AudioFlinger.cpp revision 5ba4440c11eb975ec0e104e0af1981838f42f57c
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/EffectsFactoryApi.h> 49#include <audio_effects/effect_visualizer.h> 50#include <audio_effects/effect_ns.h> 51#include <audio_effects/effect_aec.h> 52 53#include <audio_utils/primitives.h> 54 55#include <powermanager/PowerManager.h> 56 57#include <common_time/cc_helper.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <private/android_filesystem_config.h> 65 66// ---------------------------------------------------------------------------- 67 68// Note: the following macro is used for extremely verbose logging message. In 69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70// 0; but one side effect of this is to turn all LOGV's as well. Some messages 71// are so verbose that we want to suppress them even when we have ALOG_ASSERT 72// turned on. Do not uncomment the #def below unless you really know what you 73// are doing and want to see all of the extremely verbose messages. 74//#define VERY_VERY_VERBOSE_LOGGING 75#ifdef VERY_VERY_VERBOSE_LOGGING 76#define ALOGVV ALOGV 77#else 78#define ALOGVV(a...) do { } while(0) 79#endif 80 81namespace android { 82 83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85static const char kClientLockedString[] = "Client lock is taken\n"; 86 87 88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 89 90uint32_t AudioFlinger::mScreenState; 91 92#ifdef TEE_SINK 93bool AudioFlinger::mTeeSinkInputEnabled = false; 94bool AudioFlinger::mTeeSinkOutputEnabled = false; 95bool AudioFlinger::mTeeSinkTrackEnabled = false; 96 97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 100#endif 101 102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 103// we define a minimum time during which a global effect is considered enabled. 104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 105 106// ---------------------------------------------------------------------------- 107 108const char *formatToString(audio_format_t format) { 109 switch(format) { 110 case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8"; 111 case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16"; 112 case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32"; 113 case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24"; 114 case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24"; 115 case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat"; 116 case AUDIO_FORMAT_MP3: return "mp3"; 117 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 118 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 119 case AUDIO_FORMAT_AAC: return "aac"; 120 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 121 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 122 case AUDIO_FORMAT_VORBIS: return "vorbis"; 123 default: 124 break; 125 } 126 return "unknown"; 127} 128 129static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 130{ 131 const hw_module_t *mod; 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 135 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 136 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 137 if (rc) { 138 goto out; 139 } 140 rc = audio_hw_device_open(mod, dev); 141 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 142 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 143 if (rc) { 144 goto out; 145 } 146 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 147 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 148 rc = BAD_VALUE; 149 goto out; 150 } 151 return 0; 152 153out: 154 *dev = NULL; 155 return rc; 156} 157 158// ---------------------------------------------------------------------------- 159 160AudioFlinger::AudioFlinger() 161 : BnAudioFlinger(), 162 mPrimaryHardwareDev(NULL), 163 mAudioHwDevs(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), 165 mMasterVolume(1.0f), 166 mMasterMute(false), 167 mNextUniqueId(1), 168 mMode(AUDIO_MODE_INVALID), 169 mBtNrecIsOff(false), 170 mIsLowRamDevice(true), 171 mIsDeviceTypeKnown(false), 172 mGlobalEffectEnableTime(0), 173 mPrimaryOutputSampleRate(0) 174{ 175 getpid_cached = getpid(); 176 char value[PROPERTY_VALUE_MAX]; 177 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 178 if (doLog) { 179 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 180 } 181 182#ifdef TEE_SINK 183 (void) property_get("ro.debuggable", value, "0"); 184 int debuggable = atoi(value); 185 int teeEnabled = 0; 186 if (debuggable) { 187 (void) property_get("af.tee", value, "0"); 188 teeEnabled = atoi(value); 189 } 190 // FIXME symbolic constants here 191 if (teeEnabled & 1) { 192 mTeeSinkInputEnabled = true; 193 } 194 if (teeEnabled & 2) { 195 mTeeSinkOutputEnabled = true; 196 } 197 if (teeEnabled & 4) { 198 mTeeSinkTrackEnabled = true; 199 } 200#endif 201} 202 203void AudioFlinger::onFirstRef() 204{ 205 int rc = 0; 206 207 Mutex::Autolock _l(mLock); 208 209 /* TODO: move all this work into an Init() function */ 210 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 211 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 212 uint32_t int_val; 213 if (1 == sscanf(val_str, "%u", &int_val)) { 214 mStandbyTimeInNsecs = milliseconds(int_val); 215 ALOGI("Using %u mSec as standby time.", int_val); 216 } else { 217 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 218 ALOGI("Using default %u mSec as standby time.", 219 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 220 } 221 } 222 223 mPatchPanel = new PatchPanel(this); 224 225 mMode = AUDIO_MODE_NORMAL; 226} 227 228AudioFlinger::~AudioFlinger() 229{ 230 while (!mRecordThreads.isEmpty()) { 231 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 232 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 233 } 234 while (!mPlaybackThreads.isEmpty()) { 235 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 236 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 237 } 238 239 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 240 // no mHardwareLock needed, as there are no other references to this 241 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 242 delete mAudioHwDevs.valueAt(i); 243 } 244 245 // Tell media.log service about any old writers that still need to be unregistered 246 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 247 if (binder != 0) { 248 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 249 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 250 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 251 mUnregisteredWriters.pop(); 252 mediaLogService->unregisterWriter(iMemory); 253 } 254 } 255 256} 257 258static const char * const audio_interfaces[] = { 259 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 260 AUDIO_HARDWARE_MODULE_ID_A2DP, 261 AUDIO_HARDWARE_MODULE_ID_USB, 262}; 263#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 264 265AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 266 audio_module_handle_t module, 267 audio_devices_t devices) 268{ 269 // if module is 0, the request comes from an old policy manager and we should load 270 // well known modules 271 if (module == 0) { 272 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 273 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 274 loadHwModule_l(audio_interfaces[i]); 275 } 276 // then try to find a module supporting the requested device. 277 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 278 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 279 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 280 if ((dev->get_supported_devices != NULL) && 281 (dev->get_supported_devices(dev) & devices) == devices) 282 return audioHwDevice; 283 } 284 } else { 285 // check a match for the requested module handle 286 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 287 if (audioHwDevice != NULL) { 288 return audioHwDevice; 289 } 290 } 291 292 return NULL; 293} 294 295void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 296{ 297 const size_t SIZE = 256; 298 char buffer[SIZE]; 299 String8 result; 300 301 result.append("Clients:\n"); 302 for (size_t i = 0; i < mClients.size(); ++i) { 303 sp<Client> client = mClients.valueAt(i).promote(); 304 if (client != 0) { 305 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 306 result.append(buffer); 307 } 308 } 309 310 result.append("Notification Clients:\n"); 311 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 312 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 313 result.append(buffer); 314 } 315 316 result.append("Global session refs:\n"); 317 result.append(" session pid count\n"); 318 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 319 AudioSessionRef *r = mAudioSessionRefs[i]; 320 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 321 result.append(buffer); 322 } 323 write(fd, result.string(), result.size()); 324} 325 326 327void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 328{ 329 const size_t SIZE = 256; 330 char buffer[SIZE]; 331 String8 result; 332 hardware_call_state hardwareStatus = mHardwareStatus; 333 334 snprintf(buffer, SIZE, "Hardware status: %d\n" 335 "Standby Time mSec: %u\n", 336 hardwareStatus, 337 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 338 result.append(buffer); 339 write(fd, result.string(), result.size()); 340} 341 342void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 343{ 344 const size_t SIZE = 256; 345 char buffer[SIZE]; 346 String8 result; 347 snprintf(buffer, SIZE, "Permission Denial: " 348 "can't dump AudioFlinger from pid=%d, uid=%d\n", 349 IPCThreadState::self()->getCallingPid(), 350 IPCThreadState::self()->getCallingUid()); 351 result.append(buffer); 352 write(fd, result.string(), result.size()); 353} 354 355bool AudioFlinger::dumpTryLock(Mutex& mutex) 356{ 357 bool locked = false; 358 for (int i = 0; i < kDumpLockRetries; ++i) { 359 if (mutex.tryLock() == NO_ERROR) { 360 locked = true; 361 break; 362 } 363 usleep(kDumpLockSleepUs); 364 } 365 return locked; 366} 367 368status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 369{ 370 if (!dumpAllowed()) { 371 dumpPermissionDenial(fd, args); 372 } else { 373 // get