AudioFlinger.cpp revision 36d0ca16024820df9a12903d2ac443fabcc180bc
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 <common_time/cc_helper.h> 60 61#include <media/IMediaLogService.h> 62 63#include <media/nbaio/Pipe.h> 64#include <media/nbaio/PipeReader.h> 65#include <media/AudioParameter.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (format & AUDIO_FORMAT_MAIN_MASK) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 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(1), 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 char value[PROPERTY_VALUE_MAX]; 188 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 189 if (doLog) { 190 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 191 MemoryHeapBase::READ_ONLY); 192 } 193 194#ifdef TEE_SINK 195 (void) property_get("ro.debuggable", value, "0"); 196 int debuggable = atoi(value); 197 int teeEnabled = 0; 198 if (debuggable) { 199 (void) property_get("af.tee", value, "0"); 200 teeEnabled = atoi(value); 201 } 202 // FIXME symbolic constants here 203 if (teeEnabled & 1) { 204 mTeeSinkInputEnabled = true; 205 } 206 if (teeEnabled & 2) { 207 mTeeSinkOutputEnabled = true; 208 } 209 if (teeEnabled & 4) { 210 mTeeSinkTrackEnabled = true; 211 } 212#endif 213} 214 215void AudioFlinger::onFirstRef() 216{ 217 int rc = 0; 218 219 Mutex::Autolock _l(mLock); 220 221 /* TODO: move all this work into an Init() function */ 222 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 223 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 224 uint32_t int_val; 225 if (1 == sscanf(val_str, "%u", &int_val)) { 226 mStandbyTimeInNsecs = milliseconds(int_val); 227 ALOGI("Using %u mSec as standby time.", int_val); 228 } else { 229 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 230 ALOGI("Using default %u mSec as standby time.", 231 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 232 } 233 } 234 235 mPatchPanel = new PatchPanel(this); 236 237 mMode = AUDIO_MODE_NORMAL; 238} 239 240AudioFlinger::~AudioFlinger() 241{ 242 while (!mRecordThreads.isEmpty()) { 243 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 244 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 245 } 246 while (!mPlaybackThreads.isEmpty()) { 247 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 248 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 249 } 250 251 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 252 // no mHardwareLock needed, as there are no other references to this 253 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 254 delete mAudioHwDevs.valueAt(i); 255 } 256 257 // Tell media.log service about any old writers that still need to be unregistered 258 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 259 if (binder != 0) { 260 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 261 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 262 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 263 mUnregisteredWriters.pop(); 264 mediaLogService->unregisterWriter(iMemory); 265 } 266 } 267 268} 269 270static const char * const audio_interfaces[] = { 271 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 272 AUDIO_HARDWARE_MODULE_ID_A2DP, 273 AUDIO_HARDWARE_MODULE_ID_USB, 274}; 275#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 276 277AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 278 audio_module_handle_t module, 279 audio_devices_t devices) 280{ 281 // if module is 0, the request comes from an old policy manager and we should load 282 // well known modules 283 if (module == 0) { 284 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 285 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 286 loadHwModule_l(audio_interfaces[i]); 287 } 288 // then try to find a module supporting the requested device. 