AudioFlinger.cpp revision 7b81b7e026053615f93af1efc130205eef547e57
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 Mutex::Autolock _l(mLock); 223 224 /* TODO: move all this work into an Init() function */ 225 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 226 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 227 uint32_t int_val; 228 if (1 == sscanf(val_str, "%u", &int_val)) { 229 mStandbyTimeInNsecs = milliseconds(int_val); 230 ALOGI("Using %u mSec as standby time.", int_val); 231 } else { 232 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 233 ALOGI("Using default %u mSec as standby time.", 234 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 235 } 236 } 237 238 mPatchPanel = new PatchPanel(this); 239 240 mMode = AUDIO_MODE_NORMAL; 241} 242 243AudioFlinger::~AudioFlinger() 244{ 245 while (!mRecordThreads.isEmpty()) { 246 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 247 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 248 } 249 while (!mPlaybackThreads.isEmpty()) { 250 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 251 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 252 } 253 254 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 255 // no mHardwareLock needed, as there are no other references to this 256 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 257 delete mAudioHwDevs.valueAt(i); 258 } 259 260 // Tell media.log service about any old writers that still need to be unregistered 261 if (mLogMemoryDealer != 0) { 262 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 263 if (binder != 0) { 264 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 265 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 266 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 267 mUnregisteredWriters.pop(); 268 mediaLogService->unregisterWriter(iMemory); 269 } 270 } 271 } 272} 273 274static const char * const audio_interfaces[] = { 275 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 276 AUDIO_HARDWARE_MODULE_ID_A2DP, 277 AUDIO_HARDWARE_MODULE_ID_USB, 278}; 279#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 280 281AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 282 audio_module_handle_t module, 283 audio_devices_t devices) 284{ 285 // if module is 0, the request comes from an old policy manager and we should load 286 // well known modules 287 if (module == 0) { 288 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 289 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 290 loadHwModule_l(audio_interfaces[i]); 291 } 292 // then try to find a module supporting the requested device. 293 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 294 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 295 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 296 if ((dev->get_supported_devices != NULL) && 297 (dev->get_supported_devices(dev) & devices) == devices) 298 return audioHwDevice; 299 } 300 } else { 301 // check a match for the requested module handle 302 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 303 if (audioHwDevice != NULL) { 304 return audioHwDevice; 305 } 306 } 307 308 return NULL; 309} 310 311void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 312{ 313 const size_t SIZE = 256; 314 char buffer[SIZE]; 315 String8 result; 316 317 result.append("Clients:\n"); 318 for (size_t i = 0; i < mClients.size(); ++i) { 319 sp<Client> client = mClients.valueAt(i).promote(); 320 if (client != 0) { 321 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 322 result.append(buffer); 323 } 324 } 325 326 result.append("Notification Clients:\n"); 327 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 328 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 329 result.append(buffer); 330 } 331 332 result.append("Global session refs:\n"); 333 result.append(" session pid count\n"); 334 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 335 AudioSessionRef *r = mAudioSessionRefs[i]; 336 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 337 result.append(buffer); 338 } 339 write(fd, result.string(), result.size()); 340} 341 342 343void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 344{ 345 const size_t SIZE = 256; 346 char buffer[SIZE]; 347 String8 result; 348 hardware_call_state hardwareStatus = mHardwareStatus; 349 350 snprintf(buffer, SIZE, "Hardware status: %d\n" 351 "Standby Time mSec: %u\n", 352 hardwareStatus, 353 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 354 result.append(buffer); 355 write(fd, result.string(), result.size()); 356} 357 358void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 359{ 360 const size_t SIZE = 256; 361 char buffer[SIZE]; 362 String8 result; 363 snprintf(buffer, SIZE, "Permission Denial: " 364 "can't dump AudioFlinger from pid=%d, uid=%d\n", 365 IPCThreadState::self()->getCallingPid(), 366 IPCThreadState::self()->getCallingUid()); 367 result.append(buffer); 368 write(fd, result.string(), result.size()); 369} 370 371bool AudioFlinger::dumpTryLock(Mutex& mutex) 372{ 373 bool locked = false; 374 for (int i = 0; i < kDumpLockRetries; ++i) { 375 if (mutex.tryLock() == NO_ERROR) { 376 locked = true; 377 break; 378 } 379 usleep(kDumpLockSleepUs); 380 } 381 return locked; 382} 383 384status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 385{ 386 if (!dumpAllowed()) { 387 dumpPermissionDenial(fd, args); 388 } else { 389 // get state of hardware lock 390 bool hardwareLocked = dumpTryLock(mHardwareLock); 391 if (!hardwareLocked) { 392 String8 result(kHardwareLockedString); 393 write(fd, result.string(), result.size()); 394 } else { 395 mHardwareLock.unlock(); 396 } 397 398 bool locked = dumpTryLock(mLock); 399 400 // failed to lock - AudioFlinger is probably deadlocked 401 if (!locked) { 402 String8 result(kDeadlockedString); 403 write(fd, result.string(), result.size()); 404 } 405 406 bool clientLocked = dumpTryLock(mClientLock); 407 if (!clientLocked) { 408 String8 result(kClientLockedString); 409 write(fd, result.string(), result.size()); 410 } 411 412 EffectDumpEffects(fd); 413 414 dumpClients(fd, args); 415 if (clientLocked) { 416 mClientLock.unlock(); 417 } 418 419 dumpInternals(fd, args); 420 421 // dump playback threads 422 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 423 mPlaybackThreads.valueAt(i)->dump(fd, args); 424 } 425 426 // dump record threads 427 for (size_t i = 0; i < mRecordThreads.size(); i++) { 428 mRecordThreads.valueAt(i)->dump(fd, args); 429 } 430 431 // dump orphan effect chains 432 if (mOrphanEffectChains.size() != 0) { 433 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 434 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 435 mOrphanEffectChains.valueAt(i)->dump(fd, args); 436 } 437 } 438 // dump all hardware devs 439 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 440 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 441 dev->dump(dev, fd); 442 } 443 444#ifdef TEE_SINK 445 // dump the serially shared record tee sink 446 if (mRecordTeeSource != 0) { 447 dumpTee(fd, mRecordTeeSource); 448 } 449#endif 450 451 if (locked) { 452 mLock.unlock(); 453 } 454 455 // append a copy of media.log here by forwarding fd to it, but don't attempt 456 // to lookup the service if it's not running, as it will block for a second 457 if (mLogMemoryDealer != 0) { 458 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 459 if (binder != 0) { 460 dprintf(fd, "\nmedia.log:\n"); 461 Vector<String16> args; 462 binder->dump(fd, args); 463 } 464 } 465 } 466 return NO_ERROR; 467} 468 469sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 470{ 471 Mutex::Autolock _cl(mClientLock); 472 // If pid is already in the mClients wp<> map, then use that entry 473 // (for which promote() is always != 0), otherwise create a new entry and Client. 474 sp<Client> client = mClients.valueFor(pid).promote(); 475 if (client == 0) { 476 client = new Client(this, pid); 477 mClients.add(pid, client); 478 } 479 480 return client; 481} 482 483sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 484{ 485 // If there is no memory allocated for logs, return a dummy writer that does nothing 486 if (mLogMemoryDealer == 0) { 487 return new NBLog::Writer(); 488 } 489 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 490 // Similarly if we can't contact the media.log service, also return a dummy writer 491 if (binder == 0) { 492 return new NBLog::Writer(); 493 } 494 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 495 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 496 // If allocation fails, consult the vector of previously unregistered writers 497 // and garbage-collect one or more them until an allocation succeeds 498 if (shared == 0) { 499 Mutex::Autolock _l(mUnregisteredWritersLock); 500 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 501 { 502 // Pick the oldest stale writer to garbage-collect 503 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 504 mUnregisteredWriters.removeAt(0); 505 mediaLogService->unregisterWriter(iMemory); 506 // Now the media.log remote reference to IMemory is gone. When our last local 507 // reference to IMemory also drops to zero at end of this block, 508 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 509 } 510 // Re-attempt the allocation 511 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 512 if (shared != 0) { 513 goto success; 514 } 515 } 516 // Even after garbage-collecting all old writers, there is still not enough memory, 517 // so return a dummy writer 518 return new NBLog::Writer(); 519 } 520success: 521 mediaLogService->registerWriter(shared, size, name); 522 return new NBLog::Writer(size, shared); 523} 524 525void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 526{ 527 if (writer == 0) { 528 return; 529 } 530 sp<IMemory> iMemory(writer->getIMemory()); 531 if (iMemory == 0) { 532 return; 533 } 534 // Rather than removing the writer immediately, append it to a queue of old writers to 535 // be garbage-collected later. This allows us to continue to view old logs for a while. 