AudioFlinger.cpp revision 35fec5f61393124c9e13958941637b8fe386385e
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 <memunreachable/memunreachable.h> 35#include <utils/String16.h> 36#include <utils/threads.h> 37#include <utils/Atomic.h> 38 39#include <cutils/bitops.h> 40#include <cutils/properties.h> 41 42#include <system/audio.h> 43#include <hardware/audio.h> 44 45#include "AudioMixer.h" 46#include "AudioFlinger.h" 47#include "ServiceUtilities.h" 48 49#include <media/AudioResamplerPublic.h> 50 51#include <media/EffectsFactoryApi.h> 52#include <audio_effects/effect_visualizer.h> 53#include <audio_effects/effect_ns.h> 54#include <audio_effects/effect_aec.h> 55 56#include <audio_utils/primitives.h> 57 58#include <powermanager/PowerManager.h> 59 60#include <media/IMediaLogService.h> 61 62#include <media/nbaio/Pipe.h> 63#include <media/nbaio/PipeReader.h> 64#include <media/AudioParameter.h> 65#include <mediautils/BatteryNotifier.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (audio_get_main_format(format)) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 134 case AUDIO_FORMAT_IEC61937: return "iec61937"; 135 default: 136 break; 137 } 138 return "unknown"; 139} 140 141static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 142{ 143 const hw_module_t *mod; 144 int rc; 145 146 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 147 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 148 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 149 if (rc) { 150 goto out; 151 } 152 rc = audio_hw_device_open(mod, dev); 153 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 154 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 155 if (rc) { 156 goto out; 157 } 158 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 159 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 160 rc = BAD_VALUE; 161 goto out; 162 } 163 return 0; 164 165out: 166 *dev = NULL; 167 return rc; 168} 169 170// ---------------------------------------------------------------------------- 171 172AudioFlinger::AudioFlinger() 173 : BnAudioFlinger(), 174 mPrimaryHardwareDev(NULL), 175 mAudioHwDevs(NULL), 176 mHardwareStatus(AUDIO_HW_IDLE), 177 mMasterVolume(1.0f), 178 mMasterMute(false), 179 mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), // zero has a special meaning, so unavailable 180 mMode(AUDIO_MODE_INVALID), 181 mBtNrecIsOff(false), 182 mIsLowRamDevice(true), 183 mIsDeviceTypeKnown(false), 184 mGlobalEffectEnableTime(0), 185 mSystemReady(false) 186{ 187 getpid_cached = getpid(); 188 const bool doLog = property_get_bool("ro.test_harness", false); 189 if (doLog) { 190 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 191 MemoryHeapBase::READ_ONLY); 192 } 193 194 // reset battery stats. 195 // if the audio service has crashed, battery stats could be left 196 // in bad state, reset the state upon service start. 197 BatteryNotifier::getInstance().noteResetAudio(); 198 199#ifdef TEE_SINK 200 char value[PROPERTY_VALUE_MAX]; 201 (void) property_get("ro.debuggable", value, "0"); 202 int debuggable = atoi(value); 203 int teeEnabled = 0; 204 if (debuggable) { 205 (void) property_get("af.tee", value, "0"); 206 teeEnabled = atoi(value); 207 } 208 // FIXME symbolic constants here 209 if (teeEnabled & 1) { 210 mTeeSinkInputEnabled = true; 211 } 212 if (teeEnabled & 2) { 213 mTeeSinkOutputEnabled = true; 214 } 215 if (teeEnabled & 4) { 216 mTeeSinkTrackEnabled = true; 217 } 218#endif 219} 220 221void AudioFlinger::onFirstRef() 222{ 223 Mutex::Autolock _l(mLock); 224 225 /* TODO: move all this work into an Init() function */ 226 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 227 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 228 uint32_t int_val; 229 if (1 == sscanf(val_str, "%u", &int_val)) { 230 mStandbyTimeInNsecs = milliseconds(int_val); 231 ALOGI("Using %u mSec as standby time.", int_val); 232 } else { 233 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 234 ALOGI("Using default %u mSec as standby time.", 235 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 236 } 237 } 238 239 mPatchPanel = new PatchPanel(this); 240 241 mMode = AUDIO_MODE_NORMAL; 242} 243 244AudioFlinger::~AudioFlinger() 245{ 246 while (!mRecordThreads.isEmpty()) { 247 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 248 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 249 } 250 while (!mPlaybackThreads.isEmpty()) { 251 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 252 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 253 } 254 255 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 256 // no mHardwareLock needed, as there are no other references to this 257 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 258 delete mAudioHwDevs.valueAt(i); 259 } 260 261 // Tell media.log service about any old writers that still need to be unregistered 262 if (mLogMemoryDealer != 0) { 263 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 264 if (binder != 0) { 265 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 266 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 267 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 268 mUnregisteredWriters.pop(); 269 mediaLogService->unregisterWriter(iMemory); 270 } 271 } 272 } 273} 274 275static const char * const audio_interfaces[] = { 276 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 277 AUDIO_HARDWARE_MODULE_ID_A2DP, 278 AUDIO_HARDWARE_MODULE_ID_USB, 279}; 280#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 281 282AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 283 audio_module_handle_t module, 284 audio_devices_t devices) 285{ 286 // if module is 0, the request comes from an old policy manager and we should load 287 // well known modules 288 if (module == 0) { 289 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 290 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 291 loadHwModule_l(audio_interfaces[i]); 292 } 293 // then try to find a module supporting the requested device. 294 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 295 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 296 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 297 if ((dev->get_supported_devices != NULL) && 298 (dev->get_supported_devices(dev) & devices) == devices) 299 return audioHwDevice; 300 } 301 } else { 302 // check a match for the requested module handle 303 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 304 if (audioHwDevice != NULL) { 305 return audioHwDevice; 306 } 307 } 308 309 return NULL; 310} 311 312void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 313{ 314 const size_t SIZE = 256; 315 char buffer[SIZE]; 316 String8 result; 317 318 result.append("Clients:\n"); 319 for (size_t i = 0; i < mClients.size(); ++i) { 320 sp<Client> client = mClients.valueAt(i).promote(); 321 if (client != 0) { 322 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 323 result.append(buffer); 324 } 325 } 326 327 result.append("Notification Clients:\n"); 328 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 329 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 330 result.append(buffer); 331 } 332 333 result.append("Global session refs:\n"); 334 result.append(" session pid count\n"); 335 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 336 AudioSessionRef *r = mAudioSessionRefs[i]; 337 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 338 result.append(buffer); 339 } 340 write(fd, result.string(), result.size()); 341} 342 343 344void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 345{ 346 const size_t SIZE = 256; 347 char buffer[SIZE]; 348 String8 result; 349 hardware_call_state hardwareStatus = mHardwareStatus; 350 351 snprintf(buffer, SIZE, "Hardware status: %d\n" 352 "Standby Time mSec: %u\n", 353 hardwareStatus, 354 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 355 result.append(buffer); 356 write(fd, result.string(), result.size()); 357} 358 359void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 360{ 361 const size_t SIZE = 256; 362 char buffer[SIZE]; 363 String8 result; 364 snprintf(buffer, SIZE, "Permission Denial: " 365 "can't dump AudioFlinger from pid=%d, uid=%d\n", 366 IPCThreadState::self()->getCallingPid(), 367 IPCThreadState::self()->getCallingUid()); 368 result.append(buffer); 369 write(fd, result.string(), result.size()); 370} 371 372bool AudioFlinger::dumpTryLock(Mutex& mutex) 373{ 374 bool locked = false; 375 for (int i = 0; i < kDumpLockRetries; ++i) { 376 if (mutex.tryLock() == NO_ERROR) { 377 locked = true; 378 break; 379 } 380 usleep(kDumpLockSleepUs); 381 } 382 return locked; 383} 384 385status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 386{ 387 if (!dumpAllowed()) { 388 dumpPermissionDenial(fd, args); 389 } else { 390 // get state of hardware lock 391 bool hardwareLocked = dumpTryLock(mHardwareLock); 392 if (!hardwareLocked) { 393 String8 result(kHardwareLockedString); 394 write(fd, result.string(), result.size()); 395 } else { 396 mHardwareLock.unlock(); 397 } 398 399 bool locked = dumpTryLock(mLock); 400 401 // failed to lock - AudioFlinger is probably deadlocked 402 if (!locked) { 403 String8 result(kDeadlockedString); 404 write(fd, result.string(), result.size()); 405 } 406 407 bool clientLocked = dumpTryLock(mClientLock); 408 if (!clientLocked) { 409 String8 result(kClientLockedString); 410 write(fd, result.string(), result.size()); 411 } 412 413 EffectDumpEffects(fd); 414 415 dumpClients(fd, args); 416 if (clientLocked) { 417 mClientLock.unlock(); 418 } 419 420 dumpInternals(fd, args); 421 422 // dump playback threads 423 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 424 mPlaybackThreads.valueAt(i)->dump(fd, args); 425 } 426 427 // dump record threads 428 for (size_t i = 0; i < mRecordThreads.size(); i++) { 429 mRecordThreads.valueAt(i)->dump(fd, args); 430 } 431 432 // dump orphan effect chains 433 if (mOrphanEffectChains.size() != 0) { 434 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 435 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 436 mOrphanEffectChains.valueAt(i)->dump(fd, args); 437 } 438 } 439 // dump all hardware devs 440 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 441 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 442 dev->dump(dev, fd); 443 } 444 445#ifdef TEE_SINK 446 // dump the serially shared record tee sink 447 if (mRecordTeeSource != 0) { 448 dumpTee(fd, mRecordTeeSource); 449 } 450#endif 451 452 if (locked) { 453 mLock.unlock(); 454 } 455 456 // append a copy of media.log here by forwarding fd to it, but don't attempt 457 // to lookup the service if it's not running, as it will block for a second 458 if (mLogMemoryDealer != 0) { 459 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 460 if (binder != 0) { 461 dprintf(fd, "\nmedia.log:\n"); 462 Vector<String16> args; 463 binder->dump(fd, args); 464 } 465 } 466 467 // check for optional arguments 468 bool unreachableMemory = false; 469 for (const auto &arg : args) { 470 if (arg == String16("--unreachable")) { 471 unreachableMemory = true; 472 } 473 } 474 475 if (unreachableMemory) { 476 dprintf(fd, "\nDumping unreachable memory:\n"); 477 // TODO - should limit be an argument parameter? 478 std::string s = GetUnreachableMemoryString(true /* contents */, 10000 /* limit */); 479 write(fd, s.c_str(), s.size()); 480 } 481 } 482 return NO_ERROR; 483} 484 485sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 486{ 487 Mutex::Autolock _cl(mClientLock); 488 // If pid is already in the mClients wp<> map, then use that entry 489 // (for which promote() is always != 0), otherwise create a new entry and Client. 