AudioFlinger.cpp revision fe1e1449cadff4f946c33403aecc73b4b4a11e56
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/bitops.h> 45#include <cutils/properties.h> 46 47#include <system/audio.h> 48 49#include "AudioFlinger.h" 50#include "ServiceUtilities.h" 51 52#include <media/AudioResamplerPublic.h> 53 54#include <system/audio_effects/effect_visualizer.h> 55#include <system/audio_effects/effect_ns.h> 56#include <system/audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <powermanager/PowerManager.h> 61 62#include <media/IMediaLogService.h> 63#include <media/MemoryLeakTrackUtil.h> 64#include <media/nbaio/Pipe.h> 65#include <media/nbaio/PipeReader.h> 66#include <media/AudioParameter.h> 67#include <mediautils/BatteryNotifier.h> 68#include <private/android_filesystem_config.h> 69 70//#define BUFLOG_NDEBUG 0 71#include <BufLog.h> 72 73#include "TypedLogger.h" 74 75// ---------------------------------------------------------------------------- 76 77// Note: the following macro is used for extremely verbose logging message. In 78// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 79// 0; but one side effect of this is to turn all LOGV's as well. Some messages 80// are so verbose that we want to suppress them even when we have ALOG_ASSERT 81// turned on. Do not uncomment the #def below unless you really know what you 82// are doing and want to see all of the extremely verbose messages. 83//#define VERY_VERY_VERBOSE_LOGGING 84#ifdef VERY_VERY_VERBOSE_LOGGING 85#define ALOGVV ALOGV 86#else 87#define ALOGVV(a...) do { } while(0) 88#endif 89 90namespace android { 91 92static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 93static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 94static const char kClientLockedString[] = "Client lock is taken\n"; 95static const char kNoEffectsFactory[] = "Effects Factory is absent\n"; 96 97 98nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 99 100uint32_t AudioFlinger::mScreenState; 101 102#ifdef TEE_SINK 103bool AudioFlinger::mTeeSinkInputEnabled = false; 104bool AudioFlinger::mTeeSinkOutputEnabled = false; 105bool AudioFlinger::mTeeSinkTrackEnabled = false; 106 107size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 108size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 109size_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. 114static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 115 116// ---------------------------------------------------------------------------- 117 118std::string formatToString(audio_format_t format) { 119 std::string result; 120 FormatConverter::toString(format, result); 121 return result; 122} 123 124// ---------------------------------------------------------------------------- 125 126AudioFlinger::AudioFlinger() 127 : BnAudioFlinger(), 128 mPrimaryHardwareDev(NULL), 129 mAudioHwDevs(NULL), 130 mHardwareStatus(AUDIO_HW_IDLE), 131 mMasterVolume(1.0f), 132 mMasterMute(false), 133 // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), 134 mMode(AUDIO_MODE_INVALID), 135 mBtNrecIsOff(false), 136 mIsLowRamDevice(true), 137 mIsDeviceTypeKnown(false), 138 mGlobalEffectEnableTime(0), 139 mSystemReady(false) 140{ 141 // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum 142 for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) { 143 // zero ID has a special meaning, so unavailable 144 mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX; 145 } 146 147 getpid_cached = getpid(); 148 const bool doLog = property_get_bool("ro.test_harness", false); 149 if (doLog) { 150 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 151 MemoryHeapBase::READ_ONLY); 152 } 153 154 // reset battery stats. 155 // if the audio service has crashed, battery stats could be left 156 // in bad state, reset the state upon service start. 157 BatteryNotifier::getInstance().noteResetAudio(); 158 159 mDevicesFactoryHal = DevicesFactoryHalInterface::create(); 160 mEffectsFactoryHal = EffectsFactoryHalInterface::create(); 161 162#ifdef TEE_SINK 163 char value[PROPERTY_VALUE_MAX]; 164 (void) property_get("ro.debuggable", value, "0"); 165 int debuggable = atoi(value); 166 int teeEnabled = 0; 167 if (debuggable) { 168 (void) property_get("af.tee", value, "0"); 169 teeEnabled = atoi(value); 170 } 171 // FIXME symbolic constants here 172 if (teeEnabled & 1) { 173 mTeeSinkInputEnabled = true; 174 } 175 if (teeEnabled & 2) { 176 mTeeSinkOutputEnabled = true; 177 } 178 if (teeEnabled & 4) { 179 mTeeSinkTrackEnabled = true; 180 } 181#endif 182} 183 184void AudioFlinger::onFirstRef() 185{ 186 Mutex::Autolock _l(mLock); 187 188 /* TODO: move all this work into an Init() function */ 189 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 190 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 191 uint32_t int_val; 192 if (1 == sscanf(val_str, "%u", &int_val)) { 193 mStandbyTimeInNsecs = milliseconds(int_val); 194 ALOGI("Using %u mSec as standby time.", int_val); 195 } else { 196 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 197 ALOGI("Using default %u mSec as standby time.", 198 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 199 } 200 } 201 202 mPatchPanel = new PatchPanel(this); 203 204 mMode = AUDIO_MODE_NORMAL; 205} 206 207AudioFlinger::~AudioFlinger() 208{ 209 while (!mRecordThreads.isEmpty()) { 210 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 211 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 212 } 213 while (!mPlaybackThreads.isEmpty()) { 214 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 215 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 216 } 217 218 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 219 // no mHardwareLock needed, as there are no other references to this 220 delete mAudioHwDevs.valueAt(i); 221 } 222 223 // Tell media.log service about any old writers that still need to be unregistered 224 if (mLogMemoryDealer != 0) { 225 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 226 if (binder != 0) { 227 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 228 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 229 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 230 mUnregisteredWriters.pop(); 231 mediaLogService->unregisterWriter(iMemory); 232 } 233 } 234 } 235} 236 237static const char * const audio_interfaces[] = { 238 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 239 AUDIO_HARDWARE_MODULE_ID_A2DP, 240 AUDIO_HARDWARE_MODULE_ID_USB, 241}; 242#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 243 244AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 245 audio_module_handle_t module, 246 audio_devices_t devices) 247{ 248 // if module is 0, the request comes from an old policy manager and we should load 249 // well known modules 250 if (module == 0) { 251 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 252 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 253 loadHwModule_l(audio_interfaces[i]); 254 } 255 // then try to find a module supporting the requested device. 256 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 257 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 258 sp<DeviceHalInterface> dev = audioHwDevice->hwDevice(); 259 uint32_t supportedDevices; 260 if (dev->getSupportedDevices(&supportedDevices) == OK && 261 (supportedDevices & devices) == devices) { 262 return audioHwDevice; 263 } 264 } 265 } else { 266 // check a match for the requested module handle 267 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 268 if (audioHwDevice != NULL) { 269 return audioHwDevice; 270 } 271 } 272 273 return NULL; 274} 275 276void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 277{ 278 const size_t SIZE = 256; 279 char buffer[SIZE]; 280 String8 result; 281 282 result.append("Clients:\n"); 283 for (size_t i = 0; i < mClients.size(); ++i) { 284 sp<Client> client = mClients.valueAt(i).promote(); 285 if (client != 0) { 286 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 287 result.append(buffer); 288 } 289 } 290 291 result.append("Notification Clients:\n"); 292 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 293 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 294 result.append(buffer); 295 } 296 297 result.append("Global session refs:\n"); 298 result.append(" session pid count\n"); 299 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 300 AudioSessionRef *r = mAudioSessionRefs[i]; 301 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 302 result.append(buffer); 303 } 304 write(fd, result.string(), result.size()); 305} 306 307 308void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 309{ 310 const size_t SIZE = 256; 311 char buffer[SIZE]; 312 String8 result; 313 hardware_call_state hardwareStatus = mHardwareStatus; 314 315 snprintf(buffer, SIZE, "Hardware status: %d\n" 316 "Standby Time mSec: %u\n", 317 hardwareStatus, 318 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 319 result.append(buffer); 320 write(fd, result.string(), result.size()); 321} 322 323void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 324{ 325 const size_t SIZE = 256; 326 char buffer[SIZE]; 327 String8 result; 328 snprintf(buffer, SIZE, "Permission Denial: " 329 "can't dump AudioFlinger from pid=%d, uid=%d\n", 330 IPCThreadState::self()->getCallingPid(), 331 IPCThreadState::self()->getCallingUid()); 332 result.append(buffer); 333 write(fd, result.string(), result.size()); 334} 335 336bool AudioFlinger::dumpTryLock(Mutex& mutex) 337{ 338 bool locked = false; 339 for (int i = 0; i < kDumpLockRetries; ++i) { 340 if (mutex.tryLock() == NO_ERROR) { 341 locked = true; 342 break; 343 } 344 usleep(kDumpLockSleepUs); 345 } 346 return locked; 347} 348 349status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 350{ 351 if (!dumpAllowed()) { 352 dumpPermissionDenial(fd, args); 353 } else { 354 // get state of hardware lock 355 bool hardwareLocked = dumpTryLock(mHardwareLock); 356 if (!hardwareLocked) { 357 String8 result(kHardwareLockedString); 358 write(fd, result.string(), result.size()); 359 } else { 360 mHardwareLock.unlock(); 361 } 362 363 bool locked = dumpTryLock(mLock); 364 365 // failed to lock - AudioFlinger is probably deadlocked 366 if (!locked) { 367 String8 result(kDeadlockedString); 368 write(fd, result.string(), result.size()); 369 } 370 371 bool clientLocked = dumpTryLock(mClientLock); 372 if (!clientLocked) { 373 String8 result(kClientLockedString); 374 write(fd, result.string(), result.size()); 375 } 376 377 if (mEffectsFactoryHal != 0) { 378 mEffectsFactoryHal->dumpEffects(fd); 379 } else { 380 String8 result(kNoEffectsFactory); 381 write(fd, result.string(), result.size()); 382 } 383 384 dumpClients(fd, args); 385 if (clientLocked) { 386 mClientLock.unlock(); 387 } 388 389 dumpInternals(fd, args); 390 391 // dump playback threads 392 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 393 mPlaybackThreads.valueAt(i)->dump(fd, args); 394 } 395 396 // dump record threads 397 for (size_t i = 0; i < mRecordThreads.size(); i++) { 398 mRecordThreads.valueAt(i)->dump(fd, args); 399 } 400 401 // dump orphan effect chains 402 if (mOrphanEffectChains.size() != 0) { 403 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 404 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 405 mOrphanEffectChains.valueAt(i)->dump(fd, args); 406 } 407 } 408 // dump all hardware devs 409 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 410 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 411 dev->dump(fd); 412 } 413 414#ifdef TEE_SINK 415 // dump the serially shared record tee sink 416 if (mRecordTeeSource != 0) { 417 dumpTee(fd, mRecordTeeSource); 418 } 419#endif 420 421 BUFLOG_RESET; 422 423 if (locked) { 424 mLock.unlock(); 425 } 426 427 // append a copy of media.log here by forwarding fd to it, but don't attempt 428 // to lookup the service if it's not running, as it will block for a second 429 if (mLogMemoryDealer != 0) { 430 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 431 if (binder != 0) { 432 dprintf(fd, "\nmedia.log:\n"); 433 Vector<String16> args; 434 binder->dump(fd, args); 435 } 436 } 437 438 // check for optional arguments 439 bool dumpMem = false; 440 bool unreachableMemory = false; 441 for (const auto &arg : args) { 442 if (arg == String16("-m")) { 443 dumpMem = true; 444 } else if (arg == String16("--unreachable")) { 445 unreachableMemory = true; 446 } 447 } 448 449 if (dumpMem) { 450 dprintf(fd, "\nDumping memory:\n"); 451 std::string s = dumpMemoryAddresses(100 /* limit */); 452 write(fd, s.c_str(), s.size()); 453 } 454 if (unreachableMemory) { 455 dprintf(fd, "\nDumping unreachable memory:\n"); 456 // TODO - should limit be an argument parameter? 