AudioFlinger.cpp revision eb9487e10294a4e73977f460f30eeaff503acd21
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 <common_time/cc_helper.h> 60 61#include <media/IMediaLogService.h> 62 63#include <media/nbaio/Pipe.h> 64#include <media/nbaio/PipeReader.h> 65#include <media/AudioParameter.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (format & AUDIO_FORMAT_MAIN_MASK) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 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 char value[PROPERTY_VALUE_MAX]; 188 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 189 if (doLog) { 190 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 191 MemoryHeapBase::READ_ONLY); 192 } 193 194#ifdef TEE_SINK 195 (void) property_get("ro.debuggable", value, "0"); 196 int debuggable = atoi(value); 197 int teeEnabled = 0; 198 if (debuggable) { 199 (void) property_get("af.tee", value, "0"); 200 teeEnabled = atoi(value); 201 } 202 // FIXME symbolic constants here 203 if (teeEnabled & 1) { 204 mTeeSinkInputEnabled = true; 205 } 206 if (teeEnabled & 2) { 207 mTeeSinkOutputEnabled = true; 208 } 209 if (teeEnabled & 4) { 210 mTeeSinkTrackEnabled = true; 211 } 212#endif 213} 214 215void AudioFlinger::onFirstRef() 216{ 217 int rc = 0; 218 219 Mutex::Autolock _l(mLock); 220 221 /* TODO: move all this work into an Init() function */ 222 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 223 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 224 uint32_t int_val; 225 if (1 == sscanf(val_str, "%u", &int_val)) { 226 mStandbyTimeInNsecs = milliseconds(int_val); 227 ALOGI("Using %u mSec as standby time.", int_val); 228 } else { 229 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 230 ALOGI("Using default %u mSec as standby time.", 231 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 232 } 233 } 234 235 mPatchPanel = new PatchPanel(this); 236 237 mMode = AUDIO_MODE_NORMAL; 238} 239 240AudioFlinger::~AudioFlinger() 241{ 242 while (!mRecordThreads.isEmpty()) { 243 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 244 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 245 } 246 while (!mPlaybackThreads.isEmpty()) { 247 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 248 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 249 } 250 251 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 252 // no mHardwareLock needed, as there are no other references to this 253 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 254 delete mAudioHwDevs.valueAt(i); 255 } 256 257 // Tell media.log service about any old writers that still need to be unregistered 258 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 259 if (binder != 0) { 260 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 261 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 262 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 263 mUnregisteredWriters.pop(); 264 mediaLogService->unregisterWriter(iMemory); 265 } 266 } 267 268} 269 270static const char * const audio_interfaces[] = { 271 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 272 AUDIO_HARDWARE_MODULE_ID_A2DP, 273 AUDIO_HARDWARE_MODULE_ID_USB, 274}; 275#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 276 277AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 278 audio_module_handle_t module, 279 audio_devices_t devices) 280{ 281 // if module is 0, the request comes from an old policy manager and we should load 282 // well known modules 283 if (module == 0) { 284 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 285 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 286 loadHwModule_l(audio_interfaces[i]); 287 } 288 // then try to find a module supporting the requested device. 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 291 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 292 if ((dev->get_supported_devices != NULL) && 293 (dev->get_supported_devices(dev) & devices) == devices) 294 return audioHwDevice; 295 } 296 } else { 297 // check a match for the requested module handle 298 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 299 if (audioHwDevice != NULL) { 300 return audioHwDevice; 301 } 302 } 303 304 return NULL; 305} 306 307void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Notification Clients:\n"); 323 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 324 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 325 result.append(buffer); 326 } 327 328 result.append("Global session refs:\n"); 329 result.append(" session pid count\n"); 330 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 331 AudioSessionRef *r = mAudioSessionRefs[i]; 332 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 333 result.append(buffer); 334 } 335 write(fd, result.string(), result.size()); 336} 337 338 339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 340{ 341 const size_t SIZE = 256; 342 char buffer[SIZE]; 343 String8 result; 344 hardware_call_state hardwareStatus = mHardwareStatus; 345 346 snprintf(buffer, SIZE, "Hardware status: %d\n" 347 "Standby Time mSec: %u\n", 348 hardwareStatus, 349 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 355{ 356 const size_t SIZE = 256; 357 char buffer[SIZE]; 358 String8 result; 359 snprintf(buffer, SIZE, "Permission Denial: " 360 "can't dump AudioFlinger from pid=%d, uid=%d\n", 361 IPCThreadState::self()->getCallingPid(), 362 IPCThreadState::self()->getCallingUid()); 363 result.append(buffer); 364 write(fd, result.string(), result.size()); 365} 366 367bool AudioFlinger::dumpTryLock(Mutex& mutex) 368{ 369 bool locked = false; 370 for (int i = 0; i < kDumpLockRetries; ++i) { 371 if (mutex.tryLock() == NO_ERROR) { 372 locked = true; 373 break; 374 } 375 usleep(kDumpLockSleepUs); 376 } 377 return locked; 378} 379 380status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 381{ 382 if (!dumpAllowed()) { 383 dumpPermissionDenial(fd, args); 384 } else { 385 // get state of hardware lock 386 bool hardwareLocked = dumpTryLock(mHardwareLock); 387 if (!hardwareLocked) { 388 String8 result(kHardwareLockedString); 389 write(fd, result.string(), result.size()); 390 } else { 391 mHardwareLock.unlock(); 392 } 393 394 bool locked = dumpTryLock(mLock); 395 396 // failed to lock - AudioFlinger is probably deadlocked 397 if (!locked) { 398 String8 result(kDeadlockedString); 399 write(fd, result.string(), result.size()); 400 } 401 402 bool clientLocked = dumpTryLock(mClientLock); 403 if (!clientLocked) { 404 String8 result(kClientLockedString); 405 write(fd, result.string(), result.size()); 406 } 407 408 EffectDumpEffects(fd); 409 410 dumpClients(fd, args); 411 if (clientLocked) { 412 mClientLock.unlock(); 413 } 414 415 dumpInternals(fd, args); 416 417 // dump playback threads 418 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 419 mPlaybackThreads.valueAt(i)->dump(fd, args); 420 } 421 422 // dump record threads 423 for (size_t i = 0; i < mRecordThreads.size(); i++) { 424 mRecordThreads.valueAt(i)->dump(fd, args); 425 } 426 427 // dump orphan effect chains 428 if (mOrphanEffectChains.size() != 0) { 429 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 430 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 431 mOrphanEffectChains.valueAt(i)->dump(fd, args); 432 } 433 } 434 // dump all hardware devs 435 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 436 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 437 dev->dump(dev, fd); 438 } 439 440#ifdef TEE_SINK 441 // dump the serially shared record tee sink 442 if (mRecordTeeSource != 0) { 443 dumpTee(fd, mRecordTeeSource); 444 } 445#endif 446 447 if (locked) { 448 mLock.unlock(); 449 } 450 451 // append a copy of media.log here by forwarding fd to it, but don't attempt 452 // to lookup the service if it's not running, as it will block for a second 453 if (mLogMemoryDealer != 0) { 454 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 455 if (binder != 0) { 456 dprintf(fd, "\nmedia.log:\n"); 457 Vector<String16> args; 458 binder->dump(fd, args); 459 } 460 } 461 } 462 return NO_ERROR; 463} 464 465sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 466{ 467 Mutex::Autolock _cl(mClientLock); 468 // If pid is already in the mClients wp<> map, then use that entry 469 // (for which promote() is always != 0), otherwise create a new entry and Client. 470 sp<Client> client = mClients.valueFor(pid).promote(); 471 if (client == 0) { 472 client = new Client(this, pid); 473 mClients.add(pid, client); 474 } 475 476 return client; 477} 478 479sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 480{ 481 // If there is no memory allocated for logs, return a dummy writer that does nothing 482 if (mLogMemoryDealer == 0) { 483 return new NBLog::Writer(); 484 } 485 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 486 // Similarly if we can't contact the media.log service, also return a dummy writer 487 if (binder == 0) { 488 return new NBLog::Writer(); 489 } 490 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 491 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 492 // If allocation fails, consult the vector of previously unregistered writers 493 // and garbage-collect one or more them until an allocation succeeds 494 if (shared == 0) { 495 Mutex::Autolock _l(mUnregisteredWritersLock); 496 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 497 { 498 // Pick the oldest stale writer to garbage-collect 499 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 500 mUnregisteredWriters.removeAt(0); 501 mediaLogService->unregisterWriter(iMemory); 502 // Now the media.log remote reference to IMemory is gone. When our last local 503 // reference to IMemory also drops to zero at end of this block, 504 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 505 } 506 // Re-attempt the allocation 507 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 508 if (shared != 0) { 509 goto success; 510 } 511 } 512 // Even after garbage-collecting all old writers, there is still not enough memory, 513 // so return a dummy writer 514 return new NBLog::Writer(); 515 } 516success: 517 mediaLogService->registerWriter(shared, size, name); 518 return new NBLog::Writer(size, shared); 519} 520 521void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 522{ 523 if (writer == 0) { 524 return; 525 } 526 sp<IMemory> iMemory(writer->getIMemory()); 527 if (iMemory == 0) { 528 return; 529 } 530 // Rather than removing the writer immediately, append it to a queue of old writers to 531 // be garbage-collected later. This allows us to continue to view old logs for a while. 