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 <utils/String16.h> 35 #include <utils/threads.h> 36 #include <utils/Atomic.h> 37 38 #include <cutils/bitops.h> 39 #include <cutils/properties.h> 40 41 #include <system/audio.h> 42 #include <hardware/audio.h> 43 44 #include "AudioMixer.h" 45 #include "AudioFlinger.h" 46 #include "ServiceUtilities.h" 47 48 #include <media/EffectsFactoryApi.h> 49 #include <audio_effects/effect_visualizer.h> 50 #include <audio_effects/effect_ns.h> 51 #include <audio_effects/effect_aec.h> 52 53 #include <audio_utils/primitives.h> 54 55 #include <powermanager/PowerManager.h> 56 57 #include <common_time/cc_helper.h> 58 59 #include <media/IMediaLogService.h> 60 61 #include <media/nbaio/Pipe.h> 62 #include <media/nbaio/PipeReader.h> 63 #include <media/AudioParameter.h> 64 #include <private/android_filesystem_config.h> 65 66 // ---------------------------------------------------------------------------- 67 68 // Note: the following macro is used for extremely verbose logging message. In 69 // order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70 // 0; but one side effect of this is to turn all LOGV's as well. Some messages 71 // are so verbose that we want to suppress them even when we have ALOG_ASSERT 72 // turned on. Do not uncomment the #def below unless you really know what you 73 // are doing and want to see all of the extremely verbose messages. 74 //#define VERY_VERY_VERBOSE_LOGGING 75 #ifdef VERY_VERY_VERBOSE_LOGGING 76 #define ALOGVV ALOGV 77 #else 78 #define ALOGVV(a...) do { } while(0) 79 #endif 80 81 namespace android { 82 83 static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84 static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85 86 87 nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 88 89 uint32_t AudioFlinger::mScreenState; 90 91 #ifdef TEE_SINK 92 bool AudioFlinger::mTeeSinkInputEnabled = false; 93 bool AudioFlinger::mTeeSinkOutputEnabled = false; 94 bool AudioFlinger::mTeeSinkTrackEnabled = false; 95 96 size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 97 size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 98 size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 99 #endif 100 101 // In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 102 // we define a minimum time during which a global effect is considered enabled. 103 static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 104 105 // ---------------------------------------------------------------------------- 106 107 static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 108 { 109 const hw_module_t *mod; 110 int rc; 111 112 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 113 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 114 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 115 if (rc) { 116 goto out; 117 } 118 rc = audio_hw_device_open(mod, dev); 119 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 120 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 121 if (rc) { 122 goto out; 123 } 124 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 125 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 126 rc = BAD_VALUE; 127 goto out; 128 } 129 return 0; 130 131 out: 132 *dev = NULL; 133 return rc; 134 } 135 136 // ---------------------------------------------------------------------------- 137 138 AudioFlinger::AudioFlinger() 139 : BnAudioFlinger(), 140 mPrimaryHardwareDev(NULL), 141 mHardwareStatus(AUDIO_HW_IDLE), 142 mMasterVolume(1.0f), 143 mMasterMute(false), 144 mNextUniqueId(1), 145 mMode(AUDIO_MODE_INVALID), 146 mBtNrecIsOff(false), 147 mIsLowRamDevice(true), 148 mIsDeviceTypeKnown(false), 149 mGlobalEffectEnableTime(0) 150 { 151 getpid_cached = getpid(); 152 char value[PROPERTY_VALUE_MAX]; 153 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 154 if (doLog) { 155 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters"); 156 } 157 #ifdef TEE_SINK 158 (void) property_get("ro.debuggable", value, "0"); 159 int debuggable = atoi(value); 160 int teeEnabled = 0; 161 if (debuggable) { 162 (void) property_get("af.tee", value, "0"); 163 teeEnabled = atoi(value); 164 } 165 if (teeEnabled & 1) 166 mTeeSinkInputEnabled = true; 167 if (teeEnabled & 2) 168 mTeeSinkOutputEnabled = true; 169 if (teeEnabled & 4) 170 mTeeSinkTrackEnabled = true; 171 #endif 172 } 173 174 void AudioFlinger::onFirstRef() 175 { 176 int rc = 0; 177 178 Mutex::Autolock _l(mLock); 179 180 /* TODO: move all this work into an Init() function */ 181 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 182 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 183 uint32_t int_val; 184 if (1 == sscanf(val_str, "%u", &int_val)) { 185 mStandbyTimeInNsecs = milliseconds(int_val); 186 ALOGI("Using %u mSec as standby time.", int_val); 187 } else { 188 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 189 ALOGI("Using default %u mSec as standby time.", 190 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 191 } 192 } 193 194 mMode = AUDIO_MODE_NORMAL; 195 } 196 197 AudioFlinger::~AudioFlinger() 198 { 199 while (!mRecordThreads.isEmpty()) { 200 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 201 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 202 } 203 while (!mPlaybackThreads.isEmpty()) { 204 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 205 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 206 } 207 208 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 209 // no mHardwareLock needed, as there are no other references to this 210 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 211 delete mAudioHwDevs.valueAt(i); 212 } 213 } 214 215 static const char * const audio_interfaces[] = { 216 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 217 AUDIO_HARDWARE_MODULE_ID_A2DP, 218 AUDIO_HARDWARE_MODULE_ID_USB, 219 }; 220 #define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 221 222 AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 223 audio_module_handle_t module, 224 audio_devices_t devices) 225 { 226 // if module is 0, the request comes from an old policy manager and we should load 227 // well known modules 228 if (module == 0) { 229 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 230 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 231 loadHwModule_l(audio_interfaces[i]); 232 } 233 // then try to find a module supporting the requested device. 234 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 235 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 236 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 237 if ((dev->get_supported_devices != NULL) && 238 (dev->get_supported_devices(dev) & devices) == devices) 239 return audioHwDevice; 240 } 241 } else { 242 // check a match for the requested module handle 243 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 244 if (audioHwDevice != NULL) { 245 return audioHwDevice; 246 } 247 } 248 249 return NULL; 250 } 251 252 void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 253 { 254 const size_t SIZE = 256; 255 char buffer[SIZE]; 256 String8 result; 257 258 result.append("Clients:\n"); 259 for (size_t i = 0; i < mClients.size(); ++i) { 260 sp<Client> client = mClients.valueAt(i).promote(); 261 if (client != 0) { 262 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 263 result.append(buffer); 264 } 265 } 266 267 result.append("Notification Clients:\n"); 268 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 269 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 270 result.append(buffer); 271 } 272 273 result.append("Global session refs:\n"); 274 result.append(" session pid count\n"); 275 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 276 AudioSessionRef *r = mAudioSessionRefs[i]; 277 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 278 result.append(buffer); 279 } 280 write(fd, result.string(), result.size()); 281 } 282 283 284 void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 285 { 286 const size_t SIZE = 256; 287 char buffer[SIZE]; 288 String8 result; 289 hardware_call_state hardwareStatus = mHardwareStatus; 290 291 snprintf(buffer, SIZE, "Hardware status: %d\n" 292 "Standby Time mSec: %u\n", 293 hardwareStatus, 294 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 295 result.append(buffer); 296 write(fd, result.string(), result.size()); 297 } 298 299 void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 300 { 301 const size_t SIZE = 256; 302 char buffer[SIZE]; 303 String8 result; 304 snprintf(buffer, SIZE, "Permission Denial: " 305 "can't dump AudioFlinger from pid=%d, uid=%d\n", 306 IPCThreadState::self()->getCallingPid(), 307 IPCThreadState::self()->getCallingUid()); 308 result.append(buffer); 309 write(fd, result.string(), result.size()); 310 } 311 312 bool AudioFlinger::dumpTryLock(Mutex& mutex) 313 { 314 bool locked = false; 315 for (int i = 0; i < kDumpLockRetries; ++i) { 316 if (mutex.tryLock() == NO_ERROR) { 317 locked = true; 318 break; 319 } 320 usleep(kDumpLockSleepUs); 321 } 322 return locked; 323 } 324 325 status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 326 { 327 if (!dumpAllowed()) { 328 dumpPermissionDenial(fd, args); 329 } else { 330 // get state of hardware lock 331 bool hardwareLocked = dumpTryLock(mHardwareLock); 332 if (!hardwareLocked) { 333 String8 result(kHardwareLockedString); 334 write(fd, result.string(), result.size()); 335 } else { 336 mHardwareLock.unlock(); 337 } 