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 291 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 292 if ((dev->get_supported_devices != NULL) && 293 (dev->get_supported_devices(dev) & devices) == devices) 294 return audioHwDevice; 295 } 296 } else { 297 // check a match for the requested module handle 298 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 299 if (audioHwDevice != NULL) { 300 return audioHwDevice; 301 } 302 } 303 304 return NULL; 305} 306 307void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Notification Clients:\n"); 323 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 324 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 325 result.append(buffer); 326 } 327 328 result.append("Global session refs:\n"); 329 result.append(" session pid count\n"); 330 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 331 AudioSessionRef *r = mAudioSessionRefs[i]; 332 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 333 result.append(buffer); 334 } 335 write(fd, result.string(), result.size()); 336} 337 338 339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 340{ 341 const size_t SIZE = 256; 342 char buffer[SIZE]; 343 String8 result; 344 hardware_call_state hardwareStatus = mHardwareStatus; 345 346 snprintf(buffer, SIZE, "Hardware status: %d\n" 347 "Standby Time mSec: %u\n", 348 hardwareStatus, 349 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 355{ 356 const size_t SIZE = 256; 357 char buffer[SIZE]; 358 String8 result; 359 snprintf(buffer, SIZE, "Permission Denial: " 360 "can't dump AudioFlinger from pid=%d, uid=%d\n", 361 IPCThreadState::self()->getCallingPid(), 362 IPCThreadState::self()->getCallingUid()); 363 result.append(buffer); 364 write(fd, result.string(), result.size()); 365} 366 367bool AudioFlinger::dumpTryLock(Mutex& mutex) 368{ 369 bool locked = false; 370 for (int i = 0; i < kDumpLockRetries; ++i) { 371 if (mutex.tryLock() == NO_ERROR) { 372 locked = true; 373 break; 374 } 375 usleep(kDumpLockSleepUs); 376 } 377 return locked; 378} 379 380status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 381{ 382 if (!dumpAllowed()) { 383 dumpPermissionDenial(fd, args); 384 } else { 385 // get state of hardware lock 386 bool hardwareLocked = dumpTryLock(mHardwareLock); 387 if (!hardwareLocked) { 388 String8 result(kHardwareLockedString); 389 write(fd, result.string(), result.size()); 390 } else { 391 mHardwareLock.unlock(); 392 } 393 394 bool locked = dumpTryLock(mLock); 395 396 // failed to lock - AudioFlinger is probably deadlocked 397 if (!locked) { 398 String8 result(kDeadlockedString); 399 write(fd, result.string(), result.size()); 400 } 401 402 bool clientLocked = dumpTryLock(mClientLock); 403 if (!clientLocked) { 404 String8 result(kClientLockedString); 405 write(fd, result.string(), result.size()); 406 } 407 408 EffectDumpEffects(fd); 409 410 dumpClients(fd, args); 411 if (clientLocked) { 412 mClientLock.unlock(); 413 } 414 415 dumpInternals(fd, args); 416 417 // dump playback threads 418 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 419 mPlaybackThreads.valueAt(i)->dump(fd, args); 420 } 421 422 // dump record threads 423 for (size_t i = 0; i < mRecordThreads.size(); i++) { 424 mRecordThreads.valueAt(i)->dump(fd, args); 425 } 426 427 // dump orphan effect chains 428 if (mOrphanEffectChains.size() != 0) { 429 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 430 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 431 mOrphanEffectChains.valueAt(i)->dump(fd, args); 432 } 433 } 434 // dump all hardware devs 435 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 436 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 437 dev->dump(dev, fd); 438 } 439 440#ifdef TEE_SINK 441 // dump the serially shared record tee sink 442 if (mRecordTeeSource != 0) { 443 dumpTee(fd, mRecordTeeSource); 444 } 445#endif 446 447 if (locked) { 448 mLock.unlock(); 449 } 450 451 // append a copy of media.log here by forwarding fd to it, but don't attempt 452 // to lookup the service if it's not running, as it will block for a second 453 if (mLogMemoryDealer != 0) { 454 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 455 if (binder != 0) { 456 dprintf(fd, "\nmedia.log:\n"); 457 Vector<String16> args; 458 binder->dump(fd, args); 459 } 460 } 461 } 462 return NO_ERROR; 463} 464 465sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 466{ 467 Mutex::Autolock _cl(mClientLock); 468 // If pid is already in the mClients wp<> map, then use that entry 469 // (for which promote() is always != 0), otherwise create a new entry and Client. 