532 Mutex::Autolock _l(mUnregisteredWritersLock); 533 mUnregisteredWriters.push(writer); 534} 535 536// IAudioFlinger interface 537 538 539sp<IAudioTrack> AudioFlinger::createTrack( 540 audio_stream_type_t streamType, 541 uint32_t sampleRate, 542 audio_format_t format, 543 audio_channel_mask_t channelMask, 544 size_t *frameCount, 545 IAudioFlinger::track_flags_t *flags, 546 const sp<IMemory>& sharedBuffer, 547 audio_io_handle_t output, 548 pid_t tid, 549 int *sessionId, 550 int clientUid, 551 status_t *status) 552{ 553 sp<PlaybackThread::Track> track; 554 sp<TrackHandle> trackHandle; 555 sp<Client> client; 556 status_t lStatus; 557 int lSessionId; 558 559 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 560 // but if someone uses binder directly they could bypass that and cause us to crash 561 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 562 ALOGE("createTrack() invalid stream type %d", streamType); 563 lStatus = BAD_VALUE; 564 goto Exit; 565 } 566 567 // further sample rate checks are performed by createTrack_l() depending on the thread type 568 if (sampleRate == 0) { 569 ALOGE("createTrack() invalid sample rate %u", sampleRate); 570 lStatus = BAD_VALUE; 571 goto Exit; 572 } 573 574 // further channel mask checks are performed by createTrack_l() depending on the thread type 575 if (!audio_is_output_channel(channelMask)) { 576 ALOGE("createTrack() invalid channel mask %#x", channelMask); 577 lStatus = BAD_VALUE; 578 goto Exit; 579 } 580 581 // further format checks are performed by createTrack_l() depending on the thread type 582 if (!audio_is_valid_format(format)) { 583 ALOGE("createTrack() invalid format %#x", format); 584 lStatus = BAD_VALUE; 585 goto Exit; 586 } 587 588 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 589 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 { 595 Mutex::Autolock _l(mLock); 596 PlaybackThread *thread = checkPlaybackThread_l(output); 597 if (thread == NULL) { 598 ALOGE("no playback thread found for output handle %d", output); 599 lStatus = BAD_VALUE; 600 goto Exit; 601 } 602 603 pid_t pid = IPCThreadState::self()->getCallingPid(); 604 client = registerPid(pid); 605 606 PlaybackThread *effectThread = NULL; 607 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 608 lSessionId = *sessionId; 609 // check if an effect chain with the same session ID is present on another 610 // output thread and move it here. 611 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 612 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 613 if (mPlaybackThreads.keyAt(i) != output) { 614 uint32_t sessions = t->hasAudioSession(lSessionId); 615 if (sessions & PlaybackThread::EFFECT_SESSION) { 616 effectThread = t.get(); 617 break; 618 } 619 } 620 } 621 } else { 622 // if no audio session id is provided, create one here 623 lSessionId = nextUniqueId(); 624 if (sessionId != NULL) { 625 *sessionId = lSessionId; 626 } 627 } 628 ALOGV("createTrack() lSessionId: %d", lSessionId); 629 630 track = thread->createTrack_l(client, streamType, sampleRate, format, 631 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 632 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 633 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 634 635 // move effect chain to this output thread if an effect on same session was waiting 636 // for a track to be created 637 if (lStatus == NO_ERROR && effectThread != NULL) { 638 // no risk of deadlock because AudioFlinger::mLock is held 639 Mutex::Autolock _dl(thread->mLock); 640 Mutex::Autolock _sl(effectThread->mLock); 641 moveEffectChain_l(lSessionId, effectThread, thread, true); 642 } 643 644 // Look for sync events awaiting for a session to be used. 645 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 646 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 647 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 648 if (lStatus == NO_ERROR) { 649 (void) track->setSyncEvent(mPendingSyncEvents[i]); 650 } else { 651 mPendingSyncEvents[i]->cancel(); 652 } 653 mPendingSyncEvents.removeAt(i); 654 i--; 655 } 656 } 657 } 658 659 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 660 } 661 662 if (lStatus != NO_ERROR) { 663 // remove local strong reference to Client before deleting the Track so that the 664 // Client destructor is called by the TrackBase destructor with mClientLock held 665 // Don't hold mClientLock when releasing the reference on the track as the 666 // destructor will acquire it. 667 { 668 Mutex::Autolock _cl(mClientLock); 669 client.clear(); 670 } 671 track.clear(); 672 goto Exit; 673 } 674 675 // return handle to client 676 trackHandle = new TrackHandle(track); 677 678Exit: 679 *status = lStatus; 680 return trackHandle; 681} 682 683uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 684{ 685 Mutex::Autolock _l(mLock); 686 PlaybackThread *thread = checkPlaybackThread_l(output); 687 if (thread == NULL) { 688 ALOGW("sampleRate() unknown thread %d", output); 689 return 0; 690 } 691 return thread->sampleRate(); 692} 693 694audio_format_t AudioFlinger::format(audio_io_handle_t output) const 695{ 696 Mutex::Autolock _l(mLock); 697 PlaybackThread *thread = checkPlaybackThread_l(output); 698 if (thread == NULL) { 699 ALOGW("format() unknown thread %d", output); 700 return AUDIO_FORMAT_INVALID; 701 } 702 return thread->format(); 703} 704 705size_t AudioFlinger::frameCount(audio_io_handle_t output) const 706{ 707 Mutex::Autolock _l(mLock); 708 PlaybackThread *thread = checkPlaybackThread_l(output); 709 if (thread == NULL) { 710 ALOGW("frameCount() unknown thread %d", output); 711 return 0; 712 } 713 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 714 // should examine all callers and fix them to handle smaller counts 715 return thread->frameCount(); 716} 717 718uint32_t AudioFlinger::latency(audio_io_handle_t output) const 719{ 720 Mutex::Autolock _l(mLock); 721 PlaybackThread *thread = checkPlaybackThread_l(output); 722 if (thread == NULL) { 723 ALOGW("latency(): no playback thread found for output handle %d", output); 724 return 0; 725 } 726 return thread->latency(); 727} 728 729status_t AudioFlinger::setMasterVolume(float value) 730{ 731 status_t ret = initCheck(); 732 if (ret != NO_ERROR) { 733 return ret; 734 } 735 736 // check calling permissions 737 if (!settingsAllowed()) { 738 return PERMISSION_DENIED; 739 } 740 741 Mutex::Autolock _l(mLock); 742 mMasterVolume = value; 743 744 // Set master volume in the HALs which support it. 745 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 746 AutoMutex lock(mHardwareLock); 747 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 748 749 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 750 if (dev->canSetMasterVolume()) { 751 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 752 } 753 mHardwareStatus = AUDIO_HW_IDLE; 754 } 755 756 // Now set the master volume in each playback thread. Playback threads 757 // assigned to HALs which do not have master volume support will apply 758 // master volume during the mix operation. Threads with HALs which do 759 // support master volume will simply ignore the setting. 760 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 761 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 762 continue; 763 } 764 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 765 } 766 767 return NO_ERROR; 768} 769 770status_t AudioFlinger::setMode(audio_mode_t mode) 771{ 772 status_t ret = initCheck(); 773 if (ret != NO_ERROR) { 774 return ret; 775 } 776 777 // check calling permissions 778 if (!settingsAllowed()) { 779 return PERMISSION_DENIED; 780 } 781 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 782 ALOGW("Illegal value: setMode(%d)", mode); 783 return BAD_VALUE; 784 } 785 786 { // scope for the lock 787 AutoMutex lock(mHardwareLock); 788 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 789 mHardwareStatus = AUDIO_HW_SET_MODE; 790 ret = dev->set_mode(dev, mode); 791 mHardwareStatus = AUDIO_HW_IDLE; 792 } 793 794 if (NO_ERROR == ret) { 795 Mutex::Autolock _l(mLock); 796 mMode = mode; 797 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 798 mPlaybackThreads.valueAt(i)->setMode(mode); 799 } 800 801 return ret; 802} 803 804status_t AudioFlinger::setMicMute(bool state) 805{ 806 status_t ret = initCheck(); 807 if (ret != NO_ERROR) { 808 return ret; 809 } 810 811 // check calling permissions 812 if (!settingsAllowed()) { 813 return PERMISSION_DENIED; 814 } 815 816 AutoMutex lock(mHardwareLock); 817 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 818 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 819 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 820 status_t result = dev->set_mic_mute(dev, state); 821 if (result != NO_ERROR) { 822 ret = result; 823 } 824 } 825 mHardwareStatus = AUDIO_HW_IDLE; 826 return ret; 827} 828 829bool AudioFlinger::getMicMute() const 830{ 831 status_t ret = initCheck(); 832 if (ret != NO_ERROR) { 833 return false; 834 } 835 bool mute = true; 836 bool state = AUDIO_MODE_INVALID; 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 841 status_t result = dev->get_mic_mute(dev, &state); 842 if (result == NO_ERROR) { 843 mute = mute && state; 844 } 845 } 846 mHardwareStatus = AUDIO_HW_IDLE; 847 848 return mute; 849} 850 851status_t AudioFlinger::setMasterMute(bool muted) 852{ 853 status_t ret = initCheck(); 854 if (ret != NO_ERROR) { 855 return ret; 856 } 857 858 // check calling permissions 859 if (!settingsAllowed()) { 860 return PERMISSION_DENIED; 861 } 862 863 Mutex::Autolock _l(mLock); 864 mMasterMute = muted; 865 866 // Set master mute in the HALs which support it. 867 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 868 AutoMutex lock(mHardwareLock); 869 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 870 871 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 872 if (dev->canSetMasterMute()) { 873 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 874 } 875 mHardwareStatus = AUDIO_HW_IDLE; 876 } 877 878 // Now set the master mute in each playback thread. Playback threads 879 // assigned to HALs which do not have master mute support will apply master 880 // mute during the mix operation. Threads with HALs which do support master 881 // mute will simply ignore the setting. 