536 Mutex::Autolock _l(mUnregisteredWritersLock); 537 mUnregisteredWriters.push(writer); 538} 539 540// IAudioFlinger interface 541 542 543sp<IAudioTrack> AudioFlinger::createTrack( 544 audio_stream_type_t streamType, 545 uint32_t sampleRate, 546 audio_format_t format, 547 audio_channel_mask_t channelMask, 548 size_t *frameCount, 549 IAudioFlinger::track_flags_t *flags, 550 const sp<IMemory>& sharedBuffer, 551 audio_io_handle_t output, 552 pid_t tid, 553 audio_session_t *sessionId, 554 int clientUid, 555 status_t *status) 556{ 557 sp<PlaybackThread::Track> track; 558 sp<TrackHandle> trackHandle; 559 sp<Client> client; 560 status_t lStatus; 561 audio_session_t lSessionId; 562 563 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 564 // but if someone uses binder directly they could bypass that and cause us to crash 565 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 566 ALOGE("createTrack() invalid stream type %d", streamType); 567 lStatus = BAD_VALUE; 568 goto Exit; 569 } 570 571 // further sample rate checks are performed by createTrack_l() depending on the thread type 572 if (sampleRate == 0) { 573 ALOGE("createTrack() invalid sample rate %u", sampleRate); 574 lStatus = BAD_VALUE; 575 goto Exit; 576 } 577 578 // further channel mask checks are performed by createTrack_l() depending on the thread type 579 if (!audio_is_output_channel(channelMask)) { 580 ALOGE("createTrack() invalid channel mask %#x", channelMask); 581 lStatus = BAD_VALUE; 582 goto Exit; 583 } 584 585 // further format checks are performed by createTrack_l() depending on the thread type 586 if (!audio_is_valid_format(format)) { 587 ALOGE("createTrack() invalid format %#x", format); 588 lStatus = BAD_VALUE; 589 goto Exit; 590 } 591 592 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 593 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 594 lStatus = BAD_VALUE; 595 goto Exit; 596 } 597 598 { 599 Mutex::Autolock _l(mLock); 600 PlaybackThread *thread = checkPlaybackThread_l(output); 601 if (thread == NULL) { 602 ALOGE("no playback thread found for output handle %d", output); 603 lStatus = BAD_VALUE; 604 goto Exit; 605 } 606 607 pid_t pid = IPCThreadState::self()->getCallingPid(); 608 client = registerPid(pid); 609 610 PlaybackThread *effectThread = NULL; 611 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 612 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 613 ALOGE("createTrack() invalid session ID %d", *sessionId); 614 lStatus = BAD_VALUE; 615 goto Exit; 616 } 617 lSessionId = *sessionId; 618 // check if an effect chain with the same session ID is present on another 619 // output thread and move it here. 620 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 621 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 622 if (mPlaybackThreads.keyAt(i) != output) { 623 uint32_t sessions = t->hasAudioSession(lSessionId); 624 if (sessions & PlaybackThread::EFFECT_SESSION) { 625 effectThread = t.get(); 626 break; 627 } 628 } 629 } 630 } else { 631 // if no audio session id is provided, create one here 632 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 633 if (sessionId != NULL) { 634 *sessionId = lSessionId; 635 } 636 } 637 ALOGV("createTrack() lSessionId: %d", lSessionId); 638 639 track = thread->createTrack_l(client, streamType, sampleRate, format, 640 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 641 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 642 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 643 644 // move effect chain to this output thread if an effect on same session was waiting 645 // for a track to be created 646 if (lStatus == NO_ERROR && effectThread != NULL) { 647 // no risk of deadlock because AudioFlinger::mLock is held 648 Mutex::Autolock _dl(thread->mLock); 649 Mutex::Autolock _sl(effectThread->mLock); 650 moveEffectChain_l(lSessionId, effectThread, thread, true); 651 } 652 653 // Look for sync events awaiting for a session to be used. 654 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 655 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 656 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 657 if (lStatus == NO_ERROR) { 658 (void) track->setSyncEvent(mPendingSyncEvents[i]); 659 } else { 660 mPendingSyncEvents[i]->cancel(); 661 } 662 mPendingSyncEvents.removeAt(i); 663 i--; 664 } 665 } 666 } 667 668 setAudioHwSyncForSession_l(thread, lSessionId); 669 } 670 671 if (lStatus != NO_ERROR) { 672 // remove local strong reference to Client before deleting the Track so that the 673 // Client destructor is called by the TrackBase destructor with mClientLock held 674 // Don't hold mClientLock when releasing the reference on the track as the 675 // destructor will acquire it. 676 { 677 Mutex::Autolock _cl(mClientLock); 678 client.clear(); 679 } 680 track.clear(); 681 goto Exit; 682 } 683 684 // return handle to client 685 trackHandle = new TrackHandle(track); 686 687Exit: 688 *status = lStatus; 689 return trackHandle; 690} 691 692uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 693{ 694 Mutex::Autolock _l(mLock); 695 ThreadBase *thread = checkThread_l(ioHandle); 696 if (thread == NULL) { 697 ALOGW("sampleRate() unknown thread %d", ioHandle); 698 return 0; 699 } 700 return thread->sampleRate(); 701} 702 703audio_format_t AudioFlinger::format(audio_io_handle_t output) const 704{ 705 Mutex::Autolock _l(mLock); 706 PlaybackThread *thread = checkPlaybackThread_l(output); 707 if (thread == NULL) { 708 ALOGW("format() unknown thread %d", output); 709 return AUDIO_FORMAT_INVALID; 710 } 711 return thread->format(); 712} 713 714size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 715{ 716 Mutex::Autolock _l(mLock); 717 ThreadBase *thread = checkThread_l(ioHandle); 718 if (thread == NULL) { 719 ALOGW("frameCount() unknown thread %d", ioHandle); 720 return 0; 721 } 722 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 723 // should examine all callers and fix them to handle smaller counts 724 return thread->frameCount(); 725} 726 727uint32_t AudioFlinger::latency(audio_io_handle_t output) const 728{ 729 Mutex::Autolock _l(mLock); 730 PlaybackThread *thread = checkPlaybackThread_l(output); 731 if (thread == NULL) { 732 ALOGW("latency(): no playback thread found for output handle %d", output); 733 return 0; 734 } 735 return thread->latency(); 736} 737 738status_t AudioFlinger::setMasterVolume(float value) 739{ 740 status_t ret = initCheck(); 741 if (ret != NO_ERROR) { 742 return ret; 743 } 744 745 // check calling permissions 746 if (!settingsAllowed()) { 747 return PERMISSION_DENIED; 748 } 749 750 Mutex::Autolock _l(mLock); 751 mMasterVolume = value; 752 753 // Set master volume in the HALs which support it. 754 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 755 AutoMutex lock(mHardwareLock); 756 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 757 758 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 759 if (dev->canSetMasterVolume()) { 760 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 761 } 762 mHardwareStatus = AUDIO_HW_IDLE; 763 } 764 765 // Now set the master volume in each playback thread. Playback threads 766 // assigned to HALs which do not have master volume support will apply 767 // master volume during the mix operation. Threads with HALs which do 768 // support master volume will simply ignore the setting. 769 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 770 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 771 continue; 772 } 773 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 774 } 775 776 return NO_ERROR; 777} 778 779status_t AudioFlinger::setMode(audio_mode_t mode) 780{ 781 status_t ret = initCheck(); 782 if (ret != NO_ERROR) { 783 return ret; 784 } 785 786 // check calling permissions 787 if (!settingsAllowed()) { 788 return PERMISSION_DENIED; 789 } 790 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 791 ALOGW("Illegal value: setMode(%d)", mode); 792 return BAD_VALUE; 793 } 794 795 { // scope for the lock 796 AutoMutex lock(mHardwareLock); 797 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 798 mHardwareStatus = AUDIO_HW_SET_MODE; 799 ret = dev->set_mode(dev, mode); 800 mHardwareStatus = AUDIO_HW_IDLE; 801 } 802 803 if (NO_ERROR == ret) { 804 Mutex::Autolock _l(mLock); 805 mMode = mode; 806 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 807 mPlaybackThreads.valueAt(i)->setMode(mode); 808 } 809 810 return ret; 811} 812 813status_t AudioFlinger::setMicMute(bool state) 814{ 815 status_t ret = initCheck(); 816 if (ret != NO_ERROR) { 817 return ret; 818 } 819 820 // check calling permissions 821 if (!settingsAllowed()) { 822 return PERMISSION_DENIED; 823 } 824 825 AutoMutex lock(mHardwareLock); 826 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 827 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 828 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 829 status_t result = dev->set_mic_mute(dev, state); 830 if (result != NO_ERROR) { 831 ret = result; 832 } 833 } 834 mHardwareStatus = AUDIO_HW_IDLE; 835 return ret; 836} 837 838bool AudioFlinger::getMicMute() const 839{ 840 status_t ret = initCheck(); 841 if (ret != NO_ERROR) { 842 return false; 843 } 844 bool mute = true; 845 bool state = AUDIO_MODE_INVALID; 846 AutoMutex lock(mHardwareLock); 847 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 848 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 849 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 850 status_t result = dev->get_mic_mute(dev, &state); 851 if (result == NO_ERROR) { 852 mute = mute && state; 853 } 854 } 855 mHardwareStatus = AUDIO_HW_IDLE; 856 857 return mute; 858} 859 860status_t AudioFlinger::setMasterMute(bool muted) 861{ 862 status_t ret = initCheck(); 863 if (ret != NO_ERROR) { 864 return ret; 865 } 866 867 // check calling permissions 868 if (!settingsAllowed()) { 869 return PERMISSION_DENIED; 870 } 871 872 Mutex::Autolock _l(mLock); 873 mMasterMute = muted; 874 875 // Set master mute in the HALs which support it. 876 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 877 AutoMutex lock(mHardwareLock); 878 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 879 880 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 881 if (dev->canSetMasterMute()) { 882 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 883 } 884 mHardwareStatus = AUDIO_HW_IDLE; 885 } 886 887 // Now set the master mute in each playback thread. Playback threads 888 // assigned to HALs which do not have master mute support will apply master 889 // mute during the mix operation. Threads with HALs which do support master 890 // mute will simply ignore the setting. 