state of hardware lock 374 bool hardwareLocked = dumpTryLock(mHardwareLock); 375 if (!hardwareLocked) { 376 String8 result(kHardwareLockedString); 377 write(fd, result.string(), result.size()); 378 } else { 379 mHardwareLock.unlock(); 380 } 381 382 bool locked = dumpTryLock(mLock); 383 384 // failed to lock - AudioFlinger is probably deadlocked 385 if (!locked) { 386 String8 result(kDeadlockedString); 387 write(fd, result.string(), result.size()); 388 } 389 390 bool clientLocked = dumpTryLock(mClientLock); 391 if (!clientLocked) { 392 String8 result(kClientLockedString); 393 write(fd, result.string(), result.size()); 394 } 395 dumpClients(fd, args); 396 if (clientLocked) { 397 mClientLock.unlock(); 398 } 399 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 415 dev->dump(dev, fd); 416 } 417 418#ifdef TEE_SINK 419 // dump the serially shared record tee sink 420 if (mRecordTeeSource != 0) { 421 dumpTee(fd, mRecordTeeSource); 422 } 423#endif 424 425 if (locked) { 426 mLock.unlock(); 427 } 428 429 // append a copy of media.log here by forwarding fd to it, but don't attempt 430 // to lookup the service if it's not running, as it will block for a second 431 if (mLogMemoryDealer != 0) { 432 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 433 if (binder != 0) { 434 dprintf(fd, "\nmedia.log:\n"); 435 Vector<String16> args; 436 binder->dump(fd, args); 437 } 438 } 439 } 440 return NO_ERROR; 441} 442 443sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 444{ 445 Mutex::Autolock _cl(mClientLock); 446 // If pid is already in the mClients wp<> map, then use that entry 447 // (for which promote() is always != 0), otherwise create a new entry and Client. 448 sp<Client> client = mClients.valueFor(pid).promote(); 449 if (client == 0) { 450 client = new Client(this, pid); 451 mClients.add(pid, client); 452 } 453 454 return client; 455} 456 457sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 458{ 459 // If there is no memory allocated for logs, return a dummy writer that does nothing 460 if (mLogMemoryDealer == 0) { 461 return new NBLog::Writer(); 462 } 463 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 464 // Similarly if we can't contact the media.log service, also return a dummy writer 465 if (binder == 0) { 466 return new NBLog::Writer(); 467 } 468 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 469 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 470 // If allocation fails, consult the vector of previously unregistered writers 471 // and garbage-collect one or more them until an allocation succeeds 472 if (shared == 0) { 473 Mutex::Autolock _l(mUnregisteredWritersLock); 474 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 475 { 476 // Pick the oldest stale writer to garbage-collect 477 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 478 mUnregisteredWriters.removeAt(0); 479 mediaLogService->unregisterWriter(iMemory); 480 // Now the media.log remote reference to IMemory is gone. When our last local 481 // reference to IMemory also drops to zero at end of this block, 482 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 483 } 484 // Re-attempt the allocation 485 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 486 if (shared != 0) { 487 goto success; 488 } 489 } 490 // Even after garbage-collecting all old writers, there is still not enough memory, 491 // so return a dummy writer 492 return new NBLog::Writer(); 493 } 494success: 495 mediaLogService->registerWriter(shared, size, name); 496 return new NBLog::Writer(size, shared); 497} 498 499void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 500{ 501 if (writer == 0) { 502 return; 503 } 504 sp<IMemory> iMemory(writer->getIMemory()); 505 if (iMemory == 0) { 506 return; 507 } 508 // Rather than removing the writer immediately, append it to a queue of old writers to 509 // be garbage-collected later. This allows us to continue to view old logs for a while. 510 Mutex::Autolock _l(mUnregisteredWritersLock); 511 mUnregisteredWriters.push(writer); 512} 513 514// IAudioFlinger interface 515 516 517sp<IAudioTrack> AudioFlinger::createTrack( 518 audio_stream_type_t streamType, 519 uint32_t sampleRate, 520 audio_format_t format, 521 audio_channel_mask_t channelMask, 522 size_t *frameCount, 523 IAudioFlinger::track_flags_t *flags, 524 const sp<IMemory>& sharedBuffer, 525 audio_io_handle_t output, 526 pid_t tid, 527 int *sessionId, 528 int clientUid, 529 status_t *status) 530{ 531 sp<PlaybackThread::Track> track; 532 sp<TrackHandle> trackHandle; 533 sp<Client> client; 534 status_t lStatus; 535 int lSessionId; 536 537 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 538 // but if someone uses binder directly they could bypass that and cause us to crash 539 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 540 ALOGE("createTrack() invalid stream type %d", streamType); 541 lStatus = BAD_VALUE; 542 goto Exit; 543 } 544 545 // further sample rate checks are performed by createTrack_l() depending on the thread type 546 if (sampleRate == 0) { 547 ALOGE("createTrack() invalid sample rate %u", sampleRate); 548 lStatus = BAD_VALUE; 549 goto Exit; 550 } 551 552 // further channel mask checks are performed by createTrack_l() depending on the thread type 553 if (!audio_is_output_channel(channelMask)) { 554 ALOGE("createTrack() invalid channel mask %#x", channelMask); 555 lStatus = BAD_VALUE; 556 goto Exit; 557 } 558 559 // further format checks are performed by createTrack_l() depending on the thread type 560 if (!audio_is_valid_format(format)) { 561 ALOGE("createTrack() invalid format %#x", format); 562 lStatus = BAD_VALUE; 563 goto Exit; 564 } 565 566 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 567 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 568 lStatus = BAD_VALUE; 569 goto Exit; 570 } 571 572 { 573 Mutex::Autolock _l(mLock); 574 PlaybackThread *thread = checkPlaybackThread_l(output); 575 if (thread == NULL) { 576 ALOGE("no playback thread found for output handle %d", output); 577 lStatus = BAD_VALUE; 578 goto Exit; 579 } 580 581 pid_t pid = IPCThreadState::self()->getCallingPid(); 582 client = registerPid(pid); 583 584 PlaybackThread *effectThread = NULL; 585 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 586 lSessionId = *sessionId; 587 // check if an effect chain with the same session ID is present on another 588 // output thread and move it here. 589 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 590 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 591 if (mPlaybackThreads.keyAt(i) != output) { 592 uint32_t sessions = t->hasAudioSession(lSessionId); 593 if (sessions & PlaybackThread::EFFECT_SESSION) { 594 effectThread = t.get(); 595 break; 596 } 597 } 598 } 599 } else { 600 // if no audio session id is provided, create one here 601 lSessionId = nextUniqueId(); 602 if (sessionId != NULL) { 603 *sessionId = lSessionId; 604 } 605 } 606 ALOGV("createTrack() lSessionId: %d", lSessionId); 607 608 track = thread->createTrack_l(client, streamType, sampleRate, format, 609 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 610 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 611 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 612 613 // move effect chain to this output thread if an effect on same session was waiting 614 // for a track to be created 615 if (lStatus == NO_ERROR && effectThread != NULL) { 616 // no risk of deadlock because AudioFlinger::mLock is held 617 Mutex::Autolock _dl(thread->mLock); 618 Mutex::Autolock _sl(effectThread->mLock); 619 moveEffectChain_l(lSessionId, effectThread, thread, true); 620 } 621 622 // Look for sync events awaiting for a session to be used. 623 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 624 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 625 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 626 if (lStatus == NO_ERROR) { 627 (void) track->setSyncEvent(mPendingSyncEvents[i]); 628 } else { 629 mPendingSyncEvents[i]->cancel(); 630 } 631 mPendingSyncEvents.removeAt(i); 632 i--; 633 } 634 } 635 } 636 637 } 638 639 if (lStatus != NO_ERROR) { 640 // remove local strong reference to Client before deleting the Track so that the 641 // Client destructor is called by the TrackBase destructor with mClientLock held 642 // Don't hold mClientLock when releasing the reference on the track as the 643 // destructor will acquire it. 644 { 645 Mutex::Autolock _cl(mClientLock); 646 client.clear(); 647 } 648 track.clear(); 649 goto Exit; 650 } 651 652 // return handle to client 653 trackHandle = new TrackHandle(track); 654 655Exit: 656 *status = lStatus; 657 return trackHandle; 658} 659 660uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 661{ 662 Mutex::Autolock _l(mLock); 663 PlaybackThread *thread = checkPlaybackThread_l(output); 664 if (thread == NULL) { 665 ALOGW("sampleRate() unknown thread %d", output); 666 return 0; 667 } 668 return thread->sampleRate(); 669} 670 671audio_format_t AudioFlinger::format(audio_io_handle_t output) const 672{ 673 Mutex::Autolock _l(mLock); 674 PlaybackThread *thread = checkPlaybackThread_l(output); 675 if (thread == NULL) { 676 ALOGW("format() unknown thread %d", output); 677 return AUDIO_FORMAT_INVALID; 678 } 679 return thread->format(); 680} 681 682size_t AudioFlinger::frameCount(audio_io_handle_t output) const 683{ 684 Mutex::Autolock _l(mLock); 685 PlaybackThread *thread = checkPlaybackThread_l(output); 686 if (thread == NULL) { 687 ALOGW("frameCount() unknown thread %d", output); 688 return 0; 689 } 690 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 691 // should examine all callers and fix them to handle smaller counts 692 return thread->frameCount(); 693} 694 695uint32_t AudioFlinger::latency(audio_io_handle_t output) const 696{ 697 Mutex::Autolock _l(mLock); 698 PlaybackThread *thread = checkPlaybackThread_l(output); 699 if (thread == NULL) { 700 ALOGW("latency(): no playback thread found for output handle %d", output); 701 return 0; 702 } 703 return thread->latency(); 704} 705 706status_t AudioFlinger::setMasterVolume(float value) 707{ 708 status_t ret = initCheck(); 709 if (ret != NO_ERROR) { 710 return ret; 711 } 712 713 // check calling permissions 714 if (!settingsAllowed()) { 715 return PERMISSION_DENIED; 716 } 717 718 Mutex::Autolock _l(mLock); 719 mMasterVolume = value; 720 721 // Set master volume in the HALs which support it. 722 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 723 AutoMutex lock(mHardwareLock); 724 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 725 726 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 727 if (dev->canSetMasterVolume()) { 728 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 729 } 730 mHardwareStatus = AUDIO_HW_IDLE; 731 } 732 733 // Now set the master volume in each playback thread. Playback threads 734 // assigned to HALs which do not have master volume support will apply 735 // master volume during the mix operation. Threads with HALs which do 736 // support master volume will simply ignore the setting. 