); 2607 // remove all effects from the chain 2608 while (ec->mEffects.size()) { 2609 sp<EffectModule> effect = ec->mEffects[0]; 2610 effect->unPin(); 2611 t->removeEffect_l(effect, /*release*/ true); 2612 if (effect->purgeHandles()) { 2613 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2614 } 2615 AudioSystem::unregisterEffect(effect->id()); 2616 } 2617 } 2618 } 2619 return; 2620 } 2621 2622 // checkThread_l() must be called with AudioFlinger::mLock held 2623 AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2624 { 2625 ThreadBase *thread = checkMmapThread_l(ioHandle); 2626 if (thread == 0) { 2627 switch (audio_unique_id_get_use(ioHandle)) { 2628 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2629 thread = checkPlaybackThread_l(ioHandle); 2630 break; 2631 case AUDIO_UNIQUE_ID_USE_INPUT: 2632 thread = checkRecordThread_l(ioHandle); 2633 break; 2634 default: 2635 break; 2636 } 2637 } 2638 return thread; 2639 } 2640 2641 // checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2642 AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2643 { 2644 return mPlaybackThreads.valueFor(output).get(); 2645 } 2646 2647 // checkMixerThread_l() must be called with AudioFlinger::mLock held 2648 AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2649 { 2650 PlaybackThread *thread = checkPlaybackThread_l(output); 2651 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2652 } 2653 2654 // checkRecordThread_l() must be called with AudioFlinger::mLock held 2655 AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2656 { 2657 return mRecordThreads.valueFor(input).get(); 2658 } 2659 2660 // checkMmapThread_l() must be called with AudioFlinger::mLock held 2661 AudioFlinger::MmapThread *AudioFlinger::checkMmapThread_l(audio_io_handle_t io) const 2662 { 2663 return mMmapThreads.valueFor(io).get(); 2664 } 2665 2666 2667 // checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2668 AudioFlinger::VolumeInterface *AudioFlinger::getVolumeInterface_l(audio_io_handle_t output) const 2669 { 2670 VolumeInterface *volumeInterface = mPlaybackThreads.valueFor(output).get(); 2671 if (volumeInterface == nullptr) { 2672 MmapThread *mmapThread = mMmapThreads.valueFor(output).get(); 2673 if (mmapThread != nullptr) { 2674 if (mmapThread->isOutput()) { 2675 MmapPlaybackThread *mmapPlaybackThread = 2676 static_cast<MmapPlaybackThread *>(mmapThread); 2677 volumeInterface = mmapPlaybackThread; 2678 } 2679 } 2680 } 2681 return volumeInterface; 2682 } 2683 2684 Vector <AudioFlinger::VolumeInterface *> AudioFlinger::getAllVolumeInterfaces_l() const 2685 { 2686 Vector <VolumeInterface *> volumeInterfaces; 2687 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2688 volumeInterfaces.add(mPlaybackThreads.valueAt(i).get()); 2689 } 2690 for (size_t i = 0; i < mMmapThreads.size(); i++) { 2691 if (mMmapThreads.valueAt(i)->isOutput()) { 2692 MmapPlaybackThread *mmapPlaybackThread = 2693 static_cast<MmapPlaybackThread *>(mMmapThreads.valueAt(i).get()); 2694 volumeInterfaces.add(mmapPlaybackThread); 2695 } 2696 } 2697 return volumeInterfaces; 2698 } 2699 2700 audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2701 { 2702 // This is the internal API, so it is OK to assert on bad parameter. 2703 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2704 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2705 for (int retry = 0; retry < maxRetries; retry++) { 2706 // The cast allows wraparound from max positive to min negative instead of abort 2707 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2708 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2709 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2710 // allow wrap by skipping 0 and -1 for session ids 2711 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2712 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2713 return (audio_unique_id_t) (base | use); 2714 } 2715 } 2716 // We have no way of recovering from wraparound 2717 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2718 // TODO Use a floor after wraparound. This may need a mutex. 