AudioFlinger.cpp revision aaa44478a373232d8416657035a9020f9c7aa7c3
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 & AUDIO_FORMAT_MAIN_MASK) { 110 case AUDIO_FORMAT_PCM: 111 switch (format) { 112 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 113 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 114 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 115 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 116 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 117 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 118 default: 119 break; 120 } 121 break; 122 case AUDIO_FORMAT_MP3: return "mp3"; 123 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 124 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 125 case AUDIO_FORMAT_AAC: return "aac"; 126 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 127 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 128 case AUDIO_FORMAT_VORBIS: return "vorbis"; 129 case AUDIO_FORMAT_OPUS: return "opus"; 130 case AUDIO_FORMAT_AC3: return "ac-3"; 131 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 132 default: 133 break; 134 } 135 return "unknown"; 136} 137 138static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 139{ 140 const hw_module_t *mod; 141 int rc; 142 143 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 144 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 145 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 146 if (rc) { 147 goto out; 148 } 149 rc = audio_hw_device_open(mod, dev); 150 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 156 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 157 rc = BAD_VALUE; 158 goto out; 159 } 160 return 0; 161 162out: 163 *dev = NULL; 164 return rc; 165} 166 167// ---------------------------------------------------------------------------- 168 169AudioFlinger::AudioFlinger() 170 : BnAudioFlinger(), 171 mPrimaryHardwareDev(NULL), 172 mAudioHwDevs(NULL), 173 mHardwareStatus(AUDIO_HW_IDLE), 174 mMasterVolume(1.0f), 175 mMasterMute(false), 176 mNextUniqueId(1), 177 mMode(AUDIO_MODE_INVALID), 178 mBtNrecIsOff(false), 179 mIsLowRamDevice(true), 180 mIsDeviceTypeKnown(false), 181 mGlobalEffectEnableTime(0), 182 mPrimaryOutputSampleRate(0) 183{ 184 getpid_cached = getpid(); 185 char value[PROPERTY_VALUE_MAX]; 186 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 187 if (doLog) { 188 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 189 } 190 191#ifdef TEE_SINK 192 (void) property_get("ro.debuggable", value, "0"); 193 int debuggable = atoi(value); 194 int teeEnabled = 0; 195 if (debuggable) { 196 (void) property_get("af.tee", value, "0"); 197 teeEnabled = atoi(value); 198 } 199 // FIXME symbolic constants here 200 if (teeEnabled & 1) { 201 mTeeSinkInputEnabled = true; 202 } 203 if (teeEnabled & 2) { 204 mTeeSinkOutputEnabled = true; 205 } 206 if (teeEnabled & 4) { 207 mTeeSinkTrackEnabled = true; 208 } 209#endif 210} 211 212void AudioFlinger::onFirstRef() 213{ 214 int rc = 0; 215 216 Mutex::Autolock _l(mLock); 217 218 /* TODO: move all this work into an Init() function */ 219 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 220 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 221 uint32_t int_val; 222 if (1 == sscanf(val_str, "%u", &int_val)) { 223 mStandbyTimeInNsecs = milliseconds(int_val); 224 ALOGI("Using %u mSec as standby time.", int_val); 225 } else { 226 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 227 ALOGI("Using default %u mSec as standby time.", 228 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 229 } 230 } 231 232 mPatchPanel = new PatchPanel(this); 233 234 mMode = AUDIO_MODE_NORMAL; 235} 236 237AudioFlinger::~AudioFlinger() 238{ 239 while (!mRecordThreads.isEmpty()) { 240 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 241 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 242 } 243 while (!mPlaybackThreads.isEmpty()) { 244 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 245 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 246 } 247 248 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 249 // no mHardwareLock needed, as there are no other references to this 250 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 251 delete mAudioHwDevs.valueAt(i); 252 } 253 254 // Tell media.log service about any old writers that still need to be unregistered 255 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 256 if (binder != 0) { 257 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 258 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 259 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 260 mUnregisteredWriters.pop(); 261 mediaLogService->unregisterWriter(iMemory); 262 } 263 } 264 265} 266 267static const char * const audio_interfaces[] = { 268 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 269 AUDIO_HARDWARE_MODULE_ID_A2DP, 270 AUDIO_HARDWARE_MODULE_ID_USB, 271}; 272#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 273 274AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 275 audio_module_handle_t module, 276 audio_devices_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 // then try to find a module supporting the requested device. 286 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 287 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 288 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 289 if ((dev->get_supported_devices != NULL) && 290 (dev->get_supported_devices(dev) & devices) == devices) 291 return audioHwDevice; 292 } 293 } else { 294 // check a match for the requested module handle 295 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 296 if (audioHwDevice != NULL) { 297 return audioHwDevice; 298 } 299 } 300 301 return NULL; 302} 303 304void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Notification Clients:\n"); 320 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 321 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 322 result.append(buffer); 323 } 324 325 result.append("Global session refs:\n"); 326 result.append(" session pid count\n"); 327 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 328 AudioSessionRef *r = mAudioSessionRefs[i]; 329 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 330 result.append(buffer); 331 } 332 write(fd, result.string(), result.size()); 333} 334 335 336void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 337{ 338 const size_t SIZE = 256; 339 char buffer[SIZE]; 340 String8 result; 341 hardware_call_state hardwareStatus = mHardwareStatus; 342 343 snprintf(buffer, SIZE, "Hardware status: %d\n" 344 "Standby Time mSec: %u\n", 345 hardwareStatus, 346 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 347 result.append(buffer); 348 write(fd, result.string(), result.size()); 349} 350 351void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 352{ 353 const size_t SIZE = 256; 354 char buffer[SIZE]; 355 String8 result; 356 snprintf(buffer, SIZE, "Permission Denial: " 357 "can't dump AudioFlinger from pid=%d, uid=%d\n", 358 IPCThreadState::self()->getCallingPid(), 359 IPCThreadState::self()->getCallingUid()); 360 result.append(buffer); 361 write(fd, result.string(), result.size()); 362} 363 364bool AudioFlinger::dumpTryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = dumpTryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = dumpTryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 bool clientLocked = dumpTryLock(mClientLock); 400 if (!clientLocked) { 401 String8 result(kClientLockedString); 402 write(fd, result.string(), result.size()); 403 } 404 dumpClients(fd, args); 405 if (clientLocked) { 406 mClientLock.unlock(); 407 } 408 409 dumpInternals(fd, args); 410 411 // dump playback threads 412 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 413 mPlaybackThreads.valueAt(i)->dump(fd, args); 414 } 415 416 // dump record threads 417 for (size_t i = 0; i < mRecordThreads.size(); i++) { 418 mRecordThreads.valueAt(i)->dump(fd, args); 419 } 420 421 // dump orphan effect chains 422 if (mOrphanEffectChains.size() != 0) { 423 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 424 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 425 mOrphanEffectChains.valueAt(i)->dump(fd, args); 426 } 427 } 428 // dump all hardware devs 429 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 430 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 431 dev->dump(dev, fd); 432 } 433 434#ifdef TEE_SINK 435 // dump the serially shared record tee sink 436 if (mRecordTeeSource != 0) { 437 dumpTee(fd, mRecordTeeSource); 438 } 439#endif 440 441 if (locked) { 442 mLock.unlock(); 443 } 444 445 // append a copy of media.log here by forwarding fd to it, but don't attempt 446 // to lookup the service if it's not running, as it will block for a second 447 if (mLogMemoryDealer != 0) { 448 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 449 if (binder != 0) { 450 dprintf(fd, "\nmedia.log:\n"); 451 Vector<String16> args; 452 binder->dump(fd, args); 453 } 454 } 455 } 456 return NO_ERROR; 457} 458 459sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 460{ 461 Mutex::Autolock _cl(mClientLock); 462 // If pid is already in the mClients wp<> map, then use that entry 463 // (for which promote() is always != 0), otherwise create a new entry and Client. 