457 std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */); 458 write(fd, s.c_str(), s.size()); 459 } 460 } 461 return NO_ERROR; 462} 463 464sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 465{ 466 Mutex::Autolock _cl(mClientLock); 467 // If pid is already in the mClients wp<> map, then use that entry 468 // (for which promote() is always != 0), otherwise create a new entry and Client. 469 sp<Client> client = mClients.valueFor(pid).promote(); 470 if (client == 0) { 471 client = new Client(this, pid); 472 mClients.add(pid, client); 473 } 474 475 return client; 476} 477 478sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 479{ 480 // If there is no memory allocated for logs, return a dummy writer that does nothing 481 if (mLogMemoryDealer == 0) { 482 return new NBLog::Writer(); 483 } 484 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 485 // Similarly if we can't contact the media.log service, also return a dummy writer 486 if (binder == 0) { 487 return new NBLog::Writer(); 488 } 489 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 490 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 491 // If allocation fails, consult the vector of previously unregistered writers 492 // and garbage-collect one or more them until an allocation succeeds 493 if (shared == 0) { 494 Mutex::Autolock _l(mUnregisteredWritersLock); 495 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 496 { 497 // Pick the oldest stale writer to garbage-collect 498 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 499 mUnregisteredWriters.removeAt(0); 500 mediaLogService->unregisterWriter(iMemory); 501 // Now the media.log remote reference to IMemory is gone. When our last local 502 // reference to IMemory also drops to zero at end of this block, 503 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 504 } 505 // Re-attempt the allocation 506 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 507 if (shared != 0) { 508 goto success; 509 } 510 } 511 // Even after garbage-collecting all old writers, there is still not enough memory, 512 // so return a dummy writer 513 return new NBLog::Writer(); 514 } 515success: 516 NBLog::Shared *sharedRawPtr = (NBLog::Shared *) shared->pointer(); 517 new((void *) sharedRawPtr) NBLog::Shared(); // placement new here, but the corresponding 518 // explicit destructor not needed since it is POD 519 mediaLogService->registerWriter(shared, size, name); 520 return new NBLog::Writer(shared, size); 521} 522 523void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 524{ 525 if (writer == 0) { 526 return; 527 } 528 sp<IMemory> iMemory(writer->getIMemory()); 529 if (iMemory == 0) { 530 return; 531 } 532 // Rather than removing the writer immediately, append it to a queue of old writers to 533 // be garbage-collected later. This allows us to continue to view old logs for a while. 534 Mutex::Autolock _l(mUnregisteredWritersLock); 535 mUnregisteredWriters.push(writer); 536} 537 538// IAudioFlinger interface 539 540 541sp<IAudioTrack> AudioFlinger::createTrack( 542 audio_stream_type_t streamType, 543 uint32_t sampleRate, 544 audio_format_t format, 545 audio_channel_mask_t channelMask, 546 size_t *frameCount, 547 audio_output_flags_t *flags, 548 const sp<IMemory>& sharedBuffer, 549 audio_io_handle_t output, 550 pid_t pid, 551 pid_t tid, 552 audio_session_t *sessionId, 553 int clientUid, 554 status_t *status, 555 audio_port_handle_t portId) 556{ 557 sp<PlaybackThread::Track> track; 558 sp<TrackHandle> trackHandle; 559 sp<Client> client; 560 status_t lStatus; 561 audio_session_t lSessionId; 562 563 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 564 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 565 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 566 ALOGW_IF(pid != -1 && pid != callingPid, 567 "%s uid %d pid %d tried to pass itself off as pid %d", 568 __func__, callingUid, callingPid, pid); 569 pid = callingPid; 570 } 571 572 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 573 // but if someone uses binder directly they could bypass that and cause us to crash 574 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 575 ALOGE("createTrack() invalid stream type %d", streamType); 576 lStatus = BAD_VALUE; 577 goto Exit; 578 } 579 580 // further sample rate checks are performed by createTrack_l() depending on the thread type 581 if (sampleRate == 0) { 582 ALOGE("createTrack() invalid sample rate %u", sampleRate); 583 lStatus = BAD_VALUE; 584 goto Exit; 585 } 586 587 // further channel mask checks are performed by createTrack_l() depending on the thread type 588 if (!audio_is_output_channel(channelMask)) { 589 ALOGE("createTrack() invalid channel mask %#x", channelMask); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 // further format checks are performed by createTrack_l() depending on the thread type 595 if (!audio_is_valid_format(format)) { 596 ALOGE("createTrack() invalid format %#x", format); 597 lStatus = BAD_VALUE; 598 goto Exit; 599 } 600 601 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 602 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 603 lStatus = BAD_VALUE; 604 goto Exit; 605 } 606 607 { 608 Mutex::Autolock _l(mLock); 609 PlaybackThread *thread = checkPlaybackThread_l(output); 610 if (thread == NULL) { 611 ALOGE("no playback thread found for output handle %d", output); 612 lStatus = BAD_VALUE; 613 goto Exit; 614 } 615 616 client = registerPid(pid); 617 618 PlaybackThread *effectThread = NULL; 619 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 620 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 621 ALOGE("createTrack() invalid session ID %d", *sessionId); 622 lStatus = BAD_VALUE; 623 goto Exit; 624 } 625 lSessionId = *sessionId; 626 // check if an effect chain with the same session ID is present on another 627 // output thread and move it here. 628 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 629 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 630 if (mPlaybackThreads.keyAt(i) != output) { 631 uint32_t sessions = t->hasAudioSession(lSessionId); 632 if (sessions & ThreadBase::EFFECT_SESSION) { 633 effectThread = t.get(); 634 break; 635 } 636 } 637 } 638 } else { 639 // if no audio session id is provided, create one here 640 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 641 if (sessionId != NULL) { 642 *sessionId = lSessionId; 643 } 644 } 645 ALOGV("createTrack() lSessionId: %d", lSessionId); 646 647 track = thread->createTrack_l(client, streamType, sampleRate, format, 648 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, 649 clientUid, &lStatus, portId); 650 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 651 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 652 653 // move effect chain to this output thread if an effect on same session was waiting 654 // for a track to be created 655 if (lStatus == NO_ERROR && effectThread != NULL) { 656 // no risk of deadlock because AudioFlinger::mLock is held 657 Mutex::Autolock _dl(thread->mLock); 658 Mutex::Autolock _sl(effectThread->mLock); 659 moveEffectChain_l(lSessionId, effectThread, thread, true); 660 } 661 662 // Look for sync events awaiting for a session to be used. 663 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 664 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 665 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 666 if (lStatus == NO_ERROR) { 667 (void) track->setSyncEvent(mPendingSyncEvents[i]); 668 } else { 669 mPendingSyncEvents[i]->cancel(); 670 } 671 mPendingSyncEvents.removeAt(i); 672 i--; 673 } 674 } 675 } 676 677 setAudioHwSyncForSession_l(thread, lSessionId); 678 } 679 680 if (lStatus != NO_ERROR) { 681 // remove local strong reference to Client before deleting the Track so that the 682 // Client destructor is called by the TrackBase destructor with mClientLock held 683 // Don't hold mClientLock when releasing the reference on the track as the 684 // destructor will acquire it. 685 { 686 Mutex::Autolock _cl(mClientLock); 687 client.clear(); 688 } 689 track.clear(); 690 goto Exit; 691 } 692 693 // return handle to client 694 trackHandle = new TrackHandle(track); 695 696Exit: 697 *status = lStatus; 698 return trackHandle; 699} 700 701uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 702{ 703 Mutex::Autolock _l(mLock); 704 ThreadBase *thread = checkThread_l(ioHandle); 705 if (thread == NULL) { 706 ALOGW("sampleRate() unknown thread %d", ioHandle); 707 return 0; 708 } 709 return thread->sampleRate(); 710} 711 712audio_format_t AudioFlinger::format(audio_io_handle_t output) const 713{ 714 Mutex::Autolock _l(mLock); 715 PlaybackThread *thread = checkPlaybackThread_l(output); 716 if (thread == NULL) { 717 ALOGW("format() unknown thread %d", output); 718 return AUDIO_FORMAT_INVALID; 719 } 720 return thread->format(); 721} 722 723size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 724{ 725 Mutex::Autolock _l(mLock); 726 ThreadBase *thread = checkThread_l(ioHandle); 727 if (thread == NULL) { 728 ALOGW("frameCount() unknown thread %d", ioHandle); 729 return 0; 730 } 731 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 732 // should examine all callers and fix them to handle smaller counts 733 return thread->frameCount(); 734} 735 736size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const 737{ 738 Mutex::Autolock _l(mLock); 739 ThreadBase *thread = checkThread_l(ioHandle); 740 if (thread == NULL) { 741 ALOGW("frameCountHAL() unknown thread %d", ioHandle); 742 return 0; 743 } 744 return thread->frameCountHAL(); 745} 746 747uint32_t AudioFlinger::latency(audio_io_handle_t output) const 748{ 749 Mutex::Autolock _l(mLock); 750 PlaybackThread *thread = checkPlaybackThread_l(output); 751 if (thread == NULL) { 752 ALOGW("latency(): no playback thread found for output handle %d", output); 753 return 0; 754 } 755 return thread->latency(); 756} 757 758status_t AudioFlinger::setMasterVolume(float value) 759{ 760 status_t ret = initCheck(); 761 if (ret != NO_ERROR) { 762 return ret; 763 } 764 765 // check calling permissions 766 if (!settingsAllowed()) { 767 return PERMISSION_DENIED; 768 } 769 770 Mutex::Autolock _l(mLock); 771 mMasterVolume = value; 772 773 // Set master volume in the HALs which support it. 774 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 775 AutoMutex lock(mHardwareLock); 776 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 777 778 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 779 if (dev->canSetMasterVolume()) { 780 dev->hwDevice()->setMasterVolume(value); 781 } 782 mHardwareStatus = AUDIO_HW_IDLE; 783 } 784 785 // Now set the master volume in each playback thread. Playback threads 786 // assigned to HALs which do not have master volume support will apply 787 // master volume during the mix operation. Threads with HALs which do 788 // support master volume will simply ignore the setting. 