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 <dirent.h> 23 #include <math.h> 24 #include <signal.h> 25 #include <sys/time.h> 26 #include <sys/resource.h> 27 28 #include <binder/IPCThreadState.h> 29 #include <binder/IServiceManager.h> 30 #include <utils/Log.h> 31 #include <utils/Trace.h> 32 #include <binder/Parcel.h> 33 #include <utils/String16.h> 34 #include <utils/threads.h> 35 #include <utils/Atomic.h> 36 37 #include <cutils/bitops.h> 38 #include <cutils/properties.h> 39 #include <cutils/compiler.h> 40 41 //#include <private/media/AudioTrackShared.h> 42 //#include <private/media/AudioEffectShared.h> 43 44 #include <system/audio.h> 45 #include <hardware/audio.h> 46 47 #include "AudioMixer.h" 48 #include "AudioFlinger.h" 49 #include "ServiceUtilities.h" 50 51 #include <media/EffectsFactoryApi.h> 52 #include <audio_effects/effect_visualizer.h> 53 #include <audio_effects/effect_ns.h> 54 #include <audio_effects/effect_aec.h> 55 56 #include <audio_utils/primitives.h> 57 58 #include <powermanager/PowerManager.h> 59 60 #include <common_time/cc_helper.h> 61 //#include <common_time/local_clock.h> 62 63 #include <media/IMediaLogService.h> 64 65 #include <media/nbaio/Pipe.h> 66 #include <media/nbaio/PipeReader.h> 67 68 // ---------------------------------------------------------------------------- 69 70 // Note: the following macro is used for extremely verbose logging message. In 71 // order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72 // 0; but one side effect of this is to turn all LOGV's as well. Some messages 73 // are so verbose that we want to suppress them even when we have ALOG_ASSERT 74 // turned on. Do not uncomment the #def below unless you really know what you 75 // are doing and want to see all of the extremely verbose messages. 76 //#define VERY_VERY_VERBOSE_LOGGING 77 #ifdef VERY_VERY_VERBOSE_LOGGING 78 #define ALOGVV ALOGV 79 #else 80 #define ALOGVV(a...) do { } while(0) 81 #endif 82 83 namespace android { 84 85 static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86 static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87 88 89 nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 90 91 uint32_t AudioFlinger::mScreenState; 92 93 #ifdef TEE_SINK 94 bool AudioFlinger::mTeeSinkInputEnabled = false; 95 bool AudioFlinger::mTeeSinkOutputEnabled = false; 96 bool AudioFlinger::mTeeSinkTrackEnabled = false; 97 98 size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 99 size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 100 size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 101 #endif 102 103 // ---------------------------------------------------------------------------- 104 105 static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 106 { 107 const hw_module_t *mod; 108 int rc; 109 110 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 111 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 112 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 113 if (rc) { 114 goto out; 115 } 116 rc = audio_hw_device_open(mod, dev); 117 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 118 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 119 if (rc) { 120 goto out; 121 } 122 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 123 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 124 rc = BAD_VALUE; 125 goto out; 126 } 127 return 0; 128 129 out: 130 *dev = NULL; 131 return rc; 132 } 133 134 // ---------------------------------------------------------------------------- 135 136 AudioFlinger::AudioFlinger() 137 : BnAudioFlinger(), 138 mPrimaryHardwareDev(NULL), 139 mHardwareStatus(AUDIO_HW_IDLE), 140 mMasterVolume(1.0f), 141 mMasterMute(false), 142 mNextUniqueId(1), 143 mMode(AUDIO_MODE_INVALID), 144 mBtNrecIsOff(false) 145 { 146 getpid_cached = getpid(); 147 char value[PROPERTY_VALUE_MAX]; 148 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 149 if (doLog) { 150 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters"); 151 } 152 #ifdef TEE_SINK 153 (void) property_get("ro.debuggable", value, "0"); 154 int debuggable = atoi(value); 155 int teeEnabled = 0; 156 if (debuggable) { 157 (void) property_get("af.tee", value, "0"); 158 teeEnabled = atoi(value); 159 } 160 if (teeEnabled & 1) 161 mTeeSinkInputEnabled = true; 162 if (teeEnabled & 2) 163 mTeeSinkOutputEnabled = true; 164 if (teeEnabled & 4) 165 mTeeSinkTrackEnabled = true; 166 #endif 167 } 168 169 void AudioFlinger::onFirstRef() 170 { 171 int rc = 0; 172 173 Mutex::Autolock _l(mLock); 174 175 /* TODO: move all this work into an Init() function */ 176 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 177 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 178 uint32_t int_val; 179 if (1 == sscanf(val_str, "%u", &int_val)) { 180 mStandbyTimeInNsecs = milliseconds(int_val); 181 ALOGI("Using %u mSec as standby time.", int_val); 182 } else { 183 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 184 ALOGI("Using default %u mSec as standby time.", 185 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 186 } 187 } 188 189 mMode = AUDIO_MODE_NORMAL; 190 } 191 192 AudioFlinger::~AudioFlinger() 193 { 194 while (!mRecordThreads.isEmpty()) { 195 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 196 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 197 } 198 while (!mPlaybackThreads.isEmpty()) { 199 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 200 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 201 } 202 203 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 204 // no mHardwareLock needed, as there are no other references to this 205 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 206 delete mAudioHwDevs.valueAt(i); 207 } 208 } 209 210 static const char * const audio_interfaces[] = { 211 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 212 AUDIO_HARDWARE_MODULE_ID_A2DP, 213 AUDIO_HARDWARE_MODULE_ID_USB, 214 }; 215 #define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 216 217 AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 218 audio_module_handle_t module, 219 audio_devices_t devices) 220 { 221 // if module is 0, the request comes from an old policy manager and we should load 222 // well known modules 223 if (module == 0) { 224 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 225 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 226 loadHwModule_l(audio_interfaces[i]); 227 } 228 // then try to find a module supporting the requested device. 229 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 230 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 231 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 232 if ((dev->get_supported_devices != NULL) && 233 (dev->get_supported_devices(dev) & devices) == devices) 234 return audioHwDevice; 235 } 236 } else { 237 // check a match for the requested module handle 238 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 239 if (audioHwDevice != NULL) { 240 return audioHwDevice; 241 } 242 } 243 244 return NULL; 245 } 246 247 void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 248 { 249 const size_t SIZE = 256; 250 char buffer[SIZE]; 251 String8 result; 252 253 result.append("Clients:\n"); 254 for (size_t i = 0; i < mClients.size(); ++i) { 255 sp<Client> client = mClients.valueAt(i).promote(); 256 if (client != 0) { 257 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 258 result.append(buffer); 259 } 260 } 261 262 result.append("Global session refs:\n"); 263 result.append(" session pid count\n"); 264 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 265 AudioSessionRef *r = mAudioSessionRefs[i]; 266 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 267 result.append(buffer); 268 } 269 write(fd, result.string(), result.size()); 270 } 271 272 273 void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 274 { 275 const size_t SIZE = 256; 276 char buffer[SIZE]; 277 String8 result; 278 hardware_call_state hardwareStatus = mHardwareStatus; 279 280 snprintf(buffer, SIZE, "Hardware status: %d\n" 281 "Standby Time mSec: %u\n", 282 hardwareStatus, 283 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 284 result.append(buffer); 285 write(fd, result.string(), result.size()); 286 } 287 288 void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 289 { 290 const size_t SIZE = 256; 291 char buffer[SIZE]; 292 String8 result; 293 snprintf(buffer, SIZE, "Permission Denial: " 294 "can't dump AudioFlinger from pid=%d, uid=%d\n", 295 IPCThreadState::self()->getCallingPid(), 296 IPCThreadState::self()->getCallingUid()); 297 result.append(buffer); 298 write(fd, result.string(), result.size()); 299 } 300 301 bool AudioFlinger::dumpTryLock(Mutex& mutex) 302 { 303 bool locked = false; 304 for (int i = 0; i < kDumpLockRetries; ++i) { 305 if (mutex.tryLock() == NO_ERROR) { 306 locked = true; 307 break; 308 } 309 usleep(kDumpLockSleepUs); 