470 sp<Client> client = mClients.valueFor(pid).promote(); 471 if (client == 0) { 472 client = new Client(this, pid); 473 mClients.add(pid, client); 474 } 475 476 return client; 477} 478 479sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 480{ 481 // If there is no memory allocated for logs, return a dummy writer that does nothing 482 if (mLogMemoryDealer == 0) { 483 return new NBLog::Writer(); 484 } 485 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 486 // Similarly if we can't contact the media.log service, also return a dummy writer 487 if (binder == 0) { 488 return new NBLog::Writer(); 489 } 490 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 491 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 492 // If allocation fails, consult the vector of previously unregistered writers 493 // and garbage-collect one or more them until an allocation succeeds 494 if (shared == 0) { 495 Mutex::Autolock _l(mUnregisteredWritersLock); 496 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 497 { 498 // Pick the oldest stale writer to garbage-collect 499 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 500 mUnregisteredWriters.removeAt(0); 501 mediaLogService->unregisterWriter(iMemory); 502 // Now the media.log remote reference to IMemory is gone. When our last local 503 // reference to IMemory also drops to zero at end of this block, 504 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 505 } 506 // Re-attempt the allocation 507 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 508 if (shared != 0) { 509 goto success; 510 } 511 } 512 // Even after garbage-collecting all old writers, there is still not enough memory, 513 // so return a dummy writer 514 return new NBLog::Writer(); 515 } 516success: 517 mediaLogService->registerWriter(shared, size, name); 518 return new NBLog::Writer(size, shared); 519} 520 521void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 522{ 523 if (writer == 0) { 524 return; 525 } 526 sp<IMemory> iMemory(writer->getIMemory()); 527 if (iMemory == 0) { 528 return; 529 } 530 // Rather than removing the writer immediately, append it to a queue of old writers to 531 // be garbage-collected later. This allows us to continue to view old logs for a while. 532 Mutex::Autolock _l(mUnregisteredWritersLock); 533 mUnregisteredWriters.push(writer); 534} 535 536// IAudioFlinger interface 537 538 539sp<IAudioTrack> AudioFlinger::createTrack( 540 audio_stream_type_t streamType, 541 uint32_t sampleRate, 542 audio_format_t format, 543 audio_channel_mask_t channelMask, 544 size_t *frameCount, 545 IAudioFlinger::track_flags_t *flags, 546 const sp<IMemory>& sharedBuffer, 547 audio_io_handle_t output, 548 pid_t tid, 549 int *sessionId, 550 int clientUid, 551 status_t *status) 552{ 553 sp<PlaybackThread::Track> track; 554 sp<TrackHandle> trackHandle; 555 sp<Client> client; 556 status_t lStatus; 557 int lSessionId; 558 559 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 560 // but if someone uses binder directly they could bypass that and cause us to crash 561 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 562 ALOGE("createTrack() invalid stream type %d", streamType); 563 lStatus = BAD_VALUE; 564 goto Exit; 565 } 566 567 // further sample rate checks are performed by createTrack_l() depending on the thread type 568 if (sampleRate == 0) { 569 ALOGE("createTrack() invalid sample rate %u", sampleRate); 570 lStatus = BAD_VALUE; 571 goto Exit; 572 } 573 574 // further channel mask checks are performed by createTrack_l() depending on the thread type 575 if (!audio_is_output_channel(channelMask)) { 576 ALOGE("createTrack() invalid channel mask %#x", channelMask); 577 lStatus = BAD_VALUE; 578 goto Exit; 579 } 580 581 // further format checks are performed by createTrack_l() depending on the thread type 582 if (!audio_is_valid_format(format)) { 583 ALOGE("createTrack() invalid format %#x", format); 584 lStatus = BAD_VALUE; 585 goto Exit; 586 } 587 588 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 589 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 { 595 Mutex::Autolock _l(mLock); 596 PlaybackThread *thread = checkPlaybackThread_l(output); 597 if (thread == NULL) { 598 ALOGE("no playback thread found for output handle %d", output); 599 lStatus = BAD_VALUE; 600 goto Exit; 601 } 602 603 pid_t pid = IPCThreadState::self()->getCallingPid(); 604 client = registerPid(pid); 605 606 PlaybackThread *effectThread = NULL; 607 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 608 lSessionId = *sessionId; 609 // check if an effect chain with the same session ID is present on another 610 // output thread and move it here. 611 