737 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 738 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 739 740 return NO_ERROR; 741} 742 743status_t AudioFlinger::setMode(audio_mode_t mode) 744{ 745 status_t ret = initCheck(); 746 if (ret != NO_ERROR) { 747 return ret; 748 } 749 750 // check calling permissions 751 if (!settingsAllowed()) { 752 return PERMISSION_DENIED; 753 } 754 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 755 ALOGW("Illegal value: setMode(%d)", mode); 756 return BAD_VALUE; 757 } 758 759 { // scope for the lock 760 AutoMutex lock(mHardwareLock); 761 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 762 mHardwareStatus = AUDIO_HW_SET_MODE; 763 ret = dev->set_mode(dev, mode); 764 mHardwareStatus = AUDIO_HW_IDLE; 765 } 766 767 if (NO_ERROR == ret) { 768 Mutex::Autolock _l(mLock); 769 mMode = mode; 770 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 771 mPlaybackThreads.valueAt(i)->setMode(mode); 772 } 773 774 return ret; 775} 776 777status_t AudioFlinger::setMicMute(bool state) 778{ 779 status_t ret = initCheck(); 780 if (ret != NO_ERROR) { 781 return ret; 782 } 783 784 // check calling permissions 785 if (!settingsAllowed()) { 786 return PERMISSION_DENIED; 787 } 788 789 AutoMutex lock(mHardwareLock); 790 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 791 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 792 ret = dev->set_mic_mute(dev, state); 793 mHardwareStatus = AUDIO_HW_IDLE; 794 return ret; 795} 796 797bool AudioFlinger::getMicMute() const 798{ 799 status_t ret = initCheck(); 800 if (ret != NO_ERROR) { 801 return false; 802 } 803 804 bool state = AUDIO_MODE_INVALID; 805 AutoMutex lock(mHardwareLock); 806 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 807 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 808 dev->get_mic_mute(dev, &state); 809 mHardwareStatus = AUDIO_HW_IDLE; 810 return state; 811} 812 813status_t AudioFlinger::setMasterMute(bool muted) 814{ 815 status_t ret = initCheck(); 816 if (ret != NO_ERROR) { 817 return ret; 818 } 819 820 // check calling permissions 821 if (!settingsAllowed()) { 822 return PERMISSION_DENIED; 823 } 824 825 Mutex::Autolock _l(mLock); 826 mMasterMute = muted; 827 828 // Set master mute in the HALs which support it. 829 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 830 AutoMutex lock(mHardwareLock); 831 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 832 833 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 834 if (dev->canSetMasterMute()) { 835 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 836 } 837 mHardwareStatus = AUDIO_HW_IDLE; 838 } 839 840 // Now set the master mute in each playback thread. Playback threads 841 // assigned to HALs which do not have master mute support will apply master 842 // mute during the mix operation. Threads with HALs which do support master 843 // mute will simply ignore the setting. 844 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 845 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 846 847 return NO_ERROR; 848} 849 850float AudioFlinger::masterVolume() const 851{ 852 Mutex::Autolock _l(mLock); 853 return masterVolume_l(); 854} 855 856bool AudioFlinger::masterMute() const 857{ 858 Mutex::Autolock _l(mLock); 859 return masterMute_l(); 860} 861 862float AudioFlinger::masterVolume_l() const 863{ 864 return mMasterVolume; 865} 866 867bool AudioFlinger::masterMute_l() const 868{ 869 return mMasterMute; 870} 871 872status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 873 audio_io_handle_t output) 874{ 875 // check calling permissions 876 if (!settingsAllowed()) { 877 return PERMISSION_DENIED; 878 } 879 880 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 881 ALOGE("setStreamVolume() invalid stream %d", stream); 882 return BAD_VALUE; 883 } 884 885 AutoMutex lock(mLock); 886 PlaybackThread *thread = NULL; 887 if (output != AUDIO_IO_HANDLE_NONE) { 888 thread = checkPlaybackThread_l(output); 889 if (thread == NULL) { 890 return BAD_VALUE; 891 } 892 } 893 894 mStreamTypes[stream].volume = value; 895 896 if (thread == NULL) { 897 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 898 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 899 } 900 } else { 901 thread->setStreamVolume(stream, value); 902 } 903 904 return NO_ERROR; 905} 906 907status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 908{ 909 // check calling permissions 910 if (!settingsAllowed()) { 911 return PERMISSION_DENIED; 912 } 913 914 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 915 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 916 ALOGE("setStreamMute() invalid stream %d", stream); 917 return BAD_VALUE; 918 } 919 920 AutoMutex lock(mLock); 921 mStreamTypes[stream].mute = muted; 922 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 923 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 924 925 return NO_ERROR; 926} 927 928float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 929{ 930 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 931 return 0.0f; 932 } 933 934 AutoMutex lock(mLock); 935 float volume; 936 if (output != AUDIO_IO_HANDLE_NONE) { 937 PlaybackThread *thread = checkPlaybackThread_l(output); 938 if (thread == NULL) { 939 return 0.0f; 940 } 941 volume = thread->streamVolume(stream); 942 } else { 943 volume = streamVolume_l(stream); 944 } 945 946 return volume; 947} 948 949bool AudioFlinger::streamMute(audio_stream_type_t stream) const 950{ 951 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 952 return true; 953 } 954 955 AutoMutex lock(mLock); 956 return streamMute_l(stream); 957} 958 959status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 960{ 961 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 962 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 963 964 // check calling permissions 965 if (!settingsAllowed()) { 966 return PERMISSION_DENIED; 967 } 968 969 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 970 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 971 Mutex::Autolock _l(mLock); 972 status_t final_result = NO_ERROR; 973 { 974 AutoMutex lock(mHardwareLock); 975 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 976 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 977 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 978 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 979 final_result = result ?: final_result; 980 } 981 mHardwareStatus = AUDIO_HW_IDLE; 982 } 983 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 984 AudioParameter param = AudioParameter(keyValuePairs); 985 String8 value; 986 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 987 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 988 if (mBtNrecIsOff != btNrecIsOff) { 989 for (size_t i = 0; i < mRecordThreads.size(); i++) { 990 sp<RecordThread> thread = mRecordThreads.valueAt(i); 991 audio_devices_t device = thread->inDevice(); 992 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 993 // collect all of the thread's session IDs 994 KeyedVector<int, bool> ids = thread->sessionIds(); 995 // suspend effects associated with those session IDs 996 for (size_t j = 0; j < ids.size(); ++j) { 997 int sessionId = ids.keyAt(j); 998 thread->setEffectSuspended(FX_IID_AEC, 999 suspend, 1000 sessionId); 1001 thread->setEffectSuspended(FX_IID_NS, 1002 suspend, 1003 sessionId); 1004 } 1005 } 1006 mBtNrecIsOff = btNrecIsOff; 1007 } 1008 } 1009 String8 screenState; 1010 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1011 bool isOff = screenState == "off"; 1012 if (isOff != (AudioFlinger::mScreenState & 1)) { 1013 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1014 } 1015 } 1016 return final_result; 1017 } 1018 1019 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1020 // and the thread is exited once the lock is released 1021 sp<ThreadBase> thread; 1022 { 1023 Mutex::Autolock _l(mLock); 1024 thread = checkPlaybackThread_l(ioHandle); 1025 if (thread == 0) { 1026 thread = checkRecordThread_l(ioHandle); 1027 } else if (thread == primaryPlaybackThread_l()) { 1028 // indicate output device change to all input threads for pre processing 1029 AudioParameter param = AudioParameter(keyValuePairs); 1030 int value; 1031 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1032 (value != 0)) { 1033 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1034 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1035 } 1036 } 1037 } 1038 } 1039 if (thread != 0) { 1040 return thread->setParameters(keyValuePairs); 1041 } 1042 return BAD_VALUE; 1043} 1044 1045String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1046{ 1047 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1048 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1049 1050 Mutex::Autolock _l(mLock); 1051 1052 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1053 String8 out_s8; 1054 1055 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1056 char *s; 1057 { 1058 AutoMutex lock(mHardwareLock); 1059 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1060 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1061 