AudioFlinger.cpp revision ae0cff1d48b2cd10aeff9627398faf684894eece
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/AudioResamplerPublic.h> 49 50#include <media/EffectsFactoryApi.h> 51#include <audio_effects/effect_visualizer.h> 52#include <audio_effects/effect_ns.h> 53#include <audio_effects/effect_aec.h> 54 55#include <audio_utils/primitives.h> 56 57#include <powermanager/PowerManager.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 <mediautils/BatteryNotifier.h> 65#include <private/android_filesystem_config.h> 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 85static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 86static const char kClientLockedString[] = "Client lock is taken\n"; 87 88 89nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 90 91uint32_t AudioFlinger::mScreenState; 92 93#ifdef TEE_SINK 94bool AudioFlinger::mTeeSinkInputEnabled = false; 95bool AudioFlinger::mTeeSinkOutputEnabled = false; 96bool AudioFlinger::mTeeSinkTrackEnabled = false; 97 98size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 99size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 100size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 101#endif 102 103// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 104// we define a minimum time during which a global effect is considered enabled. 105static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 106 107// ---------------------------------------------------------------------------- 108 109const char *formatToString(audio_format_t format) { 110 switch (audio_get_main_format(format)) { 111 case AUDIO_FORMAT_PCM: 112 switch (format) { 113 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 114 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 115 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 116 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 117 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 118 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 119 default: 120 break; 121 } 122 break; 123 case AUDIO_FORMAT_MP3: return "mp3"; 124 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 125 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 126 case AUDIO_FORMAT_AAC: return "aac"; 127 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 128 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 129 case AUDIO_FORMAT_VORBIS: return "vorbis"; 130 case AUDIO_FORMAT_OPUS: return "opus"; 131 case AUDIO_FORMAT_AC3: return "ac-3"; 132 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 133 case AUDIO_FORMAT_IEC61937: return "iec61937"; 134 default: 135 break; 136 } 137 return "unknown"; 138} 139 140static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 141{ 142 const hw_module_t *mod; 143 int rc; 144 145 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 146 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 147 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 148 if (rc) { 149 goto out; 150 } 151 rc = audio_hw_device_open(mod, dev); 152 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 153 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 154 if (rc) { 155 goto out; 156 } 157 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 158 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 159 rc = BAD_VALUE; 160 goto out; 161 } 162 return 0; 163 164out: 165 *dev = NULL; 166 return rc; 167} 168 169// ---------------------------------------------------------------------------- 170 171AudioFlinger::AudioFlinger() 172 : BnAudioFlinger(), 173 mPrimaryHardwareDev(NULL), 174 mAudioHwDevs(NULL), 175 mHardwareStatus(AUDIO_HW_IDLE), 176 mMasterVolume(1.0f), 177 mMasterMute(false), 178 mNextUniqueId(1), 179 mMode(AUDIO_MODE_INVALID), 180 mBtNrecIsOff(false), 181 mIsLowRamDevice(true), 182 mIsDeviceTypeKnown(false), 183 mGlobalEffectEnableTime(0), 184 mSystemReady(false) 185{ 186 getpid_cached = getpid(); 187 const bool doLog = property_get_bool("ro.test_harness", false); 188 if (doLog) { 189 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 190 MemoryHeapBase::READ_ONLY); 191 } 192 193 // reset battery stats. 194 // if the audio service has crashed, battery stats could be left 195 // in bad state, reset the state upon service start. 196 BatteryNotifier::getInstance().noteResetAudio(); 197 198#ifdef TEE_SINK 199 char value[PROPERTY_VALUE_MAX]; 200 (void) property_get("ro.debuggable", value, "0"); 201 int debuggable = atoi(value); 202 int teeEnabled = 0; 203 if (debuggable) { 204 (void) property_get("af.tee", value, "0"); 205 teeEnabled = atoi(value); 206 } 207 // FIXME symbolic constants here 208 if (teeEnabled & 1) { 209 mTeeSinkInputEnabled = true; 210 } 211 if (teeEnabled & 2) { 212 mTeeSinkOutputEnabled = true; 213 } 214 if (teeEnabled & 4) { 215 mTeeSinkTrackEnabled = true; 216 } 217#endif 218} 219 220void AudioFlinger::onFirstRef() 221{ 222 int rc = 0; 223 224 Mutex::Autolock _l(mLock); 225 226 /* TODO: move all this work into an Init() function */ 227 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 228 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 229 uint32_t int_val; 230 if (1 == sscanf(val_str, "%u", &int_val)) { 231 mStandbyTimeInNsecs = milliseconds(int_val); 232 ALOGI("Using %u mSec as standby time.", int_val); 233 } else { 234 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 235 ALOGI("Using default %u mSec as standby time.", 236 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 237 } 238 } 239 240 mPatchPanel = new PatchPanel(this); 241 242 mMode = AUDIO_MODE_NORMAL; 243} 244 245AudioFlinger::~AudioFlinger() 246{ 247 while (!mRecordThreads.isEmpty()) { 248 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 249 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 250 } 251 while (!mPlaybackThreads.isEmpty()) { 252 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 253 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 254 } 255 256 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 257 // no mHardwareLock needed, as there are no other references to this 258 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 259 delete mAudioHwDevs.valueAt(i); 260 } 261 262 // Tell media.log service about any old writers that still need to be unregistered 263 if (mLogMemoryDealer != 0) { 264 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 265 if (binder != 0) { 266 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 267 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 268 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 269 mUnregisteredWriters.pop(); 270 mediaLogService->unregisterWriter(iMemory); 271 } 272 } 273 } 274} 275 276static const char * const audio_interfaces[] = { 277 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 278 AUDIO_HARDWARE_MODULE_ID_A2DP, 279 AUDIO_HARDWARE_MODULE_ID_USB, 280}; 281#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 282 283AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 284 audio_module_handle_t module, 285 audio_devices_t devices) 286{ 287 // if module is 0, the request comes from an old policy manager and we should load 288 // well known modules 289 if (module == 0) { 290 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 291 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 292 loadHwModule_l(audio_interfaces[i]); 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 297 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 298 if ((dev->get_supported_devices != NULL) && 299 (dev->get_supported_devices(dev) & devices) == devices) 300 return audioHwDevice; 301 } 302 } else { 303 // check a match for the requested module handle 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 305 if (audioHwDevice != NULL) { 306 return audioHwDevice; 307 } 308 } 309 310 return NULL; 311} 312 313void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 314{ 315 const size_t SIZE = 256; 316 char buffer[SIZE]; 317 String8 result; 318 319 result.append("Clients:\n"); 320 for (size_t i = 0; i < mClients.size(); ++i) { 321 sp<Client> client = mClients.valueAt(i).promote(); 322 if (client != 0) { 323 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 324 result.append(buffer); 325 } 326 } 327 328 result.append("Notification Clients:\n"); 329 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 330 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 331 result.append(buffer); 332 } 333 334 result.append("Global session refs:\n"); 335 result.append(" session pid count\n"); 336 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 337 AudioSessionRef *r = mAudioSessionRefs[i]; 338 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 339 result.append(buffer); 340 } 341 write(fd, result.string(), result.size()); 342} 343 344 345void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 346{ 347 const size_t SIZE = 256; 348 char buffer[SIZE]; 349 String8 result; 350 hardware_call_state hardwareStatus = mHardwareStatus; 351 352 snprintf(buffer, SIZE, "Hardware status: %d\n" 353 "Standby Time mSec: %u\n", 354 hardwareStatus, 355 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358} 359 360void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 361{ 362 const size_t SIZE = 256; 363 char buffer[SIZE]; 364 String8 result; 365 snprintf(buffer, SIZE, "Permission Denial: " 366 "can't dump AudioFlinger from pid=%d, uid=%d\n", 367 IPCThreadState::self()->getCallingPid(), 368 IPCThreadState::self()->getCallingUid()); 369 result.append(buffer); 370 write(fd, result.string(), result.size()); 371} 372 373bool AudioFlinger::dumpTryLock(Mutex& mutex) 374{ 375 bool locked = false; 376 for (int i = 0; i < kDumpLockRetries; ++i) { 377 if (mutex.tryLock() == NO_ERROR) { 378 locked = true; 379 break; 380 } 381 usleep(kDumpLockSleepUs); 382 } 383 return locked; 384} 385 386status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 387{ 388 if (!dumpAllowed()) { 389 dumpPermissionDenial(fd, args); 390 } else { 391 // get state of hardware lock 392 bool hardwareLocked = dumpTryLock(mHardwareLock); 393 if (!hardwareLocked) { 394 String8 result(kHardwareLockedString); 395 write(fd, result.string(), result.size()); 396 } else { 397 mHardwareLock.unlock(); 398 } 399 400 bool locked = dumpTryLock(mLock); 401 402 // failed to lock - AudioFlinger is probably deadlocked 403 if (!locked) { 404 String8 result(kDeadlockedString); 405 write(fd, result.string(), result.size()); 406 } 407 408 bool clientLocked = dumpTryLock(mClientLock); 409 if (!clientLocked) { 410 String8 result(kClientLockedString); 411 write(fd, result.string(), result.size()); 412 } 413 414 EffectDumpEffects(fd); 415 416 dumpClients(fd, args); 417 if (clientLocked) { 418 mClientLock.unlock(); 419 } 420 421 dumpInternals(fd, args); 422 423 // dump playback threads 424 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 425 mPlaybackThreads.valueAt(i)->dump(fd, args); 426 } 427 428 // dump record threads 429 for (size_t i = 0; i < mRecordThreads.size(); i++) { 430 mRecordThreads.valueAt(i)->dump(fd, args); 431 } 432 433 // dump orphan effect chains 434 if (mOrphanEffectChains.size() != 0) { 435 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 436 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 437 mOrphanEffectChains.valueAt(i)->dump(fd, args); 438 } 439 } 440 // dump all hardware devs 441 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 442 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 443 dev->dump(dev, fd); 444 } 445 446#ifdef TEE_SINK 447 // dump the serially shared record tee sink 448 if (mRecordTeeSource != 0) { 449 dumpTee(fd, mRecordTeeSource); 450 } 451#endif 452 453 if (locked) { 454 mLock.unlock(); 455 } 456 457 // append a copy of media.log here by forwarding fd to it, but don't attempt 458 // to lookup the service if it's not running, as it will block for a second 459 if (mLogMemoryDealer != 0) { 460 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 461 if (binder != 0) { 462 dprintf(fd, "\nmedia.log:\n"); 463 Vector<String16> args; 464 binder->dump(fd, args); 465 } 466 } 467 } 468 return NO_ERROR; 469} 470 471sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 472{ 473 Mutex::Autolock _cl(mClientLock); 474 // If pid is already in the mClients wp<> map, then use that entry 475 // (for which promote() is always != 0), otherwise create a new entry and Client. 