AudioFlinger.cpp revision d848eb48c121c119e8ba7583efc75415fe102570
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/AudioResamplerPublic.h> 49 50#include <media/EffectsFactoryApi.h> 51#include <audio_effects/effect_visualizer.h> 52#include <audio_effects/effect_ns.h> 53#include <audio_effects/effect_aec.h> 54 55#include <audio_utils/primitives.h> 56 57#include <powermanager/PowerManager.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <mediautils/BatteryNotifier.h> 65#include <private/android_filesystem_config.h> 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 85static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 86static const char kClientLockedString[] = "Client lock is taken\n"; 87 88 89nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 90 91uint32_t AudioFlinger::mScreenState; 92 93#ifdef TEE_SINK 94bool AudioFlinger::mTeeSinkInputEnabled = false; 95bool AudioFlinger::mTeeSinkOutputEnabled = false; 96bool AudioFlinger::mTeeSinkTrackEnabled = false; 97 98size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 99size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 100size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 101#endif 102 103// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 104// we define a minimum time during which a global effect is considered enabled. 105static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 106 107// ---------------------------------------------------------------------------- 108 109const char *formatToString(audio_format_t format) { 110 switch (audio_get_main_format(format)) { 111 case AUDIO_FORMAT_PCM: 112 switch (format) { 113 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 114 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 115 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 116 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 117 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 118 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 119 default: 120 break; 121 } 122 break; 123 case AUDIO_FORMAT_MP3: return "mp3"; 124 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 125 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 126 case AUDIO_FORMAT_AAC: return "aac"; 127 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 128 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 129 case AUDIO_FORMAT_VORBIS: return "vorbis"; 130 case AUDIO_FORMAT_OPUS: return "opus"; 131 case AUDIO_FORMAT_AC3: return "ac-3"; 132 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 133 case AUDIO_FORMAT_IEC61937: return "iec61937"; 134 default: 135 break; 136 } 137 return "unknown"; 138} 139 140static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 141{ 142 const hw_module_t *mod; 143 int rc; 144 145 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 146 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 147 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 148 if (rc) { 149 goto out; 150 } 151 rc = audio_hw_device_open(mod, dev); 152 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 153 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 154 if (rc) { 155 goto out; 156 } 157 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 158 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 159 rc = BAD_VALUE; 160 goto out; 161 } 162 return 0; 163 164out: 165 *dev = NULL; 166 return rc; 167} 168 169// ---------------------------------------------------------------------------- 170 171AudioFlinger::AudioFlinger() 172 : BnAudioFlinger(), 173 mPrimaryHardwareDev(NULL), 174 mAudioHwDevs(NULL), 175 mHardwareStatus(AUDIO_HW_IDLE), 176 mMasterVolume(1.0f), 177 mMasterMute(false), 178 mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), // zero has a special meaning, so unavailable 179 mMode(AUDIO_MODE_INVALID), 180 mBtNrecIsOff(false), 181 mIsLowRamDevice(true), 182 mIsDeviceTypeKnown(false), 183 mGlobalEffectEnableTime(0), 184 mSystemReady(false) 185{ 186 getpid_cached = getpid(); 187 const bool doLog = property_get_bool("ro.test_harness", false); 188 if (doLog) { 189 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 190 MemoryHeapBase::READ_ONLY); 191 } 192 193 // reset battery stats. 194 // if the audio service has crashed, battery stats could be left 195 // in bad state, reset the state upon service start. 196 BatteryNotifier::getInstance().noteResetAudio(); 197 198#ifdef TEE_SINK 199 char value[PROPERTY_VALUE_MAX]; 200 (void) property_get("ro.debuggable", value, "0"); 201 int debuggable = atoi(value); 202 int teeEnabled = 0; 203 if (debuggable) { 204 (void) property_get("af.tee", value, "0"); 205 teeEnabled = atoi(value); 206 } 207 // FIXME symbolic constants here 208 if (teeEnabled & 1) { 209 mTeeSinkInputEnabled = true; 210 } 211 if (teeEnabled & 2) { 212 mTeeSinkOutputEnabled = true; 213 } 214 if (teeEnabled & 4) { 215 mTeeSinkTrackEnabled = true; 216 } 217#endif 218} 219 220void AudioFlinger::onFirstRef() 221{ 222 int rc = 0; 223 224 Mutex::Autolock _l(mLock); 225 226 /* TODO: move all this work into an Init() function */ 227 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 228 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 229 uint32_t int_val; 230 if (1 == sscanf(val_str, "%u", &int_val)) { 231 mStandbyTimeInNsecs = milliseconds(int_val); 232 ALOGI("Using %u mSec as standby time.", int_val); 233 } else { 234 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 235 ALOGI("Using default %u mSec as standby time.", 236 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 237 } 238 } 239 240 mPatchPanel = new PatchPanel(this); 241 242 mMode = AUDIO_MODE_NORMAL; 243} 244 245AudioFlinger::~AudioFlinger() 246{ 247 while (!mRecordThreads.isEmpty()) { 248 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 249 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 250 } 251 while (!mPlaybackThreads.isEmpty()) { 252 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 253 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 254 } 255 256 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 257 // no mHardwareLock needed, as there are no other references to this 258 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 259 delete mAudioHwDevs.valueAt(i); 260 } 261 262 // Tell media.log service about any old writers that still need to be unregistered 263 if (mLogMemoryDealer != 0) { 264 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 265 if (binder != 0) { 266 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 267 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 268 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 269 mUnregisteredWriters.pop(); 270 mediaLogService->unregisterWriter(iMemory); 271 } 272 } 273 } 274} 275 276static const char * const audio_interfaces[] = { 277 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 278 AUDIO_HARDWARE_MODULE_ID_A2DP, 279 AUDIO_HARDWARE_MODULE_ID_USB, 280}; 281#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 282 283AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 284 audio_module_handle_t module, 285 audio_devices_t devices) 286{ 287 // if module is 0, the request comes from an old policy manager and we should load 288 // well known modules 289 if (module == 0) { 290 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 291 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 292 loadHwModule_l(audio_interfaces[i]); 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 297 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 298 if ((dev->get_supported_devices != NULL) && 299 (dev->get_supported_devices(dev) & devices) == devices) 300 return audioHwDevice; 301 } 302 } else { 303 // check a match for the requested module handle 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 305 if (audioHwDevice != NULL) { 306 return audioHwDevice; 307 } 308 } 309 310 return NULL; 311} 312 313void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 314{ 315 const size_t SIZE = 256; 316 char buffer[SIZE]; 317 String8 result; 318 319 result.append("Clients:\n"); 320 for (size_t i = 0; i < mClients.size(); ++i) { 321 sp<Client> client = mClients.valueAt(i).promote(); 322 if (client != 0) { 323 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 324 result.append(buffer); 325 } 326 } 327 328 result.append("Notification Clients:\n"); 329 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 330 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 331 result.append(buffer); 332 } 333 334 result.append("Global session refs:\n"); 335 result.append(" session pid count\n"); 336 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 337 AudioSessionRef *r = mAudioSessionRefs[i]; 338 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 339 result.append(buffer); 340 } 341 write(fd, result.string(), result.size()); 342} 343 344 345void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 346{ 347 const size_t SIZE = 256; 348 char buffer[SIZE]; 349 String8 result; 350 hardware_call_state hardwareStatus = mHardwareStatus; 351 352 snprintf(buffer, SIZE, "Hardware status: %d\n" 353 "Standby Time mSec: %u\n", 354 hardwareStatus, 355 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358} 359 360void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 361{ 362 const size_t SIZE = 256; 363 char buffer[SIZE]; 364 String8 result; 365 snprintf(buffer, SIZE, "Permission Denial: " 366 "can't dump AudioFlinger from pid=%d, uid=%d\n", 367 IPCThreadState::self()->getCallingPid(), 368 IPCThreadState::self()->getCallingUid()); 369 result.append(buffer); 370 write(fd, result.string(), result.size()); 371} 372 373bool AudioFlinger::dumpTryLock(Mutex& mutex) 374{ 375 bool locked = false; 376 for (int i = 0; i < kDumpLockRetries; ++i) { 377 if (mutex.tryLock() == NO_ERROR) { 378 locked = true; 379 break; 380 } 381 usleep(kDumpLockSleepUs); 382 } 383 return locked; 384} 385 386status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 387{ 388 if (!dumpAllowed()) { 389 dumpPermissionDenial(fd, args); 390 } else { 391 // get state of hardware lock 392 bool hardwareLocked = dumpTryLock(mHardwareLock); 393 if (!hardwareLocked) { 394 String8 result(kHardwareLockedString); 395 write(fd, result.string(), result.size()); 396 } else { 397 mHardwareLock.unlock(); 398 } 399 400 bool locked = dumpTryLock(mLock); 401 402 // failed to lock - AudioFlinger is probably deadlocked 403 if (!locked) { 404 String8 result(kDeadlockedString); 405 write(fd, result.string(), result.size()); 406 } 407 408 bool clientLocked = dumpTryLock(mClientLock); 409 if (!clientLocked) { 410 String8 result(kClientLockedString); 411 write(fd, result.string(), result.size()); 412 } 413 414 EffectDumpEffects(fd); 415 416 dumpClients(fd, args); 417 if (clientLocked) { 418 mClientLock.unlock(); 419 } 420 421 dumpInternals(fd, args); 422 423 // dump playback threads 424 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 425 mPlaybackThreads.valueAt(i)->dump(fd, args); 426 } 427 428 // dump record threads 429 for (size_t i = 0; i < mRecordThreads.size(); i++) { 430 mRecordThreads.valueAt(i)->dump(fd, args); 431 } 432 433 // dump orphan effect chains 434 if (mOrphanEffectChains.size() != 0) { 435 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 436 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 437 mOrphanEffectChains.valueAt(i)->dump(fd, args); 438 } 439 } 440 // dump all hardware devs 441 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 442 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 443 dev->dump(dev, fd); 444 } 445 446#ifdef TEE_SINK 447 // dump the serially shared record tee sink 448 if (mRecordTeeSource != 0) { 449 dumpTee(fd, mRecordTeeSource); 450 } 451#endif 452 453 if (locked) { 454 mLock.unlock(); 455 } 456 457 // append a copy of media.log here by forwarding fd to it, but don't attempt 458 // to lookup the service if it's not running, as it will block for a second 459 if (mLogMemoryDealer != 0) { 460 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 461 if (binder != 0) { 462 dprintf(fd, "\nmedia.log:\n"); 463 Vector<String16> args; 464 binder->dump(fd, args); 465 } 466 } 467 } 468 return NO_ERROR; 469} 470 471sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 472{ 473 Mutex::Autolock _cl(mClientLock); 474 // If pid is already in the mClients wp<> map, then use that entry 475 // (for which promote() is always != 0), otherwise create a new entry and Client. 