AudioFlinger.cpp revision 06fd24f5a3a17ee5ac3069c3e009238fcb6aab39
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/AudioResamplerPublic.h> 49 50#include <media/EffectsFactoryApi.h> 51#include <audio_effects/effect_visualizer.h> 52#include <audio_effects/effect_ns.h> 53#include <audio_effects/effect_aec.h> 54 55#include <audio_utils/primitives.h> 56 57#include <powermanager/PowerManager.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <mediautils/BatteryNotifier.h> 65#include <private/android_filesystem_config.h> 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 85static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 86static const char kClientLockedString[] = "Client lock is taken\n"; 87 88 89nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 90 91uint32_t AudioFlinger::mScreenState; 92 93#ifdef TEE_SINK 94bool AudioFlinger::mTeeSinkInputEnabled = false; 95bool AudioFlinger::mTeeSinkOutputEnabled = false; 96bool AudioFlinger::mTeeSinkTrackEnabled = false; 97 98size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 99size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 100size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 101#endif 102 103// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 104// we define a minimum time during which a global effect is considered enabled. 105static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 106 107// ---------------------------------------------------------------------------- 108 109const char *formatToString(audio_format_t format) { 110 switch (audio_get_main_format(format)) { 111 case AUDIO_FORMAT_PCM: 112 switch (format) { 113 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 114 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 115 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 116 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 117 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 118 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 119 default: 120 break; 121 } 122 break; 123 case AUDIO_FORMAT_MP3: return "mp3"; 124 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 125 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 126 case AUDIO_FORMAT_AAC: return "aac"; 127 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 128 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 129 case AUDIO_FORMAT_VORBIS: return "vorbis"; 130 case AUDIO_FORMAT_OPUS: return "opus"; 131 case AUDIO_FORMAT_AC3: return "ac-3"; 132 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 133 case AUDIO_FORMAT_IEC61937: return "iec61937"; 134 default: 135 break; 136 } 137 return "unknown"; 138} 139 140static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 141{ 142 const hw_module_t *mod; 143 int rc; 144 145 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 146 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 147 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 148 if (rc) { 149 goto out; 150 } 151 rc = audio_hw_device_open(mod, dev); 152 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 153 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 154 if (rc) { 155 goto out; 156 } 157 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 158 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 159 rc = BAD_VALUE; 160 goto out; 161 } 162 return 0; 163 164out: 165 *dev = NULL; 166 return rc; 167} 168 169// ---------------------------------------------------------------------------- 170 171AudioFlinger::AudioFlinger() 172 : BnAudioFlinger(), 173 mPrimaryHardwareDev(NULL), 174 mAudioHwDevs(NULL), 175 mHardwareStatus(AUDIO_HW_IDLE), 176 mMasterVolume(1.0f), 177 mMasterMute(false), 178 mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), // zero has a special meaning, so unavailable 179 mMode(AUDIO_MODE_INVALID), 180 mBtNrecIsOff(false), 181 mIsLowRamDevice(true), 182 mIsDeviceTypeKnown(false), 183 mGlobalEffectEnableTime(0), 184 mSystemReady(false) 185{ 186 getpid_cached = getpid(); 187 const bool doLog = property_get_bool("ro.test_harness", false); 188 if (doLog) { 189 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 190 MemoryHeapBase::READ_ONLY); 191 } 192 193 // reset battery stats. 194 // if the audio service has crashed, battery stats could be left 195 // in bad state, reset the state upon service start. 196 BatteryNotifier::getInstance().noteResetAudio(); 197 198#ifdef TEE_SINK 199 char value[PROPERTY_VALUE_MAX]; 200 (void) property_get("ro.debuggable", value, "0"); 201 int debuggable = atoi(value); 202 int teeEnabled = 0; 203 if (debuggable) { 204 (void) property_get("af.tee", value, "0"); 205 teeEnabled = atoi(value); 206 } 207 // FIXME symbolic constants here 208 if (teeEnabled & 1) { 209 mTeeSinkInputEnabled = true; 210 } 211 if (teeEnabled & 2) { 212 mTeeSinkOutputEnabled = true; 213 } 214 if (teeEnabled & 4) { 215 mTeeSinkTrackEnabled = true; 216 } 217#endif 218} 219 220void AudioFlinger::onFirstRef() 221{ 222 int rc = 0; 223 224 Mutex::Autolock _l(mLock); 225 226 /* TODO: move all this work into an Init() function */ 227 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 228 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 229 uint32_t int_val; 230 if (1 == sscanf(val_str, "%u", &int_val)) { 231 mStandbyTimeInNsecs = milliseconds(int_val); 232 ALOGI("Using %u mSec as standby time.", int_val); 233 } else { 234 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 235 ALOGI("Using default %u mSec as standby time.", 236 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 237 } 238 } 239 240 mPatchPanel = new PatchPanel(this); 241 242 mMode = AUDIO_MODE_NORMAL; 243} 244 245AudioFlinger::~AudioFlinger() 246{ 247 while (!mRecordThreads.isEmpty()) { 248 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 249 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 250 } 251 while (!mPlaybackThreads.isEmpty()) { 252 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 253 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 254 } 255 256 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 257 // no mHardwareLock needed, as there are no other references to this 258 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 259 delete mAudioHwDevs.valueAt(i); 260 } 261 262 // Tell media.log service about any old writers that still need to be unregistered 263 if (mLogMemoryDealer != 0) { 264 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 265 if (binder != 0) { 266 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 267 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 268 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 269 mUnregisteredWriters.pop(); 270 mediaLogService->unregisterWriter(iMemory); 271 } 272 } 273 } 274} 275 276static const char * const audio_interfaces[] = { 277 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 278 AUDIO_HARDWARE_MODULE_ID_A2DP, 279 AUDIO_HARDWARE_MODULE_ID_USB, 280}; 281#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 282 283AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 284 audio_module_handle_t module, 285 audio_devices_t devices) 286{ 287 // if module is 0, the request comes from an old policy manager and we should load 288 // well known modules 289 if (module == 0) { 290 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 291 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 292 loadHwModule_l(audio_interfaces[i]); 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 297 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 298 if ((dev->get_supported_devices != NULL) && 299 (dev->get_supported_devices(dev) & devices) == devices) 300 return audioHwDevice; 301 } 302 } else { 303 // check a match for the requested module handle 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 305 if (audioHwDevice != NULL) { 306 return audioHwDevice; 307 } 308 } 309 310 return NULL; 311} 312 313void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 314{ 315 const size_t SIZE = 256; 316 char buffer[SIZE]; 317 String8 result; 318 319 result.append("Clients:\n"); 320 for (size_t i = 0; i < mClients.size(); ++i) { 321 sp<Client> client = mClients.valueAt(i).promote(); 322 if (client != 0) { 323 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 324 result.append(buffer); 325 } 326 } 327 328 result.append("Notification Clients:\n"); 329 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 330 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 331 result.append(buffer); 332 } 333 334 result.append("Global session refs:\n"); 335 result.append(" session pid count\n"); 336 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 337 AudioSessionRef *r = mAudioSessionRefs[i]; 338 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 339 result.append(buffer); 340 } 341 write(fd, result.string(), result.size()); 342} 343 344 345void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 346{ 347 const size_t SIZE = 256; 348 char buffer[SIZE]; 349 String8 result; 350 hardware_call_state hardwareStatus = mHardwareStatus; 351 352 snprintf(buffer, SIZE, "Hardware status: %d\n" 353 "Standby Time mSec: %u\n", 354 hardwareStatus, 355 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358} 359 360void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 361{ 362 const size_t SIZE = 256; 363 char buffer[SIZE]; 364 String8 result; 365 snprintf(buffer, SIZE, "Permission Denial: " 366 "can't dump AudioFlinger from pid=%d, uid=%d\n", 367 IPCThreadState::self()->getCallingPid(), 368 IPCThreadState::self()->getCallingUid()); 369 result.append(buffer); 370 write(fd, result.string(), result.size()); 371} 372 373bool AudioFlinger::dumpTryLock(Mutex& mutex) 374{ 375 bool locked = false; 376 for (int i = 0; i < kDumpLockRetries; ++i) { 377 if (mutex.tryLock() == NO_ERROR) { 378 locked = true; 379 break; 380 } 381 usleep(kDumpLockSleepUs); 382 } 383 return locked; 384} 385 386status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 387{ 388 if (!dumpAllowed()) { 389 dumpPermissionDenial(fd, args); 390 } else { 391 // get state of hardware lock 392 bool hardwareLocked = dumpTryLock(mHardwareLock); 393 if (!hardwareLocked) { 394 String8 result(kHardwareLockedString); 395 write(fd, result.string(), result.size()); 396 } else { 397 mHardwareLock.unlock(); 398 } 399 400 bool locked = dumpTryLock(mLock); 401 402 // failed to lock - AudioFlinger is probably deadlocked 403 if (!locked) { 404 String8 result(kDeadlockedString); 405 write(fd, result.string(), result.size()); 406 } 407 408 bool clientLocked = dumpTryLock(mClientLock); 409 if (!clientLocked) { 410 String8 result(kClientLockedString); 411 write(fd, result.string(), result.size()); 412 } 413 414 EffectDumpEffects(fd); 415 416 dumpClients(fd, args); 417 if (clientLocked) { 418 mClientLock.unlock(); 419 } 420 421 dumpInternals(fd, args); 422 423 // dump playback threads 424 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 425 mPlaybackThreads.valueAt(i)->dump(fd, args); 426 } 427 428 // dump record threads 429 for (size_t i = 0; i < mRecordThreads.size(); i++) { 430 mRecordThreads.valueAt(i)->dump(fd, args); 431 } 432 433 // dump orphan effect chains 434 if (mOrphanEffectChains.size() != 0) { 435 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 436 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 437 mOrphanEffectChains.valueAt(i)->dump(fd, args); 438 } 439 } 440 // dump all hardware devs 441 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 442 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 443 dev->dump(dev, fd); 444 } 445 446#ifdef TEE_SINK 447 // dump the serially shared record tee sink 448 if (mRecordTeeSource != 0) { 449 dumpTee(fd, mRecordTeeSource); 450 } 451#endif 452 453 if (locked) { 454 mLock.unlock(); 455 } 456 457 // append a copy of media.log here by forwarding fd to it, but don't attempt 458 // to lookup the service if it's not running, as it will block for a second 459 if (mLogMemoryDealer != 0) { 460 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 461 if (binder != 0) { 462 dprintf(fd, "\nmedia.log:\n"); 463 Vector<String16> args; 464 binder->dump(fd, args); 465 } 466 } 467 } 468 return NO_ERROR; 469} 470 471sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 472{ 473 Mutex::Autolock _cl(mClientLock); 474 // If pid is already in the mClients wp<> map, then use that entry 475 // (for which promote() is always != 0), otherwise create a new entry and Client. 