AudioFlinger.cpp revision c263ca0ad8b6bdf5b0693996bc5f2f5916e0cd49
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 fdprintf(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 671int AudioFlinger::channelCount(audio_io_handle_t output) const 672{ 673 Mutex::Autolock _l(mLock); 674 PlaybackThread *thread = checkPlaybackThread_l(output); 675 if (thread == NULL) { 676 ALOGW("channelCount() unknown thread %d", output); 677 return 0; 678 } 679 return thread->channelCount(); 680} 681 682audio_format_t AudioFlinger::format(audio_io_handle_t output) const 683{ 684 Mutex::Autolock _l(mLock); 685 PlaybackThread *thread = checkPlaybackThread_l(output); 686 if (thread == NULL) { 687 ALOGW("format() unknown thread %d", output); 688 return AUDIO_FORMAT_INVALID; 689 } 690 return thread->format(); 691} 692 693size_t AudioFlinger::frameCount(audio_io_handle_t output) const 694{ 695 Mutex::Autolock _l(mLock); 696 PlaybackThread *thread = checkPlaybackThread_l(output); 697 if (thread == NULL) { 698 ALOGW("frameCount() unknown thread %d", output); 699 return 0; 700 } 701 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 702 // should examine all callers and fix them to handle smaller counts 703 return thread->frameCount(); 704} 705 706uint32_t AudioFlinger::latency(audio_io_handle_t output) const 707{ 708 Mutex::Autolock _l(mLock); 709 PlaybackThread *thread = checkPlaybackThread_l(output); 710 if (thread == NULL) { 711 ALOGW("latency(): no playback thread found for output handle %d", output); 712 return 0; 713 } 714 return thread->latency(); 715} 716 717status_t AudioFlinger::setMasterVolume(float value) 718{ 719 status_t ret = initCheck(); 720 if (ret != NO_ERROR) { 721 return ret; 722 } 723 724 // check calling permissions 725 if (!settingsAllowed()) { 726 return PERMISSION_DENIED; 727 } 728 729 Mutex::Autolock _l(mLock); 730 mMasterVolume = value; 731 732 // Set master volume in the HALs which support it. 733 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 734 AutoMutex lock(mHardwareLock); 735 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 736 737 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 738 if (dev->canSetMasterVolume()) { 739 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 740 } 741 mHardwareStatus = AUDIO_HW_IDLE; 742 } 743 744 // Now set the master volume in each playback thread. Playback threads 745 // assigned to HALs which do not have master volume support will apply 746 // master volume during the mix operation. Threads with HALs which do 747 // support master volume will simply ignore the setting. 748 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 749 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 750 751 return NO_ERROR; 752} 753 754status_t AudioFlinger::setMode(audio_mode_t mode) 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 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 766 ALOGW("Illegal value: setMode(%d)", mode); 767 return BAD_VALUE; 768 } 769 770 { // scope for the lock 771 AutoMutex lock(mHardwareLock); 772 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 773 mHardwareStatus = AUDIO_HW_SET_MODE; 774 ret = dev->set_mode(dev, mode); 775 mHardwareStatus = AUDIO_HW_IDLE; 776 } 777 778 if (NO_ERROR == ret) { 779 Mutex::Autolock _l(mLock); 780 mMode = mode; 781 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 782 mPlaybackThreads.valueAt(i)->setMode(mode); 783 } 784 785 return ret; 786} 787 788status_t AudioFlinger::setMicMute(bool state) 789{ 790 status_t ret = initCheck(); 791 if (ret != NO_ERROR) { 792 return ret; 793 } 794 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 AutoMutex lock(mHardwareLock); 801 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 802 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 803 ret = dev->set_mic_mute(dev, state); 804 mHardwareStatus = AUDIO_HW_IDLE; 805 return ret; 806} 807 808bool AudioFlinger::getMicMute() const 809{ 810 status_t ret = initCheck(); 811 if (ret != NO_ERROR) { 812 return false; 813 } 814 815 bool state = AUDIO_MODE_INVALID; 816 AutoMutex lock(mHardwareLock); 817 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 818 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 819 dev->get_mic_mute(dev, &state); 820 mHardwareStatus = AUDIO_HW_IDLE; 821 return state; 822} 823 824status_t AudioFlinger::setMasterMute(bool muted) 825{ 826 status_t ret = initCheck(); 827 if (ret != NO_ERROR) { 828 return ret; 829 } 830 831 // check calling permissions 832 if (!settingsAllowed()) { 833 return PERMISSION_DENIED; 834 } 835 836 Mutex::Autolock _l(mLock); 837 mMasterMute = muted; 838 839 // Set master mute in the HALs which support it. 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 AutoMutex lock(mHardwareLock); 842 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 843 844 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 845 if (dev->canSetMasterMute()) { 846 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 847 } 848 mHardwareStatus = AUDIO_HW_IDLE; 849 } 850 851 // Now set the master mute in each playback thread. Playback threads 852 // assigned to HALs which do not have master mute support will apply master 853 // mute during the mix operation. Threads with HALs which do support master 854 // mute will simply ignore the setting. 855 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 856 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 857 858 return NO_ERROR; 859} 860 861float AudioFlinger::masterVolume() const 862{ 863 Mutex::Autolock _l(mLock); 864 return masterVolume_l(); 865} 866 867bool AudioFlinger::masterMute() const 868{ 869 Mutex::Autolock _l(mLock); 870 return masterMute_l(); 871} 872 873float AudioFlinger::masterVolume_l() const 874{ 875 return mMasterVolume; 876} 877 878bool AudioFlinger::masterMute_l() const 879{ 880 return mMasterMute; 881} 882 883status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 884 audio_io_handle_t output) 885{ 886 // check calling permissions 887 if (!settingsAllowed()) { 888 return PERMISSION_DENIED; 889 } 890 891 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 892 ALOGE("setStreamVolume() invalid stream %d", stream); 893 return BAD_VALUE; 894 } 895 896 AutoMutex lock(mLock); 897 PlaybackThread *thread = NULL; 898 if (output != AUDIO_IO_HANDLE_NONE) { 899 thread = checkPlaybackThread_l(output); 900 if (thread == NULL) { 901 return BAD_VALUE; 902 } 903 } 904 905 mStreamTypes[stream].volume = value; 906 907 if (thread == NULL) { 908 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 909 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 910 } 911 } else { 912 thread->setStreamVolume(stream, value); 913 } 914 915 return NO_ERROR; 916} 917 918status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 919{ 920 // check calling permissions 921 if (!settingsAllowed()) { 922 return PERMISSION_DENIED; 923 } 924 925 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 926 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 927 ALOGE("setStreamMute() invalid stream %d", stream); 928 return BAD_VALUE; 929 } 930 931 AutoMutex lock(mLock); 932 mStreamTypes[stream].mute = muted; 933 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 934 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 935 936 return NO_ERROR; 937} 938 939float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 940{ 941 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 942 return 0.0f; 943 } 944 945 AutoMutex lock(mLock); 946 float volume; 947 if (output != AUDIO_IO_HANDLE_NONE) { 948 PlaybackThread *thread = checkPlaybackThread_l(output); 949 if (thread == NULL) { 950 return 0.0f; 951 } 952 volume = thread->streamVolume(stream); 953 } else { 954 volume = streamVolume_l(stream); 955 } 956 957 return volume; 958} 959 960bool AudioFlinger::streamMute(audio_stream_type_t stream) const 961{ 962 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 963 return true; 964 } 965 966 AutoMutex lock(mLock); 967 return streamMute_l(stream); 968} 969 970status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 971{ 972 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 973 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 974 975 // check calling permissions 976 if (!settingsAllowed()) { 977 return PERMISSION_DENIED; 978 } 979 980 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 981 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 982 Mutex::Autolock _l(mLock); 983 status_t final_result = NO_ERROR; 984 { 985 AutoMutex lock(mHardwareLock); 986 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 987 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 988 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 989 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 990 final_result = result ?: final_result; 991 } 992 mHardwareStatus = AUDIO_HW_IDLE; 993 } 994 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 995 AudioParameter param = AudioParameter(keyValuePairs); 996 String8 value; 997 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 998 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 999 if (mBtNrecIsOff != btNrecIsOff) { 1000 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1001 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1002 audio_devices_t device = thread->inDevice(); 1003 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1004 // collect all of the thread's session IDs 1005 KeyedVector<int, bool> ids = thread->sessionIds(); 1006 // suspend effects associated with those session IDs 1007 for (size_t j = 0; j < ids.size(); ++j) { 1008 int sessionId = ids.keyAt(j); 1009 thread->setEffectSuspended(FX_IID_AEC, 1010 suspend, 1011 sessionId); 1012 thread->setEffectSuspended(FX_IID_NS, 1013 suspend, 1014 sessionId); 1015 } 1016 } 1017 mBtNrecIsOff = btNrecIsOff; 1018 } 1019 } 1020 String8 screenState; 1021 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1022 bool isOff = screenState == "off"; 1023 if (isOff != (AudioFlinger::mScreenState & 1)) { 1024 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1025 } 1026 } 1027 return final_result; 1028 } 1029 1030 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1031 // and the thread is exited once the lock is released 1032 sp<ThreadBase> thread; 1033 { 1034 Mutex::Autolock _l(mLock); 1035 thread = checkPlaybackThread_l(ioHandle); 1036 if (thread == 0) { 1037 thread = checkRecordThread_l(ioHandle); 1038 } else if (thread == primaryPlaybackThread_l()) { 1039 // indicate output device change to all input threads for pre processing 1040 AudioParameter param = AudioParameter(keyValuePairs); 1041 int value; 1042 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1043 (value != 0)) { 1044 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1045 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1046 } 1047 } 1048 } 1049 } 1050 if (thread != 0) { 1051 return thread->setParameters(keyValuePairs); 1052 } 1053 return BAD_VALUE; 1054} 1055 1056String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1057{ 1058 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1059 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1060 1061 Mutex::Autolock _l(mLock); 1062 1063 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1064 String8 out_s8; 1065 1066 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1067 char *s; 1068 { 1069 AutoMutex lock(mHardwareLock); 1070 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1071 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1072 s = dev->get_parameters(dev, keys.string()); 1073 mHardwareStatus = AUDIO_HW_IDLE; 1074 } 1075 out_s8 += String8(s ? s : ""); 1076 free(s); 1077 } 1078 return out_s8; 1079 } 1080 1081 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1082 if (playbackThread != NULL) { 1083 return playbackThread->getParameters(keys); 1084 } 1085 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1086 if (recordThread != NULL) { 1087 return recordThread->getParameters(keys); 1088 } 1089 return String8(""); 1090} 1091 1092size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1093 audio_channel_mask_t channelMask) const 1094{ 1095 status_t ret = initCheck(); 1096 if (ret != NO_ERROR) { 1097 return 0; 1098 } 1099 1100 AutoMutex lock(mHardwareLock); 1101 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1102 struct audio_config config; 1103 memset(&config, 0, sizeof(config)); 1104 config.sample_rate = sampleRate; 1105 config.channel_mask = channelMask; 1106 config.format = format; 1107 1108 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1109 size_t size = dev->get_input_buffer_size(dev, &config); 1110 mHardwareStatus = AUDIO_HW_IDLE; 1111 return size; 1112} 1113 1114uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1115{ 1116 Mutex::Autolock _l(mLock); 1117 1118 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1119 if (recordThread != NULL) { 1120 return recordThread->getInputFramesLost(); 1121 } 1122 return 0; 1123} 1124 1125status_t AudioFlinger::setVoiceVolume(float value) 1126{ 1127 status_t ret = initCheck(); 1128 if (ret != NO_ERROR) { 1129 return ret; 1130 } 1131 1132 // check calling permissions 1133 if (!settingsAllowed()) { 1134 return PERMISSION_DENIED; 1135 } 1136 1137 AutoMutex lock(mHardwareLock); 1138 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1139 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1140 ret = dev->set_voice_volume(dev, value); 1141 mHardwareStatus = AUDIO_HW_IDLE; 1142 1143 return ret; 1144} 1145 1146status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1147 audio_io_handle_t output) const 1148{ 1149 status_t status; 1150 1151 Mutex::Autolock _l(mLock); 1152 1153 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1154 if (playbackThread != NULL) { 1155 return playbackThread->getRenderPosition(halFrames, dspFrames); 1156 } 1157 1158 return BAD_VALUE; 1159} 1160 1161void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1162{ 1163 Mutex::Autolock _l(mLock); 1164 bool clientAdded = false; 1165 { 1166 Mutex::Autolock _cl(mClientLock); 1167 1168 pid_t pid = IPCThreadState::self()->getCallingPid(); 1169 if (mNotificationClients.indexOfKey(pid) < 0) { 1170 sp<NotificationClient> notificationClient = new NotificationClient(this, 1171 client, 1172 pid); 1173 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1174 1175 mNotificationClients.add(pid, notificationClient); 1176 1177 sp<IBinder> binder = client->asBinder(); 1178 binder->linkToDeath(notificationClient); 1179 clientAdded = true; 1180 } 1181 } 1182 1183 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1184 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1185 if (clientAdded) { 1186 // the config change is always sent from playback or record threads to avoid deadlock 1187 // with AudioSystem::gLock 1188 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1189 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1190 } 1191 1192 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1193 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1194 } 1195 } 1196} 1197 1198void AudioFlinger::removeNotificationClient(pid_t pid) 1199{ 1200 Mutex::Autolock _l(mLock); 1201 { 1202 Mutex::Autolock _cl(mClientLock); 1203 mNotificationClients.removeItem(pid); 1204 } 1205 1206 ALOGV("%d died, releasing its sessions", pid); 1207 size_t num = mAudioSessionRefs.size(); 1208 bool removed = false; 1209 for (size_t i = 0; i< num; ) { 1210 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1211 ALOGV(" pid %d @ %d", ref->mPid, i); 1212 if (ref->mPid == pid) { 1213 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1214 mAudioSessionRefs.removeAt(i); 1215 delete ref; 1216 removed = true; 1217 num--; 1218 } else { 1219 i++; 1220 } 1221 } 1222 if (removed) { 1223 purgeStaleEffects_l(); 1224 } 1225} 1226 1227void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1228{ 1229 Mutex::Autolock _l(mClientLock); 1230 size_t size = mNotificationClients.size(); 1231 for (size_t i = 0; i < size; i++) { 1232 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1233 ioHandle, 1234 param2); 1235 } 1236} 1237 1238// removeClient_l() must be called with AudioFlinger::mClientLock held 1239void AudioFlinger::removeClient_l(pid_t pid) 1240{ 1241 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1242 IPCThreadState::self()->getCallingPid()); 1243 mClients.removeItem(pid); 1244} 1245 1246// getEffectThread_l() must be called with AudioFlinger::mLock held 1247sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1248{ 1249 sp<PlaybackThread> thread; 1250 1251 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1252 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1253 ALOG_ASSERT(thread == 0); 1254 thread = mPlaybackThreads.valueAt(i); 1255 } 1256 } 1257 1258 return thread; 1259} 1260 1261 1262 1263// ---------------------------------------------------------------------------- 1264 1265AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1266 : RefBase(), 1267 mAudioFlinger(audioFlinger), 1268 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1269 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1270 mPid(pid), 1271 mTimedTrackCount(0) 1272{ 1273 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1274} 1275 1276// Client destructor must be called with AudioFlinger::mClientLock held 1277AudioFlinger::Client::~Client() 1278{ 1279 mAudioFlinger->removeClient_l(mPid); 1280} 1281 1282sp<MemoryDealer> AudioFlinger::Client::heap() const 1283{ 1284 return mMemoryDealer; 1285} 1286 1287// Reserve one of the limited slots for a timed audio track associated 1288// with this client 1289bool AudioFlinger::Client::reserveTimedTrack() 1290{ 1291 const int kMaxTimedTracksPerClient = 4; 1292 1293 Mutex::Autolock _l(mTimedTrackLock); 1294 1295 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1296 ALOGW("can not create timed track - pid %d has exceeded the limit", 1297 mPid); 1298 return false; 1299 } 1300 1301 mTimedTrackCount++; 1302 return true; 1303} 1304 1305// Release a slot for a timed audio track 1306void AudioFlinger::Client::releaseTimedTrack() 1307{ 1308 Mutex::Autolock _l(mTimedTrackLock); 1309 mTimedTrackCount--; 1310} 1311 1312// ---------------------------------------------------------------------------- 1313 1314AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1315 const sp<IAudioFlingerClient>& client, 1316 pid_t pid) 1317 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1318{ 1319} 1320 1321AudioFlinger::NotificationClient::~NotificationClient() 1322{ 1323} 1324 1325void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1326{ 1327 sp<NotificationClient> keep(this); 1328 mAudioFlinger->removeNotificationClient(mPid); 1329} 1330 1331 1332// ---------------------------------------------------------------------------- 1333 1334static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1335 return audio_is_remote_submix_device(inDevice); 1336} 1337 1338sp<IAudioRecord> AudioFlinger::openRecord( 1339 audio_io_handle_t input, 1340 uint32_t sampleRate, 1341 audio_format_t format, 1342 audio_channel_mask_t channelMask, 1343 size_t *frameCount, 1344 IAudioFlinger::track_flags_t *flags, 1345 pid_t tid, 1346 int *sessionId, 1347 sp<IMemory>& cblk, 1348 sp<IMemory>& buffers, 1349 status_t *status) 1350{ 1351 sp<RecordThread::RecordTrack> recordTrack; 1352 sp<RecordHandle> recordHandle; 1353 sp<Client> client; 1354 status_t lStatus; 1355 int lSessionId; 1356 1357 cblk.clear(); 1358 buffers.clear(); 1359 1360 // check calling permissions 1361 if (!recordingAllowed()) { 1362 ALOGE("openRecord() permission denied: recording not allowed"); 1363 lStatus = PERMISSION_DENIED; 1364 goto Exit; 1365 } 1366 1367 // further sample rate checks are performed by createRecordTrack_l() 1368 if (sampleRate == 0) { 1369 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1370 lStatus = BAD_VALUE; 1371 goto Exit; 1372 } 1373 1374 // we don't yet support anything other than 16-bit PCM 1375 if (!(audio_is_valid_format(format) && 1376 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1377 ALOGE("openRecord() invalid format %#x", format); 1378 lStatus = BAD_VALUE; 1379 goto Exit; 1380 } 1381 1382 // further channel mask checks are performed by createRecordTrack_l() 1383 if (!audio_is_input_channel(channelMask)) { 1384 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1385 lStatus = BAD_VALUE; 1386 goto Exit; 1387 } 1388 1389 { 1390 Mutex::Autolock _l(mLock); 1391 RecordThread *thread = checkRecordThread_l(input); 1392 if (thread == NULL) { 1393 ALOGE("openRecord() checkRecordThread_l failed"); 1394 lStatus = BAD_VALUE; 1395 goto Exit; 1396 } 1397 1398 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1399 && !captureAudioOutputAllowed()) { 1400 ALOGE("openRecord() permission denied: capture not allowed"); 1401 lStatus = PERMISSION_DENIED; 1402 goto Exit; 1403 } 1404 1405 pid_t pid = IPCThreadState::self()->getCallingPid(); 1406 client = registerPid(pid); 1407 1408 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1409 lSessionId = *sessionId; 1410 } else { 1411 // if no audio session id is provided, create one here 1412 lSessionId = nextUniqueId(); 1413 if (sessionId != NULL) { 1414 *sessionId = lSessionId; 1415 } 1416 } 1417 ALOGV("openRecord() lSessionId: %d", lSessionId); 1418 1419 // TODO: the uid should be passed in as a parameter to openRecord 1420 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1421 frameCount, lSessionId, 1422 IPCThreadState::self()->getCallingUid(), 1423 flags, tid, &lStatus); 1424 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1425 } 1426 1427 if (lStatus != NO_ERROR) { 1428 // remove local strong reference to Client before deleting the RecordTrack so that the 1429 // Client destructor is called by the TrackBase destructor with mClientLock held 1430 // Don't hold mClientLock when releasing the reference on the track as the 1431 // destructor will acquire it. 1432 { 1433 Mutex::Autolock _cl(mClientLock); 1434 client.clear(); 1435 } 1436 recordTrack.clear(); 1437 goto Exit; 1438 } 1439 1440 cblk = recordTrack->getCblk(); 1441 buffers = recordTrack->getBuffers(); 1442 1443 // return handle to client 1444 recordHandle = new RecordHandle(recordTrack); 1445 1446Exit: 1447 *status = lStatus; 1448 return recordHandle; 1449} 1450 1451 1452 1453// ---------------------------------------------------------------------------- 1454 1455audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1456{ 1457 if (!settingsAllowed()) { 1458 return 0; 1459 } 1460 Mutex::Autolock _l(mLock); 1461 return loadHwModule_l(name); 1462} 1463 1464// loadHwModule_l() must be called with AudioFlinger::mLock held 1465audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1466{ 1467 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1468 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1469 ALOGW("loadHwModule() module %s already loaded", name); 1470 return mAudioHwDevs.keyAt(i); 1471 } 1472 } 1473 1474 audio_hw_device_t *dev; 1475 1476 int rc = load_audio_interface(name, &dev); 1477 if (rc) { 1478 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1479 return 0; 1480 } 1481 1482 mHardwareStatus = AUDIO_HW_INIT; 1483 rc = dev->init_check(dev); 1484 mHardwareStatus = AUDIO_HW_IDLE; 1485 if (rc) { 1486 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1487 return 0; 1488 } 1489 1490 // Check and cache this HAL's level of support for master mute and master 1491 // volume. If this is the first HAL opened, and it supports the get 1492 // methods, use the initial values provided by the HAL as the current 1493 // master mute and volume settings. 