476 sp<Client> client = mClients.valueFor(pid).promote(); 477 if (client == 0) { 478 client = new Client(this, pid); 479 mClients.add(pid, client); 480 } 481 482 return client; 483} 484 485sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 486{ 487 // If there is no memory allocated for logs, return a dummy writer that does nothing 488 if (mLogMemoryDealer == 0) { 489 return new NBLog::Writer(); 490 } 491 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 492 // Similarly if we can't contact the media.log service, also return a dummy writer 493 if (binder == 0) { 494 return new NBLog::Writer(); 495 } 496 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 497 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 498 // If allocation fails, consult the vector of previously unregistered writers 499 // and garbage-collect one or more them until an allocation succeeds 500 if (shared == 0) { 501 Mutex::Autolock _l(mUnregisteredWritersLock); 502 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 503 { 504 // Pick the oldest stale writer to garbage-collect 505 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 506 mUnregisteredWriters.removeAt(0); 507 mediaLogService->unregisterWriter(iMemory); 508 // Now the media.log remote reference to IMemory is gone. When our last local 509 // reference to IMemory also drops to zero at end of this block, 510 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 511 } 512 // Re-attempt the allocation 513 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 514 if (shared != 0) { 515 goto success; 516 } 517 } 518 // Even after garbage-collecting all old writers, there is still not enough memory, 519 // so return a dummy writer 520 return new NBLog::Writer(); 521 } 522success: 523 mediaLogService->registerWriter(shared, size, name); 524 return new NBLog::Writer(size, shared); 525} 526 527void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 528{ 529 if (writer == 0) { 530 return; 531 } 532 sp<IMemory> iMemory(writer->getIMemory()); 533 if (iMemory == 0) { 534 return; 535 } 536 // Rather than removing the writer immediately, append it to a queue of old writers to 537 // be garbage-collected later. This allows us to continue to view old logs for a while. 538 Mutex::Autolock _l(mUnregisteredWritersLock); 539 mUnregisteredWriters.push(writer); 540} 541 542// IAudioFlinger interface 543 544 545sp<IAudioTrack> AudioFlinger::createTrack( 546 audio_stream_type_t streamType, 547 uint32_t sampleRate, 548 audio_format_t format, 549 audio_channel_mask_t channelMask, 550 size_t *frameCount, 551 IAudioFlinger::track_flags_t *flags, 552 const sp<IMemory>& sharedBuffer, 553 audio_io_handle_t output, 554 pid_t tid, 555 audio_session_t *sessionId, 556 int clientUid, 557 status_t *status) 558{ 559 sp<PlaybackThread::Track> track; 560 sp<TrackHandle> trackHandle; 561 sp<Client> client; 562 status_t lStatus; 563 audio_session_t lSessionId; 564 565 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 566 // but if someone uses binder directly they could bypass that and cause us to crash 567 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 568 ALOGE("createTrack() invalid stream type %d", streamType); 569 lStatus = BAD_VALUE; 570 goto Exit; 571 } 572 573 // further sample rate checks are performed by createTrack_l() depending on the thread type 574 if (sampleRate == 0) { 575 ALOGE("createTrack() invalid sample rate %u", sampleRate); 576 lStatus = BAD_VALUE; 577 goto Exit; 578 } 579 580 // further channel mask checks are performed by createTrack_l() depending on the thread type 581 if (!audio_is_output_channel(channelMask)) { 582 ALOGE("createTrack() invalid channel mask %#x", channelMask); 583 lStatus = BAD_VALUE; 584 goto Exit; 585 } 586 587 // further format checks are performed by createTrack_l() depending on the thread type 588 if (!audio_is_valid_format(format)) { 589 ALOGE("createTrack() invalid format %#x", format); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 595 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 596 lStatus = BAD_VALUE; 597 goto Exit; 598 } 599 600 { 601 Mutex::Autolock _l(mLock); 602 PlaybackThread *thread = checkPlaybackThread_l(output); 603 if (thread == NULL) { 604 ALOGE("no playback thread found for output handle %d", output); 605 lStatus = BAD_VALUE; 606 goto Exit; 607 } 608 609 pid_t pid = IPCThreadState::self()->getCallingPid(); 610 client = registerPid(pid); 611 612 PlaybackThread *effectThread = NULL; 613 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 614 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 615 ALOGE("createTrack() invalid session ID %d", *sessionId); 616 lStatus = BAD_VALUE; 617 goto Exit; 618 } 619 lSessionId = *sessionId; 620 // check if an effect chain with the same session ID is present on another 621 // output thread and move it here. 622 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 623 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 624 if (mPlaybackThreads.keyAt(i) != output) { 625 uint32_t sessions = t->hasAudioSession(lSessionId); 626 if (sessions & PlaybackThread::EFFECT_SESSION) { 627 effectThread = t.get(); 628 break; 629 } 630 } 631 } 632 } else { 633 // if no audio session id is provided, create one here 634 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 635 if (sessionId != NULL) { 636 *sessionId = lSessionId; 637 } 638 } 639 ALOGV("createTrack() lSessionId: %d", lSessionId); 640 641 track = thread->createTrack_l(client, streamType, sampleRate, format, 642 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 643 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 644 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 645 646 // move effect chain to this output thread if an effect on same session was waiting 647 // for a track to be created 648 if (lStatus == NO_ERROR && effectThread != NULL) { 649 // no risk of deadlock because AudioFlinger::mLock is held 650 Mutex::Autolock _dl(thread->mLock); 651 Mutex::Autolock _sl(effectThread->mLock); 652 moveEffectChain_l(lSessionId, effectThread, thread, true); 653 } 654 655 // Look for sync events awaiting for a session to be used. 656 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 657 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 658 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 659 if (lStatus == NO_ERROR) { 660 (void) track->setSyncEvent(mPendingSyncEvents[i]); 661 } else { 662 mPendingSyncEvents[i]->cancel(); 663 } 664 mPendingSyncEvents.removeAt(i); 665 i--; 666 } 667 } 668 } 669 670 setAudioHwSyncForSession_l(thread, lSessionId); 671 } 672 673 if (lStatus != NO_ERROR) { 674 // remove local strong reference to Client before deleting the Track so that the 675 // Client destructor is called by the TrackBase destructor with mClientLock held 676 // Don't hold mClientLock when releasing the reference on the track as the 677 // destructor will acquire it. 678 { 679 Mutex::Autolock _cl(mClientLock); 680 client.clear(); 681 } 682 track.clear(); 683 goto Exit; 684 } 685 686 // return handle to client 687 trackHandle = new TrackHandle(track); 688 689Exit: 690 *status = lStatus; 691 return trackHandle; 692} 693 694uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 695{ 696 Mutex::Autolock _l(mLock); 697 ThreadBase *thread = checkThread_l(ioHandle); 698 if (thread == NULL) { 699 ALOGW("sampleRate() unknown thread %d", ioHandle); 700 return 0; 701 } 702 return thread->sampleRate(); 703} 704 705audio_format_t AudioFlinger::format(audio_io_handle_t output) const 706{ 707 Mutex::Autolock _l(mLock); 708 PlaybackThread *thread = checkPlaybackThread_l(output); 709 if (thread == NULL) { 710 ALOGW("format() unknown thread %d", output); 711 return AUDIO_FORMAT_INVALID; 712 } 713 return thread->format(); 714} 715 716size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 717{ 718 Mutex::Autolock _l(mLock); 719 ThreadBase *thread = checkThread_l(ioHandle); 720 if (thread == NULL) { 721 ALOGW("frameCount() unknown thread %d", ioHandle); 722 return 0; 723 } 724 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 725 // should examine all callers and fix them to handle smaller counts 726 return thread->frameCount(); 727} 728 729uint32_t AudioFlinger::latency(audio_io_handle_t output) const 730{ 731 Mutex::Autolock _l(mLock); 732 PlaybackThread *thread = checkPlaybackThread_l(output); 733 if (thread == NULL) { 734 ALOGW("latency(): no playback thread found for output handle %d", output); 735 return 0; 736 } 737 return thread->latency(); 738} 739 740status_t AudioFlinger::setMasterVolume(float value) 741{ 742 status_t ret = initCheck(); 743 if (ret != NO_ERROR) { 744 return ret; 745 } 746 747 // check calling permissions 748 if (!settingsAllowed()) { 749 return PERMISSION_DENIED; 750 } 751 752 Mutex::Autolock _l(mLock); 753 mMasterVolume = value; 754 755 // Set master volume in the HALs which support it. 756 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 757 AutoMutex lock(mHardwareLock); 758 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 759 760 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 761 if (dev->canSetMasterVolume()) { 762 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 763 } 764 mHardwareStatus = AUDIO_HW_IDLE; 765 } 766 767 // Now set the master volume in each playback thread. Playback threads 768 // assigned to HALs which do not have master volume support will apply 769 // master volume during the mix operation. Threads with HALs which do 770 // support master volume will simply ignore the setting. 