310 } 311 return locked; 312 } 313 314 status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 315 { 316 if (!dumpAllowed()) { 317 dumpPermissionDenial(fd, args); 318 } else { 319 // get state of hardware lock 320 bool hardwareLocked = dumpTryLock(mHardwareLock); 321 if (!hardwareLocked) { 322 String8 result(kHardwareLockedString); 323 write(fd, result.string(), result.size()); 324 } else { 325 mHardwareLock.unlock(); 326 } 327 328 bool locked = dumpTryLock(mLock); 329 330 // failed to lock - AudioFlinger is probably deadlocked 331 if (!locked) { 332 String8 result(kDeadlockedString); 333 write(fd, result.string(), result.size()); 334 } 335 336 dumpClients(fd, args); 337 dumpInternals(fd, args); 338 339 // dump playback threads 340 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 341 mPlaybackThreads.valueAt(i)->dump(fd, args); 342 } 343 344 // dump record threads 345 for (size_t i = 0; i < mRecordThreads.size(); i++) { 346 mRecordThreads.valueAt(i)->dump(fd, args); 347 } 348 349 // dump all hardware devs 350 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 351 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 352 dev->dump(dev, fd); 353 } 354 355 #ifdef TEE_SINK 356 // dump the serially shared record tee sink 357 if (mRecordTeeSource != 0) { 358 dumpTee(fd, mRecordTeeSource); 359 } 360 #endif 361 362 if (locked) { 363 mLock.unlock(); 364 } 365 366 // append a copy of media.log here by forwarding fd to it, but don't attempt 367 // to lookup the service if it's not running, as it will block for a second 368 if (mLogMemoryDealer != 0) { 369 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 370 if (binder != 0) { 371 fdprintf(fd, "\nmedia.log:\n"); 372 Vector<String16> args; 373 binder->dump(fd, args); 374 } 375 } 376 } 377 return NO_ERROR; 378 } 379 380 sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 381 { 382 // If pid is already in the mClients wp<> map, then use that entry 383 // (for which promote() is always != 0), otherwise create a new entry and Client. 384 sp<Client> client = mClients.valueFor(pid).promote(); 385 if (client == 0) { 386 client = new Client(this, pid); 387 mClients.add(pid, client); 388 } 389 390 return client; 391 } 392 393 sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 394 { 395 if (mLogMemoryDealer == 0) { 396 return new NBLog::Writer(); 397 } 398 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 399 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 400 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 401 if (binder != 0) { 402 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 403 } 404 return writer; 405 } 406 407 void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 408 { 409 if (writer == 0) { 410 return; 411 } 412 sp<IMemory> iMemory(writer->getIMemory()); 413 if (iMemory == 0) { 414 return; 415 } 416 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 417 if (binder != 0) { 418 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 419 // Now the media.log remote reference to IMemory is gone. 420 // When our last local reference to IMemory also drops to zero, 421 // the IMemory destructor will deallocate the region from mMemoryDealer. 422 } 423 } 424 425 // IAudioFlinger interface 426 427 428 sp<IAudioTrack> AudioFlinger::createTrack( 429 audio_stream_type_t streamType, 430 uint32_t sampleRate, 431 audio_format_t format, 432 audio_channel_mask_t channelMask, 433 size_t frameCount, 434 IAudioFlinger::track_flags_t *flags, 435 const sp<IMemory>& sharedBuffer, 436 audio_io_handle_t output, 437 pid_t tid, 438 int *sessionId, 439 status_t *status) 440 { 441 sp<PlaybackThread::Track> track; 442 sp<TrackHandle> trackHandle; 443 sp<Client> client; 444 status_t lStatus; 445 int lSessionId; 446 447 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 448 // but if someone uses binder directly they could bypass that and cause us to crash 449 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 450 ALOGE("createTrack() invalid stream type %d", streamType); 451 lStatus = BAD_VALUE; 452 goto Exit; 453 } 454 455 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 456 // and we don't yet support 8.24 or 32-bit PCM 457 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 458 ALOGE("createTrack() invalid format %d", format); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("no playback thread found for output handle %d", output); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 pid_t pid = IPCThreadState::self()->getCallingPid(); 474 client = registerPid_l(pid); 475 476 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 477 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 478 // check if an effect chain with the same session ID is present on another 479 // output thread and move it here. 480 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 481 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 482 if (mPlaybackThreads.keyAt(i) != output) { 483 uint32_t sessions = t->hasAudioSession(*sessionId); 484 if (sessions & PlaybackThread::EFFECT_SESSION) { 485 effectThread = t.get(); 486 break; 487 } 488 } 489 } 490 lSessionId = *sessionId; 491 } else { 492 // if no audio session id is provided, create one here 493 lSessionId = nextUniqueId(); 494 if (sessionId != NULL) { 495 *sessionId = lSessionId; 496 } 497 } 498 ALOGV("createTrack() lSessionId: %d", lSessionId); 499 500 track = thread->createTrack_l(client, streamType, sampleRate, format, 501 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 502 503 // move effect chain to this output thread if an effect on same session was waiting 504 // for a track to be created 505 if (lStatus == NO_ERROR && effectThread != NULL) { 506 Mutex::Autolock _dl(thread->mLock); 507 Mutex::Autolock _sl(effectThread->mLock); 508 moveEffectChain_l(lSessionId, effectThread, thread, true); 509 } 510 511 // Look for sync events awaiting for a session to be used. 512 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 513 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 514 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 515 if (lStatus == NO_ERROR) { 516 (void) track->setSyncEvent(mPendingSyncEvents[i]); 517 } else { 518 mPendingSyncEvents[i]->cancel(); 519 } 520 mPendingSyncEvents.removeAt(i); 521 i--; 522 } 523 } 524 } 525 } 526 if (lStatus == NO_ERROR) { 527 trackHandle = new TrackHandle(track); 528 } else { 529 // remove local strong reference to Client before deleting the Track so that the Client 530 // destructor is called by the TrackBase destructor with mLock held 531 client.clear(); 532 track.clear(); 533 } 534 535 Exit: 536 if (status != NULL) { 537 *status = lStatus; 538 } 539 return trackHandle; 540 } 541 542 uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 543 { 544 Mutex::Autolock _l(mLock); 545 PlaybackThread *thread = checkPlaybackThread_l(output); 546 if (thread == NULL) { 547 ALOGW("sampleRate() unknown thread %d", output); 548 return 0; 549 } 550 return thread->sampleRate(); 551 } 552 553 int AudioFlinger::channelCount(audio_io_handle_t output) const 554 { 555 Mutex::Autolock _l(mLock); 556 PlaybackThread *thread = checkPlaybackThread_l(output); 557 if (thread == NULL) { 558 ALOGW("channelCount() unknown thread %d", output); 559 return 0; 560 } 561 return thread->channelCount(); 562 } 563 564 audio_format_t AudioFlinger::format(audio_io_handle_t output) const 565 { 566 Mutex::Autolock _l(mLock); 567 PlaybackThread *thread = checkPlaybackThread_l(output); 568 if (thread == NULL) { 569 ALOGW("format() unknown thread %d", output); 570 return AUDIO_FORMAT_INVALID; 571 } 572 return thread->format(); 573 } 574 575 size_t AudioFlinger::frameCount(audio_io_handle_t output) const 576 { 577 Mutex::Autolock _l(mLock); 578 PlaybackThread *thread = checkPlaybackThread_l(output); 579 if (thread == NULL) { 580 ALOGW("frameCount() unknown thread %d", output); 581 return 0; 582 } 583 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 584 // should examine all callers and fix them to handle smaller counts 585 return thread->frameCount(); 586 } 587 588 uint32_t AudioFlinger::latency(audio_io_handle_t output) const 589 { 590 Mutex::Autolock _l(mLock); 591 PlaybackThread *thread = checkPlaybackThread_l(output); 592 if (thread == NULL) { 593 ALOGW("latency(): no playback thread found for output handle %d", output); 594 return 0; 595 } 596 return thread->latency(); 597 } 598 599 status_t AudioFlinger::setMasterVolume(float value) 600 { 601 status_t ret = initCheck(); 602 if (ret != NO_ERROR) { 603 return ret; 604 } 605 606 // check calling permissions 607 if (!settingsAllowed()) { 608 return PERMISSION_DENIED; 609 } 610 611 Mutex::Autolock _l(mLock); 612 mMasterVolume = value; 613 614 // Set master volume in the HALs which support it. 615 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 616 AutoMutex lock(mHardwareLock); 617 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 618 619 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 620 if (dev->canSetMasterVolume()) { 621 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 622 } 623 mHardwareStatus = AUDIO_HW_IDLE; 624 } 625 626 // Now set the master volume in each playback thread. Playback threads 627 // assigned to HALs which do not have master volume support will apply 628 // master volume during the mix operation. Threads with HALs which do 629 // support master volume will simply ignore the setting. 