476 sp<Client> client = mClients.valueFor(pid).promote(); 477 if (client == 0) { 478 client = new Client(this, pid); 479 mClients.add(pid, client); 480 } 481 482 return client; 483} 484 485sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 486{ 487 // If there is no memory allocated for logs, return a dummy writer that does nothing 488 if (mLogMemoryDealer == 0) { 489 return new NBLog::Writer(); 490 } 491 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 492 // Similarly if we can't contact the media.log service, also return a dummy writer 493 if (binder == 0) { 494 return new NBLog::Writer(); 495 } 496 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 497 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 498 // If allocation fails, consult the vector of previously unregistered writers 499 // and garbage-collect one or more them until an allocation succeeds 500 if (shared == 0) { 501 Mutex::Autolock _l(mUnregisteredWritersLock); 502 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 503 { 504 // Pick the oldest stale writer to garbage-collect 505 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 506 mUnregisteredWriters.removeAt(0); 507 mediaLogService->unregisterWriter(iMemory); 508 // Now the media.log remote reference to IMemory is gone. When our last local 509 // reference to IMemory also drops to zero at end of this block, 510 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 511 } 512 // Re-attempt the allocation 513 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 514 if (shared != 0) { 515 goto success; 516 } 517 } 518 // Even after garbage-collecting all old writers, there is still not enough memory, 519 // so return a dummy writer 520 return new NBLog::Writer(); 521 } 522success: 523 mediaLogService->registerWriter(shared, size, name); 524 return new NBLog::Writer(size, shared); 525} 526 527void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 528{ 529 if (writer == 0) { 530 return; 531 } 532 sp<IMemory> iMemory(writer->getIMemory()); 533 if (iMemory == 0) { 534 return; 535 } 536 // Rather than removing the writer immediately, append it to a queue of old writers to 537 // be garbage-collected later. This allows us to continue to view old logs for a while. 538 Mutex::Autolock _l(mUnregisteredWritersLock); 539 mUnregisteredWriters.push(writer); 540} 541 542// IAudioFlinger interface 543 544 545sp<IAudioTrack> AudioFlinger::createTrack( 546 audio_stream_type_t streamType, 547 uint32_t sampleRate, 548 audio_format_t format, 549 audio_channel_mask_t channelMask, 550 size_t *frameCount, 551 IAudioFlinger::track_flags_t *flags, 552 const sp<IMemory>& sharedBuffer, 553 audio_io_handle_t output, 554 pid_t tid, 555 int *sessionId, 556 int clientUid, 557 status_t *status) 558{ 559 sp<PlaybackThread::Track> track; 560 sp<TrackHandle> trackHandle; 561 sp<Client> client; 562 status_t lStatus; 563 int lSessionId; 564 565 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 566 // but if someone uses binder directly they could bypass that and cause us to crash 567 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 568 ALOGE("createTrack() invalid stream type %d", streamType); 569 lStatus = BAD_VALUE; 570 goto Exit; 571 } 572 573 // further sample rate checks are performed by createTrack_l() depending on the thread type 574 if (sampleRate == 0) { 575 ALOGE("createTrack() invalid sample rate %u", sampleRate); 576 lStatus = BAD_VALUE; 577 goto Exit; 578 } 579 580 // further channel mask checks are performed by createTrack_l() depending on the thread type 581 if (!audio_is_output_channel(channelMask)) { 582 ALOGE("createTrack() invalid channel mask %#x", channelMask); 583 lStatus = BAD_VALUE; 584 goto Exit; 585 } 586 587 // further format checks are performed by createTrack_l() depending on the thread type 588 if (!audio_is_valid_format(format)) { 589 ALOGE("createTrack() invalid format %#x", format); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 595 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 596 lStatus = BAD_VALUE; 597 goto Exit; 598 } 599 600 { 601 Mutex::Autolock _l(mLock); 602 PlaybackThread *thread = checkPlaybackThread_l(output); 603 if (thread == NULL) { 604 ALOGE("no playback thread found for output handle %d", output); 605 lStatus = BAD_VALUE; 606 goto Exit; 607 } 608 609 pid_t pid = IPCThreadState::self()->getCallingPid(); 610 client = registerPid(pid); 611 612 PlaybackThread *effectThread = NULL; 613 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 614 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 615 ALOGE("createTrack() invalid session ID %d", *sessionId); 616 lStatus = BAD_VALUE; 617 goto Exit; 618 } 619 lSessionId = *sessionId; 620 // check if an effect chain with the same session ID is present on another 621 // output thread and move it here. 622 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 623 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 624 if (mPlaybackThreads.keyAt(i) != output) { 625 uint32_t sessions = t->hasAudioSession(lSessionId); 626 if (sessions & PlaybackThread::EFFECT_SESSION) { 627 effectThread = t.get(); 628 break; 629 } 630 } 631 } 632 } else { 633 // if no audio session id is provided, create one here 634 lSessionId = nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 635 if (sessionId != NULL) { 636 *sessionId = lSessionId; 637 } 638 } 639 ALOGV("createTrack() lSessionId: %d", lSessionId); 640 641 track = thread->createTrack_l(client, streamType, sampleRate, format, 642 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 643 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 644 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 645 646 // move effect chain to this output thread if an effect on same session was waiting 647 // for a track to be created 648 if (lStatus == NO_ERROR && effectThread != NULL) { 649 // no risk of deadlock because AudioFlinger::mLock is held 650 Mutex::Autolock _dl(thread->mLock); 651 Mutex::Autolock _sl(effectThread->mLock); 652 moveEffectChain_l(lSessionId, effectThread, thread, true); 653 } 654 655 // Look for sync events awaiting for a session to be used. 656 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 657 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 658 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 659 if (lStatus == NO_ERROR) { 660 (void) track->setSyncEvent(mPendingSyncEvents[i]); 661 } else { 662 mPendingSyncEvents[i]->cancel(); 663 } 664 mPendingSyncEvents.removeAt(i); 665 i--; 666 } 667 } 668 } 669 670 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 671 } 672 673 if (lStatus != NO_ERROR) { 674 // remove local strong reference to Client before deleting the Track so that the 675 // Client destructor is called by the TrackBase destructor with mClientLock held 676 // Don't hold mClientLock when releasing the reference on the track as the 677 // destructor will acquire it. 678 { 679 Mutex::Autolock _cl(mClientLock); 680 client.clear(); 681 } 682 track.clear(); 683 goto Exit; 684 } 685 686 // return handle to client 687 trackHandle = new TrackHandle(track); 688 689Exit: 690 *status = lStatus; 691 return trackHandle; 692} 693 694uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 695{ 696 Mutex::Autolock _l(mLock); 697 ThreadBase *thread = checkThread_l(ioHandle); 698 if (thread == NULL) { 699 ALOGW("sampleRate() unknown thread %d", ioHandle); 700 return 0; 701 } 702 return thread->sampleRate(); 703} 704 705audio_format_t AudioFlinger::format(audio_io_handle_t output) const 706{ 707 Mutex::Autolock _l(mLock); 708 PlaybackThread *thread = checkPlaybackThread_l(output); 709 if (thread == NULL) { 710 ALOGW("format() unknown thread %d", output); 711 return AUDIO_FORMAT_INVALID; 712 } 713 return thread->format(); 714} 715 716size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 717{ 718 Mutex::Autolock _l(mLock); 719 ThreadBase *thread = checkThread_l(ioHandle); 720 if (thread == NULL) { 721 ALOGW("frameCount() unknown thread %d", ioHandle); 722 return 0; 723 } 724 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 725 // should examine all callers and fix them to handle smaller counts 726 return thread->frameCount(); 727} 728 729uint32_t AudioFlinger::latency(audio_io_handle_t output) const 730{ 731 Mutex::Autolock _l(mLock); 732 PlaybackThread *thread = checkPlaybackThread_l(output); 733 if (thread == NULL) { 734 ALOGW("latency(): no playback thread found for output handle %d", output); 735 return 0; 736 } 737 return thread->latency(); 738} 739 740status_t AudioFlinger::setMasterVolume(float value) 741{ 742 status_t ret = initCheck(); 743 if (ret != NO_ERROR) { 744 return ret; 745 } 746 747 // check calling permissions 748 if (!settingsAllowed()) { 749 return PERMISSION_DENIED; 750 } 751 752 Mutex::Autolock _l(mLock); 753 mMasterVolume = value; 754 755 // Set master volume in the HALs which support it. 756 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 757 AutoMutex lock(mHardwareLock); 758 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 759 760 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 761 if (dev->canSetMasterVolume()) { 762 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 763 } 764 mHardwareStatus = AUDIO_HW_IDLE; 765 } 766 767 // Now set the master volume in each playback thread. Playback threads 768 // assigned to HALs which do not have master volume support will apply 769 // master volume during the mix operation. Threads with HALs which do 770 // support master volume will simply ignore the setting. 