s = dev->get_parameters(dev, keys.string()); 1062 mHardwareStatus = AUDIO_HW_IDLE; 1063 } 1064 out_s8 += String8(s ? s : ""); 1065 free(s); 1066 } 1067 return out_s8; 1068 } 1069 1070 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1071 if (playbackThread != NULL) { 1072 return playbackThread->getParameters(keys); 1073 } 1074 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1075 if (recordThread != NULL) { 1076 return recordThread->getParameters(keys); 1077 } 1078 return String8(""); 1079} 1080 1081size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1082 audio_channel_mask_t channelMask) const 1083{ 1084 status_t ret = initCheck(); 1085 if (ret != NO_ERROR) { 1086 return 0; 1087 } 1088 1089 AutoMutex lock(mHardwareLock); 1090 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1091 struct audio_config config; 1092 memset(&config, 0, sizeof(config)); 1093 config.sample_rate = sampleRate; 1094 config.channel_mask = channelMask; 1095 config.format = format; 1096 1097 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1098 size_t size = dev->get_input_buffer_size(dev, &config); 1099 mHardwareStatus = AUDIO_HW_IDLE; 1100 return size; 1101} 1102 1103uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1104{ 1105 Mutex::Autolock _l(mLock); 1106 1107 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1108 if (recordThread != NULL) { 1109 return recordThread->getInputFramesLost(); 1110 } 1111 return 0; 1112} 1113 1114status_t AudioFlinger::setVoiceVolume(float value) 1115{ 1116 status_t ret = initCheck(); 1117 if (ret != NO_ERROR) { 1118 return ret; 1119 } 1120 1121 // check calling permissions 1122 if (!settingsAllowed()) { 1123 return PERMISSION_DENIED; 1124 } 1125 1126 AutoMutex lock(mHardwareLock); 1127 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1128 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1129 ret = dev->set_voice_volume(dev, value); 1130 mHardwareStatus = AUDIO_HW_IDLE; 1131 1132 return ret; 1133} 1134 1135status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1136 audio_io_handle_t output) const 1137{ 1138 status_t status; 1139 1140 Mutex::Autolock _l(mLock); 1141 1142 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1143 if (playbackThread != NULL) { 1144 return playbackThread->getRenderPosition(halFrames, dspFrames); 1145 } 1146 1147 return BAD_VALUE; 1148} 1149 1150void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1151{ 1152 Mutex::Autolock _l(mLock); 1153 bool clientAdded = false; 1154 { 1155 Mutex::Autolock _cl(mClientLock); 1156 1157 pid_t pid = IPCThreadState::self()->getCallingPid(); 1158 if (mNotificationClients.indexOfKey(pid) < 0) { 1159 sp<NotificationClient> notificationClient = new NotificationClient(this, 1160 client, 1161 pid); 1162 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1163 1164 mNotificationClients.add(pid, notificationClient); 1165 1166 sp<IBinder> binder = client->asBinder(); 1167 binder->linkToDeath(notificationClient); 1168 clientAdded = true; 1169 } 1170 } 1171 1172 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1173 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1174 if (clientAdded) { 1175 // the config change is always sent from playback or record threads to avoid deadlock 1176 // with AudioSystem::gLock 1177 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1178 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1179 } 1180 1181 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1182 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1183 } 1184 } 1185} 1186 1187void AudioFlinger::removeNotificationClient(pid_t pid) 1188{ 1189 Mutex::Autolock _l(mLock); 1190 { 1191 Mutex::Autolock _cl(mClientLock); 1192 mNotificationClients.removeItem(pid); 1193 } 1194 1195 ALOGV("%d died, releasing its sessions", pid); 1196 size_t num = mAudioSessionRefs.size(); 1197 bool removed = false; 1198 for (size_t i = 0; i< num; ) { 1199 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1200 ALOGV(" pid %d @ %d", ref->mPid, i); 1201 if (ref->mPid == pid) { 1202 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1203 mAudioSessionRefs.removeAt(i); 1204 delete ref; 1205 removed = true; 1206 num--; 1207 } else { 1208 i++; 1209 } 1210 } 1211 if (removed) { 1212 purgeStaleEffects_l(); 1213 } 1214} 1215 1216void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1217{ 1218 Mutex::Autolock _l(mClientLock); 1219 size_t size = mNotificationClients.size(); 1220 for (size_t i = 0; i < size; i++) { 1221 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1222 ioHandle, 1223 param2); 1224 } 1225} 1226 1227// removeClient_l() must be called with AudioFlinger::mClientLock held 1228void AudioFlinger::removeClient_l(pid_t pid) 1229{ 1230 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1231 IPCThreadState::self()->getCallingPid()); 1232 mClients.removeItem(pid); 1233} 1234 1235// getEffectThread_l() must be called with AudioFlinger::mLock held 1236sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1237{ 1238 sp<PlaybackThread> thread; 1239 1240 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1241 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1242 ALOG_ASSERT(thread == 0); 1243 thread = mPlaybackThreads.valueAt(i); 1244 } 1245 } 1246 1247 return thread; 1248} 1249 1250 1251 1252// ---------------------------------------------------------------------------- 1253 1254AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1255 : RefBase(), 1256 mAudioFlinger(audioFlinger), 1257 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1258 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1259 mPid(pid), 1260 mTimedTrackCount(0) 1261{ 1262 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1263} 1264 1265// Client destructor must be called with AudioFlinger::mClientLock held 1266AudioFlinger::Client::~Client() 1267{ 1268 mAudioFlinger->removeClient_l(mPid); 1269} 1270 1271sp<MemoryDealer> AudioFlinger::Client::heap() const 1272{ 1273 return mMemoryDealer; 1274} 1275 1276// Reserve one of the limited slots for a timed audio track associated 1277// with this client 1278bool AudioFlinger::Client::reserveTimedTrack() 1279{ 1280 const int kMaxTimedTracksPerClient = 4; 1281 1282 Mutex::Autolock _l(mTimedTrackLock); 1283 1284 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1285 ALOGW("can not create timed track - pid %d has exceeded the limit", 1286 mPid); 1287 return false; 1288 } 1289 1290 mTimedTrackCount++; 1291 return true; 1292} 1293 1294// Release a slot for a timed audio track 1295void AudioFlinger::Client::releaseTimedTrack() 1296{ 1297 Mutex::Autolock _l(mTimedTrackLock); 1298 mTimedTrackCount--; 1299} 1300 1301// ---------------------------------------------------------------------------- 1302 1303AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1304 const sp<IAudioFlingerClient>& client, 1305 pid_t pid) 1306 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1307{ 1308} 1309 1310AudioFlinger::NotificationClient::~NotificationClient() 1311{ 1312} 1313 1314void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1315{ 1316 sp<NotificationClient> keep(this); 1317 mAudioFlinger->removeNotificationClient(mPid); 1318} 1319 1320 1321// ---------------------------------------------------------------------------- 1322 1323static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1324 return audio_is_remote_submix_device(inDevice); 1325} 1326 1327sp<IAudioRecord> AudioFlinger::openRecord( 1328 audio_io_handle_t input, 1329 uint32_t sampleRate, 1330 audio_format_t format, 1331 audio_channel_mask_t channelMask, 1332 size_t *frameCount, 1333 IAudioFlinger::track_flags_t *flags, 1334 pid_t tid, 1335 int *sessionId, 1336 sp<IMemory>& cblk, 1337 sp<IMemory>& buffers, 1338 status_t *status) 1339{ 1340 sp<RecordThread::RecordTrack> recordTrack; 1341 sp<RecordHandle> recordHandle; 1342 sp<Client> client; 1343 status_t lStatus; 1344 int lSessionId; 1345 1346 cblk.clear(); 1347 buffers.clear(); 1348 1349 // check calling permissions 1350 if (!recordingAllowed()) { 1351 ALOGE("openRecord() permission denied: recording not allowed"); 1352 lStatus = PERMISSION_DENIED; 1353 goto Exit; 1354 } 1355 1356 // further sample rate checks are performed by createRecordTrack_l() 1357 if (sampleRate == 0) { 1358 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1359 lStatus = BAD_VALUE; 1360 goto Exit; 1361 } 1362 1363 // we don't yet support anything other than 16-bit PCM 1364 if (!(audio_is_valid_format(format) && 1365 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1366 ALOGE("openRecord() invalid format %#x", format); 1367 lStatus = BAD_VALUE; 1368 goto Exit; 1369 } 1370 1371 // further channel mask checks are performed by createRecordTrack_l() 1372 if (!audio_is_input_channel(channelMask)) { 1373 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1374 lStatus = BAD_VALUE; 1375 goto Exit; 1376 } 1377 1378 { 1379 Mutex::Autolock _l(mLock); 1380 RecordThread *thread = checkRecordThread_l(input); 1381 if (thread == NULL) { 1382 ALOGE("openRecord() checkRecordThread_l failed"); 1383 lStatus = BAD_VALUE; 1384 goto Exit; 1385 } 1386 1387 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1388 && !captureAudioOutputAllowed()) { 1389 ALOGE("openRecord() permission denied: capture not allowed"); 1390 lStatus = PERMISSION_DENIED; 1391 goto Exit; 1392 } 1393 1394 pid_t pid = IPCThreadState::self()->getCallingPid(); 1395 client = registerPid(pid); 1396 1397 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1398 lSessionId = *sessionId; 1399 } else { 1400 // if no audio session id is provided, create one here 1401 lSessionId = nextUniqueId(); 1402 if (sessionId != NULL) { 1403 *sessionId = lSessionId; 1404 } 1405 } 1406 ALOGV("openRecord() lSessionId: %d", lSessionId); 1407 1408 // TODO: the uid should be passed in as a parameter to openRecord 1409 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1410 frameCount, lSessionId, 1411 IPCThreadState::self()->getCallingUid(), 1412 flags, tid, &lStatus); 1413 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1414 } 1415 1416 if (lStatus != NO_ERROR) { 1417 // remove local strong reference to Client before deleting the RecordTrack so that the 1418 // Client destructor is called by the TrackBase destructor with mClientLock held 1419 // Don't hold mClientLock when releasing the reference on the track as the 1420 // destructor will acquire it. 1421 { 1422 Mutex::Autolock _cl(mClientLock); 1423 client.clear(); 1424 } 1425 recordTrack.clear(); 1426 goto Exit; 1427 } 1428 1429 cblk = recordTrack->getCblk(); 1430 buffers = recordTrack->getBuffers(); 1431 1432 // return handle to client 1433 recordHandle = new RecordHandle(recordTrack); 1434 1435Exit: 1436 *status = lStatus; 1437 return recordHandle; 1438} 1439 1440 1441 1442// ---------------------------------------------------------------------------- 1443 1444audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1445{ 1446 if (!settingsAllowed()) { 1447 return 0; 1448 } 1449 Mutex::Autolock _l(mLock); 1450 return loadHwModule_l(name); 1451} 1452 1453// loadHwModule_l() must be called with AudioFlinger::mLock held 1454audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1455{ 1456 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1457 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1458 ALOGW("loadHwModule() module %s already loaded", name); 1459 return mAudioHwDevs.keyAt(i); 1460 } 1461 } 1462 1463 audio_hw_device_t *dev; 1464 1465 int rc = load_audio_interface(name, &dev); 1466 if (rc) { 1467 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1468 return 0; 1469 } 1470 1471 mHardwareStatus = AUDIO_HW_INIT; 1472 rc = dev->init_check(dev); 1473 mHardwareStatus = AUDIO_HW_IDLE; 1474 if (rc) { 1475 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1476 return 0; 1477 } 1478 1479 // Check and cache this HAL's level of support for master mute and master 1480 // volume. If this is the first HAL opened, and it supports the get 1481 // methods, use the initial values provided by the HAL as the current 1482 // master mute and volume settings. 1483 1484 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1485 { // scope for auto-lock pattern 1486 AutoMutex lock(mHardwareLock); 1487 1488 if (0 == mAudioHwDevs.size()) { 1489 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1490 if (NULL != dev->get_master_volume) { 1491 float mv; 1492 if (OK == dev->get_master_volume(dev, &mv)) { 1493 mMasterVolume = mv; 1494 } 1495 } 1496 1497 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1498 if (NULL != dev->get_master_mute) { 1499 bool mm; 1500 if (OK == dev->get_master_mute(dev, &mm)) { 1501 mMasterMute = mm; 1502 } 1503 } 1504 } 1505 1506 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1507 if ((NULL != dev->set_master_volume) && 1508 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1509 flags = static_cast<AudioHwDevice::Flags>(flags | 1510 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1511 } 1512 1513 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1514 if ((NULL != dev->set_master_mute) && 1515 (OK == dev->set_master_mute(dev, mMasterMute))) { 1516 flags = static_cast<AudioHwDevice::Flags>(flags | 1517 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1518 } 1519 1520 mHardwareStatus = AUDIO_HW_IDLE; 1521 } 1522 1523 audio_module_handle_t handle = nextUniqueId(); 1524 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1525 1526 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1527 name, dev->common.module->name, dev->common.module->id, handle); 1528 1529 return handle; 1530 1531} 1532 1533// ---------------------------------------------------------------------------- 1534 1535uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1536{ 1537 Mutex::Autolock _l(mLock); 1538 PlaybackThread *thread = primaryPlaybackThread_l(); 1539 return thread != NULL ? thread->sampleRate() : 0; 1540} 1541 1542size_t AudioFlinger::getPrimaryOutputFrameCount() 1543{ 1544 Mutex::Autolock _l(mLock); 1545 PlaybackThread *thread = primaryPlaybackThread_l(); 1546 return thread != NULL ? thread->frameCountHAL() : 0; 1547} 1548 1549// ---------------------------------------------------------------------------- 1550 1551status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1552{ 1553 uid_t uid = IPCThreadState::self()->getCallingUid(); 1554 if (uid != AID_SYSTEM) { 1555 return PERMISSION_DENIED; 1556 } 1557 Mutex::Autolock _l(mLock); 1558 if (mIsDeviceTypeKnown) { 1559 return INVALID_OPERATION; 1560 } 1561 mIsLowRamDevice = isLowRamDevice; 1562 mIsDeviceTypeKnown = true; 1563 return NO_ERROR; 1564} 1565 1566// ---------------------------------------------------------------------------- 1567 1568audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1569 audio_devices_t *pDevices, 1570 uint32_t *pSamplingRate, 1571 audio_format_t *pFormat, 1572 audio_channel_mask_t *pChannelMask, 1573 uint32_t *pLatencyMs, 1574 audio_output_flags_t flags, 1575 const audio_offload_info_t *offloadInfo) 1576{ 1577 struct audio_config config; 1578 memset(&config, 0, sizeof(config)); 1579 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1580 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1581 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1582 if (offloadInfo != NULL) { 1583 config.offload_info = *offloadInfo; 1584 } 1585 1586 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1587 module, 1588 (pDevices != NULL) ? *pDevices : 0, 1589 config.sample_rate, 1590 config.format, 1591 config.channel_mask, 1592 flags); 1593 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1594 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version); 1595 1596 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1597 return AUDIO_IO_HANDLE_NONE; 1598 } 1599 1600 Mutex::Autolock _l(mLock); 1601 1602 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices); 1603 if (outHwDev == NULL) { 1604 return AUDIO_IO_HANDLE_NONE; 1605 } 1606 1607 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1608 audio_io_handle_t id = nextUniqueId(); 1609 1610 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1611 1612 audio_stream_out_t *outStream = NULL; 1613 1614 // FOR TESTING ONLY: 1615 // Enable increased sink precision for mixing mode if kEnableExtendedPrecision is true. 1616 if (kEnableExtendedPrecision && // Check only for Normal Mixing mode 1617 !(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1618 // Update format 1619 //config.format = AUDIO_FORMAT_PCM_FLOAT; 1620 //config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1621 //config.format = AUDIO_FORMAT_PCM_32_BIT; 1622 //config.format = AUDIO_FORMAT_PCM_8_24_BIT; 1623 // ALOGV("openOutput() upgrading format to %#08x", config.format); 1624 } 1625 1626 status_t status = hwDevHal->open_output_stream(hwDevHal, 1627 id, 1628 *pDevices, 1629 (audio_output_flags_t)flags, 1630 &config, 1631 &outStream); 1632 1633 mHardwareStatus = AUDIO_HW_IDLE; 1634 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1635 "Channels %x, status %d", 1636 outStream, 1637 config.sample_rate, 1638 config.format, 1639 config.channel_mask, 1640 status); 1641 1642 if (status == NO_ERROR && outStream != NULL) { 1643 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1644 1645 PlaybackThread *thread; 1646 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1647 thread = new OffloadThread(this, output, id, *pDevices); 1648 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1649 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1650 || !isValidPcmSinkFormat(config.format) 1651 || (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1652 thread = new DirectOutputThread(this, output, id, *pDevices); 1653 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1654 } else { 1655 thread = new MixerThread(this, output, id, *pDevices); 1656 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1657 } 1658 mPlaybackThreads.add(id, thread); 1659 1660 if (pSamplingRate != NULL) { 1661 *pSamplingRate = config.sample_rate; 1662 } 1663 if (pFormat != NULL) { 1664 *pFormat = config.format; 1665 } 1666 if (pChannelMask != NULL) { 1667 *pChannelMask = config.channel_mask; 1668 } 1669 if (pLatencyMs != NULL) { 1670 *pLatencyMs = thread->latency(); 1671 } 1672 1673 // notify client processes of the new output creation 1674 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1675 1676 // the first primary output opened designates the primary hw device 1677 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1678 ALOGI("Using module %d has the primary audio interface", module); 1679 mPrimaryHardwareDev = outHwDev; 1680 1681 AutoMutex lock(mHardwareLock); 1682 mHardwareStatus = AUDIO_HW_SET_MODE; 1683 hwDevHal->set_mode(hwDevHal, mMode); 1684 mHardwareStatus = AUDIO_HW_IDLE; 1685 1686 mPrimaryOutputSampleRate = config.sample_rate; 1687 } 1688 return id; 1689 } 1690 1691 return AUDIO_IO_HANDLE_NONE; 1692} 1693 1694audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1695 audio_io_handle_t output2) 1696{ 1697 Mutex::Autolock _l(mLock); 1698 MixerThread *thread1 = checkMixerThread_l(output1); 1699 MixerThread *thread2 = checkMixerThread_l(output2); 1700 1701 if (thread1 == NULL || thread2 == NULL) { 1702 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1703 output2); 1704 return AUDIO_IO_HANDLE_NONE; 1705 } 1706 1707 audio_io_handle_t id = nextUniqueId(); 1708 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1709 thread->addOutputTrack(thread2); 1710 mPlaybackThreads.add(id, thread); 1711 // notify client processes of the new output creation 1712 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1713 return id; 1714} 1715 1716status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1717{ 1718 return closeOutput_nonvirtual(output); 1719} 1720 1721status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1722{ 1723 // keep strong reference on the playback thread so that 1724 // it is not destroyed while exit() is executed 1725 sp<PlaybackThread> thread; 1726 { 1727 Mutex::Autolock _l(mLock); 1728 thread = checkPlaybackThread_l(output); 1729 if (thread == NULL) { 1730 return BAD_VALUE; 1731 } 1732 1733 ALOGV("closeOutput() %d", output); 1734 1735 if (thread->type() == ThreadBase::MIXER) { 1736 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1737 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1738 DuplicatingThread *dupThread = 1739 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1740 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1741 1742 } 1743 } 1744 } 1745 1746 1747 mPlaybackThreads.removeItem(output); 1748 // save all effects to the default thread 1749 if (mPlaybackThreads.size()) { 1750 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1751 if (dstThread != NULL) { 1752 // audioflinger lock is held here so the acquisition order of thread locks does not 1753 // matter 1754 Mutex::Autolock _dl(dstThread->mLock); 1755 Mutex::Autolock _sl(thread->mLock); 1756 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1757 for (size_t i = 0; i < effectChains.size(); i ++) { 1758 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1759 } 1760 } 1761 } 1762 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1763 } 1764 thread->exit(); 1765 // The thread entity (active unit of execution) is no longer running here, 1766 // but the ThreadBase container still exists. 