630 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 631 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 632 633 return NO_ERROR; 634 } 635 636 status_t AudioFlinger::setMode(audio_mode_t mode) 637 { 638 status_t ret = initCheck(); 639 if (ret != NO_ERROR) { 640 return ret; 641 } 642 643 // check calling permissions 644 if (!settingsAllowed()) { 645 return PERMISSION_DENIED; 646 } 647 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 648 ALOGW("Illegal value: setMode(%d)", mode); 649 return BAD_VALUE; 650 } 651 652 { // scope for the lock 653 AutoMutex lock(mHardwareLock); 654 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 655 mHardwareStatus = AUDIO_HW_SET_MODE; 656 ret = dev->set_mode(dev, mode); 657 mHardwareStatus = AUDIO_HW_IDLE; 658 } 659 660 if (NO_ERROR == ret) { 661 Mutex::Autolock _l(mLock); 662 mMode = mode; 663 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 664 mPlaybackThreads.valueAt(i)->setMode(mode); 665 } 666 667 return ret; 668 } 669 670 status_t AudioFlinger::setMicMute(bool state) 671 { 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return ret; 675 } 676 677 // check calling permissions 678 if (!settingsAllowed()) { 679 return PERMISSION_DENIED; 680 } 681 682 AutoMutex lock(mHardwareLock); 683 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = dev->set_mic_mute(dev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688 } 689 690 bool AudioFlinger::getMicMute() const 691 { 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 700 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 701 dev->get_mic_mute(dev, &state); 702 mHardwareStatus = AUDIO_HW_IDLE; 703 return state; 704 } 705 706 status_t AudioFlinger::setMasterMute(bool muted) 707 { 708 status_t ret = initCheck(); 709 if (ret != NO_ERROR) { 710 return ret; 711 } 712 713 // check calling permissions 714 if (!settingsAllowed()) { 715 return PERMISSION_DENIED; 716 } 717 718 Mutex::Autolock _l(mLock); 719 mMasterMute = muted; 720 721 // Set master mute in the HALs which support it. 722 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 723 AutoMutex lock(mHardwareLock); 724 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 725 726 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 727 if (dev->canSetMasterMute()) { 728 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 729 } 730 mHardwareStatus = AUDIO_HW_IDLE; 731 } 732 733 // Now set the master mute in each playback thread. Playback threads 734 // assigned to HALs which do not have master mute support will apply master 735 // mute during the mix operation. Threads with HALs which do support master 736 // mute will simply ignore the setting. 737 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 738 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 739 740 return NO_ERROR; 741 } 742 743 float AudioFlinger::masterVolume() const 744 { 745 Mutex::Autolock _l(mLock); 746 return masterVolume_l(); 747 } 748 749 bool AudioFlinger::masterMute() const 750 { 751 Mutex::Autolock _l(mLock); 752 return masterMute_l(); 753 } 754 755 float AudioFlinger::masterVolume_l() const 756 { 757 return mMasterVolume; 758 } 759 760 bool AudioFlinger::masterMute_l() const 761 { 762 return mMasterMute; 763 } 764 765 status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 766 audio_io_handle_t output) 767 { 768 // check calling permissions 769 if (!settingsAllowed()) { 770 return PERMISSION_DENIED; 771 } 772 773 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 774 ALOGE("setStreamVolume() invalid stream %d", stream); 775 return BAD_VALUE; 776 } 777 778 AutoMutex lock(mLock); 779 PlaybackThread *thread = NULL; 780 if (output) { 781 thread = checkPlaybackThread_l(output); 782 if (thread == NULL) { 783 return BAD_VALUE; 784 } 785 } 786 787 mStreamTypes[stream].volume = value; 788 789 if (thread == NULL) { 790 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 791 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 792 } 793 } else { 794 thread->setStreamVolume(stream, value); 795 } 796 797 return NO_ERROR; 798 } 799 800 status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 801 { 802 // check calling permissions 803 if (!settingsAllowed()) { 804 return PERMISSION_DENIED; 805 } 806 807 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 808 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 809 ALOGE("setStreamMute() invalid stream %d", stream); 810 return BAD_VALUE; 811 } 812 813 AutoMutex lock(mLock); 814 mStreamTypes[stream].mute = muted; 815 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 816 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 817 818 return NO_ERROR; 819 } 820 821 float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 822 { 823 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 824 return 0.0f; 825 } 826 827 AutoMutex lock(mLock); 828 float volume; 829 if (output) { 830 PlaybackThread *thread = checkPlaybackThread_l(output); 831 if (thread == NULL) { 832 return 0.0f; 833 } 834 volume = thread->streamVolume(stream); 835 } else { 836 volume = streamVolume_l(stream); 837 } 838 839 return volume; 840 } 841 842 bool AudioFlinger::streamMute(audio_stream_type_t stream) const 843 { 844 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 845 return true; 846 } 847 848 AutoMutex lock(mLock); 849 return streamMute_l(stream); 850 } 851 852 status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 853 { 854 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 855 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 856 857 // check calling permissions 858 if (!settingsAllowed()) { 859 return PERMISSION_DENIED; 860 } 861 862 // ioHandle == 0 means the parameters are global to the audio hardware interface 863 if (ioHandle == 0) { 864 Mutex::Autolock _l(mLock); 865 status_t final_result = NO_ERROR; 866 { 867 AutoMutex lock(mHardwareLock); 868 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 869 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 870 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 871 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 872 final_result = result ?: final_result; 873 } 874 mHardwareStatus = AUDIO_HW_IDLE; 875 } 876 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 877 AudioParameter param = AudioParameter(keyValuePairs); 878 String8 value; 879 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 880 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 881 if (mBtNrecIsOff != btNrecIsOff) { 882 for (size_t i = 0; i < mRecordThreads.size(); i++) { 883 sp<RecordThread> thread = mRecordThreads.valueAt(i); 884 audio_devices_t device = thread->inDevice(); 885 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 886 // collect all of the thread's session IDs 887 KeyedVector<int, bool> ids = thread->sessionIds(); 888 // suspend effects associated with those session IDs 889 for (size_t j = 0; j < ids.size(); ++j) { 890 int sessionId = ids.keyAt(j); 891 thread->setEffectSuspended(FX_IID_AEC, 892 suspend, 893 sessionId); 894 thread->setEffectSuspended(FX_IID_NS, 895 suspend, 896 sessionId); 897 } 898 } 899 mBtNrecIsOff = btNrecIsOff; 900 } 901 } 902 String8 screenState; 903 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 904 bool isOff = screenState == "off"; 905 if (isOff != (AudioFlinger::mScreenState & 1)) { 906 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 907 } 908 } 909 return final_result; 910 } 911 912 // hold a strong ref on thread in case closeOutput() or closeInput() is called 913 // and the thread is exited once the lock is released 914 sp<ThreadBase> thread; 915 { 916 Mutex::Autolock _l(mLock); 917 thread = checkPlaybackThread_l(ioHandle); 918 if (thread == 0) { 919 thread = checkRecordThread_l(ioHandle); 920 } else if (thread == primaryPlaybackThread_l()) { 921 // indicate output device change to all input threads for pre processing 922 AudioParameter param = AudioParameter(keyValuePairs); 923 int value; 924 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 925 (value != 0)) { 926 for (size_t i = 0; i < mRecordThreads.size(); i++) { 927 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 928 } 929 } 930 } 931 } 932 if (thread != 0) { 933 return thread->setParameters(keyValuePairs); 934 } 935 return