891 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 892 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 893 continue; 894 } 895 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 896 } 897 898 return NO_ERROR; 899} 900 901float AudioFlinger::masterVolume() const 902{ 903 Mutex::Autolock _l(mLock); 904 return masterVolume_l(); 905} 906 907bool AudioFlinger::masterMute() const 908{ 909 Mutex::Autolock _l(mLock); 910 return masterMute_l(); 911} 912 913float AudioFlinger::masterVolume_l() const 914{ 915 return mMasterVolume; 916} 917 918bool AudioFlinger::masterMute_l() const 919{ 920 return mMasterMute; 921} 922 923status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 924{ 925 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 926 ALOGW("setStreamVolume() invalid stream %d", stream); 927 return BAD_VALUE; 928 } 929 pid_t caller = IPCThreadState::self()->getCallingPid(); 930 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 931 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 932 return PERMISSION_DENIED; 933 } 934 935 return NO_ERROR; 936} 937 938status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 939 audio_io_handle_t output) 940{ 941 // check calling permissions 942 if (!settingsAllowed()) { 943 return PERMISSION_DENIED; 944 } 945 946 status_t status = checkStreamType(stream); 947 if (status != NO_ERROR) { 948 return status; 949 } 950 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 951 952 AutoMutex lock(mLock); 953 PlaybackThread *thread = NULL; 954 if (output != AUDIO_IO_HANDLE_NONE) { 955 thread = checkPlaybackThread_l(output); 956 if (thread == NULL) { 957 return BAD_VALUE; 958 } 959 } 960 961 mStreamTypes[stream].volume = value; 962 963 if (thread == NULL) { 964 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 965 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 966 } 967 } else { 968 thread->setStreamVolume(stream, value); 969 } 970 971 return NO_ERROR; 972} 973 974status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 975{ 976 // check calling permissions 977 if (!settingsAllowed()) { 978 return PERMISSION_DENIED; 979 } 980 981 status_t status = checkStreamType(stream); 982 if (status != NO_ERROR) { 983 return status; 984 } 985 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 986 987 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 988 ALOGE("setStreamMute() invalid stream %d", stream); 989 return BAD_VALUE; 990 } 991 992 AutoMutex lock(mLock); 993 mStreamTypes[stream].mute = muted; 994 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 995 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 996 997 return NO_ERROR; 998} 999 1000float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1001{ 1002 status_t status = checkStreamType(stream); 1003 if (status != NO_ERROR) { 1004 return 0.0f; 1005 } 1006 1007 AutoMutex lock(mLock); 1008 float volume; 1009 if (output != AUDIO_IO_HANDLE_NONE) { 1010 PlaybackThread *thread = checkPlaybackThread_l(output); 1011 if (thread == NULL) { 1012 return 0.0f; 1013 } 1014 volume = thread->streamVolume(stream); 1015 } else { 1016 volume = streamVolume_l(stream); 1017 } 1018 1019 return volume; 1020} 1021 1022bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1023{ 1024 status_t status = checkStreamType(stream); 1025 if (status != NO_ERROR) { 1026 return true; 1027 } 1028 1029 AutoMutex lock(mLock); 1030 return streamMute_l(stream); 1031} 1032 1033 1034void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1035{ 1036 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1037 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1038 } 1039} 1040 1041status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1042{ 1043 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1044 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1045 1046 // check calling permissions 1047 if (!settingsAllowed()) { 1048 return PERMISSION_DENIED; 1049 } 1050 1051 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1052 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1053 Mutex::Autolock _l(mLock); 1054 status_t final_result = NO_ERROR; 1055 { 1056 AutoMutex lock(mHardwareLock); 1057 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1058 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1059 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1060 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1061 final_result = result ?: final_result; 1062 } 1063 mHardwareStatus = AUDIO_HW_IDLE; 1064 } 1065 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1066 AudioParameter param = AudioParameter(keyValuePairs); 1067 String8 value; 1068 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1069 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1070 if (mBtNrecIsOff != btNrecIsOff) { 1071 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1072 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1073 audio_devices_t device = thread->inDevice(); 1074 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1075 // collect all of the thread's session IDs 1076 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1077 // suspend effects associated with those session IDs 1078 for (size_t j = 0; j < ids.size(); ++j) { 1079 audio_session_t sessionId = ids.keyAt(j); 1080 thread->setEffectSuspended(FX_IID_AEC, 1081 suspend, 1082 sessionId); 1083 thread->setEffectSuspended(FX_IID_NS, 1084 suspend, 1085 sessionId); 1086 } 1087 } 1088 mBtNrecIsOff = btNrecIsOff; 1089 } 1090 } 1091 String8 screenState; 1092 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1093 bool isOff = screenState == "off"; 1094 if (isOff != (AudioFlinger::mScreenState & 1)) { 1095 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1096 } 1097 } 1098 return final_result; 1099 } 1100 1101 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1102 // and the thread is exited once the lock is released 1103 sp<ThreadBase> thread; 1104 { 1105 Mutex::Autolock _l(mLock); 1106 thread = checkPlaybackThread_l(ioHandle); 1107 if (thread == 0) { 1108 thread = checkRecordThread_l(ioHandle); 1109 } else if (thread == primaryPlaybackThread_l()) { 1110 // indicate output device change to all input threads for pre processing 1111 AudioParameter param = AudioParameter(keyValuePairs); 1112 int value; 1113 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1114 (value != 0)) { 1115 broacastParametersToRecordThreads_l(keyValuePairs); 1116 } 1117 } 1118 } 1119 if (thread != 0) { 1120 return thread->setParameters(keyValuePairs); 1121 } 1122 return BAD_VALUE; 1123} 1124 1125String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1126{ 1127 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1128 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1129 1130 Mutex::Autolock _l(mLock); 1131 1132 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1133 String8 out_s8; 1134 1135 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1136 char *s; 1137 { 1138 AutoMutex lock(mHardwareLock); 1139 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1140 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1141 s = dev->get_parameters(dev, keys.string()); 1142 mHardwareStatus = AUDIO_HW_IDLE; 1143 } 1144 out_s8 += String8(s ? s : ""); 1145 free(s); 1146 } 1147 return out_s8; 1148 } 1149 1150 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1151 if (playbackThread != NULL) { 1152 return playbackThread->getParameters(keys); 1153 } 1154 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1155 if (recordThread != NULL) { 1156 return recordThread->getParameters(keys); 1157 } 1158 return String8(""); 1159} 1160 1161size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1162 audio_channel_mask_t channelMask) const 1163{ 1164 status_t ret = initCheck(); 1165 if (ret != NO_ERROR) { 1166 return 0; 1167 } 1168 if ((sampleRate == 0) || 1169 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1170 !audio_is_input_channel(channelMask)) { 1171 return 0; 1172 } 1173 1174 AutoMutex lock(mHardwareLock); 1175 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1176 audio_config_t config, proposed; 1177 memset(&proposed, 0, sizeof(proposed)); 1178 proposed.sample_rate = sampleRate; 1179 proposed.channel_mask = channelMask; 1180 proposed.format = format; 1181 1182 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1183 size_t frames; 1184 for (;;) { 1185 // Note: config is currently a const parameter for get_input_buffer_size() 1186 // but we use a copy from proposed in case config changes from the call. 1187 config = proposed; 1188 frames = dev->get_input_buffer_size(dev, &config); 1189 if (frames != 0) { 1190 break; // hal success, config is the result 1191 } 1192 // change one parameter of the configuration each iteration to a more "common" value 1193 // to see if the device will support it. 1194 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1195 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1196 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1197 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1198 } else { 1199 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1200 "format %#x, channelMask 0x%X", 1201 sampleRate, format, channelMask); 1202 break; // retries failed, break out of loop with frames == 0. 1203 } 1204 } 1205 mHardwareStatus = AUDIO_HW_IDLE; 1206 if (frames > 0 && config.sample_rate != sampleRate) { 1207 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1208 } 1209 return frames; // may be converted to bytes at the Java level. 1210} 1211 1212uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1213{ 1214 Mutex::Autolock _l(mLock); 1215 1216 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1217 if (recordThread != NULL) { 1218 return recordThread->getInputFramesLost(); 1219 } 1220 return 0; 1221} 1222 1223status_t AudioFlinger::setVoiceVolume(float value) 1224{ 1225 status_t ret = initCheck(); 1226 if (ret != NO_ERROR) { 1227 return ret; 1228 } 1229 1230 // check calling permissions 1231 if (!settingsAllowed()) { 1232 return PERMISSION_DENIED; 1233 } 1234 1235 AutoMutex lock(mHardwareLock); 1236 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1237 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1238 ret = dev->set_voice_volume(dev, value); 1239 mHardwareStatus = AUDIO_HW_IDLE; 1240 1241 return ret; 1242} 1243 1244status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1245 audio_io_handle_t output) const 1246{ 1247 Mutex::Autolock _l(mLock); 1248 1249 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1250 if (playbackThread != NULL) { 1251 return playbackThread->getRenderPosition(halFrames, dspFrames); 1252 } 1253 1254 return BAD_VALUE; 1255} 1256 1257void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1258{ 1259 Mutex::Autolock _l(mLock); 1260 if (client == 0) { 1261 return; 1262 } 1263 pid_t pid = IPCThreadState::self()->getCallingPid(); 1264 { 1265 Mutex::Autolock _cl(mClientLock); 1266 if (mNotificationClients.indexOfKey(pid) < 0) { 1267 sp<NotificationClient> notificationClient = new NotificationClient(this, 1268 client, 1269 pid); 1270 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1271 1272 mNotificationClients.add(pid, notificationClient); 1273 1274 sp<IBinder> binder = IInterface::asBinder(client); 1275 binder->linkToDeath(notificationClient); 1276 } 1277 } 1278 1279 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1280 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1281 // the config change is always sent from playback or record threads to avoid deadlock 1282 // with AudioSystem::gLock 1283 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1284 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1285 } 1286 1287 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1288 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1289 } 1290} 1291 1292void AudioFlinger::removeNotificationClient(pid_t pid) 1293{ 1294 Mutex::Autolock _l(mLock); 1295 { 1296 Mutex::Autolock _cl(mClientLock); 1297 mNotificationClients.removeItem(pid); 1298 } 1299 1300 ALOGV("%d died, releasing its sessions", pid); 1301 size_t num = mAudioSessionRefs.size(); 1302 bool removed = false; 1303 for (size_t i = 0; i< num; ) { 1304 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1305 ALOGV(" pid %d @ %zu", ref->mPid, i); 1306 if (ref->mPid == pid) { 1307 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1308 