882 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 883 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 884 continue; 885 } 886 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 887 } 888 889 return NO_ERROR; 890} 891 892float AudioFlinger::masterVolume() const 893{ 894 Mutex::Autolock _l(mLock); 895 return masterVolume_l(); 896} 897 898bool AudioFlinger::masterMute() const 899{ 900 Mutex::Autolock _l(mLock); 901 return masterMute_l(); 902} 903 904float AudioFlinger::masterVolume_l() const 905{ 906 return mMasterVolume; 907} 908 909bool AudioFlinger::masterMute_l() const 910{ 911 return mMasterMute; 912} 913 914status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 915{ 916 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 917 ALOGW("setStreamVolume() invalid stream %d", stream); 918 return BAD_VALUE; 919 } 920 pid_t caller = IPCThreadState::self()->getCallingPid(); 921 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 922 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 923 return PERMISSION_DENIED; 924 } 925 926 return NO_ERROR; 927} 928 929status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 930 audio_io_handle_t output) 931{ 932 // check calling permissions 933 if (!settingsAllowed()) { 934 return PERMISSION_DENIED; 935 } 936 937 status_t status = checkStreamType(stream); 938 if (status != NO_ERROR) { 939 return status; 940 } 941 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 942 943 AutoMutex lock(mLock); 944 PlaybackThread *thread = NULL; 945 if (output != AUDIO_IO_HANDLE_NONE) { 946 thread = checkPlaybackThread_l(output); 947 if (thread == NULL) { 948 return BAD_VALUE; 949 } 950 } 951 952 mStreamTypes[stream].volume = value; 953 954 if (thread == NULL) { 955 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 956 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 957 } 958 } else { 959 thread->setStreamVolume(stream, value); 960 } 961 962 return NO_ERROR; 963} 964 965status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 966{ 967 // check calling permissions 968 if (!settingsAllowed()) { 969 return PERMISSION_DENIED; 970 } 971 972 status_t status = checkStreamType(stream); 973 if (status != NO_ERROR) { 974 return status; 975 } 976 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 977 978 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 979 ALOGE("setStreamMute() invalid stream %d", stream); 980 return BAD_VALUE; 981 } 982 983 AutoMutex lock(mLock); 984 mStreamTypes[stream].mute = muted; 985 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 986 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 987 988 return NO_ERROR; 989} 990 991float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 992{ 993 status_t status = checkStreamType(stream); 994 if (status != NO_ERROR) { 995 return 0.0f; 996 } 997 998 AutoMutex lock(mLock); 999 float volume; 1000 if (output != AUDIO_IO_HANDLE_NONE) { 1001 PlaybackThread *thread = checkPlaybackThread_l(output); 1002 if (thread == NULL) { 1003 return 0.0f; 1004 } 1005 volume = thread->streamVolume(stream); 1006 } else { 1007 volume = streamVolume_l(stream); 1008 } 1009 1010 return volume; 1011} 1012 1013bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1014{ 1015 status_t status = checkStreamType(stream); 1016 if (status != NO_ERROR) { 1017 return true; 1018 } 1019 1020 AutoMutex lock(mLock); 1021 return streamMute_l(stream); 1022} 1023 1024 1025void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1026{ 1027 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1028 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1029 } 1030} 1031 1032status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1033{ 1034 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1035 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1036 1037 // check calling permissions 1038 if (!settingsAllowed()) { 1039 return PERMISSION_DENIED; 1040 } 1041 1042 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1043 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1044 Mutex::Autolock _l(mLock); 1045 status_t final_result = NO_ERROR; 1046 { 1047 AutoMutex lock(mHardwareLock); 1048 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1049 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1050 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1051 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1052 final_result = result ?: final_result; 1053 } 1054 mHardwareStatus = AUDIO_HW_IDLE; 1055 } 1056 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1057 AudioParameter param = AudioParameter(keyValuePairs); 1058 String8 value; 1059 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1060 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1061 if (mBtNrecIsOff != btNrecIsOff) { 1062 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1063 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1064 audio_devices_t device = thread->inDevice(); 1065 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1066 // collect all of the thread's session IDs 1067 KeyedVector<int, bool> ids = thread->sessionIds(); 1068 // suspend effects associated with those session IDs 1069 for (size_t j = 0; j < ids.size(); ++j) { 1070 int sessionId = ids.keyAt(j); 1071 thread->setEffectSuspended(FX_IID_AEC, 1072 suspend, 1073 sessionId); 1074 thread->setEffectSuspended(FX_IID_NS, 1075 suspend, 1076 sessionId); 1077 } 1078 } 1079 mBtNrecIsOff = btNrecIsOff; 1080 } 1081 } 1082 String8 screenState; 1083 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1084 bool isOff = screenState == "off"; 1085 if (isOff != (AudioFlinger::mScreenState & 1)) { 1086 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1087 } 1088 } 1089 return final_result; 1090 } 1091 1092 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1093 // and the thread is exited once the lock is released 1094 sp<ThreadBase> thread; 1095 { 1096 Mutex::Autolock _l(mLock); 1097 thread = checkPlaybackThread_l(ioHandle); 1098 if (thread == 0) { 1099 thread = checkRecordThread_l(ioHandle); 1100 } else if (thread == primaryPlaybackThread_l()) { 1101 // indicate output device change to all input threads for pre processing 1102 AudioParameter param = AudioParameter(keyValuePairs); 1103 int value; 1104 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1105 (value != 0)) { 1106 broacastParametersToRecordThreads_l(keyValuePairs); 1107 } 1108 } 1109 } 1110 if (thread != 0) { 1111 return thread->setParameters(keyValuePairs); 1112 } 1113 return BAD_VALUE; 1114} 1115 1116String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1117{ 1118 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1119 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1120 1121 Mutex::Autolock _l(mLock); 1122 1123 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1124 String8 out_s8; 1125 1126 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1127 char *s; 1128 { 1129 AutoMutex lock(mHardwareLock); 1130 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1131 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1132 s = dev->get_parameters(dev, keys.string()); 1133 mHardwareStatus = AUDIO_HW_IDLE; 1134 } 1135 out_s8 += String8(s ? s : ""); 1136 free(s); 1137 } 1138 return out_s8; 1139 } 1140 1141 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1142 if (playbackThread != NULL) { 1143 return playbackThread->getParameters(keys); 1144 } 1145 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1146 if (recordThread != NULL) { 1147 return recordThread->getParameters(keys); 1148 } 1149 return String8(""); 1150} 1151 1152size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1153 audio_channel_mask_t channelMask) const 1154{ 1155 status_t ret = initCheck(); 1156 if (ret != NO_ERROR) { 1157 return 0; 1158 } 1159 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1160 return 0; 1161 } 1162 1163 AutoMutex lock(mHardwareLock); 1164 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1165 audio_config_t config, proposed; 1166 memset(&proposed, 0, sizeof(proposed)); 1167 proposed.sample_rate = sampleRate; 1168 proposed.channel_mask = channelMask; 1169 proposed.format = format; 1170 1171 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1172 size_t frames; 1173 for (;;) { 1174 // Note: config is currently a const parameter for get_input_buffer_size() 1175 // but we use a copy from proposed in case config changes from the call. 1176 config = proposed; 1177 frames = dev->get_input_buffer_size(dev, &config); 1178 if (frames != 0) { 1179 break; // hal success, config is the result 1180 } 1181 // change one parameter of the configuration each iteration to a more "common" value 1182 // to see if the device will support it. 1183 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1184 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1185 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1186 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1187 } else { 1188 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1189 "format %#x, channelMask 0x%X", 1190 sampleRate, format, channelMask); 1191 break; // retries failed, break out of loop with frames == 0. 1192 } 1193 } 1194 mHardwareStatus = AUDIO_HW_IDLE; 1195 if (frames > 0 && config.sample_rate != sampleRate) { 1196 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1197 } 1198 return frames; // may be converted to bytes at the Java level. 1199} 1200 1201uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1202{ 1203 Mutex::Autolock _l(mLock); 1204 1205 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1206 if (recordThread != NULL) { 1207 return recordThread->getInputFramesLost(); 1208 } 1209 return 0; 1210} 1211 1212status_t AudioFlinger::setVoiceVolume(float value) 1213{ 1214 status_t ret = initCheck(); 1215 if (ret != NO_ERROR) { 1216 return ret; 1217 } 1218 1219 // check calling permissions 1220 if (!settingsAllowed()) { 1221 return PERMISSION_DENIED; 1222 } 1223 1224 AutoMutex lock(mHardwareLock); 1225 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1226 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1227 ret = dev->set_voice_volume(dev, value); 1228 mHardwareStatus = AUDIO_HW_IDLE; 1229 1230 return ret; 1231} 1232 1233status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1234 audio_io_handle_t output) const 1235{ 1236 status_t status; 1237 1238 Mutex::Autolock _l(mLock); 1239 1240 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1241 if (playbackThread != NULL) { 1242 return playbackThread->getRenderPosition(halFrames, dspFrames); 1243 } 1244 1245 return BAD_VALUE; 1246} 1247 1248void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1249{ 1250 Mutex::Autolock _l(mLock); 1251 if (client == 0) { 1252 return; 1253 } 1254 pid_t pid = IPCThreadState::self()->getCallingPid(); 1255 { 1256 Mutex::Autolock _cl(mClientLock); 1257 if (mNotificationClients.indexOfKey(pid) < 0) { 1258 sp<NotificationClient> notificationClient = new NotificationClient(this, 1259 client, 1260 pid); 1261 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1262 1263 mNotificationClients.add(pid, notificationClient); 1264 1265 sp<IBinder> binder = IInterface::asBinder(client); 1266 binder->linkToDeath(notificationClient); 1267 } 1268 } 1269 1270 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1271 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1272 // the config change is always sent from playback or record threads to avoid deadlock 1273 // with AudioSystem::gLock 1274 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1275 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1276 } 1277 1278 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1279 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1280 } 