771 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 772 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 773 continue; 774 } 775 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 776 } 777 778 return NO_ERROR; 779} 780 781status_t AudioFlinger::setMode(audio_mode_t mode) 782{ 783 status_t ret = initCheck(); 784 if (ret != NO_ERROR) { 785 return ret; 786 } 787 788 // check calling permissions 789 if (!settingsAllowed()) { 790 return PERMISSION_DENIED; 791 } 792 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 793 ALOGW("Illegal value: setMode(%d)", mode); 794 return BAD_VALUE; 795 } 796 797 { // scope for the lock 798 AutoMutex lock(mHardwareLock); 799 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 800 mHardwareStatus = AUDIO_HW_SET_MODE; 801 ret = dev->set_mode(dev, mode); 802 mHardwareStatus = AUDIO_HW_IDLE; 803 } 804 805 if (NO_ERROR == ret) { 806 Mutex::Autolock _l(mLock); 807 mMode = mode; 808 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setMode(mode); 810 } 811 812 return ret; 813} 814 815status_t AudioFlinger::setMicMute(bool state) 816{ 817 status_t ret = initCheck(); 818 if (ret != NO_ERROR) { 819 return ret; 820 } 821 822 // check calling permissions 823 if (!settingsAllowed()) { 824 return PERMISSION_DENIED; 825 } 826 827 AutoMutex lock(mHardwareLock); 828 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 829 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 830 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 831 status_t result = dev->set_mic_mute(dev, state); 832 if (result != NO_ERROR) { 833 ret = result; 834 } 835 } 836 mHardwareStatus = AUDIO_HW_IDLE; 837 return ret; 838} 839 840bool AudioFlinger::getMicMute() const 841{ 842 status_t ret = initCheck(); 843 if (ret != NO_ERROR) { 844 return false; 845 } 846 bool mute = true; 847 bool state = AUDIO_MODE_INVALID; 848 AutoMutex lock(mHardwareLock); 849 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 850 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 851 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 852 status_t result = dev->get_mic_mute(dev, &state); 853 if (result == NO_ERROR) { 854 mute = mute && state; 855 } 856 } 857 mHardwareStatus = AUDIO_HW_IDLE; 858 859 return mute; 860} 861 862status_t AudioFlinger::setMasterMute(bool muted) 863{ 864 status_t ret = initCheck(); 865 if (ret != NO_ERROR) { 866 return ret; 867 } 868 869 // check calling permissions 870 if (!settingsAllowed()) { 871 return PERMISSION_DENIED; 872 } 873 874 Mutex::Autolock _l(mLock); 875 mMasterMute = muted; 876 877 // Set master mute in the HALs which support it. 878 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 879 AutoMutex lock(mHardwareLock); 880 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 881 882 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 883 if (dev->canSetMasterMute()) { 884 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 885 } 886 mHardwareStatus = AUDIO_HW_IDLE; 887 } 888 889 // Now set the master mute in each playback thread. Playback threads 890 // assigned to HALs which do not have master mute support will apply master 891 // mute during the mix operation. Threads with HALs which do support master 892 // mute will simply ignore the setting. 893 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 894 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 895 continue; 896 } 897 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 898 } 899 900 return NO_ERROR; 901} 902 903float AudioFlinger::masterVolume() const 904{ 905 Mutex::Autolock _l(mLock); 906 return masterVolume_l(); 907} 908 909bool AudioFlinger::masterMute() const 910{ 911 Mutex::Autolock _l(mLock); 912 return masterMute_l(); 913} 914 915float AudioFlinger::masterVolume_l() const 916{ 917 return mMasterVolume; 918} 919 920bool AudioFlinger::masterMute_l() const 921{ 922 return mMasterMute; 923} 924 925status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 926{ 927 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 928 ALOGW("setStreamVolume() invalid stream %d", stream); 929 return BAD_VALUE; 930 } 931 pid_t caller = IPCThreadState::self()->getCallingPid(); 932 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 933 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 934 return PERMISSION_DENIED; 935 } 936 937 return NO_ERROR; 938} 939 940status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 941 audio_io_handle_t output) 942{ 943 // check calling permissions 944 if (!settingsAllowed()) { 945 return PERMISSION_DENIED; 946 } 947 948 status_t status = checkStreamType(stream); 949 if (status != NO_ERROR) { 950 return status; 951 } 952 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 953 954 AutoMutex lock(mLock); 955 PlaybackThread *thread = NULL; 956 if (output != AUDIO_IO_HANDLE_NONE) { 957 thread = checkPlaybackThread_l(output); 958 if (thread == NULL) { 959 return BAD_VALUE; 960 } 961 } 962 963 mStreamTypes[stream].volume = value; 964 965 if (thread == NULL) { 966 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 967 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 968 } 969 } else { 970 thread->setStreamVolume(stream, value); 971 } 972 973 return NO_ERROR; 974} 975 976status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 977{ 978 // check calling permissions 979 if (!settingsAllowed()) { 980 return PERMISSION_DENIED; 981 } 982 983 status_t status = checkStreamType(stream); 984 if (status != NO_ERROR) { 985 return status; 986 } 987 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 988 989 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 990 ALOGE("setStreamMute() invalid stream %d", stream); 991 return BAD_VALUE; 992 } 993 994 AutoMutex lock(mLock); 995 mStreamTypes[stream].mute = muted; 996 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 997 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 998 999 return NO_ERROR; 1000} 1001 1002float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1003{ 1004 status_t status = checkStreamType(stream); 1005 if (status != NO_ERROR) { 1006 return 0.0f; 1007 } 1008 1009 AutoMutex lock(mLock); 1010 float volume; 1011 if (output != AUDIO_IO_HANDLE_NONE) { 1012 PlaybackThread *thread = checkPlaybackThread_l(output); 1013 if (thread == NULL) { 1014 return 0.0f; 1015 } 1016 volume = thread->streamVolume(stream); 1017 } else { 1018 volume = streamVolume_l(stream); 1019 } 1020 1021 return volume; 1022} 1023 1024bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1025{ 1026 status_t status = checkStreamType(stream); 1027 if (status != NO_ERROR) { 1028 return true; 1029 } 1030 1031 AutoMutex lock(mLock); 1032 return streamMute_l(stream); 1033} 1034 1035 1036void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1037{ 1038 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1039 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1040 } 1041} 1042 1043status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1044{ 1045 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1046 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1047 1048 // check calling permissions 1049 if (!settingsAllowed()) { 1050 return PERMISSION_DENIED; 1051 } 1052 1053 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1054 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1055 Mutex::Autolock _l(mLock); 1056 status_t final_result = NO_ERROR; 1057 { 1058 AutoMutex lock(mHardwareLock); 1059 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1060 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1061 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1062 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1063 final_result = result ?: final_result; 1064 } 1065 mHardwareStatus = AUDIO_HW_IDLE; 1066 } 1067 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1068 AudioParameter param = AudioParameter(keyValuePairs); 1069 String8 value; 1070 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1071 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1072 if (mBtNrecIsOff != btNrecIsOff) { 1073 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1074 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1075 audio_devices_t device = thread->inDevice(); 1076 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1077 // collect all of the thread's session IDs 1078 KeyedVector<int, bool> ids = thread->sessionIds(); 1079 // suspend effects associated with those session IDs 1080 for (size_t j = 0; j < ids.size(); ++j) { 1081 int sessionId = ids.keyAt(j); 1082 thread->setEffectSuspended(FX_IID_AEC, 1083 suspend, 1084 sessionId); 1085 thread->setEffectSuspended(FX_IID_NS, 1086 suspend, 1087 sessionId); 1088 } 1089 } 1090 mBtNrecIsOff = btNrecIsOff; 1091 } 1092 } 1093 String8 screenState; 1094 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1095 bool isOff = screenState == "off"; 1096 if (isOff != (AudioFlinger::mScreenState & 1)) { 1097 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1098 } 1099 } 1100 return final_result; 1101 } 1102 1103 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1104 // and the thread is exited once the lock is released 1105 sp<ThreadBase> thread; 1106 { 1107 Mutex::Autolock _l(mLock); 1108 thread = checkPlaybackThread_l(ioHandle); 1109 if (thread == 0) { 1110 thread = checkRecordThread_l(ioHandle); 1111 } else if (thread == primaryPlaybackThread_l()) { 1112 // indicate output device change to all input threads for pre processing 1113 AudioParameter param = AudioParameter(keyValuePairs); 1114 int value; 1115 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1116 (value != 0)) { 1117 broacastParametersToRecordThreads_l(keyValuePairs); 1118 } 1119 } 1120 } 1121 if (thread != 0) { 1122 return thread->setParameters(keyValuePairs); 1123 } 1124 return BAD_VALUE; 1125} 1126 1127String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1128{ 1129 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1130 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1131 1132 Mutex::Autolock _l(mLock); 1133 1134 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1135 String8 out_s8; 1136 1137 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1138 char *s; 1139 { 1140 AutoMutex lock(mHardwareLock); 1141 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1142 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1143 s = dev->get_parameters(dev, keys.string()); 1144 mHardwareStatus = AUDIO_HW_IDLE; 1145 } 1146 out_s8 += String8(s ? s : ""); 1147 free(s); 1148 } 1149 return out_s8; 1150 } 1151 1152 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1153 if (playbackThread != NULL) { 1154 return playbackThread->getParameters(keys); 1155 } 1156 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1157 if (recordThread != NULL) { 1158 return recordThread->getParameters(keys); 1159 } 1160 return String8(""); 1161} 1162 1163size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1164 audio_channel_mask_t channelMask) const 1165{ 1166 status_t ret = initCheck(); 1167 if (ret != NO_ERROR) { 1168 return 0; 1169 } 1170 if ((sampleRate == 0) || 1171 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1172 !audio_is_input_channel(channelMask)) { 1173 return 0; 1174 } 1175 1176 AutoMutex lock(mHardwareLock); 1177 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1178 audio_config_t config, proposed; 1179 memset(&proposed, 0, sizeof(proposed)); 1180 proposed.sample_rate = sampleRate; 1181 proposed.channel_mask = channelMask; 1182 proposed.format = format; 1183 1184 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1185 size_t frames; 1186 for (;;) { 1187 // Note: config is currently a const parameter for get_input_buffer_size() 1188 // but we use a copy from proposed in case config changes from the call. 1189 config = proposed; 1190 frames = dev->get_input_buffer_size(dev, &config); 1191 if (frames != 0) { 1192 break; // hal success, config is the result 1193 } 1194 // change one parameter of the configuration each iteration to a more "common" value 1195 // to see if the device will support it. 1196 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1197 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1198 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1199 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1200 } else { 1201 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1202 "format %#x, channelMask 0x%X", 1203 sampleRate, format, channelMask); 1204 break; // retries failed, break out of loop with frames == 0. 1205 } 1206 } 1207 mHardwareStatus = AUDIO_HW_IDLE; 1208 if (frames > 0 && config.sample_rate != sampleRate) { 1209 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1210 } 1211 return frames; // may be converted to bytes at the Java level. 