338 339 bool locked = dumpTryLock(mLock); 340 341 // failed to lock - AudioFlinger is probably deadlocked 342 if (!locked) { 343 String8 result(kDeadlockedString); 344 write(fd, result.string(), result.size()); 345 } 346 347 dumpClients(fd, args); 348 dumpInternals(fd, args); 349 350 // dump playback threads 351 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 352 mPlaybackThreads.valueAt(i)->dump(fd, args); 353 } 354 355 // dump record threads 356 for (size_t i = 0; i < mRecordThreads.size(); i++) { 357 mRecordThreads.valueAt(i)->dump(fd, args); 358 } 359 360 // dump all hardware devs 361 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 362 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 363 dev->dump(dev, fd); 364 } 365 366 #ifdef TEE_SINK 367 // dump the serially shared record tee sink 368 if (mRecordTeeSource != 0) { 369 dumpTee(fd, mRecordTeeSource); 370 } 371 #endif 372 373 if (locked) { 374 mLock.unlock(); 375 } 376 377 // append a copy of media.log here by forwarding fd to it, but don't attempt 378 // to lookup the service if it's not running, as it will block for a second 379 if (mLogMemoryDealer != 0) { 380 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 381 if (binder != 0) { 382 fdprintf(fd, "\nmedia.log:\n"); 383 Vector<String16> args; 384 binder->dump(fd, args); 385 } 386 } 387 } 388 return NO_ERROR; 389 } 390 391 sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 392 { 393 // If pid is already in the mClients wp<> map, then use that entry 394 // (for which promote() is always != 0), otherwise create a new entry and Client. 395 sp<Client> client = mClients.valueFor(pid).promote(); 396 if (client == 0) { 397 client = new Client(this, pid); 398 mClients.add(pid, client); 399 } 400 401 return client; 402 } 403 404 sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 405 { 406 if (mLogMemoryDealer == 0) { 407 return new NBLog::Writer(); 408 } 409 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 410 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 411 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 412 if (binder != 0) { 413 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 414 } 415 return writer; 416 } 417 418 void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 419 { 420 if (writer == 0) { 421 return; 422 } 423 sp<IMemory> iMemory(writer->getIMemory()); 424 if (iMemory == 0) { 425 return; 426 } 427 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 428 if (binder != 0) { 429 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 430 // Now the media.log remote reference to IMemory is gone. 431 // When our last local reference to IMemory also drops to zero, 432 // the IMemory destructor will deallocate the region from mMemoryDealer. 433 } 434 } 435 436 // IAudioFlinger interface 437 438 439 sp<IAudioTrack> AudioFlinger::createTrack( 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 audio_channel_mask_t channelMask, 444 size_t frameCount, 445 IAudioFlinger::track_flags_t *flags, 446 const sp<IMemory>& sharedBuffer, 447 audio_io_handle_t output, 448 pid_t tid, 449 int *sessionId, 450 String8& name, 451 status_t *status) 452 { 453 sp<PlaybackThread::Track> track; 454 sp<TrackHandle> trackHandle; 455 sp<Client> client; 456 status_t lStatus; 457 int lSessionId; 458 459 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 460 // but if someone uses binder directly they could bypass that and cause us to crash 461 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 462 ALOGE("createTrack() invalid stream type %d", streamType); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 468 // and we don't yet support 8.24 or 32-bit PCM 469 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 470 ALOGE("createTrack() invalid format %d", format); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 { 476 Mutex::Autolock _l(mLock); 477 PlaybackThread *thread = checkPlaybackThread_l(output); 478 PlaybackThread *effectThread = NULL; 479 if (thread == NULL) { 480 ALOGE("no playback thread found for output handle %d", output); 481 lStatus = BAD_VALUE; 482 goto Exit; 483 } 484 485 pid_t pid = IPCThreadState::self()->getCallingPid(); 486 client = registerPid_l(pid); 487 488 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 489 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 490 // check if an effect chain with the same session ID is present on another 491 // output thread and move it here. 492 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 493 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 494 if (mPlaybackThreads.keyAt(i) != output) { 495 uint32_t sessions = t->hasAudioSession(*sessionId); 496 if (sessions & PlaybackThread::EFFECT_SESSION) { 497 effectThread = t.get(); 498 break; 499 } 500 } 501 } 502 lSessionId = *sessionId; 503 } else { 504 // if no audio session id is provided, create one here 505 lSessionId = nextUniqueId(); 506 if (sessionId != NULL) { 507 *sessionId = lSessionId; 508 } 509 } 510 ALOGV("createTrack() lSessionId: %d", lSessionId); 511 512 track = thread->createTrack_l(client, streamType, sampleRate, format, 513 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 514 515 // move effect chain to this output thread if an effect on same session was waiting 516 // for a track to be created 517 if (lStatus == NO_ERROR && effectThread != NULL) { 518 Mutex::Autolock _dl(thread->mLock); 519 Mutex::Autolock _sl(effectThread->mLock); 520 moveEffectChain_l(lSessionId, effectThread, thread, true); 521 } 522 523 // Look for sync events awaiting for a session to be used. 524 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 525 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 526 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 527 if (lStatus == NO_ERROR) { 528 (void) track->setSyncEvent(mPendingSyncEvents[i]); 529 } else { 530 mPendingSyncEvents[i]->cancel(); 531 } 532 mPendingSyncEvents.removeAt(i); 533 i--; 534 } 535 } 536 } 537 } 538 if (lStatus == NO_ERROR) { 539 // s for server's pid, n for normal mixer name, f for fast index 540 name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, 541 track->fastIndex()); 542 trackHandle = new TrackHandle(track); 543 } else { 544 // remove local strong reference to Client before deleting the Track so that the Client 545 // destructor is called by the TrackBase destructor with mLock held 546 client.clear(); 547 track.clear(); 548 } 549 550 Exit: 551 if (status != NULL) { 552 *status = lStatus; 553 } 554 return trackHandle; 555 } 556 557 uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 558 { 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("sampleRate() unknown thread %d", output); 563 return 0; 564 } 565 return thread->sampleRate(); 566 } 567 568 int AudioFlinger::channelCount(audio_io_handle_t output) const 569 { 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("channelCount() unknown thread %d", output); 574 return 0; 575 } 576 return thread->channelCount(); 577 } 578 579 audio_format_t AudioFlinger::format(audio_io_handle_t output) const 580 { 581 Mutex::Autolock _l(mLock); 582 PlaybackThread *thread = checkPlaybackThread_l(output); 583 if (thread == NULL) { 584 ALOGW("format() unknown thread %d", output); 585 return AUDIO_FORMAT_INVALID; 586 } 587 return thread->format(); 588 } 589 590 size_t AudioFlinger::frameCount(audio_io_handle_t output) const 591 { 592 Mutex::Autolock _l(mLock); 593 PlaybackThread *thread = checkPlaybackThread_l(output); 594 if (thread == NULL) { 595 ALOGW("frameCount() unknown thread %d", output); 596 return 0; 597 } 598 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 599 // should examine all callers and fix them to handle smaller counts 600 return thread->frameCount(); 601 } 602 603 uint32_t AudioFlinger::latency(audio_io_handle_t output) const 604 { 605 Mutex::Autolock _l(mLock); 606 PlaybackThread *thread = checkPlaybackThread_l(output); 607 if (thread == NULL) { 608 ALOGW("latency(): no playback thread found for output handle %d", output); 609 return 0; 610 } 611 return thread->latency(); 612 } 613 614 status_t AudioFlinger::setMasterVolume(float value) 615 { 616 status_t ret = initCheck(); 617 if (ret != NO_ERROR) { 618 return ret; 619 } 620 621 // check calling permissions 622 if (!settingsAllowed()) { 623 return PERMISSION_DENIED; 624 } 625 626 Mutex::Autolock _l(mLock); 627 mMasterVolume = value; 628 629 // Set master volume in the HALs which support it. 630 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 631 AutoMutex lock(mHardwareLock); 632 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 633 634 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 635 if (dev->canSetMasterVolume()) { 636 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 637 } 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 // Now set the master volume in each playback thread. Playback threads 642 // assigned to HALs which do not have master volume support will apply 643 // master volume during the mix operation. Threads with HALs which do 644 // support master volume will simply ignore the setting. 