mAudioSessionRefs.removeAt(i); 1309 delete ref; 1310 removed = true; 1311 num--; 1312 } else { 1313 i++; 1314 } 1315 } 1316 if (removed) { 1317 purgeStaleEffects_l(); 1318 } 1319} 1320 1321void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1322 const sp<AudioIoDescriptor>& ioDesc, 1323 pid_t pid) 1324{ 1325 Mutex::Autolock _l(mClientLock); 1326 size_t size = mNotificationClients.size(); 1327 for (size_t i = 0; i < size; i++) { 1328 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1329 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1330 } 1331 } 1332} 1333 1334// removeClient_l() must be called with AudioFlinger::mClientLock held 1335void AudioFlinger::removeClient_l(pid_t pid) 1336{ 1337 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1338 IPCThreadState::self()->getCallingPid()); 1339 mClients.removeItem(pid); 1340} 1341 1342// getEffectThread_l() must be called with AudioFlinger::mLock held 1343sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1344 int EffectId) 1345{ 1346 sp<PlaybackThread> thread; 1347 1348 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1349 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1350 ALOG_ASSERT(thread == 0); 1351 thread = mPlaybackThreads.valueAt(i); 1352 } 1353 } 1354 1355 return thread; 1356} 1357 1358 1359 1360// ---------------------------------------------------------------------------- 1361 1362AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1363 : RefBase(), 1364 mAudioFlinger(audioFlinger), 1365 mPid(pid) 1366{ 1367 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1368 heapSize *= 1024; 1369 if (!heapSize) { 1370 heapSize = kClientSharedHeapSizeBytes; 1371 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1372 // invalidated tracks 1373 if (!audioFlinger->isLowRamDevice()) { 1374 heapSize *= kClientSharedHeapSizeMultiplier; 1375 } 1376 } 1377 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1378} 1379 1380// Client destructor must be called with AudioFlinger::mClientLock held 1381AudioFlinger::Client::~Client() 1382{ 1383 mAudioFlinger->removeClient_l(mPid); 1384} 1385 1386sp<MemoryDealer> AudioFlinger::Client::heap() const 1387{ 1388 return mMemoryDealer; 1389} 1390 1391// ---------------------------------------------------------------------------- 1392 1393AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1394 const sp<IAudioFlingerClient>& client, 1395 pid_t pid) 1396 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1397{ 1398} 1399 1400AudioFlinger::NotificationClient::~NotificationClient() 1401{ 1402} 1403 1404void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1405{ 1406 sp<NotificationClient> keep(this); 1407 mAudioFlinger->removeNotificationClient(mPid); 1408} 1409 1410 1411// ---------------------------------------------------------------------------- 1412 1413sp<IAudioRecord> AudioFlinger::openRecord( 1414 audio_io_handle_t input, 1415 uint32_t sampleRate, 1416 audio_format_t format, 1417 audio_channel_mask_t channelMask, 1418 const String16& opPackageName, 1419 size_t *frameCount, 1420 IAudioFlinger::track_flags_t *flags, 1421 pid_t tid, 1422 int clientUid, 1423 audio_session_t *sessionId, 1424 size_t *notificationFrames, 1425 sp<IMemory>& cblk, 1426 sp<IMemory>& buffers, 1427 status_t *status) 1428{ 1429 sp<RecordThread::RecordTrack> recordTrack; 1430 sp<RecordHandle> recordHandle; 1431 sp<Client> client; 1432 status_t lStatus; 1433 audio_session_t lSessionId; 1434 1435 cblk.clear(); 1436 buffers.clear(); 1437 1438 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1439 if (!isTrustedCallingUid(callingUid)) { 1440 ALOGW_IF((uid_t)clientUid != callingUid, 1441 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1442 clientUid = callingUid; 1443 } 1444 1445 // check calling permissions 1446 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1447 ALOGE("openRecord() permission denied: recording not allowed"); 1448 lStatus = PERMISSION_DENIED; 1449 goto Exit; 1450 } 1451 1452 // further sample rate checks are performed by createRecordTrack_l() 1453 if (sampleRate == 0) { 1454 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1455 lStatus = BAD_VALUE; 1456 goto Exit; 1457 } 1458 1459 // we don't yet support anything other than linear PCM 1460 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1461 ALOGE("openRecord() invalid format %#x", format); 1462 lStatus = BAD_VALUE; 1463 goto Exit; 1464 } 1465 1466 // further channel mask checks are performed by createRecordTrack_l() 1467 if (!audio_is_input_channel(channelMask)) { 1468 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1469 lStatus = BAD_VALUE; 1470 goto Exit; 1471 } 1472 1473 { 1474 Mutex::Autolock _l(mLock); 1475 RecordThread *thread = checkRecordThread_l(input); 1476 if (thread == NULL) { 1477 ALOGE("openRecord() checkRecordThread_l failed"); 1478 lStatus = BAD_VALUE; 1479 goto Exit; 1480 } 1481 1482 pid_t pid = IPCThreadState::self()->getCallingPid(); 1483 client = registerPid(pid); 1484 1485 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1486 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1487 lStatus = BAD_VALUE; 1488 goto Exit; 1489 } 1490 lSessionId = *sessionId; 1491 } else { 1492 // if no audio session id is provided, create one here 1493 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1494 if (sessionId != NULL) { 1495 *sessionId = lSessionId; 1496 } 1497 } 1498 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1499 1500 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1501 frameCount, lSessionId, notificationFrames, 1502 clientUid, flags, tid, &lStatus); 1503 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1504 1505 if (lStatus == NO_ERROR) { 1506 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1507 // session and move it to this thread. 1508 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1509 if (chain != 0) { 1510 Mutex::Autolock _l(thread->mLock); 1511 thread->addEffectChain_l(chain); 1512 } 1513 } 1514 } 1515 1516 if (lStatus != NO_ERROR) { 1517 // remove local strong reference to Client before deleting the RecordTrack so that the 1518 // Client destructor is called by the TrackBase destructor with mClientLock held 1519 // Don't hold mClientLock when releasing the reference on the track as the 1520 // destructor will acquire it. 1521 { 1522 Mutex::Autolock _cl(mClientLock); 1523 client.clear(); 1524 } 1525 recordTrack.clear(); 1526 goto Exit; 1527 } 1528 1529 cblk = recordTrack->getCblk(); 1530 buffers = recordTrack->getBuffers(); 1531 1532 // return handle to client 1533 recordHandle = new RecordHandle(recordTrack); 1534 1535Exit: 1536 *status = lStatus; 1537 return recordHandle; 1538} 1539 1540 1541 1542// ---------------------------------------------------------------------------- 1543 1544audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1545{ 1546 if (name == NULL) { 1547 return 0; 1548 } 1549 if (!settingsAllowed()) { 1550 return 0; 1551 } 1552 Mutex::Autolock _l(mLock); 1553 return loadHwModule_l(name); 1554} 1555 1556// loadHwModule_l() must be called with AudioFlinger::mLock held 1557audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1558{ 1559 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1560 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1561 ALOGW("loadHwModule() module %s already loaded", name); 1562 return mAudioHwDevs.keyAt(i); 1563 } 1564 } 1565 1566 audio_hw_device_t *dev; 1567 1568 int rc = load_audio_interface(name, &dev); 1569 if (rc) { 1570 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1571 return 0; 1572 } 1573 1574 mHardwareStatus = AUDIO_HW_INIT; 1575 rc = dev->init_check(dev); 1576 mHardwareStatus = AUDIO_HW_IDLE; 1577 if (rc) { 1578 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1579 return 0; 1580 } 1581 1582 // Check and cache this HAL's level of support for master mute and master 1583 // volume. If this is the first HAL opened, and it supports the get 1584 // methods, use the initial values provided by the HAL as the current 1585 // master mute and volume settings. 1586 1587 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1588 { // scope for auto-lock pattern 1589 AutoMutex lock(mHardwareLock); 1590 1591 if (0 == mAudioHwDevs.size()) { 1592 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1593 if (NULL != dev->get_master_volume) { 1594 float mv; 1595 if (OK == dev->get_master_volume(dev, &mv)) { 1596 mMasterVolume = mv; 1597 } 1598 } 1599 1600 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1601 if (NULL != dev->get_master_mute) { 1602 bool mm; 1603 if (OK == dev->get_master_mute(dev, &mm)) { 1604 mMasterMute = mm; 1605 } 1606 } 1607 } 1608 1609 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1610 if ((NULL != dev->set_master_volume) && 1611 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1612 flags = static_cast<AudioHwDevice::Flags>(flags | 1613 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1614 } 1615 1616 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1617 if ((NULL != dev->set_master_mute) && 1618 (OK == dev->set_master_mute(dev, mMasterMute))) { 1619 flags = static_cast<AudioHwDevice::Flags>(flags | 1620 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1621 } 1622 1623 mHardwareStatus = AUDIO_HW_IDLE; 1624 } 1625 1626 audio_module_handle_t handle = nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1627 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1628 1629 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1630 name, dev->common.module->name, dev->common.module->id, handle); 1631 1632 return handle; 1633 1634} 1635 1636// ---------------------------------------------------------------------------- 1637 1638uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1639{ 1640 Mutex::Autolock _l(mLock); 1641 PlaybackThread *thread = primaryPlaybackThread_l(); 1642 return thread != NULL ? thread->sampleRate() : 0; 1643} 1644 1645size_t AudioFlinger::getPrimaryOutputFrameCount() 1646{ 1647 Mutex::Autolock _l(mLock); 1648 PlaybackThread *thread = primaryPlaybackThread_l(); 1649 return thread != NULL ? thread->frameCountHAL() : 0; 1650} 1651 1652// ---------------------------------------------------------------------------- 1653 1654status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1655{ 1656 uid_t uid = IPCThreadState::self()->getCallingUid(); 1657 if (uid != AID_SYSTEM) { 1658 return PERMISSION_DENIED; 1659 } 1660 Mutex::Autolock _l(mLock); 1661 if (mIsDeviceTypeKnown) { 1662 return INVALID_OPERATION; 1663 } 1664 mIsLowRamDevice = isLowRamDevice; 1665 mIsDeviceTypeKnown = true; 1666 return NO_ERROR; 1667} 1668 1669audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1670{ 1671 Mutex::Autolock _l(mLock); 1672 1673 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1674 if (index >= 0) { 1675 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1676 mHwAvSyncIds.valueAt(index), sessionId); 1677 return mHwAvSyncIds.valueAt(index); 1678 } 1679 1680 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1681 if (dev == NULL) { 1682 