1212} 1213 1214uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1215{ 1216 Mutex::Autolock _l(mLock); 1217 1218 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1219 if (recordThread != NULL) { 1220 return recordThread->getInputFramesLost(); 1221 } 1222 return 0; 1223} 1224 1225status_t AudioFlinger::setVoiceVolume(float value) 1226{ 1227 status_t ret = initCheck(); 1228 if (ret != NO_ERROR) { 1229 return ret; 1230 } 1231 1232 // check calling permissions 1233 if (!settingsAllowed()) { 1234 return PERMISSION_DENIED; 1235 } 1236 1237 AutoMutex lock(mHardwareLock); 1238 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1239 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1240 ret = dev->set_voice_volume(dev, value); 1241 mHardwareStatus = AUDIO_HW_IDLE; 1242 1243 return ret; 1244} 1245 1246status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1247 audio_io_handle_t output) const 1248{ 1249 status_t status; 1250 1251 Mutex::Autolock _l(mLock); 1252 1253 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1254 if (playbackThread != NULL) { 1255 return playbackThread->getRenderPosition(halFrames, dspFrames); 1256 } 1257 1258 return BAD_VALUE; 1259} 1260 1261void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1262{ 1263 Mutex::Autolock _l(mLock); 1264 if (client == 0) { 1265 return; 1266 } 1267 pid_t pid = IPCThreadState::self()->getCallingPid(); 1268 { 1269 Mutex::Autolock _cl(mClientLock); 1270 if (mNotificationClients.indexOfKey(pid) < 0) { 1271 sp<NotificationClient> notificationClient = new NotificationClient(this, 1272 client, 1273 pid); 1274 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1275 1276 mNotificationClients.add(pid, notificationClient); 1277 1278 sp<IBinder> binder = IInterface::asBinder(client); 1279 binder->linkToDeath(notificationClient); 1280 } 1281 } 1282 1283 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1284 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1285 // the config change is always sent from playback or record threads to avoid deadlock 1286 // with AudioSystem::gLock 1287 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1288 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1289 } 1290 1291 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1292 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1293 } 1294} 1295 1296void AudioFlinger::removeNotificationClient(pid_t pid) 1297{ 1298 Mutex::Autolock _l(mLock); 1299 { 1300 Mutex::Autolock _cl(mClientLock); 1301 mNotificationClients.removeItem(pid); 1302 } 1303 1304 ALOGV("%d died, releasing its sessions", pid); 1305 size_t num = mAudioSessionRefs.size(); 1306 bool removed = false; 1307 for (size_t i = 0; i< num; ) { 1308 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1309 ALOGV(" pid %d @ %d", ref->mPid, i); 1310 if (ref->mPid == pid) { 1311 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1312 mAudioSessionRefs.removeAt(i); 1313 delete ref; 1314 removed = true; 1315 num--; 1316 } else { 1317 i++; 1318 } 1319 } 1320 if (removed) { 1321 purgeStaleEffects_l(); 1322 } 1323} 1324 1325void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1326 const sp<AudioIoDescriptor>& ioDesc, 1327 pid_t pid) 1328{ 1329 Mutex::Autolock _l(mClientLock); 1330 size_t size = mNotificationClients.size(); 1331 for (size_t i = 0; i < size; i++) { 1332 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1333 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1334 } 1335 } 1336} 1337 1338// removeClient_l() must be called with AudioFlinger::mClientLock held 1339void AudioFlinger::removeClient_l(pid_t pid) 1340{ 1341 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1342 IPCThreadState::self()->getCallingPid()); 1343 mClients.removeItem(pid); 1344} 1345 1346// getEffectThread_l() must be called with AudioFlinger::mLock held 1347sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1348{ 1349 sp<PlaybackThread> thread; 1350 1351 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1352 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1353 ALOG_ASSERT(thread == 0); 1354 thread = mPlaybackThreads.valueAt(i); 1355 } 1356 } 1357 1358 return thread; 1359} 1360 1361 1362 1363// ---------------------------------------------------------------------------- 1364 1365AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1366 : RefBase(), 1367 mAudioFlinger(audioFlinger), 1368 mPid(pid) 1369{ 1370 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1371 heapSize *= 1024; 1372 if (!heapSize) { 1373 heapSize = kClientSharedHeapSizeBytes; 1374 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1375 // invalidated tracks 1376 if (!audioFlinger->isLowRamDevice()) { 1377 heapSize *= kClientSharedHeapSizeMultiplier; 1378 } 1379 } 1380 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1381} 1382 1383// Client destructor must be called with AudioFlinger::mClientLock held 1384AudioFlinger::Client::~Client() 1385{ 1386 mAudioFlinger->removeClient_l(mPid); 1387} 1388 1389sp<MemoryDealer> AudioFlinger::Client::heap() const 1390{ 1391 return mMemoryDealer; 1392} 1393 1394// ---------------------------------------------------------------------------- 1395 1396AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1397 const sp<IAudioFlingerClient>& client, 1398 pid_t pid) 1399 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1400{ 1401} 1402 1403AudioFlinger::NotificationClient::~NotificationClient() 1404{ 1405} 1406 1407void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1408{ 1409 sp<NotificationClient> keep(this); 1410 mAudioFlinger->removeNotificationClient(mPid); 1411} 1412 1413 1414// ---------------------------------------------------------------------------- 1415 1416static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1417 return audio_is_remote_submix_device(inDevice); 1418} 1419 1420sp<IAudioRecord> AudioFlinger::openRecord( 1421 audio_io_handle_t input, 1422 uint32_t sampleRate, 1423 audio_format_t format, 1424 audio_channel_mask_t channelMask, 1425 const String16& opPackageName, 1426 size_t *frameCount, 1427 IAudioFlinger::track_flags_t *flags, 1428 pid_t tid, 1429 int clientUid, 1430 int *sessionId, 1431 size_t *notificationFrames, 1432 sp<IMemory>& cblk, 1433 sp<IMemory>& buffers, 1434 status_t *status) 1435{ 1436 sp<RecordThread::RecordTrack> recordTrack; 1437 sp<RecordHandle> recordHandle; 1438 sp<Client> client; 1439 status_t lStatus; 1440 int lSessionId; 1441 1442 cblk.clear(); 1443 buffers.clear(); 1444 1445 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1446 if (!isTrustedCallingUid(callingUid)) { 1447 ALOGW_IF((uid_t)clientUid != callingUid, 1448 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1449 clientUid = callingUid; 1450 } 1451 1452 // check calling permissions 1453 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1454 ALOGE("openRecord() permission denied: recording not allowed"); 1455 lStatus = PERMISSION_DENIED; 1456 goto Exit; 1457 } 1458 1459 // further sample rate checks are performed by createRecordTrack_l() 1460 if (sampleRate == 0) { 1461 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1462 lStatus = BAD_VALUE; 1463 goto Exit; 1464 } 1465 1466 // we don't yet support anything other than linear PCM 1467 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1468 ALOGE("openRecord() invalid format %#x", format); 1469 lStatus = BAD_VALUE; 1470 goto Exit; 1471 } 1472 1473 // further channel mask checks are performed by createRecordTrack_l() 1474 if (!audio_is_input_channel(channelMask)) { 1475 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1476 lStatus = BAD_VALUE; 1477 goto Exit; 1478 } 1479 1480 { 1481 Mutex::Autolock _l(mLock); 1482 RecordThread *thread = checkRecordThread_l(input); 1483 if (thread == NULL) { 1484 ALOGE("openRecord() checkRecordThread_l failed"); 1485 lStatus = BAD_VALUE; 1486 goto Exit; 1487 } 1488 1489 pid_t pid = IPCThreadState::self()->getCallingPid(); 1490 client = registerPid(pid); 1491 1492 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1493 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1494 lStatus = BAD_VALUE; 1495 goto Exit; 1496 } 1497 lSessionId = *sessionId; 1498 } else { 1499 // if no audio session id is provided, create one here 1500 lSessionId = nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1501 if (sessionId != NULL) { 1502 *sessionId = lSessionId; 1503 } 1504 } 1505 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1506 1507 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1508 frameCount, lSessionId, notificationFrames, 1509 clientUid, flags, tid, &lStatus); 1510 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1511 1512 if (lStatus == NO_ERROR) { 1513 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1514 // session and move it to this thread. 1515 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1516 if (chain != 0) { 1517 Mutex::Autolock _l(thread->mLock); 1518 thread->addEffectChain_l(chain); 1519 } 1520 } 1521 } 1522 1523 if (lStatus != NO_ERROR) { 1524 // remove local strong reference to Client before deleting the RecordTrack so that the 1525 // Client destructor is called by the TrackBase destructor with mClientLock held 1526 // Don't hold mClientLock when releasing the reference on the track as the 1527 // destructor will acquire it. 1528 { 1529 Mutex::Autolock _cl(mClientLock); 1530 client.clear(); 1531 } 1532 recordTrack.clear(); 1533 goto Exit; 1534 } 1535 1536 cblk = recordTrack->getCblk(); 1537 buffers = recordTrack->getBuffers(); 1538 1539 // return handle to client 1540 recordHandle = new RecordHandle(recordTrack); 1541 1542Exit: 1543 *status = lStatus; 1544 return recordHandle; 1545} 1546 1547 1548 1549// ---------------------------------------------------------------------------- 1550 1551audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1552{ 1553 if (name == NULL) { 1554 return 0; 1555 } 1556 if (!settingsAllowed()) { 1557 return 0; 1558 } 1559 Mutex::Autolock _l(mLock); 1560 return loadHwModule_l(name); 1561} 1562 1563// loadHwModule_l() must be called with AudioFlinger::mLock held 1564audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1565{ 1566 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1567 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1568 ALOGW("loadHwModule() module %s already loaded", name); 1569 return mAudioHwDevs.keyAt(i); 1570 } 1571 } 1572 1573 audio_hw_device_t *dev; 1574 1575 int rc = load_audio_interface(name, &dev); 1576 if (rc) { 1577 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1578 return 0; 1579 } 1580 1581 mHardwareStatus = AUDIO_HW_INIT; 1582 rc = dev->init_check(dev); 1583 mHardwareStatus = AUDIO_HW_IDLE; 1584 if (rc) { 1585 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1586 return 0; 1587 } 1588 1589 // Check and cache this HAL's level of support for master mute and master 1590 // volume. If this is the first HAL opened, and it supports the get 1591 // methods, use the initial values provided by the HAL as the current 1592 // master mute and volume settings. 