2719 } 2720 2721 AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2722 { 2723 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2724 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2725 if(thread->isDuplicating()) { 2726 continue; 2727 } 2728 AudioStreamOut *output = thread->getOutput(); 2729 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2730 return thread; 2731 } 2732 } 2733 return NULL; 2734 } 2735 2736 audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2737 { 2738 PlaybackThread *thread = primaryPlaybackThread_l(); 2739 2740 if (thread == NULL) { 2741 return 0; 2742 } 2743 2744 return thread->outDevice(); 2745 } 2746 2747 AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const 2748 { 2749 size_t minFrameCount = 0; 2750 PlaybackThread *minThread = NULL; 2751 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2752 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2753 if (!thread->isDuplicating()) { 2754 size_t frameCount = thread->frameCountHAL(); 2755 if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount || 2756 (frameCount == minFrameCount && thread->hasFastMixer() && 2757 /*minThread != NULL &&*/ !minThread->hasFastMixer()))) { 2758 minFrameCount = frameCount; 2759 minThread = thread; 2760 } 2761 } 2762 } 2763 return minThread; 2764 } 2765 2766 sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2767 audio_session_t triggerSession, 2768 audio_session_t listenerSession, 2769 sync_event_callback_t callBack, 2770 const wp<RefBase>& cookie) 2771 { 2772 Mutex::Autolock _l(mLock); 2773 2774 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2775 status_t playStatus = NAME_NOT_FOUND; 2776 status_t recStatus = NAME_NOT_FOUND; 2777 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2778 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2779 if (playStatus == NO_ERROR) { 2780 return event; 2781 } 2782 } 2783 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2784 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2785 if (recStatus == NO_ERROR) { 2786 return event; 2787 } 2788 } 2789 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2790 mPendingSyncEvents.add(event); 2791 } else { 2792 ALOGV("createSyncEvent() invalid event %d", event->type()); 2793 event.clear(); 2794 } 2795 return event; 2796 } 2797 2798 // ---------------------------------------------------------------------------- 2799 // Effect management 2800 // ---------------------------------------------------------------------------- 2801 2802 sp<EffectsFactoryHalInterface> AudioFlinger::getEffectsFactory() { 2803 return mEffectsFactoryHal; 2804 } 2805 2806 status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2807 { 2808 Mutex::Autolock _l(mLock); 2809 if (mEffectsFactoryHal.get()) { 2810 return mEffectsFactoryHal->queryNumberEffects(numEffects); 2811 } else { 2812 return -ENODEV; 2813 } 2814 } 2815 2816 status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2817 { 2818 Mutex::Autolock _l(mLock); 2819 if (mEffectsFactoryHal.get()) { 2820 return mEffectsFactoryHal->getDescriptor(index, descriptor); 2821 } else { 2822 return -ENODEV; 2823 } 2824 } 2825 2826 status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2827 effect_descriptor_t *descriptor) const 2828 { 2829 Mutex::Autolock _l(mLock); 2830 if (mEffectsFactoryHal.get()) { 2831 return mEffectsFactoryHal->getDescriptor(pUuid, descriptor); 2832 } else { 2833 return -ENODEV; 2834 } 2835 } 2836 2837 2838 sp<IEffect> AudioFlinger::createEffect( 2839 effect_descriptor_t *pDesc, 2840 const sp<IEffectClient>& effectClient, 2841 int32_t priority, 2842 audio_io_handle_t io, 2843 audio_session_t sessionId, 2844 const String16& opPackageName, 2845 pid_t pid, 2846 status_t *status, 2847 int *id, 2848 int *enabled) 2849 { 2850 status_t lStatus = NO_ERROR; 2851 sp<EffectHandle> handle; 2852 effect_descriptor_t desc; 2853 2854 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 2855 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 2856 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 2857 ALOGW_IF(pid != -1 && pid != callingPid, 2858 "%s uid %d pid %d tried to pass itself off as pid %d", 2859 __func__, callingUid, callingPid, pid); 2860 pid = callingPid; 2861 } 2862 2863 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p", 2864 pid, effectClient.get(), priority, sessionId, io, mEffectsFactoryHal.get()); 2865 2866 if (pDesc == NULL) { 2867 lStatus = BAD_VALUE; 2868 goto Exit; 2869 } 2870 2871 // check audio settings permission for global effects 2872 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2873 lStatus = PERMISSION_DENIED; 2874 goto Exit; 2875 } 2876 2877 