645 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 646 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 647 648 return NO_ERROR; 649 } 650 651 status_t AudioFlinger::setMode(audio_mode_t mode) 652 { 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 663 ALOGW("Illegal value: setMode(%d)", mode); 664 return BAD_VALUE; 665 } 666 667 { // scope for the lock 668 AutoMutex lock(mHardwareLock); 669 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 670 mHardwareStatus = AUDIO_HW_SET_MODE; 671 ret = dev->set_mode(dev, mode); 672 mHardwareStatus = AUDIO_HW_IDLE; 673 } 674 675 if (NO_ERROR == ret) { 676 Mutex::Autolock _l(mLock); 677 mMode = mode; 678 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 679 mPlaybackThreads.valueAt(i)->setMode(mode); 680 } 681 682 return ret; 683 } 684 685 status_t AudioFlinger::setMicMute(bool state) 686 { 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return ret; 690 } 691 692 // check calling permissions 693 if (!settingsAllowed()) { 694 return PERMISSION_DENIED; 695 } 696 697 AutoMutex lock(mHardwareLock); 698 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 699 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 700 ret = dev->set_mic_mute(dev, state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return ret; 703 } 704 705 bool AudioFlinger::getMicMute() const 706 { 707 status_t ret = initCheck(); 708 if (ret != NO_ERROR) { 709 return false; 710 } 711 712 bool state = AUDIO_MODE_INVALID; 713 AutoMutex lock(mHardwareLock); 714 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 715 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 716 dev->get_mic_mute(dev, &state); 717 mHardwareStatus = AUDIO_HW_IDLE; 718 return state; 719 } 720 721 status_t AudioFlinger::setMasterMute(bool muted) 722 { 723 status_t ret = initCheck(); 724 if (ret != NO_ERROR) { 725 return ret; 726 } 727 728 // check calling permissions 729 if (!settingsAllowed()) { 730 return PERMISSION_DENIED; 731 } 732 733 Mutex::Autolock _l(mLock); 734 mMasterMute = muted; 735 736 // Set master mute in the HALs which support it. 737 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 738 AutoMutex lock(mHardwareLock); 739 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 740 741 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 742 if (dev->canSetMasterMute()) { 743 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 744 } 745 mHardwareStatus = AUDIO_HW_IDLE; 746 } 747 748 // Now set the master mute in each playback thread. Playback threads 749 // assigned to HALs which do not have master mute support will apply master 750 // mute during the mix operation. Threads with HALs which do support master 751 // mute will simply ignore the setting. 752 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 753 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 754 755 return NO_ERROR; 756 } 757 758 float AudioFlinger::masterVolume() const 759 { 760 Mutex::Autolock _l(mLock); 761 return masterVolume_l(); 762 } 763 764 bool AudioFlinger::masterMute() const 765 { 766 Mutex::Autolock _l(mLock); 767 return masterMute_l(); 768 } 769 770 float AudioFlinger::masterVolume_l() const 771 { 772 return mMasterVolume; 773 } 774 775 bool AudioFlinger::masterMute_l() const 776 { 777 return mMasterMute; 778 } 779 780 status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 781 audio_io_handle_t output) 782 { 783 // check calling permissions 784 if (!settingsAllowed()) { 785 return PERMISSION_DENIED; 786 } 787 788 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 789 ALOGE("setStreamVolume() invalid stream %d", stream); 790 return BAD_VALUE; 791 } 792 793 AutoMutex lock(mLock); 794 PlaybackThread *thread = NULL; 795 if (output) { 796 thread = checkPlaybackThread_l(output); 797 if (thread == NULL) { 798 return BAD_VALUE; 799 } 800 } 801 802 mStreamTypes[stream].volume = value; 803 804 if (thread == NULL) { 805 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 806 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 807 } 808 } else { 809 thread->setStreamVolume(stream, value); 810 } 811 812 return NO_ERROR; 813 } 814 815 status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 816 { 817 // check calling permissions 818 if (!settingsAllowed()) { 819 return PERMISSION_DENIED; 820 } 821 822 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 823 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 824 ALOGE("setStreamMute() invalid stream %d", stream); 825 return BAD_VALUE; 826 } 827 828 AutoMutex lock(mLock); 829 mStreamTypes[stream].mute = muted; 830 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 831 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 832 833 return NO_ERROR; 834 } 835 836 float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 837 { 838 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 839 return 0.0f; 840 } 841 842 AutoMutex lock(mLock); 843 float volume; 844 if (output) { 845 PlaybackThread *thread = checkPlaybackThread_l(output); 846 if (thread == NULL) { 847 return 0.0f; 848 } 849 volume = thread->streamVolume(stream); 850 } else { 851 volume = streamVolume_l(stream); 852 } 853 854 return volume; 855 } 856 857 bool AudioFlinger::streamMute(audio_stream_type_t stream) const 858 { 859 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 860 return true; 861 } 862 863 AutoMutex lock(mLock); 864 return streamMute_l(stream); 865 } 866 867 status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 868 { 869 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 870 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 871 872 // check calling permissions 873 if (!settingsAllowed()) { 874 return PERMISSION_DENIED; 875 } 876 877 // ioHandle == 0 means the parameters are global to the audio hardware interface 878 if (ioHandle == 0) { 879 Mutex::Autolock _l(mLock); 880 status_t final_result = NO_ERROR; 881 { 882 AutoMutex lock(mHardwareLock); 883 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 884 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 885 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 886 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 887 final_result = result ?: final_result; 888 } 889 mHardwareStatus = AUDIO_HW_IDLE; 890 } 891 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 892 AudioParameter param = AudioParameter(keyValuePairs); 893 String8 value; 894 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 895 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 896 if (mBtNrecIsOff != btNrecIsOff) { 897 for (size_t i = 0; i < mRecordThreads.size(); i++) { 898 sp<RecordThread> thread = mRecordThreads.valueAt(i); 899 audio_devices_t device = thread->inDevice(); 900 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 901 // collect all of the thread's session IDs 902 KeyedVector<int, bool> ids = thread->sessionIds(); 903 // suspend effects associated with those session IDs 904 for (size_t j = 0; j < ids.size(); ++j) { 905 int sessionId = ids.keyAt(j); 906 thread->setEffectSuspended(FX_IID_AEC, 907 suspend, 908 sessionId); 909 thread->setEffectSuspended(FX_IID_NS, 910 suspend, 911 sessionId); 912 } 913 } 914 mBtNrecIsOff = btNrecIsOff; 915 } 916 } 917 String8 screenState; 918 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 919 bool isOff = screenState == "off"; 920 if (isOff != (AudioFlinger::mScreenState & 1)) { 921 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 922 } 923 } 924 return final_result; 925 } 926 927 // hold a strong ref on thread in case closeOutput() or closeInput() is called 928 // and the thread is exited once the lock is released 929 sp<ThreadBase> thread; 930 { 931 Mutex::Autolock _l(mLock); 932 thread = checkPlaybackThread_l(ioHandle); 933 if (thread == 0) { 934 thread = checkRecordThread_l(ioHandle); 935 } else if (thread == primaryPlaybackThread_l()) { 936 // indicate output device change to all input threads for pre processing 937 AudioParameter param = AudioParameter(keyValuePairs); 938 int value; 939 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 940 (value != 0)) { 941 for (size_t i = 0; i < mRecordThreads.size(); i++) { 942 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 943 } 944 } 945 } 946 } 947 if (thread != 0) { 948 return thread->setParameters(keyValuePairs); 949 } 950 return BAD_VALUE; 951 } 952 953 String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 954 { 955 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 956 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 957 958 Mutex::Autolock _l(mLock); 959 960 if (ioHandle == 0) { 961 String8 out_s8; 962 963 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 964 char *s; 965 { 966 AutoMutex lock(mHardwareLock); 967 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 968 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 969 s = dev->get_parameters(dev, keys.string()); 970 mHardwareStatus = AUDIO_HW_IDLE; 971 } 972 out_s8 += String8(s ? s : ""); 973 free(s); 974 } 975 return out_s8; 976 } 977 978 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 979 if (playbackThread != NULL) { 980 return playbackThread->getParameters(keys); 981 } 982 RecordThread *recordThread = checkRecordThread_l(ioHandle); 983 if (recordThread != NULL) { 984 return recordThread->getParameters(keys); 985 } 986 return String8(""); 987 } 988 989 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 990 audio_channel_mask_t channelMask) const 991 { 992 status_t ret = initCheck(); 993 if (ret != NO_ERROR) { 994 return 0; 995 } 996 997 AutoMutex lock(mHardwareLock); 998 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 999 struct audio_config config; 1000 memset(&config, 0, sizeof(config)); 1001 config.sample_rate = sampleRate; 