1593 1594 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1595 { // scope for auto-lock pattern 1596 AutoMutex lock(mHardwareLock); 1597 1598 if (0 == mAudioHwDevs.size()) { 1599 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1600 if (NULL != dev->get_master_volume) { 1601 float mv; 1602 if (OK == dev->get_master_volume(dev, &mv)) { 1603 mMasterVolume = mv; 1604 } 1605 } 1606 1607 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1608 if (NULL != dev->get_master_mute) { 1609 bool mm; 1610 if (OK == dev->get_master_mute(dev, &mm)) { 1611 mMasterMute = mm; 1612 } 1613 } 1614 } 1615 1616 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1617 if ((NULL != dev->set_master_volume) && 1618 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1619 flags = static_cast<AudioHwDevice::Flags>(flags | 1620 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1621 } 1622 1623 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1624 if ((NULL != dev->set_master_mute) && 1625 (OK == dev->set_master_mute(dev, mMasterMute))) { 1626 flags = static_cast<AudioHwDevice::Flags>(flags | 1627 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1628 } 1629 1630 mHardwareStatus = AUDIO_HW_IDLE; 1631 } 1632 1633 audio_module_handle_t handle = nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1634 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1635 1636 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1637 name, dev->common.module->name, dev->common.module->id, handle); 1638 1639 return handle; 1640 1641} 1642 1643// ---------------------------------------------------------------------------- 1644 1645uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1646{ 1647 Mutex::Autolock _l(mLock); 1648 PlaybackThread *thread = primaryPlaybackThread_l(); 1649 return thread != NULL ? thread->sampleRate() : 0; 1650} 1651 1652size_t AudioFlinger::getPrimaryOutputFrameCount() 1653{ 1654 Mutex::Autolock _l(mLock); 1655 PlaybackThread *thread = primaryPlaybackThread_l(); 1656 return thread != NULL ? thread->frameCountHAL() : 0; 1657} 1658 1659// ---------------------------------------------------------------------------- 1660 1661status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1662{ 1663 uid_t uid = IPCThreadState::self()->getCallingUid(); 1664 if (uid != AID_SYSTEM) { 1665 return PERMISSION_DENIED; 1666 } 1667 Mutex::Autolock _l(mLock); 1668 if (mIsDeviceTypeKnown) { 1669 return INVALID_OPERATION; 1670 } 1671 mIsLowRamDevice = isLowRamDevice; 1672 mIsDeviceTypeKnown = true; 1673 return NO_ERROR; 1674} 1675 1676audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1677{ 1678 Mutex::Autolock _l(mLock); 1679 1680 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1681 if (index >= 0) { 1682 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1683 mHwAvSyncIds.valueAt(index), sessionId); 1684 return mHwAvSyncIds.valueAt(index); 1685 } 1686 1687 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1688 if (dev == NULL) { 1689 return AUDIO_HW_SYNC_INVALID; 1690 } 1691 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1692 AudioParameter param = AudioParameter(String8(reply)); 1693 free(reply); 1694 1695 int value; 1696 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1697 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1698 return AUDIO_HW_SYNC_INVALID; 1699 } 1700 1701 // allow only one session for a given HW A/V sync ID. 1702 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1703 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1704 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1705 value, mHwAvSyncIds.keyAt(i)); 1706 mHwAvSyncIds.removeItemsAt(i); 1707 break; 1708 } 1709 } 1710 1711 mHwAvSyncIds.add(sessionId, value); 1712 1713 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1714 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1715 uint32_t sessions = thread->hasAudioSession(sessionId); 1716 if (sessions & PlaybackThread::TRACK_SESSION) { 1717 AudioParameter param = AudioParameter(); 1718 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1719 thread->setParameters(param.toString()); 1720 break; 1721 } 1722 } 1723 1724 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1725 return (audio_hw_sync_t)value; 1726} 1727 1728status_t AudioFlinger::systemReady() 1729{ 1730 Mutex::Autolock _l(mLock); 1731 ALOGI("%s", __FUNCTION__); 1732 if (mSystemReady) { 1733 ALOGW("%s called twice", __FUNCTION__); 1734 return NO_ERROR; 1735 } 1736 mSystemReady = true; 1737 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1738 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1739 thread->systemReady(); 1740 } 1741 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1742 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1743 thread->systemReady(); 1744 } 1745 return NO_ERROR; 1746} 1747 1748// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1749void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1750{ 1751 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1752 if (index >= 0) { 1753 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1754 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1755 AudioParameter param = AudioParameter(); 1756 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1757 thread->setParameters(param.toString()); 1758 } 1759} 1760 1761 1762// ---------------------------------------------------------------------------- 1763 1764 1765sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1766 audio_io_handle_t *output, 1767 audio_config_t *config, 1768 audio_devices_t devices, 1769 const String8& address, 1770 audio_output_flags_t flags) 1771{ 1772 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1773 if (outHwDev == NULL) { 1774 return 0; 1775 } 1776 1777 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1778 1779 if (*output == AUDIO_IO_HANDLE_NONE) { 1780 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1781 } else { 1782 // Audio Policy does not currently request a specific output handle. 1783 // If this is ever needed, see openInput_l() for example code. 1784 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1785 return 0; 1786 } 1787 1788 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1789 1790 // FOR TESTING ONLY: 1791 // This if statement allows overriding the audio policy settings 1792 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1793 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1794 // Check only for Normal Mixing mode 1795 if (kEnableExtendedPrecision) { 1796 // Specify format (uncomment one below to choose) 1797 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1798 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1799 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1800 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1801 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1802 } 1803 if (kEnableExtendedChannels) { 1804 // Specify channel mask (uncomment one below to choose) 1805 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1806 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1807 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1808 } 1809 } 1810 1811 AudioStreamOut *outputStream = NULL; 1812 status_t status = outHwDev->openOutputStream( 1813 &outputStream, 1814 *output, 1815 devices, 1816 flags, 1817 config, 1818 address.string()); 1819 1820 mHardwareStatus = AUDIO_HW_IDLE; 1821 1822 if (status == NO_ERROR) { 1823 1824 PlaybackThread *thread; 1825 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1826 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady, 1827 config->offload_info.bit_rate); 1828 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1829 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1830 || !isValidPcmSinkFormat(config->format) 1831 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1832 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1833 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1834 } else { 1835 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1836 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1837 } 1838 mPlaybackThreads.add(*output, thread); 1839 return thread; 1840 } 1841 1842 return 0; 1843} 1844 1845status_t AudioFlinger::openOutput(audio_module_handle_t module, 1846 audio_io_handle_t *output, 1847 audio_config_t *config, 1848 audio_devices_t *devices, 1849 const String8& address, 1850 uint32_t *latencyMs, 1851 audio_output_flags_t flags) 1852{ 1853 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1854 module, 1855 (devices != NULL) ? *devices : 0, 1856 config->sample_rate, 1857 config->format, 1858 config->channel_mask, 1859 flags); 1860 1861 if (*devices == AUDIO_DEVICE_NONE) { 1862 return BAD_VALUE; 1863 } 1864 1865 Mutex::Autolock _l(mLock); 1866 1867 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1868 if (thread != 0) { 1869 *latencyMs = thread->latency(); 1870 1871 // notify client processes of the new output creation 1872 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1873 1874 // the first primary output opened designates the primary hw device 1875 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1876 ALOGI("Using module %d has the primary audio interface", module); 1877 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1878 1879 AutoMutex lock(mHardwareLock); 1880 mHardwareStatus = AUDIO_HW_SET_MODE; 1881 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1882 mHardwareStatus = AUDIO_HW_IDLE; 1883 } 1884 return NO_ERROR; 1885 } 1886 1887 return NO_INIT; 1888} 1889 1890audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1891 audio_io_handle_t output2) 1892{ 1893 Mutex::Autolock _l(mLock); 1894 MixerThread *thread1 = checkMixerThread_l(output1); 1895 MixerThread *thread2 = checkMixerThread_l(output2); 1896 1897 if (thread1 == NULL || thread2 == NULL) { 1898 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1899 output2); 1900 return AUDIO_IO_HANDLE_NONE; 1901 } 1902 1903 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1904 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1905 thread->addOutputTrack(thread2); 1906 mPlaybackThreads.add(id, thread); 1907 // notify client processes of the new output creation 1908 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1909 return id; 1910} 1911 1912status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1913{ 1914 return closeOutput_nonvirtual(output); 1915} 1916 1917status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1918{ 1919 // keep strong reference on the playback thread so that 1920 // it is not destroyed while exit() is executed 1921 sp<PlaybackThread> thread; 1922 { 1923 Mutex::Autolock _l(mLock); 1924 thread = checkPlaybackThread_l(output); 1925 if (thread == NULL) { 1926 return BAD_VALUE; 1927 } 1928 1929 ALOGV("closeOutput() %d", output); 1930 1931 if (thread->type() == ThreadBase::MIXER) { 1932 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1933 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1934 DuplicatingThread *dupThread = 1935 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1936 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1937 } 1938 } 1939 } 1940 1941 1942 mPlaybackThreads.removeItem(output); 1943 // save all effects to the default thread 1944 if (mPlaybackThreads.size()) { 1945 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1946 if (dstThread != NULL) { 1947 // audioflinger lock is held here so the acquisition order of thread locks does not 1948 // matter 1949 Mutex::Autolock _dl(dstThread->mLock); 1950 Mutex::Autolock _sl(thread->mLock); 1951 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1952 for (size_t i = 0; i < effectChains.size(); i ++) { 1953 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1954 } 1955 } 1956 } 1957 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1958 ioDesc->mIoHandle = output; 1959 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1960 } 1961 thread->exit(); 1962 // The thread entity (active unit of execution) is no longer running here, 1963 // but the ThreadBase container still exists. 