1767 1768 if (thread->type() != ThreadBase::DUPLICATING) { 1769 AudioStreamOut *out = thread->clearOutput(); 1770 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1771 // from now on thread->mOutput is NULL 1772 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1773 delete out; 1774 } 1775 return NO_ERROR; 1776} 1777 1778status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1779{ 1780 Mutex::Autolock _l(mLock); 1781 PlaybackThread *thread = checkPlaybackThread_l(output); 1782 1783 if (thread == NULL) { 1784 return BAD_VALUE; 1785 } 1786 1787 ALOGV("suspendOutput() %d", output); 1788 thread->suspend(); 1789 1790 return NO_ERROR; 1791} 1792 1793status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1794{ 1795 Mutex::Autolock _l(mLock); 1796 PlaybackThread *thread = checkPlaybackThread_l(output); 1797 1798 if (thread == NULL) { 1799 return BAD_VALUE; 1800 } 1801 1802 ALOGV("restoreOutput() %d", output); 1803 1804 thread->restore(); 1805 1806 return NO_ERROR; 1807} 1808 1809audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1810 audio_devices_t *pDevices, 1811 uint32_t *pSamplingRate, 1812 audio_format_t *pFormat, 1813 audio_channel_mask_t *pChannelMask) 1814{ 1815 struct audio_config config; 1816 memset(&config, 0, sizeof(config)); 1817 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1818 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1819 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1820 1821 uint32_t reqSamplingRate = config.sample_rate; 1822 audio_format_t reqFormat = config.format; 1823 audio_channel_mask_t reqChannelMask = config.channel_mask; 1824 1825 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1826 return 0; 1827 } 1828 1829 Mutex::Autolock _l(mLock); 1830 1831 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices); 1832 if (inHwDev == NULL) { 1833 return 0; 1834 } 1835 1836 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1837 audio_io_handle_t id = nextUniqueId(); 1838 1839 audio_stream_in_t *inStream = NULL; 1840 status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1841 &inStream); 1842 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, " 1843 "status %d", 1844 inStream, 1845 config.sample_rate, 1846 config.format, 1847 config.channel_mask, 1848 status); 1849 1850 // If the input could not be opened with the requested parameters and we can handle the 1851 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1852 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1853 if (status == BAD_VALUE && 1854 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1855 (config.sample_rate <= 2 * reqSamplingRate) && 1856 (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) && 1857 (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) { 1858 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1859 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1860 inStream = NULL; 1861 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1862 // FIXME log this new status; HAL should not propose any further changes 1863 } 1864 1865 if (status == NO_ERROR && inStream != NULL) { 1866 1867#ifdef TEE_SINK 1868 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1869 // or (re-)create if current Pipe is idle and does not match the new format 1870 sp<NBAIO_Sink> teeSink; 1871 enum { 1872 TEE_SINK_NO, // don't copy input 1873 TEE_SINK_NEW, // copy input using a new pipe 1874 TEE_SINK_OLD, // copy input using an existing pipe 1875 } kind; 1876 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1877 audio_channel_count_from_in_mask( 1878 inStream->common.get_channels(&inStream->common))); 1879 if (!mTeeSinkInputEnabled) { 1880 kind = TEE_SINK_NO; 1881 } else if (!Format_isValid(format)) { 1882 kind = TEE_SINK_NO; 1883 } else if (mRecordTeeSink == 0) { 1884 kind = TEE_SINK_NEW; 1885 } else if (mRecordTeeSink->getStrongCount() != 1) { 1886 kind = TEE_SINK_NO; 1887 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1888 kind = TEE_SINK_OLD; 1889 } else { 1890 kind = TEE_SINK_NEW; 1891 } 1892 switch (kind) { 1893 case TEE_SINK_NEW: { 1894 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1895 size_t numCounterOffers = 0; 1896 const NBAIO_Format offers[1] = {format}; 1897 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1898 ALOG_ASSERT(index == 0); 1899 PipeReader *pipeReader = new PipeReader(*pipe); 1900 numCounterOffers = 0; 1901 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1902 ALOG_ASSERT(index == 0); 1903 mRecordTeeSink = pipe; 1904 mRecordTeeSource = pipeReader; 1905 teeSink = pipe; 1906 } 1907 break; 1908 case TEE_SINK_OLD: 1909 teeSink = mRecordTeeSink; 1910 break; 1911 case TEE_SINK_NO: 1912 default: 1913 break; 1914 } 1915#endif 1916 1917 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1918 1919 // Start record thread 1920 // RecordThread requires both input and output device indication to forward to audio 1921 // pre processing modules 1922 RecordThread *thread = new RecordThread(this, 1923 input, 1924 id, 1925 primaryOutputDevice_l(), 1926 *pDevices 1927#ifdef TEE_SINK 1928 , teeSink 1929#endif 1930 ); 1931 mRecordThreads.add(id, thread); 1932 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1933 if (pSamplingRate != NULL) { 1934 *pSamplingRate = reqSamplingRate; 1935 } 1936 if (pFormat != NULL) { 1937 *pFormat = config.format; 1938 } 1939 if (pChannelMask != NULL) { 1940 *pChannelMask = reqChannelMask; 1941 } 1942 1943 // notify client processes of the new input creation 1944 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1945 return id; 1946 } 1947 1948 return 0; 1949} 1950 1951status_t AudioFlinger::closeInput(audio_io_handle_t input) 1952{ 1953 return closeInput_nonvirtual(input); 1954} 1955 1956status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1957{ 1958 // keep strong reference on the record thread so that 1959 // it is not destroyed while exit() is executed 1960 sp<RecordThread> thread; 1961 { 1962 Mutex::Autolock _l(mLock); 1963 thread = checkRecordThread_l(input); 1964 if (thread == 0) { 1965 return BAD_VALUE; 1966 } 1967 1968 ALOGV("closeInput() %d", input); 1969 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 1970 mRecordThreads.removeItem(input); 1971 } 1972 thread->exit(); 1973 // The thread entity (active unit of execution) is no longer running here, 1974 // but the ThreadBase container still exists. 1975 1976 AudioStreamIn *in = thread->clearInput(); 1977 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1978 // from now on thread->mInput is NULL 1979 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1980 delete in; 1981 1982 return NO_ERROR; 1983} 1984 1985status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 1986{ 1987 Mutex::Autolock _l(mLock); 1988 ALOGV("invalidateStream() stream %d", stream); 1989 1990 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1991 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1992 thread->invalidateTracks(stream); 1993 } 1994 1995 return NO_ERROR; 1996} 1997 1998 1999int AudioFlinger::newAudioSessionId() 2000{ 2001 return nextUniqueId(); 2002} 2003 2004void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2005{ 2006 Mutex::Autolock _l(mLock); 2007 pid_t caller = IPCThreadState::self()->getCallingPid(); 2008 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2009 if (pid != -1 && (caller == getpid_cached)) { 2010 caller = pid; 2011 } 2012 2013 { 2014 Mutex::Autolock _cl(mClientLock); 2015 // Ignore requests received from processes not known as notification client. The request 2016 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2017 // called from a different pid leaving a stale session reference. Also we don't know how 2018 // to clear this reference if the client process dies. 2019 if (mNotificationClients.indexOfKey(caller) < 0) { 2020 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2021 return; 2022 } 2023 } 2024 2025 size_t num = mAudioSessionRefs.size(); 2026 for (size_t i = 0; i< num; i++) { 2027 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2028 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2029 ref->mCnt++; 2030 ALOGV(" incremented refcount to %d", ref->mCnt); 2031 return; 2032 } 2033 } 2034 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2035 ALOGV(" added new entry for %d", audioSession); 2036} 2037 2038void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2039{ 2040 Mutex::Autolock _l(mLock); 2041 pid_t caller = IPCThreadState::self()->getCallingPid(); 2042 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2043 if (pid != -1 && (caller == getpid_cached)) { 2044 caller = pid; 2045 } 2046 size_t num = mAudioSessionRefs.size(); 2047 for (size_t i = 0; i< num; i++) { 2048 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2049 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2050 ref->mCnt--; 2051 ALOGV(" decremented refcount to %d", ref->mCnt); 2052 if (ref->mCnt == 0) { 2053 mAudioSessionRefs.removeAt(i); 2054 delete ref; 2055 purgeStaleEffects_l(); 2056 } 2057 return; 2058 } 2059 } 2060 // If the caller is mediaserver it is likely that the session being released was acquired 2061 // on behalf of a process not in notification clients and we ignore the warning. 