1281} 1282 1283void AudioFlinger::removeNotificationClient(pid_t pid) 1284{ 1285 Mutex::Autolock _l(mLock); 1286 { 1287 Mutex::Autolock _cl(mClientLock); 1288 mNotificationClients.removeItem(pid); 1289 } 1290 1291 ALOGV("%d died, releasing its sessions", pid); 1292 size_t num = mAudioSessionRefs.size(); 1293 bool removed = false; 1294 for (size_t i = 0; i< num; ) { 1295 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1296 ALOGV(" pid %d @ %d", ref->mPid, i); 1297 if (ref->mPid == pid) { 1298 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1299 mAudioSessionRefs.removeAt(i); 1300 delete ref; 1301 removed = true; 1302 num--; 1303 } else { 1304 i++; 1305 } 1306 } 1307 if (removed) { 1308 purgeStaleEffects_l(); 1309 } 1310} 1311 1312void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1313 const sp<AudioIoDescriptor>& ioDesc, 1314 pid_t pid) 1315{ 1316 Mutex::Autolock _l(mClientLock); 1317 size_t size = mNotificationClients.size(); 1318 for (size_t i = 0; i < size; i++) { 1319 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1320 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1321 } 1322 } 1323} 1324 1325// removeClient_l() must be called with AudioFlinger::mClientLock held 1326void AudioFlinger::removeClient_l(pid_t pid) 1327{ 1328 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1329 IPCThreadState::self()->getCallingPid()); 1330 mClients.removeItem(pid); 1331} 1332 1333// getEffectThread_l() must be called with AudioFlinger::mLock held 1334sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1335{ 1336 sp<PlaybackThread> thread; 1337 1338 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1339 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1340 ALOG_ASSERT(thread == 0); 1341 thread = mPlaybackThreads.valueAt(i); 1342 } 1343 } 1344 1345 return thread; 1346} 1347 1348 1349 1350// ---------------------------------------------------------------------------- 1351 1352AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1353 : RefBase(), 1354 mAudioFlinger(audioFlinger), 1355 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1356 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1357 mPid(pid), 1358 mTimedTrackCount(0) 1359{ 1360 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1361} 1362 1363// Client destructor must be called with AudioFlinger::mClientLock held 1364AudioFlinger::Client::~Client() 1365{ 1366 mAudioFlinger->removeClient_l(mPid); 1367} 1368 1369sp<MemoryDealer> AudioFlinger::Client::heap() const 1370{ 1371 return mMemoryDealer; 1372} 1373 1374// Reserve one of the limited slots for a timed audio track associated 1375// with this client 1376bool AudioFlinger::Client::reserveTimedTrack() 1377{ 1378 const int kMaxTimedTracksPerClient = 4; 1379 1380 Mutex::Autolock _l(mTimedTrackLock); 1381 1382 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1383 ALOGW("can not create timed track - pid %d has exceeded the limit", 1384 mPid); 1385 return false; 1386 } 1387 1388 mTimedTrackCount++; 1389 return true; 1390} 1391 1392// Release a slot for a timed audio track 1393void AudioFlinger::Client::releaseTimedTrack() 1394{ 1395 Mutex::Autolock _l(mTimedTrackLock); 1396 mTimedTrackCount--; 1397} 1398 1399// ---------------------------------------------------------------------------- 1400 1401AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1402 const sp<IAudioFlingerClient>& client, 1403 pid_t pid) 1404 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1405{ 1406} 1407 1408AudioFlinger::NotificationClient::~NotificationClient() 1409{ 1410} 1411 1412void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1413{ 1414 sp<NotificationClient> keep(this); 1415 mAudioFlinger->removeNotificationClient(mPid); 1416} 1417 1418 1419// ---------------------------------------------------------------------------- 1420 1421static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1422 return audio_is_remote_submix_device(inDevice); 1423} 1424 1425sp<IAudioRecord> AudioFlinger::openRecord( 1426 audio_io_handle_t input, 1427 uint32_t sampleRate, 1428 audio_format_t format, 1429 audio_channel_mask_t channelMask, 1430 const String16& opPackageName, 1431 size_t *frameCount, 1432 IAudioFlinger::track_flags_t *flags, 1433 pid_t tid, 1434 int clientUid, 1435 int *sessionId, 1436 size_t *notificationFrames, 1437 sp<IMemory>& cblk, 1438 sp<IMemory>& buffers, 1439 status_t *status) 1440{ 1441 sp<RecordThread::RecordTrack> recordTrack; 1442 sp<RecordHandle> recordHandle; 1443 sp<Client> client; 1444 status_t lStatus; 1445 int lSessionId; 1446 1447 cblk.clear(); 1448 buffers.clear(); 1449 1450 // check calling permissions 1451 if (!recordingAllowed(opPackageName)) { 1452 ALOGE("openRecord() permission denied: recording not allowed"); 1453 lStatus = PERMISSION_DENIED; 1454 goto Exit; 1455 } 1456 1457 // further sample rate checks are performed by createRecordTrack_l() 1458 if (sampleRate == 0) { 1459 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1460 lStatus = BAD_VALUE; 1461 goto Exit; 1462 } 1463 1464 // we don't yet support anything other than linear PCM 1465 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1466 ALOGE("openRecord() invalid format %#x", format); 1467 lStatus = BAD_VALUE; 1468 goto Exit; 1469 } 1470 1471 // further channel mask checks are performed by createRecordTrack_l() 1472 if (!audio_is_input_channel(channelMask)) { 1473 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1474 lStatus = BAD_VALUE; 1475 goto Exit; 1476 } 1477 1478 { 1479 Mutex::Autolock _l(mLock); 1480 RecordThread *thread = checkRecordThread_l(input); 1481 if (thread == NULL) { 1482 ALOGE("openRecord() checkRecordThread_l failed"); 1483 lStatus = BAD_VALUE; 1484 goto Exit; 1485 } 1486 1487 pid_t pid = IPCThreadState::self()->getCallingPid(); 1488 client = registerPid(pid); 1489 1490 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1491 lSessionId = *sessionId; 1492 } else { 1493 // if no audio session id is provided, create one here 1494 lSessionId = nextUniqueId(); 1495 if (sessionId != NULL) { 1496 *sessionId = lSessionId; 1497 } 1498 } 1499 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1500 1501 // TODO: the uid should be passed in as a parameter to openRecord 1502 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1503 frameCount, lSessionId, notificationFrames, 1504 clientUid, flags, tid, &lStatus); 1505 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1506 1507 if (lStatus == NO_ERROR) { 1508 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1509 // session and move it to this thread. 1510 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1511 if (chain != 0) { 1512 Mutex::Autolock _l(thread->mLock); 1513 thread->addEffectChain_l(chain); 1514 } 1515 } 1516 } 1517 1518 if (lStatus != NO_ERROR) { 1519 // remove local strong reference to Client before deleting the RecordTrack so that the 1520 // Client destructor is called by the TrackBase destructor with mClientLock held 1521 // Don't hold mClientLock when releasing the reference on the track as the 1522 // destructor will acquire it. 1523 { 1524 Mutex::Autolock _cl(mClientLock); 1525 client.clear(); 1526 } 1527 recordTrack.clear(); 1528 goto Exit; 1529 } 1530 1531 cblk = recordTrack->getCblk(); 1532 buffers = recordTrack->getBuffers(); 1533 1534 // return handle to client 1535 recordHandle = new RecordHandle(recordTrack); 1536 1537Exit: 1538 *status = lStatus; 1539 return recordHandle; 1540} 1541 1542 1543 1544// ---------------------------------------------------------------------------- 1545 1546audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1547{ 1548 if (name == NULL) { 1549 return 0; 1550 } 1551 if (!settingsAllowed()) { 1552 return 0; 1553 } 1554 Mutex::Autolock _l(mLock); 1555 return loadHwModule_l(name); 1556} 1557 1558// loadHwModule_l() must be called with AudioFlinger::mLock held 1559audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1560{ 1561 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1562 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1563 ALOGW("loadHwModule() module %s already loaded", name); 1564 return mAudioHwDevs.keyAt(i); 1565 } 1566 } 1567 1568 audio_hw_device_t *dev; 1569 1570 int rc = load_audio_interface(name, &dev); 1571 if (rc) { 1572 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1573 return 0; 1574 } 1575 1576 mHardwareStatus = AUDIO_HW_INIT; 1577 rc = dev->init_check(dev); 1578 mHardwareStatus = AUDIO_HW_IDLE; 1579 if (rc) { 1580 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1581 return 0; 1582 } 1583 1584 // Check and cache this HAL's level of support for master mute and master 1585 // volume. If this is the first HAL opened, and it supports the get 1586 // methods, use the initial values provided by the HAL as the current 1587 // master mute and volume settings. 1588 1589 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1590 { // scope for auto-lock pattern 1591 AutoMutex lock(mHardwareLock); 1592 1593 if (0 == mAudioHwDevs.size()) { 1594 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1595 if (NULL != dev->get_master_volume) { 1596 float mv; 1597 if (OK == dev->get_master_volume(dev, &mv)) { 1598 mMasterVolume = mv; 1599 } 1600 } 1601 1602 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1603 if (NULL != dev->get_master_mute) { 1604 bool mm; 1605 if (OK == dev->get_master_mute(dev, &mm)) { 1606 mMasterMute = mm; 1607 } 1608 } 1609 } 1610 1611 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1612 if ((NULL != dev->set_master_volume) && 1613 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1614 flags = static_cast<AudioHwDevice::Flags>(flags | 1615 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1616 } 1617 1618 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1619 if ((NULL != dev->set_master_mute) && 1620 (OK == dev->set_master_mute(dev, mMasterMute))) { 1621 flags = static_cast<AudioHwDevice::Flags>(flags | 1622 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1623 } 1624 1625 mHardwareStatus = AUDIO_HW_IDLE; 1626 } 1627 1628 audio_module_handle_t handle = nextUniqueId(); 1629 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1630 1631 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1632 name, dev->common.module->name, dev->common.module->id, handle); 1633 1634 return handle; 1635 1636} 1637 1638// ---------------------------------------------------------------------------- 1639 1640uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1641{ 1642 Mutex::Autolock _l(mLock); 1643 PlaybackThread *thread = primaryPlaybackThread_l(); 1644 return thread != NULL ? thread->sampleRate() : 0; 1645} 1646 1647size_t AudioFlinger::getPrimaryOutputFrameCount() 1648{ 1649 Mutex::Autolock _l(mLock); 1650 PlaybackThread *thread = primaryPlaybackThread_l(); 1651 return thread != NULL ? thread->frameCountHAL() : 