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 612 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 613 if (mPlaybackThreads.keyAt(i) != output) { 614 uint32_t sessions = t->hasAudioSession(lSessionId); 615 if (sessions & PlaybackThread::EFFECT_SESSION) { 616 effectThread = t.get(); 617 break; 618 } 619 } 620 } 621 } else { 622 // if no audio session id is provided, create one here 623 lSessionId = nextUniqueId(); 624 if (sessionId != NULL) { 625 *sessionId = lSessionId; 626 } 627 } 628 ALOGV("createTrack() lSessionId: %d", lSessionId); 629 630 track = thread->createTrack_l(client, streamType, sampleRate, format, 631 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 632 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 633 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 634 635 // move effect chain to this output thread if an effect on same session was waiting 636 // for a track to be created 637 if (lStatus == NO_ERROR && effectThread != NULL) { 638 // no risk of deadlock because AudioFlinger::mLock is held 639 Mutex::Autolock _dl(thread->mLock); 640 Mutex::Autolock _sl(effectThread->mLock); 641 moveEffectChain_l(lSessionId, effectThread, thread, true); 642 } 643 644 // Look for sync events awaiting for a session to be used. 645 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 646 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 647 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 648 if (lStatus == NO_ERROR) { 649 (void) track->setSyncEvent(mPendingSyncEvents[i]); 650 } else { 651 mPendingSyncEvents[i]->cancel(); 652 } 653 mPendingSyncEvents.removeAt(i); 654 i--; 655 } 656 } 657 } 658 659 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 660 } 661 662 if (lStatus != NO_ERROR) { 663 // remove local strong reference to Client before deleting the Track so that the 664 // Client destructor is called by the TrackBase destructor with mClientLock held 665 // Don't hold mClientLock when releasing the reference on the track as the 666 // destructor will acquire it. 667 { 668 Mutex::Autolock _cl(mClientLock); 669 client.clear(); 670 } 671 track.clear(); 672 goto Exit; 673 } 674 675 // return handle to client 676 trackHandle = new TrackHandle(track); 677 678Exit: 679 *status = lStatus; 680 return trackHandle; 681} 682 683uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 684{ 685 Mutex::Autolock _l(mLock); 686 PlaybackThread *thread = checkPlaybackThread_l(output); 687 if (thread == NULL) { 688 ALOGW("sampleRate() unknown thread %d", output); 689 return 0; 690 } 691 return thread->sampleRate(); 692} 693 694audio_format_t AudioFlinger::format(audio_io_handle_t output) const 695{ 696 Mutex::Autolock _l(mLock); 697 PlaybackThread *thread = checkPlaybackThread_l(output); 698 if (thread == NULL) { 699 ALOGW("format() unknown thread %d", output); 700 return AUDIO_FORMAT_INVALID; 701 } 702 return thread->format(); 703} 704 705size_t AudioFlinger::frameCount(audio_io_handle_t output) const 706{ 707 Mutex::Autolock _l(mLock); 708 PlaybackThread *thread = checkPlaybackThread_l(output); 709 if (thread == NULL) { 710 ALOGW("frameCount() unknown thread %d", output); 711 return 0; 712 } 713 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 714 // should examine all callers and fix them to handle smaller counts 715 return thread->frameCount(); 716} 717 718uint32_t AudioFlinger::latency(audio_io_handle_t output) const 719{ 720 Mutex::Autolock _l(mLock); 721 PlaybackThread *thread = checkPlaybackThread_l(output); 722 if (thread == NULL) { 723 ALOGW("latency(): no playback thread found for output handle %d", output); 724 return 0; 725 } 726 return thread->latency(); 727} 728 729status_t AudioFlinger::setMasterVolume(float value) 730{ 731 status_t ret = initCheck(); 732 if (ret != NO_ERROR) { 733 return ret; 734 } 735 736 // check calling permissions 737 if (!settingsAllowed()) { 738 return PERMISSION_DENIED; 739 } 740 741 Mutex::Autolock _l(mLock); 742 mMasterVolume = value; 743 744 // Set master volume in the HALs which support it. 745 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 746 AutoMutex lock(mHardwareLock); 747 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 748 749 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 750 if (dev->canSetMasterVolume()) { 751 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 752 } 753 mHardwareStatus = AUDIO_HW_IDLE; 754 } 755 756 // Now set the master volume in each playback thread. Playback threads 757 // assigned to HALs which do not have master volume support will apply 758 // master volume during the mix operation. Threads with HALs which do 759 // support master volume will simply ignore the setting. 