1494 1495 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1496 { // scope for auto-lock pattern 1497 AutoMutex lock(mHardwareLock); 1498 1499 if (0 == mAudioHwDevs.size()) { 1500 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1501 if (NULL != dev->get_master_volume) { 1502 float mv; 1503 if (OK == dev->get_master_volume(dev, &mv)) { 1504 mMasterVolume = mv; 1505 } 1506 } 1507 1508 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1509 if (NULL != dev->get_master_mute) { 1510 bool mm; 1511 if (OK == dev->get_master_mute(dev, &mm)) { 1512 mMasterMute = mm; 1513 } 1514 } 1515 } 1516 1517 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1518 if ((NULL != dev->set_master_volume) && 1519 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1520 flags = static_cast<AudioHwDevice::Flags>(flags | 1521 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1522 } 1523 1524 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1525 if ((NULL != dev->set_master_mute) && 1526 (OK == dev->set_master_mute(dev, mMasterMute))) { 1527 flags = static_cast<AudioHwDevice::Flags>(flags | 1528 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1529 } 1530 1531 mHardwareStatus = AUDIO_HW_IDLE; 1532 } 1533 1534 audio_module_handle_t handle = nextUniqueId(); 1535 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1536 1537 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1538 name, dev->common.module->name, dev->common.module->id, handle); 1539 1540 return handle; 1541 1542} 1543 1544// ---------------------------------------------------------------------------- 1545 1546uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1547{ 1548 Mutex::Autolock _l(mLock); 1549 PlaybackThread *thread = primaryPlaybackThread_l(); 1550 return thread != NULL ? thread->sampleRate() : 0; 1551} 1552 1553size_t AudioFlinger::getPrimaryOutputFrameCount() 1554{ 1555 Mutex::Autolock _l(mLock); 1556 PlaybackThread *thread = primaryPlaybackThread_l(); 1557 return thread != NULL ? thread->frameCountHAL() : 0; 1558} 1559 1560// ---------------------------------------------------------------------------- 1561 1562status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1563{ 1564 uid_t uid = IPCThreadState::self()->getCallingUid(); 1565 if (uid != AID_SYSTEM) { 1566 return PERMISSION_DENIED; 1567 } 1568 Mutex::Autolock _l(mLock); 1569 if (mIsDeviceTypeKnown) { 1570 return INVALID_OPERATION; 1571 } 1572 mIsLowRamDevice = isLowRamDevice; 1573 mIsDeviceTypeKnown = true; 1574 return NO_ERROR; 1575} 1576 1577// ---------------------------------------------------------------------------- 1578 1579audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1580 audio_devices_t *pDevices, 1581 uint32_t *pSamplingRate, 1582 audio_format_t *pFormat, 1583 audio_channel_mask_t *pChannelMask, 1584 uint32_t *pLatencyMs, 1585 audio_output_flags_t flags, 1586 const audio_offload_info_t *offloadInfo) 1587{ 1588 struct audio_config config; 1589 memset(&config, 0, sizeof(config)); 1590 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1591 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1592 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1593 if (offloadInfo != NULL) { 1594 config.offload_info = *offloadInfo; 1595 } 1596 1597 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1598 module, 1599 (pDevices != NULL) ? *pDevices : 0, 1600 config.sample_rate, 1601 config.format, 1602 config.channel_mask, 1603 flags); 1604 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1605 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version); 1606 1607 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1608 return AUDIO_IO_HANDLE_NONE; 1609 } 1610 1611 Mutex::Autolock _l(mLock); 1612 1613 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices); 1614 if (outHwDev == NULL) { 1615 return AUDIO_IO_HANDLE_NONE; 1616 } 1617 1618 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1619 audio_io_handle_t id = nextUniqueId(); 1620 1621 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1622 1623 audio_stream_out_t *outStream = NULL; 1624 status_t status = hwDevHal->open_output_stream(hwDevHal, 1625 id, 1626 *pDevices, 1627 (audio_output_flags_t)flags, 1628 &config, 1629 &outStream); 1630 1631 mHardwareStatus = AUDIO_HW_IDLE; 1632 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1633 "Channels %x, status %d", 1634 outStream, 1635 config.sample_rate, 1636 config.format, 1637 config.channel_mask, 1638 status); 1639 1640 if (status == NO_ERROR && outStream != NULL) { 1641 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1642 1643 PlaybackThread *thread; 1644 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1645 thread = new OffloadThread(this, output, id, *pDevices); 1646 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1647 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1648 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1649 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1650 thread = new DirectOutputThread(this, output, id, *pDevices); 1651 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1652 } else { 1653 thread = new MixerThread(this, output, id, *pDevices); 1654 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1655 } 1656 mPlaybackThreads.add(id, thread); 1657 1658 if (pSamplingRate != NULL) { 1659 *pSamplingRate = config.sample_rate; 1660 } 1661 if (pFormat != NULL) { 1662 *pFormat = config.format; 1663 } 1664 if (pChannelMask != NULL) { 1665 *pChannelMask = config.channel_mask; 1666 } 1667 if (pLatencyMs != NULL) { 1668 *pLatencyMs = thread->latency(); 1669 } 1670 1671 // notify client processes of the new output creation 1672 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1673 1674 // the first primary output opened designates the primary hw device 1675 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1676 ALOGI("Using module %d has the primary audio interface", module); 1677 mPrimaryHardwareDev = outHwDev; 1678 1679 AutoMutex lock(mHardwareLock); 1680 mHardwareStatus = AUDIO_HW_SET_MODE; 1681 hwDevHal->set_mode(hwDevHal, mMode); 1682 mHardwareStatus = AUDIO_HW_IDLE; 1683 1684 mPrimaryOutputSampleRate = config.sample_rate; 1685 } 1686 return id; 1687 } 1688 1689 return AUDIO_IO_HANDLE_NONE; 1690} 1691 1692audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1693 audio_io_handle_t output2) 1694{ 1695 Mutex::Autolock _l(mLock); 1696 MixerThread *thread1 = checkMixerThread_l(output1); 1697 MixerThread *thread2 = checkMixerThread_l(output2); 1698 1699 if (thread1 == NULL || thread2 == NULL) { 1700 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1701 output2); 1702 return AUDIO_IO_HANDLE_NONE; 1703 } 1704 1705 audio_io_handle_t id = nextUniqueId(); 1706 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1707 thread->addOutputTrack(thread2); 1708 mPlaybackThreads.add(id, thread); 1709 // notify client processes of the new output creation 1710 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1711 return id; 1712} 1713 1714status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1715{ 