0; 1652} 1653 1654// ---------------------------------------------------------------------------- 1655 1656status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1657{ 1658 uid_t uid = IPCThreadState::self()->getCallingUid(); 1659 if (uid != AID_SYSTEM) { 1660 return PERMISSION_DENIED; 1661 } 1662 Mutex::Autolock _l(mLock); 1663 if (mIsDeviceTypeKnown) { 1664 return INVALID_OPERATION; 1665 } 1666 mIsLowRamDevice = isLowRamDevice; 1667 mIsDeviceTypeKnown = true; 1668 return NO_ERROR; 1669} 1670 1671audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1672{ 1673 Mutex::Autolock _l(mLock); 1674 1675 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1676 if (index >= 0) { 1677 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1678 mHwAvSyncIds.valueAt(index), sessionId); 1679 return mHwAvSyncIds.valueAt(index); 1680 } 1681 1682 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1683 if (dev == NULL) { 1684 return AUDIO_HW_SYNC_INVALID; 1685 } 1686 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1687 AudioParameter param = AudioParameter(String8(reply)); 1688 free(reply); 1689 1690 int value; 1691 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1692 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1693 return AUDIO_HW_SYNC_INVALID; 1694 } 1695 1696 // allow only one session for a given HW A/V sync ID. 1697 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1698 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1699 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1700 value, mHwAvSyncIds.keyAt(i)); 1701 mHwAvSyncIds.removeItemsAt(i); 1702 break; 1703 } 1704 } 1705 1706 mHwAvSyncIds.add(sessionId, value); 1707 1708 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1709 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1710 uint32_t sessions = thread->hasAudioSession(sessionId); 1711 if (sessions & PlaybackThread::TRACK_SESSION) { 1712 AudioParameter param = AudioParameter(); 1713 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1714 thread->setParameters(param.toString()); 1715 break; 1716 } 1717 } 1718 1719 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1720 return (audio_hw_sync_t)value; 1721} 1722 1723status_t AudioFlinger::systemReady() 1724{ 1725 Mutex::Autolock _l(mLock); 1726 ALOGI("%s", __FUNCTION__); 1727 if (mSystemReady) { 1728 ALOGW("%s called twice", __FUNCTION__); 1729 return NO_ERROR; 1730 } 1731 mSystemReady = true; 1732 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1733 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1734 thread->systemReady(); 1735 } 1736 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1737 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1738 thread->systemReady(); 1739 } 1740 return NO_ERROR; 1741} 1742 1743// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1744void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1745{ 1746 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1747 if (index >= 0) { 1748 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1749 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1750 AudioParameter param = AudioParameter(); 1751 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1752 thread->setParameters(param.toString()); 1753 } 1754} 1755 1756 1757// ---------------------------------------------------------------------------- 1758 1759 1760sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1761 audio_io_handle_t *output, 1762 audio_config_t *config, 1763 audio_devices_t devices, 1764 const String8& address, 1765 audio_output_flags_t flags) 1766{ 1767 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1768 if (outHwDev == NULL) { 1769 return 0; 1770 } 1771 1772 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1773 if (*output == AUDIO_IO_HANDLE_NONE) { 1774 *output = nextUniqueId(); 1775 } 1776 1777 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1778 1779 // FOR TESTING ONLY: 1780 // This if statement allows overriding the audio policy settings 1781 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1782 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1783 // Check only for Normal Mixing mode 1784 if (kEnableExtendedPrecision) { 1785 // Specify format (uncomment one below to choose) 1786 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1787 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1788 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1789 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1790 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1791 } 1792 if (kEnableExtendedChannels) { 1793 // Specify channel mask (uncomment one below to choose) 1794 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1795 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1796 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1797 } 1798 } 1799 1800 AudioStreamOut *outputStream = NULL; 1801 status_t status = outHwDev->openOutputStream( 1802 &outputStream, 1803 *output, 1804 devices, 1805 flags, 1806 config, 1807 address.string()); 1808 1809 mHardwareStatus = AUDIO_HW_IDLE; 1810 1811 if (status == NO_ERROR) { 1812 1813 PlaybackThread *thread; 1814 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1815 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1816 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1817 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1818 || !isValidPcmSinkFormat(config->format) 1819 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1820 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1821 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1822 } else { 1823 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1824 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1825 } 1826 mPlaybackThreads.add(*output, thread); 1827 return thread; 1828 } 1829 1830 return 0; 1831} 1832 1833status_t AudioFlinger::openOutput(audio_module_handle_t module, 1834 audio_io_handle_t *output, 1835 audio_config_t *config, 1836 audio_devices_t *devices, 1837 const String8& address, 1838 uint32_t *latencyMs, 1839 audio_output_flags_t flags) 1840{ 1841 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1842 module, 1843 (devices != NULL) ? *devices : 0, 1844 config->sample_rate, 1845 config->format, 1846 config->channel_mask, 1847 flags); 1848 1849 if (*devices == AUDIO_DEVICE_NONE) { 1850 return BAD_VALUE; 1851 } 1852 1853 Mutex::Autolock _l(mLock); 1854 1855 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1856 if (thread != 0) { 1857 *latencyMs = thread->latency(); 1858 1859 // notify client processes of the new output creation 1860 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1861 1862 // the first primary output opened designates the primary hw device 1863 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1864 ALOGI("Using module %d has the primary audio interface", module); 1865 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1866 1867 AutoMutex lock(mHardwareLock); 1868 mHardwareStatus = AUDIO_HW_SET_MODE; 1869 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1870 mHardwareStatus = AUDIO_HW_IDLE; 1871 } 1872 return NO_ERROR; 1873 } 1874 1875 return NO_INIT; 1876} 1877 1878audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1879 audio_io_handle_t output2) 1880{ 1881 Mutex::Autolock _l(mLock); 1882 MixerThread *thread1 = checkMixerThread_l(output1); 1883 MixerThread *thread2 = checkMixerThread_l(output2); 1884 1885 if (thread1 == NULL || thread2 == NULL) { 1886 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1887 output2); 1888 return AUDIO_IO_HANDLE_NONE; 1889 } 1890 1891 audio_io_handle_t id = nextUniqueId(); 1892 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1893 thread->addOutputTrack(thread2); 1894 mPlaybackThreads.add(id, thread); 1895 // notify client processes of the new output creation 1896 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1897 return id; 1898} 1899 1900status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1901{ 1902 return closeOutput_nonvirtual(output); 1903} 1904 1905status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1906{ 1907 // keep strong reference on the playback thread so that 1908 // it is not destroyed while exit() is executed 1909 sp<PlaybackThread> thread; 1910 { 1911 Mutex::Autolock _l(mLock); 1912 thread = checkPlaybackThread_l(output); 1913 if (thread == NULL) { 1914 return BAD_VALUE; 1915 } 1916 1917 ALOGV("closeOutput() %d", output); 1918 1919 if (thread->type() == ThreadBase::MIXER) { 1920 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1921 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1922 DuplicatingThread *dupThread = 1923 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1924 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1925 } 1926 } 1927 } 1928 1929 1930 mPlaybackThreads.removeItem(output); 1931 // save all effects to the default thread 1932 if (mPlaybackThreads.size()) { 1933 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1934 if (dstThread != NULL) { 1935 // audioflinger lock is held here so the acquisition order of thread locks does not 1936 // matter 1937 Mutex::Autolock _dl(dstThread->mLock); 1938 Mutex::Autolock _sl(thread->mLock); 1939 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1940 for (size_t i = 0; i < effectChains.size(); i ++) { 1941 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1942 } 1943 } 1944 } 1945 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1946 ioDesc->mIoHandle = output; 1947 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1948 } 1949 thread->exit(); 1950 // The thread entity (active unit of execution) is no longer running here, 1951 // but the ThreadBase container still exists. 