771 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 772 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 773 continue; 774 } 775 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 776 } 777 778 return NO_ERROR; 779} 780 781status_t AudioFlinger::setMode(audio_mode_t mode) 782{ 783 status_t ret = initCheck(); 784 if (ret != NO_ERROR) { 785 return ret; 786 } 787 788 // check calling permissions 789 if (!settingsAllowed()) { 790 return PERMISSION_DENIED; 791 } 792 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 793 ALOGW("Illegal value: setMode(%d)", mode); 794 return BAD_VALUE; 795 } 796 797 { // scope for the lock 798 AutoMutex lock(mHardwareLock); 799 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 800 mHardwareStatus = AUDIO_HW_SET_MODE; 801 ret = dev->set_mode(dev, mode); 802 mHardwareStatus = AUDIO_HW_IDLE; 803 } 804 805 if (NO_ERROR == ret) { 806 Mutex::Autolock _l(mLock); 807 mMode = mode; 808 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setMode(mode); 810 } 811 812 return ret; 813} 814 815status_t AudioFlinger::setMicMute(bool state) 816{ 817 status_t ret = initCheck(); 818 if (ret != NO_ERROR) { 819 return ret; 820 } 821 822 // check calling permissions 823 if (!settingsAllowed()) { 824 return PERMISSION_DENIED; 825 } 826 827 AutoMutex lock(mHardwareLock); 828 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 829 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 830 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 831 status_t result = dev->set_mic_mute(dev, state); 832 if (result != NO_ERROR) { 833 ret = result; 834 } 835 } 836 mHardwareStatus = AUDIO_HW_IDLE; 837 return ret; 838} 839 840bool AudioFlinger::getMicMute() const 841{ 842 status_t ret = initCheck(); 843 if (ret != NO_ERROR) { 844 return false; 845 } 846 bool mute = true; 847 bool state = AUDIO_MODE_INVALID; 848 AutoMutex lock(mHardwareLock); 849 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 850 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 851 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 852 status_t result = dev->get_mic_mute(dev, &state); 853 if (result == NO_ERROR) { 854 mute = mute && state; 855 } 856 } 857 mHardwareStatus = AUDIO_HW_IDLE; 858 859 return mute; 860} 861 862status_t AudioFlinger::setMasterMute(bool muted) 863{ 864 status_t ret = initCheck(); 865 if (ret != NO_ERROR) { 866 return ret; 867 } 868 869 // check calling permissions 870 if (!settingsAllowed()) { 871 return PERMISSION_DENIED; 872 } 873 874 Mutex::Autolock _l(mLock); 875 mMasterMute = muted; 876 877 // Set master mute in the HALs which support it. 878 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 879 AutoMutex lock(mHardwareLock); 880 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 881 882 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 883 if (dev->canSetMasterMute()) { 884 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 885 } 886 mHardwareStatus = AUDIO_HW_IDLE; 887 } 888 889 // Now set the master mute in each playback thread. Playback threads 890 // assigned to HALs which do not have master mute support will apply master 891 // mute during the mix operation. Threads with HALs which do support master 892 // mute will simply ignore the setting. 893 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 894 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 895 continue; 896 } 897 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 898 } 899 900 return NO_ERROR; 901} 902 903float AudioFlinger::masterVolume() const 904{ 905 Mutex::Autolock _l(mLock); 906 return masterVolume_l(); 907} 908 909bool AudioFlinger::masterMute() const 910{ 911 Mutex::Autolock _l(mLock); 912 return masterMute_l(); 913} 914 915float AudioFlinger::masterVolume_l() const 916{ 917 return mMasterVolume; 918} 919 920bool AudioFlinger::masterMute_l() const 921{ 922 return mMasterMute; 923} 924 925status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 926{ 927 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 928 ALOGW("setStreamVolume() invalid stream %d", stream); 929 return BAD_VALUE; 930 } 931 pid_t caller = IPCThreadState::self()->getCallingPid(); 932 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 933 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 934 return PERMISSION_DENIED; 935 } 936 937 return NO_ERROR; 938} 939 940status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 941 audio_io_handle_t output) 942{ 943 // check calling permissions 944 if (!settingsAllowed()) { 945 return PERMISSION_DENIED; 946 } 947 948 status_t status = checkStreamType(stream); 949 if (status != NO_ERROR) { 950 return status; 951 } 952 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 953 954 AutoMutex lock(mLock); 955 PlaybackThread *thread = NULL; 956 if (output != AUDIO_IO_HANDLE_NONE) { 957 thread = checkPlaybackThread_l(output); 958 if (thread == NULL) { 959 return BAD_VALUE; 960 } 961 } 962 963 mStreamTypes[stream].volume = value; 964 965 if (thread == NULL) { 966 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 967 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 968 } 969 } else { 970 thread->setStreamVolume(stream, value); 971 } 972 973 return NO_ERROR; 974} 975 976status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 977{ 978 // check calling permissions 979 if (!settingsAllowed()) { 980 return PERMISSION_DENIED; 981 } 982 983 status_t status = checkStreamType(stream); 984 if (status != NO_ERROR) { 985 return status; 986 } 987 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 988 989 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 990 ALOGE("setStreamMute() invalid stream %d", stream); 991 return BAD_VALUE; 992 } 993 994 AutoMutex lock(mLock); 995 mStreamTypes[stream].mute = muted; 996 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 997 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 998 999 return NO_ERROR; 1000} 1001 1002float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1003{ 1004 status_t status = checkStreamType(stream); 1005 if (status != NO_ERROR) { 1006 return 0.0f; 1007 } 1008 1009 AutoMutex lock(mLock); 1010 float volume; 1011 if (output != AUDIO_IO_HANDLE_NONE) { 1012 PlaybackThread *thread = checkPlaybackThread_l(output); 1013 if (thread == NULL) { 1014 return 0.0f; 1015 } 1016 volume = thread->streamVolume(stream); 1017 } else { 1018 volume = streamVolume_l(stream); 1019 } 1020 1021 return volume; 1022} 1023 1024bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1025{ 1026 status_t status = checkStreamType(stream); 1027 if (status != NO_ERROR) { 1028 return true; 1029 } 1030 1031 AutoMutex lock(mLock); 1032 return streamMute_l(stream); 1033} 1034 1035 1036void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1037{ 1038 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1039 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1040 } 1041} 1042 1043status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1044{ 1045 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1046 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1047 1048 // check calling permissions 1049 if (!settingsAllowed()) { 1050 return PERMISSION_DENIED; 1051 } 1052 1053 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1054 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1055 Mutex::Autolock _l(mLock); 1056 status_t final_result = NO_ERROR; 1057 { 1058 AutoMutex lock(mHardwareLock); 1059 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1060 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1061 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1062 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1063 final_result = result ?: final_result; 1064 } 1065 mHardwareStatus = AUDIO_HW_IDLE; 1066 } 1067 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1068 AudioParameter param = AudioParameter(keyValuePairs); 1069 String8 value; 1070 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1071 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1072 if (mBtNrecIsOff != btNrecIsOff) { 1073 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1074 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1075 audio_devices_t device = thread->inDevice(); 1076 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1077 // collect all of the thread's session IDs 1078 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1079 // suspend effects associated with those session IDs 1080 for (size_t j = 0; j < ids.size(); ++j) { 1081 audio_session_t sessionId = ids.keyAt(j); 1082 thread->setEffectSuspended(FX_IID_AEC, 1083 suspend, 1084 sessionId); 1085 thread->setEffectSuspended(FX_IID_NS, 1086 suspend, 1087 sessionId); 1088 } 1089 } 1090 mBtNrecIsOff = btNrecIsOff; 1091 } 1092 } 1093 String8 screenState; 1094 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1095 bool isOff = screenState == "off"; 1096 if (isOff != (AudioFlinger::mScreenState & 1)) { 1097 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1098 } 1099 } 1100 return final_result; 1101 } 1102 1103 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1104 // and the thread is exited once the lock is released 1105 sp<ThreadBase> thread; 1106 { 1107 Mutex::Autolock _l(mLock); 1108 thread = checkPlaybackThread_l(ioHandle); 1109 if (thread == 0) { 1110 thread = checkRecordThread_l(ioHandle); 1111 } else if (thread == primaryPlaybackThread_l()) { 1112 // indicate output device change to all input threads for pre processing 1113 AudioParameter param = AudioParameter(keyValuePairs); 1114 int value; 1115 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1116 (value != 0)) { 1117 broacastParametersToRecordThreads_l(keyValuePairs); 1118 } 1119 } 1120 } 1121 if (thread != 0) { 1122 return thread->setParameters(keyValuePairs); 1123 } 1124 return BAD_VALUE; 1125} 1126 1127String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1128{ 1129 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1130 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1131 1132 Mutex::Autolock _l(mLock); 1133 1134 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1135 String8 out_s8; 1136 1137 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1138 char *s; 1139 { 1140 AutoMutex lock(mHardwareLock); 1141 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1142 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1143 s = dev->get_parameters(dev, keys.string()); 1144 mHardwareStatus = AUDIO_HW_IDLE; 1145 } 1146 out_s8 += String8(s ? s : ""); 1147 free(s); 1148 } 1149 return out_s8; 1150 } 1151 1152 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1153 if (playbackThread != NULL) { 1154 return playbackThread->getParameters(keys); 1155 } 1156 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1157 if (recordThread != NULL) { 1158 return recordThread->getParameters(keys); 1159 } 1160 return String8(""); 1161} 1162 1163size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1164 audio_channel_mask_t channelMask) const 1165{ 1166 status_t ret = initCheck(); 1167 if (ret != NO_ERROR) { 1168 return 0; 1169 } 1170 if ((sampleRate == 0) || 1171 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1172 !audio_is_input_channel(channelMask)) { 1173 return 0; 1174 } 1175 1176 AutoMutex lock(mHardwareLock); 1177 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1178 audio_config_t config, proposed; 1179 memset(&proposed, 0, sizeof(proposed)); 1180 proposed.sample_rate = sampleRate; 1181 proposed.channel_mask = channelMask; 1182 proposed.format = format; 1183 1184 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1185 size_t frames; 1186 for (;;) { 1187 // Note: config is currently a const parameter for get_input_buffer_size() 1188 // but we use a copy from proposed in case config changes from the call. 1189 config = proposed; 1190 frames = dev->get_input_buffer_size(dev, &config); 1191 if (frames != 0) { 1192 break; // hal success, config is the result 1193 } 1194 // change one parameter of the configuration each iteration to a more "common" value 1195 // to see if the device will support it. 1196 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1197 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1198 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1199 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1200 } else { 1201 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1202 "format %#x, channelMask 0x%X", 1203 sampleRate, format, channelMask); 1204 break; // retries failed, break out of loop with frames == 0. 1205 } 1206 } 1207 mHardwareStatus = AUDIO_HW_IDLE; 1208 if (frames > 0 && config.sample_rate != sampleRate) { 1209 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1210 } 1211 return frames; // may be converted to bytes at the Java level. 