return AUDIO_HW_SYNC_INVALID; 1683 } 1684 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1685 AudioParameter param = AudioParameter(String8(reply)); 1686 free(reply); 1687 1688 int value; 1689 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1690 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1691 return AUDIO_HW_SYNC_INVALID; 1692 } 1693 1694 // allow only one session for a given HW A/V sync ID. 1695 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1696 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1697 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1698 value, mHwAvSyncIds.keyAt(i)); 1699 mHwAvSyncIds.removeItemsAt(i); 1700 break; 1701 } 1702 } 1703 1704 mHwAvSyncIds.add(sessionId, value); 1705 1706 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1707 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1708 uint32_t sessions = thread->hasAudioSession(sessionId); 1709 if (sessions & PlaybackThread::TRACK_SESSION) { 1710 AudioParameter param = AudioParameter(); 1711 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1712 thread->setParameters(param.toString()); 1713 break; 1714 } 1715 } 1716 1717 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1718 return (audio_hw_sync_t)value; 1719} 1720 1721status_t AudioFlinger::systemReady() 1722{ 1723 Mutex::Autolock _l(mLock); 1724 ALOGI("%s", __FUNCTION__); 1725 if (mSystemReady) { 1726 ALOGW("%s called twice", __FUNCTION__); 1727 return NO_ERROR; 1728 } 1729 mSystemReady = true; 1730 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1731 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1732 thread->systemReady(); 1733 } 1734 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1735 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1736 thread->systemReady(); 1737 } 1738 return NO_ERROR; 1739} 1740 1741// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1742void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1743{ 1744 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1745 if (index >= 0) { 1746 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1747 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1748 AudioParameter param = AudioParameter(); 1749 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1750 thread->setParameters(param.toString()); 1751 } 1752} 1753 1754 1755// ---------------------------------------------------------------------------- 1756 1757 1758sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1759 audio_io_handle_t *output, 1760 audio_config_t *config, 1761 audio_devices_t devices, 1762 const String8& address, 1763 audio_output_flags_t flags) 1764{ 1765 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1766 if (outHwDev == NULL) { 1767 return 0; 1768 } 1769 1770 if (*output == AUDIO_IO_HANDLE_NONE) { 1771 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1772 } else { 1773 // Audio Policy does not currently request a specific output handle. 1774 // If this is ever needed, see openInput_l() for example code. 1775 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1776 return 0; 1777 } 1778 1779 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1780 1781 // FOR TESTING ONLY: 1782 // This if statement allows overriding the audio policy settings 1783 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1784 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1785 // Check only for Normal Mixing mode 1786 if (kEnableExtendedPrecision) { 1787 // Specify format (uncomment one below to choose) 1788 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1789 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1790 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1791 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1792 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1793 } 1794 if (kEnableExtendedChannels) { 1795 // Specify channel mask (uncomment one below to choose) 1796 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1797 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1798 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1799 } 1800 } 1801 1802 AudioStreamOut *outputStream = NULL; 1803 status_t status = outHwDev->openOutputStream( 1804 &outputStream, 1805 *output, 1806 devices, 1807 flags, 1808 config, 1809 address.string()); 1810 1811 mHardwareStatus = AUDIO_HW_IDLE; 1812 1813 if (status == NO_ERROR) { 1814 1815 PlaybackThread *thread; 1816 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1817 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady, 1818 config->offload_info.bit_rate); 1819 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1820 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1821 || !isValidPcmSinkFormat(config->format) 1822 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1823 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1824 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1825 } else { 1826 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1827 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1828 } 1829 mPlaybackThreads.add(*output, thread); 1830 return thread; 1831 } 1832 1833 return 0; 1834} 1835 1836status_t AudioFlinger::openOutput(audio_module_handle_t module, 1837 audio_io_handle_t *output, 1838 audio_config_t *config, 1839 audio_devices_t *devices, 1840 const String8& address, 1841 uint32_t *latencyMs, 1842 audio_output_flags_t flags) 1843{ 1844 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1845 module, 1846 (devices != NULL) ? *devices : 0, 1847 config->sample_rate, 1848 config->format, 1849 config->channel_mask, 1850 flags); 1851 1852 if (*devices == AUDIO_DEVICE_NONE) { 1853 return BAD_VALUE; 1854 } 1855 1856 Mutex::Autolock _l(mLock); 1857 1858 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1859 if (thread != 0) { 1860 *latencyMs = thread->latency(); 1861 1862 // notify client processes of the new output creation 1863 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1864 1865 // the first primary output opened designates the primary hw device 1866 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1867 ALOGI("Using module %d has the primary audio interface", module); 1868 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1869 1870 AutoMutex lock(mHardwareLock); 1871 mHardwareStatus = AUDIO_HW_SET_MODE; 1872 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1873 mHardwareStatus = AUDIO_HW_IDLE; 1874 } 1875 return NO_ERROR; 1876 } 1877 1878 return NO_INIT; 1879} 1880 1881audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1882 audio_io_handle_t output2) 1883{ 1884 Mutex::Autolock _l(mLock); 1885 MixerThread *thread1 = checkMixerThread_l(output1); 1886 MixerThread *thread2 = checkMixerThread_l(output2); 1887 1888 if (thread1 == NULL || thread2 == NULL) { 1889 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1890 output2); 1891 return AUDIO_IO_HANDLE_NONE; 1892 } 1893 1894 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1895 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1896 thread->addOutputTrack(thread2); 1897 mPlaybackThreads.add(id, thread); 1898 // notify client processes of the new output creation 1899 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1900 return id; 1901} 1902 1903status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1904{ 1905 return closeOutput_nonvirtual(output); 1906} 1907 1908status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1909{ 1910 // keep strong reference on the playback thread so that 1911 // it is not destroyed while exit() is executed 1912 sp<PlaybackThread> thread; 1913 { 1914 Mutex::Autolock _l(mLock); 1915 thread = checkPlaybackThread_l(output); 1916 if (thread == NULL) { 1917 return BAD_VALUE; 1918 } 1919 1920 ALOGV("closeOutput() %d", output); 1921 1922 if (thread->type() == ThreadBase::MIXER) { 1923 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1924 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1925 DuplicatingThread *dupThread = 1926 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1927 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1928 } 1929 } 1930 } 1931 1932 1933 mPlaybackThreads.removeItem(output); 1934 // save all effects to the default thread 1935 if (mPlaybackThreads.size()) { 1936 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1937 if (dstThread != NULL) { 1938 // audioflinger lock is held here so the acquisition order of thread locks does not 1939 // matter 1940 Mutex::Autolock _dl(dstThread->mLock); 1941 Mutex::Autolock _sl(thread->mLock); 1942 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1943 for (size_t i = 0; i < effectChains.size(); i ++) { 1944 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1945 } 1946 } 1947 } 1948 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1949 ioDesc->mIoHandle = output; 1950 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1951 } 1952 thread->exit(); 1953 // The thread entity (active unit of execution) is no longer running here, 1954 // but the ThreadBase container still exists. 1955 1956 if (!thread->isDuplicating()) { 1957 closeOutputFinish(thread); 1958 } 1959 1960 return NO_ERROR; 1961} 1962 1963void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1964{ 1965 AudioStreamOut *out = thread->clearOutput(); 1966 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1967 // from now on thread->mOutput is NULL 1968 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1969 delete out; 1970} 1971 1972void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1973{ 1974 mPlaybackThreads.removeItem(thread->mId); 1975 thread->exit(); 1976 closeOutputFinish(thread); 1977} 1978 1979status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1980{ 1981 Mutex::Autolock _l(mLock); 1982 PlaybackThread *thread = checkPlaybackThread_l(output); 1983 1984 if (thread == NULL) { 1985 return BAD_VALUE; 1986 } 1987 1988 ALOGV("suspendOutput() %d", output); 1989 thread->suspend(); 1990 1991 return NO_ERROR; 1992} 1993 1994status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1995{ 1996 Mutex::Autolock _l(mLock); 1997 PlaybackThread *thread = checkPlaybackThread_l(output); 1998 1999 if (thread == NULL) { 2000 return BAD_VALUE; 2001 } 2002 2003 ALOGV("restoreOutput() %d", output); 2004 2005 thread->restore(); 2006 2007 return NO_ERROR; 2008} 2009 2010status_t AudioFlinger::openInput(audio_module_handle_t module, 2011 audio_io_handle_t *input, 2012 audio_config_t *config, 2013 audio_devices_t *devices, 2014 const String8& address, 2015 audio_source_t source, 2016 audio_input_flags_t flags) 2017{ 2018 Mutex::Autolock _l(mLock); 2019 2020 if (*devices == AUDIO_DEVICE_NONE) { 2021 return BAD_VALUE; 2022 } 2023 2024 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2025 2026 if (thread != 0) { 2027 // notify client processes of the new input creation 2028 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2029 return NO_ERROR; 2030 } 2031 return NO_INIT; 2032} 2033 2034sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2035 audio_io_handle_t *input, 2036 audio_config_t *config, 2037 audio_devices_t devices, 2038 const String8& address, 2039 audio_source_t source, 2040 audio_input_flags_t flags) 2041{ 2042 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2043 if (inHwDev == NULL) { 2044 *input = AUDIO_IO_HANDLE_NONE; 2045 return 0; 2046 } 2047 2048 // Audio Policy can request a specific handle for hardware hotword. 