1002 config.channel_mask = channelMask; 1003 config.format = format; 1004 1005 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1006 size_t size = dev->get_input_buffer_size(dev, &config); 1007 mHardwareStatus = AUDIO_HW_IDLE; 1008 return size; 1009 } 1010 1011 unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1012 { 1013 Mutex::Autolock _l(mLock); 1014 1015 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1016 if (recordThread != NULL) { 1017 return recordThread->getInputFramesLost(); 1018 } 1019 return 0; 1020 } 1021 1022 status_t AudioFlinger::setVoiceVolume(float value) 1023 { 1024 status_t ret = initCheck(); 1025 if (ret != NO_ERROR) { 1026 return ret; 1027 } 1028 1029 // check calling permissions 1030 if (!settingsAllowed()) { 1031 return PERMISSION_DENIED; 1032 } 1033 1034 AutoMutex lock(mHardwareLock); 1035 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1036 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1037 ret = dev->set_voice_volume(dev, value); 1038 mHardwareStatus = AUDIO_HW_IDLE; 1039 1040 return ret; 1041 } 1042 1043 status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1044 audio_io_handle_t output) const 1045 { 1046 status_t status; 1047 1048 Mutex::Autolock _l(mLock); 1049 1050 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1051 if (playbackThread != NULL) { 1052 return playbackThread->getRenderPosition(halFrames, dspFrames); 1053 } 1054 1055 return BAD_VALUE; 1056 } 1057 1058 void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1059 { 1060 1061 Mutex::Autolock _l(mLock); 1062 1063 pid_t pid = IPCThreadState::self()->getCallingPid(); 1064 if (mNotificationClients.indexOfKey(pid) < 0) { 1065 sp<NotificationClient> notificationClient = new NotificationClient(this, 1066 client, 1067 pid); 1068 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1069 1070 mNotificationClients.add(pid, notificationClient); 1071 1072 sp<IBinder> binder = client->asBinder(); 1073 binder->linkToDeath(notificationClient); 1074 1075 // the config change is always sent from playback or record threads to avoid deadlock 1076 // with AudioSystem::gLock 1077 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1078 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1079 } 1080 1081 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1082 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1083 } 1084 } 1085 } 1086 1087 void AudioFlinger::removeNotificationClient(pid_t pid) 1088 { 1089 Mutex::Autolock _l(mLock); 1090 1091 mNotificationClients.removeItem(pid); 1092 1093 ALOGV("%d died, releasing its sessions", pid); 1094 size_t num = mAudioSessionRefs.size(); 1095 bool removed = false; 1096 for (size_t i = 0; i< num; ) { 1097 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1098 ALOGV(" pid %d @ %d", ref->mPid, i); 1099 if (ref->mPid == pid) { 1100 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1101 mAudioSessionRefs.removeAt(i); 1102 delete ref; 1103 removed = true; 1104 num--; 1105 } else { 1106 i++; 1107 } 1108 } 1109 if (removed) { 1110 purgeStaleEffects_l(); 1111 } 1112 } 1113 1114 // audioConfigChanged_l() must be called with AudioFlinger::mLock held 1115 void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1116 { 1117 size_t size = mNotificationClients.size(); 1118 for (size_t i = 0; i < size; i++) { 1119 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1120 param2); 1121 } 1122 } 1123 1124 // removeClient_l() must be called with AudioFlinger::mLock held 1125 void AudioFlinger::removeClient_l(pid_t pid) 1126 { 1127 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1128 IPCThreadState::self()->getCallingPid()); 1129 mClients.removeItem(pid); 1130 } 1131 1132 // getEffectThread_l() must be called with AudioFlinger::mLock held 1133 sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1134 { 1135 sp<PlaybackThread> thread; 1136 1137 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1138 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1139 ALOG_ASSERT(thread == 0); 1140 thread = mPlaybackThreads.valueAt(i); 1141 } 1142 } 1143 1144 return thread; 1145 } 1146 1147 1148 1149 // ---------------------------------------------------------------------------- 1150 1151 AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1152 : RefBase(), 1153 mAudioFlinger(audioFlinger), 1154 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1155 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1156 mPid(pid), 1157 mTimedTrackCount(0) 1158 { 1159 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1160 } 1161 1162 // Client destructor must be called with AudioFlinger::mLock held 1163 AudioFlinger::Client::~Client() 1164 { 1165 mAudioFlinger->removeClient_l(mPid); 1166 } 1167 1168 sp<MemoryDealer> AudioFlinger::Client::heap() const 1169 { 1170 return mMemoryDealer; 1171 } 1172 1173 // Reserve one of the limited slots for a timed audio track associated 1174 // with this client 1175 bool AudioFlinger::Client::reserveTimedTrack() 1176 { 1177 const int kMaxTimedTracksPerClient = 4; 1178 1179 Mutex::Autolock _l(mTimedTrackLock); 1180 1181 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1182 ALOGW("can not create timed track - pid %d has exceeded the limit", 1183 mPid); 1184 return false; 1185 } 1186 1187 mTimedTrackCount++; 1188 return true; 1189 } 1190 1191 // Release a slot for a timed audio track 1192 void AudioFlinger::Client::releaseTimedTrack() 1193 { 1194 Mutex::Autolock _l(mTimedTrackLock); 1195 mTimedTrackCount--; 1196 } 1197 1198 // ---------------------------------------------------------------------------- 1199 1200 AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1201 const sp<IAudioFlingerClient>& client, 1202 pid_t pid) 1203 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1204 { 1205 } 1206 1207 AudioFlinger::NotificationClient::~NotificationClient() 1208 { 1209 } 1210 1211 void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1212 { 1213 sp<NotificationClient> keep(this); 1214 mAudioFlinger->removeNotificationClient(mPid); 1215 } 1216 1217 1218 // ---------------------------------------------------------------------------- 1219 1220 static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1221 return audio_is_remote_submix_device(inDevice); 1222 } 1223 1224 sp<IAudioRecord> AudioFlinger::openRecord( 1225 audio_io_handle_t input, 1226 uint32_t sampleRate, 1227 audio_format_t format, 1228 audio_channel_mask_t channelMask, 1229 size_t frameCount, 1230 IAudioFlinger::track_flags_t *flags, 1231 pid_t tid, 1232 int *sessionId, 1233 status_t *status) 1234 { 1235 sp<RecordThread::RecordTrack> recordTrack; 1236 sp<RecordHandle> recordHandle; 1237 sp<Client> client; 1238 status_t lStatus; 1239 RecordThread *thread; 1240 size_t inFrameCount; 1241 int lSessionId; 1242 1243 // check calling permissions 1244 if (!recordingAllowed()) { 1245 ALOGE("openRecord() permission denied: recording not allowed"); 1246 lStatus = PERMISSION_DENIED; 1247 goto Exit; 1248 } 1249 1250 if (format != AUDIO_FORMAT_PCM_16_BIT) { 1251 ALOGE("openRecord() invalid format %d", format); 1252 lStatus = BAD_VALUE; 1253 goto Exit; 1254 } 1255 1256 // add client to list 1257 { // scope for mLock 1258 Mutex::Autolock _l(mLock); 1259 thread = checkRecordThread_l(input); 1260 if (thread == NULL) { 1261 ALOGE("openRecord() checkRecordThread_l failed"); 1262 lStatus = BAD_VALUE; 1263 goto Exit; 1264 } 1265 1266 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1267 && !captureAudioOutputAllowed()) { 1268 ALOGE("openRecord() permission denied: capture not allowed"); 1269 lStatus = PERMISSION_DENIED; 1270 goto Exit; 1271 } 1272 1273 pid_t pid = IPCThreadState::self()->getCallingPid(); 1274 client = registerPid_l(pid); 1275 1276 // If no audio session id is provided, create one here 1277 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1278 lSessionId = *sessionId; 1279 } else { 1280 lSessionId = nextUniqueId(); 1281 if (sessionId != NULL) { 1282 *sessionId = lSessionId; 1283 } 1284 } 1285 // create new record track. 1286 // The record track uses one track in mHardwareMixerThread by convention. 1287 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1288 frameCount, lSessionId, flags, tid, &lStatus); 1289 LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR)); 1290 } 1291 if (lStatus != NO_ERROR) { 1292 // remove local strong reference to Client before deleting the RecordTrack so that the 1293 // Client destructor is called by the TrackBase destructor with mLock held 1294 client.clear(); 1295 recordTrack.clear(); 1296 goto Exit; 1297 } 1298 1299 // return to handle to client 1300 recordHandle = new RecordHandle(recordTrack); 1301 lStatus = NO_ERROR; 1302 1303 Exit: 1304 if (status) { 1305 *status = lStatus; 1306 } 1307 return recordHandle; 1308 } 1309 1310 1311 1312 // ---------------------------------------------------------------------------- 1313 1314 audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1315 { 1316 if (!settingsAllowed()) { 1317 return 0; 1318 } 1319 Mutex::Autolock _l(mLock); 1320 return loadHwModule_l(name); 1321 } 1322 1323 // loadHwModule_l() must be called with AudioFlinger::mLock held 1324 audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1325 { 1326 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1327 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1328 ALOGW("loadHwModule() module %s already loaded", name); 1329 return mAudioHwDevs.keyAt(i); 1330 } 1331 } 1332 1333 audio_hw_device_t *dev; 1334 1335 int rc = load_audio_interface(name, &dev); 1336 if (rc) { 1337 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1338 return 0; 1339 } 1340 1341 mHardwareStatus = AUDIO_HW_INIT; 1342 rc = dev->init_check(dev); 1343 mHardwareStatus = AUDIO_HW_IDLE; 1344 if (rc) { 1345 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1346 return 0; 1347 } 1348 1349 // Check and cache this HAL's level of support for master mute and master 1350 // volume. If this is the first HAL opened, and it supports the get 1351 // methods, use the initial values provided by the HAL as the current 1352 // master mute and volume settings. 