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2878 // that can only be created by audio policy manager (running in same process) 2879 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2880 lStatus = PERMISSION_DENIED; 2881 goto Exit; 2882 } 2883 2884 if (mEffectsFactoryHal == 0) { 2885 lStatus = NO_INIT; 2886 goto Exit; 2887 } 2888 2889 { 2890 if (!EffectsFactoryHalInterface::isNullUuid(&pDesc->uuid)) { 2891 // if uuid is specified, request effect descriptor 2892 lStatus = mEffectsFactoryHal->getDescriptor(&pDesc->uuid, &desc); 2893 if (lStatus < 0) { 2894 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2895 goto Exit; 2896 } 2897 } else { 2898 // if uuid is not specified, look for an available implementation 2899 // of the required type in effect factory 2900 if (EffectsFactoryHalInterface::isNullUuid(&pDesc->type)) { 2901 ALOGW("createEffect() no effect type"); 2902 lStatus = BAD_VALUE; 2903 goto Exit; 2904 } 2905 uint32_t numEffects = 0; 2906 effect_descriptor_t d; 2907 d.flags = 0; // prevent compiler warning 2908 bool found = false; 2909 2910 lStatus = mEffectsFactoryHal->queryNumberEffects(&numEffects); 2911 if (lStatus < 0) { 2912 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2913 goto Exit; 2914 } 2915 for (uint32_t i = 0; i < numEffects; i++) { 2916 lStatus = mEffectsFactoryHal->getDescriptor(i, &desc); 2917 if (lStatus < 0) { 2918 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2919 continue; 2920 } 2921 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2922 // If matching type found save effect descriptor. If the session is 2923 // 0 and the effect is not auxiliary, continue enumeration in case 2924 // an auxiliary version of this effect type is available 2925 found = true; 2926 d = desc; 2927 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2928 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2929 break; 2930 } 2931 } 2932 } 2933 if (!found) { 2934 lStatus = BAD_VALUE; 2935 ALOGW("createEffect() effect not found"); 2936 goto Exit; 2937 } 2938 // For same effect type, chose auxiliary version over insert version if 2939 // connect to output mix (Compliance to OpenSL ES) 2940 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2941 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2942 desc = d; 2943 } 2944 } 2945 2946 // Do not allow auxiliary effects on a session different from 0 (output mix) 2947 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2948 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2949 lStatus = INVALID_OPERATION; 2950 goto Exit; 2951 } 2952 2953 // check recording permission for visualizer 2954 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2955 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2956 lStatus = PERMISSION_DENIED; 2957 goto Exit; 2958 } 2959 2960 // return effect descriptor 2961 *pDesc = desc; 2962 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2963 // if the output returned by getOutputForEffect() is removed before we lock the 2964 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2965 // and we will exit safely 2966 io = AudioSystem::getOutputForEffect(&desc); 2967 ALOGV("createEffect got output %d", io); 2968 } 2969 2970 Mutex::Autolock _l(mLock); 2971 2972 // If output is not specified try to find a matching audio session ID in one of the 2973 // output threads. 2974 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2975 // because of code checking output when entering the function. 2976 // Note: io is never 0 when creating an effect on an input 2977 if (io == AUDIO_IO_HANDLE_NONE) { 2978 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2979 // output must be specified by AudioPolicyManager when using session 2980 // AUDIO_SESSION_OUTPUT_STAGE 2981 lStatus = BAD_VALUE; 2982 goto Exit; 2983 } 2984 // look for the thread where the specified audio session is present 2985 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2986 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2987 io = mPlaybackThreads.keyAt(i); 2988 break; 2989 } 2990 } 2991 if (io == AUDIO_IO_HANDLE_NONE) { 2992 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2993 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2994 io = mRecordThreads.keyAt(i); 2995 break; 2996 } 2997 } 2998 } 2999 if (io == AUDIO_IO_HANDLE_NONE) { 3000 for (size_t i = 0; i < mMmapThreads.size(); i++) { 3001 if (mMmapThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 3002 io = mMmapThreads.keyAt(i); 3003 break; 3004 } 3005 } 3006 } 3007 // If no output thread contains the requested session ID, default to 3008 // first output. The effect chain will be moved to the correct output 3009 // thread when a track with the same session ID is created 3010 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 3011 io = mPlaybackThreads.keyAt(0); 3012 } 3013 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 3014 } 3015 ThreadBase *thread = checkRecordThread_l(io); 3016 if (thread == NULL) { 3017 thread = checkPlaybackThread_l(io); 3018 if (thread == NULL) { 3019 thread = checkMmapThread_l(io); 3020 if (thread == NULL) { 3021 ALOGE("createEffect() unknown output thread"); 3022 lStatus = BAD_VALUE; 3023 goto Exit; 3024 } 3025 } 3026 } else { 3027 // Check if one effect chain was awaiting for an effect to be created on this 3028 // session and used it instead of creating a new one. 3029 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 3030 if (chain != 0) { 3031 Mutex::Autolock _l(thread->mLock); 3032 thread->addEffectChain_l(chain); 3033 } 3034 } 3035 3036 sp<Client> client = registerPid(pid); 3037 3038 // create effect on selected output thread 3039 bool pinned = (sessionId > AUDIO_SESSION_OUTPUT_MIX) && isSessionAcquired_l(sessionId); 3040 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 3041 &desc, enabled, &lStatus, pinned); 3042 if (handle != 0 && id != NULL) { 3043 *id = handle->id(); 3044 } 3045 if (handle == 0) { 3046 // remove local strong reference to Client with mClientLock held 3047 Mutex::Autolock _cl(mClientLock); 3048 client.clear(); 3049 } 3050 } 3051 3052 Exit: 3053 *status = lStatus; 3054 return handle; 3055 } 3056 3057 status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 3058 audio_io_handle_t dstOutput) 3059 { 3060 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 3061 sessionId, srcOutput, dstOutput); 3062 Mutex::Autolock _l(mLock); 3063 if (srcOutput == dstOutput) { 3064 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 3065 return NO_ERROR; 3066 } 3067 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 3068 if (srcThread == NULL) { 3069 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 3070 return BAD_VALUE; 3071 } 3072 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 3073 if (dstThread == NULL) { 3074 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 3075 return BAD_VALUE; 3076 } 3077 3078 Mutex::Autolock _dl(dstThread->mLock); 3079 Mutex::Autolock _sl(srcThread->mLock); 3080 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 3081 } 3082 3083 // moveEffectChain_l must be called with both srcThread and dstThread mLocks held 3084 status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 3085 AudioFlinger::PlaybackThread *srcThread, 3086 AudioFlinger::PlaybackThread *dstThread, 3087 bool reRegister) 3088 { 3089 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 3090 sessionId, srcThread, dstThread); 3091 3092 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 3093 if (chain == 0) { 3094 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 3095 sessionId, srcThread); 3096 return INVALID_OPERATION; 3097 } 3098 3099 // Check whether the destination thread and all effects in the chain are compatible 3100 if (!chain->isCompatibleWithThread_l(dstThread)) { 3101 ALOGW("moveEffectChain_l() effect chain failed because" 3102 " destination thread %p is not compatible with effects in the chain", 3103 dstThread); 3104 return INVALID_OPERATION; 3105 } 3106 3107 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 3108 // so that a new chain is created with correct parameters when first effect is added. This is 3109 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 3110 // removed. 