2049 // The goal here is not to re-open an already opened input. 2050 // It is to use a pre-assigned I/O handle. 2051 if (*input == AUDIO_IO_HANDLE_NONE) { 2052 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2053 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2054 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2055 return 0; 2056 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2057 // This should not happen in a transient state with current design. 2058 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2059 return 0; 2060 } 2061 2062 audio_config_t halconfig = *config; 2063 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2064 audio_stream_in_t *inStream = NULL; 2065 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2066 &inStream, flags, address.string(), source); 2067 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2068 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2069 inStream, 2070 halconfig.sample_rate, 2071 halconfig.format, 2072 halconfig.channel_mask, 2073 flags, 2074 status, address.string()); 2075 2076 // If the input could not be opened with the requested parameters and we can handle the 2077 // conversion internally, try to open again with the proposed parameters. 2078 if (status == BAD_VALUE && 2079 audio_is_linear_pcm(config->format) && 2080 audio_is_linear_pcm(halconfig.format) && 2081 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2082 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2083 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2084 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2085 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2086 inStream = NULL; 2087 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2088 &inStream, flags, address.string(), source); 2089 // FIXME log this new status; HAL should not propose any further changes 2090 } 2091 2092 if (status == NO_ERROR && inStream != NULL) { 2093 2094#ifdef TEE_SINK 2095 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2096 // or (re-)create if current Pipe is idle and does not match the new format 2097 sp<NBAIO_Sink> teeSink; 2098 enum { 2099 TEE_SINK_NO, // don't copy input 2100 TEE_SINK_NEW, // copy input using a new pipe 2101 TEE_SINK_OLD, // copy input using an existing pipe 2102 } kind; 2103 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2104 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2105 if (!mTeeSinkInputEnabled) { 2106 kind = TEE_SINK_NO; 2107 } else if (!Format_isValid(format)) { 2108 kind = TEE_SINK_NO; 2109 } else if (mRecordTeeSink == 0) { 2110 kind = TEE_SINK_NEW; 2111 } else if (mRecordTeeSink->getStrongCount() != 1) { 2112 kind = TEE_SINK_NO; 2113 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2114 kind = TEE_SINK_OLD; 2115 } else { 2116 kind = TEE_SINK_NEW; 2117 } 2118 switch (kind) { 2119 case TEE_SINK_NEW: { 2120 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2121 size_t numCounterOffers = 0; 2122 const NBAIO_Format offers[1] = {format}; 2123 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2124 ALOG_ASSERT(index == 0); 2125 PipeReader *pipeReader = new PipeReader(*pipe); 2126 numCounterOffers = 0; 2127 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2128 ALOG_ASSERT(index == 0); 2129 mRecordTeeSink = pipe; 2130 mRecordTeeSource = pipeReader; 2131 teeSink = pipe; 2132 } 2133 break; 2134 case TEE_SINK_OLD: 2135 teeSink = mRecordTeeSink; 2136 break; 2137 case TEE_SINK_NO: 2138 default: 2139 break; 2140 } 2141#endif 2142 2143 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2144 2145 // Start record thread 2146 // RecordThread requires both input and output device indication to forward to audio 2147 // pre processing modules 2148 sp<RecordThread> thread = new RecordThread(this, 2149 inputStream, 2150 *input, 2151 primaryOutputDevice_l(), 2152 devices, 2153 mSystemReady 2154#ifdef TEE_SINK 2155 , teeSink 2156#endif 2157 ); 2158 mRecordThreads.add(*input, thread); 2159 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2160 return thread; 2161 } 2162 2163 *input = AUDIO_IO_HANDLE_NONE; 2164 return 0; 2165} 2166 2167status_t AudioFlinger::closeInput(audio_io_handle_t input) 2168{ 2169 return closeInput_nonvirtual(input); 2170} 2171 2172status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2173{ 2174 // keep strong reference on the record thread so that 2175 // it is not destroyed while exit() is executed 2176 sp<RecordThread> thread; 2177 { 2178 Mutex::Autolock _l(mLock); 2179 thread = checkRecordThread_l(input); 2180 if (thread == 0) { 2181 return BAD_VALUE; 2182 } 2183 2184 ALOGV("closeInput() %d", input); 2185 2186 // If we still have effect chains, it means that a client still holds a handle 2187 // on at least one effect. We must either move the chain to an existing thread with the 2188 // same session ID or put it aside in case a new record thread is opened for a 2189 // new capture on the same session 2190 sp<EffectChain> chain; 2191 { 2192 Mutex::Autolock _sl(thread->mLock); 2193 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2194 // Note: maximum one chain per record thread 2195 if (effectChains.size() != 0) { 2196 chain = effectChains[0]; 2197 } 2198 } 2199 if (chain != 0) { 2200 // first check if a record thread is already opened with a client on the same session. 2201 // This should only happen in case of overlap between one thread tear down and the 2202 // creation of its replacement 2203 size_t i; 2204 for (i = 0; i < mRecordThreads.size(); i++) { 2205 sp<RecordThread> t = mRecordThreads.valueAt(i); 2206 if (t == thread) { 2207 continue; 2208 } 2209 if (t->hasAudioSession(chain->sessionId()) != 0) { 2210 Mutex::Autolock _l(t->mLock); 2211 ALOGV("closeInput() found thread %d for effect session %d", 2212 t->id(), chain->sessionId()); 2213 t->addEffectChain_l(chain); 2214 break; 2215 } 2216 } 2217 // put the chain aside if we could not find a record thread with the same session id. 2218 if (i == mRecordThreads.size()) { 2219 putOrphanEffectChain_l(chain); 2220 } 2221 } 2222 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2223 ioDesc->mIoHandle = input; 2224 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2225 mRecordThreads.removeItem(input); 2226 } 2227 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2228 // we have a different lock for notification client 2229 closeInputFinish(thread); 2230 return NO_ERROR; 2231} 2232 2233void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2234{ 2235 thread->exit(); 2236 AudioStreamIn *in = thread->clearInput(); 2237 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2238 // from now on thread->mInput is NULL 2239 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2240 delete in; 2241} 2242 2243void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2244{ 2245 mRecordThreads.removeItem(thread->mId); 2246 closeInputFinish(thread); 2247} 2248 2249status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2250{ 2251 Mutex::Autolock _l(mLock); 2252 ALOGV("invalidateStream() stream %d", stream); 2253 2254 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2255 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2256 thread->invalidateTracks(stream); 2257 } 2258 2259 return NO_ERROR; 2260} 2261 2262 2263audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2264{ 2265 return nextUniqueId(use); 2266} 2267 2268void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2269{ 2270 Mutex::Autolock _l(mLock); 2271 pid_t caller = IPCThreadState::self()->getCallingPid(); 2272 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2273 if (pid != -1 && (caller == getpid_cached)) { 2274 caller = pid; 2275 } 2276 2277 { 2278 Mutex::Autolock _cl(mClientLock); 2279 // Ignore requests received from processes not known as notification client. The request 2280 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2281 // called from a different pid leaving a stale session reference. Also we don't know how 2282 // to clear this reference if the client process dies. 2283 if (mNotificationClients.indexOfKey(caller) < 0) { 2284 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2285 return; 2286 } 2287 } 2288 2289 size_t num = mAudioSessionRefs.size(); 2290 for (size_t i = 0; i< num; i++) { 2291 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2292 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2293 ref->mCnt++; 2294 ALOGV(" incremented refcount to %d", ref->mCnt); 2295 return; 2296 } 2297 } 2298 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2299 ALOGV(" added new entry for %d", audioSession); 2300} 2301 2302void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2303{ 2304 Mutex::Autolock _l(mLock); 2305 pid_t caller = IPCThreadState::self()->getCallingPid(); 2306 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2307 if (pid != -1 && (caller == getpid_cached)) { 2308 caller = pid; 2309 } 2310 size_t num = mAudioSessionRefs.size(); 2311 for (size_t i = 0; i< num; i++) { 2312 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2313 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2314 ref->mCnt--; 2315 ALOGV(" decremented refcount to %d", ref->mCnt); 2316 if (ref->mCnt == 0) { 2317 mAudioSessionRefs.removeAt(i); 2318 delete ref; 2319 purgeStaleEffects_l(); 2320 } 2321 return; 2322 } 2323 } 2324 // If the caller is mediaserver it is likely that the session being released was acquired 2325 // on behalf of a process not in notification clients and we ignore the warning. 