476 sp<Client> client = mClients.valueFor(pid).promote(); 477 if (client == 0) { 478 client = new Client(this, pid); 479 mClients.add(pid, client); 480 } 481 482 return client; 483} 484 485sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 486{ 487 // If there is no memory allocated for logs, return a dummy writer that does nothing 488 if (mLogMemoryDealer == 0) { 489 return new NBLog::Writer(); 490 } 491 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 492 // Similarly if we can't contact the media.log service, also return a dummy writer 493 if (binder == 0) { 494 return new NBLog::Writer(); 495 } 496 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 497 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 498 // If allocation fails, consult the vector of previously unregistered writers 499 // and garbage-collect one or more them until an allocation succeeds 500 if (shared == 0) { 501 Mutex::Autolock _l(mUnregisteredWritersLock); 502 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 503 { 504 // Pick the oldest stale writer to garbage-collect 505 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 506 mUnregisteredWriters.removeAt(0); 507 mediaLogService->unregisterWriter(iMemory); 508 // Now the media.log remote reference to IMemory is gone. When our last local 509 // reference to IMemory also drops to zero at end of this block, 510 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 511 } 512 // Re-attempt the allocation 513 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 514 if (shared != 0) { 515 goto success; 516 } 517 } 518 // Even after garbage-collecting all old writers, there is still not enough memory, 519 // so return a dummy writer 520 return new NBLog::Writer(); 521 } 522success: 523 mediaLogService->registerWriter(shared, size, name); 524 return new NBLog::Writer(size, shared); 525} 526 527void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 528{ 529 if (writer == 0) { 530 return; 531 } 532 sp<IMemory> iMemory(writer->getIMemory()); 533 if (iMemory == 0) { 534 return; 535 } 536 // Rather than removing the writer immediately, append it to a queue of old writers to 537 // be garbage-collected later. This allows us to continue to view old logs for a while. 538 Mutex::Autolock _l(mUnregisteredWritersLock); 539 mUnregisteredWriters.push(writer); 540} 541 542// IAudioFlinger interface 543 544 545sp<IAudioTrack> AudioFlinger::createTrack( 546 audio_stream_type_t streamType, 547 uint32_t sampleRate, 548 audio_format_t format, 549 audio_channel_mask_t channelMask, 550 size_t *frameCount, 551 IAudioFlinger::track_flags_t *flags, 552 const sp<IMemory>& sharedBuffer, 553 audio_io_handle_t output, 554 pid_t tid, 555 int *sessionId, 556 int clientUid, 557 status_t *status) 558{ 559 sp<PlaybackThread::Track> track; 560 sp<TrackHandle> trackHandle; 561 sp<Client> client; 562 status_t lStatus; 563 int lSessionId; 564 565 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 566 // but if someone uses binder directly they could bypass that and cause us to crash 567 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 568 ALOGE("createTrack() invalid stream type %d", streamType); 569 lStatus = BAD_VALUE; 570 goto Exit; 571 } 572 573 // further sample rate checks are performed by createTrack_l() depending on the thread type 574 if (sampleRate == 0) { 575 ALOGE("createTrack() invalid sample rate %u", sampleRate); 576 lStatus = BAD_VALUE; 577 goto Exit; 578 } 579 580 // further channel mask checks are performed by createTrack_l() depending on the thread type 581 if (!audio_is_output_channel(channelMask)) { 582 ALOGE("createTrack() invalid channel mask %#x", channelMask); 583 lStatus = BAD_VALUE; 584 goto Exit; 585 } 586 587 // further format checks are performed by createTrack_l() depending on the thread type 588 if (!audio_is_valid_format(format)) { 589 ALOGE("createTrack() invalid format %#x", format); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 595 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 596 lStatus = BAD_VALUE; 597 goto Exit; 598 } 599 600 { 601 Mutex::Autolock _l(mLock); 602 PlaybackThread *thread = checkPlaybackThread_l(output); 603 if (thread == NULL) { 604 ALOGE("no playback thread found for output handle %d", output); 605 lStatus = BAD_VALUE; 606 goto Exit; 607 } 608 609 pid_t pid = IPCThreadState::self()->getCallingPid(); 610 client = registerPid(pid); 611 612 PlaybackThread *effectThread = NULL; 613 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 614 lSessionId = *sessionId; 615 // check if an effect chain with the same session ID is present on another 616 // output thread and move it here. 617 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 618 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 619 if (mPlaybackThreads.keyAt(i) != output) { 620 uint32_t sessions = t->hasAudioSession(lSessionId); 621 if (sessions & PlaybackThread::EFFECT_SESSION) { 622 effectThread = t.get(); 623 break; 624 } 625 } 626 } 627 } else { 628 // if no audio session id is provided, create one here 629 lSessionId = nextUniqueId(); 630 if (sessionId != NULL) { 631 *sessionId = lSessionId; 632 } 633 } 634 ALOGV("createTrack() lSessionId: %d", lSessionId); 635 636 track = thread->createTrack_l(client, streamType, sampleRate, format, 637 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 638 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 639 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 640 641 // move effect chain to this output thread if an effect on same session was waiting 642 // for a track to be created 643 if (lStatus == NO_ERROR && effectThread != NULL) { 644 // no risk of deadlock because AudioFlinger::mLock is held 645 Mutex::Autolock _dl(thread->mLock); 646 Mutex::Autolock _sl(effectThread->mLock); 647 moveEffectChain_l(lSessionId, effectThread, thread, true); 648 } 649 650 // Look for sync events awaiting for a session to be used. 651 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 652 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 653 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 654 if (lStatus == NO_ERROR) { 655 (void) track->setSyncEvent(mPendingSyncEvents[i]); 656 } else { 657 mPendingSyncEvents[i]->cancel(); 658 } 659 mPendingSyncEvents.removeAt(i); 660 i--; 661 } 662 } 663 } 664 665 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 666 } 667 668 if (lStatus != NO_ERROR) { 669 // remove local strong reference to Client before deleting the Track so that the 670 // Client destructor is called by the TrackBase destructor with mClientLock held 671 // Don't hold mClientLock when releasing the reference on the track as the 672 // destructor will acquire it. 673 { 674 Mutex::Autolock _cl(mClientLock); 675 client.clear(); 676 } 677 track.clear(); 678 goto Exit; 679 } 680 681 // return handle to client 682 trackHandle = new TrackHandle(track); 683 684Exit: 685 *status = lStatus; 686 return trackHandle; 687} 688 689uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 690{ 691 Mutex::Autolock _l(mLock); 692 PlaybackThread *thread = checkPlaybackThread_l(output); 693 if (thread == NULL) { 694 ALOGW("sampleRate() unknown thread %d", output); 695 return 0; 696 } 697 return thread->sampleRate(); 698} 699 700audio_format_t AudioFlinger::format(audio_io_handle_t output) const 701{ 702 Mutex::Autolock _l(mLock); 703 PlaybackThread *thread = checkPlaybackThread_l(output); 704 if (thread == NULL) { 705 ALOGW("format() unknown thread %d", output); 706 return AUDIO_FORMAT_INVALID; 707 } 708 return thread->format(); 709} 710 711size_t AudioFlinger::frameCount(audio_io_handle_t output) const 712{ 713 Mutex::Autolock _l(mLock); 714 PlaybackThread *thread = checkPlaybackThread_l(output); 715 if (thread == NULL) { 716 ALOGW("frameCount() unknown thread %d", output); 717 return 0; 718 } 719 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 720 // should examine all callers and fix them to handle smaller counts 721 return thread->frameCount(); 722} 723 724uint32_t AudioFlinger::latency(audio_io_handle_t output) const 725{ 726 Mutex::Autolock _l(mLock); 727 PlaybackThread *thread = checkPlaybackThread_l(output); 728 if (thread == NULL) { 729 ALOGW("latency(): no playback thread found for output handle %d", output); 730 return 0; 731 } 732 return thread->latency(); 733} 734 735status_t AudioFlinger::setMasterVolume(float value) 736{ 737 status_t ret = initCheck(); 738 if (ret != NO_ERROR) { 739 return ret; 740 } 741 742 // check calling permissions 743 if (!settingsAllowed()) { 744 return PERMISSION_DENIED; 745 } 746 747 Mutex::Autolock _l(mLock); 748 mMasterVolume = value; 749 750 // Set master volume in the HALs which support it. 751 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 752 AutoMutex lock(mHardwareLock); 753 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 754 755 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 756 if (dev->canSetMasterVolume()) { 757 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 758 } 759 mHardwareStatus = AUDIO_HW_IDLE; 760 } 761 762 // Now set the master volume in each playback thread. Playback threads 763 // assigned to HALs which do not have master volume support will apply 764 // master volume during the mix operation. Threads with HALs which do 765 // support master volume will simply ignore the setting. 