BAD_VALUE; 936 } 937 938 String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 939 { 940 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 941 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 942 943 Mutex::Autolock _l(mLock); 944 945 if (ioHandle == 0) { 946 String8 out_s8; 947 948 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 949 char *s; 950 { 951 AutoMutex lock(mHardwareLock); 952 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 953 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 954 s = dev->get_parameters(dev, keys.string()); 955 mHardwareStatus = AUDIO_HW_IDLE; 956 } 957 out_s8 += String8(s ? s : ""); 958 free(s); 959 } 960 return out_s8; 961 } 962 963 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 964 if (playbackThread != NULL) { 965 return playbackThread->getParameters(keys); 966 } 967 RecordThread *recordThread = checkRecordThread_l(ioHandle); 968 if (recordThread != NULL) { 969 return recordThread->getParameters(keys); 970 } 971 return String8(""); 972 } 973 974 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 975 audio_channel_mask_t channelMask) const 976 { 977 status_t ret = initCheck(); 978 if (ret != NO_ERROR) { 979 return 0; 980 } 981 982 AutoMutex lock(mHardwareLock); 983 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 984 struct audio_config config = { 985 sample_rate: sampleRate, 986 channel_mask: channelMask, 987 format: format, 988 }; 989 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 990 size_t size = dev->get_input_buffer_size(dev, &config); 991 mHardwareStatus = AUDIO_HW_IDLE; 992 return size; 993 } 994 995 unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 996 { 997 Mutex::Autolock _l(mLock); 998 999 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1000 if (recordThread != NULL) { 1001 return recordThread->getInputFramesLost(); 1002 } 1003 return 0; 1004 } 1005 1006 status_t AudioFlinger::setVoiceVolume(float value) 1007 { 1008 status_t ret = initCheck(); 1009 if (ret != NO_ERROR) { 1010 return ret; 1011 } 1012 1013 // check calling permissions 1014 if (!settingsAllowed()) { 1015 return PERMISSION_DENIED; 1016 } 1017 1018 AutoMutex lock(mHardwareLock); 1019 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1020 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1021 ret = dev->set_voice_volume(dev, value); 1022 mHardwareStatus = AUDIO_HW_IDLE; 1023 1024 return ret; 1025 } 1026 1027 status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1028 audio_io_handle_t output) const 1029 { 1030 status_t status; 1031 1032 Mutex::Autolock _l(mLock); 1033 1034 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1035 if (playbackThread != NULL) { 1036 return playbackThread->getRenderPosition(halFrames, dspFrames); 1037 } 1038 1039 return BAD_VALUE; 1040 } 1041 1042 void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1043 { 1044 1045 Mutex::Autolock _l(mLock); 1046 1047 pid_t pid = IPCThreadState::self()->getCallingPid(); 1048 if (mNotificationClients.indexOfKey(pid) < 0) { 1049 sp<NotificationClient> notificationClient = new NotificationClient(this, 1050 client, 1051 pid); 1052 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1053 1054 mNotificationClients.add(pid, notificationClient); 1055 1056 sp<IBinder> binder = client->asBinder(); 1057 binder->linkToDeath(notificationClient); 1058 1059 // the config change is always sent from playback or record threads to avoid deadlock 1060 // with AudioSystem::gLock 1061 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1062 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1063 } 1064 1065 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1066 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1067 } 1068 } 1069 } 1070 1071 void AudioFlinger::removeNotificationClient(pid_t pid) 1072 { 1073 Mutex::Autolock _l(mLock); 1074 1075 mNotificationClients.removeItem(pid); 1076 1077 ALOGV("%d died, releasing its sessions", pid); 1078 size_t num = mAudioSessionRefs.size(); 1079 bool removed = false; 1080 for (size_t i = 0; i< num; ) { 1081 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1082 ALOGV(" pid %d @ %d", ref->mPid, i); 1083 if (ref->mPid == pid) { 1084 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1085 mAudioSessionRefs.removeAt(i); 1086 delete ref; 1087 removed = true; 1088 num--; 1089 } else { 1090 i++; 1091 } 1092 } 1093 if (removed) { 1094 purgeStaleEffects_l(); 1095 } 1096 } 1097 1098 // audioConfigChanged_l() must be called with AudioFlinger::mLock held 1099 void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1100 { 1101 size_t size = mNotificationClients.size(); 1102 for (size_t i = 0; i < size; i++) { 1103 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1104 param2); 1105 } 1106 } 1107 1108 // removeClient_l() must be called with AudioFlinger::mLock held 1109 void AudioFlinger::removeClient_l(pid_t pid) 1110 { 1111 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1112 IPCThreadState::self()->getCallingPid()); 1113 mClients.removeItem(pid); 1114 } 1115 1116 // getEffectThread_l() must be called with AudioFlinger::mLock held 1117 sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1118 { 1119 sp<PlaybackThread> thread; 1120 1121 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1122 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1123 ALOG_ASSERT(thread == 0); 1124 thread = mPlaybackThreads.valueAt(i); 1125 } 1126 } 1127 1128 return thread; 1129 } 1130 1131 1132 1133 // ---------------------------------------------------------------------------- 1134 1135 AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1136 : RefBase(), 1137 mAudioFlinger(audioFlinger), 1138 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1139 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1140 mPid(pid), 1141 mTimedTrackCount(0) 1142 { 1143 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1144 } 1145 1146 // Client destructor must be called with AudioFlinger::mLock held 1147 AudioFlinger::Client::~Client() 1148 { 1149 mAudioFlinger->removeClient_l(mPid); 1150 } 1151 1152 sp<MemoryDealer> AudioFlinger::Client::heap() const 1153 { 1154 return mMemoryDealer; 1155 } 1156 1157 // Reserve one of the limited slots for a timed audio track associated 1158 // with this client 1159 bool AudioFlinger::Client::reserveTimedTrack() 1160 { 1161 const int kMaxTimedTracksPerClient = 4; 1162 1163 Mutex::Autolock _l(mTimedTrackLock); 1164 1165 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1166 ALOGW("can not create timed track - pid %d has exceeded the limit", 1167 mPid); 1168 return false; 1169 } 1170 1171 mTimedTrackCount++; 1172 return true; 1173 } 1174 1175 // Release a slot for a timed audio track 1176 void AudioFlinger::Client::releaseTimedTrack() 1177 { 1178 Mutex::Autolock _l(mTimedTrackLock); 1179 mTimedTrackCount--; 1180 } 1181 1182 // ---------------------------------------------------------------------------- 1183 1184 AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1185 const sp<IAudioFlingerClient>& client, 1186 pid_t pid) 1187 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1188 { 1189 } 1190 1191 AudioFlinger::NotificationClient::~NotificationClient() 1192 { 1193 } 1194 1195 void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1196 { 1197 sp<NotificationClient> keep(this); 1198 mAudioFlinger->removeNotificationClient(mPid); 1199 } 1200 1201 1202 // ---------------------------------------------------------------------------- 1203 1204 sp<IAudioRecord> AudioFlinger::openRecord( 1205 audio_io_handle_t input, 1206 uint32_t sampleRate, 1207 audio_format_t format, 1208 audio_channel_mask_t channelMask, 1209 size_t frameCount, 1210 IAudioFlinger::track_flags_t flags, 1211 pid_t tid, 1212 int *sessionId, 1213 status_t *status) 1214 { 1215 sp<RecordThread::RecordTrack> recordTrack; 1216 sp<RecordHandle> recordHandle; 1217 sp<Client> client; 1218 status_t lStatus; 1219 RecordThread *thread; 1220 size_t inFrameCount; 1221 int lSessionId; 1222 1223 // check calling permissions 1224 if (!recordingAllowed()) { 1225 lStatus = PERMISSION_DENIED; 1226 goto Exit; 1227 } 1228 1229 // add client to list 1230 { // scope for mLock 1231 Mutex::Autolock _l(mLock); 1232 thread = checkRecordThread_l(input); 1233 if (thread == NULL) { 1234 lStatus = BAD_VALUE; 1235 goto Exit; 1236 } 1237 1238 pid_t pid = IPCThreadState::self()->getCallingPid(); 1239 client = registerPid_l(pid); 1240 1241 // If no audio session id is provided, create one here 1242 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1243 lSessionId = *sessionId; 1244 } else { 1245 lSessionId = nextUniqueId(); 1246 if (sessionId != NULL) { 1247 *sessionId = lSessionId; 1248 } 1249 } 1250 // create new record track. 1251 // The record track uses one track in mHardwareMixerThread by convention. 