1353 1354 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1355 { // scope for auto-lock pattern 1356 AutoMutex lock(mHardwareLock); 1357 1358 if (0 == mAudioHwDevs.size()) { 1359 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1360 if (NULL != dev->get_master_volume) { 1361 float mv; 1362 if (OK == dev->get_master_volume(dev, &mv)) { 1363 mMasterVolume = mv; 1364 } 1365 } 1366 1367 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1368 if (NULL != dev->get_master_mute) { 1369 bool mm; 1370 if (OK == dev->get_master_mute(dev, &mm)) { 1371 mMasterMute = mm; 1372 } 1373 } 1374 } 1375 1376 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1377 if ((NULL != dev->set_master_volume) && 1378 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1379 flags = static_cast<AudioHwDevice::Flags>(flags | 1380 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1381 } 1382 1383 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1384 if ((NULL != dev->set_master_mute) && 1385 (OK == dev->set_master_mute(dev, mMasterMute))) { 1386 flags = static_cast<AudioHwDevice::Flags>(flags | 1387 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1388 } 1389 1390 mHardwareStatus = AUDIO_HW_IDLE; 1391 } 1392 1393 audio_module_handle_t handle = nextUniqueId(); 1394 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1395 1396 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1397 name, dev->common.module->name, dev->common.module->id, handle); 1398 1399 return handle; 1400 1401 } 1402 1403 // ---------------------------------------------------------------------------- 1404 1405 uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1406 { 1407 Mutex::Autolock _l(mLock); 1408 PlaybackThread *thread = primaryPlaybackThread_l(); 1409 return thread != NULL ? thread->sampleRate() : 0; 1410 } 1411 1412 size_t AudioFlinger::getPrimaryOutputFrameCount() 1413 { 1414 Mutex::Autolock _l(mLock); 1415 PlaybackThread *thread = primaryPlaybackThread_l(); 1416 return thread != NULL ? thread->frameCountHAL() : 0; 1417 } 1418 1419 // ---------------------------------------------------------------------------- 1420 1421 status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1422 { 1423 uid_t uid = IPCThreadState::self()->getCallingUid(); 1424 if (uid != AID_SYSTEM) { 1425 return PERMISSION_DENIED; 1426 } 1427 Mutex::Autolock _l(mLock); 1428 if (mIsDeviceTypeKnown) { 1429 return INVALID_OPERATION; 1430 } 1431 mIsLowRamDevice = isLowRamDevice; 1432 mIsDeviceTypeKnown = true; 1433 return NO_ERROR; 1434 } 1435 1436 // ---------------------------------------------------------------------------- 1437 1438 audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1439 audio_devices_t *pDevices, 1440 uint32_t *pSamplingRate, 1441 audio_format_t *pFormat, 1442 audio_channel_mask_t *pChannelMask, 1443 uint32_t *pLatencyMs, 1444 audio_output_flags_t flags, 1445 const audio_offload_info_t *offloadInfo) 1446 { 1447 PlaybackThread *thread = NULL; 1448 struct audio_config config; 1449 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1450 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1451 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1452 if (offloadInfo) { 1453 config.offload_info = *offloadInfo; 1454 } 1455 1456 audio_stream_out_t *outStream = NULL; 1457 AudioHwDevice *outHwDev; 1458 1459 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1460 module, 1461 (pDevices != NULL) ? *pDevices : 0, 1462 config.sample_rate, 1463 config.format, 1464 config.channel_mask, 1465 flags); 1466 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1467 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); 1468 1469 if (pDevices == NULL || *pDevices == 0) { 1470 return 0; 1471 } 1472 1473 Mutex::Autolock _l(mLock); 1474 1475 outHwDev = findSuitableHwDev_l(module, *pDevices); 1476 if (outHwDev == NULL) 1477 return 0; 1478 1479 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1480 audio_io_handle_t id = nextUniqueId(); 1481 1482 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1483 1484 status_t status = hwDevHal->open_output_stream(hwDevHal, 1485 id, 1486 *pDevices, 1487 (audio_output_flags_t)flags, 1488 &config, 1489 &outStream); 1490 1491 mHardwareStatus = AUDIO_HW_IDLE; 1492 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1493 "Channels %x, status %d", 1494 outStream, 1495 config.sample_rate, 1496 config.format, 1497 config.channel_mask, 1498 status); 1499 1500 if (status == NO_ERROR && outStream != NULL) { 1501 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1502 1503 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1504 thread = new OffloadThread(this, output, id, *pDevices); 1505 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1506 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1507 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1508 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1509 thread = new DirectOutputThread(this, output, id, *pDevices); 1510 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1511 } else { 1512 thread = new MixerThread(this, output, id, *pDevices); 1513 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1514 } 1515 mPlaybackThreads.add(id, thread); 1516 1517 if (pSamplingRate != NULL) { 1518 *pSamplingRate = config.sample_rate; 1519 } 1520 if (pFormat != NULL) { 1521 *pFormat = config.format; 1522 } 1523 if (pChannelMask != NULL) { 1524 *pChannelMask = config.channel_mask; 1525 } 1526 if (pLatencyMs != NULL) { 1527 *pLatencyMs = thread->latency(); 1528 } 1529 1530 // notify client processes of the new output creation 1531 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1532 1533 // the first primary output opened designates the primary hw device 1534 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1535 ALOGI("Using module %d has the primary audio interface", module); 1536 mPrimaryHardwareDev = outHwDev; 1537 1538 AutoMutex lock(mHardwareLock); 1539 mHardwareStatus = AUDIO_HW_SET_MODE; 1540 hwDevHal->set_mode(hwDevHal, mMode); 1541 mHardwareStatus = AUDIO_HW_IDLE; 1542 } 1543 return id; 1544 } 1545 1546 return 0; 1547 } 1548 1549 audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1550 audio_io_handle_t output2) 1551 { 1552 Mutex::Autolock _l(mLock); 1553 MixerThread *thread1 = checkMixerThread_l(output1); 1554 MixerThread *thread2 = checkMixerThread_l(output2); 1555 1556 if (thread1 == NULL || thread2 == NULL) { 1557 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1558 output2); 1559 return 0; 1560 } 1561 1562 audio_io_handle_t id = nextUniqueId(); 1563 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1564 thread->addOutputTrack(thread2); 1565 mPlaybackThreads.add(id, thread); 1566 // notify client processes of the new output creation 1567 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1568 return id; 1569 } 1570 1571 status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1572 { 1573 return closeOutput_nonvirtual(output); 1574 } 1575 1576 status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1577 { 1578 // keep strong reference on the playback thread so that 1579 // it is not destroyed while exit() is executed 1580 sp<PlaybackThread> thread; 1581 { 1582 Mutex::Autolock _l(mLock); 1583 thread = checkPlaybackThread_l(output); 1584 if (thread == NULL) { 1585 return BAD_VALUE; 1586 } 1587 1588 ALOGV("closeOutput() %d", output); 1589 1590 if (thread->type() == ThreadBase::MIXER) { 1591 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1592 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1593 DuplicatingThread *dupThread = 1594 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1595 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1596 1597 } 1598 } 1599 } 1600 1601 1602 mPlaybackThreads.removeItem(output); 1603 // save all effects to the default thread 1604 if (mPlaybackThreads.size()) { 1605 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1606 if (dstThread != NULL) { 1607 // audioflinger lock is held here so the acquisition order of thread locks does not 1608 // matter 1609 Mutex::Autolock _dl(dstThread->mLock); 1610 Mutex::Autolock _sl(thread->mLock); 1611 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1612 for (size_t i = 0; i < effectChains.size(); i ++) { 1613 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1614 } 1615 } 1616 } 1617 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1618 } 1619 thread->exit(); 1620 // The thread entity (active unit of execution) is no longer running here, 1621 // but the ThreadBase container still exists. 