464 sp<Client> client = mClients.valueFor(pid).promote(); 465 if (client == 0) { 466 client = new Client(this, pid); 467 mClients.add(pid, client); 468 } 469 470 return client; 471} 472 473sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 474{ 475 // If there is no memory allocated for logs, return a dummy writer that does nothing 476 if (mLogMemoryDealer == 0) { 477 return new NBLog::Writer(); 478 } 479 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 480 // Similarly if we can't contact the media.log service, also return a dummy writer 481 if (binder == 0) { 482 return new NBLog::Writer(); 483 } 484 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 485 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 486 // If allocation fails, consult the vector of previously unregistered writers 487 // and garbage-collect one or more them until an allocation succeeds 488 if (shared == 0) { 489 Mutex::Autolock _l(mUnregisteredWritersLock); 490 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 491 { 492 // Pick the oldest stale writer to garbage-collect 493 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 494 mUnregisteredWriters.removeAt(0); 495 mediaLogService->unregisterWriter(iMemory); 496 // Now the media.log remote reference to IMemory is gone. When our last local 497 // reference to IMemory also drops to zero at end of this block, 498 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 499 } 500 // Re-attempt the allocation 501 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 502 if (shared != 0) { 503 goto success; 504 } 505 } 506 // Even after garbage-collecting all old writers, there is still not enough memory, 507 // so return a dummy writer 508 return new NBLog::Writer(); 509 } 510success: 511 mediaLogService->registerWriter(shared, size, name); 512 return new NBLog::Writer(size, shared); 513} 514 515void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 516{ 517 if (writer == 0) { 518 return; 519 } 520 sp<IMemory> iMemory(writer->getIMemory()); 521 if (iMemory == 0) { 522 return; 523 } 524 // Rather than removing the writer immediately, append it to a queue of old writers to 525 // be garbage-collected later. This allows us to continue to view old logs for a while. 526 Mutex::Autolock _l(mUnregisteredWritersLock); 527 mUnregisteredWriters.push(writer); 528} 529 530// IAudioFlinger interface 531 532 533sp<IAudioTrack> AudioFlinger::createTrack( 534 audio_stream_type_t streamType, 535 uint32_t sampleRate, 536 audio_format_t format, 537 audio_channel_mask_t channelMask, 538 size_t *frameCount, 539 IAudioFlinger::track_flags_t *flags, 540 const sp<IMemory>& sharedBuffer, 541 audio_io_handle_t output, 542 pid_t tid, 543 int *sessionId, 544 int clientUid, 545 status_t *status) 546{ 547 sp<PlaybackThread::Track> track; 548 sp<TrackHandle> trackHandle; 549 sp<Client> client; 550 status_t lStatus; 551 int lSessionId; 552 553 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 554 // but if someone uses binder directly they could bypass that and cause us to crash 555 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 556 ALOGE("createTrack() invalid stream type %d", streamType); 557 lStatus = BAD_VALUE; 558 goto Exit; 559 } 560 561 // further sample rate checks are performed by createTrack_l() depending on the thread type 562 if (sampleRate == 0) { 563 ALOGE("createTrack() invalid sample rate %u", sampleRate); 564 lStatus = BAD_VALUE; 565 goto Exit; 566 } 567 568 // further channel mask checks are performed by createTrack_l() depending on the thread type 569 if (!audio_is_output_channel(channelMask)) { 570 ALOGE("createTrack() invalid channel mask %#x", channelMask); 571 lStatus = BAD_VALUE; 572 goto Exit; 573 } 574 575 // further format checks are performed by createTrack_l() depending on the thread type 576 if (!audio_is_valid_format(format)) { 577 ALOGE("createTrack() invalid format %#x", format); 578 lStatus = BAD_VALUE; 579 goto Exit; 580 } 581 582 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 583 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 584 lStatus = BAD_VALUE; 585 goto Exit; 586 } 587 588 { 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGE("no playback thread found for output handle %d", output); 593 lStatus = BAD_VALUE; 594 goto Exit; 595 } 596 597 pid_t pid = IPCThreadState::self()->getCallingPid(); 598 client = registerPid(pid); 599 600 PlaybackThread *effectThread = NULL; 601 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 602 lSessionId = *sessionId; 603 // check if an effect chain with the same session ID is present on another 604 // output thread and move it here. 605 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 606 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 607 if (mPlaybackThreads.keyAt(i) != output) { 608 uint32_t sessions = t->hasAudioSession(lSessionId); 609 if (sessions & PlaybackThread::EFFECT_SESSION) { 610 effectThread = t.get(); 611 break; 612 } 613 } 614 } 615 } else { 616 // if no audio session id is provided, create one here 617 lSessionId = nextUniqueId(); 618 if (sessionId != NULL) { 619 *sessionId = lSessionId; 620 } 621 } 622 ALOGV("createTrack() lSessionId: %d", lSessionId); 623 624 track = thread->createTrack_l(client, streamType, sampleRate, format, 625 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 626 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 627 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 628 629 // move effect chain to this output thread if an effect on same session was waiting 630 // for a track to be created 631 if (lStatus == NO_ERROR && effectThread != NULL) { 632 // no risk of deadlock because AudioFlinger::mLock is held 633 Mutex::Autolock _dl(thread->mLock); 634 Mutex::Autolock _sl(effectThread->mLock); 635 moveEffectChain_l(lSessionId, effectThread, thread, true); 636 } 637 638 // Look for sync events awaiting for a session to be used. 639 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 640 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 641 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 642 if (lStatus == NO_ERROR) { 643 (void) track->setSyncEvent(mPendingSyncEvents[i]); 644 } else { 645 mPendingSyncEvents[i]->cancel(); 646 } 647 mPendingSyncEvents.removeAt(i); 648 i--; 649 } 650 } 651 } 652 653 } 654 655 if (lStatus != NO_ERROR) { 656 // remove local strong reference to Client before deleting the Track so that the 657 // Client destructor is called by the TrackBase destructor with mClientLock held 658 // Don't hold mClientLock when releasing the reference on the track as the 659 // destructor will acquire it. 660 { 661 Mutex::Autolock _cl(mClientLock); 662 client.clear(); 663 } 664 track.clear(); 665 goto Exit; 666 } 667 668 // return handle to client 669 trackHandle = new TrackHandle(track); 670 671Exit: 672 *status = lStatus; 673 return trackHandle; 674} 675 676uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 677{ 678 Mutex::Autolock _l(mLock); 679 PlaybackThread *thread = checkPlaybackThread_l(output); 680 if (thread == NULL) { 681 ALOGW("sampleRate() unknown thread %d", output); 682 return 0; 683 } 684 return thread->sampleRate(); 685} 686 687audio_format_t AudioFlinger::format(audio_io_handle_t output) const 688{ 689 Mutex::Autolock _l(mLock); 690 PlaybackThread *thread = checkPlaybackThread_l(output); 691 if (thread == NULL) { 692 ALOGW("format() unknown thread %d", output); 693 return AUDIO_FORMAT_INVALID; 694 } 695 return thread->format(); 696} 697 698size_t AudioFlinger::frameCount(audio_io_handle_t output) const 699{ 700 Mutex::Autolock _l(mLock); 701 PlaybackThread *thread = checkPlaybackThread_l(output); 702 if (thread == NULL) { 703 ALOGW("frameCount() unknown thread %d", output); 704 return 0; 705 } 706 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 707 // should examine all callers and fix them to handle smaller counts 708 return thread->frameCount(); 709} 710 711uint32_t AudioFlinger::latency(audio_io_handle_t output) const 712{ 713 Mutex::Autolock _l(mLock); 714 PlaybackThread *thread = checkPlaybackThread_l(output); 715 if (thread == NULL) { 716 ALOGW("latency(): no playback thread found for output handle %d", output); 717 return 0; 718 } 719 return thread->latency(); 720} 721 722status_t AudioFlinger::setMasterVolume(float value) 723{ 724 status_t ret = initCheck(); 725 if (ret != NO_ERROR) { 726 return ret; 727 } 728 729 // check calling permissions 730 if (!settingsAllowed()) { 731 return PERMISSION_DENIED; 732 } 733 734 Mutex::Autolock _l(mLock); 735 mMasterVolume = value; 736 737 // Set master volume in the HALs which support it. 738 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 739 AutoMutex lock(mHardwareLock); 740 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 741 742 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 743 if (dev->canSetMasterVolume()) { 744 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 745 } 746 mHardwareStatus = AUDIO_HW_IDLE; 747 } 748 749 // Now set the master volume in each playback thread. Playback threads 750 // assigned to HALs which do not have master volume support will apply 751 // master volume during the mix operation. Threads with HALs which do 752 // support master volume will simply ignore the setting. 753 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 754 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 755 756 return NO_ERROR; 757} 758 759status_t AudioFlinger::setMode(audio_mode_t mode) 760{ 761 status_t ret = initCheck(); 762 if (ret != NO_ERROR) { 763 return ret; 764 } 765 766 // check calling permissions 767 if (!settingsAllowed()) { 768 return PERMISSION_DENIED; 769 } 770 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 771 ALOGW("Illegal value: setMode(%d)", mode); 772 return BAD_VALUE; 773 } 774 775 { // scope for the lock 776 AutoMutex lock(mHardwareLock); 777 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 778 mHardwareStatus = AUDIO_HW_SET_MODE; 779 ret = dev->set_mode(dev, mode); 780 mHardwareStatus = AUDIO_HW_IDLE; 781 } 782 783 if (NO_ERROR == ret) { 784 Mutex::Autolock _l(mLock); 785 mMode = mode; 786 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 787 mPlaybackThreads.valueAt(i)->setMode(mode); 788 } 789 790 return ret; 791} 792 793status_t AudioFlinger::setMicMute(bool state) 794{ 795 status_t ret = initCheck(); 796 if (ret != NO_ERROR) { 797 return ret; 798 } 799 800 // check calling permissions 801 if (!settingsAllowed()) { 802 return PERMISSION_DENIED; 803 } 804 805 AutoMutex lock(mHardwareLock); 806 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 807 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 808 ret = dev->set_mic_mute(dev, state); 809 mHardwareStatus = AUDIO_HW_IDLE; 810 return ret; 811} 812 813bool AudioFlinger::getMicMute() const 814{ 815 status_t ret = initCheck(); 816 if (ret != NO_ERROR) { 817 return false; 818 } 819 820 bool state = AUDIO_MODE_INVALID; 821 AutoMutex lock(mHardwareLock); 822 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 823 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 824 dev->get_mic_mute(dev, &state); 825 mHardwareStatus = AUDIO_HW_IDLE; 826 return state; 827} 828 829status_t AudioFlinger::setMasterMute(bool muted) 830{ 831 status_t ret = initCheck(); 832 if (ret != NO_ERROR) { 833 return ret; 834 } 835 836 // check calling permissions 837 if (!settingsAllowed()) { 838 return PERMISSION_DENIED; 839 } 840 841 Mutex::Autolock _l(mLock); 842 mMasterMute = muted; 843 844 // Set master mute in the HALs which support it. 845 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 846 AutoMutex lock(mHardwareLock); 847 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 848 849 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 850 if (dev->canSetMasterMute()) { 851 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 852 } 853 mHardwareStatus = AUDIO_HW_IDLE; 854 } 855 856 // Now set the master mute in each playback thread. Playback threads 857 // assigned to HALs which do not have master mute support will apply master 858 // mute during the mix operation. Threads with HALs which do support master 859 // mute will simply ignore the setting. 