789 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 790 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 791 continue; 792 } 793 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 794 } 795 796 return NO_ERROR; 797} 798 799status_t AudioFlinger::setMode(audio_mode_t mode) 800{ 801 status_t ret = initCheck(); 802 if (ret != NO_ERROR) { 803 return ret; 804 } 805 806 // check calling permissions 807 if (!settingsAllowed()) { 808 return PERMISSION_DENIED; 809 } 810 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 811 ALOGW("Illegal value: setMode(%d)", mode); 812 return BAD_VALUE; 813 } 814 815 { // scope for the lock 816 AutoMutex lock(mHardwareLock); 817 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 818 mHardwareStatus = AUDIO_HW_SET_MODE; 819 ret = dev->setMode(mode); 820 mHardwareStatus = AUDIO_HW_IDLE; 821 } 822 823 if (NO_ERROR == ret) { 824 Mutex::Autolock _l(mLock); 825 mMode = mode; 826 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 827 mPlaybackThreads.valueAt(i)->setMode(mode); 828 } 829 830 return ret; 831} 832 833status_t AudioFlinger::setMicMute(bool state) 834{ 835 status_t ret = initCheck(); 836 if (ret != NO_ERROR) { 837 return ret; 838 } 839 840 // check calling permissions 841 if (!settingsAllowed()) { 842 return PERMISSION_DENIED; 843 } 844 845 AutoMutex lock(mHardwareLock); 846 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 847 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 848 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 849 status_t result = dev->setMicMute(state); 850 if (result != NO_ERROR) { 851 ret = result; 852 } 853 } 854 mHardwareStatus = AUDIO_HW_IDLE; 855 return ret; 856} 857 858bool AudioFlinger::getMicMute() const 859{ 860 status_t ret = initCheck(); 861 if (ret != NO_ERROR) { 862 return false; 863 } 864 bool mute = true; 865 bool state = AUDIO_MODE_INVALID; 866 AutoMutex lock(mHardwareLock); 867 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 868 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 869 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 870 status_t result = dev->getMicMute(&state); 871 if (result == NO_ERROR) { 872 mute = mute && state; 873 } 874 } 875 mHardwareStatus = AUDIO_HW_IDLE; 876 877 return mute; 878} 879 880status_t AudioFlinger::setMasterMute(bool muted) 881{ 882 status_t ret = initCheck(); 883 if (ret != NO_ERROR) { 884 return ret; 885 } 886 887 // check calling permissions 888 if (!settingsAllowed()) { 889 return PERMISSION_DENIED; 890 } 891 892 Mutex::Autolock _l(mLock); 893 mMasterMute = muted; 894 895 // Set master mute in the HALs which support it. 896 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 897 AutoMutex lock(mHardwareLock); 898 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 899 900 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 901 if (dev->canSetMasterMute()) { 902 dev->hwDevice()->setMasterMute(muted); 903 } 904 mHardwareStatus = AUDIO_HW_IDLE; 905 } 906 907 // Now set the master mute in each playback thread. Playback threads 908 // assigned to HALs which do not have master mute support will apply master 909 // mute during the mix operation. Threads with HALs which do support master 910 // mute will simply ignore the setting. 911 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 912 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 913 continue; 914 } 915 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 916 } 917 918 return NO_ERROR; 919} 920 921float AudioFlinger::masterVolume() const 922{ 923 Mutex::Autolock _l(mLock); 924 return masterVolume_l(); 925} 926 927bool AudioFlinger::masterMute() const 928{ 929 Mutex::Autolock _l(mLock); 930 return masterMute_l(); 931} 932 933float AudioFlinger::masterVolume_l() const 934{ 935 return mMasterVolume; 936} 937 938bool AudioFlinger::masterMute_l() const 939{ 940 return mMasterMute; 941} 942 943status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 944{ 945 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 946 ALOGW("setStreamVolume() invalid stream %d", stream); 947 return BAD_VALUE; 948 } 949 pid_t caller = IPCThreadState::self()->getCallingPid(); 950 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 951 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 952 return PERMISSION_DENIED; 953 } 954 955 return NO_ERROR; 956} 957 958status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 959 audio_io_handle_t output) 960{ 961 // check calling permissions 962 if (!settingsAllowed()) { 963 return PERMISSION_DENIED; 964 } 965 966 status_t status = checkStreamType(stream); 967 if (status != NO_ERROR) { 968 return status; 969 } 970 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 971 972 AutoMutex lock(mLock); 973 PlaybackThread *thread = NULL; 974 if (output != AUDIO_IO_HANDLE_NONE) { 975 thread = checkPlaybackThread_l(output); 976 if (thread == NULL) { 977 return BAD_VALUE; 978 } 979 } 980 981 mStreamTypes[stream].volume = value; 982 983 if (thread == NULL) { 984 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 985 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 986 } 987 } else { 988 thread->setStreamVolume(stream, value); 989 } 990 991 return NO_ERROR; 992} 993 994status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 995{ 996 // check calling permissions 997 if (!settingsAllowed()) { 998 return PERMISSION_DENIED; 999 } 1000 1001 status_t status = checkStreamType(stream); 1002 if (status != NO_ERROR) { 1003 return status; 1004 } 1005 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1006 1007 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1008 ALOGE("setStreamMute() invalid stream %d", stream); 1009 return BAD_VALUE; 1010 } 1011 1012 AutoMutex lock(mLock); 1013 mStreamTypes[stream].mute = muted; 1014 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 1015 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 1016 1017 return NO_ERROR; 1018} 1019 1020float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1021{ 1022 status_t status = checkStreamType(stream); 1023 if (status != NO_ERROR) { 1024 return 0.0f; 1025 } 1026 1027 AutoMutex lock(mLock); 1028 float volume; 1029 if (output != AUDIO_IO_HANDLE_NONE) { 1030 PlaybackThread *thread = checkPlaybackThread_l(output); 1031 if (thread == NULL) { 1032 return 0.0f; 1033 } 1034 volume = thread->streamVolume(stream); 1035 } else { 1036 volume = streamVolume_l(stream); 1037 } 1038 1039 return volume; 1040} 1041 1042bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1043{ 1044 status_t status = checkStreamType(stream); 1045 if (status != NO_ERROR) { 1046 return true; 1047 } 1048 1049 AutoMutex lock(mLock); 1050 return streamMute_l(stream); 1051} 1052 1053 1054void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1055{ 1056 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1057 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1058 } 1059} 1060 1061status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1062{ 1063 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1064 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1065 1066 // check calling permissions 1067 if (!settingsAllowed()) { 1068 return PERMISSION_DENIED; 1069 } 1070 1071 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1072 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1073 Mutex::Autolock _l(mLock); 1074 // result will remain NO_INIT if no audio device is present 1075 status_t final_result = NO_INIT; 1076 { 1077 AutoMutex lock(mHardwareLock); 1078 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1079 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1080 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1081 status_t result = dev->setParameters(keyValuePairs); 1082 // return success if at least one audio device accepts the parameters as not all 1083 // HALs are requested to support all parameters. If no audio device supports the 1084 // requested parameters, the last error is reported. 1085 if (final_result != NO_ERROR) { 1086 final_result = result; 1087 } 1088 } 1089 mHardwareStatus = AUDIO_HW_IDLE; 1090 } 1091 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1092 AudioParameter param = AudioParameter(keyValuePairs); 1093 String8 value; 1094 if (param.get(String8(AudioParameter::keyBtNrec), value) == NO_ERROR) { 1095 bool btNrecIsOff = (value == AudioParameter::valueOff); 1096 if (mBtNrecIsOff != btNrecIsOff) { 1097 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1098 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1099 audio_devices_t device = thread->inDevice(); 1100 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1101 // collect all of the thread's session IDs 1102 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1103 // suspend effects associated with those session IDs 1104 for (size_t j = 0; j < ids.size(); ++j) { 1105 audio_session_t sessionId = ids.keyAt(j); 1106 thread->setEffectSuspended(FX_IID_AEC, 1107 suspend, 1108 sessionId); 1109 thread->setEffectSuspended(FX_IID_NS, 1110 suspend, 1111 sessionId); 1112 } 1113 } 1114 mBtNrecIsOff = btNrecIsOff; 1115 } 1116 } 1117 String8 screenState; 1118 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1119 bool isOff = (screenState == AudioParameter::valueOff); 1120 if (isOff != (AudioFlinger::mScreenState & 1)) { 1121 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1122 } 1123 } 1124 return final_result; 1125 } 1126 1127 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1128 // and the thread is exited once the lock is released 1129 sp<ThreadBase> thread; 1130 { 1131 Mutex::Autolock _l(mLock); 1132 thread = checkPlaybackThread_l(ioHandle); 1133 if (thread == 0) { 1134 thread = checkRecordThread_l(ioHandle); 1135 } else if (thread == primaryPlaybackThread_l()) { 1136 // indicate output device change to all input threads for pre processing 1137 AudioParameter param = AudioParameter(keyValuePairs); 1138 int value; 1139 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1140 (value != 0)) { 1141 broacastParametersToRecordThreads_l(keyValuePairs); 1142 } 1143 } 1144 } 1145 if (thread != 0) { 1146 return thread->setParameters(keyValuePairs); 1147 } 1148 return BAD_VALUE; 1149} 1150 1151String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1152{ 1153 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1154 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1155 1156 Mutex::Autolock _l(mLock); 1157 1158 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1159 String8 out_s8; 1160 1161 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1162 String8 s; 1163 status_t result; 1164 { 1165 AutoMutex lock(mHardwareLock); 1166 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1167 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1168 result = dev->getParameters(keys, &s); 1169 mHardwareStatus = AUDIO_HW_IDLE; 1170 } 1171 if (result == OK) out_s8 += s; 1172 } 1173 return out_s8; 1174 } 1175 1176 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1177 if (playbackThread != NULL) { 1178 return playbackThread->getParameters(keys); 1179 } 1180 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1181 if (recordThread != NULL) { 1182 return recordThread->getParameters(keys); 1183 } 1184 return String8(""); 1185} 1186 1187size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1188 audio_channel_mask_t channelMask) const 1189{ 1190 status_t ret = initCheck(); 1191 if (ret != NO_ERROR) { 1192 return 0; 1193 } 1194 if ((sampleRate == 0) || 1195 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1196 !audio_is_input_channel(channelMask)) { 1197 return 0; 1198 } 1199 1200 AutoMutex lock(mHardwareLock); 1201 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1202 audio_config_t config, proposed; 1203 memset(&proposed, 0, sizeof(proposed)); 1204 proposed.sample_rate = sampleRate; 1205 proposed.channel_mask = channelMask; 1206 proposed.format = format; 1207 1208 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1209 size_t frames; 1210 for (;;) { 1211 // Note: config is currently a const parameter for get_input_buffer_size() 1212 // but we use a copy from proposed in case config changes from the call. 1213 config = proposed; 1214 status_t result = dev->getInputBufferSize(&config, &frames); 1215 if (result == OK && frames != 0) { 1216 break; // hal success, config is the result 1217 } 1218 // change one parameter of the configuration each iteration to a more "common" value 1219 // to see if the device will support it. 1220 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1221 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1222 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1223 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1224 } else { 1225 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1226 "format %#x, channelMask 0x%X", 1227 sampleRate, format, channelMask); 1228 break; // retries failed, break out of loop with frames == 0. 1229 } 1230 } 1231 mHardwareStatus = AUDIO_HW_IDLE; 1232 if (frames > 0 && config.sample_rate != sampleRate) { 1233 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1234 } 1235 return frames; // may be converted to bytes at the Java level. 