490 sp<Client> client = mClients.valueFor(pid).promote(); 491 if (client == 0) { 492 client = new Client(this, pid); 493 mClients.add(pid, client); 494 } 495 496 return client; 497} 498 499sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 500{ 501 // If there is no memory allocated for logs, return a dummy writer that does nothing 502 if (mLogMemoryDealer == 0) { 503 return new NBLog::Writer(); 504 } 505 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 506 // Similarly if we can't contact the media.log service, also return a dummy writer 507 if (binder == 0) { 508 return new NBLog::Writer(); 509 } 510 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 511 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 512 // If allocation fails, consult the vector of previously unregistered writers 513 // and garbage-collect one or more them until an allocation succeeds 514 if (shared == 0) { 515 Mutex::Autolock _l(mUnregisteredWritersLock); 516 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 517 { 518 // Pick the oldest stale writer to garbage-collect 519 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 520 mUnregisteredWriters.removeAt(0); 521 mediaLogService->unregisterWriter(iMemory); 522 // Now the media.log remote reference to IMemory is gone. When our last local 523 // reference to IMemory also drops to zero at end of this block, 524 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 525 } 526 // Re-attempt the allocation 527 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 528 if (shared != 0) { 529 goto success; 530 } 531 } 532 // Even after garbage-collecting all old writers, there is still not enough memory, 533 // so return a dummy writer 534 return new NBLog::Writer(); 535 } 536success: 537 mediaLogService->registerWriter(shared, size, name); 538 return new NBLog::Writer(size, shared); 539} 540 541void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 542{ 543 if (writer == 0) { 544 return; 545 } 546 sp<IMemory> iMemory(writer->getIMemory()); 547 if (iMemory == 0) { 548 return; 549 } 550 // Rather than removing the writer immediately, append it to a queue of old writers to 551 // be garbage-collected later. This allows us to continue to view old logs for a while. 552 Mutex::Autolock _l(mUnregisteredWritersLock); 553 mUnregisteredWriters.push(writer); 554} 555 556// IAudioFlinger interface 557 558 559sp<IAudioTrack> AudioFlinger::createTrack( 560 audio_stream_type_t streamType, 561 uint32_t sampleRate, 562 audio_format_t format, 563 audio_channel_mask_t channelMask, 564 size_t *frameCount, 565 IAudioFlinger::track_flags_t *flags, 566 const sp<IMemory>& sharedBuffer, 567 audio_io_handle_t output, 568 pid_t tid, 569 audio_session_t *sessionId, 570 int clientUid, 571 status_t *status) 572{ 573 sp<PlaybackThread::Track> track; 574 sp<TrackHandle> trackHandle; 575 sp<Client> client; 576 status_t lStatus; 577 audio_session_t lSessionId; 578 579 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 580 // but if someone uses binder directly they could bypass that and cause us to crash 581 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 582 ALOGE("createTrack() invalid stream type %d", streamType); 583 lStatus = BAD_VALUE; 584 goto Exit; 585 } 586 587 // further sample rate checks are performed by createTrack_l() depending on the thread type 588 if (sampleRate == 0) { 589 ALOGE("createTrack() invalid sample rate %u", sampleRate); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 // further channel mask checks are performed by createTrack_l() depending on the thread type 595 if (!audio_is_output_channel(channelMask)) { 596 ALOGE("createTrack() invalid channel mask %#x", channelMask); 597 lStatus = BAD_VALUE; 598 goto Exit; 599 } 600 601 // further format checks are performed by createTrack_l() depending on the thread type 602 if (!audio_is_valid_format(format)) { 603 ALOGE("createTrack() invalid format %#x", format); 604 lStatus = BAD_VALUE; 605 goto Exit; 606 } 607 608 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 609 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 610 lStatus = BAD_VALUE; 611 goto Exit; 612 } 613 614 { 615 Mutex::Autolock _l(mLock); 616 PlaybackThread *thread = checkPlaybackThread_l(output); 617 if (thread == NULL) { 618 ALOGE("no playback thread found for output handle %d", output); 619 lStatus = BAD_VALUE; 620 goto Exit; 621 } 622 623 pid_t pid = IPCThreadState::self()->getCallingPid(); 624 client = registerPid(pid); 625 626 PlaybackThread *effectThread = NULL; 627 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 628 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 629 ALOGE("createTrack() invalid session ID %d", *sessionId); 630 lStatus = BAD_VALUE; 631 goto Exit; 632 } 633 lSessionId = *sessionId; 634 // check if an effect chain with the same session ID is present on another 635 // output thread and move it here. 636 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 637 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 638 if (mPlaybackThreads.keyAt(i) != output) { 639 uint32_t sessions = t->hasAudioSession(lSessionId); 640 if (sessions & PlaybackThread::EFFECT_SESSION) { 641 effectThread = t.get(); 642 break; 643 } 644 } 645 } 646 } else { 647 // if no audio session id is provided, create one here 648 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 649 if (sessionId != NULL) { 650 *sessionId = lSessionId; 651 } 652 } 653 ALOGV("createTrack() lSessionId: %d", lSessionId); 654 655 track = thread->createTrack_l(client, streamType, sampleRate, format, 656 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 657 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 658 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 659 660 // move effect chain to this output thread if an effect on same session was waiting 661 // for a track to be created 662 if (lStatus == NO_ERROR && effectThread != NULL) { 663 // no risk of deadlock because AudioFlinger::mLock is held 664 Mutex::Autolock _dl(thread->mLock); 665 Mutex::Autolock _sl(effectThread->mLock); 666 moveEffectChain_l(lSessionId, effectThread, thread, true); 667 } 668 669 // Look for sync events awaiting for a session to be used. 670 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 671 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 672 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 673 if (lStatus == NO_ERROR) { 674 (void) track->setSyncEvent(mPendingSyncEvents[i]); 675 } else { 676 mPendingSyncEvents[i]->cancel(); 677 } 678 mPendingSyncEvents.removeAt(i); 679 i--; 680 } 681 } 682 } 683 684 setAudioHwSyncForSession_l(thread, lSessionId); 685 } 686 687 if (lStatus != NO_ERROR) { 688 // remove local strong reference to Client before deleting the Track so that the 689 // Client destructor is called by the TrackBase destructor with mClientLock held 690 // Don't hold mClientLock when releasing the reference on the track as the 691 // destructor will acquire it. 692 { 693 Mutex::Autolock _cl(mClientLock); 694 client.clear(); 695 } 696 track.clear(); 697 goto Exit; 698 } 699 700 // return handle to client 701 trackHandle = new TrackHandle(track); 702 703Exit: 704 *status = lStatus; 705 return trackHandle; 706} 707 708uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 709{ 710 Mutex::Autolock _l(mLock); 711 ThreadBase *thread = checkThread_l(ioHandle); 712 if (thread == NULL) { 713 ALOGW("sampleRate() unknown thread %d", ioHandle); 714 return 0; 715 } 716 return thread->sampleRate(); 717} 718 719audio_format_t AudioFlinger::format(audio_io_handle_t output) const 720{ 721 Mutex::Autolock _l(mLock); 722 PlaybackThread *thread = checkPlaybackThread_l(output); 723 if (thread == NULL) { 724 ALOGW("format() unknown thread %d", output); 725 return AUDIO_FORMAT_INVALID; 726 } 727 return thread->format(); 728} 729 730size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 731{ 732 Mutex::Autolock _l(mLock); 733 ThreadBase *thread = checkThread_l(ioHandle); 734 if (thread == NULL) { 735 ALOGW("frameCount() unknown thread %d", ioHandle); 736 return 0; 737 } 738 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 739 // should examine all callers and fix them to handle smaller counts 740 return thread->frameCount(); 741} 742 743uint32_t AudioFlinger::latency(audio_io_handle_t output) const 744{ 745 Mutex::Autolock _l(mLock); 746 PlaybackThread *thread = checkPlaybackThread_l(output); 747 if (thread == NULL) { 748 ALOGW("latency(): no playback thread found for output handle %d", output); 749 return 0; 750 } 751 return thread->latency(); 752} 753 754status_t AudioFlinger::setMasterVolume(float value) 755{ 756 status_t ret = initCheck(); 757 if (ret != NO_ERROR) { 758 return ret; 759 } 760 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 Mutex::Autolock _l(mLock); 767 mMasterVolume = value; 768 769 // Set master volume in the HALs which support it. 770 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 771 AutoMutex lock(mHardwareLock); 772 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 773 774 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 775 if (dev->canSetMasterVolume()) { 776 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 777 } 778 mHardwareStatus = AUDIO_HW_IDLE; 779 } 780 781 // Now set the master volume in each playback thread. Playback threads 782 // assigned to HALs which do not have master volume support will apply 783 // master volume during the mix operation. Threads with HALs which do 784 // support master volume will simply ignore the setting. 785 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 786 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 787 continue; 788 } 789 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 790 } 791 792 return NO_ERROR; 793} 794 795status_t AudioFlinger::setMode(audio_mode_t mode) 796{ 797 status_t ret = initCheck(); 798 if (ret != NO_ERROR) { 799 return ret; 800 } 801 802 // check calling permissions 803 if (!settingsAllowed()) { 804 return PERMISSION_DENIED; 805 } 806 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 807 ALOGW("Illegal value: setMode(%d)", mode); 808 return BAD_VALUE; 809 } 810 811 { // scope for the lock 812 AutoMutex lock(mHardwareLock); 813 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 814 mHardwareStatus = AUDIO_HW_SET_MODE; 815 ret = dev->set_mode(dev, mode); 816 mHardwareStatus = AUDIO_HW_IDLE; 817 } 818 819 if (NO_ERROR == ret) { 820 Mutex::Autolock _l(mLock); 821 mMode = mode; 822 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 823 mPlaybackThreads.valueAt(i)->setMode(mode); 824 } 825 826 return ret; 827} 828 829status_t AudioFlinger::setMicMute(bool state) 830{ 831 status_t ret = initCheck(); 832 if (ret != NO_ERROR) { 833 return ret; 834 } 835 836 // check calling permissions 837 if (!settingsAllowed()) { 838 return PERMISSION_DENIED; 839 } 840 841 AutoMutex lock(mHardwareLock); 842 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 843 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 844 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 845 status_t result = dev->set_mic_mute(dev, state); 846 if (result != NO_ERROR) { 847 ret = result; 848 } 849 } 850 mHardwareStatus = AUDIO_HW_IDLE; 851 return ret; 852} 853 854bool AudioFlinger::getMicMute() const 855{ 856 status_t ret = initCheck(); 857 if (ret != NO_ERROR) { 858 return false; 859 } 860 bool mute = true; 861 bool state = AUDIO_MODE_INVALID; 862 AutoMutex lock(mHardwareLock); 863 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 864 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 865 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 866 status_t result = dev->get_mic_mute(dev, &state); 867 if (result == NO_ERROR) { 868 mute = mute && state; 869 } 870 } 871 mHardwareStatus = AUDIO_HW_IDLE; 872 873 return mute; 874} 875 876status_t AudioFlinger::setMasterMute(bool muted) 877{ 878 status_t ret = initCheck(); 879 if (ret != NO_ERROR) { 880 return ret; 881 } 882 883 // check calling permissions 884 if (!settingsAllowed()) { 885 return PERMISSION_DENIED; 886 } 887 888 Mutex::Autolock _l(mLock); 889 mMasterMute = muted; 890 891 // Set master mute in the HALs which support it. 892 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 893 AutoMutex lock(mHardwareLock); 894 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 895 896 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 897 if (dev->canSetMasterMute()) { 898 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 899 } 900 mHardwareStatus = AUDIO_HW_IDLE; 901 } 902 903 // Now set the master mute in each playback thread. Playback threads 904 // assigned to HALs which do not have master mute support will apply master 905 // mute during the mix operation. Threads with HALs which do support master 906 // mute will simply ignore the setting. 