1964 1965 if (!thread->isDuplicating()) { 1966 closeOutputFinish(thread); 1967 } 1968 1969 return NO_ERROR; 1970} 1971 1972void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1973{ 1974 AudioStreamOut *out = thread->clearOutput(); 1975 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1976 // from now on thread->mOutput is NULL 1977 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1978 delete out; 1979} 1980 1981void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1982{ 1983 mPlaybackThreads.removeItem(thread->mId); 1984 thread->exit(); 1985 closeOutputFinish(thread); 1986} 1987 1988status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1989{ 1990 Mutex::Autolock _l(mLock); 1991 PlaybackThread *thread = checkPlaybackThread_l(output); 1992 1993 if (thread == NULL) { 1994 return BAD_VALUE; 1995 } 1996 1997 ALOGV("suspendOutput() %d", output); 1998 thread->suspend(); 1999 2000 return NO_ERROR; 2001} 2002 2003status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2004{ 2005 Mutex::Autolock _l(mLock); 2006 PlaybackThread *thread = checkPlaybackThread_l(output); 2007 2008 if (thread == NULL) { 2009 return BAD_VALUE; 2010 } 2011 2012 ALOGV("restoreOutput() %d", output); 2013 2014 thread->restore(); 2015 2016 return NO_ERROR; 2017} 2018 2019status_t AudioFlinger::openInput(audio_module_handle_t module, 2020 audio_io_handle_t *input, 2021 audio_config_t *config, 2022 audio_devices_t *devices, 2023 const String8& address, 2024 audio_source_t source, 2025 audio_input_flags_t flags) 2026{ 2027 Mutex::Autolock _l(mLock); 2028 2029 if (*devices == AUDIO_DEVICE_NONE) { 2030 return BAD_VALUE; 2031 } 2032 2033 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2034 2035 if (thread != 0) { 2036 // notify client processes of the new input creation 2037 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2038 return NO_ERROR; 2039 } 2040 return NO_INIT; 2041} 2042 2043sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2044 audio_io_handle_t *input, 2045 audio_config_t *config, 2046 audio_devices_t devices, 2047 const String8& address, 2048 audio_source_t source, 2049 audio_input_flags_t flags) 2050{ 2051 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2052 if (inHwDev == NULL) { 2053 *input = AUDIO_IO_HANDLE_NONE; 2054 return 0; 2055 } 2056 2057 // Audio Policy can request a specific handle for hardware hotword. 2058 // The goal here is not to re-open an already opened input. 2059 // It is to use a pre-assigned I/O handle. 2060 if (*input == AUDIO_IO_HANDLE_NONE) { 2061 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2062 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2063 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2064 return 0; 2065 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2066 // This should not happen in a transient state with current design. 2067 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2068 return 0; 2069 } 2070 2071 audio_config_t halconfig = *config; 2072 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2073 audio_stream_in_t *inStream = NULL; 2074 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2075 &inStream, flags, address.string(), source); 2076 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2077 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2078 inStream, 2079 halconfig.sample_rate, 2080 halconfig.format, 2081 halconfig.channel_mask, 2082 flags, 2083 status, address.string()); 2084 2085 // If the input could not be opened with the requested parameters and we can handle the 2086 // conversion internally, try to open again with the proposed parameters. 2087 if (status == BAD_VALUE && 2088 audio_is_linear_pcm(config->format) && 2089 audio_is_linear_pcm(halconfig.format) && 2090 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2091 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2092 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2093 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2094 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2095 inStream = NULL; 2096 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2097 &inStream, flags, address.string(), source); 2098 // FIXME log this new status; HAL should not propose any further changes 2099 } 2100 2101 if (status == NO_ERROR && inStream != NULL) { 2102 2103#ifdef TEE_SINK 2104 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2105 // or (re-)create if current Pipe is idle and does not match the new format 2106 sp<NBAIO_Sink> teeSink; 2107 enum { 2108 TEE_SINK_NO, // don't copy input 2109 TEE_SINK_NEW, // copy input using a new pipe 2110 TEE_SINK_OLD, // copy input using an existing pipe 2111 } kind; 2112 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2113 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2114 if (!mTeeSinkInputEnabled) { 2115 kind = TEE_SINK_NO; 2116 } else if (!Format_isValid(format)) { 2117 kind = TEE_SINK_NO; 2118 } else if (mRecordTeeSink == 0) { 2119 kind = TEE_SINK_NEW; 2120 } else if (mRecordTeeSink->getStrongCount() != 1) { 2121 kind = TEE_SINK_NO; 2122 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2123 kind = TEE_SINK_OLD; 2124 } else { 2125 kind = TEE_SINK_NEW; 2126 } 2127 switch (kind) { 2128 case TEE_SINK_NEW: { 2129 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2130 size_t numCounterOffers = 0; 2131 const NBAIO_Format offers[1] = {format}; 2132 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2133 ALOG_ASSERT(index == 0); 2134 PipeReader *pipeReader = new PipeReader(*pipe); 2135 numCounterOffers = 0; 2136 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2137 ALOG_ASSERT(index == 0); 2138 mRecordTeeSink = pipe; 2139 mRecordTeeSource = pipeReader; 2140 teeSink = pipe; 2141 } 2142 break; 2143 case TEE_SINK_OLD: 2144 teeSink = mRecordTeeSink; 2145 break; 2146 case TEE_SINK_NO: 2147 default: 2148 break; 2149 } 2150#endif 2151 2152 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2153 2154 // Start record thread 2155 // RecordThread requires both input and output device indication to forward to audio 2156 // pre processing modules 2157 sp<RecordThread> thread = new RecordThread(this, 2158 inputStream, 2159 *input, 2160 primaryOutputDevice_l(), 2161 devices, 2162 mSystemReady 2163#ifdef TEE_SINK 2164 , teeSink 2165#endif 2166 ); 2167 mRecordThreads.add(*input, thread); 2168 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2169 return thread; 2170 } 2171 2172 *input = AUDIO_IO_HANDLE_NONE; 2173 return 0; 2174} 2175 2176status_t AudioFlinger::closeInput(audio_io_handle_t input) 2177{ 2178 return closeInput_nonvirtual(input); 2179} 2180 2181status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2182{ 2183 // keep strong reference on the record thread so that 2184 // it is not destroyed while exit() is executed 2185 sp<RecordThread> thread; 2186 { 2187 Mutex::Autolock _l(mLock); 2188 thread = checkRecordThread_l(input); 2189 if (thread == 0) { 2190 return BAD_VALUE; 2191 } 2192 2193 ALOGV("closeInput() %d", input); 2194 2195 // If we still have effect chains, it means that a client still holds a handle 2196 // on at least one effect. We must either move the chain to an existing thread with the 2197 // same session ID or put it aside in case a new record thread is opened for a 2198 // new capture on the same session 2199 sp<EffectChain> chain; 2200 { 2201 Mutex::Autolock _sl(thread->mLock); 2202 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2203 // Note: maximum one chain per record thread 2204 if (effectChains.size() != 0) { 2205 chain = effectChains[0]; 2206 } 2207 } 2208 if (chain != 0) { 2209 // first check if a record thread is already opened with a client on the same session. 2210 // This should only happen in case of overlap between one thread tear down and the 2211 // creation of its replacement 2212 size_t i; 2213 for (i = 0; i < mRecordThreads.size(); i++) { 2214 sp<RecordThread> t = mRecordThreads.valueAt(i); 2215 if (t == thread) { 2216 continue; 2217 } 2218 if (t->hasAudioSession(chain->sessionId()) != 0) { 2219 Mutex::Autolock _l(t->mLock); 2220 ALOGV("closeInput() found thread %d for effect session %d", 2221 t->id(), chain->sessionId()); 2222 t->addEffectChain_l(chain); 2223 break; 2224 } 2225 } 2226 // put the chain aside if we could not find a record thread with the same session id. 2227 if (i == mRecordThreads.size()) { 2228 putOrphanEffectChain_l(chain); 2229 } 2230 } 2231 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2232 ioDesc->mIoHandle = input; 2233 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2234 mRecordThreads.removeItem(input); 2235 } 2236 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2237 // we have a different lock for notification client 2238 closeInputFinish(thread); 2239 return NO_ERROR; 2240} 2241 2242void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2243{ 2244 thread->exit(); 2245 AudioStreamIn *in = thread->clearInput(); 2246 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2247 // from now on thread->mInput is NULL 2248 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2249 delete in; 2250} 2251 2252void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2253{ 2254 mRecordThreads.removeItem(thread->mId); 2255 closeInputFinish(thread); 2256} 2257 2258status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2259{ 2260 Mutex::Autolock _l(mLock); 2261 ALOGV("invalidateStream() stream %d", stream); 2262 2263 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2264 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2265 thread->invalidateTracks(stream); 2266 } 2267 2268 return NO_ERROR; 2269} 2270 2271 2272audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2273{ 2274 return nextUniqueId(use); 2275} 2276 2277void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2278{ 2279 Mutex::Autolock _l(mLock); 2280 pid_t caller = IPCThreadState::self()->getCallingPid(); 2281 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2282 if (pid != -1 && (caller == getpid_cached)) { 2283 caller = pid; 2284 } 2285 2286 { 2287 Mutex::Autolock _cl(mClientLock); 2288 // Ignore requests received from processes not known as notification client. The request 2289 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2290 // called from a different pid leaving a stale session reference. Also we don't know how 2291 // to clear this reference if the client process dies. 2292 if (mNotificationClients.indexOfKey(caller) < 0) { 2293 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2294 return; 2295 } 2296 } 2297 2298 size_t num = mAudioSessionRefs.size(); 2299 for (size_t i = 0; i< num; i++) { 2300 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2301 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2302 ref->mCnt++; 2303 ALOGV(" incremented refcount to %d", ref->mCnt); 2304 return; 2305 } 2306 } 2307 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2308 ALOGV(" added new entry for %d", audioSession); 2309} 2310 2311void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2312{ 2313 Mutex::Autolock _l(mLock); 2314 pid_t caller = IPCThreadState::self()->getCallingPid(); 2315 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2316 if (pid != -1 && (caller == getpid_cached)) { 2317 caller = pid; 2318 } 2319 size_t num = mAudioSessionRefs.size(); 2320 for (size_t i = 0; i< num; i++) { 2321 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2322 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2323 ref->mCnt--; 2324 ALOGV(" decremented refcount to %d", ref->mCnt); 2325 if (ref->mCnt == 0) { 2326 mAudioSessionRefs.removeAt(i); 2327 delete ref; 2328 purgeStaleEffects_l(); 2329 } 2330 return; 2331 } 2332 } 2333 // If the caller is mediaserver it is likely that the session being released was acquired 2334 // on behalf of a process not in notification clients and we ignore the warning. 