1236} 1237 1238uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1239{ 1240 Mutex::Autolock _l(mLock); 1241 1242 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1243 if (recordThread != NULL) { 1244 return recordThread->getInputFramesLost(); 1245 } 1246 return 0; 1247} 1248 1249status_t AudioFlinger::setVoiceVolume(float value) 1250{ 1251 status_t ret = initCheck(); 1252 if (ret != NO_ERROR) { 1253 return ret; 1254 } 1255 1256 // check calling permissions 1257 if (!settingsAllowed()) { 1258 return PERMISSION_DENIED; 1259 } 1260 1261 AutoMutex lock(mHardwareLock); 1262 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1263 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1264 ret = dev->setVoiceVolume(value); 1265 mHardwareStatus = AUDIO_HW_IDLE; 1266 1267 return ret; 1268} 1269 1270status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1271 audio_io_handle_t output) const 1272{ 1273 Mutex::Autolock _l(mLock); 1274 1275 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1276 if (playbackThread != NULL) { 1277 return playbackThread->getRenderPosition(halFrames, dspFrames); 1278 } 1279 1280 return BAD_VALUE; 1281} 1282 1283void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1284{ 1285 Mutex::Autolock _l(mLock); 1286 if (client == 0) { 1287 return; 1288 } 1289 pid_t pid = IPCThreadState::self()->getCallingPid(); 1290 { 1291 Mutex::Autolock _cl(mClientLock); 1292 if (mNotificationClients.indexOfKey(pid) < 0) { 1293 sp<NotificationClient> notificationClient = new NotificationClient(this, 1294 client, 1295 pid); 1296 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1297 1298 mNotificationClients.add(pid, notificationClient); 1299 1300 sp<IBinder> binder = IInterface::asBinder(client); 1301 binder->linkToDeath(notificationClient); 1302 } 1303 } 1304 1305 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1306 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1307 // the config change is always sent from playback or record threads to avoid deadlock 1308 // with AudioSystem::gLock 1309 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1310 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1311 } 1312 1313 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1314 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1315 } 1316} 1317 1318void AudioFlinger::removeNotificationClient(pid_t pid) 1319{ 1320 Mutex::Autolock _l(mLock); 1321 { 1322 Mutex::Autolock _cl(mClientLock); 1323 mNotificationClients.removeItem(pid); 1324 } 1325 1326 ALOGV("%d died, releasing its sessions", pid); 1327 size_t num = mAudioSessionRefs.size(); 1328 bool removed = false; 1329 for (size_t i = 0; i < num; ) { 1330 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1331 ALOGV(" pid %d @ %zu", ref->mPid, i); 1332 if (ref->mPid == pid) { 1333 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1334 mAudioSessionRefs.removeAt(i); 1335 delete ref; 1336 removed = true; 1337 num--; 1338 } else { 1339 i++; 1340 } 1341 } 1342 if (removed) { 1343 purgeStaleEffects_l(); 1344 } 1345} 1346 1347void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1348 const sp<AudioIoDescriptor>& ioDesc, 1349 pid_t pid) 1350{ 1351 Mutex::Autolock _l(mClientLock); 1352 size_t size = mNotificationClients.size(); 1353 for (size_t i = 0; i < size; i++) { 1354 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1355 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1356 } 1357 } 1358} 1359 1360// removeClient_l() must be called with AudioFlinger::mClientLock held 1361void AudioFlinger::removeClient_l(pid_t pid) 1362{ 1363 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1364 IPCThreadState::self()->getCallingPid()); 1365 mClients.removeItem(pid); 1366} 1367 1368// getEffectThread_l() must be called with AudioFlinger::mLock held 1369sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1370 int EffectId) 1371{ 1372 sp<PlaybackThread> thread; 1373 1374 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1375 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1376 ALOG_ASSERT(thread == 0); 1377 thread = mPlaybackThreads.valueAt(i); 1378 } 1379 } 1380 1381 return thread; 1382} 1383 1384 1385 1386// ---------------------------------------------------------------------------- 1387 1388AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1389 : RefBase(), 1390 mAudioFlinger(audioFlinger), 1391 mPid(pid) 1392{ 1393 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1394 heapSize *= 1024; 1395 if (!heapSize) { 1396 heapSize = kClientSharedHeapSizeBytes; 1397 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1398 // invalidated tracks 1399 if (!audioFlinger->isLowRamDevice()) { 1400 heapSize *= kClientSharedHeapSizeMultiplier; 1401 } 1402 } 1403 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1404} 1405 1406// Client destructor must be called with AudioFlinger::mClientLock held 1407AudioFlinger::Client::~Client() 1408{ 1409 mAudioFlinger->removeClient_l(mPid); 1410} 1411 1412sp<MemoryDealer> AudioFlinger::Client::heap() const 1413{ 1414 return mMemoryDealer; 1415} 1416 1417// ---------------------------------------------------------------------------- 1418 1419AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1420 const sp<IAudioFlingerClient>& client, 1421 pid_t pid) 1422 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1423{ 1424} 1425 1426AudioFlinger::NotificationClient::~NotificationClient() 1427{ 1428} 1429 1430void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1431{ 1432 sp<NotificationClient> keep(this); 1433 mAudioFlinger->removeNotificationClient(mPid); 1434} 1435 1436 1437// ---------------------------------------------------------------------------- 1438 1439sp<IAudioRecord> AudioFlinger::openRecord( 1440 audio_io_handle_t input, 1441 uint32_t sampleRate, 1442 audio_format_t format, 1443 audio_channel_mask_t channelMask, 1444 const String16& opPackageName, 1445 size_t *frameCount, 1446 audio_input_flags_t *flags, 1447 pid_t pid, 1448 pid_t tid, 1449 int clientUid, 1450 audio_session_t *sessionId, 1451 size_t *notificationFrames, 1452 sp<IMemory>& cblk, 1453 sp<IMemory>& buffers, 1454 status_t *status, 1455 audio_port_handle_t portId) 1456{ 1457 sp<RecordThread::RecordTrack> recordTrack; 1458 sp<RecordHandle> recordHandle; 1459 sp<Client> client; 1460 status_t lStatus; 1461 audio_session_t lSessionId; 1462 1463 cblk.clear(); 1464 buffers.clear(); 1465 1466 bool updatePid = (pid == -1); 1467 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1468 if (!isTrustedCallingUid(callingUid)) { 1469 ALOGW_IF((uid_t)clientUid != callingUid, 1470 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1471 clientUid = callingUid; 1472 updatePid = true; 1473 } 1474 1475 if (updatePid) { 1476 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1477 ALOGW_IF(pid != -1 && pid != callingPid, 1478 "%s uid %d pid %d tried to pass itself off as pid %d", 1479 __func__, callingUid, callingPid, pid); 1480 pid = callingPid; 1481 } 1482 1483 // check calling permissions 1484 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1485 ALOGE("openRecord() permission denied: recording not allowed"); 1486 lStatus = PERMISSION_DENIED; 1487 goto Exit; 1488 } 1489 1490 // further sample rate checks are performed by createRecordTrack_l() 1491 if (sampleRate == 0) { 1492 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1493 lStatus = BAD_VALUE; 1494 goto Exit; 1495 } 1496 1497 // we don't yet support anything other than linear PCM 1498 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1499 ALOGE("openRecord() invalid format %#x", format); 1500 lStatus = BAD_VALUE; 1501 goto Exit; 1502 } 1503 1504 // further channel mask checks are performed by createRecordTrack_l() 1505 if (!audio_is_input_channel(channelMask)) { 1506 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1507 lStatus = BAD_VALUE; 1508 goto Exit; 1509 } 1510 1511 { 1512 Mutex::Autolock _l(mLock); 1513 RecordThread *thread = checkRecordThread_l(input); 1514 if (thread == NULL) { 1515 ALOGE("openRecord() checkRecordThread_l failed"); 1516 lStatus = BAD_VALUE; 1517 goto Exit; 1518 } 1519 1520 client = registerPid(pid); 1521 1522 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1523 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1524 lStatus = BAD_VALUE; 1525 goto Exit; 1526 } 1527 lSessionId = *sessionId; 1528 } else { 1529 // if no audio session id is provided, create one here 1530 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1531 if (sessionId != NULL) { 1532 *sessionId = lSessionId; 1533 } 1534 } 1535 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1536 1537 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1538 frameCount, lSessionId, notificationFrames, 1539 clientUid, flags, tid, &lStatus, portId); 1540 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1541 1542 if (lStatus == NO_ERROR) { 1543 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1544 // session and move it to this thread. 1545 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1546 if (chain != 0) { 1547 Mutex::Autolock _l(thread->mLock); 1548 thread->addEffectChain_l(chain); 1549 } 1550 } 1551 } 1552 1553 if (lStatus != NO_ERROR) { 1554 // remove local strong reference to Client before deleting the RecordTrack so that the 1555 // Client destructor is called by the TrackBase destructor with mClientLock held 1556 // Don't hold mClientLock when releasing the reference on the track as the 1557 // destructor will acquire it. 1558 { 1559 Mutex::Autolock _cl(mClientLock); 1560 client.clear(); 1561 } 1562 recordTrack.clear(); 1563 goto Exit; 1564 } 1565 1566 cblk = recordTrack->getCblk(); 1567 buffers = recordTrack->getBuffers(); 1568 1569 // return handle to client 1570 recordHandle = new RecordHandle(recordTrack); 1571 1572Exit: 1573 *status = lStatus; 1574 return recordHandle; 1575} 1576 1577 1578 1579// ---------------------------------------------------------------------------- 1580 1581audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1582{ 1583 if (name == NULL) { 1584 return AUDIO_MODULE_HANDLE_NONE; 1585 } 1586 if (!settingsAllowed()) { 1587 return AUDIO_MODULE_HANDLE_NONE; 1588 } 1589 Mutex::Autolock _l(mLock); 1590 return loadHwModule_l(name); 1591} 1592 1593// loadHwModule_l() must be called with AudioFlinger::mLock held 1594audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1595{ 1596 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1597 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1598 ALOGW("loadHwModule() module %s already loaded", name); 1599 return mAudioHwDevs.keyAt(i); 1600 } 1601 } 1602 1603 sp<DeviceHalInterface> dev; 1604 1605 int rc = mDevicesFactoryHal->openDevice(name, &dev); 1606 if (rc) { 1607 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1608 return AUDIO_MODULE_HANDLE_NONE; 1609 } 1610 1611 mHardwareStatus = AUDIO_HW_INIT; 1612 rc = dev->initCheck(); 1613 mHardwareStatus = AUDIO_HW_IDLE; 1614 if (rc) { 1615 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1616 return AUDIO_MODULE_HANDLE_NONE; 1617 } 1618 1619 // Check and cache this HAL's level of support for master mute and master 1620 // volume. If this is the first HAL opened, and it supports the get 1621 // methods, use the initial values provided by the HAL as the current 1622 // master mute and volume settings. 