1252 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1253 frameCount, lSessionId, flags, tid, &lStatus); 1254 } 1255 if (lStatus != NO_ERROR) { 1256 // remove local strong reference to Client before deleting the RecordTrack so that the 1257 // Client destructor is called by the TrackBase destructor with mLock held 1258 client.clear(); 1259 recordTrack.clear(); 1260 goto Exit; 1261 } 1262 1263 // return to handle to client 1264 recordHandle = new RecordHandle(recordTrack); 1265 lStatus = NO_ERROR; 1266 1267 Exit: 1268 if (status) { 1269 *status = lStatus; 1270 } 1271 return recordHandle; 1272 } 1273 1274 1275 1276 // ---------------------------------------------------------------------------- 1277 1278 audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1279 { 1280 if (!settingsAllowed()) { 1281 return 0; 1282 } 1283 Mutex::Autolock _l(mLock); 1284 return loadHwModule_l(name); 1285 } 1286 1287 // loadHwModule_l() must be called with AudioFlinger::mLock held 1288 audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1289 { 1290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1291 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1292 ALOGW("loadHwModule() module %s already loaded", name); 1293 return mAudioHwDevs.keyAt(i); 1294 } 1295 } 1296 1297 audio_hw_device_t *dev; 1298 1299 int rc = load_audio_interface(name, &dev); 1300 if (rc) { 1301 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1302 return 0; 1303 } 1304 1305 mHardwareStatus = AUDIO_HW_INIT; 1306 rc = dev->init_check(dev); 1307 mHardwareStatus = AUDIO_HW_IDLE; 1308 if (rc) { 1309 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1310 return 0; 1311 } 1312 1313 // Check and cache this HAL's level of support for master mute and master 1314 // volume. If this is the first HAL opened, and it supports the get 1315 // methods, use the initial values provided by the HAL as the current 1316 // master mute and volume settings. 1317 1318 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1319 { // scope for auto-lock pattern 1320 AutoMutex lock(mHardwareLock); 1321 1322 if (0 == mAudioHwDevs.size()) { 1323 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1324 if (NULL != dev->get_master_volume) { 1325 float mv; 1326 if (OK == dev->get_master_volume(dev, &mv)) { 1327 mMasterVolume = mv; 1328 } 1329 } 1330 1331 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1332 if (NULL != dev->get_master_mute) { 1333 bool mm; 1334 if (OK == dev->get_master_mute(dev, &mm)) { 1335 mMasterMute = mm; 1336 } 1337 } 1338 } 1339 1340 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1341 if ((NULL != dev->set_master_volume) && 1342 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1343 flags = static_cast<AudioHwDevice::Flags>(flags | 1344 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1345 } 1346 1347 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1348 if ((NULL != dev->set_master_mute) && 1349 (OK == dev->set_master_mute(dev, mMasterMute))) { 1350 flags = static_cast<AudioHwDevice::Flags>(flags | 1351 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1352 } 1353 1354 mHardwareStatus = AUDIO_HW_IDLE; 1355 } 1356 1357 audio_module_handle_t handle = nextUniqueId(); 1358 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1359 1360 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1361 name, dev->common.module->name, dev->common.module->id, handle); 1362 1363 return handle; 1364 1365 } 1366 1367 // ---------------------------------------------------------------------------- 1368 1369 uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1370 { 1371 Mutex::Autolock _l(mLock); 1372 PlaybackThread *thread = primaryPlaybackThread_l(); 1373 return thread != NULL ? thread->sampleRate() : 0; 1374 } 1375 1376 size_t AudioFlinger::getPrimaryOutputFrameCount() 1377 { 1378 Mutex::Autolock _l(mLock); 1379 PlaybackThread *thread = primaryPlaybackThread_l(); 1380 return thread != NULL ? thread->frameCountHAL() : 0; 1381 } 1382 1383 // ---------------------------------------------------------------------------- 1384 1385 audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1386 audio_devices_t *pDevices, 1387 uint32_t *pSamplingRate, 1388 audio_format_t *pFormat, 1389 audio_channel_mask_t *pChannelMask, 1390 uint32_t *pLatencyMs, 1391 audio_output_flags_t flags) 1392 { 1393 status_t status; 1394 PlaybackThread *thread = NULL; 1395 struct audio_config config = { 1396 sample_rate: pSamplingRate ? *pSamplingRate : 0, 1397 channel_mask: pChannelMask ? *pChannelMask : 0, 1398 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 1399 }; 1400 audio_stream_out_t *outStream = NULL; 1401 AudioHwDevice *outHwDev; 1402 1403 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 1404 module, 1405 (pDevices != NULL) ? *pDevices : 0, 1406 config.sample_rate, 1407 config.format, 1408 config.channel_mask, 1409 flags); 1410 1411 if (pDevices == NULL || *pDevices == 0) { 1412 return 0; 1413 } 1414 1415 Mutex::Autolock _l(mLock); 1416 1417 outHwDev = findSuitableHwDev_l(module, *pDevices); 1418 if (outHwDev == NULL) 1419 return 0; 1420 1421 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1422 audio_io_handle_t id = nextUniqueId(); 1423 1424 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1425 1426 status = hwDevHal->open_output_stream(hwDevHal, 1427 id, 1428 *pDevices, 1429 (audio_output_flags_t)flags, 1430 &config, 1431 &outStream); 1432 1433 mHardwareStatus = AUDIO_HW_IDLE; 1434 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 1435 "Channels %x, status %d", 1436 outStream, 1437 config.sample_rate, 1438 config.format, 1439 config.channel_mask, 1440 status); 1441 1442 if (status == NO_ERROR && outStream != NULL) { 1443 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 1444 1445 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1446 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1447 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1448 thread = new DirectOutputThread(this, output, id, *pDevices); 1449 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1450 } else { 1451 thread = new MixerThread(this, output, id, *pDevices); 1452 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1453 } 1454 mPlaybackThreads.add(id, thread); 1455 1456 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 1457 if (pFormat != NULL) *pFormat = config.format; 1458 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 1459 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 1460 1461 // notify client processes of the new output creation 1462 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1463 1464 // the first primary output opened designates the primary hw device 1465 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1466 ALOGI("Using module %d has the primary audio interface", module); 1467 mPrimaryHardwareDev = outHwDev; 1468 1469 AutoMutex lock(mHardwareLock); 1470 mHardwareStatus = AUDIO_HW_SET_MODE; 1471 hwDevHal->set_mode(hwDevHal, mMode); 1472 mHardwareStatus = AUDIO_HW_IDLE; 1473 } 1474 return id; 1475 } 1476 1477 return 0; 1478 } 1479 1480 audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1481 audio_io_handle_t output2) 1482 { 1483 Mutex::Autolock _l(mLock); 1484 MixerThread *thread1 = checkMixerThread_l(output1); 1485 MixerThread *thread2 = checkMixerThread_l(output2); 1486 1487 if (thread1 == NULL || thread2 == NULL) { 1488 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1489 output2); 1490 return 0; 1491 } 1492 1493 audio_io_handle_t id = nextUniqueId(); 1494 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1495 thread->addOutputTrack(thread2); 1496 mPlaybackThreads.add(id, thread); 1497 // notify client processes of the new output creation 1498 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1499 return id; 1500 } 1501 1502 status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1503 { 1504 return closeOutput_nonvirtual(output); 1505 } 1506 1507 status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1508 { 1509 // keep strong reference on the playback thread so that 1510 // it is not destroyed while exit() is executed 1511 sp<PlaybackThread> thread; 1512 { 1513 Mutex::Autolock _l(mLock); 1514 thread = checkPlaybackThread_l(output); 1515 if (thread == NULL) { 1516 return BAD_VALUE; 1517 } 1518 1519 ALOGV("closeOutput() %d", output); 1520 1521 if (thread->type() == ThreadBase::MIXER) { 1522 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1523 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1524 DuplicatingThread *dupThread = 1525 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1526 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1527 } 1528 } 1529 } 1530 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1531 mPlaybackThreads.removeItem(output); 1532 } 1533 thread->exit(); 1534 // The thread entity (active unit of execution) is no longer running here, 1535 // but the ThreadBase container still exists. 