2062 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2063} 2064 2065void AudioFlinger::purgeStaleEffects_l() { 2066 2067 ALOGV("purging stale effects"); 2068 2069 Vector< sp<EffectChain> > chains; 2070 2071 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2072 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2073 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2074 sp<EffectChain> ec = t->mEffectChains[j]; 2075 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2076 chains.push(ec); 2077 } 2078 } 2079 } 2080 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2081 sp<RecordThread> t = mRecordThreads.valueAt(i); 2082 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2083 sp<EffectChain> ec = t->mEffectChains[j]; 2084 chains.push(ec); 2085 } 2086 } 2087 2088 for (size_t i = 0; i < chains.size(); i++) { 2089 sp<EffectChain> ec = chains[i]; 2090 int sessionid = ec->sessionId(); 2091 sp<ThreadBase> t = ec->mThread.promote(); 2092 if (t == 0) { 2093 continue; 2094 } 2095 size_t numsessionrefs = mAudioSessionRefs.size(); 2096 bool found = false; 2097 for (size_t k = 0; k < numsessionrefs; k++) { 2098 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2099 if (ref->mSessionid == sessionid) { 2100 ALOGV(" session %d still exists for %d with %d refs", 2101 sessionid, ref->mPid, ref->mCnt); 2102 found = true; 2103 break; 2104 } 2105 } 2106 if (!found) { 2107 Mutex::Autolock _l(t->mLock); 2108 // remove all effects from the chain 2109 while (ec->mEffects.size()) { 2110 sp<EffectModule> effect = ec->mEffects[0]; 2111 effect->unPin(); 2112 t->removeEffect_l(effect); 2113 if (effect->purgeHandles()) { 2114 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2115 } 2116 AudioSystem::unregisterEffect(effect->id()); 2117 } 2118 } 2119 } 2120 return; 2121} 2122 2123// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2124AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2125{ 2126 return mPlaybackThreads.valueFor(output).get(); 2127} 2128 2129// checkMixerThread_l() must be called with AudioFlinger::mLock held 2130AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2131{ 2132 PlaybackThread *thread = checkPlaybackThread_l(output); 2133 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2134} 2135 2136// checkRecordThread_l() must be called with AudioFlinger::mLock held 2137AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2138{ 2139 return mRecordThreads.valueFor(input).get(); 2140} 2141 2142uint32_t AudioFlinger::nextUniqueId() 2143{ 2144 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2145} 2146 2147AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2148{ 2149 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2150 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2151 AudioStreamOut *output = thread->getOutput(); 2152 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2153 return thread; 2154 } 2155 } 2156 return NULL; 2157} 2158 2159audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2160{ 2161 PlaybackThread *thread = primaryPlaybackThread_l(); 2162 2163 if (thread == NULL) { 2164 return 0; 2165 } 2166 2167 return thread->outDevice(); 2168} 2169 2170sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2171 int triggerSession, 2172 int listenerSession, 2173 sync_event_callback_t callBack, 2174 wp<RefBase> cookie) 2175{ 2176 Mutex::Autolock _l(mLock); 2177 2178 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2179 status_t playStatus = NAME_NOT_FOUND; 2180 status_t recStatus = NAME_NOT_FOUND; 2181 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2182 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2183 if (playStatus == NO_ERROR) { 2184 return event; 2185 } 2186 } 2187 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2188 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2189 if (recStatus == NO_ERROR) { 2190 return event; 2191 } 2192 } 2193 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2194 mPendingSyncEvents.add(event); 2195 } else { 2196 ALOGV("createSyncEvent() invalid event %d", event->type()); 2197 event.clear(); 2198 } 2199 return event; 2200} 2201 2202// ---------------------------------------------------------------------------- 2203// Effect management 2204// ---------------------------------------------------------------------------- 2205 2206 2207status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2208{ 2209 Mutex::Autolock _l(mLock); 2210 return EffectQueryNumberEffects(numEffects); 2211} 2212 2213status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2214{ 2215 Mutex::Autolock _l(mLock); 2216 return EffectQueryEffect(index, descriptor); 2217} 2218 2219status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2220 effect_descriptor_t *descriptor) const 2221{ 2222 Mutex::Autolock _l(mLock); 2223 return EffectGetDescriptor(pUuid, descriptor); 2224} 2225 2226 2227sp<IEffect> AudioFlinger::createEffect( 2228 effect_descriptor_t *pDesc, 2229 const sp<IEffectClient>& effectClient, 2230 int32_t priority, 2231 audio_io_handle_t io, 2232 int sessionId, 2233 status_t *status, 2234 int *id, 2235 int *enabled) 2236{ 2237 status_t lStatus = NO_ERROR; 2238 sp<EffectHandle> handle; 2239 effect_descriptor_t desc; 2240 2241 pid_t pid = IPCThreadState::self()->getCallingPid(); 2242 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2243 pid, effectClient.get(), priority, sessionId, io); 2244 2245 if (pDesc == NULL) { 2246 lStatus = BAD_VALUE; 2247 goto Exit; 2248 } 2249 2250 // check audio settings permission for global effects 2251 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2252 lStatus = PERMISSION_DENIED; 2253 goto Exit; 2254 } 2255 2256 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2257 // that can only be created by audio policy manager (running in same process) 2258 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2259 lStatus = PERMISSION_DENIED; 2260 goto Exit; 2261 } 2262 2263 { 2264 if (!EffectIsNullUuid(&pDesc->uuid)) { 2265 // if uuid is specified, request effect descriptor 2266 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2267 if (lStatus < 0) { 2268 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2269 goto Exit; 2270 } 2271 } else { 2272 // if uuid is not specified, look for an available implementation 2273 // of the required type in effect factory 2274 if (EffectIsNullUuid(&pDesc->type)) { 2275 ALOGW("createEffect() no effect type"); 2276 lStatus = BAD_VALUE; 2277 goto Exit; 2278 } 2279 uint32_t numEffects = 0; 2280 effect_descriptor_t d; 2281 d.flags = 0; // prevent compiler warning 2282 bool found = false; 2283 2284 lStatus = EffectQueryNumberEffects(&numEffects); 2285 if (lStatus < 0) { 2286 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2287 goto Exit; 2288 } 2289 for (uint32_t i = 0; i < numEffects; i++) { 2290 lStatus = EffectQueryEffect(i, &desc); 2291 if (lStatus < 0) { 2292 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2293 continue; 2294 } 2295 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2296 // If matching type found save effect descriptor. If the session is 2297 // 0 and the effect is not auxiliary, continue enumeration in case 2298 // an auxiliary version of this effect type is available 2299 found = true; 2300 d = desc; 2301 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2302 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2303 break; 2304 } 2305 } 2306 } 2307 if (!found) { 2308 lStatus = BAD_VALUE; 2309 ALOGW("createEffect() effect not found"); 2310 goto Exit; 2311 } 2312 // For same effect type, chose auxiliary version over insert version if 2313 // connect to output mix (Compliance to OpenSL ES) 2314 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2315 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2316 desc = d; 2317 } 2318 } 2319 2320 // Do not allow auxiliary effects on a session different from 0 (output mix) 2321 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2322 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2323 lStatus = INVALID_OPERATION; 2324 goto Exit; 2325 } 2326 2327 // check recording permission for visualizer 2328 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2329 !recordingAllowed()) { 2330 lStatus = PERMISSION_DENIED; 2331 goto Exit; 2332 } 2333 2334 // return effect descriptor 2335 *pDesc = desc; 2336 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2337 // if the output returned by getOutputForEffect() is removed before we lock the 2338 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2339 // and we will exit safely 2340 io = AudioSystem::getOutputForEffect(&desc); 2341 ALOGV("createEffect got output %d", io); 2342 } 2343 2344 Mutex::Autolock _l(mLock); 2345 2346 // If output is not specified try to find a matching audio session ID in one of the 2347 // output threads. 2348 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2349 // because of code checking output when entering the function. 