760 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 761 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 762 continue; 763 } 764 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 765 } 766 767 return NO_ERROR; 768} 769 770status_t AudioFlinger::setMode(audio_mode_t mode) 771{ 772 status_t ret = initCheck(); 773 if (ret != NO_ERROR) { 774 return ret; 775 } 776 777 // check calling permissions 778 if (!settingsAllowed()) { 779 return PERMISSION_DENIED; 780 } 781 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 782 ALOGW("Illegal value: setMode(%d)", mode); 783 return BAD_VALUE; 784 } 785 786 { // scope for the lock 787 AutoMutex lock(mHardwareLock); 788 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 789 mHardwareStatus = AUDIO_HW_SET_MODE; 790 ret = dev->set_mode(dev, mode); 791 mHardwareStatus = AUDIO_HW_IDLE; 792 } 793 794 if (NO_ERROR == ret) { 795 Mutex::Autolock _l(mLock); 796 mMode = mode; 797 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 798 mPlaybackThreads.valueAt(i)->setMode(mode); 799 } 800 801 return ret; 802} 803 804status_t AudioFlinger::setMicMute(bool state) 805{ 806 status_t ret = initCheck(); 807 if (ret != NO_ERROR) { 808 return ret; 809 } 810 811 // check calling permissions 812 if (!settingsAllowed()) { 813 return PERMISSION_DENIED; 814 } 815 816 AutoMutex lock(mHardwareLock); 817 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 818 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 819 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 820 status_t result = dev->set_mic_mute(dev, state); 821 if (result != NO_ERROR) { 822 ret = result; 823 } 824 } 825 mHardwareStatus = AUDIO_HW_IDLE; 826 return ret; 827} 828 829bool AudioFlinger::getMicMute() const 830{ 831 status_t ret = initCheck(); 832 if (ret != NO_ERROR) { 833 return false; 834 } 835 bool mute = true; 836 bool state = AUDIO_MODE_INVALID; 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 841 status_t result = dev->get_mic_mute(dev, &state); 842 if (result == NO_ERROR) { 843 mute = mute && state; 844 } 845 } 846 mHardwareStatus = AUDIO_HW_IDLE; 847 848 return mute; 849} 850 851status_t AudioFlinger::setMasterMute(bool muted) 852{ 853 status_t ret = initCheck(); 854 if (ret != NO_ERROR) { 855 return ret; 856 } 857 858 // check calling permissions 859 if (!settingsAllowed()) { 860 return PERMISSION_DENIED; 861 } 862 863 Mutex::Autolock _l(mLock); 864 mMasterMute = muted; 865 866 // Set master mute in the HALs which support it. 867 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 868 AutoMutex lock(mHardwareLock); 869 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 870 871 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 872 if (dev->canSetMasterMute()) { 873 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 874 } 875 mHardwareStatus = AUDIO_HW_IDLE; 876 } 877 878 // Now set the master mute in each playback thread. Playback threads 879 // assigned to HALs which do not have master mute support will apply master 880 // mute during the mix operation. Threads with HALs which do support master 881 // mute will simply ignore the setting. 882 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 883 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 884 continue; 885 } 886 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 887 } 888 889 return NO_ERROR; 890} 891 892float AudioFlinger::masterVolume() const 893{ 894 Mutex::Autolock _l(mLock); 895 return masterVolume_l(); 896} 897 898bool AudioFlinger::masterMute() const 899{ 900 Mutex::Autolock _l(mLock); 901 return masterMute_l(); 902} 903 904float AudioFlinger::masterVolume_l() const 905{ 906 return mMasterVolume; 907} 908 909bool AudioFlinger::masterMute_l() const 910{ 911 return mMasterMute; 912} 913 914status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 915{ 916 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 917 ALOGW("setStreamVolume() invalid stream %d", stream); 918 return BAD_VALUE; 919 } 920 pid_t