766 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 767 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 768 continue; 769 } 770 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 771 } 772 773 return NO_ERROR; 774} 775 776status_t AudioFlinger::setMode(audio_mode_t mode) 777{ 778 status_t ret = initCheck(); 779 if (ret != NO_ERROR) { 780 return ret; 781 } 782 783 // check calling permissions 784 if (!settingsAllowed()) { 785 return PERMISSION_DENIED; 786 } 787 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 788 ALOGW("Illegal value: setMode(%d)", mode); 789 return BAD_VALUE; 790 } 791 792 { // scope for the lock 793 AutoMutex lock(mHardwareLock); 794 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 795 mHardwareStatus = AUDIO_HW_SET_MODE; 796 ret = dev->set_mode(dev, mode); 797 mHardwareStatus = AUDIO_HW_IDLE; 798 } 799 800 if (NO_ERROR == ret) { 801 Mutex::Autolock _l(mLock); 802 mMode = mode; 803 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 804 mPlaybackThreads.valueAt(i)->setMode(mode); 805 } 806 807 return ret; 808} 809 810status_t AudioFlinger::setMicMute(bool state) 811{ 812 status_t ret = initCheck(); 813 if (ret != NO_ERROR) { 814 return ret; 815 } 816 817 // check calling permissions 818 if (!settingsAllowed()) { 819 return PERMISSION_DENIED; 820 } 821 822 AutoMutex lock(mHardwareLock); 823 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 824 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 825 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 826 status_t result = dev->set_mic_mute(dev, state); 827 if (result != NO_ERROR) { 828 ret = result; 829 } 830 } 831 mHardwareStatus = AUDIO_HW_IDLE; 832 return ret; 833} 834 835bool AudioFlinger::getMicMute() const 836{ 837 status_t ret = initCheck(); 838 if (ret != NO_ERROR) { 839 return false; 840 } 841 bool mute = true; 842 bool state = AUDIO_MODE_INVALID; 843 AutoMutex lock(mHardwareLock); 844 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 845 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 846 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 847 status_t result = dev->get_mic_mute(dev, &state); 848 if (result == NO_ERROR) { 849 mute = mute && state; 850 } 851 } 852 mHardwareStatus = AUDIO_HW_IDLE; 853 854 return mute; 855} 856 857status_t AudioFlinger::setMasterMute(bool muted) 858{ 859 status_t ret = initCheck(); 860 if (ret != NO_ERROR) { 861 return ret; 862 } 863 864 // check calling permissions 865 if (!settingsAllowed()) { 866 return PERMISSION_DENIED; 867 } 868 869 Mutex::Autolock _l(mLock); 870 mMasterMute = muted; 871 872 // Set master mute in the HALs which support it. 873 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 874 AutoMutex lock(mHardwareLock); 875 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 876 877 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 878 if (dev->canSetMasterMute()) { 879 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 880 } 881 mHardwareStatus = AUDIO_HW_IDLE; 882 } 883 884 // Now set the master mute in each playback thread. Playback threads 885 // assigned to HALs which do not have master mute support will apply master 886 // mute during the mix operation. Threads with HALs which do support master 887 // mute will simply ignore the setting. 888 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 889 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 890 continue; 891 } 892 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 893 } 894 895 return NO_ERROR; 896} 897 898float AudioFlinger::masterVolume() const 899{ 900 Mutex::Autolock _l(mLock); 901 return masterVolume_l(); 902} 903 904bool AudioFlinger::masterMute() const 905{ 906 Mutex::Autolock _l(mLock); 907 return masterMute_l(); 908} 909 910float AudioFlinger::masterVolume_l() const 911{ 912 return mMasterVolume; 913} 914 915bool AudioFlinger::masterMute_l() const 916{ 917 return mMasterMute; 918} 919 920status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 921{ 922 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 923 ALOGW("setStreamVolume() invalid stream %d", stream); 924 return BAD_VALUE; 925 } 926 pid_t caller = IPCThreadState::self()->getCallingPid(); 927 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 928 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 929 return PERMISSION_DENIED; 930 } 931 932 return NO_ERROR; 933} 934 935status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 936 audio_io_handle_t output) 937{ 938 // check calling permissions 939 if (!settingsAllowed()) { 940 return PERMISSION_DENIED; 941 } 942 943 status_t status = checkStreamType(stream); 944 if (status != NO_ERROR) { 945 return status; 946 } 947 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 948 949 AutoMutex lock(mLock); 950 PlaybackThread *thread = NULL; 951 if (output != AUDIO_IO_HANDLE_NONE) { 952 thread = checkPlaybackThread_l(output); 953 if (thread == NULL) { 954 return BAD_VALUE; 955 } 956 } 957 958 mStreamTypes[stream].volume = value; 959 960 if (thread == NULL) { 961 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 962 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 963 } 964 } else { 965 thread->setStreamVolume(stream, value); 966 } 967 968 return NO_ERROR; 969} 970 971status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 972{ 973 // check calling permissions 974 if (!settingsAllowed()) { 975 return PERMISSION_DENIED; 976 } 977 978 status_t status = checkStreamType(stream); 979 if (status != NO_ERROR) { 980 return status; 981 } 982 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 983 984 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 985 ALOGE("setStreamMute() invalid stream %d", stream); 986 return BAD_VALUE; 987 } 988 989 AutoMutex lock(mLock); 990 mStreamTypes[stream].mute = muted; 991 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 992 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 993 994 return NO_ERROR; 995} 996 997float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 998{ 999 status_t status = checkStreamType(stream); 1000 if (status != NO_ERROR) { 1001 return 0.0f; 1002 } 1003 1004 AutoMutex lock(mLock); 1005 float volume; 1006 if (output != AUDIO_IO_HANDLE_NONE) { 1007 PlaybackThread *thread = checkPlaybackThread_l(output); 1008 if (thread == NULL) { 1009 return 0.0f; 1010 } 1011 volume = thread->streamVolume(stream); 1012 } else { 1013 volume = streamVolume_l(stream); 1014 } 1015 1016 return volume; 1017} 1018 1019bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1020{ 1021 status_t status = checkStreamType(stream); 1022 if (status != NO_ERROR) { 1023 return true; 1024 } 1025 1026 AutoMutex lock(mLock); 1027 return streamMute_l(stream); 1028} 1029 1030 1031void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1032{ 1033 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1034 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1035 } 1036} 1037 1038status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1039{ 1040 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1041 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1042 1043 // check calling permissions 1044 if (!settingsAllowed()) { 1045 return PERMISSION_DENIED; 1046 } 1047 1048 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1049 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1050 Mutex::Autolock _l(mLock); 1051 status_t final_result = NO_ERROR; 1052 { 1053 AutoMutex lock(mHardwareLock); 1054 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1055 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1056 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1057 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1058 final_result = result ?: final_result; 1059 } 1060 mHardwareStatus = AUDIO_HW_IDLE; 1061 } 1062 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1063 AudioParameter param = AudioParameter(keyValuePairs); 1064 String8 value; 1065 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1066 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1067 if (mBtNrecIsOff != btNrecIsOff) { 1068 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1069 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1070 audio_devices_t device = thread->inDevice(); 1071 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1072 // collect all of the thread's session IDs 1073 KeyedVector<int, bool> ids = thread->sessionIds(); 1074 // suspend effects associated with those session IDs 1075 for (size_t j = 0; j < ids.size(); ++j) { 1076 int sessionId = ids.keyAt(j); 1077 thread->setEffectSuspended(FX_IID_AEC, 1078 suspend, 1079 sessionId); 1080 thread->setEffectSuspended(FX_IID_NS, 1081 suspend, 1082 sessionId); 1083 } 1084 } 1085 mBtNrecIsOff = btNrecIsOff; 1086 } 1087 } 1088 String8 screenState; 1089 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1090 bool isOff = screenState == "off"; 1091 if (isOff != (AudioFlinger::mScreenState & 1)) { 1092 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1093 } 1094 } 1095 return final_result; 1096 } 1097 1098 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1099 // and the thread is exited once the lock is released 1100 sp<ThreadBase> thread; 1101 { 1102 Mutex::Autolock _l(mLock); 1103 thread = checkPlaybackThread_l(ioHandle); 1104 if (thread == 0) { 1105 thread = checkRecordThread_l(ioHandle); 1106 } else if (thread == primaryPlaybackThread_l()) { 1107 // indicate output device change to all input threads for pre processing 1108 AudioParameter param = AudioParameter(keyValuePairs); 1109 int value; 1110 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1111 (value != 0)) { 1112 broacastParametersToRecordThreads_l(keyValuePairs); 1113 } 1114 } 1115 } 1116 if (thread != 0) { 1117 return thread->setParameters(keyValuePairs); 1118 } 1119 return BAD_VALUE; 1120} 1121 1122String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1123{ 1124 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1125 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1126 1127 Mutex::Autolock _l(mLock); 1128 1129 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1130 String8 out_s8; 1131 1132 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1133 char *s; 1134 { 1135 AutoMutex lock(mHardwareLock); 1136 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1137 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1138 s = dev->get_parameters(dev, keys.string()); 1139 mHardwareStatus = AUDIO_HW_IDLE; 1140 } 1141 out_s8 += String8(s ? s : ""); 1142 free(s); 1143 } 1144 return out_s8; 1145 } 1146 1147 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1148 if (playbackThread != NULL) { 1149 return playbackThread->getParameters(keys); 1150 } 1151 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1152 if (recordThread != NULL) { 1153 return recordThread->getParameters(keys); 1154 } 1155 return String8(""); 1156} 1157 1158size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1159 audio_channel_mask_t channelMask) const 1160{ 1161 status_t ret = initCheck(); 1162 if (ret != NO_ERROR) { 1163 return 0; 1164 } 1165 if ((sampleRate == 0) || 1166 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1167 !audio_is_input_channel(channelMask)) { 1168 return 0; 1169 } 1170 1171 AutoMutex lock(mHardwareLock); 1172 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1173 audio_config_t config, proposed; 1174 memset(&proposed, 0, sizeof(proposed)); 1175 proposed.sample_rate = sampleRate; 1176 proposed.channel_mask = channelMask; 1177 proposed.format = format; 1178 1179 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1180 size_t frames; 1181 for (;;) { 1182 // Note: config is currently a const parameter for get_input_buffer_size() 1183 // but we use a copy from proposed in case config changes from the call. 1184 config = proposed; 1185 frames = dev->get_input_buffer_size(dev, &config); 1186 if (frames != 0) { 1187 break; // hal success, config is the result 1188 } 1189 // change one parameter of the configuration each iteration to a more "common" value 1190 // to see if the device will support it. 1191 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1192 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1193 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1194 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1195 } else { 1196 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1197 "format %#x, channelMask 0x%X", 1198 sampleRate, format, channelMask); 1199 break; // retries failed, break out of loop with frames == 0. 