1212} 1213 1214uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1215{ 1216 Mutex::Autolock _l(mLock); 1217 1218 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1219 if (recordThread != NULL) { 1220 return recordThread->getInputFramesLost(); 1221 } 1222 return 0; 1223} 1224 1225status_t AudioFlinger::setVoiceVolume(float value) 1226{ 1227 status_t ret = initCheck(); 1228 if (ret != NO_ERROR) { 1229 return ret; 1230 } 1231 1232 // check calling permissions 1233 if (!settingsAllowed()) { 1234 return PERMISSION_DENIED; 1235 } 1236 1237 AutoMutex lock(mHardwareLock); 1238 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1239 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1240 ret = dev->set_voice_volume(dev, value); 1241 mHardwareStatus = AUDIO_HW_IDLE; 1242 1243 return ret; 1244} 1245 1246status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1247 audio_io_handle_t output) const 1248{ 1249 status_t status; 1250 1251 Mutex::Autolock _l(mLock); 1252 1253 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1254 if (playbackThread != NULL) { 1255 return playbackThread->getRenderPosition(halFrames, dspFrames); 1256 } 1257 1258 return BAD_VALUE; 1259} 1260 1261void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1262{ 1263 Mutex::Autolock _l(mLock); 1264 if (client == 0) { 1265 return; 1266 } 1267 pid_t pid = IPCThreadState::self()->getCallingPid(); 1268 { 1269 Mutex::Autolock _cl(mClientLock); 1270 if (mNotificationClients.indexOfKey(pid) < 0) { 1271 sp<NotificationClient> notificationClient = new NotificationClient(this, 1272 client, 1273 pid); 1274 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1275 1276 mNotificationClients.add(pid, notificationClient); 1277 1278 sp<IBinder> binder = IInterface::asBinder(client); 1279 binder->linkToDeath(notificationClient); 1280 } 1281 } 1282 1283 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1284 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1285 // the config change is always sent from playback or record threads to avoid deadlock 1286 // with AudioSystem::gLock 1287 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1288 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1289 } 1290 1291 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1292 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1293 } 1294} 1295 1296void AudioFlinger::removeNotificationClient(pid_t pid) 1297{ 1298 Mutex::Autolock _l(mLock); 1299 { 1300 Mutex::Autolock _cl(mClientLock); 1301 mNotificationClients.removeItem(pid); 1302 } 1303 1304 ALOGV("%d died, releasing its sessions", pid); 1305 size_t num = mAudioSessionRefs.size(); 1306 bool removed = false; 1307 for (size_t i = 0; i< num; ) { 1308 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1309 ALOGV(" pid %d @ %d", ref->mPid, i); 1310 if (ref->mPid == pid) { 1311 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1312 mAudioSessionRefs.removeAt(i); 1313 delete ref; 1314 removed = true; 1315 num--; 1316 } else { 1317 i++; 1318 } 1319 } 1320 if (removed) { 1321 purgeStaleEffects_l(); 1322 } 1323} 1324 1325void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1326 const sp<AudioIoDescriptor>& ioDesc, 1327 pid_t pid) 1328{ 1329 Mutex::Autolock _l(mClientLock); 1330 size_t size = mNotificationClients.size(); 1331 for (size_t i = 0; i < size; i++) { 1332 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1333 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1334 } 1335 } 1336} 1337 1338// removeClient_l() must be called with AudioFlinger::mClientLock held 1339void AudioFlinger::removeClient_l(pid_t pid) 1340{ 1341 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1342 IPCThreadState::self()->getCallingPid()); 1343 mClients.removeItem(pid); 1344} 1345 1346// getEffectThread_l() must be called with AudioFlinger::mLock held 1347sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1348 int EffectId) 1349{ 1350 sp<PlaybackThread> thread; 1351 1352 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1353 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1354 ALOG_ASSERT(thread == 0); 1355 thread = mPlaybackThreads.valueAt(i); 1356 } 1357 } 1358 1359 return thread; 1360} 1361 1362 1363 1364// ---------------------------------------------------------------------------- 1365 1366AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1367 : RefBase(), 1368 mAudioFlinger(audioFlinger), 1369 mPid(pid) 1370{ 1371 size_t heapSize = kClientSharedHeapSizeBytes; 1372 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1373 // invalidated tracks 1374 if (!audioFlinger->isLowRamDevice()) { 1375 heapSize *= kClientSharedHeapSizeMultiplier; 1376 } 1377 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1378} 1379 1380// Client destructor must be called with AudioFlinger::mClientLock held 1381AudioFlinger::Client::~Client() 1382{ 1383 mAudioFlinger->removeClient_l(mPid); 1384} 1385 1386sp<MemoryDealer> AudioFlinger::Client::heap() const 1387{ 1388 return mMemoryDealer; 1389} 1390 1391// ---------------------------------------------------------------------------- 1392 1393AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1394 const sp<IAudioFlingerClient>& client, 1395 pid_t pid) 1396 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1397{ 1398} 1399 1400AudioFlinger::NotificationClient::~NotificationClient() 1401{ 1402} 1403 1404void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1405{ 1406 sp<NotificationClient> keep(this); 1407 mAudioFlinger->removeNotificationClient(mPid); 1408} 1409 1410 1411// ---------------------------------------------------------------------------- 1412 1413static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1414 return audio_is_remote_submix_device(inDevice); 1415} 1416 1417sp<IAudioRecord> AudioFlinger::openRecord( 1418 audio_io_handle_t input, 1419 uint32_t sampleRate, 1420 audio_format_t format, 1421 audio_channel_mask_t channelMask, 1422 const String16& opPackageName, 1423 size_t *frameCount, 1424 IAudioFlinger::track_flags_t *flags, 1425 pid_t tid, 1426 int clientUid, 1427 audio_session_t *sessionId, 1428 size_t *notificationFrames, 1429 sp<IMemory>& cblk, 1430 sp<IMemory>& buffers, 1431 status_t *status) 1432{ 1433 sp<RecordThread::RecordTrack> recordTrack; 1434 sp<RecordHandle> recordHandle; 1435 sp<Client> client; 1436 status_t lStatus; 1437 audio_session_t lSessionId; 1438 1439 cblk.clear(); 1440 buffers.clear(); 1441 1442 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1443 if (!isTrustedCallingUid(callingUid)) { 1444 ALOGW_IF((uid_t)clientUid != callingUid, 1445 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1446 clientUid = callingUid; 1447 } 1448 1449 // check calling permissions 1450 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1451 ALOGE("openRecord() permission denied: recording not allowed"); 1452 lStatus = PERMISSION_DENIED; 1453 goto Exit; 1454 } 1455 1456 // further sample rate checks are performed by createRecordTrack_l() 1457 if (sampleRate == 0) { 1458 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1459 lStatus = BAD_VALUE; 1460 goto Exit; 1461 } 1462 1463 // we don't yet support anything other than linear PCM 1464 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1465 ALOGE("openRecord() invalid format %#x", format); 1466 lStatus = BAD_VALUE; 1467 goto Exit; 1468 } 1469 1470 // further channel mask checks are performed by createRecordTrack_l() 1471 if (!audio_is_input_channel(channelMask)) { 1472 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1473 lStatus = BAD_VALUE; 1474 goto Exit; 1475 } 1476 1477 { 1478 Mutex::Autolock _l(mLock); 1479 RecordThread *thread = checkRecordThread_l(input); 1480 if (thread == NULL) { 1481 ALOGE("openRecord() checkRecordThread_l failed"); 1482 lStatus = BAD_VALUE; 1483 goto Exit; 1484 } 1485 1486 pid_t pid = IPCThreadState::self()->getCallingPid(); 1487 client = registerPid(pid); 1488 1489 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1490 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1491 lStatus = BAD_VALUE; 1492 goto Exit; 1493 } 1494 lSessionId = *sessionId; 1495 } else { 1496 // if no audio session id is provided, create one here 1497 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1498 if (sessionId != NULL) { 1499 *sessionId = lSessionId; 1500 } 1501 } 1502 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1503 1504 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1505 frameCount, lSessionId, notificationFrames, 1506 clientUid, flags, tid, &lStatus); 1507 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1508 1509 if (lStatus == NO_ERROR) { 1510 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1511 // session and move it to this thread. 1512 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1513 if (chain != 0) { 1514 Mutex::Autolock _l(thread->mLock); 1515 thread->addEffectChain_l(chain); 1516 } 1517 } 1518 } 1519 1520 if (lStatus != NO_ERROR) { 1521 // remove local strong reference to Client before deleting the RecordTrack so that the 1522 // Client destructor is called by the TrackBase destructor with mClientLock held 1523 // Don't hold mClientLock when releasing the reference on the track as the 1524 // destructor will acquire it. 1525 { 1526 Mutex::Autolock _cl(mClientLock); 1527 client.clear(); 1528 } 1529 recordTrack.clear(); 1530 goto Exit; 1531 } 1532 1533 cblk = recordTrack->getCblk(); 1534 buffers = recordTrack->getBuffers(); 1535 1536 // return handle to client 1537 recordHandle = new RecordHandle(recordTrack); 1538 1539Exit: 1540 *status = lStatus; 1541 return recordHandle; 1542} 1543 1544 1545 1546// ---------------------------------------------------------------------------- 1547 1548audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1549{ 1550 if (name == NULL) { 1551 return 0; 1552 } 1553 if (!settingsAllowed()) { 1554 return 0; 1555 } 1556 Mutex::Autolock _l(mLock); 1557 return loadHwModule_l(name); 1558} 1559 1560// loadHwModule_l() must be called with AudioFlinger::mLock held 1561audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1562{ 1563 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1564 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1565 ALOGW("loadHwModule() module %s already loaded", name); 1566 return mAudioHwDevs.keyAt(i); 1567 } 1568 } 1569 1570 audio_hw_device_t *dev; 1571 1572 int rc = load_audio_interface(name, &dev); 1573 if (rc) { 1574 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1575 return 0; 1576 } 1577 1578 mHardwareStatus = AUDIO_HW_INIT; 1579 rc = dev->init_check(dev); 1580 mHardwareStatus = AUDIO_HW_IDLE; 1581 if (rc) { 1582 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1583 return 0; 1584 } 1585 1586 // Check and cache this HAL's level of support for master mute and master 1587 // volume. If this is the first HAL opened, and it supports the get 1588 // methods, use the initial values provided by the HAL as the current 1589 // master mute and volume settings. 