907 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 908 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 909 continue; 910 } 911 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 912 } 913 914 return NO_ERROR; 915} 916 917float AudioFlinger::masterVolume() const 918{ 919 Mutex::Autolock _l(mLock); 920 return masterVolume_l(); 921} 922 923bool AudioFlinger::masterMute() const 924{ 925 Mutex::Autolock _l(mLock); 926 return masterMute_l(); 927} 928 929float AudioFlinger::masterVolume_l() const 930{ 931 return mMasterVolume; 932} 933 934bool AudioFlinger::masterMute_l() const 935{ 936 return mMasterMute; 937} 938 939status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 940{ 941 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 942 ALOGW("setStreamVolume() invalid stream %d", stream); 943 return BAD_VALUE; 944 } 945 pid_t caller = IPCThreadState::self()->getCallingPid(); 946 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 947 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 948 return PERMISSION_DENIED; 949 } 950 951 return NO_ERROR; 952} 953 954status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 955 audio_io_handle_t output) 956{ 957 // check calling permissions 958 if (!settingsAllowed()) { 959 return PERMISSION_DENIED; 960 } 961 962 status_t status = checkStreamType(stream); 963 if (status != NO_ERROR) { 964 return status; 965 } 966 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 967 968 AutoMutex lock(mLock); 969 PlaybackThread *thread = NULL; 970 if (output != AUDIO_IO_HANDLE_NONE) { 971 thread = checkPlaybackThread_l(output); 972 if (thread == NULL) { 973 return BAD_VALUE; 974 } 975 } 976 977 mStreamTypes[stream].volume = value; 978 979 if (thread == NULL) { 980 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 981 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 982 } 983 } else { 984 thread->setStreamVolume(stream, value); 985 } 986 987 return NO_ERROR; 988} 989 990status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 991{ 992 // check calling permissions 993 if (!settingsAllowed()) { 994 return PERMISSION_DENIED; 995 } 996 997 status_t status = checkStreamType(stream); 998 if (status != NO_ERROR) { 999 return status; 1000 } 1001 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1002 1003 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1004 ALOGE("setStreamMute() invalid stream %d", stream); 1005 return BAD_VALUE; 1006 } 1007 1008 AutoMutex lock(mLock); 1009 mStreamTypes[stream].mute = muted; 1010 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 1011 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 1012 1013 return NO_ERROR; 1014} 1015 1016float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1017{ 1018 status_t status = checkStreamType(stream); 1019 if (status != NO_ERROR) { 1020 return 0.0f; 1021 } 1022 1023 AutoMutex lock(mLock); 1024 float volume; 1025 if (output != AUDIO_IO_HANDLE_NONE) { 1026 PlaybackThread *thread = checkPlaybackThread_l(output); 1027 if (thread == NULL) { 1028 return 0.0f; 1029 } 1030 volume = thread->streamVolume(stream); 1031 } else { 1032 volume = streamVolume_l(stream); 1033 } 1034 1035 return volume; 1036} 1037 1038bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1039{ 1040 status_t status = checkStreamType(stream); 1041 if (status != NO_ERROR) { 1042 return true; 1043 } 1044 1045 AutoMutex lock(mLock); 1046 return streamMute_l(stream); 1047} 1048 1049 1050void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1051{ 1052 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1053 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1054 } 1055} 1056 1057status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1058{ 1059 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1060 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1061 1062 // check calling permissions 1063 if (!settingsAllowed()) { 1064 return PERMISSION_DENIED; 1065 } 1066 1067 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1068 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1069 Mutex::Autolock _l(mLock); 1070 status_t final_result = NO_ERROR; 1071 { 1072 AutoMutex lock(mHardwareLock); 1073 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1074 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1075 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1076 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1077 final_result = result ?: final_result; 1078 } 1079 mHardwareStatus = AUDIO_HW_IDLE; 1080 } 1081 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1082 AudioParameter param = AudioParameter(keyValuePairs); 1083 String8 value; 1084 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1085 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1086 if (mBtNrecIsOff != btNrecIsOff) { 1087 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1088 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1089 audio_devices_t device = thread->inDevice(); 1090 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1091 // collect all of the thread's session IDs 1092 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1093 // suspend effects associated with those session IDs 1094 for (size_t j = 0; j < ids.size(); ++j) { 1095 audio_session_t sessionId = ids.keyAt(j); 1096 thread->setEffectSuspended(FX_IID_AEC, 1097 suspend, 1098 sessionId); 1099 thread->setEffectSuspended(FX_IID_NS, 1100 suspend, 1101 sessionId); 1102 } 1103 } 1104 mBtNrecIsOff = btNrecIsOff; 1105 } 1106 } 1107 String8 screenState; 1108 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1109 bool isOff = screenState == "off"; 1110 if (isOff != (AudioFlinger::mScreenState & 1)) { 1111 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1112 } 1113 } 1114 return final_result; 1115 } 1116 1117 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1118 // and the thread is exited once the lock is released 1119 sp<ThreadBase> thread; 1120 { 1121 Mutex::Autolock _l(mLock); 1122 thread = checkPlaybackThread_l(ioHandle); 1123 if (thread == 0) { 1124 thread = checkRecordThread_l(ioHandle); 1125 } else if (thread == primaryPlaybackThread_l()) { 1126 // indicate output device change to all input threads for pre processing 1127 AudioParameter param = AudioParameter(keyValuePairs); 1128 int value; 1129 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1130 (value != 0)) { 1131 broacastParametersToRecordThreads_l(keyValuePairs); 1132 } 1133 } 1134 } 1135 if (thread != 0) { 1136 return thread->setParameters(keyValuePairs); 1137 } 1138 return BAD_VALUE; 1139} 1140 1141String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1142{ 1143 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1144 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1145 1146 Mutex::Autolock _l(mLock); 1147 1148 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1149 String8 out_s8; 1150 1151 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1152 char *s; 1153 { 1154 AutoMutex lock(mHardwareLock); 1155 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1156 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1157 s = dev->get_parameters(dev, keys.string()); 1158 mHardwareStatus = AUDIO_HW_IDLE; 1159 } 1160 out_s8 += String8(s ? s : ""); 1161 free(s); 1162 } 1163 return out_s8; 1164 } 1165 1166 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1167 if (playbackThread != NULL) { 1168 return playbackThread->getParameters(keys); 1169 } 1170 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1171 if (recordThread != NULL) { 1172 return recordThread->getParameters(keys); 1173 } 1174 return String8(""); 1175} 1176 1177size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1178 audio_channel_mask_t channelMask) const 1179{ 1180 status_t ret = initCheck(); 1181 if (ret != NO_ERROR) { 1182 return 0; 1183 } 1184 if ((sampleRate == 0) || 1185 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1186 !audio_is_input_channel(channelMask)) { 1187 return 0; 1188 } 1189 1190 AutoMutex lock(mHardwareLock); 1191 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1192 audio_config_t config, proposed; 1193 memset(&proposed, 0, sizeof(proposed)); 1194 proposed.sample_rate = sampleRate; 1195 proposed.channel_mask = channelMask; 1196 proposed.format = format; 1197 1198 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1199 size_t frames; 1200 for (;;) { 1201 // Note: config is currently a const parameter for get_input_buffer_size() 1202 // but we use a copy from proposed in case config changes from the call. 1203 config = proposed; 1204 frames = dev->get_input_buffer_size(dev, &config); 1205 if (frames != 0) { 1206 break; // hal success, config is the result 1207 } 1208 // change one parameter of the configuration each iteration to a more "common" value 1209 // to see if the device will support it. 1210 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1211 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1212 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1213 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1214 } else { 1215 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1216 "format %#x, channelMask 0x%X", 1217 sampleRate, format, channelMask); 1218 break; // retries failed, break out of loop with frames == 0. 1219 } 1220 } 1221 mHardwareStatus = AUDIO_HW_IDLE; 1222 if (frames > 0 && config.sample_rate != sampleRate) { 1223 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1224 } 1225 return frames; // may be converted to bytes at the Java level. 