1536 1537 if (thread->type() != ThreadBase::DUPLICATING) { 1538 AudioStreamOut *out = thread->clearOutput(); 1539 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1540 // from now on thread->mOutput is NULL 1541 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1542 delete out; 1543 } 1544 return NO_ERROR; 1545 } 1546 1547 status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1548 { 1549 Mutex::Autolock _l(mLock); 1550 PlaybackThread *thread = checkPlaybackThread_l(output); 1551 1552 if (thread == NULL) { 1553 return BAD_VALUE; 1554 } 1555 1556 ALOGV("suspendOutput() %d", output); 1557 thread->suspend(); 1558 1559 return NO_ERROR; 1560 } 1561 1562 status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1563 { 1564 Mutex::Autolock _l(mLock); 1565 PlaybackThread *thread = checkPlaybackThread_l(output); 1566 1567 if (thread == NULL) { 1568 return BAD_VALUE; 1569 } 1570 1571 ALOGV("restoreOutput() %d", output); 1572 1573 thread->restore(); 1574 1575 return NO_ERROR; 1576 } 1577 1578 audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1579 audio_devices_t *pDevices, 1580 uint32_t *pSamplingRate, 1581 audio_format_t *pFormat, 1582 audio_channel_mask_t *pChannelMask) 1583 { 1584 status_t status; 1585 RecordThread *thread = NULL; 1586 struct audio_config config = { 1587 sample_rate: pSamplingRate ? *pSamplingRate : 0, 1588 channel_mask: pChannelMask ? *pChannelMask : 0, 1589 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 1590 }; 1591 uint32_t reqSamplingRate = config.sample_rate; 1592 audio_format_t reqFormat = config.format; 1593 audio_channel_mask_t reqChannels = config.channel_mask; 1594 audio_stream_in_t *inStream = NULL; 1595 AudioHwDevice *inHwDev; 1596 1597 if (pDevices == NULL || *pDevices == 0) { 1598 return 0; 1599 } 1600 1601 Mutex::Autolock _l(mLock); 1602 1603 inHwDev = findSuitableHwDev_l(module, *pDevices); 1604 if (inHwDev == NULL) 1605 return 0; 1606 1607 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1608 audio_io_handle_t id = nextUniqueId(); 1609 1610 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1611 &inStream); 1612 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1613 "status %d", 1614 inStream, 1615 config.sample_rate, 1616 config.format, 1617 config.channel_mask, 1618 status); 1619 1620 // If the input could not be opened with the requested parameters and we can handle the 1621 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1622 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1623 if (status == BAD_VALUE && 1624 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1625 (config.sample_rate <= 2 * reqSamplingRate) && 1626 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 1627 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1628 inStream = NULL; 1629 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1630 } 1631 1632 if (status == NO_ERROR && inStream != NULL) { 1633 1634 #ifdef TEE_SINK 1635 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1636 // or (re-)create if current Pipe is idle and does not match the new format 1637 sp<NBAIO_Sink> teeSink; 1638 enum { 1639 TEE_SINK_NO, // don't copy input 1640 TEE_SINK_NEW, // copy input using a new pipe 1641 TEE_SINK_OLD, // copy input using an existing pipe 1642 } kind; 1643 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1644 popcount(inStream->common.get_channels(&inStream->common))); 1645 if (!mTeeSinkInputEnabled) { 1646 kind = TEE_SINK_NO; 1647 } else if (format == Format_Invalid) { 1648 kind = TEE_SINK_NO; 1649 } else if (mRecordTeeSink == 0) { 1650 kind = TEE_SINK_NEW; 1651 } else if (mRecordTeeSink->getStrongCount() != 1) { 1652 kind = TEE_SINK_NO; 1653 } else if (format == mRecordTeeSink->format()) { 1654 kind = TEE_SINK_OLD; 1655 } else { 1656 kind = TEE_SINK_NEW; 1657 } 1658 switch (kind) { 1659 case TEE_SINK_NEW: { 1660 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1661 size_t numCounterOffers = 0; 1662 const NBAIO_Format offers[1] = {format}; 1663 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1664 ALOG_ASSERT(index == 0); 1665 PipeReader *pipeReader = new PipeReader(*pipe); 1666 numCounterOffers = 0; 1667 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1668 ALOG_ASSERT(index == 0); 1669 mRecordTeeSink = pipe; 1670 mRecordTeeSource = pipeReader; 1671 teeSink = pipe; 1672 } 1673 break; 1674 case TEE_SINK_OLD: 1675 teeSink = mRecordTeeSink; 1676 break; 1677 case TEE_SINK_NO: 1678 default: 1679 break; 1680 } 1681 #endif 1682 1683 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1684 1685 // Start record thread 1686 // RecorThread require both input and output device indication to forward to audio 1687 // pre processing modules 1688 thread = new RecordThread(this, 1689 input, 1690 reqSamplingRate, 1691 reqChannels, 1692 id, 1693 primaryOutputDevice_l(), 1694 *pDevices 1695 #ifdef TEE_SINK 1696 , teeSink 1697 #endif 1698 ); 1699 mRecordThreads.add(id, thread); 1700 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1701 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 1702 if (pFormat != NULL) *pFormat = config.format; 1703 if (pChannelMask != NULL) *pChannelMask = reqChannels; 1704 1705 // notify client processes of the new input creation 1706 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1707 return id; 1708 } 1709 1710 return 0; 1711 } 1712 1713 status_t AudioFlinger::closeInput(audio_io_handle_t input) 1714 { 1715 return closeInput_nonvirtual(input); 1716 } 1717 1718 status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1719 { 1720 // keep strong reference on the record thread so that 1721 // it is not destroyed while exit() is executed 1722 sp<RecordThread> thread; 1723 { 1724 Mutex::Autolock _l(mLock); 1725 thread = checkRecordThread_l(input); 1726 if (thread == 0) { 1727 return BAD_VALUE; 1728 } 1729 1730 ALOGV("closeInput() %d", input); 1731 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1732 mRecordThreads.removeItem(input); 1733 } 1734 thread->exit(); 1735 // The thread entity (active unit of execution) is no longer running here, 1736 // but the ThreadBase container still exists. 1737 1738 AudioStreamIn *in = thread->clearInput(); 1739 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1740 // from now on thread->mInput is NULL 1741 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1742 delete in; 1743 1744 return NO_ERROR; 1745 } 1746 1747 status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1748 { 1749 Mutex::Autolock _l(mLock); 1750 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1751 1752 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1753 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1754 thread->invalidateTracks(stream); 1755 } 1756 1757 return NO_ERROR; 1758 } 1759 1760 1761 int AudioFlinger::newAudioSessionId() 1762 { 1763 return nextUniqueId(); 1764 } 1765 1766 void AudioFlinger::acquireAudioSessionId(int audioSession) 1767 { 1768 Mutex::Autolock _l(mLock); 1769 pid_t caller = IPCThreadState::self()->getCallingPid(); 1770 ALOGV("acquiring %d from %d", audioSession, caller); 1771 size_t num = mAudioSessionRefs.size(); 1772 for (size_t i = 0; i< num; i++) { 1773 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1774 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1775 ref->mCnt++; 1776 ALOGV(" incremented refcount to %d", ref->mCnt); 1777 return; 1778 } 1779 } 1780 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1781 ALOGV(" added new entry for %d", audioSession); 1782 } 1783 1784 void AudioFlinger::releaseAudioSessionId(int audioSession) 1785 { 1786 Mutex::Autolock _l(mLock); 1787 pid_t caller = IPCThreadState::self()->getCallingPid(); 1788 ALOGV("releasing %d from %d", audioSession, caller); 1789 size_t num = mAudioSessionRefs.size(); 1790 for (size_t i = 0; i< num; i++) { 1791 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1792 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1793 ref->mCnt--; 1794 ALOGV(" decremented refcount to %d", ref->mCnt); 1795 if (ref->mCnt == 0) { 1796 mAudioSessionRefs.removeAt(i); 1797 delete ref; 1798 purgeStaleEffects_l(); 1799 } 1800 return; 1801 } 1802 } 1803 ALOGW("session id %d not found for pid %d", audioSession, caller); 1804 } 1805 1806 void AudioFlinger::purgeStaleEffects_l() { 1807 1808 ALOGV("purging stale effects"); 1809 1810 Vector< sp<EffectChain> > chains; 1811 1812 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1813 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1814 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1815 sp<EffectChain> ec = t->mEffectChains[j]; 1816 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1817 chains.push(ec); 1818 } 1819 } 1820 } 1821 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1822 sp<RecordThread> t = mRecordThreads.valueAt(i); 