1590 1591 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1592 { // scope for auto-lock pattern 1593 AutoMutex lock(mHardwareLock); 1594 1595 if (0 == mAudioHwDevs.size()) { 1596 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1597 if (NULL != dev->get_master_volume) { 1598 float mv; 1599 if (OK == dev->get_master_volume(dev, &mv)) { 1600 mMasterVolume = mv; 1601 } 1602 } 1603 1604 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1605 if (NULL != dev->get_master_mute) { 1606 bool mm; 1607 if (OK == dev->get_master_mute(dev, &mm)) { 1608 mMasterMute = mm; 1609 } 1610 } 1611 } 1612 1613 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1614 if ((NULL != dev->set_master_volume) && 1615 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1616 flags = static_cast<AudioHwDevice::Flags>(flags | 1617 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1618 } 1619 1620 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1621 if ((NULL != dev->set_master_mute) && 1622 (OK == dev->set_master_mute(dev, mMasterMute))) { 1623 flags = static_cast<AudioHwDevice::Flags>(flags | 1624 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1625 } 1626 1627 mHardwareStatus = AUDIO_HW_IDLE; 1628 } 1629 1630 audio_module_handle_t handle = nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1631 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1632 1633 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1634 name, dev->common.module->name, dev->common.module->id, handle); 1635 1636 return handle; 1637 1638} 1639 1640// ---------------------------------------------------------------------------- 1641 1642uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1643{ 1644 Mutex::Autolock _l(mLock); 1645 PlaybackThread *thread = primaryPlaybackThread_l(); 1646 return thread != NULL ? thread->sampleRate() : 0; 1647} 1648 1649size_t AudioFlinger::getPrimaryOutputFrameCount() 1650{ 1651 Mutex::Autolock _l(mLock); 1652 PlaybackThread *thread = primaryPlaybackThread_l(); 1653 return thread != NULL ? thread->frameCountHAL() : 0; 1654} 1655 1656// ---------------------------------------------------------------------------- 1657 1658status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1659{ 1660 uid_t uid = IPCThreadState::self()->getCallingUid(); 1661 if (uid != AID_SYSTEM) { 1662 return PERMISSION_DENIED; 1663 } 1664 Mutex::Autolock _l(mLock); 1665 if (mIsDeviceTypeKnown) { 1666 return INVALID_OPERATION; 1667 } 1668 mIsLowRamDevice = isLowRamDevice; 1669 mIsDeviceTypeKnown = true; 1670 return NO_ERROR; 1671} 1672 1673audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1674{ 1675 Mutex::Autolock _l(mLock); 1676 1677 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1678 if (index >= 0) { 1679 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1680 mHwAvSyncIds.valueAt(index), sessionId); 1681 return mHwAvSyncIds.valueAt(index); 1682 } 1683 1684 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1685 if (dev == NULL) { 1686 return AUDIO_HW_SYNC_INVALID; 1687 } 1688 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1689 AudioParameter param = AudioParameter(String8(reply)); 1690 free(reply); 1691 1692 int value; 1693 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1694 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1695 return AUDIO_HW_SYNC_INVALID; 1696 } 1697 1698 // allow only one session for a given HW A/V sync ID. 1699 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1700 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1701 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1702 value, mHwAvSyncIds.keyAt(i)); 1703 mHwAvSyncIds.removeItemsAt(i); 1704 break; 1705 } 1706 } 1707 1708 mHwAvSyncIds.add(sessionId, value); 1709 1710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1711 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1712 uint32_t sessions = thread->hasAudioSession(sessionId); 1713 if (sessions & PlaybackThread::TRACK_SESSION) { 1714 AudioParameter param = AudioParameter(); 1715 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1716 thread->setParameters(param.toString()); 1717 break; 1718 } 1719 } 1720 1721 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1722 return (audio_hw_sync_t)value; 1723} 1724 1725status_t AudioFlinger::systemReady() 1726{ 1727 Mutex::Autolock _l(mLock); 1728 ALOGI("%s", __FUNCTION__); 1729 if (mSystemReady) { 1730 ALOGW("%s called twice", __FUNCTION__); 1731 return NO_ERROR; 1732 } 1733 mSystemReady = true; 1734 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1735 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1736 thread->systemReady(); 1737 } 1738 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1739 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1740 thread->systemReady(); 1741 } 1742 return NO_ERROR; 1743} 1744 1745// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1746void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1747{ 1748 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1749 if (index >= 0) { 1750 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1751 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1752 AudioParameter param = AudioParameter(); 1753 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1754 thread->setParameters(param.toString()); 1755 } 1756} 1757 1758 1759// ---------------------------------------------------------------------------- 1760 1761 1762sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1763 audio_io_handle_t *output, 1764 audio_config_t *config, 1765 audio_devices_t devices, 1766 const String8& address, 1767 audio_output_flags_t flags) 1768{ 1769 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1770 if (outHwDev == NULL) { 1771 return 0; 1772 } 1773 1774 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1775 1776 if (*output == AUDIO_IO_HANDLE_NONE) { 1777 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1778 } else { 1779 // Audio Policy does not currently request a specific output handle. 1780 // If this is ever needed, see openInput_l() for example code. 1781 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1782 return 0; 1783 } 1784 1785 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1786 1787 // FOR TESTING ONLY: 1788 // This if statement allows overriding the audio policy settings 1789 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1790 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1791 // Check only for Normal Mixing mode 1792 if (kEnableExtendedPrecision) { 1793 // Specify format (uncomment one below to choose) 1794 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1795 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1796 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1797 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1798 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1799 } 1800 if (kEnableExtendedChannels) { 1801 // Specify channel mask (uncomment one below to choose) 1802 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1803 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1804 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1805 } 1806 } 1807 1808 AudioStreamOut *outputStream = NULL; 1809 status_t status = outHwDev->openOutputStream( 1810 &outputStream, 1811 *output, 1812 devices, 1813 flags, 1814 config, 1815 address.string()); 1816 1817 mHardwareStatus = AUDIO_HW_IDLE; 1818 1819 if (status == NO_ERROR) { 1820 1821 PlaybackThread *thread; 1822 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1823 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady, 1824 config->offload_info.bit_rate); 1825 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1826 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1827 || !isValidPcmSinkFormat(config->format) 1828 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1829 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1830 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1831 } else { 1832 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1833 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1834 } 1835 mPlaybackThreads.add(*output, thread); 1836 return thread; 1837 } 1838 1839 return 0; 1840} 1841 1842status_t AudioFlinger::openOutput(audio_module_handle_t module, 1843 audio_io_handle_t *output, 1844 audio_config_t *config, 1845 audio_devices_t *devices, 1846 const String8& address, 1847 uint32_t *latencyMs, 1848 audio_output_flags_t flags) 1849{ 1850 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1851 module, 1852 (devices != NULL) ? *devices : 0, 1853 config->sample_rate, 1854 config->format, 1855 config->channel_mask, 1856 flags); 1857 1858 if (*devices == AUDIO_DEVICE_NONE) { 1859 return BAD_VALUE; 1860 } 1861 1862 Mutex::Autolock _l(mLock); 1863 1864 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1865 if (thread != 0) { 1866 *latencyMs = thread->latency(); 1867 1868 // notify client processes of the new output creation 1869 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1870 1871 // the first primary output opened designates the primary hw device 1872 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1873 ALOGI("Using module %d has the primary audio interface", module); 1874 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1875 1876 AutoMutex lock(mHardwareLock); 1877 mHardwareStatus = AUDIO_HW_SET_MODE; 1878 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1879 mHardwareStatus = AUDIO_HW_IDLE; 1880 } 1881 return NO_ERROR; 1882 } 1883 1884 return NO_INIT; 1885} 1886 1887audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1888 audio_io_handle_t output2) 1889{ 1890 Mutex::Autolock _l(mLock); 1891 MixerThread *thread1 = checkMixerThread_l(output1); 1892 MixerThread *thread2 = checkMixerThread_l(output2); 1893 1894 if (thread1 == NULL || thread2 == NULL) { 1895 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1896 output2); 1897 return AUDIO_IO_HANDLE_NONE; 1898 } 1899 1900 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1901 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1902 thread->addOutputTrack(thread2); 1903 mPlaybackThreads.add(id, thread); 1904 // notify client processes of the new output creation 1905 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1906 return id; 1907} 1908 1909status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1910{ 1911 return closeOutput_nonvirtual(output); 1912} 1913 1914status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1915{ 1916 // keep strong reference on the playback thread so that 1917 // it is not destroyed while exit() is executed 1918 sp<PlaybackThread> thread; 1919 { 1920 Mutex::Autolock _l(mLock); 1921 thread = checkPlaybackThread_l(output); 1922 if (thread == NULL) { 1923 return BAD_VALUE; 1924 } 1925 1926 ALOGV("closeOutput() %d", output); 1927 1928 if (thread->type() == ThreadBase::MIXER) { 1929 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1930 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1931 DuplicatingThread *dupThread = 1932 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1933 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1934 } 1935 } 1936 } 1937 1938 1939 mPlaybackThreads.removeItem(output); 1940 // save all effects to the default thread 1941 if (mPlaybackThreads.size()) { 1942 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1943 if (dstThread != NULL) { 1944 // audioflinger lock is held here so the acquisition order of thread locks does not 1945 // matter 1946 Mutex::Autolock _dl(dstThread->mLock); 1947 Mutex::Autolock _sl(thread->mLock); 1948 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1949 for (size_t i = 0; i < effectChains.size(); i ++) { 1950 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1951 } 1952 } 1953 } 1954 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1955 ioDesc->mIoHandle = output; 1956 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1957 } 1958 thread->exit(); 1959 // The thread entity (active unit of execution) is no longer running here, 1960 // but the ThreadBase container still exists. 