860 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 861 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 862 863 return NO_ERROR; 864} 865 866float AudioFlinger::masterVolume() const 867{ 868 Mutex::Autolock _l(mLock); 869 return masterVolume_l(); 870} 871 872bool AudioFlinger::masterMute() const 873{ 874 Mutex::Autolock _l(mLock); 875 return masterMute_l(); 876} 877 878float AudioFlinger::masterVolume_l() const 879{ 880 return mMasterVolume; 881} 882 883bool AudioFlinger::masterMute_l() const 884{ 885 return mMasterMute; 886} 887 888status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 889 audio_io_handle_t output) 890{ 891 // check calling permissions 892 if (!settingsAllowed()) { 893 return PERMISSION_DENIED; 894 } 895 896 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 897 ALOGE("setStreamVolume() invalid stream %d", stream); 898 return BAD_VALUE; 899 } 900 901 AutoMutex lock(mLock); 902 PlaybackThread *thread = NULL; 903 if (output != AUDIO_IO_HANDLE_NONE) { 904 thread = checkPlaybackThread_l(output); 905 if (thread == NULL) { 906 return BAD_VALUE; 907 } 908 } 909 910 mStreamTypes[stream].volume = value; 911 912 if (thread == NULL) { 913 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 914 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 915 } 916 } else { 917 thread->setStreamVolume(stream, value); 918 } 919 920 return NO_ERROR; 921} 922 923status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 924{ 925 // check calling permissions 926 if (!settingsAllowed()) { 927 return PERMISSION_DENIED; 928 } 929 930 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 931 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 932 ALOGE("setStreamMute() invalid stream %d", stream); 933 return BAD_VALUE; 934 } 935 936 AutoMutex lock(mLock); 937 mStreamTypes[stream].mute = muted; 938 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 939 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 940 941 return NO_ERROR; 942} 943 944float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 945{ 946 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 947 return 0.0f; 948 } 949 950 AutoMutex lock(mLock); 951 float volume; 952 if (output != AUDIO_IO_HANDLE_NONE) { 953 PlaybackThread *thread = checkPlaybackThread_l(output); 954 if (thread == NULL) { 955 return 0.0f; 956 } 957 volume = thread->streamVolume(stream); 958 } else { 959 volume = streamVolume_l(stream); 960 } 961 962 return volume; 963} 964 965bool AudioFlinger::streamMute(audio_stream_type_t stream) const 966{ 967 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 968 return true; 969 } 970 971 AutoMutex lock(mLock); 972 return streamMute_l(stream); 973} 974 975status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 976{ 977 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 978 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 979 980 // check calling permissions 981 if (!settingsAllowed()) { 982 return PERMISSION_DENIED; 983 } 984 985 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 986 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 987 Mutex::Autolock _l(mLock); 988 status_t final_result = NO_ERROR; 989 { 990 AutoMutex lock(mHardwareLock); 991 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 992 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 993 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 994 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 995 final_result = result ?: final_result; 996 } 997 mHardwareStatus = AUDIO_HW_IDLE; 998 } 999 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1000 AudioParameter param = AudioParameter(keyValuePairs); 1001 String8 value; 1002 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1003 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1004 if (mBtNrecIsOff != btNrecIsOff) { 1005 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1006 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1007 audio_devices_t device = thread->inDevice(); 1008 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1009 // collect all of the thread's session IDs 1010 KeyedVector<int, bool> ids = thread->sessionIds(); 1011 // suspend effects associated with those session IDs 1012 for (size_t j = 0; j < ids.size(); ++j) { 1013 int sessionId = ids.keyAt(j); 1014 thread->setEffectSuspended(FX_IID_AEC, 1015 suspend, 1016 sessionId); 1017 thread->setEffectSuspended(FX_IID_NS, 1018 suspend, 1019 sessionId); 1020 } 1021 } 1022 mBtNrecIsOff = btNrecIsOff; 1023 } 1024 } 1025 String8 screenState; 1026 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1027 bool isOff = screenState == "off"; 1028 if (isOff != (AudioFlinger::mScreenState & 1)) { 1029 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1030 } 1031 } 1032 return final_result; 1033 } 1034 1035 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1036 // and the thread is exited once the lock is released 1037 sp<ThreadBase> thread; 1038 { 1039 Mutex::Autolock _l(mLock); 1040 thread = checkPlaybackThread_l(ioHandle); 1041 if (thread == 0) { 1042 thread = checkRecordThread_l(ioHandle); 1043 } else if (thread == primaryPlaybackThread_l()) { 1044 // indicate output device change to all input threads for pre processing 1045 AudioParameter param = AudioParameter(keyValuePairs); 1046 int value; 1047 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1048 (value != 0)) { 1049 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1050 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1051 } 1052 } 1053 } 1054 } 1055 if (thread != 0) { 1056 return thread->setParameters(keyValuePairs); 1057 } 1058 return BAD_VALUE; 1059} 1060 1061String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1062{ 1063 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1064 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1065 1066 Mutex::Autolock _l(mLock); 1067 1068 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1069 String8 out_s8; 1070 1071 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1072 char *s; 1073 { 1074 AutoMutex lock(mHardwareLock); 1075 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1076 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1077 s = dev->get_parameters(dev, keys.string()); 1078 mHardwareStatus = AUDIO_HW_IDLE; 1079 } 1080 out_s8 += String8(s ? s : ""); 1081 free(s); 1082 } 1083 return out_s8; 1084 } 1085 1086 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1087 if (playbackThread != NULL) { 1088 return playbackThread->getParameters(keys); 1089 } 1090 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1091 if (recordThread != NULL) { 1092 return recordThread->getParameters(keys); 1093 } 1094 return String8(""); 1095} 1096 1097size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1098 audio_channel_mask_t channelMask) const 1099{ 1100 status_t ret = initCheck(); 1101 if (ret != NO_ERROR) { 1102 return 0; 1103 } 1104 1105 AutoMutex lock(mHardwareLock); 1106 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1107 audio_config_t config; 1108 memset(&config, 0, sizeof(config)); 1109 config.sample_rate = sampleRate; 1110 config.channel_mask = channelMask; 1111 config.format = format; 1112 1113 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1114 size_t size = dev->get_input_buffer_size(dev, &config); 1115 mHardwareStatus = AUDIO_HW_IDLE; 1116 return size; 1117} 1118 1119uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1120{ 1121 Mutex::Autolock _l(mLock); 1122 1123 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1124 if (recordThread != NULL) { 1125 return recordThread->getInputFramesLost(); 1126 } 1127 return 0; 1128} 1129 1130status_t AudioFlinger::setVoiceVolume(float value) 1131{ 1132 status_t ret = initCheck(); 1133 if (ret != NO_ERROR) { 1134 return ret; 1135 } 1136 1137 // check calling permissions 1138 if (!settingsAllowed()) { 1139 return PERMISSION_DENIED; 1140 } 1141 1142 AutoMutex lock(mHardwareLock); 1143 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1144 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1145 ret = dev->set_voice_volume(dev, value); 1146 mHardwareStatus = AUDIO_HW_IDLE; 1147 1148 return ret; 1149} 1150 1151status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1152 audio_io_handle_t output) const 1153{ 1154 status_t status; 1155 1156 Mutex::Autolock _l(mLock); 1157 1158 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1159 if (playbackThread != NULL) { 1160 return playbackThread->getRenderPosition(halFrames, dspFrames); 1161 } 1162 1163 return BAD_VALUE; 1164} 1165 1166void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1167{ 1168 Mutex::Autolock _l(mLock); 1169 if (client == 0) { 1170 return; 1171 } 1172 bool clientAdded = false; 1173 { 1174 Mutex::Autolock _cl(mClientLock); 1175 1176 pid_t pid = IPCThreadState::self()->getCallingPid(); 1177 if (mNotificationClients.indexOfKey(pid) < 0) { 1178 sp<NotificationClient> notificationClient = new NotificationClient(this, 1179 client, 1180 pid); 1181 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1182 1183 mNotificationClients.add(pid, notificationClient); 1184 1185 sp<IBinder> binder = client->asBinder(); 1186 binder->linkToDeath(notificationClient); 1187 clientAdded = true; 1188 } 1189 } 1190 1191 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1192 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1193 if (clientAdded) { 1194 // the config change is always sent from playback or record threads to avoid deadlock 1195 // with AudioSystem::gLock 1196 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1197 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1198 } 1199 1200 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1201 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1202 } 1203 } 1204} 1205 1206void AudioFlinger::removeNotificationClient(pid_t pid) 1207{ 1208 Mutex::Autolock _l(mLock); 1209 { 1210 Mutex::Autolock _cl(mClientLock); 1211 mNotificationClients.removeItem(pid); 1212 } 1213 1214 ALOGV("%d died, releasing its sessions", pid); 1215 size_t num = mAudioSessionRefs.size(); 1216 bool removed = false; 1217 for (size_t i = 0; i< num; ) { 1218 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1219 ALOGV(" pid %d @ %d", ref->mPid, i); 1220 if (ref->mPid == pid) { 1221 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1222 mAudioSessionRefs.removeAt(i); 1223 delete ref; 1224 removed = true; 1225 num--; 1226 } else { 1227 i++; 1228 } 1229 } 1230 if (removed) { 1231 purgeStaleEffects_l(); 1232 } 1233} 1234 1235void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1236{ 1237 Mutex::Autolock _l(mClientLock); 1238 size_t size = mNotificationClients.size(); 1239 for (size_t i = 0; i < size; i++) { 1240 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1241 ioHandle, 1242 param2); 1243 } 1244} 1245 1246// removeClient_l() must be called with AudioFlinger::mClientLock held 1247void AudioFlinger::removeClient_l(pid_t pid) 1248{ 1249 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1250 IPCThreadState::self()->getCallingPid()); 1251 mClients.removeItem(pid); 1252} 1253 1254// getEffectThread_l() must be called with AudioFlinger::mLock held 1255sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1256{ 1257 sp<PlaybackThread> thread; 1258 1259 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1260 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1261 ALOG_ASSERT(thread == 0); 1262 thread = mPlaybackThreads.valueAt(i); 1263 } 1264 } 1265 1266 return thread; 1267} 1268 1269 1270 1271// ---------------------------------------------------------------------------- 1272 1273AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1274 : RefBase(), 1275 mAudioFlinger(audioFlinger), 1276 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1277 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1278 mPid(pid), 1279 mTimedTrackCount(0) 1280{ 1281 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1282} 1283 1284// Client destructor must be called with AudioFlinger::mClientLock held 1285AudioFlinger::Client::~Client() 1286{ 1287 mAudioFlinger->removeClient_l(mPid); 1288} 1289 1290sp<MemoryDealer> AudioFlinger::Client::heap() const 1291{ 1292 return mMemoryDealer; 1293} 1294 1295// Reserve one of the limited slots for a timed audio track associated 1296// with this client 1297bool AudioFlinger::Client::reserveTimedTrack() 1298{ 1299 const int kMaxTimedTracksPerClient = 4; 1300 1301 Mutex::Autolock _l(mTimedTrackLock); 1302 1303 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1304 ALOGW("can not create timed track - pid %d has exceeded the limit", 1305 mPid); 1306 return false; 1307 } 1308 1309 mTimedTrackCount++; 1310 return true; 1311} 1312 1313// Release a slot for a timed audio track 1314void AudioFlinger::Client::releaseTimedTrack() 1315{ 1316 Mutex::Autolock _l(mTimedTrackLock); 1317 mTimedTrackCount--; 1318} 1319 1320// ---------------------------------------------------------------------------- 1321 1322AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1323 const sp<IAudioFlingerClient>& client, 1324 pid_t pid) 1325 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1326{ 1327} 1328 1329AudioFlinger::NotificationClient::~NotificationClient() 1330{ 1331} 1332 1333void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1334{ 1335 sp<NotificationClient> keep(this); 1336 mAudioFlinger->removeNotificationClient(mPid); 1337} 1338 1339 1340// ---------------------------------------------------------------------------- 1341 1342static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1343 return audio_is_remote_submix_device(inDevice); 1344} 1345 1346sp<IAudioRecord> AudioFlinger::openRecord( 1347 audio_io_handle_t input, 1348 uint32_t sampleRate, 1349 audio_format_t format, 1350 audio_channel_mask_t channelMask, 1351 size_t *frameCount, 1352 IAudioFlinger::track_flags_t *flags, 1353 pid_t tid, 1354 int *sessionId, 1355 size_t *notificationFrames, 1356 sp<IMemory>& cblk, 1357 sp<IMemory>& buffers, 1358 status_t *status) 1359{ 1360 sp<RecordThread::RecordTrack> recordTrack; 1361 sp<RecordHandle> recordHandle; 1362 sp<Client> client; 1363 status_t lStatus; 1364 int lSessionId; 1365 1366 cblk.clear(); 1367 buffers.clear(); 1368 1369 // check calling permissions 1370 if (!recordingAllowed()) { 1371 ALOGE("openRecord() permission denied: recording not allowed"); 1372 lStatus = PERMISSION_DENIED; 1373 goto Exit; 1374 } 1375 1376 // further sample rate checks are performed by createRecordTrack_l() 1377 if (sampleRate == 0) { 1378 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1379 lStatus = BAD_VALUE; 1380 goto Exit; 1381 } 1382 1383 // we don't yet support anything other than 16-bit PCM 1384 if (!(audio_is_valid_format(format) && 1385 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1386 ALOGE("openRecord() invalid format %#x", format); 1387 lStatus = BAD_VALUE; 1388 goto Exit; 1389 } 1390 1391 // further channel mask checks are performed by createRecordTrack_l() 1392 if (!audio_is_input_channel(channelMask)) { 1393 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1394 lStatus = BAD_VALUE; 1395 goto Exit; 1396 } 1397 1398 { 1399 Mutex::Autolock _l(mLock); 1400 RecordThread *thread = checkRecordThread_l(input); 1401 if (thread == NULL) { 1402 ALOGE("openRecord() checkRecordThread_l failed"); 1403 lStatus = BAD_VALUE; 1404 goto Exit; 1405 } 1406 1407 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1408 && !captureAudioOutputAllowed()) { 1409 ALOGE("openRecord() permission denied: capture not allowed"); 1410 lStatus = PERMISSION_DENIED; 1411 goto Exit; 1412 } 1413 1414 pid_t pid = IPCThreadState::self()->getCallingPid(); 1415 client = registerPid(pid); 1416 1417 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1418 lSessionId = *sessionId; 1419 } else { 1420 // if no audio session id is provided, create one here 1421 lSessionId = nextUniqueId(); 1422 if (sessionId != NULL) { 1423 *sessionId = lSessionId; 1424 } 1425 } 1426 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1427 1428 // TODO: the uid should be passed in as a parameter to openRecord 1429 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1430 frameCount, lSessionId, notificationFrames, 1431 IPCThreadState::self()->getCallingUid(), 1432 flags, tid, &lStatus); 1433 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1434 } 1435 1436 if (lStatus != NO_ERROR) { 1437 // remove local strong reference to Client before deleting the RecordTrack so that the 1438 // Client destructor is called by the TrackBase destructor with mClientLock held 1439 // Don't hold mClientLock when releasing the reference on the track as the 1440 // destructor will acquire it. 1441 { 1442 Mutex::Autolock _cl(mClientLock); 1443 client.clear(); 1444 } 1445 recordTrack.clear(); 1446 goto Exit; 1447 } 1448 1449 cblk = recordTrack->getCblk(); 1450 buffers = recordTrack->getBuffers(); 1451 1452 // return handle to client 1453 recordHandle = new RecordHandle(recordTrack); 1454 1455Exit: 1456 *status = lStatus; 1457 return recordHandle; 1458} 1459 1460 1461 1462// ---------------------------------------------------------------------------- 1463 1464audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1465{ 1466 if (name == NULL) { 1467 return 0; 1468 } 1469 if (!settingsAllowed()) { 1470 return 0; 1471 } 1472 Mutex::Autolock _l(mLock); 1473 return loadHwModule_l(name); 1474} 1475 1476// loadHwModule_l() must be called with AudioFlinger::mLock held 1477audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1478{ 1479 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1480 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1481 ALOGW("loadHwModule() module %s already loaded", name); 1482 return mAudioHwDevs.keyAt(i); 1483 } 1484 } 1485 1486 audio_hw_device_t *dev; 1487 1488 int rc = load_audio_interface(name, &dev); 1489 if (rc) { 1490 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1491 return 0; 1492 } 1493 1494 mHardwareStatus = AUDIO_HW_INIT; 1495 rc = dev->init_check(dev); 1496 mHardwareStatus = AUDIO_HW_IDLE; 1497 if (rc) { 1498 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1499 return 0; 1500 } 1501 1502 // Check and cache this HAL's level of support for master mute and master 1503 // volume. If this is the first HAL opened, and it supports the get 1504 // methods, use the initial values provided by the HAL as the current 1505 // master mute and volume settings. 