1226} 1227 1228uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1229{ 1230 Mutex::Autolock _l(mLock); 1231 1232 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1233 if (recordThread != NULL) { 1234 return recordThread->getInputFramesLost(); 1235 } 1236 return 0; 1237} 1238 1239status_t AudioFlinger::setVoiceVolume(float value) 1240{ 1241 status_t ret = initCheck(); 1242 if (ret != NO_ERROR) { 1243 return ret; 1244 } 1245 1246 // check calling permissions 1247 if (!settingsAllowed()) { 1248 return PERMISSION_DENIED; 1249 } 1250 1251 AutoMutex lock(mHardwareLock); 1252 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1253 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1254 ret = dev->set_voice_volume(dev, value); 1255 mHardwareStatus = AUDIO_HW_IDLE; 1256 1257 return ret; 1258} 1259 1260status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1261 audio_io_handle_t output) const 1262{ 1263 Mutex::Autolock _l(mLock); 1264 1265 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1266 if (playbackThread != NULL) { 1267 return playbackThread->getRenderPosition(halFrames, dspFrames); 1268 } 1269 1270 return BAD_VALUE; 1271} 1272 1273void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1274{ 1275 Mutex::Autolock _l(mLock); 1276 if (client == 0) { 1277 return; 1278 } 1279 pid_t pid = IPCThreadState::self()->getCallingPid(); 1280 { 1281 Mutex::Autolock _cl(mClientLock); 1282 if (mNotificationClients.indexOfKey(pid) < 0) { 1283 sp<NotificationClient> notificationClient = new NotificationClient(this, 1284 client, 1285 pid); 1286 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1287 1288 mNotificationClients.add(pid, notificationClient); 1289 1290 sp<IBinder> binder = IInterface::asBinder(client); 1291 binder->linkToDeath(notificationClient); 1292 } 1293 } 1294 1295 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1296 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1297 // the config change is always sent from playback or record threads to avoid deadlock 1298 // with AudioSystem::gLock 1299 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1300 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1301 } 1302 1303 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1304 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1305 } 1306} 1307 1308void AudioFlinger::removeNotificationClient(pid_t pid) 1309{ 1310 Mutex::Autolock _l(mLock); 1311 { 1312 Mutex::Autolock _cl(mClientLock); 1313 mNotificationClients.removeItem(pid); 1314 } 1315 1316 ALOGV("%d died, releasing its sessions", pid); 1317 size_t num = mAudioSessionRefs.size(); 1318 bool removed = false; 1319 for (size_t i = 0; i< num; ) { 1320 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1321 ALOGV(" pid %d @ %zu", ref->mPid, i); 1322 if (ref->mPid == pid) { 1323 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1324 mAudioSessionRefs.removeAt(i); 1325 delete ref; 1326 removed = true; 1327 num--; 1328 } else { 1329 i++; 1330 } 1331 } 1332 if (removed) { 1333 purgeStaleEffects_l(); 1334 } 1335} 1336 1337void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1338 const sp<AudioIoDescriptor>& ioDesc, 1339 pid_t pid) 1340{ 1341 Mutex::Autolock _l(mClientLock); 1342 size_t size = mNotificationClients.size(); 1343 for (size_t i = 0; i < size; i++) { 1344 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1345 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1346 } 1347 } 1348} 1349 1350// removeClient_l() must be called with AudioFlinger::mClientLock held 1351void AudioFlinger::removeClient_l(pid_t pid) 1352{ 1353 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1354 IPCThreadState::self()->getCallingPid()); 1355 mClients.removeItem(pid); 1356} 1357 1358// getEffectThread_l() must be called with AudioFlinger::mLock held 1359sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1360 int EffectId) 1361{ 1362 sp<PlaybackThread> thread; 1363 1364 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1365 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1366 ALOG_ASSERT(thread == 0); 1367 thread = mPlaybackThreads.valueAt(i); 1368 } 1369 } 1370 1371 return thread; 1372} 1373 1374 1375 1376// ---------------------------------------------------------------------------- 1377 1378AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1379 : RefBase(), 1380 mAudioFlinger(audioFlinger), 1381 mPid(pid) 1382{ 1383 size_t heapSize = kClientSharedHeapSizeBytes; 1384 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1385 // invalidated tracks 1386 if (!audioFlinger->isLowRamDevice()) { 1387 heapSize *= kClientSharedHeapSizeMultiplier; 1388 } 1389 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1390} 1391 1392// Client destructor must be called with AudioFlinger::mClientLock held 1393AudioFlinger::Client::~Client() 1394{ 1395 mAudioFlinger->removeClient_l(mPid); 1396} 1397 1398sp<MemoryDealer> AudioFlinger::Client::heap() const 1399{ 1400 return mMemoryDealer; 1401} 1402 1403// ---------------------------------------------------------------------------- 1404 1405AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1406 const sp<IAudioFlingerClient>& client, 1407 pid_t pid) 1408 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1409{ 1410} 1411 1412AudioFlinger::NotificationClient::~NotificationClient() 1413{ 1414} 1415 1416void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1417{ 1418 sp<NotificationClient> keep(this); 1419 mAudioFlinger->removeNotificationClient(mPid); 1420} 1421 1422 1423// ---------------------------------------------------------------------------- 1424 1425sp<IAudioRecord> AudioFlinger::openRecord( 1426 audio_io_handle_t input, 1427 uint32_t sampleRate, 1428 audio_format_t format, 1429 audio_channel_mask_t channelMask, 1430 const String16& opPackageName, 1431 size_t *frameCount, 1432 IAudioFlinger::track_flags_t *flags, 1433 pid_t tid, 1434 int clientUid, 1435 audio_session_t *sessionId, 1436 size_t *notificationFrames, 1437 sp<IMemory>& cblk, 1438 sp<IMemory>& buffers, 1439 status_t *status) 1440{ 1441 sp<RecordThread::RecordTrack> recordTrack; 1442 sp<RecordHandle> recordHandle; 1443 sp<Client> client; 1444 status_t lStatus; 1445 audio_session_t lSessionId; 1446 1447 cblk.clear(); 1448 buffers.clear(); 1449 1450 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1451 if (!isTrustedCallingUid(callingUid)) { 1452 ALOGW_IF((uid_t)clientUid != callingUid, 1453 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1454 clientUid = callingUid; 1455 } 1456 1457 // check calling permissions 1458 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1459 ALOGE("openRecord() permission denied: recording not allowed"); 1460 lStatus = PERMISSION_DENIED; 1461 goto Exit; 1462 } 1463 1464 // further sample rate checks are performed by createRecordTrack_l() 1465 if (sampleRate == 0) { 1466 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1467 lStatus = BAD_VALUE; 1468 goto Exit; 1469 } 1470 1471 // we don't yet support anything other than linear PCM 1472 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1473 ALOGE("openRecord() invalid format %#x", format); 1474 lStatus = BAD_VALUE; 1475 goto Exit; 1476 } 1477 1478 // further channel mask checks are performed by createRecordTrack_l() 1479 if (!audio_is_input_channel(channelMask)) { 1480 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1481 lStatus = BAD_VALUE; 1482 goto Exit; 1483 } 1484 1485 { 1486 Mutex::Autolock _l(mLock); 1487 RecordThread *thread = checkRecordThread_l(input); 1488 if (thread == NULL) { 1489 ALOGE("openRecord() checkRecordThread_l failed"); 1490 lStatus = BAD_VALUE; 1491 goto Exit; 1492 } 1493 1494 pid_t pid = IPCThreadState::self()->getCallingPid(); 1495 client = registerPid(pid); 1496 1497 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1498 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1499 lStatus = BAD_VALUE; 1500 goto Exit; 1501 } 1502 lSessionId = *sessionId; 1503 } else { 1504 // if no audio session id is provided, create one here 1505 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1506 if (sessionId != NULL) { 1507 *sessionId = lSessionId; 1508 } 1509 } 1510 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1511 1512 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1513 frameCount, lSessionId, notificationFrames, 1514 clientUid, flags, tid, &lStatus); 1515 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1516 1517 if (lStatus == NO_ERROR) { 1518 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1519 // session and move it to this thread. 1520 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1521 if (chain != 0) { 1522 Mutex::Autolock _l(thread->mLock); 1523 thread->addEffectChain_l(chain); 1524 } 1525 } 1526 } 1527 1528 if (lStatus != NO_ERROR) { 1529 // remove local strong reference to Client before deleting the RecordTrack so that the 1530 // Client destructor is called by the TrackBase destructor with mClientLock held 1531 // Don't hold mClientLock when releasing the reference on the track as the 1532 // destructor will acquire it. 1533 { 1534 Mutex::Autolock _cl(mClientLock); 1535 client.clear(); 1536 } 1537 recordTrack.clear(); 1538 goto Exit; 1539 } 1540 1541 cblk = recordTrack->getCblk(); 1542 buffers = recordTrack->getBuffers(); 1543 1544 // return handle to client 1545 recordHandle = new RecordHandle(recordTrack); 1546 1547Exit: 1548 *status = lStatus; 1549 return recordHandle; 1550} 1551 1552 1553 1554// ---------------------------------------------------------------------------- 1555 1556audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1557{ 1558 if (name == NULL) { 1559 return AUDIO_MODULE_HANDLE_NONE; 1560 } 1561 if (!settingsAllowed()) { 1562 return AUDIO_MODULE_HANDLE_NONE; 1563 } 1564 Mutex::Autolock _l(mLock); 1565 return loadHwModule_l(name); 1566} 1567 1568// loadHwModule_l() must be called with AudioFlinger::mLock held 1569audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1570{ 1571 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1572 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1573 ALOGW("loadHwModule() module %s already loaded", name); 1574 return mAudioHwDevs.keyAt(i); 1575 } 1576 } 1577 1578 audio_hw_device_t *dev; 1579 1580 int rc = load_audio_interface(name, &dev); 1581 if (rc) { 1582 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1583 return AUDIO_MODULE_HANDLE_NONE; 1584 } 1585 1586 mHardwareStatus = AUDIO_HW_INIT; 1587 rc = dev->init_check(dev); 1588 mHardwareStatus = AUDIO_HW_IDLE; 1589 if (rc) { 1590 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1591 return AUDIO_MODULE_HANDLE_NONE; 1592 } 1593 1594 // Check and cache this HAL's level of support for master mute and master 1595 // volume. If this is the first HAL opened, and it supports the get 1596 // methods, use the initial values provided by the HAL as the current 1597 // master mute and volume settings. 