1622 1623 if (thread->type() != ThreadBase::DUPLICATING) { 1624 AudioStreamOut *out = thread->clearOutput(); 1625 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1626 // from now on thread->mOutput is NULL 1627 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1628 delete out; 1629 } 1630 return NO_ERROR; 1631 } 1632 1633 status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1634 { 1635 Mutex::Autolock _l(mLock); 1636 PlaybackThread *thread = checkPlaybackThread_l(output); 1637 1638 if (thread == NULL) { 1639 return BAD_VALUE; 1640 } 1641 1642 ALOGV("suspendOutput() %d", output); 1643 thread->suspend(); 1644 1645 return NO_ERROR; 1646 } 1647 1648 status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1649 { 1650 Mutex::Autolock _l(mLock); 1651 PlaybackThread *thread = checkPlaybackThread_l(output); 1652 1653 if (thread == NULL) { 1654 return BAD_VALUE; 1655 } 1656 1657 ALOGV("restoreOutput() %d", output); 1658 1659 thread->restore(); 1660 1661 return NO_ERROR; 1662 } 1663 1664 audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1665 audio_devices_t *pDevices, 1666 uint32_t *pSamplingRate, 1667 audio_format_t *pFormat, 1668 audio_channel_mask_t *pChannelMask) 1669 { 1670 status_t status; 1671 RecordThread *thread = NULL; 1672 struct audio_config config; 1673 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1674 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1675 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1676 1677 uint32_t reqSamplingRate = config.sample_rate; 1678 audio_format_t reqFormat = config.format; 1679 audio_channel_mask_t reqChannels = config.channel_mask; 1680 audio_stream_in_t *inStream = NULL; 1681 AudioHwDevice *inHwDev; 1682 1683 if (pDevices == NULL || *pDevices == 0) { 1684 return 0; 1685 } 1686 1687 Mutex::Autolock _l(mLock); 1688 1689 inHwDev = findSuitableHwDev_l(module, *pDevices); 1690 if (inHwDev == NULL) 1691 return 0; 1692 1693 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1694 audio_io_handle_t id = nextUniqueId(); 1695 1696 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1697 &inStream); 1698 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1699 "status %d", 1700 inStream, 1701 config.sample_rate, 1702 config.format, 1703 config.channel_mask, 1704 status); 1705 1706 // If the input could not be opened with the requested parameters and we can handle the 1707 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1708 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1709 if (status == BAD_VALUE && 1710 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1711 (config.sample_rate <= 2 * reqSamplingRate) && 1712 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 1713 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1714 inStream = NULL; 1715 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1716 } 1717 1718 if (status == NO_ERROR && inStream != NULL) { 1719 1720 #ifdef TEE_SINK 1721 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1722 // or (re-)create if current Pipe is idle and does not match the new format 1723 sp<NBAIO_Sink> teeSink; 1724 enum { 1725 TEE_SINK_NO, // don't copy input 1726 TEE_SINK_NEW, // copy input using a new pipe 1727 TEE_SINK_OLD, // copy input using an existing pipe 1728 } kind; 1729 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1730 popcount(inStream->common.get_channels(&inStream->common))); 1731 if (!mTeeSinkInputEnabled) { 1732 kind = TEE_SINK_NO; 1733 } else if (format == Format_Invalid) { 1734 kind = TEE_SINK_NO; 1735 } else if (mRecordTeeSink == 0) { 1736 kind = TEE_SINK_NEW; 1737 } else if (mRecordTeeSink->getStrongCount() != 1) { 1738 kind = TEE_SINK_NO; 1739 } else if (format == mRecordTeeSink->format()) { 1740 kind = TEE_SINK_OLD; 1741 } else { 1742 kind = TEE_SINK_NEW; 1743 } 1744 switch (kind) { 1745 case TEE_SINK_NEW: { 1746 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1747 size_t numCounterOffers = 0; 1748 const NBAIO_Format offers[1] = {format}; 1749 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1750 ALOG_ASSERT(index == 0); 1751 PipeReader *pipeReader = new PipeReader(*pipe); 1752 numCounterOffers = 0; 1753 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1754 ALOG_ASSERT(index == 0); 1755 mRecordTeeSink = pipe; 1756 mRecordTeeSource = pipeReader; 1757 teeSink = pipe; 1758 } 1759 break; 1760 case TEE_SINK_OLD: 1761 teeSink = mRecordTeeSink; 1762 break; 1763 case TEE_SINK_NO: 1764 default: 1765 break; 1766 } 1767 #endif 1768 1769 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1770 1771 // Start record thread 1772 // RecordThread requires both input and output device indication to forward to audio 1773 // pre processing modules 1774 thread = new RecordThread(this, 1775 input, 1776 reqSamplingRate, 1777 reqChannels, 1778 id, 1779 primaryOutputDevice_l(), 1780 *pDevices 1781 #ifdef TEE_SINK 1782 , teeSink 1783 #endif 1784 ); 1785 mRecordThreads.add(id, thread); 1786 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1787 if (pSamplingRate != NULL) { 1788 *pSamplingRate = reqSamplingRate; 1789 } 1790 if (pFormat != NULL) { 1791 *pFormat = config.format; 1792 } 1793 if (pChannelMask != NULL) { 1794 *pChannelMask = reqChannels; 1795 } 1796 1797 // notify client processes of the new input creation 1798 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1799 return id; 1800 } 1801 1802 return 0; 1803 } 1804 1805 status_t AudioFlinger::closeInput(audio_io_handle_t input) 1806 { 1807 return closeInput_nonvirtual(input); 1808 } 1809 1810 status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1811 { 1812 // keep strong reference on the record thread so that 1813 // it is not destroyed while exit() is executed 1814 sp<RecordThread> thread; 1815 { 1816 Mutex::Autolock _l(mLock); 1817 thread = checkRecordThread_l(input); 1818 if (thread == 0) { 1819 return BAD_VALUE; 1820 } 1821 1822 ALOGV("closeInput() %d", input); 1823 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1824 mRecordThreads.removeItem(input); 1825 } 1826 thread->exit(); 1827 // The thread entity (active unit of execution) is no longer running here, 1828 // but the ThreadBase container still exists. 1829 1830 AudioStreamIn *in = thread->clearInput(); 1831 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1832 // from now on thread->mInput is NULL 1833 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1834 delete in; 1835 1836 return NO_ERROR; 1837 } 1838 1839 status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1840 { 1841 Mutex::Autolock _l(mLock); 1842 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1843 1844 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1845 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1846 thread->invalidateTracks(stream); 1847 } 1848 1849 return NO_ERROR; 1850 } 1851 1852 1853 int AudioFlinger::newAudioSessionId() 1854 { 1855 return nextUniqueId(); 1856 } 1857 1858 void AudioFlinger::acquireAudioSessionId(int audioSession) 1859 { 1860 Mutex::Autolock _l(mLock); 1861 pid_t caller = IPCThreadState::self()->getCallingPid(); 1862 ALOGV("acquiring %d from %d", audioSession, caller); 1863 1864 // Ignore requests received from processes not known as notification client. The request 1865 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 1866 // called from a different pid leaving a stale session reference. Also we don't know how 1867 // to clear this reference if the client process dies. 1868 if (mNotificationClients.indexOfKey(caller) < 0) { 1869 ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 1870 return; 1871 } 1872 1873 size_t num = mAudioSessionRefs.size(); 1874 for (size_t i = 0; i< num; i++) { 1875 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1876 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1877 ref->mCnt++; 1878 ALOGV(" incremented refcount to %d", ref->mCnt); 1879 return; 1880 } 1881 } 1882 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1883 ALOGV(" added new entry for %d", audioSession); 1884 } 1885 1886 void AudioFlinger::releaseAudioSessionId(int audioSession) 1887 { 1888 Mutex::Autolock _l(mLock); 1889 pid_t caller = IPCThreadState::self()->getCallingPid(); 1890 ALOGV("releasing %d from %d", audioSession, caller); 1891 size_t num = mAudioSessionRefs.size(); 1892 for (size_t i = 0; i< num; i++) { 1893 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1894 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1895 ref->mCnt--; 1896 ALOGV(" decremented refcount to %d", ref->mCnt); 1897 if (ref->mCnt == 0) { 1898 mAudioSessionRefs.removeAt(i); 1899 delete ref; 1900 purgeStaleEffects_l(); 1901 } 1902 return; 1903 } 1904 } 1905 // If the caller is mediaserver it is likely that the session being released was acquired 1906 // on behalf of a process not in notification clients and we ignore the warning. 