2326 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2327} 2328 2329void AudioFlinger::purgeStaleEffects_l() { 2330 2331 ALOGV("purging stale effects"); 2332 2333 Vector< sp<EffectChain> > chains; 2334 2335 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2336 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2337 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2338 sp<EffectChain> ec = t->mEffectChains[j]; 2339 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2340 chains.push(ec); 2341 } 2342 } 2343 } 2344 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2345 sp<RecordThread> t = mRecordThreads.valueAt(i); 2346 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2347 sp<EffectChain> ec = t->mEffectChains[j]; 2348 chains.push(ec); 2349 } 2350 } 2351 2352 for (size_t i = 0; i < chains.size(); i++) { 2353 sp<EffectChain> ec = chains[i]; 2354 int sessionid = ec->sessionId(); 2355 sp<ThreadBase> t = ec->mThread.promote(); 2356 if (t == 0) { 2357 continue; 2358 } 2359 size_t numsessionrefs = mAudioSessionRefs.size(); 2360 bool found = false; 2361 for (size_t k = 0; k < numsessionrefs; k++) { 2362 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2363 if (ref->mSessionid == sessionid) { 2364 ALOGV(" session %d still exists for %d with %d refs", 2365 sessionid, ref->mPid, ref->mCnt); 2366 found = true; 2367 break; 2368 } 2369 } 2370 if (!found) { 2371 Mutex::Autolock _l(t->mLock); 2372 // remove all effects from the chain 2373 while (ec->mEffects.size()) { 2374 sp<EffectModule> effect = ec->mEffects[0]; 2375 effect->unPin(); 2376 t->removeEffect_l(effect); 2377 if (effect->purgeHandles()) { 2378 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2379 } 2380 AudioSystem::unregisterEffect(effect->id()); 2381 } 2382 } 2383 } 2384 return; 2385} 2386 2387// checkThread_l() must be called with AudioFlinger::mLock held 2388AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2389{ 2390 ThreadBase *thread = NULL; 2391 switch (audio_unique_id_get_use(ioHandle)) { 2392 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2393 thread = checkPlaybackThread_l(ioHandle); 2394 break; 2395 case AUDIO_UNIQUE_ID_USE_INPUT: 2396 thread = checkRecordThread_l(ioHandle); 2397 break; 2398 default: 2399 break; 2400 } 2401 return thread; 2402} 2403 2404// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2405AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2406{ 2407 return mPlaybackThreads.valueFor(output).get(); 2408} 2409 2410// checkMixerThread_l() must be called with AudioFlinger::mLock held 2411AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2412{ 2413 PlaybackThread *thread = checkPlaybackThread_l(output); 2414 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2415} 2416 2417// checkRecordThread_l() must be called with AudioFlinger::mLock held 2418AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2419{ 2420 return mRecordThreads.valueFor(input).get(); 2421} 2422 2423audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2424{ 2425 int32_t base = android_atomic_add(AUDIO_UNIQUE_ID_USE_MAX, &mNextUniqueId); 2426 // We have no way of recovering from wraparound 2427 LOG_ALWAYS_FATAL_IF(base == 0, "unique ID overflow"); 2428 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2429 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2430 return (audio_unique_id_t) (base | use); 2431} 2432 2433AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2434{ 2435 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2436 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2437 if(thread->isDuplicating()) { 2438 continue; 2439 } 2440 AudioStreamOut *output = thread->getOutput(); 2441 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2442 return thread; 2443 } 2444 } 2445 return NULL; 2446} 2447 2448audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2449{ 2450 PlaybackThread *thread = primaryPlaybackThread_l(); 2451 2452 if (thread == NULL) { 2453 return 0; 2454 } 2455 2456 return thread->outDevice(); 2457} 2458 2459sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2460 audio_session_t triggerSession, 2461 audio_session_t listenerSession, 2462 sync_event_callback_t callBack, 2463 wp<RefBase> cookie) 2464{ 2465 Mutex::Autolock _l(mLock); 2466 2467 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2468 status_t playStatus = NAME_NOT_FOUND; 2469 status_t recStatus = NAME_NOT_FOUND; 2470 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2471 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2472 if (playStatus == NO_ERROR) { 2473 return event; 2474 } 2475 } 2476 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2477 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2478 if (recStatus == NO_ERROR) { 2479 return event; 2480 } 2481 } 2482 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2483 mPendingSyncEvents.add(event); 2484 } else { 2485 ALOGV("createSyncEvent() invalid event %d", event->type()); 2486 event.clear(); 2487 } 2488 return event; 2489} 2490 2491// ---------------------------------------------------------------------------- 2492// Effect management 2493// ---------------------------------------------------------------------------- 2494 2495 2496status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2497{ 2498 Mutex::Autolock _l(mLock); 2499 return EffectQueryNumberEffects(numEffects); 2500} 2501 2502status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2503{ 2504 Mutex::Autolock _l(mLock); 2505 return EffectQueryEffect(index, descriptor); 2506} 2507 2508status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2509 effect_descriptor_t *descriptor) const 2510{ 2511 Mutex::Autolock _l(mLock); 2512 return EffectGetDescriptor(pUuid, descriptor); 2513} 2514 2515 2516sp<IEffect> AudioFlinger::createEffect( 2517 effect_descriptor_t *pDesc, 2518 const sp<IEffectClient>& effectClient, 2519 int32_t priority, 2520 audio_io_handle_t io, 2521 audio_session_t sessionId, 2522 const String16& opPackageName, 2523 status_t *status, 2524 int *id, 2525 int *enabled) 2526{ 2527 status_t lStatus = NO_ERROR; 2528 sp<EffectHandle> handle; 2529 effect_descriptor_t desc; 2530 2531 pid_t pid = IPCThreadState::self()->getCallingPid(); 2532 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2533 pid, effectClient.get(), priority, sessionId, io); 2534 2535 if (pDesc == NULL) { 2536 lStatus = BAD_VALUE; 2537 goto Exit; 2538 } 2539 2540 // check audio settings permission for global effects 2541 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2542 lStatus = PERMISSION_DENIED; 2543 goto Exit; 2544 } 2545 2546 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2547 // that can only be created by audio policy manager (running in same process) 2548 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2549 lStatus = PERMISSION_DENIED; 2550 goto Exit; 2551 } 2552 2553 { 2554 if (!EffectIsNullUuid(&pDesc->uuid)) { 2555 // if uuid is specified, request effect descriptor 2556 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2557 if (lStatus < 0) { 2558 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2559 goto Exit; 2560 } 2561 } else { 2562 // if uuid is not specified, look for an available implementation 2563 // of the required type in effect factory 2564 if (EffectIsNullUuid(&pDesc->type)) { 2565 ALOGW("createEffect() no effect type"); 2566 lStatus = BAD_VALUE; 2567 goto Exit; 2568 } 2569 uint32_t numEffects = 0; 2570 effect_descriptor_t d; 2571 d.flags = 0; // prevent compiler warning 2572 bool found = false; 2573 2574 lStatus = EffectQueryNumberEffects(&numEffects); 2575 if (lStatus < 0) { 2576 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2577 goto Exit; 2578 } 2579 for (uint32_t i = 0; i < numEffects; i++) { 2580 lStatus = EffectQueryEffect(i, &desc); 2581 if (lStatus < 0) { 2582 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2583 continue; 2584 } 2585 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2586 // If matching type found save effect descriptor. If the session is 2587 // 0 and the effect is not auxiliary, continue enumeration in case 2588 // an auxiliary version of this effect type is available 2589 found = true; 2590 d = desc; 2591 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2592 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2593 break; 2594 } 2595 } 2596 } 2597 if (!found) { 2598 lStatus = BAD_VALUE; 2599 ALOGW("createEffect() effect not found"); 2600 goto Exit; 2601 } 2602 // For same effect type, chose auxiliary version over insert version if 2603 // connect to output mix (Compliance to OpenSL ES) 2604 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2605 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2606 desc = d; 2607 } 2608 } 2609 2610 // Do not allow auxiliary effects on a session different from 0 (output mix) 2611 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2612 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2613 lStatus = INVALID_OPERATION; 2614 goto Exit; 2615 } 2616 2617 // check recording permission for visualizer 2618 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2619 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2620 lStatus = PERMISSION_DENIED; 2621 goto Exit; 2622 } 2623 2624 // return effect descriptor 2625 *pDesc = desc; 2626 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2627 // if the output returned by getOutputForEffect() is removed before we lock the 2628 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2629 // and we will exit safely 2630 io = AudioSystem::getOutputForEffect(&desc); 2631 ALOGV("createEffect got output %d", io); 2632 } 2633 2634 Mutex::Autolock _l(mLock); 2635 2636 // If output is not specified try to find a matching audio session ID in one of the 2637 // output threads. 2638 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2639 // because of code checking output when entering the function. 2640 // Note: io is never 0 when creating an effect on an input 2641 if (io == AUDIO_IO_HANDLE_NONE) { 2642 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2643 // output must be specified by AudioPolicyManager when using session 2644 // AUDIO_SESSION_OUTPUT_STAGE 2645 lStatus = BAD_VALUE; 2646 goto Exit; 2647 } 2648 // look for the thread where the specified audio session is present 2649 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2650 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2651 io = mPlaybackThreads.keyAt(i); 2652 break; 2653 } 2654 } 2655 if (io == 0) { 2656 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2657 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2658 io = mRecordThreads.keyAt(i); 2659 break; 2660 } 2661 } 2662 } 2663 // If no output thread contains the requested session ID, default to 2664 // first output. The effect chain will be moved to the correct output 2665 // thread when a track with the same session ID is created 2666 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2667 io = mPlaybackThreads.keyAt(0); 2668 } 2669 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2670 } 2671 ThreadBase *thread = checkRecordThread_l(io); 2672 if (thread == NULL) { 2673 thread = checkPlaybackThread_l(io); 2674 if (thread == NULL) { 2675 ALOGE("createEffect() unknown output thread"); 2676 lStatus = BAD_VALUE; 2677 goto Exit; 2678 } 2679 } else { 2680 // Check if one effect chain was awaiting for an effect to be created on this 2681 // session and used it instead of creating a new one. 2682 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2683 if (chain != 0) { 2684 Mutex::Autolock _l(thread->mLock); 2685 thread->addEffectChain_l(chain); 2686 } 2687 } 2688 2689 sp<Client> client = registerPid(pid); 2690 2691 // create effect on selected output thread 2692 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2693 &desc, enabled, &lStatus); 2694 if (handle != 0 && id != NULL) { 2695 *id = handle->id(); 2696 } 2697 if (handle == 0) { 2698 // remove local strong reference to Client with mClientLock held 2699 Mutex::Autolock _cl(mClientLock); 2700 client.clear(); 2701 } 2702 } 2703 2704Exit: 2705 *status = lStatus; 2706 return handle; 2707} 2708 2709status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2710 audio_io_handle_t dstOutput) 2711{ 2712 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2713 sessionId, srcOutput, dstOutput); 2714 Mutex::Autolock _l(mLock); 2715 if (srcOutput == dstOutput) { 2716 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2717 return NO_ERROR; 2718 } 2719 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2720 if (srcThread == NULL) { 2721 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2722 return BAD_VALUE; 2723 } 2724 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2725 if (dstThread == NULL) { 2726 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2727 return BAD_VALUE; 2728 } 2729 2730 Mutex::Autolock _dl(dstThread->mLock); 2731 Mutex::Autolock _sl(srcThread->mLock); 2732 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2733} 2734 2735// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2736status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2737 AudioFlinger::PlaybackThread *srcThread, 2738 AudioFlinger::PlaybackThread *dstThread, 2739 bool reRegister) 2740{ 2741 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2742 sessionId, srcThread, dstThread); 2743 2744 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2745 if (chain == 0) { 2746 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2747 sessionId, srcThread); 2748 return INVALID_OPERATION; 2749 } 2750 2751 // Check whether the destination thread has a channel count of FCC_2, which is 2752 // currently required for (most) effects. Prevent moving the effect chain here rather 2753 // than disabling the addEffect_l() call in dstThread below. 