1598 1599 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1600 { // scope for auto-lock pattern 1601 AutoMutex lock(mHardwareLock); 1602 1603 if (0 == mAudioHwDevs.size()) { 1604 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1605 if (NULL != dev->get_master_volume) { 1606 float mv; 1607 if (OK == dev->get_master_volume(dev, &mv)) { 1608 mMasterVolume = mv; 1609 } 1610 } 1611 1612 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1613 if (NULL != dev->get_master_mute) { 1614 bool mm; 1615 if (OK == dev->get_master_mute(dev, &mm)) { 1616 mMasterMute = mm; 1617 } 1618 } 1619 } 1620 1621 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1622 if ((NULL != dev->set_master_volume) && 1623 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1624 flags = static_cast<AudioHwDevice::Flags>(flags | 1625 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1626 } 1627 1628 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1629 if ((NULL != dev->set_master_mute) && 1630 (OK == dev->set_master_mute(dev, mMasterMute))) { 1631 flags = static_cast<AudioHwDevice::Flags>(flags | 1632 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1633 } 1634 1635 mHardwareStatus = AUDIO_HW_IDLE; 1636 } 1637 1638 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1639 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1640 1641 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1642 name, dev->common.module->name, dev->common.module->id, handle); 1643 1644 return handle; 1645 1646} 1647 1648// ---------------------------------------------------------------------------- 1649 1650uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1651{ 1652 Mutex::Autolock _l(mLock); 1653 PlaybackThread *thread = primaryPlaybackThread_l(); 1654 return thread != NULL ? thread->sampleRate() : 0; 1655} 1656 1657size_t AudioFlinger::getPrimaryOutputFrameCount() 1658{ 1659 Mutex::Autolock _l(mLock); 1660 PlaybackThread *thread = primaryPlaybackThread_l(); 1661 return thread != NULL ? thread->frameCountHAL() : 0; 1662} 1663 1664// ---------------------------------------------------------------------------- 1665 1666status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1667{ 1668 uid_t uid = IPCThreadState::self()->getCallingUid(); 1669 if (uid != AID_SYSTEM) { 1670 return PERMISSION_DENIED; 1671 } 1672 Mutex::Autolock _l(mLock); 1673 if (mIsDeviceTypeKnown) { 1674 return INVALID_OPERATION; 1675 } 1676 mIsLowRamDevice = isLowRamDevice; 1677 mIsDeviceTypeKnown = true; 1678 return NO_ERROR; 1679} 1680 1681audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1682{ 1683 Mutex::Autolock _l(mLock); 1684 1685 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1686 if (index >= 0) { 1687 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1688 mHwAvSyncIds.valueAt(index), sessionId); 1689 return mHwAvSyncIds.valueAt(index); 1690 } 1691 1692 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1693 if (dev == NULL) { 1694 return AUDIO_HW_SYNC_INVALID; 1695 } 1696 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1697 AudioParameter param = AudioParameter(String8(reply)); 1698 free(reply); 1699 1700 int value; 1701 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1702 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1703 return AUDIO_HW_SYNC_INVALID; 1704 } 1705 1706 // allow only one session for a given HW A/V sync ID. 1707 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1708 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1709 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1710 value, mHwAvSyncIds.keyAt(i)); 1711 mHwAvSyncIds.removeItemsAt(i); 1712 break; 1713 } 1714 } 1715 1716 mHwAvSyncIds.add(sessionId, value); 1717 1718 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1719 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1720 uint32_t sessions = thread->hasAudioSession(sessionId); 1721 if (sessions & PlaybackThread::TRACK_SESSION) { 1722 AudioParameter param = AudioParameter(); 1723 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1724 thread->setParameters(param.toString()); 1725 break; 1726 } 1727 } 1728 1729 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1730 return (audio_hw_sync_t)value; 1731} 1732 1733status_t AudioFlinger::systemReady() 1734{ 1735 Mutex::Autolock _l(mLock); 1736 ALOGI("%s", __FUNCTION__); 1737 if (mSystemReady) { 1738 ALOGW("%s called twice", __FUNCTION__); 1739 return NO_ERROR; 1740 } 1741 mSystemReady = true; 1742 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1743 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1744 thread->systemReady(); 1745 } 1746 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1747 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1748 thread->systemReady(); 1749 } 1750 return NO_ERROR; 1751} 1752 1753// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1754void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1755{ 1756 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1757 if (index >= 0) { 1758 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1759 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1760 AudioParameter param = AudioParameter(); 1761 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1762 thread->setParameters(param.toString()); 1763 } 1764} 1765 1766 1767// ---------------------------------------------------------------------------- 1768 1769 1770sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1771 audio_io_handle_t *output, 1772 audio_config_t *config, 1773 audio_devices_t devices, 1774 const String8& address, 1775 audio_output_flags_t flags) 1776{ 1777 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1778 if (outHwDev == NULL) { 1779 return 0; 1780 } 1781 1782 if (*output == AUDIO_IO_HANDLE_NONE) { 1783 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1784 } else { 1785 // Audio Policy does not currently request a specific output handle. 1786 // If this is ever needed, see openInput_l() for example code. 1787 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1788 return 0; 1789 } 1790 1791 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1792 1793 // FOR TESTING ONLY: 1794 // This if statement allows overriding the audio policy settings 1795 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1796 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1797 // Check only for Normal Mixing mode 1798 if (kEnableExtendedPrecision) { 1799 // Specify format (uncomment one below to choose) 1800 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1801 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1802 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1803 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1804 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1805 } 1806 if (kEnableExtendedChannels) { 1807 // Specify channel mask (uncomment one below to choose) 1808 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1809 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1810 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1811 } 1812 } 1813 1814 AudioStreamOut *outputStream = NULL; 1815 status_t status = outHwDev->openOutputStream( 1816 &outputStream, 1817 *output, 1818 devices, 1819 flags, 1820 config, 1821 address.string()); 1822 1823 mHardwareStatus = AUDIO_HW_IDLE; 1824 1825 if (status == NO_ERROR) { 1826 1827 PlaybackThread *thread; 1828 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1829 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady, 1830 config->offload_info.bit_rate); 1831 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1832 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1833 || !isValidPcmSinkFormat(config->format) 1834 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1835 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1836 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1837 } else { 1838 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1839 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1840 } 1841 mPlaybackThreads.add(*output, thread); 1842 return thread; 1843 } 1844 1845 return 0; 1846} 1847 1848status_t AudioFlinger::openOutput(audio_module_handle_t module, 1849 audio_io_handle_t *output, 1850 audio_config_t *config, 1851 audio_devices_t *devices, 1852 const String8& address, 1853 uint32_t *latencyMs, 1854 audio_output_flags_t flags) 1855{ 1856 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1857 module, 1858 (devices != NULL) ? *devices : 0, 1859 config->sample_rate, 1860 config->format, 1861 config->channel_mask, 1862 flags); 1863 1864 if (*devices == AUDIO_DEVICE_NONE) { 1865 return BAD_VALUE; 1866 } 1867 1868 Mutex::Autolock _l(mLock); 1869 1870 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1871 if (thread != 0) { 1872 *latencyMs = thread->latency(); 1873 1874 // notify client processes of the new output creation 1875 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1876 1877 // the first primary output opened designates the primary hw device 1878 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1879 ALOGI("Using module %d has the primary audio interface", module); 1880 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1881 1882 AutoMutex lock(mHardwareLock); 1883 mHardwareStatus = AUDIO_HW_SET_MODE; 1884 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1885 mHardwareStatus = AUDIO_HW_IDLE; 1886 } 1887 return NO_ERROR; 1888 } 1889 1890 return NO_INIT; 1891} 1892 1893audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1894 audio_io_handle_t output2) 1895{ 1896 Mutex::Autolock _l(mLock); 1897 MixerThread *thread1 = checkMixerThread_l(output1); 1898 MixerThread *thread2 = checkMixerThread_l(output2); 1899 1900 if (thread1 == NULL || thread2 == NULL) { 1901 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1902 output2); 1903 return AUDIO_IO_HANDLE_NONE; 1904 } 1905 1906 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1907 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1908 thread->addOutputTrack(thread2); 1909 mPlaybackThreads.add(id, thread); 1910 // notify client processes of the new output creation 1911 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1912 return id; 1913} 1914 1915status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1916{ 1917 return closeOutput_nonvirtual(output); 1918} 1919 1920status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1921{ 1922 // keep strong reference on the playback thread so that 1923 // it is not destroyed while exit() is executed 1924 sp<PlaybackThread> thread; 1925 { 1926 Mutex::Autolock _l(mLock); 1927 thread = checkPlaybackThread_l(output); 1928 if (thread == NULL) { 1929 return BAD_VALUE; 1930 } 1931 1932 ALOGV("closeOutput() %d", output); 1933 1934 if (thread->type() == ThreadBase::MIXER) { 1935 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1936 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1937 DuplicatingThread *dupThread = 1938 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1939 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1940 } 1941 } 1942 } 1943 1944 1945 mPlaybackThreads.removeItem(output); 1946 // save all effects to the default thread 1947 if (mPlaybackThreads.size()) { 1948 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1949 if (dstThread != NULL) { 1950 // audioflinger lock is held here so the acquisition order of thread locks does not 1951 // matter 1952 Mutex::Autolock _dl(dstThread->mLock); 1953 Mutex::Autolock _sl(thread->mLock); 1954 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1955 for (size_t i = 0; i < effectChains.size(); i ++) { 1956 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1957 } 1958 } 1959 } 1960 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1961 ioDesc->mIoHandle = output; 1962 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1963 } 1964 thread->exit(); 1965 // The thread entity (active unit of execution) is no longer running here, 1966 // but the ThreadBase container still exists. 