2335 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2336} 2337 2338void AudioFlinger::purgeStaleEffects_l() { 2339 2340 ALOGV("purging stale effects"); 2341 2342 Vector< sp<EffectChain> > chains; 2343 2344 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2345 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2346 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2347 sp<EffectChain> ec = t->mEffectChains[j]; 2348 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2349 chains.push(ec); 2350 } 2351 } 2352 } 2353 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2354 sp<RecordThread> t = mRecordThreads.valueAt(i); 2355 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2356 sp<EffectChain> ec = t->mEffectChains[j]; 2357 chains.push(ec); 2358 } 2359 } 2360 2361 for (size_t i = 0; i < chains.size(); i++) { 2362 sp<EffectChain> ec = chains[i]; 2363 int sessionid = ec->sessionId(); 2364 sp<ThreadBase> t = ec->mThread.promote(); 2365 if (t == 0) { 2366 continue; 2367 } 2368 size_t numsessionrefs = mAudioSessionRefs.size(); 2369 bool found = false; 2370 for (size_t k = 0; k < numsessionrefs; k++) { 2371 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2372 if (ref->mSessionid == sessionid) { 2373 ALOGV(" session %d still exists for %d with %d refs", 2374 sessionid, ref->mPid, ref->mCnt); 2375 found = true; 2376 break; 2377 } 2378 } 2379 if (!found) { 2380 Mutex::Autolock _l(t->mLock); 2381 // remove all effects from the chain 2382 while (ec->mEffects.size()) { 2383 sp<EffectModule> effect = ec->mEffects[0]; 2384 effect->unPin(); 2385 t->removeEffect_l(effect); 2386 if (effect->purgeHandles()) { 2387 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2388 } 2389 AudioSystem::unregisterEffect(effect->id()); 2390 } 2391 } 2392 } 2393 return; 2394} 2395 2396// checkThread_l() must be called with AudioFlinger::mLock held 2397AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2398{ 2399 ThreadBase *thread = NULL; 2400 switch (audio_unique_id_get_use(ioHandle)) { 2401 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2402 thread = checkPlaybackThread_l(ioHandle); 2403 break; 2404 case AUDIO_UNIQUE_ID_USE_INPUT: 2405 thread = checkRecordThread_l(ioHandle); 2406 break; 2407 default: 2408 break; 2409 } 2410 return thread; 2411} 2412 2413// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2414AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2415{ 2416 return mPlaybackThreads.valueFor(output).get(); 2417} 2418 2419// checkMixerThread_l() must be called with AudioFlinger::mLock held 2420AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2421{ 2422 PlaybackThread *thread = checkPlaybackThread_l(output); 2423 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2424} 2425 2426// checkRecordThread_l() must be called with AudioFlinger::mLock held 2427AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2428{ 2429 return mRecordThreads.valueFor(input).get(); 2430} 2431 2432audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2433{ 2434 int32_t base = android_atomic_add(AUDIO_UNIQUE_ID_USE_MAX, &mNextUniqueId); 2435 // We have no way of recovering from wraparound 2436 LOG_ALWAYS_FATAL_IF(base == 0, "unique ID overflow"); 2437 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2438 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2439 return (audio_unique_id_t) (base | use); 2440} 2441 2442AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2443{ 2444 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2445 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2446 if(thread->isDuplicating()) { 2447 continue; 2448 } 2449 AudioStreamOut *output = thread->getOutput(); 2450 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2451 return thread; 2452 } 2453 } 2454 return NULL; 2455} 2456 2457audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2458{ 2459 PlaybackThread *thread = primaryPlaybackThread_l(); 2460 2461 if (thread == NULL) { 2462 return 0; 2463 } 2464 2465 return thread->outDevice(); 2466} 2467 2468sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2469 int triggerSession, 2470 int listenerSession, 2471 sync_event_callback_t callBack, 2472 wp<RefBase> cookie) 2473{ 2474 Mutex::Autolock _l(mLock); 2475 2476 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2477 status_t playStatus = NAME_NOT_FOUND; 2478 status_t recStatus = NAME_NOT_FOUND; 2479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2480 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2481 if (playStatus == NO_ERROR) { 2482 return event; 2483 } 2484 } 2485 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2486 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2487 if (recStatus == NO_ERROR) { 2488 return event; 2489 } 2490 } 2491 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2492 mPendingSyncEvents.add(event); 2493 } else { 2494 ALOGV("createSyncEvent() invalid event %d", event->type()); 2495 event.clear(); 2496 } 2497 return event; 2498} 2499 2500// ---------------------------------------------------------------------------- 2501// Effect management 2502// ---------------------------------------------------------------------------- 2503 2504 2505status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2506{ 2507 Mutex::Autolock _l(mLock); 2508 return EffectQueryNumberEffects(numEffects); 2509} 2510 2511status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2512{ 2513 Mutex::Autolock _l(mLock); 2514 return EffectQueryEffect(index, descriptor); 2515} 2516 2517status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2518 effect_descriptor_t *descriptor) const 2519{ 2520 Mutex::Autolock _l(mLock); 2521 return EffectGetDescriptor(pUuid, descriptor); 2522} 2523 2524 2525sp<IEffect> AudioFlinger::createEffect( 2526 effect_descriptor_t *pDesc, 2527 const sp<IEffectClient>& effectClient, 2528 int32_t priority, 2529 audio_io_handle_t io, 2530 int sessionId, 2531 const String16& opPackageName, 2532 status_t *status, 2533 int *id, 2534 int *enabled) 2535{ 2536 status_t lStatus = NO_ERROR; 2537 sp<EffectHandle> handle; 2538 effect_descriptor_t desc; 2539 2540 pid_t pid = IPCThreadState::self()->getCallingPid(); 2541 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2542 pid, effectClient.get(), priority, sessionId, io); 2543 2544 if (pDesc == NULL) { 2545 lStatus = BAD_VALUE; 2546 goto Exit; 2547 } 2548 2549 // check audio settings permission for global effects 2550 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2551 lStatus = PERMISSION_DENIED; 2552 goto Exit; 2553 } 2554 2555 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2556 // that can only be created by audio policy manager (running in same process) 2557 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2558 lStatus = PERMISSION_DENIED; 2559 goto Exit; 2560 } 2561 2562 { 2563 if (!EffectIsNullUuid(&pDesc->uuid)) { 2564 // if uuid is specified, request effect descriptor 2565 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2566 if (lStatus < 0) { 2567 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2568 goto Exit; 2569 } 2570 } else { 2571 // if uuid is not specified, look for an available implementation 2572 // of the required type in effect factory 2573 if (EffectIsNullUuid(&pDesc->type)) { 2574 ALOGW("createEffect() no effect type"); 2575 lStatus = BAD_VALUE; 2576 goto Exit; 2577 } 2578 uint32_t numEffects = 0; 2579 effect_descriptor_t d; 2580 d.flags = 0; // prevent compiler warning 2581 bool found = false; 2582 2583 lStatus = EffectQueryNumberEffects(&numEffects); 2584 if (lStatus < 0) { 2585 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2586 goto Exit; 2587 } 2588 for (uint32_t i = 0; i < numEffects; i++) { 2589 lStatus = EffectQueryEffect(i, &desc); 2590 if (lStatus < 0) { 2591 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2592 continue; 2593 } 2594 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2595 // If matching type found save effect descriptor. If the session is 2596 // 0 and the effect is not auxiliary, continue enumeration in case 2597 // an auxiliary version of this effect type is available 2598 found = true; 2599 d = desc; 2600 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2601 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2602 break; 2603 } 2604 } 2605 } 2606 if (!found) { 2607 lStatus = BAD_VALUE; 2608 ALOGW("createEffect() effect not found"); 2609 goto Exit; 2610 } 2611 // For same effect type, chose auxiliary version over insert version if 2612 // connect to output mix (Compliance to OpenSL ES) 2613 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2614 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2615 desc = d; 2616 } 2617 } 2618 2619 // Do not allow auxiliary effects on a session different from 0 (output mix) 2620 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2621 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2622 lStatus = INVALID_OPERATION; 2623 goto Exit; 2624 } 2625 2626 // check recording permission for visualizer 2627 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2628 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2629 lStatus = PERMISSION_DENIED; 2630 goto Exit; 2631 } 2632 2633 // return effect descriptor 2634 *pDesc = desc; 2635 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2636 // if the output returned by getOutputForEffect() is removed before we lock the 2637 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2638 // and we will exit safely 2639 io = AudioSystem::getOutputForEffect(&desc); 2640 ALOGV("createEffect got output %d", io); 2641 } 2642 2643 Mutex::Autolock _l(mLock); 2644 2645 // If output is not specified try to find a matching audio session ID in one of the 2646 // output threads. 2647 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2648 // because of code checking output when entering the function. 2649 // Note: io is never 0 when creating an effect on an input 2650 if (io == AUDIO_IO_HANDLE_NONE) { 2651 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2652 // output must be specified by AudioPolicyManager when using session 2653 // AUDIO_SESSION_OUTPUT_STAGE 2654 lStatus = BAD_VALUE; 2655 goto Exit; 2656 } 2657 // look for the thread where the specified audio session is present 2658 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2659 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2660 io = mPlaybackThreads.keyAt(i); 2661 break; 2662 } 2663 } 2664 if (io == 0) { 2665 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2666 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2667 io = mRecordThreads.keyAt(i); 2668 break; 2669 } 2670 } 2671 } 2672 // If no output thread contains the requested session ID, default to 2673 // first output. The effect chain will be moved to the correct output 2674 // thread when a track with the same session ID is created 2675 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2676 io = mPlaybackThreads.keyAt(0); 2677 } 2678 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2679 } 2680 ThreadBase *thread = checkRecordThread_l(io); 2681 if (thread == NULL) { 2682 thread = checkPlaybackThread_l(io); 2683 if (thread == NULL) { 2684 ALOGE("createEffect() unknown output thread"); 2685 lStatus = BAD_VALUE; 2686 goto Exit; 2687 } 2688 } else { 2689 // Check if one effect chain was awaiting for an effect to be created on this 2690 // session and used it instead of creating a new one. 2691 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2692 if (chain != 0) { 2693 Mutex::Autolock _l(thread->mLock); 2694 thread->addEffectChain_l(chain); 2695 } 2696 } 2697 2698 