1506 1507 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1508 { // scope for auto-lock pattern 1509 AutoMutex lock(mHardwareLock); 1510 1511 if (0 == mAudioHwDevs.size()) { 1512 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1513 if (NULL != dev->get_master_volume) { 1514 float mv; 1515 if (OK == dev->get_master_volume(dev, &mv)) { 1516 mMasterVolume = mv; 1517 } 1518 } 1519 1520 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1521 if (NULL != dev->get_master_mute) { 1522 bool mm; 1523 if (OK == dev->get_master_mute(dev, &mm)) { 1524 mMasterMute = mm; 1525 } 1526 } 1527 } 1528 1529 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1530 if ((NULL != dev->set_master_volume) && 1531 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1532 flags = static_cast<AudioHwDevice::Flags>(flags | 1533 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1534 } 1535 1536 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1537 if ((NULL != dev->set_master_mute) && 1538 (OK == dev->set_master_mute(dev, mMasterMute))) { 1539 flags = static_cast<AudioHwDevice::Flags>(flags | 1540 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1541 } 1542 1543 mHardwareStatus = AUDIO_HW_IDLE; 1544 } 1545 1546 audio_module_handle_t handle = nextUniqueId(); 1547 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1548 1549 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1550 name, dev->common.module->name, dev->common.module->id, handle); 1551 1552 return handle; 1553 1554} 1555 1556// ---------------------------------------------------------------------------- 1557 1558uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1559{ 1560 Mutex::Autolock _l(mLock); 1561 PlaybackThread *thread = primaryPlaybackThread_l(); 1562 return thread != NULL ? thread->sampleRate() : 0; 1563} 1564 1565size_t AudioFlinger::getPrimaryOutputFrameCount() 1566{ 1567 Mutex::Autolock _l(mLock); 1568 PlaybackThread *thread = primaryPlaybackThread_l(); 1569 return thread != NULL ? thread->frameCountHAL() : 0; 1570} 1571 1572// ---------------------------------------------------------------------------- 1573 1574status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1575{ 1576 uid_t uid = IPCThreadState::self()->getCallingUid(); 1577 if (uid != AID_SYSTEM) { 1578 return PERMISSION_DENIED; 1579 } 1580 Mutex::Autolock _l(mLock); 1581 if (mIsDeviceTypeKnown) { 1582 return INVALID_OPERATION; 1583 } 1584 mIsLowRamDevice = isLowRamDevice; 1585 mIsDeviceTypeKnown = true; 1586 return NO_ERROR; 1587} 1588 1589audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1590{ 1591 Mutex::Autolock _l(mLock); 1592 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1593 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1594 if ((thread->hasAudioSession(sessionId) & ThreadBase::TRACK_SESSION) != 0) { 1595 // A session can only be on one thread, so exit after first match 1596 String8 reply = thread->getParameters(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC)); 1597 AudioParameter param = AudioParameter(reply); 1598 int value; 1599 if (param.getInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value) == NO_ERROR) { 1600 return value; 1601 } 1602 break; 1603 } 1604 } 1605 return AUDIO_HW_SYNC_INVALID; 1606} 1607 1608// ---------------------------------------------------------------------------- 1609 1610 1611sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1612 audio_io_handle_t *output, 1613 audio_config_t *config, 1614 audio_devices_t devices, 1615 const String8& address, 1616 audio_output_flags_t flags) 1617{ 1618 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1619 if (outHwDev == NULL) { 1620 return 0; 1621 } 1622 1623 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1624 if (*output == AUDIO_IO_HANDLE_NONE) { 1625 *output = nextUniqueId(); 1626 } 1627 1628 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1629 1630 audio_stream_out_t *outStream = NULL; 1631 1632 // FOR TESTING ONLY: 1633 // This if statement allows overriding the audio policy settings 1634 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1635 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1636 // Check only for Normal Mixing mode 1637 if (kEnableExtendedPrecision) { 1638 // Specify format (uncomment one below to choose) 1639 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1640 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1641 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1642 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1643 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1644 } 1645 if (kEnableExtendedChannels) { 1646 // Specify channel mask (uncomment one below to choose) 1647 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1648 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1649 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1650 } 1651 } 1652 1653 status_t status = hwDevHal->open_output_stream(hwDevHal, 1654 *output, 1655 devices, 1656 flags, 1657 config, 1658 &outStream, 1659 address.string()); 1660 1661 mHardwareStatus = AUDIO_HW_IDLE; 1662 ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, " 1663 "channelMask %#x, status %d", 1664 outStream, 1665 config->sample_rate, 1666 config->format, 1667 config->channel_mask, 1668 status); 1669 1670 if (status == NO_ERROR && outStream != NULL) { 1671 AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags); 1672 1673 PlaybackThread *thread; 1674 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1675 thread = new OffloadThread(this, outputStream, *output, devices); 1676 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1677 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1678 || !isValidPcmSinkFormat(config->format) 1679 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1680 thread = new DirectOutputThread(this, outputStream, *output, devices); 1681 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1682 } else { 1683 thread = new MixerThread(this, outputStream, *output, devices); 1684 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1685 } 1686 mPlaybackThreads.add(*output, thread); 1687 return thread; 1688 } 1689 1690 return 0; 1691} 1692 1693status_t AudioFlinger::openOutput(audio_module_handle_t module, 1694 audio_io_handle_t *output, 1695 audio_config_t *config, 1696 audio_devices_t *devices, 1697 const String8& address, 1698 uint32_t *latencyMs, 1699 audio_output_flags_t flags) 1700{ 1701 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1702 module, 1703 (devices != NULL) ? *devices : 0, 1704 config->sample_rate, 1705 config->format, 1706 config->channel_mask, 1707 flags); 1708 1709 if (*devices == AUDIO_DEVICE_NONE) { 1710 return BAD_VALUE; 1711 } 1712 1713 Mutex::Autolock _l(mLock); 1714 1715 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1716 if (thread != 0) { 1717 *latencyMs = thread->latency(); 1718 1719 // notify client processes of the new output creation 1720 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1721 1722 // the first primary output opened designates the primary hw device 1723 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1724 ALOGI("Using module %d has the primary audio interface", module); 1725 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1726 1727 AutoMutex lock(mHardwareLock); 1728 mHardwareStatus = AUDIO_HW_SET_MODE; 1729 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1730 mHardwareStatus = AUDIO_HW_IDLE; 1731 1732 mPrimaryOutputSampleRate = config->sample_rate; 1733 } 1734 return NO_ERROR; 1735 } 1736 1737 return NO_INIT; 1738} 1739 1740audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1741 audio_io_handle_t output2) 1742{ 1743 Mutex::Autolock _l(mLock); 1744 MixerThread *thread1 = checkMixerThread_l(output1); 1745 MixerThread *thread2 = checkMixerThread_l(output2); 1746 1747 if (thread1 == NULL || thread2 == NULL) { 1748 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1749 output2); 1750 return AUDIO_IO_HANDLE_NONE; 1751 } 1752 1753 audio_io_handle_t id = nextUniqueId(); 1754 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1755 thread->addOutputTrack(thread2); 1756 mPlaybackThreads.add(id, thread); 1757 // notify client processes of the new output creation 1758 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1759 return id; 1760} 1761 1762status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1763{ 1764 return closeOutput_nonvirtual(output); 1765} 1766 1767status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1768{ 1769 // keep strong reference on the playback thread so that 1770 // it is not destroyed while exit() is executed 1771 sp<PlaybackThread> thread; 1772 { 1773 Mutex::Autolock _l(mLock); 1774 thread = checkPlaybackThread_l(output); 1775 if (thread == NULL) { 1776 return BAD_VALUE; 1777 } 1778 1779 ALOGV("closeOutput() %d", output); 1780 1781 if (thread->type() == ThreadBase::MIXER) { 1782 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1783 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1784 DuplicatingThread *dupThread = 1785 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1786 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1787 1788 } 1789 } 1790 } 1791 1792 1793 mPlaybackThreads.removeItem(output); 1794 // save all effects to the default thread 1795 if (mPlaybackThreads.size()) { 1796 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1797 if (dstThread != NULL) { 1798 // audioflinger lock is held here so the acquisition order of thread locks does not 1799 // matter 1800 Mutex::Autolock _dl(dstThread->mLock); 1801 Mutex::Autolock _sl(thread->mLock); 1802 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1803 for (size_t i = 0; i < effectChains.size(); i ++) { 1804 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1805 } 1806 } 1807 } 1808 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1809 } 1810 thread->exit(); 1811 // The thread entity (active unit of execution) is no longer running here, 1812 // but the ThreadBase container still exists. 1813 1814 if (thread->type() != ThreadBase::DUPLICATING) { 1815 closeOutputFinish(thread); 1816 } 1817 1818 return NO_ERROR; 1819} 1820 1821void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1822{ 1823 AudioStreamOut *out = thread->clearOutput(); 1824 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1825 // from now on thread->mOutput is NULL 1826 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1827 delete out; 1828} 1829 1830void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1831{ 1832 mPlaybackThreads.removeItem(thread->mId); 1833 thread->exit(); 1834 closeOutputFinish(thread); 1835} 1836 1837status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1838{ 1839 Mutex::Autolock _l(mLock); 1840 PlaybackThread *thread = checkPlaybackThread_l(output); 1841 1842 if (thread == NULL) { 1843 return BAD_VALUE; 1844 } 1845 1846 ALOGV("suspendOutput() %d", output); 1847 thread->suspend(); 1848 1849 return NO_ERROR; 1850} 1851 1852status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1853{ 1854 Mutex::Autolock _l(mLock); 1855 PlaybackThread *thread = checkPlaybackThread_l(output); 1856 1857 if (thread == NULL) { 1858 return BAD_VALUE; 1859 } 1860 1861 ALOGV("restoreOutput() %d", output); 1862 1863 thread->restore(); 1864 1865 return NO_ERROR; 1866} 1867 1868status_t AudioFlinger::openInput(audio_module_handle_t module, 1869 audio_io_handle_t *input, 1870 audio_config_t *config, 1871 audio_devices_t *device, 1872 const String8& address, 1873 audio_source_t source, 1874 audio_input_flags_t flags) 1875{ 