1623 1624 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1625 { // scope for auto-lock pattern 1626 AutoMutex lock(mHardwareLock); 1627 1628 if (0 == mAudioHwDevs.size()) { 1629 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1630 float mv; 1631 if (OK == dev->getMasterVolume(&mv)) { 1632 mMasterVolume = mv; 1633 } 1634 1635 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1636 bool mm; 1637 if (OK == dev->getMasterMute(&mm)) { 1638 mMasterMute = mm; 1639 } 1640 } 1641 1642 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1643 if (OK == dev->setMasterVolume(mMasterVolume)) { 1644 flags = static_cast<AudioHwDevice::Flags>(flags | 1645 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1646 } 1647 1648 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1649 if (OK == dev->setMasterMute(mMasterMute)) { 1650 flags = static_cast<AudioHwDevice::Flags>(flags | 1651 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1652 } 1653 1654 mHardwareStatus = AUDIO_HW_IDLE; 1655 } 1656 1657 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1658 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1659 1660 ALOGI("loadHwModule() Loaded %s audio interface, handle %d", name, handle); 1661 1662 return handle; 1663 1664} 1665 1666// ---------------------------------------------------------------------------- 1667 1668uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1669{ 1670 Mutex::Autolock _l(mLock); 1671 PlaybackThread *thread = fastPlaybackThread_l(); 1672 return thread != NULL ? thread->sampleRate() : 0; 1673} 1674 1675size_t AudioFlinger::getPrimaryOutputFrameCount() 1676{ 1677 Mutex::Autolock _l(mLock); 1678 PlaybackThread *thread = fastPlaybackThread_l(); 1679 return thread != NULL ? thread->frameCountHAL() : 0; 1680} 1681 1682// ---------------------------------------------------------------------------- 1683 1684status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1685{ 1686 uid_t uid = IPCThreadState::self()->getCallingUid(); 1687 if (uid != AID_SYSTEM) { 1688 return PERMISSION_DENIED; 1689 } 1690 Mutex::Autolock _l(mLock); 1691 if (mIsDeviceTypeKnown) { 1692 return INVALID_OPERATION; 1693 } 1694 mIsLowRamDevice = isLowRamDevice; 1695 mIsDeviceTypeKnown = true; 1696 return NO_ERROR; 1697} 1698 1699audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1700{ 1701 Mutex::Autolock _l(mLock); 1702 1703 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1704 if (index >= 0) { 1705 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1706 mHwAvSyncIds.valueAt(index), sessionId); 1707 return mHwAvSyncIds.valueAt(index); 1708 } 1709 1710 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1711 if (dev == NULL) { 1712 return AUDIO_HW_SYNC_INVALID; 1713 } 1714 String8 reply; 1715 AudioParameter param; 1716 if (dev->getParameters(String8(AudioParameter::keyHwAvSync), &reply) == OK) { 1717 param = AudioParameter(reply); 1718 } 1719 1720 int value; 1721 if (param.getInt(String8(AudioParameter::keyHwAvSync), value) != NO_ERROR) { 1722 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1723 return AUDIO_HW_SYNC_INVALID; 1724 } 1725 1726 // allow only one session for a given HW A/V sync ID. 1727 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1728 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1729 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1730 value, mHwAvSyncIds.keyAt(i)); 1731 mHwAvSyncIds.removeItemsAt(i); 1732 break; 1733 } 1734 } 1735 1736 mHwAvSyncIds.add(sessionId, value); 1737 1738 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1739 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1740 uint32_t sessions = thread->hasAudioSession(sessionId); 1741 if (sessions & ThreadBase::TRACK_SESSION) { 1742 AudioParameter param = AudioParameter(); 1743 param.addInt(String8(AudioParameter::keyStreamHwAvSync), value); 1744 thread->setParameters(param.toString()); 1745 break; 1746 } 1747 } 1748 1749 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1750 return (audio_hw_sync_t)value; 1751} 1752 1753status_t AudioFlinger::systemReady() 1754{ 1755 Mutex::Autolock _l(mLock); 1756 ALOGI("%s", __FUNCTION__); 1757 if (mSystemReady) { 1758 ALOGW("%s called twice", __FUNCTION__); 1759 return NO_ERROR; 1760 } 1761 mSystemReady = true; 1762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1763 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1764 thread->systemReady(); 1765 } 1766 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1767 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1768 thread->systemReady(); 1769 } 1770 return NO_ERROR; 1771} 1772 1773// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1774void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1775{ 1776 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1777 if (index >= 0) { 1778 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1779 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1780 AudioParameter param = AudioParameter(); 1781 param.addInt(String8(AudioParameter::keyStreamHwAvSync), syncId); 1782 thread->setParameters(param.toString()); 1783 } 1784} 1785 1786 1787// ---------------------------------------------------------------------------- 1788 1789 1790sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1791 audio_io_handle_t *output, 1792 audio_config_t *config, 1793 audio_devices_t devices, 1794 const String8& address, 1795 audio_output_flags_t flags) 1796{ 1797 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1798 if (outHwDev == NULL) { 1799 return 0; 1800 } 1801 1802 if (*output == AUDIO_IO_HANDLE_NONE) { 1803 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1804 } else { 1805 // Audio Policy does not currently request a specific output handle. 1806 // If this is ever needed, see openInput_l() for example code. 1807 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1808 return 0; 1809 } 1810 1811 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1812 1813 // FOR TESTING ONLY: 1814 // This if statement allows overriding the audio policy settings 1815 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1816 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1817 // Check only for Normal Mixing mode 1818 if (kEnableExtendedPrecision) { 1819 // Specify format (uncomment one below to choose) 1820 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1821 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1822 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1823 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1824 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1825 } 1826 if (kEnableExtendedChannels) { 1827 // Specify channel mask (uncomment one below to choose) 1828 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1829 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1830 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1831 } 1832 } 1833 1834 AudioStreamOut *outputStream = NULL; 1835 status_t status = outHwDev->openOutputStream( 1836 &outputStream, 1837 *output, 1838 devices, 1839 flags, 1840 config, 1841 address.string()); 1842 1843 mHardwareStatus = AUDIO_HW_IDLE; 1844 1845 if (status == NO_ERROR) { 1846 1847 PlaybackThread *thread; 1848 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1849 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1850 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1851 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1852 || !isValidPcmSinkFormat(config->format) 1853 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1854 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1855 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1856 } else { 1857 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1858 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1859 } 1860 mPlaybackThreads.add(*output, thread); 1861 return thread; 1862 } 1863 1864 return 0; 1865} 1866 1867status_t AudioFlinger::openOutput(audio_module_handle_t module, 1868 audio_io_handle_t *output, 1869 audio_config_t *config, 1870 audio_devices_t *devices, 1871 const String8& address, 1872 uint32_t *latencyMs, 1873 audio_output_flags_t flags) 1874{ 1875 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1876 module, 1877 (devices != NULL) ? *devices : 0, 1878 config->sample_rate, 1879 config->format, 1880 config->channel_mask, 1881 flags); 1882 1883 if (*devices == AUDIO_DEVICE_NONE) { 1884 return BAD_VALUE; 1885 } 1886 1887 Mutex::Autolock _l(mLock); 1888 1889 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1890 if (thread != 0) { 1891 *latencyMs = thread->latency(); 1892 1893 // notify client processes of the new output creation 1894 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1895 1896 // the first primary output opened designates the primary hw device 1897 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1898 ALOGI("Using module %d has the primary audio interface", module); 1899 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1900 1901 AutoMutex lock(mHardwareLock); 1902 mHardwareStatus = AUDIO_HW_SET_MODE; 1903 mPrimaryHardwareDev->hwDevice()->setMode(mMode); 1904 mHardwareStatus = AUDIO_HW_IDLE; 1905 } 1906 return NO_ERROR; 1907 } 1908 1909 return NO_INIT; 1910} 1911 1912audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1913 audio_io_handle_t output2) 1914{ 1915 Mutex::Autolock _l(mLock); 1916 MixerThread *thread1 = checkMixerThread_l(output1); 1917 MixerThread *thread2 = checkMixerThread_l(output2); 1918 1919 if (thread1 == NULL || thread2 == NULL) { 1920 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1921 output2); 1922 return AUDIO_IO_HANDLE_NONE; 1923 } 1924 1925 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1926 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1927 thread->addOutputTrack(thread2); 1928 mPlaybackThreads.add(id, thread); 1929 // notify client processes of the new output creation 1930 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1931 return id; 1932} 1933 1934status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1935{ 1936 return closeOutput_nonvirtual(output); 1937} 1938 1939status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1940{ 1941 // keep strong reference on the playback thread so that 1942 // it is not destroyed while exit() is executed 1943 sp<PlaybackThread> thread; 1944 { 1945 Mutex::Autolock _l(mLock); 1946 thread = checkPlaybackThread_l(output); 1947 if (thread == NULL) { 1948 return BAD_VALUE; 1949 } 1950 1951 ALOGV("closeOutput() %d", output); 1952 1953 if (thread->type() == ThreadBase::MIXER) { 1954 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1955 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1956 DuplicatingThread *dupThread = 1957 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1958 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1959 } 1960 } 1961 } 1962 1963 1964 mPlaybackThreads.removeItem(output); 1965 // save all effects to the default thread 1966 if (mPlaybackThreads.size()) { 1967 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1968 if (dstThread != NULL) { 1969 // audioflinger lock is held here so the acquisition order of thread locks does not 1970 // matter 1971 Mutex::Autolock _dl(dstThread->mLock); 1972 Mutex::Autolock _sl(thread->mLock); 1973 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1974 for (size_t i = 0; i < effectChains.size(); i ++) { 1975 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1976 } 1977 } 1978 } 1979 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1980 ioDesc->mIoHandle = output; 1981 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1982 } 1983 thread->exit(); 1984 // The thread entity (active unit of execution) is no longer running here, 1985 // but the ThreadBase container still exists. 1986 1987 if (!thread->isDuplicating()) { 1988 closeOutputFinish(thread); 1989 } 1990 1991 return NO_ERROR; 1992} 1993 1994void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread) 1995{ 1996 AudioStreamOut *out = thread->clearOutput(); 1997 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1998 // from now on thread->mOutput is NULL 1999 delete out; 2000} 2001 2002void AudioFlinger::closeOutputInternal_l(const sp<PlaybackThread>& thread) 2003{ 2004 mPlaybackThreads.removeItem(thread->mId); 2005 thread->exit(); 2006 closeOutputFinish(thread); 2007} 2008 2009status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2010{ 2011 Mutex::Autolock _l(mLock); 2012 PlaybackThread *thread = checkPlaybackThread_l(output); 2013 2014 if (thread == NULL) { 2015 return BAD_VALUE; 2016 } 2017 2018 ALOGV("suspendOutput() %d", output); 2019 thread->suspend(); 2020 2021 return NO_ERROR; 2022} 2023 2024status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2025{ 2026 Mutex::Autolock _l(mLock); 2027 PlaybackThread *thread = checkPlaybackThread_l(output); 2028 2029 if (thread == NULL) { 2030 return BAD_VALUE; 2031 } 2032 2033 ALOGV("restoreOutput() %d", output); 2034 2035 thread->restore(); 2036 2037 return NO_ERROR; 2038} 2039 2040status_t AudioFlinger::openInput(audio_module_handle_t module, 2041 audio_io_handle_t *input, 2042 audio_config_t *config, 2043 audio_devices_t *devices, 2044 const String8& address, 2045 audio_source_t source, 2046 audio_input_flags_t flags) 2047{ 2048 Mutex::Autolock _l(mLock); 2049 2050 if (*devices == AUDIO_DEVICE_NONE) { 2051 return BAD_VALUE; 2052 } 2053 2054 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2055 2056 if (thread != 0) { 2057 // notify client processes of the new input creation 2058 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2059 return NO_ERROR; 2060 } 2061 return NO_INIT; 2062} 2063 2064sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2065 audio_io_handle_t *input, 2066 audio_config_t *config, 2067 audio_devices_t devices, 2068 const String8& address, 2069 audio_source_t source, 2070 audio_input_flags_t flags) 2071{ 2072 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2073 if (inHwDev == NULL) { 2074 *input = AUDIO_IO_HANDLE_NONE; 2075 return 0; 2076 } 2077 2078 // Audio Policy can request a specific handle for hardware hotword. 