1200 } 1201 } 1202 mHardwareStatus = AUDIO_HW_IDLE; 1203 if (frames > 0 && config.sample_rate != sampleRate) { 1204 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1205 } 1206 return frames; // may be converted to bytes at the Java level. 1207} 1208 1209uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1210{ 1211 Mutex::Autolock _l(mLock); 1212 1213 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1214 if (recordThread != NULL) { 1215 return recordThread->getInputFramesLost(); 1216 } 1217 return 0; 1218} 1219 1220status_t AudioFlinger::setVoiceVolume(float value) 1221{ 1222 status_t ret = initCheck(); 1223 if (ret != NO_ERROR) { 1224 return ret; 1225 } 1226 1227 // check calling permissions 1228 if (!settingsAllowed()) { 1229 return PERMISSION_DENIED; 1230 } 1231 1232 AutoMutex lock(mHardwareLock); 1233 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1234 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1235 ret = dev->set_voice_volume(dev, value); 1236 mHardwareStatus = AUDIO_HW_IDLE; 1237 1238 return ret; 1239} 1240 1241status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1242 audio_io_handle_t output) const 1243{ 1244 status_t status; 1245 1246 Mutex::Autolock _l(mLock); 1247 1248 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1249 if (playbackThread != NULL) { 1250 return playbackThread->getRenderPosition(halFrames, dspFrames); 1251 } 1252 1253 return BAD_VALUE; 1254} 1255 1256void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1257{ 1258 Mutex::Autolock _l(mLock); 1259 if (client == 0) { 1260 return; 1261 } 1262 pid_t pid = IPCThreadState::self()->getCallingPid(); 1263 { 1264 Mutex::Autolock _cl(mClientLock); 1265 if (mNotificationClients.indexOfKey(pid) < 0) { 1266 sp<NotificationClient> notificationClient = new NotificationClient(this, 1267 client, 1268 pid); 1269 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1270 1271 mNotificationClients.add(pid, notificationClient); 1272 1273 sp<IBinder> binder = IInterface::asBinder(client); 1274 binder->linkToDeath(notificationClient); 1275 } 1276 } 1277 1278 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1279 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1280 // the config change is always sent from playback or record threads to avoid deadlock 1281 // with AudioSystem::gLock 1282 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1283 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1284 } 1285 1286 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1287 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1288 } 1289} 1290 1291void AudioFlinger::removeNotificationClient(pid_t pid) 1292{ 1293 Mutex::Autolock _l(mLock); 1294 { 1295 Mutex::Autolock _cl(mClientLock); 1296 mNotificationClients.removeItem(pid); 1297 } 1298 1299 ALOGV("%d died, releasing its sessions", pid); 1300 size_t num = mAudioSessionRefs.size(); 1301 bool removed = false; 1302 for (size_t i = 0; i< num; ) { 1303 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1304 ALOGV(" pid %d @ %d", ref->mPid, i); 1305 if (ref->mPid == pid) { 1306 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1307 mAudioSessionRefs.removeAt(i); 1308 delete ref; 1309 removed = true; 1310 num--; 1311 } else { 1312 i++; 1313 } 1314 } 1315 if (removed) { 1316 purgeStaleEffects_l(); 1317 } 1318} 1319 1320void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1321 const sp<AudioIoDescriptor>& ioDesc, 1322 pid_t pid) 1323{ 1324 Mutex::Autolock _l(mClientLock); 1325 size_t size = mNotificationClients.size(); 1326 for (size_t i = 0; i < size; i++) { 1327 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1328 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1329 } 1330 } 1331} 1332 1333// removeClient_l() must be called with AudioFlinger::mClientLock held 1334void AudioFlinger::removeClient_l(pid_t pid) 1335{ 1336 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1337 IPCThreadState::self()->getCallingPid()); 1338 mClients.removeItem(pid); 1339} 1340 1341// getEffectThread_l() must be called with AudioFlinger::mLock held 1342sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1343{ 1344 sp<PlaybackThread> thread; 1345 1346 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1347 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1348 ALOG_ASSERT(thread == 0); 1349 thread = mPlaybackThreads.valueAt(i); 1350 } 1351 } 1352 1353 return thread; 1354} 1355 1356 1357 1358// ---------------------------------------------------------------------------- 1359 1360AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1361 : RefBase(), 1362 mAudioFlinger(audioFlinger), 1363 mPid(pid) 1364{ 1365 size_t heapSize = kClientSharedHeapSizeBytes; 1366 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1367 // invalidated tracks 1368 if (!audioFlinger->isLowRamDevice()) { 1369 heapSize *= kClientSharedHeapSizeMultiplier; 1370 } 1371 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1372} 1373 1374// Client destructor must be called with AudioFlinger::mClientLock held 1375AudioFlinger::Client::~Client() 1376{ 1377 mAudioFlinger->removeClient_l(mPid); 1378} 1379 1380sp<MemoryDealer> AudioFlinger::Client::heap() const 1381{ 1382 return mMemoryDealer; 1383} 1384 1385// ---------------------------------------------------------------------------- 1386 1387AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1388 const sp<IAudioFlingerClient>& client, 1389 pid_t pid) 1390 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1391{ 1392} 1393 1394AudioFlinger::NotificationClient::~NotificationClient() 1395{ 1396} 1397 1398void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1399{ 1400 sp<NotificationClient> keep(this); 1401 mAudioFlinger->removeNotificationClient(mPid); 1402} 1403 1404 1405// ---------------------------------------------------------------------------- 1406 1407static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1408 return audio_is_remote_submix_device(inDevice); 1409} 1410 1411sp<IAudioRecord> AudioFlinger::openRecord( 1412 audio_io_handle_t input, 1413 uint32_t sampleRate, 1414 audio_format_t format, 1415 audio_channel_mask_t channelMask, 1416 const String16& opPackageName, 1417 size_t *frameCount, 1418 IAudioFlinger::track_flags_t *flags, 1419 pid_t tid, 1420 int clientUid, 1421 int *sessionId, 1422 size_t *notificationFrames, 1423 sp<IMemory>& cblk, 1424 sp<IMemory>& buffers, 1425 status_t *status) 1426{ 1427 sp<RecordThread::RecordTrack> recordTrack; 1428 sp<RecordHandle> recordHandle; 1429 sp<Client> client; 1430 status_t lStatus; 1431 int lSessionId; 1432 1433 cblk.clear(); 1434 buffers.clear(); 1435 1436 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1437 if (!isTrustedCallingUid(callingUid)) { 1438 ALOGW_IF((uid_t)clientUid != callingUid, 1439 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1440 clientUid = callingUid; 1441 } 1442 1443 // check calling permissions 1444 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1445 ALOGE("openRecord() permission denied: recording not allowed"); 1446 lStatus = PERMISSION_DENIED; 1447 goto Exit; 1448 } 1449 1450 // further sample rate checks are performed by createRecordTrack_l() 1451 if (sampleRate == 0) { 1452 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1453 lStatus = BAD_VALUE; 1454 goto Exit; 1455 } 1456 1457 // we don't yet support anything other than linear PCM 1458 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1459 ALOGE("openRecord() invalid format %#x", format); 1460 lStatus = BAD_VALUE; 1461 goto Exit; 1462 } 1463 1464 // further channel mask checks are performed by createRecordTrack_l() 1465 if (!audio_is_input_channel(channelMask)) { 1466 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1467 lStatus = BAD_VALUE; 1468 goto Exit; 1469 } 1470 1471 { 1472 Mutex::Autolock _l(mLock); 1473 RecordThread *thread = checkRecordThread_l(input); 1474 if (thread == NULL) { 1475 ALOGE("openRecord() checkRecordThread_l failed"); 1476 lStatus = BAD_VALUE; 1477 goto Exit; 1478 } 1479 1480 pid_t pid = IPCThreadState::self()->getCallingPid(); 1481 client = registerPid(pid); 1482 1483 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1484 lSessionId = *sessionId; 1485 } else { 1486 // if no audio session id is provided, create one here 1487 lSessionId = nextUniqueId(); 1488 if (sessionId != NULL) { 1489 *sessionId = lSessionId; 1490 } 1491 } 1492 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1493 1494 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1495 frameCount, lSessionId, notificationFrames, 1496 clientUid, flags, tid, &lStatus); 1497 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1498 1499 if (lStatus == NO_ERROR) { 1500 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1501 // session and move it to this thread. 1502 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1503 if (chain != 0) { 1504 Mutex::Autolock _l(thread->mLock); 1505 thread->addEffectChain_l(chain); 1506 } 1507 } 1508 } 1509 1510 if (lStatus != NO_ERROR) { 1511 // remove local strong reference to Client before deleting the RecordTrack so that the 1512 // Client destructor is called by the TrackBase destructor with mClientLock held 1513 // Don't hold mClientLock when releasing the reference on the track as the 1514 // destructor will acquire it. 1515 { 1516 Mutex::Autolock _cl(mClientLock); 1517 client.clear(); 1518 } 1519 recordTrack.clear(); 1520 goto Exit; 1521 } 1522 1523 cblk = recordTrack->getCblk(); 1524 buffers = recordTrack->getBuffers(); 1525 1526 // return handle to client 1527 recordHandle = new RecordHandle(recordTrack); 1528 1529Exit: 1530 *status = lStatus; 1531 return recordHandle; 1532} 1533 1534 1535 1536// ---------------------------------------------------------------------------- 1537 1538audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1539{ 1540 if (name == NULL) { 1541 return 0; 1542 } 1543 if (!settingsAllowed()) { 1544 return 0; 1545 } 1546 Mutex::Autolock _l(mLock); 1547 return loadHwModule_l(name); 1548} 1549 1550// loadHwModule_l() must be called with AudioFlinger::mLock held 1551audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1552{ 1553 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1554 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1555 ALOGW("loadHwModule() module %s already loaded", name); 1556 return mAudioHwDevs.keyAt(i); 1557 } 1558 } 1559 1560 audio_hw_device_t *dev; 1561 1562 int rc = load_audio_interface(name, &dev); 1563 if (rc) { 1564 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1565 return 0; 1566 } 1567 1568 mHardwareStatus = AUDIO_HW_INIT; 1569 rc = dev->init_check(dev); 1570 mHardwareStatus = AUDIO_HW_IDLE; 1571 if (rc) { 1572 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1573 return 0; 1574 } 1575 1576 // Check and cache this HAL's level of support for master mute and master 1577 // volume. If this is the first HAL opened, and it supports the get 1578 // methods, use the initial values provided by the HAL as the current 1579 // master mute and volume settings. 