sp<Client> client = registerPid(pid); 2699 2700 // create effect on selected output thread 2701 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2702 &desc, enabled, &lStatus); 2703 if (handle != 0 && id != NULL) { 2704 *id = handle->id(); 2705 } 2706 if (handle == 0) { 2707 // remove local strong reference to Client with mClientLock held 2708 Mutex::Autolock _cl(mClientLock); 2709 client.clear(); 2710 } 2711 } 2712 2713Exit: 2714 *status = lStatus; 2715 return handle; 2716} 2717 2718status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2719 audio_io_handle_t dstOutput) 2720{ 2721 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2722 sessionId, srcOutput, dstOutput); 2723 Mutex::Autolock _l(mLock); 2724 if (srcOutput == dstOutput) { 2725 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2726 return NO_ERROR; 2727 } 2728 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2729 if (srcThread == NULL) { 2730 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2731 return BAD_VALUE; 2732 } 2733 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2734 if (dstThread == NULL) { 2735 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2736 return BAD_VALUE; 2737 } 2738 2739 Mutex::Autolock _dl(dstThread->mLock); 2740 Mutex::Autolock _sl(srcThread->mLock); 2741 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2742} 2743 2744// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2745status_t AudioFlinger::moveEffectChain_l(int sessionId, 2746 AudioFlinger::PlaybackThread *srcThread, 2747 AudioFlinger::PlaybackThread *dstThread, 2748 bool reRegister) 2749{ 2750 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2751 sessionId, srcThread, dstThread); 2752 2753 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2754 if (chain == 0) { 2755 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2756 sessionId, srcThread); 2757 return INVALID_OPERATION; 2758 } 2759 2760 // Check whether the destination thread has a channel count of FCC_2, which is 2761 // currently required for (most) effects. Prevent moving the effect chain here rather 2762 // than disabling the addEffect_l() call in dstThread below. 2763 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2764 dstThread->mChannelCount != FCC_2) { 2765 ALOGW("moveEffectChain_l() effect chain failed because" 2766 " destination thread %p channel count(%u) != %u", 2767 dstThread, dstThread->mChannelCount, FCC_2); 2768 return INVALID_OPERATION; 2769 } 2770 2771 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2772 // so that a new chain is created with correct parameters when first effect is added. This is 2773 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2774 // removed. 2775 srcThread->removeEffectChain_l(chain); 2776 2777 // transfer all effects one by one so that new effect chain is created on new thread with 2778 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2779 sp<EffectChain> dstChain; 2780 uint32_t strategy = 0; // prevent compiler warning 2781 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2782 Vector< sp<EffectModule> > removed; 2783 status_t status = NO_ERROR; 2784 while (effect != 0) { 2785 srcThread->removeEffect_l(effect); 2786 removed.add(effect); 2787 status = dstThread->addEffect_l(effect); 2788 if (status != NO_ERROR) { 2789 break; 2790 } 2791 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2792 if (effect->state() == EffectModule::ACTIVE || 2793 effect->state() == EffectModule::STOPPING) { 2794 effect->start(); 2795 } 2796 // if the move request is not received from audio policy manager, the effect must be 2797 // re-registered with the new strategy and output 2798 if (dstChain == 0) { 2799 dstChain = effect->chain().promote(); 2800 if (dstChain == 0) { 2801 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2802 status = NO_INIT; 2803 break; 2804 } 2805 strategy = dstChain->strategy(); 2806 } 2807 if (reRegister) { 2808 AudioSystem::unregisterEffect(effect->id()); 2809 AudioSystem::registerEffect(&effect->desc(), 2810 dstThread->id(), 2811 strategy, 2812 sessionId, 2813 effect->id()); 2814 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2815 } 2816 effect = chain->getEffectFromId_l(0); 2817 } 2818 2819 if (status != NO_ERROR) { 2820 for (size_t i = 0; i < removed.size(); i++) { 2821 srcThread->addEffect_l(removed[i]); 2822 if (dstChain != 0 && reRegister) { 2823 AudioSystem::unregisterEffect(removed[i]->id()); 2824 AudioSystem::registerEffect(&removed[i]->desc(), 2825 srcThread->id(), 2826 strategy, 2827 sessionId, 2828 removed[i]->id()); 2829 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2830 } 2831 } 2832 } 2833 2834 return status; 2835} 2836 2837bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2838{ 2839 if (mGlobalEffectEnableTime != 0 && 2840 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2841 return true; 2842 } 2843 2844 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2845 sp<EffectChain> ec = 2846 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2847 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2848 return true; 2849 } 2850 } 2851 return false; 2852} 2853 2854void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2855{ 2856 Mutex::Autolock _l(mLock); 2857 2858 mGlobalEffectEnableTime = systemTime(); 2859 2860 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2861 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2862 if (t->mType == ThreadBase::OFFLOAD) { 2863 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2864 } 2865 } 2866 2867} 2868 2869status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2870{ 2871 audio_session_t session = (audio_session_t)chain->sessionId(); 2872 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2873 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2874 if (index >= 0) { 2875 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2876 return ALREADY_EXISTS; 2877 } 2878 mOrphanEffectChains.add(session, chain); 2879 return NO_ERROR; 2880} 2881 2882sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2883{ 2884 sp<EffectChain> chain; 2885 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2886 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2887 if (index >= 0) { 2888 chain = mOrphanEffectChains.valueAt(index); 2889 mOrphanEffectChains.removeItemsAt(index); 2890 } 2891 return chain; 2892} 2893 2894bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2895{ 2896 Mutex::Autolock _l(mLock); 2897 audio_session_t session = (audio_session_t)effect->sessionId(); 2898 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2899 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2900 if (index >= 0) { 2901 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2902 if (chain->removeEffect_l(effect) == 0) { 2903 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2904 mOrphanEffectChains.removeItemsAt(index); 2905 } 2906 return true; 2907 } 2908 return false; 2909} 2910 2911 2912struct Entry { 2913#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2914 char mFileName[TEE_MAX_FILENAME]; 2915}; 2916 2917int comparEntry(const void *p1, const void *p2) 2918{ 2919 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2920} 2921 2922#ifdef TEE_SINK 2923void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2924{ 2925 NBAIO_Source *teeSource = source.get(); 2926 if (teeSource != NULL) { 2927 // .wav rotation 2928 // There is a benign race condition if 2 threads call this simultaneously. 2929 // They would both traverse the directory, but the result would simply be 2930 // failures at unlink() which are ignored. It's also unlikely since 2931 // normally dumpsys is only done by bugreport or from the command line. 2932 char teePath[32+256]; 2933 strcpy(teePath, "/data/misc/audioserver"); 2934 size_t teePathLen = strlen(teePath); 2935 DIR *dir = opendir(teePath); 2936 teePath[teePathLen++] = '/'; 2937 if (dir != NULL) { 2938#define TEE_MAX_SORT 20 // number of entries to sort 2939#define TEE_MAX_KEEP 10 // number of entries to keep 2940 struct Entry entries[TEE_MAX_SORT]; 2941 size_t entryCount = 0; 2942 while (entryCount < TEE_MAX_SORT) { 2943 struct dirent de; 2944 struct dirent *result = NULL; 2945 int rc = readdir_r(dir, &de, &result); 2946 if (rc != 0) { 2947 ALOGW("readdir_r failed %d", rc); 2948 break; 2949 } 2950 if (result == NULL) { 2951 break; 2952 } 2953 if (result != &de) { 2954 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2955 break; 2956 } 2957 // ignore non .wav file entries 2958 size_t nameLen = strlen(de.d_name); 2959 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2960 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2961 continue; 2962 } 2963 strcpy(entries[entryCount++].mFileName, de.d_name); 2964 } 2965 (void) closedir(dir); 2966 if (entryCount > TEE_MAX_KEEP) { 2967 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2968 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2969 strcpy(&teePath[teePathLen], entries[i].mFileName); 2970 (void) unlink(teePath); 2971 } 2972 } 2973 } else { 2974 if (fd >= 0) { 2975 dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath, 2976 strerror(errno)); 2977 } 2978 } 2979 char teeTime[16]; 2980 struct timeval tv; 2981 gettimeofday(&tv, NULL); 2982 struct tm tm; 2983 localtime_r(&tv.tv_sec, &tm); 2984 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2985 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2986 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2987 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2988 if (teeFd >= 0) { 2989 // FIXME use libsndfile 2990 char wavHeader[44]; 2991 memcpy(wavHeader, 2992 "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", 2993 sizeof(wavHeader)); 2994 NBAIO_Format format = teeSource->format(); 2995 unsigned channelCount = Format_channelCount(format); 2996 uint32_t sampleRate = Format_sampleRate(format); 2997 size_t frameSize = Format_frameSize(format); 2998 wavHeader[22] = channelCount; // number of channels 2999 wavHeader[24] = sampleRate; // sample rate 3000 wavHeader[25] = sampleRate >> 8; 3001 wavHeader[32] = frameSize; // block alignment 3002 wavHeader[33] = frameSize >> 8; 3003 write(teeFd, wavHeader, sizeof(wavHeader)); 3004 size_t total = 0; 3005 bool firstRead = true; 3006#define TEE_SINK_READ 1024 // frames per I/O operation 3007 void *buffer = malloc(TEE_SINK_READ * frameSize); 3008 for (;;) { 3009 size_t count = TEE_SINK_READ; 3010 ssize_t actual = teeSource->read(buffer, count); 3011 bool wasFirstRead = firstRead; 3012 firstRead = false; 3013 if (actual <= 0) { 3014 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3015 continue; 3016 } 3017 break; 3018 } 3019 ALOG_ASSERT(actual <= (ssize_t)count); 3020 write(teeFd, buffer, actual * frameSize); 3021 total += actual; 3022 } 3023 free(buffer); 3024 lseek(teeFd, (off_t) 4, SEEK_SET); 3025 uint32_t temp = 44 + total * frameSize - 8; 3026 // FIXME not big-endian safe 3027 write(teeFd, &temp, sizeof(temp)); 3028 lseek(teeFd, (off_t) 40, SEEK_SET); 3029 temp = total * frameSize; 3030 // FIXME not big-endian safe 3031 write(teeFd, &temp, sizeof(temp)); 3032 close(teeFd); 3033 if (fd >= 0) { 3034 dprintf(fd, "tee copied to %s\n", teePath); 3035 } 3036 } else { 3037 if (fd >= 0) { 3038 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3039 } 3040 } 3041 } 3042} 3043#endif 3044 3045// ---------------------------------------------------------------------------- 3046 3047status_t AudioFlinger::onTransact( 3048 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3049{ 3050 return BnAudioFlinger::onTransact(code, data, reply, flags); 3051} 3052 3053} // namespace android 3054