1823 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1824 sp<EffectChain> ec = t->mEffectChains[j]; 1825 chains.push(ec); 1826 } 1827 } 1828 1829 for (size_t i = 0; i < chains.size(); i++) { 1830 sp<EffectChain> ec = chains[i]; 1831 int sessionid = ec->sessionId(); 1832 sp<ThreadBase> t = ec->mThread.promote(); 1833 if (t == 0) { 1834 continue; 1835 } 1836 size_t numsessionrefs = mAudioSessionRefs.size(); 1837 bool found = false; 1838 for (size_t k = 0; k < numsessionrefs; k++) { 1839 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1840 if (ref->mSessionid == sessionid) { 1841 ALOGV(" session %d still exists for %d with %d refs", 1842 sessionid, ref->mPid, ref->mCnt); 1843 found = true; 1844 break; 1845 } 1846 } 1847 if (!found) { 1848 Mutex::Autolock _l (t->mLock); 1849 // remove all effects from the chain 1850 while (ec->mEffects.size()) { 1851 sp<EffectModule> effect = ec->mEffects[0]; 1852 effect->unPin(); 1853 t->removeEffect_l(effect); 1854 if (effect->purgeHandles()) { 1855 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1856 } 1857 AudioSystem::unregisterEffect(effect->id()); 1858 } 1859 } 1860 } 1861 return; 1862 } 1863 1864 // checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1865 AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1866 { 1867 return mPlaybackThreads.valueFor(output).get(); 1868 } 1869 1870 // checkMixerThread_l() must be called with AudioFlinger::mLock held 1871 AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1872 { 1873 PlaybackThread *thread = checkPlaybackThread_l(output); 1874 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1875 } 1876 1877 // checkRecordThread_l() must be called with AudioFlinger::mLock held 1878 AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1879 { 1880 return mRecordThreads.valueFor(input).get(); 1881 } 1882 1883 uint32_t AudioFlinger::nextUniqueId() 1884 { 1885 return android_atomic_inc(&mNextUniqueId); 1886 } 1887 1888 AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1889 { 1890 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1891 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1892 AudioStreamOut *output = thread->getOutput(); 1893 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1894 return thread; 1895 } 1896 } 1897 return NULL; 1898 } 1899 1900 audio_devices_t AudioFlinger::primaryOutputDevice_l() const 1901 { 1902 PlaybackThread *thread = primaryPlaybackThread_l(); 1903 1904 if (thread == NULL) { 1905 return 0; 1906 } 1907 1908 return thread->outDevice(); 1909 } 1910 1911 sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 1912 int triggerSession, 1913 int listenerSession, 1914 sync_event_callback_t callBack, 1915 void *cookie) 1916 { 1917 Mutex::Autolock _l(mLock); 1918 1919 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 1920 status_t playStatus = NAME_NOT_FOUND; 1921 status_t recStatus = NAME_NOT_FOUND; 1922 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1923 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 1924 if (playStatus == NO_ERROR) { 1925 return event; 1926 } 1927 } 1928 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1929 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 1930 if (recStatus == NO_ERROR) { 1931 return event; 1932 } 1933 } 1934 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 1935 mPendingSyncEvents.add(event); 1936 } else { 1937 ALOGV("createSyncEvent() invalid event %d", event->type()); 1938 event.clear(); 1939 } 1940 return event; 1941 } 1942 1943 // ---------------------------------------------------------------------------- 1944 // Effect management 1945 // ---------------------------------------------------------------------------- 1946 1947 1948 status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 1949 { 1950 Mutex::Autolock _l(mLock); 1951 return EffectQueryNumberEffects(numEffects); 1952 } 1953 1954 status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 1955 { 1956 Mutex::Autolock _l(mLock); 1957 return EffectQueryEffect(index, descriptor); 1958 } 1959 1960 status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 1961 effect_descriptor_t *descriptor) const 1962 { 1963 Mutex::Autolock _l(mLock); 1964 return EffectGetDescriptor(pUuid, descriptor); 1965 } 1966 1967 1968 sp<IEffect> AudioFlinger::createEffect( 1969 effect_descriptor_t *pDesc, 1970 const sp<IEffectClient>& effectClient, 1971 int32_t priority, 1972 audio_io_handle_t io, 1973 int sessionId, 1974 status_t *status, 1975 int *id, 1976 int *enabled) 1977 { 1978 status_t lStatus = NO_ERROR; 1979 sp<EffectHandle> handle; 1980 effect_descriptor_t desc; 1981 1982 pid_t pid = IPCThreadState::self()->getCallingPid(); 1983 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 1984 pid, effectClient.get(), priority, sessionId, io); 1985 1986 if (pDesc == NULL) { 1987 lStatus = BAD_VALUE; 1988 goto Exit; 1989 } 1990 1991 // check audio settings permission for global effects 1992 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 1993 lStatus = PERMISSION_DENIED; 1994 goto Exit; 1995 } 1996 1997 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 1998 // that can only be created by audio policy manager (running in same process) 1999 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2000 lStatus = PERMISSION_DENIED; 2001 goto Exit; 2002 } 2003 2004 if (io == 0) { 2005 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2006 // output must be specified by AudioPolicyManager when using session 2007 // AUDIO_SESSION_OUTPUT_STAGE 2008 lStatus = BAD_VALUE; 2009 goto Exit; 2010 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2011 // if the output returned by getOutputForEffect() is removed before we lock the 2012 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2013 // and we will exit safely 2014 io = AudioSystem::getOutputForEffect(&desc); 2015 } 2016 } 2017 2018 { 2019 Mutex::Autolock _l(mLock); 2020 2021 2022 if (!EffectIsNullUuid(&pDesc->uuid)) { 2023 // if uuid is specified, request effect descriptor 2024 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2025 if (lStatus < 0) { 2026 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2027 goto Exit; 2028 } 2029 } else { 2030 // if uuid is not specified, look for an available implementation 2031 // of the required type in effect factory 2032 if (EffectIsNullUuid(&pDesc->type)) { 2033 ALOGW("createEffect() no effect type"); 2034 lStatus = BAD_VALUE; 2035 goto Exit; 2036 } 2037 uint32_t numEffects = 0; 2038 effect_descriptor_t d; 2039 d.flags = 0; // prevent compiler warning 2040 bool found = false; 2041 2042 lStatus = EffectQueryNumberEffects(&numEffects); 2043 if (lStatus < 0) { 2044 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2045 goto Exit; 2046 } 2047 for (uint32_t i = 0; i < numEffects; i++) { 2048 lStatus = EffectQueryEffect(i, &desc); 2049 if (lStatus < 0) { 2050 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2051 continue; 2052 } 2053 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2054 // If matching type found save effect descriptor. If the session is 2055 // 0 and the effect is not auxiliary, continue enumeration in case 2056 // an auxiliary version of this effect type is available 2057 found = true; 2058 d = desc; 2059 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2060 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2061 break; 2062 } 2063 } 2064 } 2065 if (!found) { 2066 lStatus = BAD_VALUE; 2067 ALOGW("createEffect() effect not found"); 2068 goto Exit; 2069 } 2070 // For same effect type, chose auxiliary version over insert version if 2071 // connect to output mix (Compliance to OpenSL ES) 2072 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2073 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2074 desc = d; 2075 } 2076 } 2077 2078 // Do not allow auxiliary effects on a session different from 0 (output mix) 2079 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2080 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2081 lStatus = INVALID_OPERATION; 2082 goto Exit; 2083 } 2084 2085 // check recording permission for visualizer 2086 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2087 !recordingAllowed()) { 2088 lStatus = PERMISSION_DENIED; 2089 goto Exit; 2090 } 2091 2092 // return effect descriptor 2093 *pDesc = desc; 2094 2095 // If output is not specified try to find a matching audio session ID in one of the 2096 // output threads. 2097 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2098 // because of code checking output when entering the function. 