1716 return closeOutput_nonvirtual(output); 1717} 1718 1719status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1720{ 1721 // keep strong reference on the playback thread so that 1722 // it is not destroyed while exit() is executed 1723 sp<PlaybackThread> thread; 1724 { 1725 Mutex::Autolock _l(mLock); 1726 thread = checkPlaybackThread_l(output); 1727 if (thread == NULL) { 1728 return BAD_VALUE; 1729 } 1730 1731 ALOGV("closeOutput() %d", output); 1732 1733 if (thread->type() == ThreadBase::MIXER) { 1734 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1735 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1736 DuplicatingThread *dupThread = 1737 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1738 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1739 1740 } 1741 } 1742 } 1743 1744 1745 mPlaybackThreads.removeItem(output); 1746 // save all effects to the default thread 1747 if (mPlaybackThreads.size()) { 1748 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1749 if (dstThread != NULL) { 1750 // audioflinger lock is held here so the acquisition order of thread locks does not 1751 // matter 1752 Mutex::Autolock _dl(dstThread->mLock); 1753 Mutex::Autolock _sl(thread->mLock); 1754 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1755 for (size_t i = 0; i < effectChains.size(); i ++) { 1756 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1757 } 1758 } 1759 } 1760 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1761 } 1762 thread->exit(); 1763 // The thread entity (active unit of execution) is no longer running here, 1764 // but the ThreadBase container still exists. 1765 1766 if (thread->type() != ThreadBase::DUPLICATING) { 1767 AudioStreamOut *out = thread->clearOutput(); 1768 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1769 // from now on thread->mOutput is NULL 1770 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1771 delete out; 1772 } 1773 return NO_ERROR; 1774} 1775 1776status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1777{ 1778 Mutex::Autolock _l(mLock); 1779 PlaybackThread *thread = checkPlaybackThread_l(output); 1780 1781 if (thread == NULL) { 1782 return BAD_VALUE; 1783 } 1784 1785 ALOGV("suspendOutput() %d", output); 1786 thread->suspend(); 1787 1788 return NO_ERROR; 1789} 1790 1791status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1792{ 1793 Mutex::Autolock _l(mLock); 1794 PlaybackThread *thread = checkPlaybackThread_l(output); 1795 1796 if (thread == NULL) { 1797 return BAD_VALUE; 1798 } 1799 1800 ALOGV("restoreOutput() %d", output); 1801 1802 thread->restore(); 1803 1804 return NO_ERROR; 1805} 1806 1807audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1808 audio_devices_t *pDevices, 1809 uint32_t *pSamplingRate, 1810 audio_format_t *pFormat, 1811 audio_channel_mask_t *pChannelMask) 1812{ 1813 struct audio_config config; 1814 memset(&config, 0, sizeof(config)); 1815 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1816 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1817 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1818 1819 uint32_t reqSamplingRate = config.sample_rate; 1820 audio_format_t reqFormat = config.format; 1821 audio_channel_mask_t reqChannelMask = config.channel_mask; 1822 1823 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1824 return 0; 1825 } 1826 1827 Mutex::Autolock _l(mLock); 1828 1829 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices); 1830 if (inHwDev == NULL) { 1831 return 0; 1832 } 1833 1834 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1835 audio_io_handle_t id = nextUniqueId(); 1836 1837 audio_stream_in_t *inStream = NULL; 1838 status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1839 &inStream); 1840 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, " 1841 "status %d", 1842 inStream, 1843 config.sample_rate, 1844 config.format, 1845 config.channel_mask, 1846 status); 1847 1848 // If the input could not be opened with the requested parameters and we can handle the 1849 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1850 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1851 if (status == BAD_VALUE && 1852 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1853 (config.sample_rate <= 2 * reqSamplingRate) && 1854 (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) && 1855 (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) { 1856 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1857 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1858 inStream = NULL; 1859 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1860 // FIXME log this new status; HAL should not propose any further changes 1861 } 1862 1863 if (status == NO_ERROR && inStream != NULL) { 1864 1865#ifdef TEE_SINK 1866 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1867 // or (re-)create if current Pipe is idle and does not match the new format 1868 sp<NBAIO_Sink> teeSink; 1869 enum { 1870 TEE_SINK_NO, // don't copy input 1871 TEE_SINK_NEW, // copy input using a new pipe 1872 TEE_SINK_OLD, // copy input using an existing pipe 1873 } kind; 1874 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1875 audio_channel_count_from_in_mask( 1876 inStream->common.get_channels(&inStream->common))); 1877 if (!mTeeSinkInputEnabled) { 1878 kind = TEE_SINK_NO; 1879 } else if (!Format_isValid(format)) { 1880 kind = TEE_SINK_NO; 1881 } else if (mRecordTeeSink == 0) { 1882 kind = TEE_SINK_NEW; 1883 } else if (mRecordTeeSink->getStrongCount() != 1) { 1884 kind = TEE_SINK_NO; 1885 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1886 kind = TEE_SINK_OLD; 1887 } else { 1888 kind = TEE_SINK_NEW; 1889 } 1890 switch (kind) { 1891 case TEE_SINK_NEW: { 1892 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1893 size_t numCounterOffers = 0; 1894 const NBAIO_Format offers[1] = {format}; 1895 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1896 ALOG_ASSERT(index == 0); 1897 PipeReader *pipeReader = new PipeReader(*pipe); 1898 numCounterOffers = 0; 1899 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1900 ALOG_ASSERT(index == 0); 1901 mRecordTeeSink = pipe; 1902 mRecordTeeSource = pipeReader; 1903 teeSink = pipe; 1904 } 1905 break; 1906 case TEE_SINK_OLD: 1907 teeSink = mRecordTeeSink; 1908 break; 1909 case TEE_SINK_NO: 1910 default: 1911 break; 1912 } 1913#endif 1914 1915 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1916 1917 // Start record thread 1918 // RecordThread requires both input and output device indication to forward to audio 1919 // pre processing modules 1920 RecordThread *thread = new RecordThread(this, 1921 input, 1922 id, 1923 primaryOutputDevice_l(), 1924 *pDevices 1925#ifdef TEE_SINK 1926 , teeSink 1927#endif 1928 ); 1929 mRecordThreads.add(id, thread); 1930 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1931 if (pSamplingRate != NULL) { 1932 *pSamplingRate = reqSamplingRate; 1933 } 1934 if (pFormat != NULL) { 1935 *pFormat = config.format; 1936 } 1937 if (pChannelMask != NULL) { 1938 *pChannelMask = reqChannelMask; 1939 } 1940 1941 // notify client processes of the new input creation 1942 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1943 return id; 1944 } 1945 1946 return 0; 1947} 1948 1949status_t AudioFlinger::closeInput(audio_io_handle_t input) 1950{ 1951 return closeInput_nonvirtual(input); 1952} 1953 1954status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1955{ 1956 // keep strong reference on the record thread so that 1957 // it is not destroyed while exit() is executed 1958 sp<RecordThread> thread; 1959 { 1960 Mutex::Autolock _l(mLock); 1961 thread = checkRecordThread_l(input); 1962 if (thread == 0) { 1963 return BAD_VALUE; 1964 } 1965 1966 ALOGV("closeInput() %d", input); 1967 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 1968 mRecordThreads.removeItem(input); 1969 } 1970 thread->exit(); 1971 // The thread entity (active unit of execution) is no longer running here, 1972 // but the ThreadBase container still exists. 