2079 // The goal here is not to re-open an already opened input. 2080 // It is to use a pre-assigned I/O handle. 2081 if (*input == AUDIO_IO_HANDLE_NONE) { 2082 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2083 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2084 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2085 return 0; 2086 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2087 // This should not happen in a transient state with current design. 2088 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2089 return 0; 2090 } 2091 2092 audio_config_t halconfig = *config; 2093 sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice(); 2094 sp<StreamInHalInterface> inStream; 2095 status_t status = inHwHal->openInputStream( 2096 *input, devices, &halconfig, flags, address.string(), source, &inStream); 2097 ALOGV("openInput_l() openInputStream returned input %p, devices %x, SamplingRate %d" 2098 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2099 inStream.get(), 2100 devices, 2101 halconfig.sample_rate, 2102 halconfig.format, 2103 halconfig.channel_mask, 2104 flags, 2105 status, address.string()); 2106 2107 // If the input could not be opened with the requested parameters and we can handle the 2108 // conversion internally, try to open again with the proposed parameters. 2109 if (status == BAD_VALUE && 2110 audio_is_linear_pcm(config->format) && 2111 audio_is_linear_pcm(halconfig.format) && 2112 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2113 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2114 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2115 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2116 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2117 inStream.clear(); 2118 status = inHwHal->openInputStream( 2119 *input, devices, &halconfig, flags, address.string(), source, &inStream); 2120 // FIXME log this new status; HAL should not propose any further changes 2121 } 2122 2123 if (status == NO_ERROR && inStream != 0) { 2124 2125#ifdef TEE_SINK 2126 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2127 // or (re-)create if current Pipe is idle and does not match the new format 2128 sp<NBAIO_Sink> teeSink; 2129 enum { 2130 TEE_SINK_NO, // don't copy input 2131 TEE_SINK_NEW, // copy input using a new pipe 2132 TEE_SINK_OLD, // copy input using an existing pipe 2133 } kind; 2134 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2135 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2136 if (!mTeeSinkInputEnabled) { 2137 kind = TEE_SINK_NO; 2138 } else if (!Format_isValid(format)) { 2139 kind = TEE_SINK_NO; 2140 } else if (mRecordTeeSink == 0) { 2141 kind = TEE_SINK_NEW; 2142 } else if (mRecordTeeSink->getStrongCount() != 1) { 2143 kind = TEE_SINK_NO; 2144 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2145 kind = TEE_SINK_OLD; 2146 } else { 2147 kind = TEE_SINK_NEW; 2148 } 2149 switch (kind) { 2150 case TEE_SINK_NEW: { 2151 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2152 size_t numCounterOffers = 0; 2153 const NBAIO_Format offers[1] = {format}; 2154 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2155 ALOG_ASSERT(index == 0); 2156 PipeReader *pipeReader = new PipeReader(*pipe); 2157 numCounterOffers = 0; 2158 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2159 ALOG_ASSERT(index == 0); 2160 mRecordTeeSink = pipe; 2161 mRecordTeeSource = pipeReader; 2162 teeSink = pipe; 2163 } 2164 break; 2165 case TEE_SINK_OLD: 2166 teeSink = mRecordTeeSink; 2167 break; 2168 case TEE_SINK_NO: 2169 default: 2170 break; 2171 } 2172#endif 2173 2174 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags); 2175 2176 // Start record thread 2177 // RecordThread requires both input and output device indication to forward to audio 2178 // pre processing modules 2179 sp<RecordThread> thread = new RecordThread(this, 2180 inputStream, 2181 *input, 2182 primaryOutputDevice_l(), 2183 devices, 2184 mSystemReady 2185#ifdef TEE_SINK 2186 , teeSink 2187#endif 2188 ); 2189 mRecordThreads.add(*input, thread); 2190 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2191 return thread; 2192 } 2193 2194 *input = AUDIO_IO_HANDLE_NONE; 2195 return 0; 2196} 2197 2198status_t AudioFlinger::closeInput(audio_io_handle_t input) 2199{ 2200 return closeInput_nonvirtual(input); 2201} 2202 2203status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2204{ 2205 // keep strong reference on the record thread so that 2206 // it is not destroyed while exit() is executed 2207 sp<RecordThread> thread; 2208 { 2209 Mutex::Autolock _l(mLock); 2210 thread = checkRecordThread_l(input); 2211 if (thread == 0) { 2212 return BAD_VALUE; 2213 } 2214 2215 ALOGV("closeInput() %d", input); 2216 2217 // If we still have effect chains, it means that a client still holds a handle 2218 // on at least one effect. We must either move the chain to an existing thread with the 2219 // same session ID or put it aside in case a new record thread is opened for a 2220 // new capture on the same session 2221 sp<EffectChain> chain; 2222 { 2223 Mutex::Autolock _sl(thread->mLock); 2224 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2225 // Note: maximum one chain per record thread 2226 if (effectChains.size() != 0) { 2227 chain = effectChains[0]; 2228 } 2229 } 2230 if (chain != 0) { 2231 // first check if a record thread is already opened with a client on the same session. 2232 // This should only happen in case of overlap between one thread tear down and the 2233 // creation of its replacement 2234 size_t i; 2235 for (i = 0; i < mRecordThreads.size(); i++) { 2236 sp<RecordThread> t = mRecordThreads.valueAt(i); 2237 if (t == thread) { 2238 continue; 2239 } 2240 if (t->hasAudioSession(chain->sessionId()) != 0) { 2241 Mutex::Autolock _l(t->mLock); 2242 ALOGV("closeInput() found thread %d for effect session %d", 2243 t->id(), chain->sessionId()); 2244 t->addEffectChain_l(chain); 2245 break; 2246 } 2247 } 2248 // put the chain aside if we could not find a record thread with the same session id. 2249 if (i == mRecordThreads.size()) { 2250 putOrphanEffectChain_l(chain); 2251 } 2252 } 2253 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2254 ioDesc->mIoHandle = input; 2255 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2256 mRecordThreads.removeItem(input); 2257 } 2258 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2259 // we have a different lock for notification client 2260 closeInputFinish(thread); 2261 return NO_ERROR; 2262} 2263 2264void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread) 2265{ 2266 thread->exit(); 2267 AudioStreamIn *in = thread->clearInput(); 2268 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2269 // from now on thread->mInput is NULL 2270 delete in; 2271} 2272 2273void AudioFlinger::closeInputInternal_l(const sp<RecordThread>& thread) 2274{ 2275 mRecordThreads.removeItem(thread->mId); 2276 closeInputFinish(thread); 2277} 2278 2279status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2280{ 2281 Mutex::Autolock _l(mLock); 2282 ALOGV("invalidateStream() stream %d", stream); 2283 2284 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2285 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2286 thread->invalidateTracks(stream); 2287 } 2288 2289 return NO_ERROR; 2290} 2291 2292 2293audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2294{ 2295 // This is a binder API, so a malicious client could pass in a bad parameter. 2296 // Check for that before calling the internal API nextUniqueId(). 2297 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2298 ALOGE("newAudioUniqueId invalid use %d", use); 2299 return AUDIO_UNIQUE_ID_ALLOCATE; 2300 } 2301 return nextUniqueId(use); 2302} 2303 2304void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2305{ 2306 Mutex::Autolock _l(mLock); 2307 pid_t caller = IPCThreadState::self()->getCallingPid(); 2308 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2309 if (pid != -1 && (caller == getpid_cached)) { 2310 caller = pid; 2311 } 2312 2313 { 2314 Mutex::Autolock _cl(mClientLock); 2315 // Ignore requests received from processes not known as notification client. The request 2316 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2317 // called from a different pid leaving a stale session reference. Also we don't know how 2318 // to clear this reference if the client process dies. 2319 if (mNotificationClients.indexOfKey(caller) < 0) { 2320 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2321 return; 2322 } 2323 } 2324 2325 size_t num = mAudioSessionRefs.size(); 2326 for (size_t i = 0; i < num; i++) { 2327 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2328 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2329 ref->mCnt++; 2330 ALOGV(" incremented refcount to %d", ref->mCnt); 2331 return; 2332 } 2333 } 2334 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2335 ALOGV(" added new entry for %d", audioSession); 2336} 2337 2338void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2339{ 2340 Mutex::Autolock _l(mLock); 2341 pid_t caller = IPCThreadState::self()->getCallingPid(); 2342 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2343 if (pid != -1 && (caller == getpid_cached)) { 2344 caller = pid; 2345 } 2346 size_t num = mAudioSessionRefs.size(); 2347 for (size_t i = 0; i < num; i++) { 2348 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2349 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2350 ref->mCnt--; 2351 ALOGV(" decremented refcount to %d", ref->mCnt); 2352 if (ref->mCnt == 0) { 2353 mAudioSessionRefs.removeAt(i); 2354 delete ref; 2355 purgeStaleEffects_l(); 2356 } 2357 return; 2358 } 2359 } 2360 // If the caller is mediaserver it is likely that the session being released was acquired 2361 // on behalf of a process not in notification clients and we ignore the warning. 