1876 Mutex::Autolock _l(mLock); 1877 1878 if (*device == AUDIO_DEVICE_NONE) { 1879 return BAD_VALUE; 1880 } 1881 1882 sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags); 1883 1884 if (thread != 0) { 1885 // notify client processes of the new input creation 1886 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1887 return NO_ERROR; 1888 } 1889 return NO_INIT; 1890} 1891 1892sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 1893 audio_io_handle_t *input, 1894 audio_config_t *config, 1895 audio_devices_t device, 1896 const String8& address, 1897 audio_source_t source, 1898 audio_input_flags_t flags) 1899{ 1900 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device); 1901 if (inHwDev == NULL) { 1902 *input = AUDIO_IO_HANDLE_NONE; 1903 return 0; 1904 } 1905 1906 if (*input == AUDIO_IO_HANDLE_NONE) { 1907 *input = nextUniqueId(); 1908 } 1909 1910 audio_config_t halconfig = *config; 1911 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1912 audio_stream_in_t *inStream = NULL; 1913 status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1914 &inStream, flags, address.string(), source); 1915 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 1916 ", Format %#x, Channels %x, flags %#x, status %d", 1917 inStream, 1918 halconfig.sample_rate, 1919 halconfig.format, 1920 halconfig.channel_mask, 1921 flags, 1922 status); 1923 1924 // If the input could not be opened with the requested parameters and we can handle the 1925 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1926 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1927 if (status == BAD_VALUE && 1928 config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT && 1929 (halconfig.sample_rate <= 2 * config->sample_rate) && 1930 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 1931 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 1932 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1933 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 1934 inStream = NULL; 1935 status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1936 &inStream, flags, address.string(), source); 1937 // FIXME log this new status; HAL should not propose any further changes 1938 } 1939 1940 if (status == NO_ERROR && inStream != NULL) { 1941 1942#ifdef TEE_SINK 1943 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1944 // or (re-)create if current Pipe is idle and does not match the new format 1945 sp<NBAIO_Sink> teeSink; 1946 enum { 1947 TEE_SINK_NO, // don't copy input 1948 TEE_SINK_NEW, // copy input using a new pipe 1949 TEE_SINK_OLD, // copy input using an existing pipe 1950 } kind; 1951 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 1952 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 1953 if (!mTeeSinkInputEnabled) { 1954 kind = TEE_SINK_NO; 1955 } else if (!Format_isValid(format)) { 1956 kind = TEE_SINK_NO; 1957 } else if (mRecordTeeSink == 0) { 1958 kind = TEE_SINK_NEW; 1959 } else if (mRecordTeeSink->getStrongCount() != 1) { 1960 kind = TEE_SINK_NO; 1961 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1962 kind = TEE_SINK_OLD; 1963 } else { 1964 kind = TEE_SINK_NEW; 1965 } 1966 switch (kind) { 1967 case TEE_SINK_NEW: { 1968 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1969 size_t numCounterOffers = 0; 1970 const NBAIO_Format offers[1] = {format}; 1971 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1972 ALOG_ASSERT(index == 0); 1973 PipeReader *pipeReader = new PipeReader(*pipe); 1974 numCounterOffers = 0; 1975 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1976 ALOG_ASSERT(index == 0); 1977 mRecordTeeSink = pipe; 1978 mRecordTeeSource = pipeReader; 1979 teeSink = pipe; 1980 } 1981 break; 1982 case TEE_SINK_OLD: 1983 teeSink = mRecordTeeSink; 1984 break; 1985 case TEE_SINK_NO: 1986 default: 1987 break; 1988 } 1989#endif 1990 1991 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 1992 1993 // Start record thread 1994 // RecordThread requires both input and output device indication to forward to audio 1995 // pre processing modules 1996 sp<RecordThread> thread = new RecordThread(this, 1997 inputStream, 1998 *input, 1999 primaryOutputDevice_l(), 2000 device 2001#ifdef TEE_SINK 2002 , teeSink 2003#endif 2004 ); 2005 mRecordThreads.add(*input, thread); 2006 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2007 return thread; 2008 } 2009 2010 *input = AUDIO_IO_HANDLE_NONE; 2011 return 0; 2012} 2013 2014status_t AudioFlinger::closeInput(audio_io_handle_t input) 2015{ 2016 return closeInput_nonvirtual(input); 2017} 2018 2019status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2020{ 2021 // keep strong reference on the record thread so that 2022 // it is not destroyed while exit() is executed 2023 sp<RecordThread> thread; 2024 { 2025 Mutex::Autolock _l(mLock); 2026 thread = checkRecordThread_l(input); 2027 if (thread == 0) { 2028 return BAD_VALUE; 2029 } 2030 2031 ALOGV("closeInput() %d", input); 2032 { 2033 // If we still have effect chains, it means that a client still holds a handle 2034 // on at least one effect. We must keep the chain alive in case a new record 2035 // thread is opened for a new capture on the same session 2036 Mutex::Autolock _sl(thread->mLock); 2037 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2038 for (size_t i = 0; i < effectChains.size(); i++) { 2039 putOrphanEffectChain_l(effectChains[i]); 2040 } 2041 } 2042 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 2043 mRecordThreads.removeItem(input); 2044 } 2045 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2046 // we have a different lock for notification client 2047 closeInputFinish(thread); 2048 return NO_ERROR; 2049} 2050 2051void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2052{ 2053 thread->exit(); 2054 AudioStreamIn *in = thread->clearInput(); 2055 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2056 // from now on thread->mInput is NULL 2057 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2058 delete in; 2059} 2060 2061void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2062{ 2063 mRecordThreads.removeItem(thread->mId); 2064 closeInputFinish(thread); 2065} 2066 2067status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2068{ 2069 Mutex::Autolock _l(mLock); 2070 ALOGV("invalidateStream() stream %d", stream); 2071 2072 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2073 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2074 thread->invalidateTracks(stream); 2075 } 2076 2077 return NO_ERROR; 2078} 2079 2080 2081audio_unique_id_t AudioFlinger::newAudioUniqueId() 2082{ 2083 return nextUniqueId(); 2084} 2085 2086void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2087{ 2088 Mutex::Autolock _l(mLock); 2089 pid_t caller = IPCThreadState::self()->getCallingPid(); 2090 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2091 if (pid != -1 && (caller == getpid_cached)) { 2092 caller = pid; 2093 } 2094 2095 { 2096 Mutex::Autolock _cl(mClientLock); 2097 // Ignore requests received from processes not known as notification client. The request 2098 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2099 // called from a different pid leaving a stale session reference. Also we don't know how 2100 // to clear this reference if the client process dies. 2101 if (mNotificationClients.indexOfKey(caller) < 0) { 2102 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2103 return; 2104 } 2105 } 2106 2107 size_t num = mAudioSessionRefs.size(); 2108 for (size_t i = 0; i< num; i++) { 2109 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2110 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2111 ref->mCnt++; 2112 ALOGV(" incremented refcount to %d", ref->mCnt); 2113 return; 2114 } 2115 } 2116 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2117 ALOGV(" added new entry for %d", audioSession); 2118} 2119 2120void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2121{ 2122 Mutex::Autolock _l(mLock); 2123 pid_t caller = IPCThreadState::self()->getCallingPid(); 2124 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2125 if (pid != -1 && (caller == getpid_cached)) { 2126 caller = pid; 2127 } 2128 size_t num = mAudioSessionRefs.size(); 2129 for (size_t i = 0; i< num; i++) { 2130 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2131 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2132 ref->mCnt--; 2133 ALOGV(" decremented refcount to %d", ref->mCnt); 2134 if (ref->mCnt == 0) { 2135 mAudioSessionRefs.removeAt(i); 2136 delete ref; 2137 purgeStaleEffects_l(); 2138 } 2139 return; 2140 } 2141 } 2142 // If the caller is mediaserver it is likely that the session being released was acquired 2143 // on behalf of a process not in notification clients and we ignore the warning. 2144 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2145} 2146 2147void AudioFlinger::purgeStaleEffects_l() { 2148 2149 ALOGV("purging stale effects"); 2150 2151 Vector< sp<EffectChain> > chains; 2152 2153 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2154 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2155 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2156 sp<EffectChain> ec = t->mEffectChains[j]; 2157 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2158 chains.push(ec); 2159 } 2160 } 2161 } 2162 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2163 sp<RecordThread> t = mRecordThreads.valueAt(i); 2164 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2165 sp<EffectChain> ec = t->mEffectChains[j]; 2166 chains.push(ec); 2167 } 2168 } 2169 2170 for (size_t i = 0; i < chains.size(); i++) { 2171 sp<EffectChain> ec = chains[i]; 2172 int sessionid = ec->sessionId(); 2173 sp<ThreadBase> t = ec->mThread.promote(); 2174 if (t == 0) { 2175 continue; 2176 } 2177 size_t numsessionrefs = mAudioSessionRefs.size(); 2178 bool found = false; 2179 for (size_t k = 0; k < numsessionrefs; k++) { 2180 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2181 if (ref->mSessionid == sessionid) { 2182 ALOGV(" session %d still exists for %d with %d refs", 2183 sessionid, ref->mPid, ref->mCnt); 2184 found = true; 2185 break; 2186 } 2187 } 2188 if (!found) { 2189 Mutex::Autolock _l(t->mLock); 2190 // remove all effects from the chain 2191 while (ec->mEffects.size()) { 2192 sp<EffectModule> effect = ec->mEffects[0]; 2193 