1907 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 1908 } 1909 1910 void AudioFlinger::purgeStaleEffects_l() { 1911 1912 ALOGV("purging stale effects"); 1913 1914 Vector< sp<EffectChain> > chains; 1915 1916 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1917 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1918 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1919 sp<EffectChain> ec = t->mEffectChains[j]; 1920 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1921 chains.push(ec); 1922 } 1923 } 1924 } 1925 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1926 sp<RecordThread> t = mRecordThreads.valueAt(i); 1927 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1928 sp<EffectChain> ec = t->mEffectChains[j]; 1929 chains.push(ec); 1930 } 1931 } 1932 1933 for (size_t i = 0; i < chains.size(); i++) { 1934 sp<EffectChain> ec = chains[i]; 1935 int sessionid = ec->sessionId(); 1936 sp<ThreadBase> t = ec->mThread.promote(); 1937 if (t == 0) { 1938 continue; 1939 } 1940 size_t numsessionrefs = mAudioSessionRefs.size(); 1941 bool found = false; 1942 for (size_t k = 0; k < numsessionrefs; k++) { 1943 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1944 if (ref->mSessionid == sessionid) { 1945 ALOGV(" session %d still exists for %d with %d refs", 1946 sessionid, ref->mPid, ref->mCnt); 1947 found = true; 1948 break; 1949 } 1950 } 1951 if (!found) { 1952 Mutex::Autolock _l (t->mLock); 1953 // remove all effects from the chain 1954 while (ec->mEffects.size()) { 1955 sp<EffectModule> effect = ec->mEffects[0]; 1956 effect->unPin(); 1957 t->removeEffect_l(effect); 1958 if (effect->purgeHandles()) { 1959 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1960 } 1961 AudioSystem::unregisterEffect(effect->id()); 1962 } 1963 } 1964 } 1965 return; 1966 } 1967 1968 // checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1969 AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1970 { 1971 return mPlaybackThreads.valueFor(output).get(); 1972 } 1973 1974 // checkMixerThread_l() must be called with AudioFlinger::mLock held 1975 AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1976 { 1977 PlaybackThread *thread = checkPlaybackThread_l(output); 1978 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1979 } 1980 1981 // checkRecordThread_l() must be called with AudioFlinger::mLock held 1982 AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1983 { 1984 return mRecordThreads.valueFor(input).get(); 1985 } 1986 1987 uint32_t AudioFlinger::nextUniqueId() 1988 { 1989 return android_atomic_inc(&mNextUniqueId); 1990 } 1991 1992 AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1993 { 1994 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1995 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1996 AudioStreamOut *output = thread->getOutput(); 1997 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1998 return thread; 1999 } 2000 } 2001 return NULL; 2002 } 2003 2004 audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2005 { 2006 PlaybackThread *thread = primaryPlaybackThread_l(); 2007 2008 if (thread == NULL) { 2009 return 0; 2010 } 2011 2012 return thread->outDevice(); 2013 } 2014 2015 sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2016 int triggerSession, 2017 int listenerSession, 2018 sync_event_callback_t callBack, 2019 void *cookie) 2020 { 2021 Mutex::Autolock _l(mLock); 2022 2023 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2024 status_t playStatus = NAME_NOT_FOUND; 2025 status_t recStatus = NAME_NOT_FOUND; 2026 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2027 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2028 if (playStatus == NO_ERROR) { 2029 return event; 2030 } 2031 } 2032 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2033 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2034 if (recStatus == NO_ERROR) { 2035 return event; 2036 } 2037 } 2038 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2039 mPendingSyncEvents.add(event); 2040 } else { 2041 ALOGV("createSyncEvent() invalid event %d", event->type()); 2042 event.clear(); 2043 } 2044 return event; 2045 } 2046 2047 // ---------------------------------------------------------------------------- 2048 // Effect management 2049 // ---------------------------------------------------------------------------- 2050 2051 2052 status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2053 { 2054 Mutex::Autolock _l(mLock); 2055 return EffectQueryNumberEffects(numEffects); 2056 } 2057 2058 status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2059 { 2060 Mutex::Autolock _l(mLock); 2061 return EffectQueryEffect(index, descriptor); 2062 } 2063 2064 status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2065 effect_descriptor_t *descriptor) const 2066 { 2067 Mutex::Autolock _l(mLock); 2068 return EffectGetDescriptor(pUuid, descriptor); 2069 } 2070 2071 2072 sp<IEffect> AudioFlinger::createEffect( 2073 effect_descriptor_t *pDesc, 2074 const sp<IEffectClient>& effectClient, 2075 int32_t priority, 2076 audio_io_handle_t io, 2077 int sessionId, 2078 status_t *status, 2079 int *id, 2080 int *enabled) 2081 { 2082 status_t lStatus = NO_ERROR; 2083 sp<EffectHandle> handle; 2084 effect_descriptor_t desc; 2085 2086 pid_t pid = IPCThreadState::self()->getCallingPid(); 2087 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2088 pid, effectClient.get(), priority, sessionId, io); 2089 2090 if (pDesc == NULL) { 2091 lStatus = BAD_VALUE; 2092 goto Exit; 2093 } 2094 2095 // check audio settings permission for global effects 2096 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2097 lStatus = PERMISSION_DENIED; 2098 goto Exit; 2099 } 2100 2101 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2102 // that can only be created by audio policy manager (running in same process) 2103 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2104 lStatus = PERMISSION_DENIED; 2105 goto Exit; 2106 } 2107 2108 { 2109 if (!EffectIsNullUuid(&pDesc->uuid)) { 2110 // if uuid is specified, request effect descriptor 2111 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2112 if (lStatus < 0) { 2113 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2114 goto Exit; 2115 } 2116 } else { 2117 // if uuid is not specified, look for an available implementation 2118 // of the required type in effect factory 2119 if (EffectIsNullUuid(&pDesc->type)) { 2120 ALOGW("createEffect() no effect type"); 2121 lStatus = BAD_VALUE; 2122 goto Exit; 2123 } 2124 uint32_t numEffects = 0; 2125 effect_descriptor_t d; 2126 d.flags = 0; // prevent compiler warning 2127 bool found = false; 2128 2129 lStatus = EffectQueryNumberEffects(&numEffects); 2130 if (lStatus < 0) { 2131 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2132 goto Exit; 2133 } 2134 for (uint32_t i = 0; i < numEffects; i++) { 2135 lStatus = EffectQueryEffect(i, &desc); 2136 if (lStatus < 0) { 2137 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2138 continue; 2139 } 2140 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2141 // If matching type found save effect descriptor. If the session is 2142 // 0 and the effect is not auxiliary, continue enumeration in case 2143 // an auxiliary version of this effect type is available 2144 found = true; 2145 d = desc; 2146 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2147 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2148 break; 2149 } 2150 } 2151 } 2152 if (!found) { 2153 lStatus = BAD_VALUE; 2154 ALOGW("createEffect() effect not found"); 2155 goto Exit; 2156 } 2157 // For same effect type, chose auxiliary version over insert version if 2158 // connect to output mix (Compliance to OpenSL ES) 2159 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2160 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2161 desc = d; 2162 } 2163 } 2164 2165 // Do not allow auxiliary effects on a session different from 0 (output mix) 2166 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2167 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2168 lStatus = INVALID_OPERATION; 2169 goto Exit; 2170 } 2171 2172 // check recording permission for visualizer 2173 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2174 !recordingAllowed()) { 2175 lStatus = PERMISSION_DENIED; 2176 goto Exit; 2177 } 2178 2179 // return effect descriptor 2180 *pDesc = desc; 2181 if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2182 // if the output returned by getOutputForEffect() is removed before we lock the 2183 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2184 // and we will exit safely 2185 io = AudioSystem::getOutputForEffect(&desc); 2186 ALOGV("createEffect got output %d", io); 2187 } 2188 2189 Mutex::Autolock _l(mLock); 2190 2191 // If output is not specified try to find a matching audio session ID in one of the 2192 // output threads. 2193 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2194 // because of code checking output when entering the function. 