2362 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2363} 2364 2365bool AudioFlinger::isSessionAcquired_l(audio_session_t audioSession) 2366{ 2367 size_t num = mAudioSessionRefs.size(); 2368 for (size_t i = 0; i < num; i++) { 2369 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2370 if (ref->mSessionid == audioSession) { 2371 return true; 2372 } 2373 } 2374 return false; 2375} 2376 2377void AudioFlinger::purgeStaleEffects_l() { 2378 2379 ALOGV("purging stale effects"); 2380 2381 Vector< sp<EffectChain> > chains; 2382 2383 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2384 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2385 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2386 sp<EffectChain> ec = t->mEffectChains[j]; 2387 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2388 chains.push(ec); 2389 } 2390 } 2391 } 2392 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2393 sp<RecordThread> t = mRecordThreads.valueAt(i); 2394 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2395 sp<EffectChain> ec = t->mEffectChains[j]; 2396 chains.push(ec); 2397 } 2398 } 2399 2400 for (size_t i = 0; i < chains.size(); i++) { 2401 sp<EffectChain> ec = chains[i]; 2402 int sessionid = ec->sessionId(); 2403 sp<ThreadBase> t = ec->mThread.promote(); 2404 if (t == 0) { 2405 continue; 2406 } 2407 size_t numsessionrefs = mAudioSessionRefs.size(); 2408 bool found = false; 2409 for (size_t k = 0; k < numsessionrefs; k++) { 2410 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2411 if (ref->mSessionid == sessionid) { 2412 ALOGV(" session %d still exists for %d with %d refs", 2413 sessionid, ref->mPid, ref->mCnt); 2414 found = true; 2415 break; 2416 } 2417 } 2418 if (!found) { 2419 Mutex::Autolock _l(t->mLock); 2420 // remove all effects from the chain 2421 while (ec->mEffects.size()) { 2422 sp<EffectModule> effect = ec->mEffects[0]; 2423 effect->unPin(); 2424 t->removeEffect_l(effect); 2425 if (effect->purgeHandles()) { 2426 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2427 } 2428 AudioSystem::unregisterEffect(effect->id()); 2429 } 2430 } 2431 } 2432 return; 2433} 2434 2435// checkThread_l() must be called with AudioFlinger::mLock held 2436AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2437{ 2438 ThreadBase *thread = NULL; 2439 switch (audio_unique_id_get_use(ioHandle)) { 2440 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2441 thread = checkPlaybackThread_l(ioHandle); 2442 break; 2443 case AUDIO_UNIQUE_ID_USE_INPUT: 2444 thread = checkRecordThread_l(ioHandle); 2445 break; 2446 default: 2447 break; 2448 } 2449 return thread; 2450} 2451 2452// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2453AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2454{ 2455 return mPlaybackThreads.valueFor(output).get(); 2456} 2457 2458// checkMixerThread_l() must be called with AudioFlinger::mLock held 2459AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2460{ 2461 PlaybackThread *thread = checkPlaybackThread_l(output); 2462 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2463} 2464 2465// checkRecordThread_l() must be called with AudioFlinger::mLock held 2466AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2467{ 2468 return mRecordThreads.valueFor(input).get(); 2469} 2470 2471audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2472{ 2473 // This is the internal API, so it is OK to assert on bad parameter. 2474 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2475 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2476 for (int retry = 0; retry < maxRetries; retry++) { 2477 // The cast allows wraparound from max positive to min negative instead of abort 2478 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2479 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2480 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2481 // allow wrap by skipping 0 and -1 for session ids 2482 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2483 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2484 return (audio_unique_id_t) (base | use); 2485 } 2486 } 2487 // We have no way of recovering from wraparound 2488 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2489 // TODO Use a floor after wraparound. This may need a mutex. 2490} 2491 2492AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2493{ 2494 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2495 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2496 if(thread->isDuplicating()) { 2497 continue; 2498 } 2499 AudioStreamOut *output = thread->getOutput(); 2500 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2501 return thread; 2502 } 2503 } 2504 return NULL; 2505} 2506 2507audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2508{ 2509 PlaybackThread *thread = primaryPlaybackThread_l(); 2510 2511 if (thread == NULL) { 2512 return 0; 2513 } 2514 2515 return thread->outDevice(); 2516} 2517 2518AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const 2519{ 2520 size_t minFrameCount = 0; 2521 PlaybackThread *minThread = NULL; 2522 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2523 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2524 if (!thread->isDuplicating()) { 2525 size_t frameCount = thread->frameCountHAL(); 2526 if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount || 2527 (frameCount == minFrameCount && thread->hasFastMixer() && 2528 /*minThread != NULL &&*/ !minThread->hasFastMixer()))) { 2529 minFrameCount = frameCount; 2530 minThread = thread; 2531 } 2532 } 2533 } 2534 return minThread; 2535} 2536 2537sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2538 audio_session_t triggerSession, 2539 audio_session_t listenerSession, 2540 sync_event_callback_t callBack, 2541 const wp<RefBase>& cookie) 2542{ 2543 Mutex::Autolock _l(mLock); 2544 2545 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2546 status_t playStatus = NAME_NOT_FOUND; 2547 status_t recStatus = NAME_NOT_FOUND; 2548 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2549 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2550 if (playStatus == NO_ERROR) { 2551 return event; 2552 } 2553 } 2554 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2555 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2556 if (recStatus == NO_ERROR) { 2557 return event; 2558 } 2559 } 2560 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2561 mPendingSyncEvents.add(event); 2562 } else { 2563 ALOGV("createSyncEvent() invalid event %d", event->type()); 2564 event.clear(); 2565 } 2566 return event; 2567} 2568 2569// ---------------------------------------------------------------------------- 2570// Effect management 2571// ---------------------------------------------------------------------------- 2572 2573sp<EffectsFactoryHalInterface> AudioFlinger::getEffectsFactory() { 2574 return mEffectsFactoryHal; 2575} 2576 2577status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2578{ 2579 Mutex::Autolock _l(mLock); 2580 if (mEffectsFactoryHal.get()) { 2581 return mEffectsFactoryHal->queryNumberEffects(numEffects); 2582 } else { 2583 return -ENODEV; 2584 } 2585} 2586 2587status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2588{ 2589 Mutex::Autolock _l(mLock); 2590 if (mEffectsFactoryHal.get()) { 2591 return mEffectsFactoryHal->getDescriptor(index, descriptor); 2592 } else { 2593 return -ENODEV; 2594 } 2595} 2596 2597status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2598 effect_descriptor_t *descriptor) const 2599{ 2600 Mutex::Autolock _l(mLock); 2601 if (mEffectsFactoryHal.get()) { 2602 return mEffectsFactoryHal->getDescriptor(pUuid, descriptor); 2603 } else { 2604 return -ENODEV; 2605 } 2606} 2607 2608 2609sp<IEffect> AudioFlinger::createEffect( 2610 effect_descriptor_t *pDesc, 2611 const sp<IEffectClient>& effectClient, 2612 int32_t priority, 2613 audio_io_handle_t io, 2614 audio_session_t sessionId, 2615 const String16& opPackageName, 2616 pid_t pid, 2617 status_t *status, 2618 int *id, 2619 int *enabled) 2620{ 2621 status_t lStatus = NO_ERROR; 2622 sp<EffectHandle> handle; 2623 effect_descriptor_t desc; 2624 2625 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 2626 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 2627 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 2628 ALOGW_IF(pid != -1 && pid != callingPid, 2629 "%s uid %d pid %d tried to pass itself off as pid %d", 2630 __func__, callingUid, callingPid, pid); 2631 pid = callingPid; 2632 } 2633 2634 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p", 2635 pid, effectClient.get(), priority, sessionId, io, mEffectsFactoryHal.get()); 2636 2637 if (pDesc == NULL) { 2638 lStatus = BAD_VALUE; 2639 goto Exit; 2640 } 2641 2642 // check audio settings permission for global effects 2643 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2644 lStatus = PERMISSION_DENIED; 2645 goto Exit; 2646 } 2647 2648 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2649 // that can only be created by audio policy manager (running in same process) 2650 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2651 lStatus = PERMISSION_DENIED; 2652 goto Exit; 2653 } 2654 2655 if (mEffectsFactoryHal == 0) { 2656 lStatus = NO_INIT; 2657 goto Exit; 2658 } 2659 2660 { 2661 if (!EffectsFactoryHalInterface::isNullUuid(&pDesc->uuid)) { 2662 // if uuid is specified, request effect descriptor 2663 lStatus = mEffectsFactoryHal->getDescriptor(&pDesc->uuid, &desc); 2664 if (lStatus < 0) { 2665 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2666 goto Exit; 2667 } 2668 } else { 2669 // if uuid is not specified, look for an available implementation 2670 // of the required type in effect factory 2671 if (EffectsFactoryHalInterface::isNullUuid(&pDesc->type)) { 2672 ALOGW("createEffect() no effect type"); 2673 lStatus = BAD_VALUE; 2674 goto Exit; 2675 } 2676 uint32_t numEffects = 0; 2677 effect_descriptor_t d; 2678 d.flags = 0; // prevent compiler warning 2679 bool found = false; 2680 2681 lStatus = mEffectsFactoryHal->queryNumberEffects(&numEffects); 2682 if (lStatus < 0) { 2683 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2684 goto Exit; 2685 } 2686 for (uint32_t i = 0; i < numEffects; i++) { 2687 lStatus = mEffectsFactoryHal->getDescriptor(i, &desc); 2688 if (lStatus < 0) { 2689 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2690 continue; 2691 } 2692 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2693 // If matching type found save effect descriptor. If the session is 2694 // 0 and the effect is not auxiliary, continue enumeration in case 2695 // an auxiliary version of this effect type is available 2696 found = true; 2697 d = desc; 2698 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2699 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2700 break; 2701 } 2702 } 2703 } 2704 if (!found) { 2705 lStatus = BAD_VALUE; 2706 ALOGW("createEffect() effect not found"); 2707 goto Exit; 2708 } 2709 // For same effect type, chose auxiliary version over insert version if 2710 // connect to output mix (Compliance to OpenSL ES) 2711 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2712 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2713 desc = d; 2714 } 2715 } 2716 2717 // Do not allow auxiliary effects on a session different from 0 (output mix) 2718 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2719 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2720 lStatus = INVALID_OPERATION; 2721 goto Exit; 2722 } 2723 2724 // check recording permission for visualizer 2725 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2726 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2727 lStatus = PERMISSION_DENIED; 2728 goto Exit; 2729 } 2730 2731 // return effect descriptor 2732 *pDesc = desc; 2733 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2734 // if the output returned by getOutputForEffect() is removed before we lock the 2735 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2736 // and we will exit safely 2737 io = AudioSystem::getOutputForEffect(&desc); 2738 ALOGV("createEffect got output %d", io); 2739 } 2740 2741 Mutex::Autolock _l(mLock); 2742 2743 // If output is not specified try to find a matching audio session ID in one of the 2744 // output threads. 2745 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2746 // because of code checking output when entering the function. 