2099 // Note: io is never 0 when creating an effect on an input 2100 if (io == 0) { 2101 // look for the thread where the specified audio session is present 2102 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2103 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2104 io = mPlaybackThreads.keyAt(i); 2105 break; 2106 } 2107 } 2108 if (io == 0) { 2109 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2110 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2111 io = mRecordThreads.keyAt(i); 2112 break; 2113 } 2114 } 2115 } 2116 // If no output thread contains the requested session ID, default to 2117 // first output. The effect chain will be moved to the correct output 2118 // thread when a track with the same session ID is created 2119 if (io == 0 && mPlaybackThreads.size()) { 2120 io = mPlaybackThreads.keyAt(0); 2121 } 2122 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2123 } 2124 ThreadBase *thread = checkRecordThread_l(io); 2125 if (thread == NULL) { 2126 thread = checkPlaybackThread_l(io); 2127 if (thread == NULL) { 2128 ALOGE("createEffect() unknown output thread"); 2129 lStatus = BAD_VALUE; 2130 goto Exit; 2131 } 2132 } 2133 2134 sp<Client> client = registerPid_l(pid); 2135 2136 // create effect on selected output thread 2137 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2138 &desc, enabled, &lStatus); 2139 if (handle != 0 && id != NULL) { 2140 *id = handle->id(); 2141 } 2142 } 2143 2144 Exit: 2145 if (status != NULL) { 2146 *status = lStatus; 2147 } 2148 return handle; 2149 } 2150 2151 status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2152 audio_io_handle_t dstOutput) 2153 { 2154 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2155 sessionId, srcOutput, dstOutput); 2156 Mutex::Autolock _l(mLock); 2157 if (srcOutput == dstOutput) { 2158 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2159 return NO_ERROR; 2160 } 2161 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2162 if (srcThread == NULL) { 2163 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2164 return BAD_VALUE; 2165 } 2166 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2167 if (dstThread == NULL) { 2168 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2169 return BAD_VALUE; 2170 } 2171 2172 Mutex::Autolock _dl(dstThread->mLock); 2173 Mutex::Autolock _sl(srcThread->mLock); 2174 moveEffectChain_l(sessionId, srcThread, dstThread, false); 2175 2176 return NO_ERROR; 2177 } 2178 2179 // moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2180 status_t AudioFlinger::moveEffectChain_l(int sessionId, 2181 AudioFlinger::PlaybackThread *srcThread, 2182 AudioFlinger::PlaybackThread *dstThread, 2183 bool reRegister) 2184 { 2185 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2186 sessionId, srcThread, dstThread); 2187 2188 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2189 if (chain == 0) { 2190 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2191 sessionId, srcThread); 2192 return INVALID_OPERATION; 2193 } 2194 2195 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2196 // so that a new chain is created with correct parameters when first effect is added. This is 2197 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2198 // removed. 2199 srcThread->removeEffectChain_l(chain); 2200 2201 // transfer all effects one by one so that new effect chain is created on new thread with 2202 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2203 audio_io_handle_t dstOutput = dstThread->id(); 2204 sp<EffectChain> dstChain; 2205 uint32_t strategy = 0; // prevent compiler warning 2206 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2207 while (effect != 0) { 2208 srcThread->removeEffect_l(effect); 2209 dstThread->addEffect_l(effect); 2210 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2211 if (effect->state() == EffectModule::ACTIVE || 2212 effect->state() == EffectModule::STOPPING) { 2213 effect->start(); 2214 } 2215 // if the move request is not received from audio policy manager, the effect must be 2216 // re-registered with the new strategy and output 2217 if (dstChain == 0) { 2218 dstChain = effect->chain().promote(); 2219 if (dstChain == 0) { 2220 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2221 srcThread->addEffect_l(effect); 2222 return NO_INIT; 2223 } 2224 strategy = dstChain->strategy(); 2225 } 2226 if (reRegister) { 2227 AudioSystem::unregisterEffect(effect->id()); 2228 AudioSystem::registerEffect(&effect->desc(), 2229 dstOutput, 2230 strategy, 2231 sessionId, 2232 effect->id()); 2233 } 2234 effect = chain->getEffectFromId_l(0); 2235 } 2236 2237 return NO_ERROR; 2238 } 2239 2240 struct Entry { 2241 #define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2242 char mName[MAX_NAME]; 2243 }; 2244 2245 int comparEntry(const void *p1, const void *p2) 2246 { 2247 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2248 } 2249 2250 #ifdef TEE_SINK 2251 void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2252 { 2253 NBAIO_Source *teeSource = source.get(); 2254 if (teeSource != NULL) { 2255 // .wav rotation 2256 // There is a benign race condition if 2 threads call this simultaneously. 2257 // They would both traverse the directory, but the result would simply be 2258 // failures at unlink() which are ignored. It's also unlikely since 2259 // normally dumpsys is only done by bugreport or from the command line. 2260 char teePath[32+256]; 2261 strcpy(teePath, "/data/misc/media"); 2262 size_t teePathLen = strlen(teePath); 2263 DIR *dir = opendir(teePath); 2264 teePath[teePathLen++] = '/'; 2265 if (dir != NULL) { 2266 #define MAX_SORT 20 // number of entries to sort 2267 #define MAX_KEEP 10 // number of entries to keep 2268 struct Entry entries[MAX_SORT]; 2269 size_t entryCount = 0; 2270 while (entryCount < MAX_SORT) { 2271 struct dirent de; 2272 struct dirent *result = NULL; 2273 int rc = readdir_r(dir, &de, &result); 2274 if (rc != 0) { 2275 ALOGW("readdir_r failed %d", rc); 2276 break; 2277 } 2278 if (result == NULL) { 2279 break; 2280 } 2281 if (result != &de) { 2282 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2283 break; 2284 } 2285 // ignore non .wav file entries 2286 size_t nameLen = strlen(de.d_name); 2287 if (nameLen <= 4 || nameLen >= MAX_NAME || 2288 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2289 continue; 2290 } 2291 strcpy(entries[entryCount++].mName, de.d_name); 2292 } 2293 (void) closedir(dir); 2294 if (entryCount > MAX_KEEP) { 2295 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2296 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2297 strcpy(&teePath[teePathLen], entries[i].mName); 2298 (void) unlink(teePath); 2299 } 2300 } 2301 } else { 2302 if (fd >= 0) { 2303 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2304 } 2305 } 2306 char teeTime[16]; 2307 struct timeval tv; 2308 gettimeofday(&tv, NULL); 2309 struct tm tm; 2310 localtime_r(&tv.tv_sec, &tm); 2311 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2312 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2313 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2314 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2315 if (teeFd >= 0) { 2316 char wavHeader[44]; 2317 memcpy(wavHeader, 2318 "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", 2319 sizeof(wavHeader)); 2320 NBAIO_Format format = teeSource->format(); 2321 unsigned channelCount = Format_channelCount(format); 2322 ALOG_ASSERT(channelCount <= FCC_2); 2323 uint32_t sampleRate = Format_sampleRate(format); 2324 wavHeader[22] = channelCount; // number of channels 2325 wavHeader[24] = sampleRate; // sample rate 2326 wavHeader[25] = sampleRate >> 8; 2327 wavHeader[32] = channelCount * 2; // block alignment 2328 write(teeFd, wavHeader, sizeof(wavHeader)); 2329 size_t total = 0; 2330 bool firstRead = true; 2331 for (;;) { 2332 #define TEE_SINK_READ 1024 2333 short buffer[TEE_SINK_READ * FCC_2]; 2334 size_t count = TEE_SINK_READ; 2335 ssize_t actual = teeSource->read(buffer, count, 2336 AudioBufferProvider::kInvalidPTS); 2337 bool wasFirstRead = firstRead; 2338 firstRead = false; 2339 if (actual <= 0) { 2340 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2341 continue; 2342 } 2343 break; 2344 } 2345 ALOG_ASSERT(actual <= (ssize_t)count); 2346 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2347 total += actual; 2348 } 2349 lseek(teeFd, (off_t) 4, SEEK_SET); 2350 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2351 write(teeFd, &temp, sizeof(temp)); 2352 lseek(teeFd, (off_t) 40, SEEK_SET); 2353 temp = total * channelCount * sizeof(short); 2354 write(teeFd, &temp, sizeof(temp)); 2355 close(teeFd); 2356 if (fd >= 0) { 2357 fdprintf(fd, "tee copied to %s\n", teePath); 2358 } 2359 } else { 2360 if (fd >= 0) { 2361 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2362 } 2363 } 2364 } 2365 } 2366 #endif 2367 2368 // ---------------------------------------------------------------------------- 2369 2370 status_t AudioFlinger::onTransact( 2371 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2372 { 2373 return BnAudioFlinger::onTransact(code, data, reply, flags); 2374 } 2375 2376 }; // namespace android 2377