1973 1974 AudioStreamIn *in = thread->clearInput(); 1975 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1976 // from now on thread->mInput is NULL 1977 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1978 delete in; 1979 1980 return NO_ERROR; 1981} 1982 1983status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 1984{ 1985 Mutex::Autolock _l(mLock); 1986 ALOGV("invalidateStream() stream %d", stream); 1987 1988 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1989 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1990 thread->invalidateTracks(stream); 1991 } 1992 1993 return NO_ERROR; 1994} 1995 1996 1997int AudioFlinger::newAudioSessionId() 1998{ 1999 return nextUniqueId(); 2000} 2001 2002void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2003{ 2004 Mutex::Autolock _l(mLock); 2005 pid_t caller = IPCThreadState::self()->getCallingPid(); 2006 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2007 if (pid != -1 && (caller == getpid_cached)) { 2008 caller = pid; 2009 } 2010 2011 { 2012 Mutex::Autolock _cl(mClientLock); 2013 // Ignore requests received from processes not known as notification client. The request 2014 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2015 // called from a different pid leaving a stale session reference. Also we don't know how 2016 // to clear this reference if the client process dies. 2017 if (mNotificationClients.indexOfKey(caller) < 0) { 2018 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2019 return; 2020 } 2021 } 2022 2023 size_t num = mAudioSessionRefs.size(); 2024 for (size_t i = 0; i< num; i++) { 2025 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2026 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2027 ref->mCnt++; 2028 ALOGV(" incremented refcount to %d", ref->mCnt); 2029 return; 2030 } 2031 } 2032 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2033 ALOGV(" added new entry for %d", audioSession); 2034} 2035 2036void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2037{ 2038 Mutex::Autolock _l(mLock); 2039 pid_t caller = IPCThreadState::self()->getCallingPid(); 2040 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2041 if (pid != -1 && (caller == getpid_cached)) { 2042 caller = pid; 2043 } 2044 size_t num = mAudioSessionRefs.size(); 2045 for (size_t i = 0; i< num; i++) { 2046 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2047 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2048 ref->mCnt--; 2049 ALOGV(" decremented refcount to %d", ref->mCnt); 2050 if (ref->mCnt == 0) { 2051 mAudioSessionRefs.removeAt(i); 2052 delete ref; 2053 purgeStaleEffects_l(); 2054 } 2055 return; 2056 } 2057 } 2058 // If the caller is mediaserver it is likely that the session being released was acquired 2059 // on behalf of a process not in notification clients and we ignore the warning. 2060 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2061} 2062 2063void AudioFlinger::purgeStaleEffects_l() { 2064 2065 ALOGV("purging stale effects"); 2066 2067 Vector< sp<EffectChain> > chains; 2068 2069 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2070 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2071 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2072 sp<EffectChain> ec = t->mEffectChains[j]; 2073 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2074 chains.push(ec); 2075 } 2076 } 2077 } 2078 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2079 sp<RecordThread> t = mRecordThreads.valueAt(i); 2080 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2081 sp<EffectChain> ec = t->mEffectChains[j]; 2082 chains.push(ec); 2083 } 2084 } 2085 2086 for (size_t i = 0; i < chains.size(); i++) { 2087 sp<EffectChain> ec = chains[i]; 2088 int sessionid = ec->sessionId(); 2089 sp<ThreadBase> t = ec->mThread.promote(); 2090 if (t == 0) { 2091 continue; 2092 } 2093 size_t numsessionrefs = mAudioSessionRefs.size(); 2094 bool found = false; 2095 for (size_t k = 0; k < numsessionrefs; k++) { 2096 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2097 if (ref->mSessionid == sessionid) { 2098 ALOGV(" session %d still exists for %d with %d refs", 2099 sessionid, ref->mPid, ref->mCnt); 2100 found = true; 2101 break; 2102 } 2103 } 2104 if (!found) { 2105 Mutex::Autolock _l(t->mLock); 2106 // remove all effects from the chain 2107 while (ec->mEffects.size()) { 2108 sp<EffectModule> effect = ec->mEffects[0]; 2109 effect->unPin(); 2110 t->removeEffect_l(effect); 2111 if (effect->purgeHandles()) { 2112 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2113 } 2114 AudioSystem::unregisterEffect(effect->id()); 2115 } 2116 } 2117 } 2118 return; 2119} 2120 2121// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2122AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2123{ 2124 return mPlaybackThreads.valueFor(output).get(); 2125} 2126 2127// checkMixerThread_l() must be called with AudioFlinger::mLock held 2128AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2129{ 2130 PlaybackThread *thread = checkPlaybackThread_l(output); 2131 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2132} 2133 2134// checkRecordThread_l() must be called with AudioFlinger::mLock held 2135AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2136{ 2137 return mRecordThreads.valueFor(input).get(); 2138} 2139 2140uint32_t AudioFlinger::nextUniqueId() 2141{ 2142 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2143} 2144 2145AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2146{ 2147 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2148 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2149 AudioStreamOut *output = thread->getOutput(); 2150 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2151 return thread; 2152 } 2153 } 2154 return NULL; 2155} 2156 2157audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2158{ 2159 PlaybackThread *thread = primaryPlaybackThread_l(); 2160 2161 if (thread == NULL) { 2162 return 0; 2163 } 2164 2165 return thread->outDevice(); 2166} 2167 2168sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2169 int triggerSession, 2170 int listenerSession, 2171 sync_event_callback_t callBack, 2172 wp<RefBase> cookie) 2173{ 2174 Mutex::Autolock _l(mLock); 2175 2176 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2177 status_t playStatus = NAME_NOT_FOUND; 2178 status_t recStatus = NAME_NOT_FOUND; 2179 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2180 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2181 if (playStatus == NO_ERROR) { 2182 return event; 2183 } 2184 } 2185 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2186 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2187 if (recStatus == NO_ERROR) { 2188 return event; 2189 } 2190 } 2191 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2192 mPendingSyncEvents.add(event); 2193 } else { 2194 ALOGV("createSyncEvent() invalid event %d", event->type()); 2195 event.clear(); 2196 } 2197 return event; 2198} 2199 2200// ---------------------------------------------------------------------------- 2201// Effect management 2202// ---------------------------------------------------------------------------- 2203 2204 2205status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2206{ 2207 Mutex::Autolock _l(mLock); 2208 return EffectQueryNumberEffects(numEffects); 2209} 2210 