1961 1962 if (!thread->isDuplicating()) { 1963 closeOutputFinish(thread); 1964 } 1965 1966 return NO_ERROR; 1967} 1968 1969void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1970{ 1971 AudioStreamOut *out = thread->clearOutput(); 1972 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1973 // from now on thread->mOutput is NULL 1974 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1975 delete out; 1976} 1977 1978void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1979{ 1980 mPlaybackThreads.removeItem(thread->mId); 1981 thread->exit(); 1982 closeOutputFinish(thread); 1983} 1984 1985status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1986{ 1987 Mutex::Autolock _l(mLock); 1988 PlaybackThread *thread = checkPlaybackThread_l(output); 1989 1990 if (thread == NULL) { 1991 return BAD_VALUE; 1992 } 1993 1994 ALOGV("suspendOutput() %d", output); 1995 thread->suspend(); 1996 1997 return NO_ERROR; 1998} 1999 2000status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2001{ 2002 Mutex::Autolock _l(mLock); 2003 PlaybackThread *thread = checkPlaybackThread_l(output); 2004 2005 if (thread == NULL) { 2006 return BAD_VALUE; 2007 } 2008 2009 ALOGV("restoreOutput() %d", output); 2010 2011 thread->restore(); 2012 2013 return NO_ERROR; 2014} 2015 2016status_t AudioFlinger::openInput(audio_module_handle_t module, 2017 audio_io_handle_t *input, 2018 audio_config_t *config, 2019 audio_devices_t *devices, 2020 const String8& address, 2021 audio_source_t source, 2022 audio_input_flags_t flags) 2023{ 2024 Mutex::Autolock _l(mLock); 2025 2026 if (*devices == AUDIO_DEVICE_NONE) { 2027 return BAD_VALUE; 2028 } 2029 2030 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2031 2032 if (thread != 0) { 2033 // notify client processes of the new input creation 2034 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2035 return NO_ERROR; 2036 } 2037 return NO_INIT; 2038} 2039 2040sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2041 audio_io_handle_t *input, 2042 audio_config_t *config, 2043 audio_devices_t devices, 2044 const String8& address, 2045 audio_source_t source, 2046 audio_input_flags_t flags) 2047{ 2048 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2049 if (inHwDev == NULL) { 2050 *input = AUDIO_IO_HANDLE_NONE; 2051 return 0; 2052 } 2053 2054 // Audio Policy can request a specific handle for hardware hotword. 2055 // The goal here is not to re-open an already opened input. 2056 // It is to use a pre-assigned I/O handle. 2057 if (*input == AUDIO_IO_HANDLE_NONE) { 2058 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2059 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2060 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2061 return 0; 2062 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2063 // This should not happen in a transient state with current design. 2064 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2065 return 0; 2066 } 2067 2068 audio_config_t halconfig = *config; 2069 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2070 audio_stream_in_t *inStream = NULL; 2071 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2072 &inStream, flags, address.string(), source); 2073 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2074 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2075 inStream, 2076 halconfig.sample_rate, 2077 halconfig.format, 2078 halconfig.channel_mask, 2079 flags, 2080 status, address.string()); 2081 2082 // If the input could not be opened with the requested parameters and we can handle the 2083 // conversion internally, try to open again with the proposed parameters. 2084 if (status == BAD_VALUE && 2085 audio_is_linear_pcm(config->format) && 2086 audio_is_linear_pcm(halconfig.format) && 2087 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2088 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2089 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2090 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2091 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2092 inStream = NULL; 2093 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2094 &inStream, flags, address.string(), source); 2095 // FIXME log this new status; HAL should not propose any further changes 2096 } 2097 2098 if (status == NO_ERROR && inStream != NULL) { 2099 2100#ifdef TEE_SINK 2101 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2102 // or (re-)create if current Pipe is idle and does not match the new format 2103 sp<NBAIO_Sink> teeSink; 2104 enum { 2105 TEE_SINK_NO, // don't copy input 2106 TEE_SINK_NEW, // copy input using a new pipe 2107 TEE_SINK_OLD, // copy input using an existing pipe 2108 } kind; 2109 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2110 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2111 if (!mTeeSinkInputEnabled) { 2112 kind = TEE_SINK_NO; 2113 } else if (!Format_isValid(format)) { 2114 kind = TEE_SINK_NO; 2115 } else if (mRecordTeeSink == 0) { 2116 kind = TEE_SINK_NEW; 2117 } else if (mRecordTeeSink->getStrongCount() != 1) { 2118 kind = TEE_SINK_NO; 2119 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2120 kind = TEE_SINK_OLD; 2121 } else { 2122 kind = TEE_SINK_NEW; 2123 } 2124 switch (kind) { 2125 case TEE_SINK_NEW: { 2126 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2127 size_t numCounterOffers = 0; 2128 const NBAIO_Format offers[1] = {format}; 2129 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2130 ALOG_ASSERT(index == 0); 2131 PipeReader *pipeReader = new PipeReader(*pipe); 2132 numCounterOffers = 0; 2133 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2134 ALOG_ASSERT(index == 0); 2135 mRecordTeeSink = pipe; 2136 mRecordTeeSource = pipeReader; 2137 teeSink = pipe; 2138 } 2139 break; 2140 case TEE_SINK_OLD: 2141 teeSink = mRecordTeeSink; 2142 break; 2143 case TEE_SINK_NO: 2144 default: 2145 break; 2146 } 2147#endif 2148 2149 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2150 2151 // Start record thread 2152 // RecordThread requires both input and output device indication to forward to audio 2153 // pre processing modules 2154 sp<RecordThread> thread = new RecordThread(this, 2155 inputStream, 2156 *input, 2157 primaryOutputDevice_l(), 2158 devices, 2159 mSystemReady 2160#ifdef TEE_SINK 2161 , teeSink 2162#endif 2163 ); 2164 mRecordThreads.add(*input, thread); 2165 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2166 return thread; 2167 } 2168 2169 *input = AUDIO_IO_HANDLE_NONE; 2170 return 0; 2171} 2172 2173status_t AudioFlinger::closeInput(audio_io_handle_t input) 2174{ 2175 return closeInput_nonvirtual(input); 2176} 2177 2178status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2179{ 2180 // keep strong reference on the record thread so that 2181 // it is not destroyed while exit() is executed 2182 sp<RecordThread> thread; 2183 { 2184 Mutex::Autolock _l(mLock); 2185 thread = checkRecordThread_l(input); 2186 if (thread == 0) { 2187 return BAD_VALUE; 2188 } 2189 2190 ALOGV("closeInput() %d", input); 2191 2192 // If we still have effect chains, it means that a client still holds a handle 2193 // on at least one effect. We must either move the chain to an existing thread with the 2194 // same session ID or put it aside in case a new record thread is opened for a 2195 // new capture on the same session 2196 sp<EffectChain> chain; 2197 { 2198 Mutex::Autolock _sl(thread->mLock); 2199 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2200 // Note: maximum one chain per record thread 2201 if (effectChains.size() != 0) { 2202 chain = effectChains[0]; 2203 } 2204 } 2205 if (chain != 0) { 2206 // first check if a record thread is already opened with a client on the same session. 2207 // This should only happen in case of overlap between one thread tear down and the 2208 // creation of its replacement 2209 size_t i; 2210 for (i = 0; i < mRecordThreads.size(); i++) { 2211 sp<RecordThread> t = mRecordThreads.valueAt(i); 2212 if (t == thread) { 2213 continue; 2214 } 2215 if (t->hasAudioSession(chain->sessionId()) != 0) { 2216 Mutex::Autolock _l(t->mLock); 2217 ALOGV("closeInput() found thread %d for effect session %d", 2218 t->id(), chain->sessionId()); 2219 t->addEffectChain_l(chain); 2220 break; 2221 } 2222 } 2223 // put the chain aside if we could not find a record thread with the same session id. 2224 if (i == mRecordThreads.size()) { 2225 putOrphanEffectChain_l(chain); 2226 } 2227 } 2228 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2229 ioDesc->mIoHandle = input; 2230 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2231 mRecordThreads.removeItem(input); 2232 } 2233 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2234 // we have a different lock for notification client 2235 closeInputFinish(thread); 2236 return NO_ERROR; 2237} 2238 2239void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2240{ 2241 thread->exit(); 2242 AudioStreamIn *in = thread->clearInput(); 2243 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2244 // from now on thread->mInput is NULL 2245 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2246 delete in; 2247} 2248 2249void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2250{ 2251 mRecordThreads.removeItem(thread->mId); 2252 closeInputFinish(thread); 2253} 2254 2255status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2256{ 2257 Mutex::Autolock _l(mLock); 2258 ALOGV("invalidateStream() stream %d", stream); 2259 2260 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2261 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2262 thread->invalidateTracks(stream); 2263 } 2264 2265 return NO_ERROR; 2266} 2267 2268 2269audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2270{ 2271 return nextUniqueId(use); 2272} 2273 2274void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2275{ 2276 Mutex::Autolock _l(mLock); 2277 pid_t caller = IPCThreadState::self()->getCallingPid(); 2278 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2279 if (pid != -1 && (caller == getpid_cached)) { 2280 caller = pid; 2281 } 2282 2283 { 2284 Mutex::Autolock _cl(mClientLock); 2285 // Ignore requests received from processes not known as notification client. The request 2286 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2287 // called from a different pid leaving a stale session reference. Also we don't know how 2288 // to clear this reference if the client process dies. 2289 if (mNotificationClients.indexOfKey(caller) < 0) { 2290 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2291 return; 2292 } 2293 } 2294 2295 size_t num = mAudioSessionRefs.size(); 2296 for (size_t i = 0; i< num; i++) { 2297 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2298 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2299 ref->mCnt++; 2300 ALOGV(" incremented refcount to %d", ref->mCnt); 2301 return; 2302 } 2303 } 2304 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2305 ALOGV(" added new entry for %d", audioSession); 2306} 2307 2308void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2309{ 2310 Mutex::Autolock _l(mLock); 2311 pid_t caller = IPCThreadState::self()->getCallingPid(); 2312 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2313 if (pid != -1 && (caller == getpid_cached)) { 2314 caller = pid; 2315 } 2316 size_t num = mAudioSessionRefs.size(); 2317 for (size_t i = 0; i< num; i++) { 2318 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2319 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2320 ref->mCnt--; 2321 ALOGV(" decremented refcount to %d", ref->mCnt); 2322 if (ref->mCnt == 0) { 2323 mAudioSessionRefs.removeAt(i); 2324 delete ref; 2325 purgeStaleEffects_l(); 2326 } 2327 return; 2328 } 2329 } 2330 // If the caller is mediaserver it is likely that the session being released was acquired 2331 // on behalf of a process not in notification clients and we ignore the warning. 