2195 // Note: io is never 0 when creating an effect on an input 2196 if (io == 0) { 2197 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2198 // output must be specified by AudioPolicyManager when using session 2199 // AUDIO_SESSION_OUTPUT_STAGE 2200 lStatus = BAD_VALUE; 2201 goto Exit; 2202 } 2203 // look for the thread where the specified audio session is present 2204 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2205 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2206 io = mPlaybackThreads.keyAt(i); 2207 break; 2208 } 2209 } 2210 if (io == 0) { 2211 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2212 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2213 io = mRecordThreads.keyAt(i); 2214 break; 2215 } 2216 } 2217 } 2218 // If no output thread contains the requested session ID, default to 2219 // first output. The effect chain will be moved to the correct output 2220 // thread when a track with the same session ID is created 2221 if (io == 0 && mPlaybackThreads.size()) { 2222 io = mPlaybackThreads.keyAt(0); 2223 } 2224 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2225 } 2226 ThreadBase *thread = checkRecordThread_l(io); 2227 if (thread == NULL) { 2228 thread = checkPlaybackThread_l(io); 2229 if (thread == NULL) { 2230 ALOGE("createEffect() unknown output thread"); 2231 lStatus = BAD_VALUE; 2232 goto Exit; 2233 } 2234 } 2235 2236 sp<Client> client = registerPid_l(pid); 2237 2238 // create effect on selected output thread 2239 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2240 &desc, enabled, &lStatus); 2241 if (handle != 0 && id != NULL) { 2242 *id = handle->id(); 2243 } 2244 } 2245 2246 Exit: 2247 if (status != NULL) { 2248 *status = lStatus; 2249 } 2250 return handle; 2251 } 2252 2253 status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2254 audio_io_handle_t dstOutput) 2255 { 2256 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2257 sessionId, srcOutput, dstOutput); 2258 Mutex::Autolock _l(mLock); 2259 if (srcOutput == dstOutput) { 2260 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2261 return NO_ERROR; 2262 } 2263 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2264 if (srcThread == NULL) { 2265 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2266 return BAD_VALUE; 2267 } 2268 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2269 if (dstThread == NULL) { 2270 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2271 return BAD_VALUE; 2272 } 2273 2274 Mutex::Autolock _dl(dstThread->mLock); 2275 Mutex::Autolock _sl(srcThread->mLock); 2276 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2277 } 2278 2279 // moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2280 status_t AudioFlinger::moveEffectChain_l(int sessionId, 2281 AudioFlinger::PlaybackThread *srcThread, 2282 AudioFlinger::PlaybackThread *dstThread, 2283 bool reRegister) 2284 { 2285 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2286 sessionId, srcThread, dstThread); 2287 2288 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2289 if (chain == 0) { 2290 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2291 sessionId, srcThread); 2292 return INVALID_OPERATION; 2293 } 2294 2295 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2296 // so that a new chain is created with correct parameters when first effect is added. This is 2297 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2298 // removed. 2299 srcThread->removeEffectChain_l(chain); 2300 2301 // transfer all effects one by one so that new effect chain is created on new thread with 2302 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2303 sp<EffectChain> dstChain; 2304 uint32_t strategy = 0; // prevent compiler warning 2305 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2306 Vector< sp<EffectModule> > removed; 2307 status_t status = NO_ERROR; 2308 while (effect != 0) { 2309 srcThread->removeEffect_l(effect); 2310 removed.add(effect); 2311 status = dstThread->addEffect_l(effect); 2312 if (status != NO_ERROR) { 2313 break; 2314 } 2315 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2316 if (effect->state() == EffectModule::ACTIVE || 2317 effect->state() == EffectModule::STOPPING) { 2318 effect->start(); 2319 } 2320 // if the move request is not received from audio policy manager, the effect must be 2321 // re-registered with the new strategy and output 2322 if (dstChain == 0) { 2323 dstChain = effect->chain().promote(); 2324 if (dstChain == 0) { 2325 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2326 status = NO_INIT; 2327 break; 2328 } 2329 strategy = dstChain->strategy(); 2330 } 2331 if (reRegister) { 2332 AudioSystem::unregisterEffect(effect->id()); 2333 AudioSystem::registerEffect(&effect->desc(), 2334 dstThread->id(), 2335 strategy, 2336 sessionId, 2337 effect->id()); 2338 } 2339 effect = chain->getEffectFromId_l(0); 2340 } 2341 2342 if (status != NO_ERROR) { 2343 for (size_t i = 0; i < removed.size(); i++) { 2344 srcThread->addEffect_l(removed[i]); 2345 if (dstChain != 0 && reRegister) { 2346 AudioSystem::unregisterEffect(removed[i]->id()); 2347 AudioSystem::registerEffect(&removed[i]->desc(), 2348 srcThread->id(), 2349 strategy, 2350 sessionId, 2351 removed[i]->id()); 2352 } 2353 } 2354 } 2355 2356 return status; 2357 } 2358 2359 bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2360 { 2361 if (mGlobalEffectEnableTime != 0 && 2362 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2363 return true; 2364 } 2365 2366 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2367 sp<EffectChain> ec = 2368 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2369 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2370 return true; 2371 } 2372 } 2373 return false; 2374 } 2375 2376 void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2377 { 2378 Mutex::Autolock _l(mLock); 2379 2380 mGlobalEffectEnableTime = systemTime(); 2381 2382 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2383 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2384 if (t->mType == ThreadBase::OFFLOAD) { 2385 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2386 } 2387 } 2388 2389 } 2390 2391 struct Entry { 2392 #define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2393 char mName[MAX_NAME]; 2394 }; 2395 2396 int comparEntry(const void *p1, const void *p2) 2397 { 2398 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2399 } 2400 2401 #ifdef TEE_SINK 2402 void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2403 { 2404 NBAIO_Source *teeSource = source.get(); 2405 if (teeSource != NULL) { 2406 // .wav rotation 2407 // There is a benign race condition if 2 threads call this simultaneously. 2408 // They would both traverse the directory, but the result would simply be 2409 // failures at unlink() which are ignored. It's also unlikely since 2410 // normally dumpsys is only done by bugreport or from the command line. 2411 char teePath[32+256]; 2412 strcpy(teePath, "/data/misc/media"); 2413 size_t teePathLen = strlen(teePath); 2414 DIR *dir = opendir(teePath); 2415 teePath[teePathLen++] = '/'; 2416 if (dir != NULL) { 2417 #define MAX_SORT 20 // number of entries to sort 2418 #define MAX_KEEP 10 // number of entries to keep 2419 struct Entry entries[MAX_SORT]; 2420 size_t entryCount = 0; 2421 while (entryCount < MAX_SORT) { 2422 struct dirent de; 2423 struct dirent *result = NULL; 2424 int rc = readdir_r(dir, &de, &result); 2425 if (rc != 0) { 2426 ALOGW("readdir_r failed %d", rc); 2427 break; 2428 } 2429 if (result == NULL) { 2430 break; 2431 } 2432 if (result != &de) { 2433 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2434 break; 2435 } 2436 // ignore non .wav file entries 2437 size_t nameLen = strlen(de.d_name); 2438 if (nameLen <= 4 || nameLen >= MAX_NAME || 2439 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2440 continue; 2441 } 2442 strcpy(entries[entryCount++].mName, de.d_name); 2443 } 2444 (void) closedir(dir); 2445 if (entryCount > MAX_KEEP) { 2446 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2447 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2448 strcpy(&teePath[teePathLen], entries[i].mName); 2449 (void) unlink(teePath); 2450 } 2451 } 2452 } else { 2453 if (fd >= 0) { 2454 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2455 } 2456 } 2457 char teeTime[16]; 2458 struct timeval tv; 2459 gettimeofday(&tv, NULL); 2460 struct tm tm; 2461 localtime_r(&tv.tv_sec, &tm); 2462 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2463 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2464 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2465 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2466 if (teeFd >= 0) { 2467 char wavHeader[44]; 2468 memcpy(wavHeader, 2469 "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", 2470 sizeof(wavHeader)); 2471 NBAIO_Format format = teeSource->format(); 2472 unsigned channelCount = Format_channelCount(format); 2473 ALOG_ASSERT(channelCount <= FCC_2); 2474 uint32_t sampleRate = Format_sampleRate(format); 2475 wavHeader[22] = channelCount; // number of channels 2476 wavHeader[24] = sampleRate; // sample rate 2477 wavHeader[25] = sampleRate >> 8; 2478 wavHeader[32] = channelCount * 2; // block alignment 2479 write(teeFd, wavHeader, sizeof(wavHeader)); 2480 size_t total = 0; 2481 bool firstRead = true; 2482 for (;;) { 2483 #define TEE_SINK_READ 1024 2484 short buffer[TEE_SINK_READ * FCC_2]; 2485 size_t count = TEE_SINK_READ; 2486 ssize_t actual = teeSource->read(buffer, count, 2487 AudioBufferProvider::kInvalidPTS); 2488 bool wasFirstRead = firstRead; 2489 firstRead = false; 2490 if (actual <= 0) { 2491 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2492 continue; 2493 } 2494 break; 2495 } 2496 ALOG_ASSERT(actual <= (ssize_t)count); 2497 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2498 total += actual; 2499 } 2500 lseek(teeFd, (off_t) 4, SEEK_SET); 2501 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2502 write(teeFd, &temp, sizeof(temp)); 2503 lseek(teeFd, (off_t) 40, SEEK_SET); 2504 temp = total * channelCount * sizeof(short); 2505 write(teeFd, &temp, sizeof(temp)); 2506 close(teeFd); 2507 if (fd >= 0) { 2508 fdprintf(fd, "tee copied to %s\n", teePath); 2509 } 2510 } else { 2511 if (fd >= 0) { 2512 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2513 } 2514 } 2515 } 2516 } 2517 #endif 2518 2519 // ---------------------------------------------------------------------------- 2520 2521 status_t AudioFlinger::onTransact( 2522 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2523 { 2524 return BnAudioFlinger::onTransact(code, data, reply, flags); 2525 } 2526 2527 }; // namespace android 2528