2747 // Note: io is never 0 when creating an effect on an input 2748 if (io == AUDIO_IO_HANDLE_NONE) { 2749 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2750 // output must be specified by AudioPolicyManager when using session 2751 // AUDIO_SESSION_OUTPUT_STAGE 2752 lStatus = BAD_VALUE; 2753 goto Exit; 2754 } 2755 // look for the thread where the specified audio session is present 2756 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2757 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2758 io = mPlaybackThreads.keyAt(i); 2759 break; 2760 } 2761 } 2762 if (io == 0) { 2763 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2764 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2765 io = mRecordThreads.keyAt(i); 2766 break; 2767 } 2768 } 2769 } 2770 // If no output thread contains the requested session ID, default to 2771 // first output. The effect chain will be moved to the correct output 2772 // thread when a track with the same session ID is created 2773 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2774 io = mPlaybackThreads.keyAt(0); 2775 } 2776 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2777 } 2778 ThreadBase *thread = checkRecordThread_l(io); 2779 if (thread == NULL) { 2780 thread = checkPlaybackThread_l(io); 2781 if (thread == NULL) { 2782 ALOGE("createEffect() unknown output thread"); 2783 lStatus = BAD_VALUE; 2784 goto Exit; 2785 } 2786 } else { 2787 // Check if one effect chain was awaiting for an effect to be created on this 2788 // session and used it instead of creating a new one. 2789 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2790 if (chain != 0) { 2791 Mutex::Autolock _l(thread->mLock); 2792 thread->addEffectChain_l(chain); 2793 } 2794 } 2795 2796 sp<Client> client = registerPid(pid); 2797 2798 // create effect on selected output thread 2799 bool pinned = (sessionId > AUDIO_SESSION_OUTPUT_MIX) && isSessionAcquired_l(sessionId); 2800 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2801 &desc, enabled, &lStatus, pinned); 2802 if (handle != 0 && id != NULL) { 2803 *id = handle->id(); 2804 } 2805 if (handle == 0) { 2806 // remove local strong reference to Client with mClientLock held 2807 Mutex::Autolock _cl(mClientLock); 2808 client.clear(); 2809 } 2810 } 2811 2812Exit: 2813 *status = lStatus; 2814 return handle; 2815} 2816 2817status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2818 audio_io_handle_t dstOutput) 2819{ 2820 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2821 sessionId, srcOutput, dstOutput); 2822 Mutex::Autolock _l(mLock); 2823 if (srcOutput == dstOutput) { 2824 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2825 return NO_ERROR; 2826 } 2827 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2828 if (srcThread == NULL) { 2829 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2830 return BAD_VALUE; 2831 } 2832 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2833 if (dstThread == NULL) { 2834 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2835 return BAD_VALUE; 2836 } 2837 2838 Mutex::Autolock _dl(dstThread->mLock); 2839 Mutex::Autolock _sl(srcThread->mLock); 2840 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2841} 2842 2843// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2844status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2845 AudioFlinger::PlaybackThread *srcThread, 2846 AudioFlinger::PlaybackThread *dstThread, 2847 bool reRegister) 2848{ 2849 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2850 sessionId, srcThread, dstThread); 2851 2852 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2853 if (chain == 0) { 2854 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2855 sessionId, srcThread); 2856 return INVALID_OPERATION; 2857 } 2858 2859 // Check whether the destination thread and all effects in the chain are compatible 2860 if (!chain->isCompatibleWithThread_l(dstThread)) { 2861 ALOGW("moveEffectChain_l() effect chain failed because" 2862 " destination thread %p is not compatible with effects in the chain", 2863 dstThread); 2864 return INVALID_OPERATION; 2865 } 2866 2867 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2868 // so that a new chain is created with correct parameters when first effect is added. This is 2869 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2870 // removed. 2871 srcThread->removeEffectChain_l(chain); 2872 2873 // transfer all effects one by one so that new effect chain is created on new thread with 2874 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2875 sp<EffectChain> dstChain; 2876 uint32_t strategy = 0; // prevent compiler warning 2877 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2878 Vector< sp<EffectModule> > removed; 2879 status_t status = NO_ERROR; 2880 while (effect != 0) { 2881 srcThread->removeEffect_l(effect); 2882 removed.add(effect); 2883 status = dstThread->addEffect_l(effect); 2884 if (status != NO_ERROR) { 2885 break; 2886 } 2887 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2888 if (effect->state() == EffectModule::ACTIVE || 2889 effect->state() == EffectModule::STOPPING) { 2890 effect->start(); 2891 } 2892 // if the move request is not received from audio policy manager, the effect must be 2893 // re-registered with the new strategy and output 2894 if (dstChain == 0) { 2895 dstChain = effect->chain().promote(); 2896 if (dstChain == 0) { 2897 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2898 status = NO_INIT; 2899 break; 2900 } 2901 strategy = dstChain->strategy(); 2902 } 2903 if (reRegister) { 2904 AudioSystem::unregisterEffect(effect->id()); 2905 AudioSystem::registerEffect(&effect->desc(), 2906 dstThread->id(), 2907 strategy, 2908 sessionId, 2909 effect->id()); 2910 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2911 } 2912 effect = chain->getEffectFromId_l(0); 2913 } 2914 2915 if (status != NO_ERROR) { 2916 for (size_t i = 0; i < removed.size(); i++) { 2917 srcThread->addEffect_l(removed[i]); 2918 if (dstChain != 0 && reRegister) { 2919 AudioSystem::unregisterEffect(removed[i]->id()); 2920 AudioSystem::registerEffect(&removed[i]->desc(), 2921 srcThread->id(), 2922 strategy, 2923 sessionId, 2924 removed[i]->id()); 2925 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2926 } 2927 } 2928 } 2929 2930 return status; 2931} 2932 2933bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2934{ 2935 if (mGlobalEffectEnableTime != 0 && 2936 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2937 return true; 2938 } 2939 2940 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2941 sp<EffectChain> ec = 2942 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2943 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2944 return true; 2945 } 2946 } 2947 return false; 2948} 2949 2950void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2951{ 2952 Mutex::Autolock _l(mLock); 2953 2954 mGlobalEffectEnableTime = systemTime(); 2955 2956 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2957 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2958 if (t->mType == ThreadBase::OFFLOAD) { 2959 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2960 } 2961 } 2962 2963} 2964 2965status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2966{ 2967 audio_session_t session = chain->sessionId(); 2968 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2969 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2970 if (index >= 0) { 2971 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2972 return ALREADY_EXISTS; 2973 } 2974 mOrphanEffectChains.add(session, chain); 2975 return NO_ERROR; 2976} 2977 2978sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2979{ 2980 sp<EffectChain> chain; 2981 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2982 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2983 if (index >= 0) { 2984 chain = mOrphanEffectChains.valueAt(index); 2985 mOrphanEffectChains.removeItemsAt(index); 2986 } 2987 return chain; 2988} 2989 2990bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2991{ 2992 Mutex::Autolock _l(mLock); 2993 audio_session_t session = effect->sessionId(); 2994 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2995 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2996 if (index >= 0) { 2997 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2998 if (chain->removeEffect_l(effect, true) == 0) { 2999 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 3000 mOrphanEffectChains.removeItemsAt(index); 3001 } 3002 return true; 3003 } 3004 return false; 3005} 3006 3007 3008struct Entry { 3009#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 3010 char mFileName[TEE_MAX_FILENAME]; 3011}; 3012 3013int comparEntry(const void *p1, const void *p2) 3014{ 3015 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 3016} 3017 3018#ifdef TEE_SINK 3019void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3020{ 3021 NBAIO_Source *teeSource = source.get(); 3022 if (teeSource != NULL) { 3023 // .wav rotation 3024 // There is a benign race condition if 2 threads call this simultaneously. 3025 // They would both traverse the directory, but the result would simply be 3026 // failures at unlink() which are ignored. It's also unlikely since 3027 // normally dumpsys is only done by bugreport or from the command line. 3028 char teePath[32+256]; 3029 strcpy(teePath, "/data/misc/audioserver"); 3030 size_t teePathLen = strlen(teePath); 3031 DIR *dir = opendir(teePath); 3032 teePath[teePathLen++] = '/'; 3033 if (dir != NULL) { 3034#define TEE_MAX_SORT 20 // number of entries to sort 3035#define TEE_MAX_KEEP 10 // number of entries to keep 3036 struct Entry entries[TEE_MAX_SORT]; 3037 size_t entryCount = 0; 3038 while (entryCount < TEE_MAX_SORT) { 3039 struct dirent de; 3040 struct dirent *result = NULL; 3041 int rc = readdir_r(dir, &de, &result); 3042 if (rc != 0) { 3043 ALOGW("readdir_r failed %d", rc); 3044 break; 3045 } 3046 if (result == NULL) { 3047 break; 3048 } 3049 if (result != &de) { 3050 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 3051 break; 3052 } 3053 // ignore non .wav file entries 3054 size_t nameLen = strlen(de.d_name); 3055 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 3056 strcmp(&de.d_name[nameLen - 4], ".wav")) { 3057 continue; 3058 } 3059 strcpy(entries[entryCount++].mFileName, de.d_name); 3060 } 3061 (void) closedir(dir); 3062 if (entryCount > TEE_MAX_KEEP) { 3063 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3064 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3065 strcpy(&teePath[teePathLen], entries[i].mFileName); 3066 (void) unlink(teePath); 3067 } 3068 } 3069 } else { 3070 if (fd >= 0) { 3071 dprintf(fd, "unable to rotate tees in %.*s: %s\n", (int) teePathLen, teePath, 3072 strerror(errno)); 3073 } 3074 } 3075 char teeTime[16]; 3076 struct timeval tv; 3077 gettimeofday(&tv, NULL); 3078 struct tm tm; 3079 localtime_r(&tv.tv_sec, &tm); 3080 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3081 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 3082 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3083 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3084 if (teeFd >= 0) { 3085 // FIXME use libsndfile 3086 char wavHeader[44]; 3087 memcpy(wavHeader, 3088 "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", 3089 sizeof(wavHeader)); 3090 NBAIO_Format format = teeSource->format(); 3091 unsigned channelCount = Format_channelCount(format); 3092 uint32_t sampleRate = Format_sampleRate(format); 3093 size_t frameSize = Format_frameSize(format); 3094 wavHeader[22] = channelCount; // number of channels 3095 wavHeader[24] = sampleRate; // sample rate 3096 wavHeader[25] = sampleRate >> 8; 3097 wavHeader[32] = frameSize; // block alignment 3098 wavHeader[33] = frameSize >> 8; 3099 write(teeFd, wavHeader, sizeof(wavHeader)); 3100 size_t total = 0; 3101 bool firstRead = true; 3102#define TEE_SINK_READ 1024 // frames per I/O operation 3103 void *buffer = malloc(TEE_SINK_READ * frameSize); 3104 for (;;) { 3105 size_t count = TEE_SINK_READ; 3106 ssize_t actual = teeSource->read(buffer, count); 3107 bool wasFirstRead = firstRead; 3108 firstRead = false; 3109 if (actual <= 0) { 3110 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3111 continue; 3112 } 3113 break; 3114 } 3115 ALOG_ASSERT(actual <= (ssize_t)count); 3116 write(teeFd, buffer, actual * frameSize); 3117 total += actual; 3118 } 3119 free(buffer); 3120 lseek(teeFd, (off_t) 4, SEEK_SET); 3121 uint32_t temp = 44 + total * frameSize - 8; 3122 // FIXME not big-endian safe 3123 write(teeFd, &temp, sizeof(temp)); 3124 lseek(teeFd, (off_t) 40, SEEK_SET); 3125 temp = total * frameSize; 3126 // FIXME not big-endian safe 3127 write(teeFd, &temp, sizeof(temp)); 3128 close(teeFd); 3129 if (fd >= 0) { 3130 dprintf(fd, "tee copied to %s\n", teePath); 3131 } 3132 } else { 3133 if (fd >= 0) { 3134 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3135 } 3136 } 3137 } 3138} 3139#endif 3140 3141// ---------------------------------------------------------------------------- 3142 3143status_t AudioFlinger::onTransact( 3144 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3145{ 3146 return BnAudioFlinger::onTransact(code, data, reply, flags); 3147} 3148 3149} // namespace android 3150