2332 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2333} 2334 2335void AudioFlinger::purgeStaleEffects_l() { 2336 2337 ALOGV("purging stale effects"); 2338 2339 Vector< sp<EffectChain> > chains; 2340 2341 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2342 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2343 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2344 sp<EffectChain> ec = t->mEffectChains[j]; 2345 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2346 chains.push(ec); 2347 } 2348 } 2349 } 2350 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2351 sp<RecordThread> t = mRecordThreads.valueAt(i); 2352 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2353 sp<EffectChain> ec = t->mEffectChains[j]; 2354 chains.push(ec); 2355 } 2356 } 2357 2358 for (size_t i = 0; i < chains.size(); i++) { 2359 sp<EffectChain> ec = chains[i]; 2360 int sessionid = ec->sessionId(); 2361 sp<ThreadBase> t = ec->mThread.promote(); 2362 if (t == 0) { 2363 continue; 2364 } 2365 size_t numsessionrefs = mAudioSessionRefs.size(); 2366 bool found = false; 2367 for (size_t k = 0; k < numsessionrefs; k++) { 2368 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2369 if (ref->mSessionid == sessionid) { 2370 ALOGV(" session %d still exists for %d with %d refs", 2371 sessionid, ref->mPid, ref->mCnt); 2372 found = true; 2373 break; 2374 } 2375 } 2376 if (!found) { 2377 Mutex::Autolock _l(t->mLock); 2378 // remove all effects from the chain 2379 while (ec->mEffects.size()) { 2380 sp<EffectModule> effect = ec->mEffects[0]; 2381 effect->unPin(); 2382 t->removeEffect_l(effect); 2383 if (effect->purgeHandles()) { 2384 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2385 } 2386 AudioSystem::unregisterEffect(effect->id()); 2387 } 2388 } 2389 } 2390 return; 2391} 2392 2393// checkThread_l() must be called with AudioFlinger::mLock held 2394AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2395{ 2396 ThreadBase *thread = NULL; 2397 switch (audio_unique_id_get_use(ioHandle)) { 2398 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2399 thread = checkPlaybackThread_l(ioHandle); 2400 break; 2401 case AUDIO_UNIQUE_ID_USE_INPUT: 2402 thread = checkRecordThread_l(ioHandle); 2403 break; 2404 default: 2405 break; 2406 } 2407 return thread; 2408} 2409 2410// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2411AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2412{ 2413 return mPlaybackThreads.valueFor(output).get(); 2414} 2415 2416// checkMixerThread_l() must be called with AudioFlinger::mLock held 2417AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2418{ 2419 PlaybackThread *thread = checkPlaybackThread_l(output); 2420 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2421} 2422 2423// checkRecordThread_l() must be called with AudioFlinger::mLock held 2424AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2425{ 2426 return mRecordThreads.valueFor(input).get(); 2427} 2428 2429audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2430{ 2431 int32_t base = android_atomic_add(AUDIO_UNIQUE_ID_USE_MAX, &mNextUniqueId); 2432 // We have no way of recovering from wraparound 2433 LOG_ALWAYS_FATAL_IF(base == 0, "unique ID overflow"); 2434 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2435 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2436 return (audio_unique_id_t) (base | use); 2437} 2438 2439AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2440{ 2441 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2442 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2443 if(thread->isDuplicating()) { 2444 continue; 2445 } 2446 AudioStreamOut *output = thread->getOutput(); 2447 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2448 return thread; 2449 } 2450 } 2451 return NULL; 2452} 2453 2454audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2455{ 2456 PlaybackThread *thread = primaryPlaybackThread_l(); 2457 2458 if (thread == NULL) { 2459 return 0; 2460 } 2461 2462 return thread->outDevice(); 2463} 2464 2465sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2466 audio_session_t triggerSession, 2467 audio_session_t listenerSession, 2468 sync_event_callback_t callBack, 2469 wp<RefBase> cookie) 2470{ 2471 Mutex::Autolock _l(mLock); 2472 2473 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2474 status_t playStatus = NAME_NOT_FOUND; 2475 status_t recStatus = NAME_NOT_FOUND; 2476 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2477 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2478 if (playStatus == NO_ERROR) { 2479 return event; 2480 } 2481 } 2482 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2483 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2484 if (recStatus == NO_ERROR) { 2485 return event; 2486 } 2487 } 2488 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2489 mPendingSyncEvents.add(event); 2490 } else { 2491 ALOGV("createSyncEvent() invalid event %d", event->type()); 2492 event.clear(); 2493 } 2494 return event; 2495} 2496 2497// ---------------------------------------------------------------------------- 2498// Effect management 2499// ---------------------------------------------------------------------------- 2500 2501 2502status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2503{ 2504 Mutex::Autolock _l(mLock); 2505 return EffectQueryNumberEffects(numEffects); 2506} 2507 2508status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2509{ 2510 Mutex::Autolock _l(mLock); 2511 return EffectQueryEffect(index, descriptor); 2512} 2513 2514status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2515 effect_descriptor_t *descriptor) const 2516{ 2517 Mutex::Autolock _l(mLock); 2518 return EffectGetDescriptor(pUuid, descriptor); 2519} 2520 2521 2522sp<IEffect> AudioFlinger::createEffect( 2523 effect_descriptor_t *pDesc, 2524 const sp<IEffectClient>& effectClient, 2525 int32_t priority, 2526 audio_io_handle_t io, 2527 audio_session_t sessionId, 2528 const String16& opPackageName, 2529 status_t *status, 2530 int *id, 2531 int *enabled) 2532{ 2533 status_t lStatus = NO_ERROR; 2534 sp<EffectHandle> handle; 2535 effect_descriptor_t desc; 2536 2537 pid_t pid = IPCThreadState::self()->getCallingPid(); 2538 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2539 pid, effectClient.get(), priority, sessionId, io); 2540 2541 if (pDesc == NULL) { 2542 lStatus = BAD_VALUE; 2543 goto Exit; 2544 } 2545 2546 // check audio settings permission for global effects 2547 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2548 lStatus = PERMISSION_DENIED; 2549 goto Exit; 2550 } 2551 2552 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2553 // that can only be created by audio policy manager (running in same process) 2554 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2555 lStatus = PERMISSION_DENIED; 2556 goto Exit; 2557 } 2558 2559 { 2560 if (!EffectIsNullUuid(&pDesc->uuid)) { 2561 // if uuid is specified, request effect descriptor 2562 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2563 if (lStatus < 0) { 2564 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2565 goto Exit; 2566 } 2567 } else { 2568 // if uuid is not specified, look for an available implementation 2569 // of the required type in effect factory 2570 if (EffectIsNullUuid(&pDesc->type)) { 2571 ALOGW("createEffect() no effect type"); 2572 lStatus = BAD_VALUE; 2573 goto Exit; 2574 } 2575 uint32_t numEffects = 0; 2576 effect_descriptor_t d; 2577 d.flags = 0; // prevent compiler warning 2578 bool found = false; 2579 2580 lStatus = EffectQueryNumberEffects(&numEffects); 2581 if (lStatus < 0) { 2582 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2583 goto Exit; 2584 } 2585 for (uint32_t i = 0; i < numEffects; i++) { 2586 lStatus = EffectQueryEffect(i, &desc); 2587 if (lStatus < 0) { 2588 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2589 continue; 2590 } 2591 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2592 // If matching type found save effect descriptor. If the session is 2593 // 0 and the effect is not auxiliary, continue enumeration in case 2594 // an auxiliary version of this effect type is available 2595 found = true; 2596 d = desc; 2597 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2598 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2599 break; 2600 } 2601 } 2602 } 2603 if (!found) { 2604 lStatus = BAD_VALUE; 2605 ALOGW("createEffect() effect not found"); 2606 goto Exit; 2607 } 2608 // For same effect type, chose auxiliary version over insert version if 2609 // connect to output mix (Compliance to OpenSL ES) 2610 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2611 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2612 desc = d; 2613 } 2614 } 2615 2616 // Do not allow auxiliary effects on a session different from 0 (output mix) 2617 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2618 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2619 lStatus = INVALID_OPERATION; 2620 goto Exit; 2621 } 2622 2623 // check recording permission for visualizer 2624 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2625 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2626 lStatus = PERMISSION_DENIED; 2627 goto Exit; 2628 } 2629 2630 // return effect descriptor 2631 *pDesc = desc; 2632 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2633 // if the output returned by getOutputForEffect() is removed before we lock the 2634 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2635 // and we will exit safely 2636 io = AudioSystem::getOutputForEffect(&desc); 2637 ALOGV("createEffect got output %d", io); 2638 } 2639 2640 Mutex::Autolock _l(mLock); 2641 2642 // If output is not specified try to find a matching audio session ID in one of the 2643 // output threads. 2644 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2645 // because of code checking output when entering the function. 