1580 1581 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1582 { // scope for auto-lock pattern 1583 AutoMutex lock(mHardwareLock); 1584 1585 if (0 == mAudioHwDevs.size()) { 1586 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1587 if (NULL != dev->get_master_volume) { 1588 float mv; 1589 if (OK == dev->get_master_volume(dev, &mv)) { 1590 mMasterVolume = mv; 1591 } 1592 } 1593 1594 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1595 if (NULL != dev->get_master_mute) { 1596 bool mm; 1597 if (OK == dev->get_master_mute(dev, &mm)) { 1598 mMasterMute = mm; 1599 } 1600 } 1601 } 1602 1603 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1604 if ((NULL != dev->set_master_volume) && 1605 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1606 flags = static_cast<AudioHwDevice::Flags>(flags | 1607 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1608 } 1609 1610 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1611 if ((NULL != dev->set_master_mute) && 1612 (OK == dev->set_master_mute(dev, mMasterMute))) { 1613 flags = static_cast<AudioHwDevice::Flags>(flags | 1614 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1615 } 1616 1617 mHardwareStatus = AUDIO_HW_IDLE; 1618 } 1619 1620 audio_module_handle_t handle = nextUniqueId(); 1621 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1622 1623 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1624 name, dev->common.module->name, dev->common.module->id, handle); 1625 1626 return handle; 1627 1628} 1629 1630// ---------------------------------------------------------------------------- 1631 1632uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1633{ 1634 Mutex::Autolock _l(mLock); 1635 PlaybackThread *thread = primaryPlaybackThread_l(); 1636 return thread != NULL ? thread->sampleRate() : 0; 1637} 1638 1639size_t AudioFlinger::getPrimaryOutputFrameCount() 1640{ 1641 Mutex::Autolock _l(mLock); 1642 PlaybackThread *thread = primaryPlaybackThread_l(); 1643 return thread != NULL ? thread->frameCountHAL() : 0; 1644} 1645 1646// ---------------------------------------------------------------------------- 1647 1648status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1649{ 1650 uid_t uid = IPCThreadState::self()->getCallingUid(); 1651 if (uid != AID_SYSTEM) { 1652 return PERMISSION_DENIED; 1653 } 1654 Mutex::Autolock _l(mLock); 1655 if (mIsDeviceTypeKnown) { 1656 return INVALID_OPERATION; 1657 } 1658 mIsLowRamDevice = isLowRamDevice; 1659 mIsDeviceTypeKnown = true; 1660 return NO_ERROR; 1661} 1662 1663audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1664{ 1665 Mutex::Autolock _l(mLock); 1666 1667 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1668 if (index >= 0) { 1669 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1670 mHwAvSyncIds.valueAt(index), sessionId); 1671 return mHwAvSyncIds.valueAt(index); 1672 } 1673 1674 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1675 if (dev == NULL) { 1676 return AUDIO_HW_SYNC_INVALID; 1677 } 1678 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1679 AudioParameter param = AudioParameter(String8(reply)); 1680 free(reply); 1681 1682 int value; 1683 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1684 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1685 return AUDIO_HW_SYNC_INVALID; 1686 } 1687 1688 // allow only one session for a given HW A/V sync ID. 1689 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1690 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1691 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1692 value, mHwAvSyncIds.keyAt(i)); 1693 mHwAvSyncIds.removeItemsAt(i); 1694 break; 1695 } 1696 } 1697 1698 mHwAvSyncIds.add(sessionId, value); 1699 1700 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1701 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1702 uint32_t sessions = thread->hasAudioSession(sessionId); 1703 if (sessions & PlaybackThread::TRACK_SESSION) { 1704 AudioParameter param = AudioParameter(); 1705 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1706 thread->setParameters(param.toString()); 1707 break; 1708 } 1709 } 1710 1711 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1712 return (audio_hw_sync_t)value; 1713} 1714 1715status_t AudioFlinger::systemReady() 1716{ 1717 Mutex::Autolock _l(mLock); 1718 ALOGI("%s", __FUNCTION__); 1719 if (mSystemReady) { 1720 ALOGW("%s called twice", __FUNCTION__); 1721 return NO_ERROR; 1722 } 1723 mSystemReady = true; 1724 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1725 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1726 thread->systemReady(); 1727 } 1728 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1729 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1730 thread->systemReady(); 1731 } 1732 return NO_ERROR; 1733} 1734 1735// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1736void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1737{ 1738 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1739 if (index >= 0) { 1740 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1741 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1742 AudioParameter param = AudioParameter(); 1743 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1744 thread->setParameters(param.toString()); 1745 } 1746} 1747 1748 1749// ---------------------------------------------------------------------------- 1750 1751 1752sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1753 audio_io_handle_t *output, 1754 audio_config_t *config, 1755 audio_devices_t devices, 1756 const String8& address, 1757 audio_output_flags_t flags) 1758{ 1759 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1760 if (outHwDev == NULL) { 1761 return 0; 1762 } 1763 1764 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1765 if (*output == AUDIO_IO_HANDLE_NONE) { 1766 *output = nextUniqueId(); 1767 } 1768 1769 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1770 1771 // FOR TESTING ONLY: 1772 // This if statement allows overriding the audio policy settings 1773 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1774 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1775 // Check only for Normal Mixing mode 1776 if (kEnableExtendedPrecision) { 1777 // Specify format (uncomment one below to choose) 1778 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1779 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1780 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1781 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1782 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1783 } 1784 if (kEnableExtendedChannels) { 1785 // Specify channel mask (uncomment one below to choose) 1786 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1787 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1788 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1789 } 1790 } 1791 1792 AudioStreamOut *outputStream = NULL; 1793 status_t status = outHwDev->openOutputStream( 1794 &outputStream, 1795 *output, 1796 devices, 1797 flags, 1798 config, 1799 address.string()); 1800 1801 mHardwareStatus = AUDIO_HW_IDLE; 1802 1803 if (status == NO_ERROR) { 1804 1805 PlaybackThread *thread; 1806 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1807 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1808 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1809 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1810 || !isValidPcmSinkFormat(config->format) 1811 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1812 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1813 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1814 } else { 1815 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1816 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1817 } 1818 mPlaybackThreads.add(*output, thread); 1819 return thread; 1820 } 1821 1822 return 0; 1823} 1824 1825status_t AudioFlinger::openOutput(audio_module_handle_t module, 1826 audio_io_handle_t *output, 1827 audio_config_t *config, 1828 audio_devices_t *devices, 1829 const String8& address, 1830 uint32_t *latencyMs, 1831 audio_output_flags_t flags) 1832{ 1833 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1834 module, 1835 (devices != NULL) ? *devices : 0, 1836 config->sample_rate, 1837 config->format, 1838 config->channel_mask, 1839 flags); 1840 1841 if (*devices == AUDIO_DEVICE_NONE) { 1842 return BAD_VALUE; 1843 } 1844 1845 Mutex::Autolock _l(mLock); 1846 1847 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1848 if (thread != 0) { 1849 *latencyMs = thread->latency(); 1850 1851 // notify client processes of the new output creation 1852 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1853 1854 // the first primary output opened designates the primary hw device 1855 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1856 ALOGI("Using module %d has the primary audio interface", module); 1857 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1858 1859 AutoMutex lock(mHardwareLock); 1860 mHardwareStatus = AUDIO_HW_SET_MODE; 1861 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1862 mHardwareStatus = AUDIO_HW_IDLE; 1863 } 1864 return NO_ERROR; 1865 } 1866 1867 return NO_INIT; 1868} 1869 1870audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1871 audio_io_handle_t output2) 1872{ 1873 Mutex::Autolock _l(mLock); 1874 MixerThread *thread1 = checkMixerThread_l(output1); 1875 MixerThread *thread2 = checkMixerThread_l(output2); 1876 1877 if (thread1 == NULL || thread2 == NULL) { 1878 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1879 output2); 1880 return AUDIO_IO_HANDLE_NONE; 1881 } 1882 1883 audio_io_handle_t id = nextUniqueId(); 1884 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1885 thread->addOutputTrack(thread2); 1886 mPlaybackThreads.add(id, thread); 1887 // notify client processes of the new output creation 1888 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1889 return id; 1890} 1891 1892status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1893{ 1894 return closeOutput_nonvirtual(output); 1895} 1896 1897status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1898{ 1899 // keep strong reference on the playback thread so that 1900 // it is not destroyed while exit() is executed 1901 sp<PlaybackThread> thread; 1902 { 1903 Mutex::Autolock _l(mLock); 1904 thread = checkPlaybackThread_l(output); 1905 if (thread == NULL) { 1906 return BAD_VALUE; 1907 } 1908 1909 ALOGV("closeOutput() %d", output); 1910 1911 if (thread->type() == ThreadBase::MIXER) { 1912 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1913 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1914 DuplicatingThread *dupThread = 1915 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1916 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1917 } 1918 } 1919 } 1920 1921 1922 mPlaybackThreads.removeItem(output); 1923 // save all effects to the default thread 1924 if (mPlaybackThreads.size()) { 1925 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1926 if (dstThread != NULL) { 1927 // audioflinger lock is held here so the acquisition order of thread locks does not 1928 // matter 1929 Mutex::Autolock _dl(dstThread->mLock); 1930 Mutex::Autolock _sl(thread->mLock); 1931 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1932 for (size_t i = 0; i < effectChains.size(); i ++) { 1933 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1934 } 1935 } 1936 } 1937 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1938 ioDesc->mIoHandle = output; 1939 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1940 } 1941 thread->exit(); 1942 // The thread entity (active unit of execution) is no longer running here, 1943 // but the ThreadBase container still exists. 