effect->unPin(); 2194 t->removeEffect_l(effect); 2195 if (effect->purgeHandles()) { 2196 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2197 } 2198 AudioSystem::unregisterEffect(effect->id()); 2199 } 2200 } 2201 } 2202 return; 2203} 2204 2205// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2206AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2207{ 2208 return mPlaybackThreads.valueFor(output).get(); 2209} 2210 2211// checkMixerThread_l() must be called with AudioFlinger::mLock held 2212AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2213{ 2214 PlaybackThread *thread = checkPlaybackThread_l(output); 2215 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2216} 2217 2218// checkRecordThread_l() must be called with AudioFlinger::mLock held 2219AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2220{ 2221 return mRecordThreads.valueFor(input).get(); 2222} 2223 2224uint32_t AudioFlinger::nextUniqueId() 2225{ 2226 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2227} 2228 2229AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2230{ 2231 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2232 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2233 AudioStreamOut *output = thread->getOutput(); 2234 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2235 return thread; 2236 } 2237 } 2238 return NULL; 2239} 2240 2241audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2242{ 2243 PlaybackThread *thread = primaryPlaybackThread_l(); 2244 2245 if (thread == NULL) { 2246 return 0; 2247 } 2248 2249 return thread->outDevice(); 2250} 2251 2252sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2253 int triggerSession, 2254 int listenerSession, 2255 sync_event_callback_t callBack, 2256 wp<RefBase> cookie) 2257{ 2258 Mutex::Autolock _l(mLock); 2259 2260 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2261 status_t playStatus = NAME_NOT_FOUND; 2262 status_t recStatus = NAME_NOT_FOUND; 2263 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2264 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2265 if (playStatus == NO_ERROR) { 2266 return event; 2267 } 2268 } 2269 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2270 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2271 if (recStatus == NO_ERROR) { 2272 return event; 2273 } 2274 } 2275 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2276 mPendingSyncEvents.add(event); 2277 } else { 2278 ALOGV("createSyncEvent() invalid event %d", event->type()); 2279 event.clear(); 2280 } 2281 return event; 2282} 2283 2284// ---------------------------------------------------------------------------- 2285// Effect management 2286// ---------------------------------------------------------------------------- 2287 2288 2289status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2290{ 2291 Mutex::Autolock _l(mLock); 2292 return EffectQueryNumberEffects(numEffects); 2293} 2294 2295status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2296{ 2297 Mutex::Autolock _l(mLock); 2298 return EffectQueryEffect(index, descriptor); 2299} 2300 2301status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2302 effect_descriptor_t *descriptor) const 2303{ 2304 Mutex::Autolock _l(mLock); 2305 return EffectGetDescriptor(pUuid, descriptor); 2306} 2307 2308 2309sp<IEffect> AudioFlinger::createEffect( 2310 effect_descriptor_t *pDesc, 2311 const sp<IEffectClient>& effectClient, 2312 int32_t priority, 2313 audio_io_handle_t io, 2314 int sessionId, 2315 status_t *status, 2316 int *id, 2317 int *enabled) 2318{ 2319 status_t lStatus = NO_ERROR; 2320 sp<EffectHandle> handle; 2321 effect_descriptor_t desc; 2322 2323 pid_t pid = IPCThreadState::self()->getCallingPid(); 2324 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2325 pid, effectClient.get(), priority, sessionId, io); 2326 2327 if (pDesc == NULL) { 2328 lStatus = BAD_VALUE; 2329 goto Exit; 2330 } 2331 2332 // check audio settings permission for global effects 2333 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2334 lStatus = PERMISSION_DENIED; 2335 goto Exit; 2336 } 2337 2338 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2339 // that can only be created by audio policy manager (running in same process) 2340 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2341 lStatus = PERMISSION_DENIED; 2342 goto Exit; 2343 } 2344 2345 { 2346 if (!EffectIsNullUuid(&pDesc->uuid)) { 2347 // if uuid is specified, request effect descriptor 2348 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2349 if (lStatus < 0) { 2350 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2351 goto Exit; 2352 } 2353 } else { 2354 // if uuid is not specified, look for an available implementation 2355 // of the required type in effect factory 2356 if (EffectIsNullUuid(&pDesc->type)) { 2357 ALOGW("createEffect() no effect type"); 2358 lStatus = BAD_VALUE; 2359 goto Exit; 2360 } 2361 uint32_t numEffects = 0; 2362 effect_descriptor_t d; 2363 d.flags = 0; // prevent compiler warning 2364 bool found = false; 2365 2366 lStatus = EffectQueryNumberEffects(&numEffects); 2367 if (lStatus < 0) { 2368 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2369 goto Exit; 2370 } 2371 for (uint32_t i = 0; i < numEffects; i++) { 2372 lStatus = EffectQueryEffect(i, &desc); 2373 if (lStatus < 0) { 2374 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2375 continue; 2376 } 2377 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2378 // If matching type found save effect descriptor. If the session is 2379 // 0 and the effect is not auxiliary, continue enumeration in case 2380 // an auxiliary version of this effect type is available 2381 found = true; 2382 d = desc; 2383 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2384 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2385 break; 2386 } 2387 } 2388 } 2389 if (!found) { 2390 lStatus = BAD_VALUE; 2391 ALOGW("createEffect() effect not found"); 2392 goto Exit; 2393 } 2394 // For same effect type, chose auxiliary version over insert version if 2395 // connect to output mix (Compliance to OpenSL ES) 2396 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2397 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2398 desc = d; 2399 } 2400 } 2401 2402 // Do not allow auxiliary effects on a session different from 0 (output mix) 2403 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2404 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2405 lStatus = INVALID_OPERATION; 2406 goto Exit; 2407 } 2408 2409 // check recording permission for visualizer 2410 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2411 !recordingAllowed()) { 2412 lStatus = PERMISSION_DENIED; 2413 goto Exit; 2414 } 2415 2416 // return effect descriptor 2417 *pDesc = desc; 2418 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2419 // if the output returned by getOutputForEffect() is removed before we lock the 2420 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2421 // and we will exit safely 2422 io = AudioSystem::getOutputForEffect(&desc); 2423 ALOGV("createEffect got output %d", io); 2424 } 2425 2426 Mutex::Autolock _l(mLock); 2427 2428 // If output is not specified try to find a matching audio session ID in one of the 2429 // output threads. 2430 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2431 // because of code checking output when entering the function. 2432 // Note: io is never 0 when creating an effect on an input 2433 if (io == AUDIO_IO_HANDLE_NONE) { 2434 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2435 // output must be specified by AudioPolicyManager when using session 2436 // AUDIO_SESSION_OUTPUT_STAGE 2437 lStatus = BAD_VALUE; 2438 goto Exit; 2439 } 2440 // look for the thread where the specified audio session is present 2441 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2442 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2443 io = mPlaybackThreads.keyAt(i); 2444 break; 2445 } 2446 } 2447 if (io == 0) { 2448 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2449 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2450 io = mRecordThreads.keyAt(i); 2451 break; 2452 } 2453 } 2454 } 2455 // If no output thread contains the requested session ID, default to 2456 // first output. The effect chain will be moved to the correct output 2457 // thread when a track with the same session ID is created 2458 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2459 io = mPlaybackThreads.keyAt(0); 2460 } 2461 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2462 } 2463 ThreadBase *thread = checkRecordThread_l(io); 2464 if (thread == NULL) { 2465 thread = checkPlaybackThread_l(io); 2466 if (thread == NULL) { 2467 ALOGE("createEffect() unknown output thread"); 2468 lStatus = BAD_VALUE; 2469 goto Exit; 2470 } 2471 } else { 2472 // Check if one effect chain was awaiting for an effect to be created on this 2473 // session and used it instead of creating a new one. 2474 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2475 if (chain != 0) { 2476 thread->addEffectChain_l(chain); 2477 } 2478 } 2479 2480 sp<Client> client = registerPid(pid); 2481 2482 // create effect on selected output thread 2483 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2484 &desc, enabled, &lStatus); 2485 if (handle != 0 && id != NULL) { 2486 *id = handle->id(); 2487 } 2488 if (handle == 0) { 2489 // remove local strong reference to Client with mClientLock held 2490 Mutex::Autolock _cl(mClientLock); 2491 client.clear(); 2492 } 2493 } 2494 2495Exit: 2496 *status = lStatus; 2497 return handle; 2498} 2499 2500status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2501 audio_io_handle_t dstOutput) 2502{ 2503 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2504 sessionId, srcOutput, dstOutput); 2505 Mutex::Autolock _l(mLock); 2506 if (srcOutput == dstOutput) { 2507 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2508 return NO_ERROR; 2509 } 2510 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2511 if (srcThread == NULL) { 2512 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2513 return BAD_VALUE; 2514 } 2515 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2516 if (dstThread == NULL) { 2517 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2518 return BAD_VALUE; 2519 } 2520 2521 Mutex::Autolock _dl(dstThread->mLock); 2522 Mutex::Autolock _sl(srcThread->mLock); 2523 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2524} 2525 2526// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2527status_t AudioFlinger::moveEffectChain_l(int sessionId, 2528 AudioFlinger::PlaybackThread *srcThread, 2529 AudioFlinger::PlaybackThread *dstThread, 2530 bool reRegister) 2531{ 2532 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2533 sessionId, srcThread, dstThread); 2534 2535 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2536 if (chain == 0) { 2537 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2538 sessionId, srcThread); 2539 return INVALID_OPERATION; 2540 } 2541 2542 // Check whether the destination thread has a channel count of FCC_2, which is 2543 // currently required for (most) effects. Prevent moving the effect chain here rather 2544 // than disabling the addEffect_l() call in dstThread below. 