1952 1953 if (!thread->isDuplicating()) { 1954 closeOutputFinish(thread); 1955 } 1956 1957 return NO_ERROR; 1958} 1959 1960void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1961{ 1962 AudioStreamOut *out = thread->clearOutput(); 1963 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1964 // from now on thread->mOutput is NULL 1965 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1966 delete out; 1967} 1968 1969void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1970{ 1971 mPlaybackThreads.removeItem(thread->mId); 1972 thread->exit(); 1973 closeOutputFinish(thread); 1974} 1975 1976status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1977{ 1978 Mutex::Autolock _l(mLock); 1979 PlaybackThread *thread = checkPlaybackThread_l(output); 1980 1981 if (thread == NULL) { 1982 return BAD_VALUE; 1983 } 1984 1985 ALOGV("suspendOutput() %d", output); 1986 thread->suspend(); 1987 1988 return NO_ERROR; 1989} 1990 1991status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1992{ 1993 Mutex::Autolock _l(mLock); 1994 PlaybackThread *thread = checkPlaybackThread_l(output); 1995 1996 if (thread == NULL) { 1997 return BAD_VALUE; 1998 } 1999 2000 ALOGV("restoreOutput() %d", output); 2001 2002 thread->restore(); 2003 2004 return NO_ERROR; 2005} 2006 2007status_t AudioFlinger::openInput(audio_module_handle_t module, 2008 audio_io_handle_t *input, 2009 audio_config_t *config, 2010 audio_devices_t *devices, 2011 const String8& address, 2012 audio_source_t source, 2013 audio_input_flags_t flags) 2014{ 2015 Mutex::Autolock _l(mLock); 2016 2017 if (*devices == AUDIO_DEVICE_NONE) { 2018 return BAD_VALUE; 2019 } 2020 2021 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2022 2023 if (thread != 0) { 2024 // notify client processes of the new input creation 2025 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2026 return NO_ERROR; 2027 } 2028 return NO_INIT; 2029} 2030 2031sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2032 audio_io_handle_t *input, 2033 audio_config_t *config, 2034 audio_devices_t devices, 2035 const String8& address, 2036 audio_source_t source, 2037 audio_input_flags_t flags) 2038{ 2039 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2040 if (inHwDev == NULL) { 2041 *input = AUDIO_IO_HANDLE_NONE; 2042 return 0; 2043 } 2044 2045 if (*input == AUDIO_IO_HANDLE_NONE) { 2046 *input = nextUniqueId(); 2047 } 2048 2049 audio_config_t halconfig = *config; 2050 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2051 audio_stream_in_t *inStream = NULL; 2052 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2053 &inStream, flags, address.string(), source); 2054 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2055 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2056 inStream, 2057 halconfig.sample_rate, 2058 halconfig.format, 2059 halconfig.channel_mask, 2060 flags, 2061 status, address.string()); 2062 2063 // If the input could not be opened with the requested parameters and we can handle the 2064 // conversion internally, try to open again with the proposed parameters. 2065 if (status == BAD_VALUE && 2066 audio_is_linear_pcm(config->format) && 2067 audio_is_linear_pcm(halconfig.format) && 2068 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2069 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2070 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2071 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2072 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2073 inStream = NULL; 2074 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2075 &inStream, flags, address.string(), source); 2076 // FIXME log this new status; HAL should not propose any further changes 2077 } 2078 2079 if (status == NO_ERROR && inStream != NULL) { 2080 2081#ifdef TEE_SINK 2082 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2083 // or (re-)create if current Pipe is idle and does not match the new format 2084 sp<NBAIO_Sink> teeSink; 2085 enum { 2086 TEE_SINK_NO, // don't copy input 2087 TEE_SINK_NEW, // copy input using a new pipe 2088 TEE_SINK_OLD, // copy input using an existing pipe 2089 } kind; 2090 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2091 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2092 if (!mTeeSinkInputEnabled) { 2093 kind = TEE_SINK_NO; 2094 } else if (!Format_isValid(format)) { 2095 kind = TEE_SINK_NO; 2096 } else if (mRecordTeeSink == 0) { 2097 kind = TEE_SINK_NEW; 2098 } else if (mRecordTeeSink->getStrongCount() != 1) { 2099 kind = TEE_SINK_NO; 2100 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2101 kind = TEE_SINK_OLD; 2102 } else { 2103 kind = TEE_SINK_NEW; 2104 } 2105 switch (kind) { 2106 case TEE_SINK_NEW: { 2107 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2108 size_t numCounterOffers = 0; 2109 const NBAIO_Format offers[1] = {format}; 2110 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2111 ALOG_ASSERT(index == 0); 2112 PipeReader *pipeReader = new PipeReader(*pipe); 2113 numCounterOffers = 0; 2114 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2115 ALOG_ASSERT(index == 0); 2116 mRecordTeeSink = pipe; 2117 mRecordTeeSource = pipeReader; 2118 teeSink = pipe; 2119 } 2120 break; 2121 case TEE_SINK_OLD: 2122 teeSink = mRecordTeeSink; 2123 break; 2124 case TEE_SINK_NO: 2125 default: 2126 break; 2127 } 2128#endif 2129 2130 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2131 2132 // Start record thread 2133 // RecordThread requires both input and output device indication to forward to audio 2134 // pre processing modules 2135 sp<RecordThread> thread = new RecordThread(this, 2136 inputStream, 2137 *input, 2138 primaryOutputDevice_l(), 2139 devices, 2140 mSystemReady 2141#ifdef TEE_SINK 2142 , teeSink 2143#endif 2144 ); 2145 mRecordThreads.add(*input, thread); 2146 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2147 return thread; 2148 } 2149 2150 *input = AUDIO_IO_HANDLE_NONE; 2151 return 0; 2152} 2153 2154status_t AudioFlinger::closeInput(audio_io_handle_t input) 2155{ 2156 return closeInput_nonvirtual(input); 2157} 2158 2159status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2160{ 2161 // keep strong reference on the record thread so that 2162 // it is not destroyed while exit() is executed 2163 sp<RecordThread> thread; 2164 { 2165 Mutex::Autolock _l(mLock); 2166 thread = checkRecordThread_l(input); 2167 if (thread == 0) { 2168 return BAD_VALUE; 2169 } 2170 2171 ALOGV("closeInput() %d", input); 2172 2173 // If we still have effect chains, it means that a client still holds a handle 2174 // on at least one effect. We must either move the chain to an existing thread with the 2175 // same session ID or put it aside in case a new record thread is opened for a 2176 // new capture on the same session 2177 sp<EffectChain> chain; 2178 { 2179 Mutex::Autolock _sl(thread->mLock); 2180 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2181 // Note: maximum one chain per record thread 2182 if (effectChains.size() != 0) { 2183 chain = effectChains[0]; 2184 } 2185 } 2186 if (chain != 0) { 2187 // first check if a record thread is already opened with a client on the same session. 2188 // This should only happen in case of overlap between one thread tear down and the 2189 // creation of its replacement 2190 size_t i; 2191 for (i = 0; i < mRecordThreads.size(); i++) { 2192 sp<RecordThread> t = mRecordThreads.valueAt(i); 2193 if (t == thread) { 2194 continue; 2195 } 2196 if (t->hasAudioSession(chain->sessionId()) != 0) { 2197 Mutex::Autolock _l(t->mLock); 2198 ALOGV("closeInput() found thread %d for effect session %d", 2199 t->id(), chain->sessionId()); 2200 t->addEffectChain_l(chain); 2201 break; 2202 } 2203 } 2204 // put the chain aside if we could not find a record thread with the same session id. 2205 if (i == mRecordThreads.size()) { 2206 putOrphanEffectChain_l(chain); 2207 } 2208 } 2209 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2210 ioDesc->mIoHandle = input; 2211 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2212 mRecordThreads.removeItem(input); 2213 } 2214 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2215 // we have a different lock for notification client 2216 closeInputFinish(thread); 2217 return NO_ERROR; 2218} 2219 2220void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2221{ 2222 thread->exit(); 2223 AudioStreamIn *in = thread->clearInput(); 2224 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2225 // from now on thread->mInput is NULL 2226 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2227 delete in; 2228} 2229 2230void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2231{ 2232 mRecordThreads.removeItem(thread->mId); 2233 closeInputFinish(thread); 2234} 2235 2236status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2237{ 2238 Mutex::Autolock _l(mLock); 2239 ALOGV("invalidateStream() stream %d", stream); 2240 2241 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2242 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2243 thread->invalidateTracks(stream); 2244 } 2245 2246 return NO_ERROR; 2247} 2248 2249 2250audio_unique_id_t AudioFlinger::newAudioUniqueId() 2251{ 2252 return nextUniqueId(); 2253} 2254 2255void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2256{ 2257 Mutex::Autolock _l(mLock); 2258 pid_t caller = IPCThreadState::self()->getCallingPid(); 2259 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2260 if (pid != -1 && (caller == getpid_cached)) { 2261 caller = pid; 2262 } 2263 2264 { 2265 Mutex::Autolock _cl(mClientLock); 2266 // Ignore requests received from processes not known as notification client. The request 2267 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2268 // called from a different pid leaving a stale session reference. Also we don't know how 2269 // to clear this reference if the client process dies. 2270 if (mNotificationClients.indexOfKey(caller) < 0) { 2271 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2272 return; 2273 } 2274 } 2275 2276 size_t num = mAudioSessionRefs.size(); 2277 for (size_t i = 0; i< num; i++) { 2278 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2279 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2280 ref->mCnt++; 2281 ALOGV(" incremented refcount to %d", ref->mCnt); 2282 return; 2283 } 2284 } 2285 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2286 ALOGV(" added new entry for %d", audioSession); 2287} 2288 2289void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2290{ 2291 Mutex::Autolock _l(mLock); 2292 pid_t caller = IPCThreadState::self()->getCallingPid(); 2293 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2294 if (pid != -1 && (caller == getpid_cached)) { 2295 caller = pid; 2296 } 2297 size_t num = mAudioSessionRefs.size(); 2298 for (size_t i = 0; i< num; i++) { 2299 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2300 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2301 ref->mCnt--; 2302 ALOGV(" decremented refcount to %d", ref->mCnt); 2303 if (ref->mCnt == 0) { 2304 mAudioSessionRefs.removeAt(i); 2305 delete ref; 2306 purgeStaleEffects_l(); 2307 } 2308 return; 2309 } 2310 } 2311 // If the caller is mediaserver it is likely that the session being released was acquired 2312 // on behalf of a process not in notification clients and we ignore the warning. 