3111 srcThread->removeEffectChain_l(chain); 3112 3113 // transfer all effects one by one so that new effect chain is created on new thread with 3114 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 3115 sp<EffectChain> dstChain; 3116 uint32_t strategy = 0; // prevent compiler warning 3117 sp<EffectModule> effect = chain->getEffectFromId_l(0); 3118 Vector< sp<EffectModule> > removed; 3119 status_t status = NO_ERROR; 3120 while (effect != 0) { 3121 srcThread->removeEffect_l(effect); 3122 removed.add(effect); 3123 status = dstThread->addEffect_l(effect); 3124 if (status != NO_ERROR) { 3125 break; 3126 } 3127 // removeEffect_l() has stopped the effect if it was active so it must be restarted 3128 if (effect->state() == EffectModule::ACTIVE || 3129 effect->state() == EffectModule::STOPPING) { 3130 effect->start(); 3131 } 3132 // if the move request is not received from audio policy manager, the effect must be 3133 // re-registered with the new strategy and output 3134 if (dstChain == 0) { 3135 dstChain = effect->chain().promote(); 3136 if (dstChain == 0) { 3137 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 3138 status = NO_INIT; 3139 break; 3140 } 3141 strategy = dstChain->strategy(); 3142 } 3143 if (reRegister) { 3144 AudioSystem::unregisterEffect(effect->id()); 3145 AudioSystem::registerEffect(&effect->desc(), 3146 dstThread->id(), 3147 strategy, 3148 sessionId, 3149 effect->id()); 3150 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 3151 } 3152 effect = chain->getEffectFromId_l(0); 3153 } 3154 3155 if (status != NO_ERROR) { 3156 for (size_t i = 0; i < removed.size(); i++) { 3157 srcThread->addEffect_l(removed[i]); 3158 if (dstChain != 0 && reRegister) { 3159 AudioSystem::unregisterEffect(removed[i]->id()); 3160 AudioSystem::registerEffect(&removed[i]->desc(), 3161 srcThread->id(), 3162 strategy, 3163 sessionId, 3164 removed[i]->id()); 3165 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 3166 } 3167 } 3168 } 3169 3170 return status; 3171 } 3172 3173 bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 3174 { 3175 if (mGlobalEffectEnableTime != 0 && 3176 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 3177 return true; 3178 } 3179 3180 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 3181 sp<EffectChain> ec = 3182 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3183 if (ec != 0 && ec->isNonOffloadableEnabled()) { 3184 return true; 3185 } 3186 } 3187 return false; 3188 } 3189 3190 void AudioFlinger::onNonOffloadableGlobalEffectEnable() 3191 { 3192 Mutex::Autolock _l(mLock); 3193 3194 mGlobalEffectEnableTime = systemTime(); 3195 3196 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 3197 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 3198 if (t->mType == ThreadBase::OFFLOAD) { 3199 t->invalidateTracks(AUDIO_STREAM_MUSIC); 3200 } 3201 } 3202 3203 } 3204 3205 status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 3206 { 3207 // clear possible suspended state before parking the chain so that it starts in default state 3208 // when attached to a new record thread 3209 chain->setEffectSuspended_l(FX_IID_AEC, false); 3210 chain->setEffectSuspended_l(FX_IID_NS, false); 3211 3212 audio_session_t session = chain->sessionId(); 3213 ssize_t index = mOrphanEffectChains.indexOfKey(session); 3214 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 3215 if (index >= 0) { 3216 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 3217 return ALREADY_EXISTS; 3218 } 3219 mOrphanEffectChains.add(session, chain); 3220 return NO_ERROR; 3221 } 3222 3223 sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 3224 { 3225 sp<EffectChain> chain; 3226 ssize_t index = mOrphanEffectChains.indexOfKey(session); 3227 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 3228 if (index >= 0) { 3229 chain = mOrphanEffectChains.valueAt(index); 3230 mOrphanEffectChains.removeItemsAt(index); 3231 } 3232 return chain; 3233 } 3234 3235 bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 3236 { 3237 Mutex::Autolock _l(mLock); 3238 audio_session_t session = effect->sessionId(); 3239 ssize_t index = mOrphanEffectChains.indexOfKey(session); 3240 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 3241 if (index >= 0) { 3242 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 3243 if (chain->removeEffect_l(effect, true) == 0) { 3244 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 3245 mOrphanEffectChains.removeItemsAt(index); 3246 } 3247 return true; 3248 } 3249 return false; 3250 } 3251 3252 3253 struct Entry { 3254 #define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 3255 char mFileName[TEE_MAX_FILENAME]; 3256 }; 3257 3258 int comparEntry(const void *p1, const void *p2) 3259 { 3260 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 3261 } 3262 3263 #ifdef TEE_SINK 3264 void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id, char suffix) 3265 { 3266 NBAIO_Source *teeSource = source.get(); 3267 if (teeSource != NULL) { 3268 // .wav rotation 3269 // There is a benign race condition if 2 threads call this simultaneously. 