1944 1945 if (!thread->isDuplicating()) { 1946 closeOutputFinish(thread); 1947 } 1948 1949 return NO_ERROR; 1950} 1951 1952void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1953{ 1954 AudioStreamOut *out = thread->clearOutput(); 1955 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1956 // from now on thread->mOutput is NULL 1957 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1958 delete out; 1959} 1960 1961void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1962{ 1963 mPlaybackThreads.removeItem(thread->mId); 1964 thread->exit(); 1965 closeOutputFinish(thread); 1966} 1967 1968status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1969{ 1970 Mutex::Autolock _l(mLock); 1971 PlaybackThread *thread = checkPlaybackThread_l(output); 1972 1973 if (thread == NULL) { 1974 return BAD_VALUE; 1975 } 1976 1977 ALOGV("suspendOutput() %d", output); 1978 thread->suspend(); 1979 1980 return NO_ERROR; 1981} 1982 1983status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1984{ 1985 Mutex::Autolock _l(mLock); 1986 PlaybackThread *thread = checkPlaybackThread_l(output); 1987 1988 if (thread == NULL) { 1989 return BAD_VALUE; 1990 } 1991 1992 ALOGV("restoreOutput() %d", output); 1993 1994 thread->restore(); 1995 1996 return NO_ERROR; 1997} 1998 1999status_t AudioFlinger::openInput(audio_module_handle_t module, 2000 audio_io_handle_t *input, 2001 audio_config_t *config, 2002 audio_devices_t *devices, 2003 const String8& address, 2004 audio_source_t source, 2005 audio_input_flags_t flags) 2006{ 2007 Mutex::Autolock _l(mLock); 2008 2009 if (*devices == AUDIO_DEVICE_NONE) { 2010 return BAD_VALUE; 2011 } 2012 2013 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2014 2015 if (thread != 0) { 2016 // notify client processes of the new input creation 2017 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2018 return NO_ERROR; 2019 } 2020 return NO_INIT; 2021} 2022 2023sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2024 audio_io_handle_t *input, 2025 audio_config_t *config, 2026 audio_devices_t devices, 2027 const String8& address, 2028 audio_source_t source, 2029 audio_input_flags_t flags) 2030{ 2031 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2032 if (inHwDev == NULL) { 2033 *input = AUDIO_IO_HANDLE_NONE; 2034 return 0; 2035 } 2036 2037 if (*input == AUDIO_IO_HANDLE_NONE) { 2038 *input = nextUniqueId(); 2039 } 2040 2041 audio_config_t halconfig = *config; 2042 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2043 audio_stream_in_t *inStream = NULL; 2044 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2045 &inStream, flags, address.string(), source); 2046 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2047 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2048 inStream, 2049 halconfig.sample_rate, 2050 halconfig.format, 2051 halconfig.channel_mask, 2052 flags, 2053 status, address.string()); 2054 2055 // If the input could not be opened with the requested parameters and we can handle the 2056 // conversion internally, try to open again with the proposed parameters. 2057 if (status == BAD_VALUE && 2058 audio_is_linear_pcm(config->format) && 2059 audio_is_linear_pcm(halconfig.format) && 2060 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2061 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2062 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2063 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2064 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2065 inStream = NULL; 2066 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2067 &inStream, flags, address.string(), source); 2068 // FIXME log this new status; HAL should not propose any further changes 2069 } 2070 2071 if (status == NO_ERROR && inStream != NULL) { 2072 2073#ifdef TEE_SINK 2074 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2075 // or (re-)create if current Pipe is idle and does not match the new format 2076 sp<NBAIO_Sink> teeSink; 2077 enum { 2078 TEE_SINK_NO, // don't copy input 2079 TEE_SINK_NEW, // copy input using a new pipe 2080 TEE_SINK_OLD, // copy input using an existing pipe 2081 } kind; 2082 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2083 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2084 if (!mTeeSinkInputEnabled) { 2085 kind = TEE_SINK_NO; 2086 } else if (!Format_isValid(format)) { 2087 kind = TEE_SINK_NO; 2088 } else if (mRecordTeeSink == 0) { 2089 kind = TEE_SINK_NEW; 2090 } else if (mRecordTeeSink->getStrongCount() != 1) { 2091 kind = TEE_SINK_NO; 2092 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2093 kind = TEE_SINK_OLD; 2094 } else { 2095 kind = TEE_SINK_NEW; 2096 } 2097 switch (kind) { 2098 case TEE_SINK_NEW: { 2099 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2100 size_t numCounterOffers = 0; 2101 const NBAIO_Format offers[1] = {format}; 2102 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2103 ALOG_ASSERT(index == 0); 2104 PipeReader *pipeReader = new PipeReader(*pipe); 2105 numCounterOffers = 0; 2106 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2107 ALOG_ASSERT(index == 0); 2108 mRecordTeeSink = pipe; 2109 mRecordTeeSource = pipeReader; 2110 teeSink = pipe; 2111 } 2112 break; 2113 case TEE_SINK_OLD: 2114 teeSink = mRecordTeeSink; 2115 break; 2116 case TEE_SINK_NO: 2117 default: 2118 break; 2119 } 2120#endif 2121 2122 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2123 2124 // Start record thread 2125 // RecordThread requires both input and output device indication to forward to audio 2126 // pre processing modules 2127 sp<RecordThread> thread = new RecordThread(this, 2128 inputStream, 2129 *input, 2130 primaryOutputDevice_l(), 2131 devices, 2132 mSystemReady 2133#ifdef TEE_SINK 2134 , teeSink 2135#endif 2136 ); 2137 mRecordThreads.add(*input, thread); 2138 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2139 return thread; 2140 } 2141 2142 *input = AUDIO_IO_HANDLE_NONE; 2143 return 0; 2144} 2145 2146status_t AudioFlinger::closeInput(audio_io_handle_t input) 2147{ 2148 return closeInput_nonvirtual(input); 2149} 2150 2151status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2152{ 2153 // keep strong reference on the record thread so that 2154 // it is not destroyed while exit() is executed 2155 sp<RecordThread> thread; 2156 { 2157 Mutex::Autolock _l(mLock); 2158 thread = checkRecordThread_l(input); 2159 if (thread == 0) { 2160 return BAD_VALUE; 2161 } 2162 2163 ALOGV("closeInput() %d", input); 2164 2165 // If we still have effect chains, it means that a client still holds a handle 2166 // on at least one effect. We must either move the chain to an existing thread with the 2167 // same session ID or put it aside in case a new record thread is opened for a 2168 // new capture on the same session 2169 sp<EffectChain> chain; 2170 { 2171 Mutex::Autolock _sl(thread->mLock); 2172 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2173 // Note: maximum one chain per record thread 2174 if (effectChains.size() != 0) { 2175 chain = effectChains[0]; 2176 } 2177 } 2178 if (chain != 0) { 2179 // first check if a record thread is already opened with a client on the same session. 2180 // This should only happen in case of overlap between one thread tear down and the 2181 // creation of its replacement 2182 size_t i; 2183 for (i = 0; i < mRecordThreads.size(); i++) { 2184 sp<RecordThread> t = mRecordThreads.valueAt(i); 2185 if (t == thread) { 2186 continue; 2187 } 2188 if (t->hasAudioSession(chain->sessionId()) != 0) { 2189 Mutex::Autolock _l(t->mLock); 2190 ALOGV("closeInput() found thread %d for effect session %d", 2191 t->id(), chain->sessionId()); 2192 t->addEffectChain_l(chain); 2193 break; 2194 } 2195 } 2196 // put the chain aside if we could not find a record thread with the same session id. 2197 if (i == mRecordThreads.size()) { 2198 putOrphanEffectChain_l(chain); 2199 } 2200 } 2201 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2202 ioDesc->mIoHandle = input; 2203 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2204 mRecordThreads.removeItem(input); 2205 } 2206 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2207 // we have a different lock for notification client 2208 closeInputFinish(thread); 2209 return NO_ERROR; 2210} 2211 2212void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2213{ 2214 thread->exit(); 2215 AudioStreamIn *in = thread->clearInput(); 2216 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2217 // from now on thread->mInput is NULL 2218 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2219 delete in; 2220} 2221 2222void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2223{ 2224 mRecordThreads.removeItem(thread->mId); 2225 closeInputFinish(thread); 2226} 2227 2228status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2229{ 2230 Mutex::Autolock _l(mLock); 2231 ALOGV("invalidateStream() stream %d", stream); 2232 2233 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2234 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2235 thread->invalidateTracks(stream); 2236 } 2237 2238 return NO_ERROR; 2239} 2240 2241 2242audio_unique_id_t AudioFlinger::newAudioUniqueId() 2243{ 2244 return nextUniqueId(); 2245} 2246 2247void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2248{ 2249 Mutex::Autolock _l(mLock); 2250 pid_t caller = IPCThreadState::self()->getCallingPid(); 2251 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2252 if (pid != -1 && (caller == getpid_cached)) { 2253 caller = pid; 2254 } 2255 2256 { 2257 Mutex::Autolock _cl(mClientLock); 2258 // Ignore requests received from processes not known as notification client. The request 2259 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2260 // called from a different pid leaving a stale session reference. Also we don't know how 2261 // to clear this reference if the client process dies. 2262 if (mNotificationClients.indexOfKey(caller) < 0) { 2263 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2264 return; 2265 } 2266 } 2267 2268 size_t num = mAudioSessionRefs.size(); 2269 for (size_t i = 0; i< num; i++) { 2270 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2271 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2272 ref->mCnt++; 2273 ALOGV(" incremented refcount to %d", ref->mCnt); 2274 return; 2275 } 2276 } 2277 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2278 ALOGV(" added new entry for %d", audioSession); 2279} 2280 2281void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2282{ 2283 Mutex::Autolock _l(mLock); 2284 pid_t caller = IPCThreadState::self()->getCallingPid(); 2285 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2286 if (pid != -1 && (caller == getpid_cached)) { 2287 caller = pid; 2288 } 2289 size_t num = mAudioSessionRefs.size(); 2290 for (size_t i = 0; i< num; i++) { 2291 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2292 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2293 ref->mCnt--; 2294 ALOGV(" decremented refcount to %d", ref->mCnt); 2295 if (ref->mCnt == 0) { 2296 mAudioSessionRefs.removeAt(i); 2297 delete ref; 2298 purgeStaleEffects_l(); 2299 } 2300 return; 2301 } 2302 } 2303 // If the caller is mediaserver it is likely that the session being released was acquired 2304 // on behalf of a process not in notification clients and we ignore the warning. 