2313 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2314} 2315 2316void AudioFlinger::purgeStaleEffects_l() { 2317 2318 ALOGV("purging stale effects"); 2319 2320 Vector< sp<EffectChain> > chains; 2321 2322 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2323 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2324 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2325 sp<EffectChain> ec = t->mEffectChains[j]; 2326 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2327 chains.push(ec); 2328 } 2329 } 2330 } 2331 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2332 sp<RecordThread> t = mRecordThreads.valueAt(i); 2333 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2334 sp<EffectChain> ec = t->mEffectChains[j]; 2335 chains.push(ec); 2336 } 2337 } 2338 2339 for (size_t i = 0; i < chains.size(); i++) { 2340 sp<EffectChain> ec = chains[i]; 2341 int sessionid = ec->sessionId(); 2342 sp<ThreadBase> t = ec->mThread.promote(); 2343 if (t == 0) { 2344 continue; 2345 } 2346 size_t numsessionrefs = mAudioSessionRefs.size(); 2347 bool found = false; 2348 for (size_t k = 0; k < numsessionrefs; k++) { 2349 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2350 if (ref->mSessionid == sessionid) { 2351 ALOGV(" session %d still exists for %d with %d refs", 2352 sessionid, ref->mPid, ref->mCnt); 2353 found = true; 2354 break; 2355 } 2356 } 2357 if (!found) { 2358 Mutex::Autolock _l(t->mLock); 2359 // remove all effects from the chain 2360 while (ec->mEffects.size()) { 2361 sp<EffectModule> effect = ec->mEffects[0]; 2362 effect->unPin(); 2363 t->removeEffect_l(effect); 2364 if (effect->purgeHandles()) { 2365 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2366 } 2367 AudioSystem::unregisterEffect(effect->id()); 2368 } 2369 } 2370 } 2371 return; 2372} 2373 2374// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2375AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2376{ 2377 return mPlaybackThreads.valueFor(output).get(); 2378} 2379 2380// checkMixerThread_l() must be called with AudioFlinger::mLock held 2381AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2382{ 2383 PlaybackThread *thread = checkPlaybackThread_l(output); 2384 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2385} 2386 2387// checkRecordThread_l() must be called with AudioFlinger::mLock held 2388AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2389{ 2390 return mRecordThreads.valueFor(input).get(); 2391} 2392 2393uint32_t AudioFlinger::nextUniqueId() 2394{ 2395 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2396} 2397 2398AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2399{ 2400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2401 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2402 if(thread->isDuplicating()) { 2403 continue; 2404 } 2405 AudioStreamOut *output = thread->getOutput(); 2406 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2407 return thread; 2408 } 2409 } 2410 return NULL; 2411} 2412 2413audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2414{ 2415 PlaybackThread *thread = primaryPlaybackThread_l(); 2416 2417 if (thread == NULL) { 2418 return 0; 2419 } 2420 2421 return thread->outDevice(); 2422} 2423 2424sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2425 int triggerSession, 2426 int listenerSession, 2427 sync_event_callback_t callBack, 2428 wp<RefBase> cookie) 2429{ 2430 Mutex::Autolock _l(mLock); 2431 2432 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2433 status_t playStatus = NAME_NOT_FOUND; 2434 status_t recStatus = NAME_NOT_FOUND; 2435 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2436 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2437 if (playStatus == NO_ERROR) { 2438 return event; 2439 } 2440 } 2441 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2442 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2443 if (recStatus == NO_ERROR) { 2444 return event; 2445 } 2446 } 2447 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2448 mPendingSyncEvents.add(event); 2449 } else { 2450 ALOGV("createSyncEvent() invalid event %d", event->type()); 2451 event.clear(); 2452 } 2453 return event; 2454} 2455 2456// ---------------------------------------------------------------------------- 2457// Effect management 2458// ---------------------------------------------------------------------------- 2459 2460 2461status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2462{ 2463 Mutex::Autolock _l(mLock); 2464 return EffectQueryNumberEffects(numEffects); 2465} 2466 2467status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2468{ 2469 Mutex::Autolock _l(mLock); 2470 return EffectQueryEffect(index, descriptor); 2471} 2472 2473status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2474 effect_descriptor_t *descriptor) const 2475{ 2476 Mutex::Autolock _l(mLock); 2477 return EffectGetDescriptor(pUuid, descriptor); 2478} 2479 2480 2481sp<IEffect> AudioFlinger::createEffect( 2482 effect_descriptor_t *pDesc, 2483 const sp<IEffectClient>& effectClient, 2484 int32_t priority, 2485 audio_io_handle_t io, 2486 int sessionId, 2487 const String16& opPackageName, 2488 status_t *status, 2489 int *id, 2490 int *enabled) 2491{ 2492 status_t lStatus = NO_ERROR; 2493 sp<EffectHandle> handle; 2494 effect_descriptor_t desc; 2495 2496 pid_t pid = IPCThreadState::self()->getCallingPid(); 2497 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2498 pid, effectClient.get(), priority, sessionId, io); 2499 2500 if (pDesc == NULL) { 2501 lStatus = BAD_VALUE; 2502 goto Exit; 2503 } 2504 2505 // check audio settings permission for global effects 2506 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2507 lStatus = PERMISSION_DENIED; 2508 goto Exit; 2509 } 2510 2511 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2512 // that can only be created by audio policy manager (running in same process) 2513 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2514 lStatus = PERMISSION_DENIED; 2515 goto Exit; 2516 } 2517 2518 { 2519 if (!EffectIsNullUuid(&pDesc->uuid)) { 2520 // if uuid is specified, request effect descriptor 2521 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2522 if (lStatus < 0) { 2523 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2524 goto Exit; 2525 } 2526 } else { 2527 // if uuid is not specified, look for an available implementation 2528 // of the required type in effect factory 2529 if (EffectIsNullUuid(&pDesc->type)) { 2530 ALOGW("createEffect() no effect type"); 2531 lStatus = BAD_VALUE; 2532 goto Exit; 2533 } 2534 uint32_t numEffects = 0; 2535 effect_descriptor_t d; 2536 d.flags = 0; // prevent compiler warning 2537 bool found = false; 2538 2539 lStatus = EffectQueryNumberEffects(&numEffects); 2540 if (lStatus < 0) { 2541 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2542 goto Exit; 2543 } 2544 for (uint32_t i = 0; i < numEffects; i++) { 2545 lStatus = EffectQueryEffect(i, &desc); 2546 if (lStatus < 0) { 2547 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2548 continue; 2549 } 2550 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2551 // If matching type found save effect descriptor. If the session is 2552 // 0 and the effect is not auxiliary, continue enumeration in case 2553 // an auxiliary version of this effect type is available 2554 found = true; 2555 d = desc; 2556 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2557 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2558 break; 2559 } 2560 } 2561 } 2562 if (!found) { 2563 lStatus = BAD_VALUE; 2564 ALOGW("createEffect() effect not found"); 2565 goto Exit; 2566 } 2567 // For same effect type, chose auxiliary version over insert version if 2568 // connect to output mix (Compliance to OpenSL ES) 2569 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2570 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2571 desc = d; 2572 } 2573 } 2574 2575 // Do not allow auxiliary effects on a session different from 0 (output mix) 2576 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2577 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2578 lStatus = INVALID_OPERATION; 2579 goto Exit; 2580 } 2581 2582 // check recording permission for visualizer 2583 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2584 !recordingAllowed(opPackageName)) { 2585 lStatus = PERMISSION_DENIED; 2586 goto Exit; 2587 } 2588 2589 // return effect descriptor 2590 *pDesc = desc; 2591 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2592 // if the output returned by getOutputForEffect() is removed before we lock the 2593 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2594 // and we will exit safely 2595 io = AudioSystem::getOutputForEffect(&desc); 2596 ALOGV("createEffect got output %d", io); 2597 } 2598 2599 Mutex::Autolock _l(mLock); 2600 2601 // If output is not specified try to find a matching audio session ID in one of the 2602 // output threads. 2603 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2604 // because of code checking output when entering the function. 2605 // Note: io is never 0 when creating an effect on an input 2606 if (io == AUDIO_IO_HANDLE_NONE) { 2607 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2608 // output must be specified by AudioPolicyManager when using session 2609 // AUDIO_SESSION_OUTPUT_STAGE 2610 lStatus = BAD_VALUE; 2611 goto Exit; 2612 } 2613 // look for the thread where the specified audio session is present 2614 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2615 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2616 io = mPlaybackThreads.keyAt(i); 2617 break; 2618 } 2619 } 2620 if (io == 0) { 2621 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2622 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2623 io = mRecordThreads.keyAt(i); 2624 break; 2625 } 2626 } 2627 } 2628 // If no output thread contains the requested session ID, default to 2629 // first output. The effect chain will be moved to the correct output 2630 // thread when a track with the same session ID is created 2631 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2632 io = mPlaybackThreads.keyAt(0); 2633 } 2634 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2635 } 2636 ThreadBase *thread = checkRecordThread_l(io); 2637 if (thread == NULL) { 2638 thread = checkPlaybackThread_l(io); 2639 if (thread == NULL) { 2640 ALOGE("createEffect() unknown output thread"); 2641 lStatus = BAD_VALUE; 2642 goto Exit; 2643 } 2644 } else { 2645 // Check if one effect chain was awaiting for an effect to be created on this 2646 // session and used it instead of creating a new one. 2647 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2648 if (chain != 0) { 2649 Mutex::Autolock _l(thread->mLock); 2650 thread->addEffectChain_l(chain); 2651 } 2652 } 2653 2654 sp<Client> client = registerPid(pid); 2655 2656 // create effect on selected output thread 2657 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2658 &desc, enabled, &lStatus); 2659 if (handle != 0 && id != NULL) { 2660 *id = handle->id(); 2661 } 2662 if (handle == 0) { 2663 // remove local strong reference to Client with mClientLock held 2664 Mutex::Autolock _cl(mClientLock); 2665 client.clear(); 2666 } 2667 } 2668 2669Exit: 2670 *status = lStatus; 2671 return handle; 2672} 2673 2674status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2675 audio_io_handle_t dstOutput) 2676{ 2677 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2678 sessionId, srcOutput, dstOutput); 2679 Mutex::Autolock _l(mLock); 2680 if (srcOutput == dstOutput) { 2681 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2682 return NO_ERROR; 2683 } 2684 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2685 if (srcThread == NULL) { 2686 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2687 return BAD_VALUE; 2688 } 2689 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2690 if (dstThread == NULL) { 2691 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2692 return BAD_VALUE; 2693 } 2694 2695 Mutex::Autolock _dl(dstThread->mLock); 2696 Mutex::Autolock _sl(srcThread->mLock); 2697 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2698} 2699 2700// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2701status_t AudioFlinger::moveEffectChain_l(int sessionId, 2702 AudioFlinger::PlaybackThread *srcThread, 2703 AudioFlinger::PlaybackThread *dstThread, 2704 bool reRegister) 2705{ 2706 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2707 sessionId, srcThread, dstThread); 2708 2709 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2710 if (chain == 0) { 2711 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2712 sessionId, srcThread); 2713 return INVALID_OPERATION; 2714 } 2715 2716 // Check whether the destination thread has a channel count of FCC_2, which is 2717 // currently required for (most) effects. Prevent moving the effect chain here rather 2718 // than disabling the addEffect_l() call in dstThread below. 