3270 // They would both traverse the directory, but the result would simply be 3271 // failures at unlink() which are ignored. It's also unlikely since 3272 // normally dumpsys is only done by bugreport or from the command line. 3273 char teePath[32+256]; 3274 strcpy(teePath, "/data/misc/audioserver"); 3275 size_t teePathLen = strlen(teePath); 3276 DIR *dir = opendir(teePath); 3277 teePath[teePathLen++] = '/'; 3278 if (dir != NULL) { 3279 #define TEE_MAX_SORT 20 // number of entries to sort 3280 #define TEE_MAX_KEEP 10 // number of entries to keep 3281 struct Entry entries[TEE_MAX_SORT]; 3282 size_t entryCount = 0; 3283 while (entryCount < TEE_MAX_SORT) { 3284 struct dirent de; 3285 struct dirent *result = NULL; 3286 int rc = readdir_r(dir, &de, &result); 3287 if (rc != 0) { 3288 ALOGW("readdir_r failed %d", rc); 3289 break; 3290 } 3291 if (result == NULL) { 3292 break; 3293 } 3294 if (result != &de) { 3295 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 3296 break; 3297 } 3298 // ignore non .wav file entries 3299 size_t nameLen = strlen(de.d_name); 3300 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 3301 strcmp(&de.d_name[nameLen - 4], ".wav")) { 3302 continue; 3303 } 3304 strcpy(entries[entryCount++].mFileName, de.d_name); 3305 } 3306 (void) closedir(dir); 3307 if (entryCount > TEE_MAX_KEEP) { 3308 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3309 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3310 strcpy(&teePath[teePathLen], entries[i].mFileName); 3311 (void) unlink(teePath); 3312 } 3313 } 3314 } else { 3315 if (fd >= 0) { 3316 dprintf(fd, "unable to rotate tees in %.*s: %s\n", (int) teePathLen, teePath, 3317 strerror(errno)); 3318 } 3319 } 3320 char teeTime[16]; 3321 struct timeval tv; 3322 gettimeofday(&tv, NULL); 3323 struct tm tm; 3324 localtime_r(&tv.tv_sec, &tm); 3325 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3326 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d_%c.wav", teeTime, id, 3327 suffix); 3328 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3329 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3330 if (teeFd >= 0) { 3331 // FIXME use libsndfile 3332 char wavHeader[44]; 3333 memcpy(wavHeader, 3334 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 3335 sizeof(wavHeader)); 3336 NBAIO_Format format = teeSource->format(); 3337 unsigned channelCount = Format_channelCount(format); 3338 uint32_t sampleRate = Format_sampleRate(format); 3339 size_t frameSize = Format_frameSize(format); 3340 wavHeader[22] = channelCount; // number of channels 3341 wavHeader[24] = sampleRate; // sample rate 3342 wavHeader[25] = sampleRate >> 8; 3343 wavHeader[32] = frameSize; // block alignment 3344 wavHeader[33] = frameSize >> 8; 3345 write(teeFd, wavHeader, sizeof(wavHeader)); 3346 size_t total = 0; 3347 bool firstRead = true; 3348 #define TEE_SINK_READ 1024 // frames per I/O operation 3349 void *buffer = malloc(TEE_SINK_READ * frameSize); 3350 for (;;) { 3351 size_t count = TEE_SINK_READ; 3352 ssize_t actual = teeSource->read(buffer, count); 3353 bool wasFirstRead = firstRead; 3354 firstRead = false; 3355 if (actual <= 0) { 3356 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3357 continue; 3358 } 3359 break; 3360 } 3361 ALOG_ASSERT(actual <= (ssize_t)count); 3362 write(teeFd, buffer, actual * frameSize); 3363 total += actual; 3364 } 3365 free(buffer); 3366 lseek(teeFd, (off_t) 4, SEEK_SET); 3367 uint32_t temp = 44 + total * frameSize - 8; 3368 // FIXME not big-endian safe 3369 write(teeFd, &temp, sizeof(temp)); 3370 lseek(teeFd, (off_t) 40, SEEK_SET); 3371 temp = total * frameSize; 3372 // FIXME not big-endian safe 3373 write(teeFd, &temp, sizeof(temp)); 3374 close(teeFd); 3375 if (fd >= 0) { 3376 dprintf(fd, "tee copied to %s\n", teePath); 3377 } 3378 } else { 3379 if (fd >= 0) { 3380 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3381 } 3382 } 3383 } 3384 } 3385 #endif 3386 3387 // ---------------------------------------------------------------------------- 3388 3389 status_t AudioFlinger::onTransact( 3390 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3391 { 3392 return BnAudioFlinger::onTransact(code, data, reply, flags); 3393 } 3394 3395 } // namespace android 3396