2350 // Note: io is never 0 when creating an effect on an input 2351 if (io == AUDIO_IO_HANDLE_NONE) { 2352 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2353 // output must be specified by AudioPolicyManager when using session 2354 // AUDIO_SESSION_OUTPUT_STAGE 2355 lStatus = BAD_VALUE; 2356 goto Exit; 2357 } 2358 // look for the thread where the specified audio session is present 2359 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2360 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2361 io = mPlaybackThreads.keyAt(i); 2362 break; 2363 } 2364 } 2365 if (io == 0) { 2366 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2367 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2368 io = mRecordThreads.keyAt(i); 2369 break; 2370 } 2371 } 2372 } 2373 // If no output thread contains the requested session ID, default to 2374 // first output. The effect chain will be moved to the correct output 2375 // thread when a track with the same session ID is created 2376 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2377 io = mPlaybackThreads.keyAt(0); 2378 } 2379 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2380 } 2381 ThreadBase *thread = checkRecordThread_l(io); 2382 if (thread == NULL) { 2383 thread = checkPlaybackThread_l(io); 2384 if (thread == NULL) { 2385 ALOGE("createEffect() unknown output thread"); 2386 lStatus = BAD_VALUE; 2387 goto Exit; 2388 } 2389 } 2390 2391 sp<Client> client = registerPid(pid); 2392 2393 // create effect on selected output thread 2394 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2395 &desc, enabled, &lStatus); 2396 if (handle != 0 && id != NULL) { 2397 *id = handle->id(); 2398 } 2399 if (handle == 0) { 2400 // remove local strong reference to Client with mClientLock held 2401 Mutex::Autolock _cl(mClientLock); 2402 client.clear(); 2403 } 2404 } 2405 2406Exit: 2407 *status = lStatus; 2408 return handle; 2409} 2410 2411status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2412 audio_io_handle_t dstOutput) 2413{ 2414 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2415 sessionId, srcOutput, dstOutput); 2416 Mutex::Autolock _l(mLock); 2417 if (srcOutput == dstOutput) { 2418 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2419 return NO_ERROR; 2420 } 2421 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2422 if (srcThread == NULL) { 2423 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2424 return BAD_VALUE; 2425 } 2426 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2427 if (dstThread == NULL) { 2428 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2429 return BAD_VALUE; 2430 } 2431 2432 Mutex::Autolock _dl(dstThread->mLock); 2433 Mutex::Autolock _sl(srcThread->mLock); 2434 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2435} 2436 2437// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2438status_t AudioFlinger::moveEffectChain_l(int sessionId, 2439 AudioFlinger::PlaybackThread *srcThread, 2440 AudioFlinger::PlaybackThread *dstThread, 2441 bool reRegister) 2442{ 2443 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2444 sessionId, srcThread, dstThread); 2445 2446 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2447 if (chain == 0) { 2448 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2449 sessionId, srcThread); 2450 return INVALID_OPERATION; 2451 } 2452 2453 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2454 // so that a new chain is created with correct parameters when first effect is added. This is 2455 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2456 // removed. 2457 srcThread->removeEffectChain_l(chain); 2458 2459 // transfer all effects one by one so that new effect chain is created on new thread with 2460 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2461 sp<EffectChain> dstChain; 2462 uint32_t strategy = 0; // prevent compiler warning 2463 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2464 Vector< sp<EffectModule> > removed; 2465 status_t status = NO_ERROR; 2466 while (effect != 0) { 2467 srcThread->removeEffect_l(effect); 2468 removed.add(effect); 2469 status = dstThread->addEffect_l(effect); 2470 if (status != NO_ERROR) { 2471 break; 2472 } 2473 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2474 if (effect->state() == EffectModule::ACTIVE || 2475 effect->state() == EffectModule::STOPPING) { 2476 effect->start(); 2477 } 2478 // if the move request is not received from audio policy manager, the effect must be 2479 // re-registered with the new strategy and output 2480 if (dstChain == 0) { 2481 dstChain = effect->chain().promote(); 2482 if (dstChain == 0) { 2483 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2484 status = NO_INIT; 2485 break; 2486 } 2487 strategy = dstChain->strategy(); 2488 } 2489 if (reRegister) { 2490 AudioSystem::unregisterEffect(effect->id()); 2491 AudioSystem::registerEffect(&effect->desc(), 2492 dstThread->id(), 2493 strategy, 2494 sessionId, 2495 effect->id()); 2496 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2497 } 2498 effect = chain->getEffectFromId_l(0); 2499 } 2500 2501 if (status != NO_ERROR) { 2502 for (size_t i = 0; i < removed.size(); i++) { 2503 srcThread->addEffect_l(removed[i]); 2504 if (dstChain != 0 && reRegister) { 2505 AudioSystem::unregisterEffect(removed[i]->id()); 2506 AudioSystem::registerEffect(&removed[i]->desc(), 2507 srcThread->id(), 2508 strategy, 2509 sessionId, 2510 removed[i]->id()); 2511 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2512 } 2513 } 2514 } 2515 2516 return status; 2517} 2518 2519bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2520{ 2521 if (mGlobalEffectEnableTime != 0 && 2522 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2523 return true; 2524 } 2525 2526 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2527 sp<EffectChain> ec = 2528 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2529 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2530 return true; 2531 } 2532 } 2533 return false; 2534} 2535 2536void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2537{ 2538 Mutex::Autolock _l(mLock); 2539 2540 mGlobalEffectEnableTime = systemTime(); 2541 2542 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2543 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2544 if (t->mType == ThreadBase::OFFLOAD) { 2545 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2546 } 2547 } 2548 2549} 2550 2551struct Entry { 2552#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2553 char mName[MAX_NAME]; 2554}; 2555 2556int comparEntry(const void *p1, const void *p2) 2557{ 2558 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2559} 2560 2561#ifdef TEE_SINK 2562void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2563{ 2564 NBAIO_Source *teeSource = source.get(); 2565 if (teeSource != NULL) { 2566 // .wav rotation 2567 // There is a benign race condition if 2 threads call this simultaneously. 2568 // They would both traverse the directory, but the result would simply be 2569 // failures at unlink() which are ignored. It's also unlikely since 2570 // normally dumpsys is only done by bugreport or from the command line. 2571 char teePath[32+256]; 2572 strcpy(teePath, "/data/misc/media"); 2573 size_t teePathLen = strlen(teePath); 2574 DIR *dir = opendir(teePath); 2575 teePath[teePathLen++] = '/'; 2576 if (dir != NULL) { 2577#define MAX_SORT 20 // number of entries to sort 2578#define MAX_KEEP 10 // number of entries to keep 2579 struct Entry entries[MAX_SORT]; 2580 size_t entryCount = 0; 2581 while (entryCount < MAX_SORT) { 2582 struct dirent de; 2583 struct dirent *result = NULL; 2584 int rc = readdir_r(dir, &de, &result); 2585 if (rc != 0) { 2586 ALOGW("readdir_r failed %d", rc); 2587 break; 2588 } 2589 if (result == NULL) { 2590 break; 2591 } 2592 if (result != &de) { 2593 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2594 break; 2595 } 2596 // ignore non .wav file entries 2597 size_t nameLen = strlen(de.d_name); 2598 if (nameLen <= 4 || nameLen >= MAX_NAME || 2599 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2600 continue; 2601 } 2602 strcpy(entries[entryCount++].mName, de.d_name); 2603 } 2604 (void) closedir(dir); 2605 if (entryCount > MAX_KEEP) { 2606 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2607 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2608 strcpy(&teePath[teePathLen], entries[i].mName); 2609 (void) unlink(teePath); 2610 } 2611 } 2612 } else { 2613 if (fd >= 0) { 2614 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2615 } 2616 } 2617 char teeTime[16]; 2618 struct timeval tv; 2619 gettimeofday(&tv, NULL); 2620 struct tm tm; 2621 localtime_r(&tv.tv_sec, &tm); 2622 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2623 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2624 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2625 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2626 if (teeFd >= 0) { 2627 char wavHeader[44]; 2628 memcpy(wavHeader, 2629 "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", 2630 sizeof(wavHeader)); 2631 NBAIO_Format format = teeSource->format(); 2632 unsigned channelCount = Format_channelCount(format); 2633 ALOG_ASSERT(channelCount <= FCC_2); 2634 uint32_t sampleRate = Format_sampleRate(format); 2635 wavHeader[22] = channelCount; // number of channels 2636 wavHeader[24] = sampleRate; // sample rate 2637 wavHeader[25] = sampleRate >> 8; 2638 wavHeader[32] = channelCount * 2; // block alignment 2639 write(teeFd, wavHeader, sizeof(wavHeader)); 2640 size_t total = 0; 2641 bool firstRead = true; 2642 for (;;) { 2643#define TEE_SINK_READ 1024 2644 short buffer[TEE_SINK_READ * FCC_2]; 2645 size_t count = TEE_SINK_READ; 2646 ssize_t actual = teeSource->read(buffer, count, 2647 AudioBufferProvider::kInvalidPTS); 2648 bool wasFirstRead = firstRead; 2649 firstRead = false; 2650 if (actual <= 0) { 2651 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2652 continue; 2653 } 2654 break; 2655 } 2656 ALOG_ASSERT(actual <= (ssize_t)count); 2657 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2658 total += actual; 2659 } 2660 lseek(teeFd, (off_t) 4, SEEK_SET); 2661 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2662 write(teeFd, &temp, sizeof(temp)); 2663 lseek(teeFd, (off_t) 40, SEEK_SET); 2664 temp = total * channelCount * sizeof(short); 2665 write(teeFd, &temp, sizeof(temp)); 2666 close(teeFd); 2667 if (fd >= 0) { 2668 dprintf(fd, "tee copied to %s\n", teePath); 2669 } 2670 } else { 2671 if (fd >= 0) { 2672 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2673 } 2674 } 2675 } 2676} 2677#endif 2678 2679// ---------------------------------------------------------------------------- 2680 2681status_t AudioFlinger::onTransact( 2682 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2683{ 2684 return BnAudioFlinger::onTransact(code, data, reply, flags); 2685} 2686 2687}; // namespace android 2688