2211status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2212{ 2213 Mutex::Autolock _l(mLock); 2214 return EffectQueryEffect(index, descriptor); 2215} 2216 2217status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2218 effect_descriptor_t *descriptor) const 2219{ 2220 Mutex::Autolock _l(mLock); 2221 return EffectGetDescriptor(pUuid, descriptor); 2222} 2223 2224 2225sp<IEffect> AudioFlinger::createEffect( 2226 effect_descriptor_t *pDesc, 2227 const sp<IEffectClient>& effectClient, 2228 int32_t priority, 2229 audio_io_handle_t io, 2230 int sessionId, 2231 status_t *status, 2232 int *id, 2233 int *enabled) 2234{ 2235 status_t lStatus = NO_ERROR; 2236 sp<EffectHandle> handle; 2237 effect_descriptor_t desc; 2238 2239 pid_t pid = IPCThreadState::self()->getCallingPid(); 2240 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2241 pid, effectClient.get(), priority, sessionId, io); 2242 2243 if (pDesc == NULL) { 2244 lStatus = BAD_VALUE; 2245 goto Exit; 2246 } 2247 2248 // check audio settings permission for global effects 2249 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2250 lStatus = PERMISSION_DENIED; 2251 goto Exit; 2252 } 2253 2254 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2255 // that can only be created by audio policy manager (running in same process) 2256 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2257 lStatus = PERMISSION_DENIED; 2258 goto Exit; 2259 } 2260 2261 { 2262 if (!EffectIsNullUuid(&pDesc->uuid)) { 2263 // if uuid is specified, request effect descriptor 2264 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2265 if (lStatus < 0) { 2266 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2267 goto Exit; 2268 } 2269 } else { 2270 // if uuid is not specified, look for an available implementation 2271 // of the required type in effect factory 2272 if (EffectIsNullUuid(&pDesc->type)) { 2273 ALOGW("createEffect() no effect type"); 2274 lStatus = BAD_VALUE; 2275 goto Exit; 2276 } 2277 uint32_t numEffects = 0; 2278 effect_descriptor_t d; 2279 d.flags = 0; // prevent compiler warning 2280 bool found = false; 2281 2282 lStatus = EffectQueryNumberEffects(&numEffects); 2283 if (lStatus < 0) { 2284 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2285 goto Exit; 2286 } 2287 for (uint32_t i = 0; i < numEffects; i++) { 2288 lStatus = EffectQueryEffect(i, &desc); 2289 if (lStatus < 0) { 2290 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2291 continue; 2292 } 2293 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2294 // If matching type found save effect descriptor. If the session is 2295 // 0 and the effect is not auxiliary, continue enumeration in case 2296 // an auxiliary version of this effect type is available 2297 found = true; 2298 d = desc; 2299 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2300 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2301 break; 2302 } 2303 } 2304 } 2305 if (!found) { 2306 lStatus = BAD_VALUE; 2307 ALOGW("createEffect() effect not found"); 2308 goto Exit; 2309 } 2310 // For same effect type, chose auxiliary version over insert version if 2311 // connect to output mix (Compliance to OpenSL ES) 2312 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2313 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2314 desc = d; 2315 } 2316 } 2317 2318 // Do not allow auxiliary effects on a session different from 0 (output mix) 2319 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2320 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2321 lStatus = INVALID_OPERATION; 2322 goto Exit; 2323 } 2324 2325 // check recording permission for visualizer 2326 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2327 !recordingAllowed()) { 2328 lStatus = PERMISSION_DENIED; 2329 goto Exit; 2330 } 2331 2332 // return effect descriptor 2333 *pDesc = desc; 2334 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2335 // if the output returned by getOutputForEffect() is removed before we lock the 2336 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2337 // and we will exit safely 2338 io = AudioSystem::getOutputForEffect(&desc); 2339 ALOGV("createEffect got output %d", io); 2340 } 2341 2342 Mutex::Autolock _l(mLock); 2343 2344 // If output is not specified try to find a matching audio session ID in one of the 2345 // output threads. 2346 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2347 // because of code checking output when entering the function. 2348 // Note: io is never 0 when creating an effect on an input 2349 if (io == AUDIO_IO_HANDLE_NONE) { 2350 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2351 // output must be specified by AudioPolicyManager when using session 2352 // AUDIO_SESSION_OUTPUT_STAGE 2353 lStatus = BAD_VALUE; 2354 goto Exit; 2355 } 2356 // look for the thread where the specified audio session is present 2357 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2358 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2359 io = mPlaybackThreads.keyAt(i); 2360 break; 2361 } 2362 } 2363 if (io == 0) { 2364 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2365 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2366 io = mRecordThreads.keyAt(i); 2367 break; 2368 } 2369 } 2370 } 2371 // If no output thread contains the requested session ID, default to 2372 // first output. The effect chain will be moved to the correct output 2373 // thread when a track with the same session ID is created 2374 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2375 io = mPlaybackThreads.keyAt(0); 2376 } 2377 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2378 } 2379 ThreadBase *thread = checkRecordThread_l(io); 2380 if (thread == NULL) { 2381 thread = checkPlaybackThread_l(io); 2382 if (thread == NULL) { 2383 ALOGE("createEffect() unknown output thread"); 2384 lStatus = BAD_VALUE; 2385 goto Exit; 2386 } 2387 } 2388 2389 sp<Client> client = registerPid(pid); 2390 2391 // create effect on selected output thread 2392 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2393 &desc, enabled, &lStatus); 2394 if (handle != 0 && id != NULL) { 2395 *id = handle->id(); 2396 } 2397 if (handle == 0) { 2398 // remove local strong reference to Client with mClientLock held 2399 Mutex::Autolock _cl(mClientLock); 2400 client.clear(); 2401 } 2402 } 2403 2404Exit: 2405 *status = lStatus; 2406 return handle; 2407} 2408 2409status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2410 audio_io_handle_t dstOutput) 2411{ 2412 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2413 sessionId, srcOutput, dstOutput); 2414 Mutex::Autolock _l(mLock); 2415 if (srcOutput == dstOutput) { 2416 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2417 return NO_ERROR; 2418 } 2419 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2420 if (srcThread == NULL) { 2421 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2422 return BAD_VALUE; 2423 } 2424 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2425 if (dstThread == NULL) { 2426 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2427 return BAD_VALUE; 2428 } 2429 2430 Mutex::Autolock _dl(dstThread->mLock); 2431 Mutex::Autolock _sl(srcThread->mLock); 2432 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2433} 2434 2435// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2436status_t AudioFlinger::moveEffectChain_l(int sessionId, 2437 AudioFlinger::PlaybackThread *srcThread, 2438 AudioFlinger::PlaybackThread *dstThread, 2439 bool reRegister) 2440{ 2441 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2442 sessionId, srcThread, dstThread); 2443 2444 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2445 if (chain == 0) { 2446 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2447 sessionId, srcThread); 2448 return INVALID_OPERATION; 2449 } 2450 2451 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2452 // so that a new chain is created with correct parameters when first effect is added. This is 2453 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2454 // removed. 