2646 // Note: io is never 0 when creating an effect on an input 2647 if (io == AUDIO_IO_HANDLE_NONE) { 2648 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2649 // output must be specified by AudioPolicyManager when using session 2650 // AUDIO_SESSION_OUTPUT_STAGE 2651 lStatus = BAD_VALUE; 2652 goto Exit; 2653 } 2654 // look for the thread where the specified audio session is present 2655 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2656 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2657 io = mPlaybackThreads.keyAt(i); 2658 break; 2659 } 2660 } 2661 if (io == 0) { 2662 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2663 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2664 io = mRecordThreads.keyAt(i); 2665 break; 2666 } 2667 } 2668 } 2669 // If no output thread contains the requested session ID, default to 2670 // first output. The effect chain will be moved to the correct output 2671 // thread when a track with the same session ID is created 2672 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2673 io = mPlaybackThreads.keyAt(0); 2674 } 2675 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2676 } 2677 ThreadBase *thread = checkRecordThread_l(io); 2678 if (thread == NULL) { 2679 thread = checkPlaybackThread_l(io); 2680 if (thread == NULL) { 2681 ALOGE("createEffect() unknown output thread"); 2682 lStatus = BAD_VALUE; 2683 goto Exit; 2684 } 2685 } else { 2686 // Check if one effect chain was awaiting for an effect to be created on this 2687 // session and used it instead of creating a new one. 2688 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2689 if (chain != 0) { 2690 Mutex::Autolock _l(thread->mLock); 2691 thread->addEffectChain_l(chain); 2692 } 2693 } 2694 2695 sp<Client> client = registerPid(pid); 2696 2697 // create effect on selected output thread 2698 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2699 &desc, enabled, &lStatus); 2700 if (handle != 0 && id != NULL) { 2701 *id = handle->id(); 2702 } 2703 if (handle == 0) { 2704 // remove local strong reference to Client with mClientLock held 2705 Mutex::Autolock _cl(mClientLock); 2706 client.clear(); 2707 } 2708 } 2709 2710Exit: 2711 *status = lStatus; 2712 return handle; 2713} 2714 2715status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2716 audio_io_handle_t dstOutput) 2717{ 2718 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2719 sessionId, srcOutput, dstOutput); 2720 Mutex::Autolock _l(mLock); 2721 if (srcOutput == dstOutput) { 2722 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2723 return NO_ERROR; 2724 } 2725 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2726 if (srcThread == NULL) { 2727 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2728 return BAD_VALUE; 2729 } 2730 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2731 if (dstThread == NULL) { 2732 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2733 return BAD_VALUE; 2734 } 2735 2736 Mutex::Autolock _dl(dstThread->mLock); 2737 Mutex::Autolock _sl(srcThread->mLock); 2738 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2739} 2740 2741// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2742status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2743 AudioFlinger::PlaybackThread *srcThread, 2744 AudioFlinger::PlaybackThread *dstThread, 2745 bool reRegister) 2746{ 2747 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2748 sessionId, srcThread, dstThread); 2749 2750 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2751 if (chain == 0) { 2752 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2753 sessionId, srcThread); 2754 return INVALID_OPERATION; 2755 } 2756 2757 // Check whether the destination thread has a channel count of FCC_2, which is 2758 // currently required for (most) effects. Prevent moving the effect chain here rather 2759 // than disabling the addEffect_l() call in dstThread below. 2760 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2761 dstThread->mChannelCount != FCC_2) { 2762 ALOGW("moveEffectChain_l() effect chain failed because" 2763 " destination thread %p channel count(%u) != %u", 2764 dstThread, dstThread->mChannelCount, FCC_2); 2765 return INVALID_OPERATION; 2766 } 2767 2768 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2769 // so that a new chain is created with correct parameters when first effect is added. This is 2770 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2771 // removed. 2772 srcThread->removeEffectChain_l(chain); 2773 2774 // transfer all effects one by one so that new effect chain is created on new thread with 2775 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2776 sp<EffectChain> dstChain; 2777 uint32_t strategy = 0; // prevent compiler warning 2778 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2779 Vector< sp<EffectModule> > removed; 2780 status_t status = NO_ERROR; 2781 while (effect != 0) { 2782 srcThread->removeEffect_l(effect); 2783 removed.add(effect); 2784 status = dstThread->addEffect_l(effect); 2785 if (status != NO_ERROR) { 2786 break; 2787 } 2788 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2789 if (effect->state() == EffectModule::ACTIVE || 2790 effect->state() == EffectModule::STOPPING) { 2791 effect->start(); 2792 } 2793 // if the move request is not received from audio policy manager, the effect must be 2794 // re-registered with the new strategy and output 2795 if (dstChain == 0) { 2796 dstChain = effect->chain().promote(); 2797 if (dstChain == 0) { 2798 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2799 status = NO_INIT; 2800 break; 2801 } 2802 strategy = dstChain->strategy(); 2803 } 2804 if (reRegister) { 2805 AudioSystem::unregisterEffect(effect->id()); 2806 AudioSystem::registerEffect(&effect->desc(), 2807 dstThread->id(), 2808 strategy, 2809 sessionId, 2810 effect->id()); 2811 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2812 } 2813 effect = chain->getEffectFromId_l(0); 2814 } 2815 2816 if (status != NO_ERROR) { 2817 for (size_t i = 0; i < removed.size(); i++) { 2818 srcThread->addEffect_l(removed[i]); 2819 if (dstChain != 0 && reRegister) { 2820 AudioSystem::unregisterEffect(removed[i]->id()); 2821 AudioSystem::registerEffect(&removed[i]->desc(), 2822 srcThread->id(), 2823 strategy, 2824 sessionId, 2825 removed[i]->id()); 2826 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2827 } 2828 } 2829 } 2830 2831 return status; 2832} 2833 2834bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2835{ 2836 if (mGlobalEffectEnableTime != 0 && 2837 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2838 return true; 2839 } 2840 2841 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2842 sp<EffectChain> ec = 2843 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2844 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2845 return true; 2846 } 2847 } 2848 return false; 2849} 2850 2851void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2852{ 2853 Mutex::Autolock _l(mLock); 2854 2855 mGlobalEffectEnableTime = systemTime(); 2856 2857 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2858 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2859 if (t->mType == ThreadBase::OFFLOAD) { 2860 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2861 } 2862 } 2863 2864} 2865 2866status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2867{ 2868 audio_session_t session = chain->sessionId(); 2869 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2870 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2871 if (index >= 0) { 2872 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2873 return ALREADY_EXISTS; 2874 } 2875 mOrphanEffectChains.add(session, chain); 2876 return NO_ERROR; 2877} 2878 2879sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2880{ 2881 sp<EffectChain> chain; 2882 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2883 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2884 if (index >= 0) { 2885 chain = mOrphanEffectChains.valueAt(index); 2886 mOrphanEffectChains.removeItemsAt(index); 2887 } 2888 return chain; 2889} 2890 2891bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2892{ 2893 Mutex::Autolock _l(mLock); 2894 audio_session_t session = effect->sessionId(); 2895 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2896 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2897 if (index >= 0) { 2898 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2899 if (chain->removeEffect_l(effect) == 0) { 2900 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2901 mOrphanEffectChains.removeItemsAt(index); 2902 } 2903 return true; 2904 } 2905 return false; 2906} 2907 2908 2909struct Entry { 2910#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2911 char mFileName[TEE_MAX_FILENAME]; 2912}; 2913 2914int comparEntry(const void *p1, const void *p2) 2915{ 2916 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2917} 2918 2919#ifdef TEE_SINK 2920void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2921{ 2922 NBAIO_Source *teeSource = source.get(); 2923 if (teeSource != NULL) { 2924 // .wav rotation 2925 // There is a benign race condition if 2 threads call this simultaneously. 2926 // They would both traverse the directory, but the result would simply be 2927 // failures at unlink() which are ignored. It's also unlikely since 2928 // normally dumpsys is only done by bugreport or from the command line. 2929 char teePath[32+256]; 2930 strcpy(teePath, "/data/misc/audioserver"); 2931 size_t teePathLen = strlen(teePath); 2932 DIR *dir = opendir(teePath); 2933 teePath[teePathLen++] = '/'; 2934 if (dir != NULL) { 2935#define TEE_MAX_SORT 20 // number of entries to sort 2936#define TEE_MAX_KEEP 10 // number of entries to keep 2937 struct Entry entries[TEE_MAX_SORT]; 2938 size_t entryCount = 0; 2939 while (entryCount < TEE_MAX_SORT) { 2940 struct dirent de; 2941 struct dirent *result = NULL; 2942 int rc = readdir_r(dir, &de, &result); 2943 if (rc != 0) { 2944 ALOGW("readdir_r failed %d", rc); 2945 break; 2946 } 2947 if (result == NULL) { 2948 break; 2949 } 2950 if (result != &de) { 2951 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2952 break; 2953 } 2954 // ignore non .wav file entries 2955 size_t nameLen = strlen(de.d_name); 2956 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2957 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2958 continue; 2959 } 2960 strcpy(entries[entryCount++].mFileName, de.d_name); 2961 } 2962 (void) closedir(dir); 2963 if (entryCount > TEE_MAX_KEEP) { 2964 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2965 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2966 strcpy(&teePath[teePathLen], entries[i].mFileName); 2967 (void) unlink(teePath); 2968 } 2969 } 2970 } else { 2971 if (fd >= 0) { 2972 dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath, 2973 strerror(errno)); 2974 } 2975 } 2976 char teeTime[16]; 2977 struct timeval tv; 2978 gettimeofday(&tv, NULL); 2979 struct tm tm; 2980 localtime_r(&tv.tv_sec, &tm); 2981 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2982 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2983 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2984 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2985 if (teeFd >= 0) { 2986 // FIXME use libsndfile 2987 char wavHeader[44]; 2988 memcpy(wavHeader, 2989 "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", 2990 sizeof(wavHeader)); 2991 NBAIO_Format format = teeSource->format(); 2992 unsigned channelCount = Format_channelCount(format); 2993 uint32_t sampleRate = Format_sampleRate(format); 2994 size_t frameSize = Format_frameSize(format); 2995 wavHeader[22] = channelCount; // number of channels 2996 wavHeader[24] = sampleRate; // sample rate 2997 wavHeader[25] = sampleRate >> 8; 2998 wavHeader[32] = frameSize; // block alignment 2999 wavHeader[33] = frameSize >> 8; 3000 write(teeFd, wavHeader, sizeof(wavHeader)); 3001 size_t total = 0; 3002 bool firstRead = true; 3003#define TEE_SINK_READ 1024 // frames per I/O operation 3004 void *buffer = malloc(TEE_SINK_READ * frameSize); 3005 for (;;) { 3006 size_t count = TEE_SINK_READ; 3007 ssize_t actual = teeSource->read(buffer, count); 3008 bool wasFirstRead = firstRead; 3009 firstRead = false; 3010 if (actual <= 0) { 3011 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3012 continue; 3013 } 3014 break; 3015 } 3016 ALOG_ASSERT(actual <= (ssize_t)count); 3017 write(teeFd, buffer, actual * frameSize); 3018 total += actual; 3019 } 3020 free(buffer); 3021 lseek(teeFd, (off_t) 4, SEEK_SET); 3022 uint32_t temp = 44 + total * frameSize - 8; 3023 // FIXME not big-endian safe 3024 write(teeFd, &temp, sizeof(temp)); 3025 lseek(teeFd, (off_t) 40, SEEK_SET); 3026 temp = total * frameSize; 3027 // FIXME not big-endian safe 3028 write(teeFd, &temp, sizeof(temp)); 3029 close(teeFd); 3030 if (fd >= 0) { 3031 dprintf(fd, "tee copied to %s\n", teePath); 3032 } 3033 } else { 3034 if (fd >= 0) { 3035 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3036 } 3037 } 3038 } 3039} 3040#endif 3041 3042// ---------------------------------------------------------------------------- 3043 3044status_t AudioFlinger::onTransact( 3045 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3046{ 3047 return BnAudioFlinger::onTransact(code, data, reply, flags); 3048} 3049 3050} // namespace android 3051