2305 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2306} 2307 2308void AudioFlinger::purgeStaleEffects_l() { 2309 2310 ALOGV("purging stale effects"); 2311 2312 Vector< sp<EffectChain> > chains; 2313 2314 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2315 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2316 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2317 sp<EffectChain> ec = t->mEffectChains[j]; 2318 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2319 chains.push(ec); 2320 } 2321 } 2322 } 2323 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2324 sp<RecordThread> t = mRecordThreads.valueAt(i); 2325 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2326 sp<EffectChain> ec = t->mEffectChains[j]; 2327 chains.push(ec); 2328 } 2329 } 2330 2331 for (size_t i = 0; i < chains.size(); i++) { 2332 sp<EffectChain> ec = chains[i]; 2333 int sessionid = ec->sessionId(); 2334 sp<ThreadBase> t = ec->mThread.promote(); 2335 if (t == 0) { 2336 continue; 2337 } 2338 size_t numsessionrefs = mAudioSessionRefs.size(); 2339 bool found = false; 2340 for (size_t k = 0; k < numsessionrefs; k++) { 2341 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2342 if (ref->mSessionid == sessionid) { 2343 ALOGV(" session %d still exists for %d with %d refs", 2344 sessionid, ref->mPid, ref->mCnt); 2345 found = true; 2346 break; 2347 } 2348 } 2349 if (!found) { 2350 Mutex::Autolock _l(t->mLock); 2351 // remove all effects from the chain 2352 while (ec->mEffects.size()) { 2353 sp<EffectModule> effect = ec->mEffects[0]; 2354 effect->unPin(); 2355 t->removeEffect_l(effect); 2356 if (effect->purgeHandles()) { 2357 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2358 } 2359 AudioSystem::unregisterEffect(effect->id()); 2360 } 2361 } 2362 } 2363 return; 2364} 2365 2366// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2367AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2368{ 2369 return mPlaybackThreads.valueFor(output).get(); 2370} 2371 2372// checkMixerThread_l() must be called with AudioFlinger::mLock held 2373AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2374{ 2375 PlaybackThread *thread = checkPlaybackThread_l(output); 2376 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2377} 2378 2379// checkRecordThread_l() must be called with AudioFlinger::mLock held 2380AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2381{ 2382 return mRecordThreads.valueFor(input).get(); 2383} 2384 2385uint32_t AudioFlinger::nextUniqueId() 2386{ 2387 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2388} 2389 2390AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2391{ 2392 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2393 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2394 if(thread->isDuplicating()) { 2395 continue; 2396 } 2397 AudioStreamOut *output = thread->getOutput(); 2398 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2399 return thread; 2400 } 2401 } 2402 return NULL; 2403} 2404 2405audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2406{ 2407 PlaybackThread *thread = primaryPlaybackThread_l(); 2408 2409 if (thread == NULL) { 2410 return 0; 2411 } 2412 2413 return thread->outDevice(); 2414} 2415 2416sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2417 int triggerSession, 2418 int listenerSession, 2419 sync_event_callback_t callBack, 2420 wp<RefBase> cookie) 2421{ 2422 Mutex::Autolock _l(mLock); 2423 2424 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2425 status_t playStatus = NAME_NOT_FOUND; 2426 status_t recStatus = NAME_NOT_FOUND; 2427 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2428 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2429 if (playStatus == NO_ERROR) { 2430 return event; 2431 } 2432 } 2433 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2434 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2435 if (recStatus == NO_ERROR) { 2436 return event; 2437 } 2438 } 2439 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2440 mPendingSyncEvents.add(event); 2441 } else { 2442 ALOGV("createSyncEvent() invalid event %d", event->type()); 2443 event.clear(); 2444 } 2445 return event; 2446} 2447 2448// ---------------------------------------------------------------------------- 2449// Effect management 2450// ---------------------------------------------------------------------------- 2451 2452 2453status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2454{ 2455 Mutex::Autolock _l(mLock); 2456 return EffectQueryNumberEffects(numEffects); 2457} 2458 2459status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2460{ 2461 Mutex::Autolock _l(mLock); 2462 return EffectQueryEffect(index, descriptor); 2463} 2464 2465status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2466 effect_descriptor_t *descriptor) const 2467{ 2468 Mutex::Autolock _l(mLock); 2469 return EffectGetDescriptor(pUuid, descriptor); 2470} 2471 2472 2473sp<IEffect> AudioFlinger::createEffect( 2474 effect_descriptor_t *pDesc, 2475 const sp<IEffectClient>& effectClient, 2476 int32_t priority, 2477 audio_io_handle_t io, 2478 int sessionId, 2479 const String16& opPackageName, 2480 status_t *status, 2481 int *id, 2482 int *enabled) 2483{ 2484 status_t lStatus = NO_ERROR; 2485 sp<EffectHandle> handle; 2486 effect_descriptor_t desc; 2487 2488 pid_t pid = IPCThreadState::self()->getCallingPid(); 2489 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2490 pid, effectClient.get(), priority, sessionId, io); 2491 2492 if (pDesc == NULL) { 2493 lStatus = BAD_VALUE; 2494 goto Exit; 2495 } 2496 2497 // check audio settings permission for global effects 2498 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2499 lStatus = PERMISSION_DENIED; 2500 goto Exit; 2501 } 2502 2503 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2504 // that can only be created by audio policy manager (running in same process) 2505 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2506 lStatus = PERMISSION_DENIED; 2507 goto Exit; 2508 } 2509 2510 { 2511 if (!EffectIsNullUuid(&pDesc->uuid)) { 2512 // if uuid is specified, request effect descriptor 2513 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2514 if (lStatus < 0) { 2515 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2516 goto Exit; 2517 } 2518 } else { 2519 // if uuid is not specified, look for an available implementation 2520 // of the required type in effect factory 2521 if (EffectIsNullUuid(&pDesc->type)) { 2522 ALOGW("createEffect() no effect type"); 2523 lStatus = BAD_VALUE; 2524 goto Exit; 2525 } 2526 uint32_t numEffects = 0; 2527 effect_descriptor_t d; 2528 d.flags = 0; // prevent compiler warning 2529 bool found = false; 2530 2531 lStatus = EffectQueryNumberEffects(&numEffects); 2532 if (lStatus < 0) { 2533 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2534 goto Exit; 2535 } 2536 for (uint32_t i = 0; i < numEffects; i++) { 2537 lStatus = EffectQueryEffect(i, &desc); 2538 if (lStatus < 0) { 2539 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2540 continue; 2541 } 2542 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2543 // If matching type found save effect descriptor. If the session is 2544 // 0 and the effect is not auxiliary, continue enumeration in case 2545 // an auxiliary version of this effect type is available 2546 found = true; 2547 d = desc; 2548 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2549 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2550 break; 2551 } 2552 } 2553 } 2554 if (!found) { 2555 lStatus = BAD_VALUE; 2556 ALOGW("createEffect() effect not found"); 2557 goto Exit; 2558 } 2559 // For same effect type, chose auxiliary version over insert version if 2560 // connect to output mix (Compliance to OpenSL ES) 2561 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2562 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2563 desc = d; 2564 } 2565 } 2566 2567 // Do not allow auxiliary effects on a session different from 0 (output mix) 2568 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2569 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2570 lStatus = INVALID_OPERATION; 2571 goto Exit; 2572 } 2573 2574 // check recording permission for visualizer 2575 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2576 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2577 lStatus = PERMISSION_DENIED; 2578 goto Exit; 2579 } 2580 2581 // return effect descriptor 2582 *pDesc = desc; 2583 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2584 // if the output returned by getOutputForEffect() is removed before we lock the 2585 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2586 // and we will exit safely 2587 io = AudioSystem::getOutputForEffect(&desc); 2588 ALOGV("createEffect got output %d", io); 2589 } 2590 2591 Mutex::Autolock _l(mLock); 2592 2593 // If output is not specified try to find a matching audio session ID in one of the 2594 // output threads. 2595 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2596 // because of code checking output when entering the function. 2597 // Note: io is never 0 when creating an effect on an input 2598 if (io == AUDIO_IO_HANDLE_NONE) { 2599 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2600 // output must be specified by AudioPolicyManager when using session 2601 // AUDIO_SESSION_OUTPUT_STAGE 2602 lStatus = BAD_VALUE; 2603 goto Exit; 2604 } 2605 // look for the thread where the specified audio session is present 2606 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2607 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2608 io = mPlaybackThreads.keyAt(i); 2609 break; 2610 } 2611 } 2612 if (io == 0) { 2613 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2614 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2615 io = mRecordThreads.keyAt(i); 2616 break; 2617 } 2618 } 2619 } 2620 // If no output thread contains the requested session ID, default to 2621 // first output. The effect chain will be moved to the correct output 2622 // thread when a track with the same session ID is created 2623 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2624 io = mPlaybackThreads.keyAt(0); 2625 } 2626 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2627 } 2628 ThreadBase *thread = checkRecordThread_l(io); 2629 if (thread == NULL) { 2630 thread = checkPlaybackThread_l(io); 2631 if (thread == NULL) { 2632 ALOGE("createEffect() unknown output thread"); 2633 lStatus = BAD_VALUE; 2634 goto Exit; 2635 } 2636 } else { 2637 // Check if one effect chain was awaiting for an effect to be created on this 2638 // session and used it instead of creating a new one. 2639 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2640 if (chain != 0) { 2641 Mutex::Autolock _l(thread->mLock); 2642 thread->addEffectChain_l(chain); 2643 } 2644 } 2645 2646 sp<Client> client = registerPid(pid); 2647 2648 // create effect on selected output thread 2649 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2650 &desc, enabled, &lStatus); 2651 if (handle != 0 && id != NULL) { 2652 *id = handle->id(); 2653 } 2654 if (handle == 0) { 2655 // remove local strong reference to Client with mClientLock held 2656 Mutex::Autolock _cl(mClientLock); 2657 client.clear(); 2658 } 2659 } 2660 2661Exit: 2662 *status = lStatus; 2663 return handle; 2664} 2665 2666status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2667 audio_io_handle_t dstOutput) 2668{ 2669 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2670 sessionId, srcOutput, dstOutput); 2671 Mutex::Autolock _l(mLock); 2672 if (srcOutput == dstOutput) { 2673 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2674 return NO_ERROR; 2675 } 2676 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2677 if (srcThread == NULL) { 2678 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2679 return BAD_VALUE; 2680 } 2681 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2682 if (dstThread == NULL) { 2683 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2684 return BAD_VALUE; 2685 } 2686 2687 Mutex::Autolock _dl(dstThread->mLock); 2688 Mutex::Autolock _sl(srcThread->mLock); 2689 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2690} 2691 2692// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2693status_t AudioFlinger::moveEffectChain_l(int sessionId, 2694 AudioFlinger::PlaybackThread *srcThread, 2695 AudioFlinger::PlaybackThread *dstThread, 2696 bool reRegister) 2697{ 2698 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2699 sessionId, srcThread, dstThread); 2700 2701 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2702 if (chain == 0) { 2703 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2704 sessionId, srcThread); 2705 return INVALID_OPERATION; 2706 } 2707 2708 // Check whether the destination thread has a channel count of FCC_2, which is 2709 // currently required for (most) effects. Prevent moving the effect chain here rather 2710 // than disabling the addEffect_l() call in dstThread below. 2711 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2712 dstThread->mChannelCount != FCC_2) { 2713 ALOGW("moveEffectChain_l() effect chain failed because" 2714 " destination thread %p channel count(%u) != %u", 2715 dstThread, dstThread->mChannelCount, FCC_2); 2716 return INVALID_OPERATION; 2717 } 2718 2719 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2720 // so that a new chain is created with correct parameters when first effect is added. This is 2721 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2722 // removed. 2723 srcThread->removeEffectChain_l(chain); 2724 2725 // transfer all effects one by one so that new effect chain is created on new thread with 2726 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2727 sp<EffectChain> dstChain; 2728 uint32_t strategy = 0; // prevent compiler warning 2729 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2730 Vector< sp<EffectModule> > removed; 2731 status_t status = NO_ERROR; 2732 while (effect != 0) { 2733 srcThread->removeEffect_l(effect); 2734 removed.add(effect); 2735 status = dstThread->addEffect_l(effect); 2736 if (status != NO_ERROR) { 2737 break; 2738 } 2739 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2740 if (effect->state() == EffectModule::ACTIVE || 2741 effect->state() == EffectModule::STOPPING) { 2742 effect->start(); 2743 } 2744 // if the move request is not received from audio policy manager, the effect must be 2745 // re-registered with the new strategy and output 2746 if (dstChain == 0) { 2747 dstChain = effect->chain().promote(); 2748 if (dstChain == 0) { 2749 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2750 status = NO_INIT; 2751 break; 2752 } 2753 strategy = dstChain->strategy(); 2754 } 2755 if (reRegister) { 2756 AudioSystem::unregisterEffect(effect->id()); 2757 AudioSystem::registerEffect(&effect->desc(), 2758 dstThread->id(), 2759 strategy, 2760 sessionId, 2761 effect->id()); 2762 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2763 } 2764 effect = chain->getEffectFromId_l(0); 2765 } 2766 2767 if (status != NO_ERROR) { 2768 for (size_t i = 0; i < removed.size(); i++) { 2769 srcThread->addEffect_l(removed[i]); 2770 if (dstChain != 0 && reRegister) { 2771 AudioSystem::unregisterEffect(removed[i]->id()); 2772 AudioSystem::registerEffect(&removed[i]->desc(), 2773 srcThread->id(), 2774 strategy, 2775 sessionId, 2776 removed[i]->id()); 2777 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2778 } 2779 } 2780 } 2781 2782 return status; 2783} 2784 2785bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2786{ 2787 if (mGlobalEffectEnableTime != 0 && 2788 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2789 return true; 2790 } 2791 2792 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2793 sp<EffectChain> ec = 2794 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2795 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2796 return true; 2797 } 2798 } 2799 return false; 2800} 2801 2802void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2803{ 2804 Mutex::Autolock _l(mLock); 2805 2806 mGlobalEffectEnableTime = systemTime(); 2807 2808 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2809 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2810 if (t->mType == ThreadBase::OFFLOAD) { 2811 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2812 } 2813 } 2814 2815} 2816 2817status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2818{ 2819 audio_session_t session = (audio_session_t)chain->sessionId(); 2820 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2821 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2822 if (index >= 0) { 2823 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2824 return ALREADY_EXISTS; 2825 } 2826 mOrphanEffectChains.add(session, chain); 2827 return NO_ERROR; 2828} 2829 2830sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2831{ 2832 sp<EffectChain> chain; 2833 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2834 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2835 if (index >= 0) { 2836 chain = mOrphanEffectChains.valueAt(index); 2837 mOrphanEffectChains.removeItemsAt(index); 2838 } 2839 return chain; 2840} 2841 2842bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2843{ 2844 Mutex::Autolock _l(mLock); 2845 audio_session_t session = (audio_session_t)effect->sessionId(); 2846 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2847 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2848 if (index >= 0) { 2849 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2850 if (chain->removeEffect_l(effect) == 0) { 2851 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2852 mOrphanEffectChains.removeItemsAt(index); 2853 } 2854 return true; 2855 } 2856 return false; 2857} 2858 2859 2860struct Entry { 2861#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2862 char mFileName[TEE_MAX_FILENAME]; 2863}; 2864 2865int comparEntry(const void *p1, const void *p2) 2866{ 2867 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2868} 2869 2870#ifdef TEE_SINK 2871void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2872{ 2873 NBAIO_Source *teeSource = source.get(); 2874 if (teeSource != NULL) { 2875 // .wav rotation 2876 // There is a benign race condition if 2 threads call this simultaneously. 2877 // They would both traverse the directory, but the result would simply be 2878 // failures at unlink() which are ignored. It's also unlikely since 2879 // normally dumpsys is only done by bugreport or from the command line. 2880 char teePath[32+256]; 2881 strcpy(teePath, "/data/misc/audioserver"); 2882 size_t teePathLen = strlen(teePath); 2883 DIR *dir = opendir(teePath); 2884 teePath[teePathLen++] = '/'; 2885 if (dir != NULL) { 2886#define TEE_MAX_SORT 20 // number of entries to sort 2887#define TEE_MAX_KEEP 10 // number of entries to keep 2888 struct Entry entries[TEE_MAX_SORT]; 2889 size_t entryCount = 0; 2890 while (entryCount < TEE_MAX_SORT) { 2891 struct dirent de; 2892 struct dirent *result = NULL; 2893 int rc = readdir_r(dir, &de, &result); 2894 if (rc != 0) { 2895 ALOGW("readdir_r failed %d", rc); 2896 break; 2897 } 2898 if (result == NULL) { 2899 break; 2900 } 2901 if (result != &de) { 2902 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2903 break; 2904 } 2905 // ignore non .wav file entries 2906 size_t nameLen = strlen(de.d_name); 2907 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2908 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2909 continue; 2910 } 2911 strcpy(entries[entryCount++].mFileName, de.d_name); 2912 } 2913 (void) closedir(dir); 2914 if (entryCount > TEE_MAX_KEEP) { 2915 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2916 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2917 strcpy(&teePath[teePathLen], entries[i].mFileName); 2918 (void) unlink(teePath); 2919 } 2920 } 2921 } else { 2922 if (fd >= 0) { 2923 dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath, 2924 strerror(errno)); 2925 } 2926 } 2927 char teeTime[16]; 2928 struct timeval tv; 2929 gettimeofday(&tv, NULL); 2930 struct tm tm; 2931 localtime_r(&tv.tv_sec, &tm); 2932 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2933 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2934 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2935 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2936 if (teeFd >= 0) { 2937 // FIXME use libsndfile 2938 char wavHeader[44]; 2939 memcpy(wavHeader, 2940 "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", 2941 sizeof(wavHeader)); 2942 NBAIO_Format format = teeSource->format(); 2943 unsigned channelCount = Format_channelCount(format); 2944 uint32_t sampleRate = Format_sampleRate(format); 2945 size_t frameSize = Format_frameSize(format); 2946 wavHeader[22] = channelCount; // number of channels 2947 wavHeader[24] = sampleRate; // sample rate 2948 wavHeader[25] = sampleRate >> 8; 2949 wavHeader[32] = frameSize; // block alignment 2950 wavHeader[33] = frameSize >> 8; 2951 write(teeFd, wavHeader, sizeof(wavHeader)); 2952 size_t total = 0; 2953 bool firstRead = true; 2954#define TEE_SINK_READ 1024 // frames per I/O operation 2955 void *buffer = malloc(TEE_SINK_READ * frameSize); 2956 for (;;) { 2957 size_t count = TEE_SINK_READ; 2958 ssize_t actual = teeSource->read(buffer, count); 2959 bool wasFirstRead = firstRead; 2960 firstRead = false; 2961 if (actual <= 0) { 2962 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2963 continue; 2964 } 2965 break; 2966 } 2967 ALOG_ASSERT(actual <= (ssize_t)count); 2968 write(teeFd, buffer, actual * frameSize); 2969 total += actual; 2970 } 2971 free(buffer); 2972 lseek(teeFd, (off_t) 4, SEEK_SET); 2973 uint32_t temp = 44 + total * frameSize - 8; 2974 // FIXME not big-endian safe 2975 write(teeFd, &temp, sizeof(temp)); 2976 lseek(teeFd, (off_t) 40, SEEK_SET); 2977 temp = total * frameSize; 2978 // FIXME not big-endian safe 2979 write(teeFd, &temp, sizeof(temp)); 2980 close(teeFd); 2981 if (fd >= 0) { 2982 dprintf(fd, "tee copied to %s\n", teePath); 2983 } 2984 } else { 2985 if (fd >= 0) { 2986 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2987 } 2988 } 2989 } 2990} 2991#endif 2992 2993// ---------------------------------------------------------------------------- 2994 2995status_t AudioFlinger::onTransact( 2996 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2997{ 2998 return BnAudioFlinger::onTransact(code, data, reply, flags); 2999} 3000 3001} // namespace android 3002