2545 if (dstThread->mChannelCount != FCC_2) { 2546 ALOGW("moveEffectChain_l() effect chain failed because" 2547 " destination thread %p channel count(%u) != %u", 2548 dstThread, dstThread->mChannelCount, FCC_2); 2549 return INVALID_OPERATION; 2550 } 2551 2552 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2553 // so that a new chain is created with correct parameters when first effect is added. This is 2554 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2555 // removed. 2556 srcThread->removeEffectChain_l(chain); 2557 2558 // transfer all effects one by one so that new effect chain is created on new thread with 2559 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2560 sp<EffectChain> dstChain; 2561 uint32_t strategy = 0; // prevent compiler warning 2562 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2563 Vector< sp<EffectModule> > removed; 2564 status_t status = NO_ERROR; 2565 while (effect != 0) { 2566 srcThread->removeEffect_l(effect); 2567 removed.add(effect); 2568 status = dstThread->addEffect_l(effect); 2569 if (status != NO_ERROR) { 2570 break; 2571 } 2572 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2573 if (effect->state() == EffectModule::ACTIVE || 2574 effect->state() == EffectModule::STOPPING) { 2575 effect->start(); 2576 } 2577 // if the move request is not received from audio policy manager, the effect must be 2578 // re-registered with the new strategy and output 2579 if (dstChain == 0) { 2580 dstChain = effect->chain().promote(); 2581 if (dstChain == 0) { 2582 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2583 status = NO_INIT; 2584 break; 2585 } 2586 strategy = dstChain->strategy(); 2587 } 2588 if (reRegister) { 2589 AudioSystem::unregisterEffect(effect->id()); 2590 AudioSystem::registerEffect(&effect->desc(), 2591 dstThread->id(), 2592 strategy, 2593 sessionId, 2594 effect->id()); 2595 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2596 } 2597 effect = chain->getEffectFromId_l(0); 2598 } 2599 2600 if (status != NO_ERROR) { 2601 for (size_t i = 0; i < removed.size(); i++) { 2602 srcThread->addEffect_l(removed[i]); 2603 if (dstChain != 0 && reRegister) { 2604 AudioSystem::unregisterEffect(removed[i]->id()); 2605 AudioSystem::registerEffect(&removed[i]->desc(), 2606 srcThread->id(), 2607 strategy, 2608 sessionId, 2609 removed[i]->id()); 2610 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2611 } 2612 } 2613 } 2614 2615 return status; 2616} 2617 2618bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2619{ 2620 if (mGlobalEffectEnableTime != 0 && 2621 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2622 return true; 2623 } 2624 2625 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2626 sp<EffectChain> ec = 2627 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2628 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2629 return true; 2630 } 2631 } 2632 return false; 2633} 2634 2635void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2636{ 2637 Mutex::Autolock _l(mLock); 2638 2639 mGlobalEffectEnableTime = systemTime(); 2640 2641 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2642 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2643 if (t->mType == ThreadBase::OFFLOAD) { 2644 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2645 } 2646 } 2647 2648} 2649 2650status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2651{ 2652 audio_session_t session = (audio_session_t)chain->sessionId(); 2653 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2654 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2655 if (index >= 0) { 2656 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2657 return ALREADY_EXISTS; 2658 } 2659 mOrphanEffectChains.add(session, chain); 2660 return NO_ERROR; 2661} 2662 2663sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2664{ 2665 sp<EffectChain> chain; 2666 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2667 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2668 if (index >= 0) { 2669 chain = mOrphanEffectChains.valueAt(index); 2670 mOrphanEffectChains.removeItemsAt(index); 2671 } 2672 return chain; 2673} 2674 2675bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2676{ 2677 Mutex::Autolock _l(mLock); 2678 audio_session_t session = (audio_session_t)effect->sessionId(); 2679 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2680 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2681 if (index >= 0) { 2682 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2683 if (chain->removeEffect_l(effect) == 0) { 2684 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2685 mOrphanEffectChains.removeItemsAt(index); 2686 } 2687 return true; 2688 } 2689 return false; 2690} 2691 2692 2693struct Entry { 2694#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2695 char mName[MAX_NAME]; 2696}; 2697 2698int comparEntry(const void *p1, const void *p2) 2699{ 2700 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2701} 2702 2703#ifdef TEE_SINK 2704void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2705{ 2706 NBAIO_Source *teeSource = source.get(); 2707 if (teeSource != NULL) { 2708 // .wav rotation 2709 // There is a benign race condition if 2 threads call this simultaneously. 2710 // They would both traverse the directory, but the result would simply be 2711 // failures at unlink() which are ignored. It's also unlikely since 2712 // normally dumpsys is only done by bugreport or from the command line. 2713 char teePath[32+256]; 2714 strcpy(teePath, "/data/misc/media"); 2715 size_t teePathLen = strlen(teePath); 2716 DIR *dir = opendir(teePath); 2717 teePath[teePathLen++] = '/'; 2718 if (dir != NULL) { 2719#define MAX_SORT 20 // number of entries to sort 2720#define MAX_KEEP 10 // number of entries to keep 2721 struct Entry entries[MAX_SORT]; 2722 size_t entryCount = 0; 2723 while (entryCount < MAX_SORT) { 2724 struct dirent de; 2725 struct dirent *result = NULL; 2726 int rc = readdir_r(dir, &de, &result); 2727 if (rc != 0) { 2728 ALOGW("readdir_r failed %d", rc); 2729 break; 2730 } 2731 if (result == NULL) { 2732 break; 2733 } 2734 if (result != &de) { 2735 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2736 break; 2737 } 2738 // ignore non .wav file entries 2739 size_t nameLen = strlen(de.d_name); 2740 if (nameLen <= 4 || nameLen >= MAX_NAME || 2741 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2742 continue; 2743 } 2744 strcpy(entries[entryCount++].mName, de.d_name); 2745 } 2746 (void) closedir(dir); 2747 if (entryCount > MAX_KEEP) { 2748 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2749 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2750 strcpy(&teePath[teePathLen], entries[i].mName); 2751 (void) unlink(teePath); 2752 } 2753 } 2754 } else { 2755 if (fd >= 0) { 2756 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2757 } 2758 } 2759 char teeTime[16]; 2760 struct timeval tv; 2761 gettimeofday(&tv, NULL); 2762 struct tm tm; 2763 localtime_r(&tv.tv_sec, &tm); 2764 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2765 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2766 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2767 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2768 if (teeFd >= 0) { 2769 // FIXME use libsndfile 2770 char wavHeader[44]; 2771 memcpy(wavHeader, 2772 "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", 2773 sizeof(wavHeader)); 2774 NBAIO_Format format = teeSource->format(); 2775 unsigned channelCount = Format_channelCount(format); 2776 uint32_t sampleRate = Format_sampleRate(format); 2777 size_t frameSize = Format_frameSize(format); 2778 wavHeader[22] = channelCount; // number of channels 2779 wavHeader[24] = sampleRate; // sample rate 2780 wavHeader[25] = sampleRate >> 8; 2781 wavHeader[32] = frameSize; // block alignment 2782 wavHeader[33] = frameSize >> 8; 2783 write(teeFd, wavHeader, sizeof(wavHeader)); 2784 size_t total = 0; 2785 bool firstRead = true; 2786#define TEE_SINK_READ 1024 // frames per I/O operation 2787 void *buffer = malloc(TEE_SINK_READ * frameSize); 2788 for (;;) { 2789 size_t count = TEE_SINK_READ; 2790 ssize_t actual = teeSource->read(buffer, count, 2791 AudioBufferProvider::kInvalidPTS); 2792 bool wasFirstRead = firstRead; 2793 firstRead = false; 2794 if (actual <= 0) { 2795 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2796 continue; 2797 } 2798 break; 2799 } 2800 ALOG_ASSERT(actual <= (ssize_t)count); 2801 write(teeFd, buffer, actual * frameSize); 2802 total += actual; 2803 } 2804 free(buffer); 2805 lseek(teeFd, (off_t) 4, SEEK_SET); 2806 uint32_t temp = 44 + total * frameSize - 8; 2807 // FIXME not big-endian safe 2808 write(teeFd, &temp, sizeof(temp)); 2809 lseek(teeFd, (off_t) 40, SEEK_SET); 2810 temp = total * frameSize; 2811 // FIXME not big-endian safe 2812 write(teeFd, &temp, sizeof(temp)); 2813 close(teeFd); 2814 if (fd >= 0) { 2815 dprintf(fd, "tee copied to %s\n", teePath); 2816 } 2817 } else { 2818 if (fd >= 0) { 2819 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2820 } 2821 } 2822 } 2823} 2824#endif 2825 2826// ---------------------------------------------------------------------------- 2827 2828status_t AudioFlinger::onTransact( 2829 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2830{ 2831 return BnAudioFlinger::onTransact(code, data, reply, flags); 2832} 2833 2834}; // namespace android 2835