1967 1968 if (!thread->isDuplicating()) { 1969 closeOutputFinish(thread); 1970 } 1971 1972 return NO_ERROR; 1973} 1974 1975void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1976{ 1977 AudioStreamOut *out = thread->clearOutput(); 1978 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1979 // from now on thread->mOutput is NULL 1980 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1981 delete out; 1982} 1983 1984void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1985{ 1986 mPlaybackThreads.removeItem(thread->mId); 1987 thread->exit(); 1988 closeOutputFinish(thread); 1989} 1990 1991status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1992{ 1993 Mutex::Autolock _l(mLock); 1994 PlaybackThread *thread = checkPlaybackThread_l(output); 1995 1996 if (thread == NULL) { 1997 return BAD_VALUE; 1998 } 1999 2000 ALOGV("suspendOutput() %d", output); 2001 thread->suspend(); 2002 2003 return NO_ERROR; 2004} 2005 2006status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2007{ 2008 Mutex::Autolock _l(mLock); 2009 PlaybackThread *thread = checkPlaybackThread_l(output); 2010 2011 if (thread == NULL) { 2012 return BAD_VALUE; 2013 } 2014 2015 ALOGV("restoreOutput() %d", output); 2016 2017 thread->restore(); 2018 2019 return NO_ERROR; 2020} 2021 2022status_t AudioFlinger::openInput(audio_module_handle_t module, 2023 audio_io_handle_t *input, 2024 audio_config_t *config, 2025 audio_devices_t *devices, 2026 const String8& address, 2027 audio_source_t source, 2028 audio_input_flags_t flags) 2029{ 2030 Mutex::Autolock _l(mLock); 2031 2032 if (*devices == AUDIO_DEVICE_NONE) { 2033 return BAD_VALUE; 2034 } 2035 2036 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2037 2038 if (thread != 0) { 2039 // notify client processes of the new input creation 2040 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2041 return NO_ERROR; 2042 } 2043 return NO_INIT; 2044} 2045 2046sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2047 audio_io_handle_t *input, 2048 audio_config_t *config, 2049 audio_devices_t devices, 2050 const String8& address, 2051 audio_source_t source, 2052 audio_input_flags_t flags) 2053{ 2054 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2055 if (inHwDev == NULL) { 2056 *input = AUDIO_IO_HANDLE_NONE; 2057 return 0; 2058 } 2059 2060 // Audio Policy can request a specific handle for hardware hotword. 2061 // The goal here is not to re-open an already opened input. 2062 // It is to use a pre-assigned I/O handle. 2063 if (*input == AUDIO_IO_HANDLE_NONE) { 2064 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2065 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2066 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2067 return 0; 2068 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2069 // This should not happen in a transient state with current design. 2070 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2071 return 0; 2072 } 2073 2074 audio_config_t halconfig = *config; 2075 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2076 audio_stream_in_t *inStream = NULL; 2077 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2078 &inStream, flags, address.string(), source); 2079 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2080 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2081 inStream, 2082 halconfig.sample_rate, 2083 halconfig.format, 2084 halconfig.channel_mask, 2085 flags, 2086 status, address.string()); 2087 2088 // If the input could not be opened with the requested parameters and we can handle the 2089 // conversion internally, try to open again with the proposed parameters. 2090 if (status == BAD_VALUE && 2091 audio_is_linear_pcm(config->format) && 2092 audio_is_linear_pcm(halconfig.format) && 2093 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2094 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2095 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2096 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2097 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2098 inStream = NULL; 2099 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2100 &inStream, flags, address.string(), source); 2101 // FIXME log this new status; HAL should not propose any further changes 2102 } 2103 2104 if (status == NO_ERROR && inStream != NULL) { 2105 2106#ifdef TEE_SINK 2107 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2108 // or (re-)create if current Pipe is idle and does not match the new format 2109 sp<NBAIO_Sink> teeSink; 2110 enum { 2111 TEE_SINK_NO, // don't copy input 2112 TEE_SINK_NEW, // copy input using a new pipe 2113 TEE_SINK_OLD, // copy input using an existing pipe 2114 } kind; 2115 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2116 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2117 if (!mTeeSinkInputEnabled) { 2118 kind = TEE_SINK_NO; 2119 } else if (!Format_isValid(format)) { 2120 kind = TEE_SINK_NO; 2121 } else if (mRecordTeeSink == 0) { 2122 kind = TEE_SINK_NEW; 2123 } else if (mRecordTeeSink->getStrongCount() != 1) { 2124 kind = TEE_SINK_NO; 2125 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2126 kind = TEE_SINK_OLD; 2127 } else { 2128 kind = TEE_SINK_NEW; 2129 } 2130 switch (kind) { 2131 case TEE_SINK_NEW: { 2132 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2133 size_t numCounterOffers = 0; 2134 const NBAIO_Format offers[1] = {format}; 2135 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2136 ALOG_ASSERT(index == 0); 2137 PipeReader *pipeReader = new PipeReader(*pipe); 2138 numCounterOffers = 0; 2139 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2140 ALOG_ASSERT(index == 0); 2141 mRecordTeeSink = pipe; 2142 mRecordTeeSource = pipeReader; 2143 teeSink = pipe; 2144 } 2145 break; 2146 case TEE_SINK_OLD: 2147 teeSink = mRecordTeeSink; 2148 break; 2149 case TEE_SINK_NO: 2150 default: 2151 break; 2152 } 2153#endif 2154 2155 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2156 2157 // Start record thread 2158 // RecordThread requires both input and output device indication to forward to audio 2159 // pre processing modules 2160 sp<RecordThread> thread = new RecordThread(this, 2161 inputStream, 2162 *input, 2163 primaryOutputDevice_l(), 2164 devices, 2165 mSystemReady 2166#ifdef TEE_SINK 2167 , teeSink 2168#endif 2169 ); 2170 mRecordThreads.add(*input, thread); 2171 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2172 return thread; 2173 } 2174 2175 *input = AUDIO_IO_HANDLE_NONE; 2176 return 0; 2177} 2178 2179status_t AudioFlinger::closeInput(audio_io_handle_t input) 2180{ 2181 return closeInput_nonvirtual(input); 2182} 2183 2184status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2185{ 2186 // keep strong reference on the record thread so that 2187 // it is not destroyed while exit() is executed 2188 sp<RecordThread> thread; 2189 { 2190 Mutex::Autolock _l(mLock); 2191 thread = checkRecordThread_l(input); 2192 if (thread == 0) { 2193 return BAD_VALUE; 2194 } 2195 2196 ALOGV("closeInput() %d", input); 2197 2198 // If we still have effect chains, it means that a client still holds a handle 2199 // on at least one effect. We must either move the chain to an existing thread with the 2200 // same session ID or put it aside in case a new record thread is opened for a 2201 // new capture on the same session 2202 sp<EffectChain> chain; 2203 { 2204 Mutex::Autolock _sl(thread->mLock); 2205 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2206 // Note: maximum one chain per record thread 2207 if (effectChains.size() != 0) { 2208 chain = effectChains[0]; 2209 } 2210 } 2211 if (chain != 0) { 2212 // first check if a record thread is already opened with a client on the same session. 2213 // This should only happen in case of overlap between one thread tear down and the 2214 // creation of its replacement 2215 size_t i; 2216 for (i = 0; i < mRecordThreads.size(); i++) { 2217 sp<RecordThread> t = mRecordThreads.valueAt(i); 2218 if (t == thread) { 2219 continue; 2220 } 2221 if (t->hasAudioSession(chain->sessionId()) != 0) { 2222 Mutex::Autolock _l(t->mLock); 2223 ALOGV("closeInput() found thread %d for effect session %d", 2224 t->id(), chain->sessionId()); 2225 t->addEffectChain_l(chain); 2226 break; 2227 } 2228 } 2229 // put the chain aside if we could not find a record thread with the same session id. 2230 if (i == mRecordThreads.size()) { 2231 putOrphanEffectChain_l(chain); 2232 } 2233 } 2234 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2235 ioDesc->mIoHandle = input; 2236 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2237 mRecordThreads.removeItem(input); 2238 } 2239 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2240 // we have a different lock for notification client 2241 closeInputFinish(thread); 2242 return NO_ERROR; 2243} 2244 2245void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2246{ 2247 thread->exit(); 2248 AudioStreamIn *in = thread->clearInput(); 2249 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2250 // from now on thread->mInput is NULL 2251 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2252 delete in; 2253} 2254 2255void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2256{ 2257 mRecordThreads.removeItem(thread->mId); 2258 closeInputFinish(thread); 2259} 2260 2261status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2262{ 2263 Mutex::Autolock _l(mLock); 2264 ALOGV("invalidateStream() stream %d", stream); 2265 2266 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2267 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2268 thread->invalidateTracks(stream); 2269 } 2270 2271 return NO_ERROR; 2272} 2273 2274 2275audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2276{ 2277 // This is a binder API, so a malicious client could pass in a bad parameter. 2278 // Check for that before calling the internal API nextUniqueId(). 2279 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2280 ALOGE("newAudioUniqueId invalid use %d", use); 2281 return AUDIO_UNIQUE_ID_ALLOCATE; 2282 } 2283 return nextUniqueId(use); 2284} 2285 2286void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2287{ 2288 Mutex::Autolock _l(mLock); 2289 pid_t caller = IPCThreadState::self()->getCallingPid(); 2290 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2291 if (pid != -1 && (caller == getpid_cached)) { 2292 caller = pid; 2293 } 2294 2295 { 2296 Mutex::Autolock _cl(mClientLock); 2297 // Ignore requests received from processes not known as notification client. The request 2298 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2299 // called from a different pid leaving a stale session reference. Also we don't know how 2300 // to clear this reference if the client process dies. 2301 if (mNotificationClients.indexOfKey(caller) < 0) { 2302 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2303 return; 2304 } 2305 } 2306 2307 size_t num = mAudioSessionRefs.size(); 2308 for (size_t i = 0; i< num; i++) { 2309 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2310 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2311 ref->mCnt++; 2312 ALOGV(" incremented refcount to %d", ref->mCnt); 2313 return; 2314 } 2315 } 2316 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2317 ALOGV(" added new entry for %d", audioSession); 2318} 2319 2320void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2321{ 2322 Mutex::Autolock _l(mLock); 2323 pid_t caller = IPCThreadState::self()->getCallingPid(); 2324 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2325 if (pid != -1 && (caller == getpid_cached)) { 2326 caller = pid; 2327 } 2328 size_t num = mAudioSessionRefs.size(); 2329 for (size_t i = 0; i< num; i++) { 2330 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2331 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2332 ref->mCnt--; 2333 ALOGV(" decremented refcount to %d", ref->mCnt); 2334 if (ref->mCnt == 0) { 2335 mAudioSessionRefs.removeAt(i); 2336 delete ref; 2337 purgeStaleEffects_l(); 2338 } 2339 return; 2340 } 2341 } 2342 // If the caller is mediaserver it is likely that the session being released was acquired 2343 // on behalf of a process not in notification clients and we ignore the warning. 