2455 srcThread->removeEffectChain_l(chain); 2456 2457 // transfer all effects one by one so that new effect chain is created on new thread with 2458 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2459 sp<EffectChain> dstChain; 2460 uint32_t strategy = 0; // prevent compiler warning 2461 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2462 Vector< sp<EffectModule> > removed; 2463 status_t status = NO_ERROR; 2464 while (effect != 0) { 2465 srcThread->removeEffect_l(effect); 2466 removed.add(effect); 2467 status = dstThread->addEffect_l(effect); 2468 if (status != NO_ERROR) { 2469 break; 2470 } 2471 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2472 if (effect->state() == EffectModule::ACTIVE || 2473 effect->state() == EffectModule::STOPPING) { 2474 effect->start(); 2475 } 2476 // if the move request is not received from audio policy manager, the effect must be 2477 // re-registered with the new strategy and output 2478 if (dstChain == 0) { 2479 dstChain = effect->chain().promote(); 2480 if (dstChain == 0) { 2481 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2482 status = NO_INIT; 2483 break; 2484 } 2485 strategy = dstChain->strategy(); 2486 } 2487 if (reRegister) { 2488 AudioSystem::unregisterEffect(effect->id()); 2489 AudioSystem::registerEffect(&effect->desc(), 2490 dstThread->id(), 2491 strategy, 2492 sessionId, 2493 effect->id()); 2494 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2495 } 2496 effect = chain->getEffectFromId_l(0); 2497 } 2498 2499 if (status != NO_ERROR) { 2500 for (size_t i = 0; i < removed.size(); i++) { 2501 srcThread->addEffect_l(removed[i]); 2502 if (dstChain != 0 && reRegister) { 2503 AudioSystem::unregisterEffect(removed[i]->id()); 2504 AudioSystem::registerEffect(&removed[i]->desc(), 2505 srcThread->id(), 2506 strategy, 2507 sessionId, 2508 removed[i]->id()); 2509 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2510 } 2511 } 2512 } 2513 2514 return status; 2515} 2516 2517bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2518{ 2519 if (mGlobalEffectEnableTime != 0 && 2520 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2521 return true; 2522 } 2523 2524 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2525 sp<EffectChain> ec = 2526 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2527 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2528 return true; 2529 } 2530 } 2531 return false; 2532} 2533 2534void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2535{ 2536 Mutex::Autolock _l(mLock); 2537 2538 mGlobalEffectEnableTime = systemTime(); 2539 2540 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2541 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2542 if (t->mType == ThreadBase::OFFLOAD) { 2543 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2544 } 2545 } 2546 2547} 2548 2549struct Entry { 2550#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2551 char mName[MAX_NAME]; 2552}; 2553 2554int comparEntry(const void *p1, const void *p2) 2555{ 2556 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2557} 2558 2559#ifdef TEE_SINK 2560void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2561{ 2562 NBAIO_Source *teeSource = source.get(); 2563 if (teeSource != NULL) { 2564 // .wav rotation 2565 // There is a benign race condition if 2 threads call this simultaneously. 2566 // They would both traverse the directory, but the result would simply be 2567 // failures at unlink() which are ignored. It's also unlikely since 2568 // normally dumpsys is only done by bugreport or from the command line. 2569 char teePath[32+256]; 2570 strcpy(teePath, "/data/misc/media"); 2571 size_t teePathLen = strlen(teePath); 2572 DIR *dir = opendir(teePath); 2573 teePath[teePathLen++] = '/'; 2574 if (dir != NULL) { 2575#define MAX_SORT 20 // number of entries to sort 2576#define MAX_KEEP 10 // number of entries to keep 2577 struct Entry entries[MAX_SORT]; 2578 size_t entryCount = 0; 2579 while (entryCount < MAX_SORT) { 2580 struct dirent de; 2581 struct dirent *result = NULL; 2582 int rc = readdir_r(dir, &de, &result); 2583 if (rc != 0) { 2584 ALOGW("readdir_r failed %d", rc); 2585 break; 2586 } 2587 if (result == NULL) { 2588 break; 2589 } 2590 if (result != &de) { 2591 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2592 break; 2593 } 2594 // ignore non .wav file entries 2595 size_t nameLen = strlen(de.d_name); 2596 if (nameLen <= 4 || nameLen >= MAX_NAME || 2597 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2598 continue; 2599 } 2600 strcpy(entries[entryCount++].mName, de.d_name); 2601 } 2602 (void) closedir(dir); 2603 if (entryCount > MAX_KEEP) { 2604 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2605 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2606 strcpy(&teePath[teePathLen], entries[i].mName); 2607 (void) unlink(teePath); 2608 } 2609 } 2610 } else { 2611 if (fd >= 0) { 2612 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2613 } 2614 } 2615 char teeTime[16]; 2616 struct timeval tv; 2617 gettimeofday(&tv, NULL); 2618 struct tm tm; 2619 localtime_r(&tv.tv_sec, &tm); 2620 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2621 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2622 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2623 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2624 if (teeFd >= 0) { 2625 char wavHeader[44]; 2626 memcpy(wavHeader, 2627 "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", 2628 sizeof(wavHeader)); 2629 NBAIO_Format format = teeSource->format(); 2630 unsigned channelCount = Format_channelCount(format); 2631 ALOG_ASSERT(channelCount <= FCC_2); 2632 uint32_t sampleRate = Format_sampleRate(format); 2633 wavHeader[22] = channelCount; // number of channels 2634 wavHeader[24] = sampleRate; // sample rate 2635 wavHeader[25] = sampleRate >> 8; 2636 wavHeader[32] = channelCount * 2; // block alignment 2637 write(teeFd, wavHeader, sizeof(wavHeader)); 2638 size_t total = 0; 2639 bool firstRead = true; 2640 for (;;) { 2641#define TEE_SINK_READ 1024 2642 short buffer[TEE_SINK_READ * FCC_2]; 2643 size_t count = TEE_SINK_READ; 2644 ssize_t actual = teeSource->read(buffer, count, 2645 AudioBufferProvider::kInvalidPTS); 2646 bool wasFirstRead = firstRead; 2647 firstRead = false; 2648 if (actual <= 0) { 2649 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2650 continue; 2651 } 2652 break; 2653 } 2654 ALOG_ASSERT(actual <= (ssize_t)count); 2655 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2656 total += actual; 2657 } 2658 lseek(teeFd, (off_t) 4, SEEK_SET); 2659 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2660 write(teeFd, &temp, sizeof(temp)); 2661 lseek(teeFd, (off_t) 40, SEEK_SET); 2662 temp = total * channelCount * sizeof(short); 2663 write(teeFd, &temp, sizeof(temp)); 2664 close(teeFd); 2665 if (fd >= 0) { 2666 fdprintf(fd, "tee copied to %s\n", teePath); 2667 } 2668 } else { 2669 if (fd >= 0) { 2670 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2671 } 2672 } 2673 } 2674} 2675#endif 2676 2677// ---------------------------------------------------------------------------- 2678 2679status_t AudioFlinger::onTransact( 2680 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2681{ 2682 return BnAudioFlinger::onTransact(code, data, reply, flags); 2683} 2684 2685}; // namespace android 2686