2754 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2755 dstThread->mChannelCount != FCC_2) { 2756 ALOGW("moveEffectChain_l() effect chain failed because" 2757 " destination thread %p channel count(%u) != %u", 2758 dstThread, dstThread->mChannelCount, FCC_2); 2759 return INVALID_OPERATION; 2760 } 2761 2762 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2763 // so that a new chain is created with correct parameters when first effect is added. This is 2764 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2765 // removed. 2766 srcThread->removeEffectChain_l(chain); 2767 2768 // transfer all effects one by one so that new effect chain is created on new thread with 2769 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2770 sp<EffectChain> dstChain; 2771 uint32_t strategy = 0; // prevent compiler warning 2772 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2773 Vector< sp<EffectModule> > removed; 2774 status_t status = NO_ERROR; 2775 while (effect != 0) { 2776 srcThread->removeEffect_l(effect); 2777 removed.add(effect); 2778 status = dstThread->addEffect_l(effect); 2779 if (status != NO_ERROR) { 2780 break; 2781 } 2782 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2783 if (effect->state() == EffectModule::ACTIVE || 2784 effect->state() == EffectModule::STOPPING) { 2785 effect->start(); 2786 } 2787 // if the move request is not received from audio policy manager, the effect must be 2788 // re-registered with the new strategy and output 2789 if (dstChain == 0) { 2790 dstChain = effect->chain().promote(); 2791 if (dstChain == 0) { 2792 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2793 status = NO_INIT; 2794 break; 2795 } 2796 strategy = dstChain->strategy(); 2797 } 2798 if (reRegister) { 2799 AudioSystem::unregisterEffect(effect->id()); 2800 AudioSystem::registerEffect(&effect->desc(), 2801 dstThread->id(), 2802 strategy, 2803 sessionId, 2804 effect->id()); 2805 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2806 } 2807 effect = chain->getEffectFromId_l(0); 2808 } 2809 2810 if (status != NO_ERROR) { 2811 for (size_t i = 0; i < removed.size(); i++) { 2812 srcThread->addEffect_l(removed[i]); 2813 if (dstChain != 0 && reRegister) { 2814 AudioSystem::unregisterEffect(removed[i]->id()); 2815 AudioSystem::registerEffect(&removed[i]->desc(), 2816 srcThread->id(), 2817 strategy, 2818 sessionId, 2819 removed[i]->id()); 2820 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2821 } 2822 } 2823 } 2824 2825 return status; 2826} 2827 2828bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2829{ 2830 if (mGlobalEffectEnableTime != 0 && 2831 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2832 return true; 2833 } 2834 2835 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2836 sp<EffectChain> ec = 2837 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2838 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2839 return true; 2840 } 2841 } 2842 return false; 2843} 2844 2845void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2846{ 2847 Mutex::Autolock _l(mLock); 2848 2849 mGlobalEffectEnableTime = systemTime(); 2850 2851 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2852 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2853 if (t->mType == ThreadBase::OFFLOAD) { 2854 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2855 } 2856 } 2857 2858} 2859 2860status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2861{ 2862 audio_session_t session = chain->sessionId(); 2863 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2864 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2865 if (index >= 0) { 2866 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2867 return ALREADY_EXISTS; 2868 } 2869 mOrphanEffectChains.add(session, chain); 2870 return NO_ERROR; 2871} 2872 2873sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2874{ 2875 sp<EffectChain> chain; 2876 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2877 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2878 if (index >= 0) { 2879 chain = mOrphanEffectChains.valueAt(index); 2880 mOrphanEffectChains.removeItemsAt(index); 2881 } 2882 return chain; 2883} 2884 2885bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2886{ 2887 Mutex::Autolock _l(mLock); 2888 audio_session_t session = effect->sessionId(); 2889 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2890 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2891 if (index >= 0) { 2892 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2893 if (chain->removeEffect_l(effect) == 0) { 2894 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2895 mOrphanEffectChains.removeItemsAt(index); 2896 } 2897 return true; 2898 } 2899 return false; 2900} 2901 2902 2903struct Entry { 2904#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2905 char mFileName[TEE_MAX_FILENAME]; 2906}; 2907 2908int comparEntry(const void *p1, const void *p2) 2909{ 2910 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2911} 2912 2913#ifdef TEE_SINK 2914void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2915{ 2916 NBAIO_Source *teeSource = source.get(); 2917 if (teeSource != NULL) { 2918 // .wav rotation 2919 // There is a benign race condition if 2 threads call this simultaneously. 2920 // They would both traverse the directory, but the result would simply be 2921 // failures at unlink() which are ignored. It's also unlikely since 2922 // normally dumpsys is only done by bugreport or from the command line. 2923 char teePath[32+256]; 2924 strcpy(teePath, "/data/misc/audioserver"); 2925 size_t teePathLen = strlen(teePath); 2926 DIR *dir = opendir(teePath); 2927 teePath[teePathLen++] = '/'; 2928 if (dir != NULL) { 2929#define TEE_MAX_SORT 20 // number of entries to sort 2930#define TEE_MAX_KEEP 10 // number of entries to keep 2931 struct Entry entries[TEE_MAX_SORT]; 2932 size_t entryCount = 0; 2933 while (entryCount < TEE_MAX_SORT) { 2934 struct dirent de; 2935 struct dirent *result = NULL; 2936 int rc = readdir_r(dir, &de, &result); 2937 if (rc != 0) { 2938 ALOGW("readdir_r failed %d", rc); 2939 break; 2940 } 2941 if (result == NULL) { 2942 break; 2943 } 2944 if (result != &de) { 2945 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2946 break; 2947 } 2948 // ignore non .wav file entries 2949 size_t nameLen = strlen(de.d_name); 2950 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2951 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2952 continue; 2953 } 2954 strcpy(entries[entryCount++].mFileName, de.d_name); 2955 } 2956 (void) closedir(dir); 2957 if (entryCount > TEE_MAX_KEEP) { 2958 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2959 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2960 strcpy(&teePath[teePathLen], entries[i].mFileName); 2961 (void) unlink(teePath); 2962 } 2963 } 2964 } else { 2965 if (fd >= 0) { 2966 dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath, 2967 strerror(errno)); 2968 } 2969 } 2970 char teeTime[16]; 2971 struct timeval tv; 2972 gettimeofday(&tv, NULL); 2973 struct tm tm; 2974 localtime_r(&tv.tv_sec, &tm); 2975 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2976 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2977 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2978 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2979 if (teeFd >= 0) { 2980 // FIXME use libsndfile 2981 char wavHeader[44]; 2982 memcpy(wavHeader, 2983 "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", 2984 sizeof(wavHeader)); 2985 NBAIO_Format format = teeSource->format(); 2986 unsigned channelCount = Format_channelCount(format); 2987 uint32_t sampleRate = Format_sampleRate(format); 2988 size_t frameSize = Format_frameSize(format); 2989 wavHeader[22] = channelCount; // number of channels 2990 wavHeader[24] = sampleRate; // sample rate 2991 wavHeader[25] = sampleRate >> 8; 2992 wavHeader[32] = frameSize; // block alignment 2993 wavHeader[33] = frameSize >> 8; 2994 write(teeFd, wavHeader, sizeof(wavHeader)); 2995 size_t total = 0; 2996 bool firstRead = true; 2997#define TEE_SINK_READ 1024 // frames per I/O operation 2998 void *buffer = malloc(TEE_SINK_READ * frameSize); 2999 for (;;) { 3000 size_t count = TEE_SINK_READ; 3001 ssize_t actual = teeSource->read(buffer, count); 3002 bool wasFirstRead = firstRead; 3003 firstRead = false; 3004 if (actual <= 0) { 3005 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3006 continue; 3007 } 3008 break; 3009 } 3010 ALOG_ASSERT(actual <= (ssize_t)count); 3011 write(teeFd, buffer, actual * frameSize); 3012 total += actual; 3013 } 3014 free(buffer); 3015 lseek(teeFd, (off_t) 4, SEEK_SET); 3016 uint32_t temp = 44 + total * frameSize - 8; 3017 // FIXME not big-endian safe 3018 write(teeFd, &temp, sizeof(temp)); 3019 lseek(teeFd, (off_t) 40, SEEK_SET); 3020 temp = total * frameSize; 3021 // FIXME not big-endian safe 3022 write(teeFd, &temp, sizeof(temp)); 3023 close(teeFd); 3024 if (fd >= 0) { 3025 dprintf(fd, "tee copied to %s\n", teePath); 3026 } 3027 } else { 3028 if (fd >= 0) { 3029 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3030 } 3031 } 3032 } 3033} 3034#endif 3035 3036// ---------------------------------------------------------------------------- 3037 3038status_t AudioFlinger::onTransact( 3039 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3040{ 3041 return BnAudioFlinger::onTransact(code, data, reply, flags); 3042} 3043 3044} // namespace android 3045