2344 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2345} 2346 2347void AudioFlinger::purgeStaleEffects_l() { 2348 2349 ALOGV("purging stale effects"); 2350 2351 Vector< sp<EffectChain> > chains; 2352 2353 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2354 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2355 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2356 sp<EffectChain> ec = t->mEffectChains[j]; 2357 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2358 chains.push(ec); 2359 } 2360 } 2361 } 2362 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2363 sp<RecordThread> t = mRecordThreads.valueAt(i); 2364 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2365 sp<EffectChain> ec = t->mEffectChains[j]; 2366 chains.push(ec); 2367 } 2368 } 2369 2370 for (size_t i = 0; i < chains.size(); i++) { 2371 sp<EffectChain> ec = chains[i]; 2372 int sessionid = ec->sessionId(); 2373 sp<ThreadBase> t = ec->mThread.promote(); 2374 if (t == 0) { 2375 continue; 2376 } 2377 size_t numsessionrefs = mAudioSessionRefs.size(); 2378 bool found = false; 2379 for (size_t k = 0; k < numsessionrefs; k++) { 2380 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2381 if (ref->mSessionid == sessionid) { 2382 ALOGV(" session %d still exists for %d with %d refs", 2383 sessionid, ref->mPid, ref->mCnt); 2384 found = true; 2385 break; 2386 } 2387 } 2388 if (!found) { 2389 Mutex::Autolock _l(t->mLock); 2390 // remove all effects from the chain 2391 while (ec->mEffects.size()) { 2392 sp<EffectModule> effect = ec->mEffects[0]; 2393 effect->unPin(); 2394 t->removeEffect_l(effect); 2395 if (effect->purgeHandles()) { 2396 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2397 } 2398 AudioSystem::unregisterEffect(effect->id()); 2399 } 2400 } 2401 } 2402 return; 2403} 2404 2405// checkThread_l() must be called with AudioFlinger::mLock held 2406AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2407{ 2408 ThreadBase *thread = NULL; 2409 switch (audio_unique_id_get_use(ioHandle)) { 2410 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2411 thread = checkPlaybackThread_l(ioHandle); 2412 break; 2413 case AUDIO_UNIQUE_ID_USE_INPUT: 2414 thread = checkRecordThread_l(ioHandle); 2415 break; 2416 default: 2417 break; 2418 } 2419 return thread; 2420} 2421 2422// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2423AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2424{ 2425 return mPlaybackThreads.valueFor(output).get(); 2426} 2427 2428// checkMixerThread_l() must be called with AudioFlinger::mLock held 2429AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2430{ 2431 PlaybackThread *thread = checkPlaybackThread_l(output); 2432 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2433} 2434 2435// checkRecordThread_l() must be called with AudioFlinger::mLock held 2436AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2437{ 2438 return mRecordThreads.valueFor(input).get(); 2439} 2440 2441audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2442{ 2443 int32_t base = android_atomic_add(AUDIO_UNIQUE_ID_USE_MAX, &mNextUniqueId); 2444 // We have no way of recovering from wraparound 2445 LOG_ALWAYS_FATAL_IF(base == 0, "unique ID overflow"); 2446 // This is the internal API, so it is OK to assert on bad parameter. 2447 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2448 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2449 return (audio_unique_id_t) (base | use); 2450} 2451 2452AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2453{ 2454 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2455 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2456 if(thread->isDuplicating()) { 2457 continue; 2458 } 2459 AudioStreamOut *output = thread->getOutput(); 2460 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2461 return thread; 2462 } 2463 } 2464 return NULL; 2465} 2466 2467audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2468{ 2469 PlaybackThread *thread = primaryPlaybackThread_l(); 2470 2471 if (thread == NULL) { 2472 return 0; 2473 } 2474 2475 return thread->outDevice(); 2476} 2477 2478sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2479 audio_session_t triggerSession, 2480 audio_session_t listenerSession, 2481 sync_event_callback_t callBack, 2482 wp<RefBase> cookie) 2483{ 2484 Mutex::Autolock _l(mLock); 2485 2486 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2487 status_t playStatus = NAME_NOT_FOUND; 2488 status_t recStatus = NAME_NOT_FOUND; 2489 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2490 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2491 if (playStatus == NO_ERROR) { 2492 return event; 2493 } 2494 } 2495 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2496 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2497 if (recStatus == NO_ERROR) { 2498 return event; 2499 } 2500 } 2501 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2502 mPendingSyncEvents.add(event); 2503 } else { 2504 ALOGV("createSyncEvent() invalid event %d", event->type()); 2505 event.clear(); 2506 } 2507 return event; 2508} 2509 2510// ---------------------------------------------------------------------------- 2511// Effect management 2512// ---------------------------------------------------------------------------- 2513 2514 2515status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2516{ 2517 Mutex::Autolock _l(mLock); 2518 return EffectQueryNumberEffects(numEffects); 2519} 2520 2521status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2522{ 2523 Mutex::Autolock _l(mLock); 2524 return EffectQueryEffect(index, descriptor); 2525} 2526 2527status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2528 effect_descriptor_t *descriptor) const 2529{ 2530 Mutex::Autolock _l(mLock); 2531 return EffectGetDescriptor(pUuid, descriptor); 2532} 2533 2534 2535sp<IEffect> AudioFlinger::createEffect( 2536 effect_descriptor_t *pDesc, 2537 const sp<IEffectClient>& effectClient, 2538 int32_t priority, 2539 audio_io_handle_t io, 2540 audio_session_t sessionId, 2541 const String16& opPackageName, 2542 status_t *status, 2543 int *id, 2544 int *enabled) 2545{ 2546 status_t lStatus = NO_ERROR; 2547 sp<EffectHandle> handle; 2548 effect_descriptor_t desc; 2549 2550 pid_t pid = IPCThreadState::self()->getCallingPid(); 2551 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2552 pid, effectClient.get(), priority, sessionId, io); 2553 2554 if (pDesc == NULL) { 2555 lStatus = BAD_VALUE; 2556 goto Exit; 2557 } 2558 2559 // check audio settings permission for global effects 2560 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2561 lStatus = PERMISSION_DENIED; 2562 goto Exit; 2563 } 2564 2565 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2566 // that can only be created by audio policy manager (running in same process) 2567 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2568 lStatus = PERMISSION_DENIED; 2569 goto Exit; 2570 } 2571 2572 { 2573 if (!EffectIsNullUuid(&pDesc->uuid)) { 2574 // if uuid is specified, request effect descriptor 2575 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2576 if (lStatus < 0) { 2577 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2578 goto Exit; 2579 } 2580 } else { 2581 // if uuid is not specified, look for an available implementation 2582 // of the required type in effect factory 2583 if (EffectIsNullUuid(&pDesc->type)) { 2584 ALOGW("createEffect() no effect type"); 2585 lStatus = BAD_VALUE; 2586 goto Exit; 2587 } 2588 uint32_t numEffects = 0; 2589 effect_descriptor_t d; 2590 d.flags = 0; // prevent compiler warning 2591 bool found = false; 2592 2593 lStatus = EffectQueryNumberEffects(&numEffects); 2594 if (lStatus < 0) { 2595 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2596 goto Exit; 2597 } 2598 for (uint32_t i = 0; i < numEffects; i++) { 2599 lStatus = EffectQueryEffect(i, &desc); 2600 if (lStatus < 0) { 2601 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2602 continue; 2603 } 2604 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2605 // If matching type found save effect descriptor. If the session is 2606 // 0 and the effect is not auxiliary, continue enumeration in case 2607 // an auxiliary version of this effect type is available 2608 found = true; 2609 d = desc; 2610 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2611 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2612 break; 2613 } 2614 } 2615 } 2616 if (!found) { 2617 lStatus = BAD_VALUE; 2618 ALOGW("createEffect() effect not found"); 2619 goto Exit; 2620 } 2621 // For same effect type, chose auxiliary version over insert version if 2622 // connect to output mix (Compliance to OpenSL ES) 2623 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2624 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2625 desc = d; 2626 } 2627 } 2628 2629 // Do not allow auxiliary effects on a session different from 0 (output mix) 2630 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2631 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2632 lStatus = INVALID_OPERATION; 2633 goto Exit; 2634 } 2635 2636 // check recording permission for visualizer 2637 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2638 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2639 lStatus = PERMISSION_DENIED; 2640 goto Exit; 2641 } 2642 2643 // return effect descriptor 2644 *pDesc = desc; 2645 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2646 // if the output returned by getOutputForEffect() is removed before we lock the 2647 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2648 // and we will exit safely 2649 io = AudioSystem::getOutputForEffect(&desc); 2650 ALOGV("createEffect got output %d", io); 2651 } 2652 2653 Mutex::Autolock _l(mLock); 2654 2655 // If output is not specified try to find a matching audio session ID in one of the 2656 // output threads. 2657 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2658 // because of code checking output when entering the function. 2659 // Note: io is never 0 when creating an effect on an input 2660 if (io == AUDIO_IO_HANDLE_NONE) { 2661 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2662 // output must be specified by AudioPolicyManager when using session 2663 // AUDIO_SESSION_OUTPUT_STAGE 2664 lStatus = BAD_VALUE; 2665 goto Exit; 2666 } 2667 // look for the thread where the specified audio session is present 2668 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2669 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2670 io = mPlaybackThreads.keyAt(i); 2671 break; 2672 } 2673 } 2674 if (io == 0) { 2675 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2676 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2677 io = mRecordThreads.keyAt(i); 2678 break; 2679 } 2680 } 2681 } 2682 // If no output thread contains the requested session ID, default to 2683 // first output. The effect chain will be moved to the correct output 2684 // thread when a track with the same session ID is created 2685 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2686 io = mPlaybackThreads.keyAt(0); 2687 } 2688 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2689 } 2690 ThreadBase *thread = checkRecordThread_l(io); 2691 if (thread == NULL) { 2692 thread = checkPlaybackThread_l(io); 2693 if (thread == NULL) { 2694 ALOGE("createEffect() unknown output thread"); 2695 lStatus = BAD_VALUE; 2696 goto Exit; 2697 } 2698 } else { 2699 // Check if one effect chain was awaiting for an effect to be created on this 2700 // session and used it instead of creating a new one. 2701 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2702 if (chain != 0) { 2703 Mutex::Autolock _l(thread->mLock); 2704 thread->addEffectChain_l(chain); 2705 } 2706 } 2707 2708 sp<Client> client = registerPid(pid); 2709 2710 // create effect on selected output thread 2711 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2712 &desc, enabled, &lStatus); 2713 if (handle != 0 && id != NULL) { 2714 *id = handle->id(); 2715 } 2716 if (handle == 0) { 2717 // remove local strong reference to Client with mClientLock held 2718 Mutex::Autolock _cl(mClientLock); 2719 client.clear(); 2720 } 2721 } 2722 2723Exit: 2724 *status = lStatus; 2725 return handle; 2726} 2727 2728status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2729 audio_io_handle_t dstOutput) 2730{ 2731 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2732 sessionId, srcOutput, dstOutput); 2733 Mutex::Autolock _l(mLock); 2734 if (srcOutput == dstOutput) { 2735 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2736 return NO_ERROR; 2737 } 2738 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2739 if (srcThread == NULL) { 2740 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2741 return BAD_VALUE; 2742 } 2743 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2744 if (dstThread == NULL) { 2745 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2746 return BAD_VALUE; 2747 } 2748 2749 Mutex::Autolock _dl(dstThread->mLock); 2750 Mutex::Autolock _sl(srcThread->mLock); 2751 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2752} 2753 2754// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2755status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2756 AudioFlinger::PlaybackThread *srcThread, 2757 AudioFlinger::PlaybackThread *dstThread, 2758 bool reRegister) 2759{ 2760 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2761 sessionId, srcThread, dstThread); 2762 2763 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2764 if (chain == 0) { 2765 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2766 sessionId, srcThread); 2767 return INVALID_OPERATION; 2768 } 2769 2770 // Check whether the destination thread has a channel count of FCC_2, which is 2771 // currently required for (most) effects. Prevent moving the effect chain here rather 2772 // than disabling the addEffect_l() call in dstThread below. 2773 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2774 dstThread->mChannelCount != FCC_2) { 2775 ALOGW("moveEffectChain_l() effect chain failed because" 2776 " destination thread %p channel count(%u) != %u", 2777 dstThread, dstThread->mChannelCount, FCC_2); 2778 return INVALID_OPERATION; 2779 } 2780 2781 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2782 // so that a new chain is created with correct parameters when first effect is added. This is 2783 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2784 // removed. 2785 srcThread->removeEffectChain_l(chain); 2786 2787 // transfer all effects one by one so that new effect chain is created on new thread with 2788 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2789 sp<EffectChain> dstChain; 2790 uint32_t strategy = 0; // prevent compiler warning 2791 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2792 Vector< sp<EffectModule> > removed; 2793 status_t status = NO_ERROR; 2794 while (effect != 0) { 2795 srcThread->removeEffect_l(effect); 2796 removed.add(effect); 2797 status = dstThread->addEffect_l(effect); 2798 if (status != NO_ERROR) { 2799 break; 2800 } 2801 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2802 if (effect->state() == EffectModule::ACTIVE || 2803 effect->state() == EffectModule::STOPPING) { 2804 effect->start(); 2805 } 2806 // if the move request is not received from audio policy manager, the effect must be 2807 // re-registered with the new strategy and output 2808 if (dstChain == 0) { 2809 dstChain = effect->chain().promote(); 2810 if (dstChain == 0) { 2811 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2812 status = NO_INIT; 2813 break; 2814 } 2815 strategy = dstChain->strategy(); 2816 } 2817 if (reRegister) { 2818 AudioSystem::unregisterEffect(effect->id()); 2819 AudioSystem::registerEffect(&effect->desc(), 2820 dstThread->id(), 2821 strategy, 2822 sessionId, 2823 effect->id()); 2824 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2825 } 2826 effect = chain->getEffectFromId_l(0); 2827 } 2828 2829 if (status != NO_ERROR) { 2830 for (size_t i = 0; i < removed.size(); i++) { 2831 srcThread->addEffect_l(removed[i]); 2832 if (dstChain != 0 && reRegister) { 2833 AudioSystem::unregisterEffect(removed[i]->id()); 2834 AudioSystem::registerEffect(&removed[i]->desc(), 2835 srcThread->id(), 2836 strategy, 2837 sessionId, 2838 removed[i]->id()); 2839 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2840 } 2841 } 2842 } 2843 2844 return status; 2845} 2846 2847bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2848{ 2849 if (mGlobalEffectEnableTime != 0 && 2850 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2851 return true; 2852 } 2853 2854 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2855 sp<EffectChain> ec = 2856 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2857 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2858 return true; 2859 } 2860 } 2861 return false; 2862} 2863 2864void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2865{ 2866 Mutex::Autolock _l(mLock); 2867 2868 mGlobalEffectEnableTime = systemTime(); 2869 2870 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2871 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2872 if (t->mType == ThreadBase::OFFLOAD) { 2873 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2874 } 2875 } 2876 2877} 2878 2879status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2880{ 2881 audio_session_t session = chain->sessionId(); 2882 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2883 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2884 if (index >= 0) { 2885 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2886 return ALREADY_EXISTS; 2887 } 2888 mOrphanEffectChains.add(session, chain); 2889 return NO_ERROR; 2890} 2891 2892sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2893{ 2894 sp<EffectChain> chain; 2895 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2896 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2897 if (index >= 0) { 2898 chain = mOrphanEffectChains.valueAt(index); 2899 mOrphanEffectChains.removeItemsAt(index); 2900 } 2901 return chain; 2902} 2903 2904bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2905{ 2906 Mutex::Autolock _l(mLock); 2907 audio_session_t session = effect->sessionId(); 2908 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2909 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2910 if (index >= 0) { 2911 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2912 if (chain->removeEffect_l(effect) == 0) { 2913 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2914 mOrphanEffectChains.removeItemsAt(index); 2915 } 2916 return true; 2917 } 2918 return false; 2919} 2920 2921 2922struct Entry { 2923#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2924 char mFileName[TEE_MAX_FILENAME]; 2925}; 2926 2927int comparEntry(const void *p1, const void *p2) 2928{ 2929 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2930} 2931 2932#ifdef TEE_SINK 2933void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2934{ 2935 NBAIO_Source *teeSource = source.get(); 2936 if (teeSource != NULL) { 2937 // .wav rotation 2938 // There is a benign race condition if 2 threads call this simultaneously. 2939 // They would both traverse the directory, but the result would simply be 2940 // failures at unlink() which are ignored. It's also unlikely since 2941 // normally dumpsys is only done by bugreport or from the command line. 2942 char teePath[32+256]; 2943 strcpy(teePath, "/data/misc/audioserver"); 2944 size_t teePathLen = strlen(teePath); 2945 DIR *dir = opendir(teePath); 2946 teePath[teePathLen++] = '/'; 2947 if (dir != NULL) { 2948#define TEE_MAX_SORT 20 // number of entries to sort 2949#define TEE_MAX_KEEP 10 // number of entries to keep 2950 struct Entry entries[TEE_MAX_SORT]; 2951 size_t entryCount = 0; 2952 while (entryCount < TEE_MAX_SORT) { 2953 struct dirent de; 2954 struct dirent *result = NULL; 2955 int rc = readdir_r(dir, &de, &result); 2956 if (rc != 0) { 2957 ALOGW("readdir_r failed %d", rc); 2958 break; 2959 } 2960 if (result == NULL) { 2961 break; 2962 } 2963 if (result != &de) { 2964 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2965 break; 2966 } 2967 // ignore non .wav file entries 2968 size_t nameLen = strlen(de.d_name); 2969 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2970 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2971 continue; 2972 } 2973 strcpy(entries[entryCount++].mFileName, de.d_name); 2974 } 2975 (void) closedir(dir); 2976 if (entryCount > TEE_MAX_KEEP) { 2977 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2978 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2979 strcpy(&teePath[teePathLen], entries[i].mFileName); 2980 (void) unlink(teePath); 2981 } 2982 } 2983 } else { 2984 if (fd >= 0) { 2985 dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath, 2986 strerror(errno)); 2987 } 2988 } 2989 char teeTime[16]; 2990 struct timeval tv; 2991 gettimeofday(&tv, NULL); 2992 struct tm tm; 2993 localtime_r(&tv.tv_sec, &tm); 2994 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2995 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2996 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2997 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2998 if (teeFd >= 0) { 2999 // FIXME use libsndfile 3000 char wavHeader[44]; 3001 memcpy(wavHeader, 3002 "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", 3003 sizeof(wavHeader)); 3004 NBAIO_Format format = teeSource->format(); 3005 unsigned channelCount = Format_channelCount(format); 3006 uint32_t sampleRate = Format_sampleRate(format); 3007 size_t frameSize = Format_frameSize(format); 3008 wavHeader[22] = channelCount; // number of channels 3009 wavHeader[24] = sampleRate; // sample rate 3010 wavHeader[25] = sampleRate >> 8; 3011 wavHeader[32] = frameSize; // block alignment 3012 wavHeader[33] = frameSize >> 8; 3013 write(teeFd, wavHeader, sizeof(wavHeader)); 3014 size_t total = 0; 3015 bool firstRead = true; 3016#define TEE_SINK_READ 1024 // frames per I/O operation 3017 void *buffer = malloc(TEE_SINK_READ * frameSize); 3018 for (;;) { 3019 size_t count = TEE_SINK_READ; 3020 ssize_t actual = teeSource->read(buffer, count); 3021 bool wasFirstRead = firstRead; 3022 firstRead = false; 3023 if (actual <= 0) { 3024 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3025 continue; 3026 } 3027 break; 3028 } 3029 ALOG_ASSERT(actual <= (ssize_t)count); 3030 write(teeFd, buffer, actual * frameSize); 3031 total += actual; 3032 } 3033 free(buffer); 3034 lseek(teeFd, (off_t) 4, SEEK_SET); 3035 uint32_t temp = 44 + total * frameSize - 8; 3036 // FIXME not big-endian safe 3037 write(teeFd, &temp, sizeof(temp)); 3038 lseek(teeFd, (off_t) 40, SEEK_SET); 3039 temp = total * frameSize; 3040 // FIXME not big-endian safe 3041 write(teeFd, &temp, sizeof(temp)); 3042 close(teeFd); 3043 if (fd >= 0) { 3044 dprintf(fd, "tee copied to %s\n", teePath); 3045 } 3046 } else { 3047 if (fd >= 0) { 3048 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3049 } 3050 } 3051 } 3052} 3053#endif 3054 3055// ---------------------------------------------------------------------------- 3056 3057status_t AudioFlinger::onTransact( 3058 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3059{ 3060 return BnAudioFlinger::onTransact(code, data, reply, flags); 3061} 3062 3063} // namespace android 3064