2719 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2720 dstThread->mChannelCount != FCC_2) { 2721 ALOGW("moveEffectChain_l() effect chain failed because" 2722 " destination thread %p channel count(%u) != %u", 2723 dstThread, dstThread->mChannelCount, FCC_2); 2724 return INVALID_OPERATION; 2725 } 2726 2727 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2728 // so that a new chain is created with correct parameters when first effect is added. This is 2729 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2730 // removed. 2731 srcThread->removeEffectChain_l(chain); 2732 2733 // transfer all effects one by one so that new effect chain is created on new thread with 2734 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2735 sp<EffectChain> dstChain; 2736 uint32_t strategy = 0; // prevent compiler warning 2737 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2738 Vector< sp<EffectModule> > removed; 2739 status_t status = NO_ERROR; 2740 while (effect != 0) { 2741 srcThread->removeEffect_l(effect); 2742 removed.add(effect); 2743 status = dstThread->addEffect_l(effect); 2744 if (status != NO_ERROR) { 2745 break; 2746 } 2747 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2748 if (effect->state() == EffectModule::ACTIVE || 2749 effect->state() == EffectModule::STOPPING) { 2750 effect->start(); 2751 } 2752 // if the move request is not received from audio policy manager, the effect must be 2753 // re-registered with the new strategy and output 2754 if (dstChain == 0) { 2755 dstChain = effect->chain().promote(); 2756 if (dstChain == 0) { 2757 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2758 status = NO_INIT; 2759 break; 2760 } 2761 strategy = dstChain->strategy(); 2762 } 2763 if (reRegister) { 2764 AudioSystem::unregisterEffect(effect->id()); 2765 AudioSystem::registerEffect(&effect->desc(), 2766 dstThread->id(), 2767 strategy, 2768 sessionId, 2769 effect->id()); 2770 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2771 } 2772 effect = chain->getEffectFromId_l(0); 2773 } 2774 2775 if (status != NO_ERROR) { 2776 for (size_t i = 0; i < removed.size(); i++) { 2777 srcThread->addEffect_l(removed[i]); 2778 if (dstChain != 0 && reRegister) { 2779 AudioSystem::unregisterEffect(removed[i]->id()); 2780 AudioSystem::registerEffect(&removed[i]->desc(), 2781 srcThread->id(), 2782 strategy, 2783 sessionId, 2784 removed[i]->id()); 2785 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2786 } 2787 } 2788 } 2789 2790 return status; 2791} 2792 2793bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2794{ 2795 if (mGlobalEffectEnableTime != 0 && 2796 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2797 return true; 2798 } 2799 2800 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2801 sp<EffectChain> ec = 2802 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2803 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2804 return true; 2805 } 2806 } 2807 return false; 2808} 2809 2810void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2811{ 2812 Mutex::Autolock _l(mLock); 2813 2814 mGlobalEffectEnableTime = systemTime(); 2815 2816 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2817 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2818 if (t->mType == ThreadBase::OFFLOAD) { 2819 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2820 } 2821 } 2822 2823} 2824 2825status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2826{ 2827 audio_session_t session = (audio_session_t)chain->sessionId(); 2828 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2829 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2830 if (index >= 0) { 2831 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2832 return ALREADY_EXISTS; 2833 } 2834 mOrphanEffectChains.add(session, chain); 2835 return NO_ERROR; 2836} 2837 2838sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2839{ 2840 sp<EffectChain> chain; 2841 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2842 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2843 if (index >= 0) { 2844 chain = mOrphanEffectChains.valueAt(index); 2845 mOrphanEffectChains.removeItemsAt(index); 2846 } 2847 return chain; 2848} 2849 2850bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2851{ 2852 Mutex::Autolock _l(mLock); 2853 audio_session_t session = (audio_session_t)effect->sessionId(); 2854 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2855 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2856 if (index >= 0) { 2857 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2858 if (chain->removeEffect_l(effect) == 0) { 2859 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2860 mOrphanEffectChains.removeItemsAt(index); 2861 } 2862 return true; 2863 } 2864 return false; 2865} 2866 2867 2868struct Entry { 2869#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2870 char mFileName[TEE_MAX_FILENAME]; 2871}; 2872 2873int comparEntry(const void *p1, const void *p2) 2874{ 2875 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2876} 2877 2878#ifdef TEE_SINK 2879void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2880{ 2881 NBAIO_Source *teeSource = source.get(); 2882 if (teeSource != NULL) { 2883 // .wav rotation 2884 // There is a benign race condition if 2 threads call this simultaneously. 2885 // They would both traverse the directory, but the result would simply be 2886 // failures at unlink() which are ignored. It's also unlikely since 2887 // normally dumpsys is only done by bugreport or from the command line. 2888 char teePath[32+256]; 2889 strcpy(teePath, "/data/misc/media"); 2890 size_t teePathLen = strlen(teePath); 2891 DIR *dir = opendir(teePath); 2892 teePath[teePathLen++] = '/'; 2893 if (dir != NULL) { 2894#define TEE_MAX_SORT 20 // number of entries to sort 2895#define TEE_MAX_KEEP 10 // number of entries to keep 2896 struct Entry entries[TEE_MAX_SORT]; 2897 size_t entryCount = 0; 2898 while (entryCount < TEE_MAX_SORT) { 2899 struct dirent de; 2900 struct dirent *result = NULL; 2901 int rc = readdir_r(dir, &de, &result); 2902 if (rc != 0) { 2903 ALOGW("readdir_r failed %d", rc); 2904 break; 2905 } 2906 if (result == NULL) { 2907 break; 2908 } 2909 if (result != &de) { 2910 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2911 break; 2912 } 2913 // ignore non .wav file entries 2914 size_t nameLen = strlen(de.d_name); 2915 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2916 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2917 continue; 2918 } 2919 strcpy(entries[entryCount++].mFileName, de.d_name); 2920 } 2921 (void) closedir(dir); 2922 if (entryCount > TEE_MAX_KEEP) { 2923 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2924 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2925 strcpy(&teePath[teePathLen], entries[i].mFileName); 2926 (void) unlink(teePath); 2927 } 2928 } 2929 } else { 2930 if (fd >= 0) { 2931 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2932 } 2933 } 2934 char teeTime[16]; 2935 struct timeval tv; 2936 gettimeofday(&tv, NULL); 2937 struct tm tm; 2938 localtime_r(&tv.tv_sec, &tm); 2939 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2940 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2941 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2942 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2943 if (teeFd >= 0) { 2944 // FIXME use libsndfile 2945 char wavHeader[44]; 2946 memcpy(wavHeader, 2947 "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", 2948 sizeof(wavHeader)); 2949 NBAIO_Format format = teeSource->format(); 2950 unsigned channelCount = Format_channelCount(format); 2951 uint32_t sampleRate = Format_sampleRate(format); 2952 size_t frameSize = Format_frameSize(format); 2953 wavHeader[22] = channelCount; // number of channels 2954 wavHeader[24] = sampleRate; // sample rate 2955 wavHeader[25] = sampleRate >> 8; 2956 wavHeader[32] = frameSize; // block alignment 2957 wavHeader[33] = frameSize >> 8; 2958 write(teeFd, wavHeader, sizeof(wavHeader)); 2959 size_t total = 0; 2960 bool firstRead = true; 2961#define TEE_SINK_READ 1024 // frames per I/O operation 2962 void *buffer = malloc(TEE_SINK_READ * frameSize); 2963 for (;;) { 2964 size_t count = TEE_SINK_READ; 2965 ssize_t actual = teeSource->read(buffer, count, 2966 AudioBufferProvider::kInvalidPTS); 2967 bool wasFirstRead = firstRead; 2968 firstRead = false; 2969 if (actual <= 0) { 2970 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2971 continue; 2972 } 2973 break; 2974 } 2975 ALOG_ASSERT(actual <= (ssize_t)count); 2976 write(teeFd, buffer, actual * frameSize); 2977 total += actual; 2978 } 2979 free(buffer); 2980 lseek(teeFd, (off_t) 4, SEEK_SET); 2981 uint32_t temp = 44 + total * frameSize - 8; 2982 // FIXME not big-endian safe 2983 write(teeFd, &temp, sizeof(temp)); 2984 lseek(teeFd, (off_t) 40, SEEK_SET); 2985 temp = total * frameSize; 2986 // FIXME not big-endian safe 2987 write(teeFd, &temp, sizeof(temp)); 2988 close(teeFd); 2989 if (fd >= 0) { 2990 dprintf(fd, "tee copied to %s\n", teePath); 2991 } 2992 } else { 2993 if (fd >= 0) { 2994 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2995 } 2996 } 2997 } 2998} 2999#endif 3000 3001// ---------------------------------------------------------------------------- 3002 3003status_t AudioFlinger::onTransact( 3004 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3005{ 3006 return BnAudioFlinger::onTransact(code, data, reply, flags); 3007} 3008 3009} // namespace android 3010