caller = IPCThreadState::self()->getCallingPid(); 921 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 922 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 923 return PERMISSION_DENIED; 924 } 925 926 return NO_ERROR; 927} 928 929status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 930 audio_io_handle_t output) 931{ 932 // check calling permissions 933 if (!settingsAllowed()) { 934 return PERMISSION_DENIED; 935 } 936 937 status_t status = checkStreamType(stream); 938 if (status != NO_ERROR) { 939 return status; 940 } 941 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 942 943 AutoMutex lock(mLock); 944 PlaybackThread *thread = NULL; 945 if (output != AUDIO_IO_HANDLE_NONE) { 946 thread = checkPlaybackThread_l(output); 947 if (thread == NULL) { 948 return BAD_VALUE; 949 } 950 } 951 952 mStreamTypes[stream].volume = value; 953 954 if (thread == NULL) { 955 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 956 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 957 } 958 } else { 959 thread->setStreamVolume(stream, value); 960 } 961 962 return NO_ERROR; 963} 964 965status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 966{ 967 // check calling permissions 968 if (!settingsAllowed()) { 969 return PERMISSION_DENIED; 970 } 971 972 status_t status = checkStreamType(stream); 973 if (status != NO_ERROR) { 974 return status; 975 } 976 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 977 978 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 979 ALOGE("setStreamMute() invalid stream %d", stream); 980 return BAD_VALUE; 981 } 982 983 AutoMutex lock(mLock); 984 mStreamTypes[stream].mute = muted; 985 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 986 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 987 988 return NO_ERROR; 989} 990 991float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 992{ 993 status_t status = checkStreamType(stream); 994 if (status != NO_ERROR) { 995 return 0.0f; 996 } 997 998 AutoMutex lock(mLock); 999 float volume; 1000 if (output != AUDIO_IO_HANDLE_NONE) { 1001 PlaybackThread *thread = checkPlaybackThread_l(output); 1002 if (thread == NULL) { 1003 return 0.0f; 1004 } 1005 volume = thread->streamVolume(stream); 1006 } else { 1007 volume = streamVolume_l(stream); 1008 } 1009 1010 return volume; 1011} 1012 1013bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1014{ 1015 status_t status = checkStreamType(stream); 1016 if (status != NO_ERROR) { 1017 return true; 1018 } 1019 1020 AutoMutex lock(mLock); 1021 return streamMute_l(stream); 1022} 1023 1024 1025void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1026{ 1027 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1028 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1029 } 1030} 1031 1032status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1033{ 1034 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1035 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1036 1037 // check calling permissions 1038 if (!settingsAllowed()) { 1039 return PERMISSION_DENIED; 1040 } 1041 1042 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1043 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1044 Mutex::Autolock _l(mLock); 1045 status_t final_result = NO_ERROR; 1046 { 1047 AutoMutex lock(mHardwareLock); 1048 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1049 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1050 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1051 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1052 final_result = result ?: final_result; 1053 } 1054 mHardwareStatus = AUDIO_HW_IDLE; 1055 } 1056 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1057 AudioParameter param = AudioParameter(keyValuePairs); 1058 String8 value; 1059 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1060 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1061 if (mBtNrecIsOff != btNrecIsOff) { 1062 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1063 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1064 audio_devices_t device = thread->inDevice(); 1065 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1066 // collect all of the thread's session IDs 1067 KeyedVector<int, bool> ids = thread->sessionIds(); 1068 // suspend effects associated with those session IDs 1069 for (size_t j = 0; j < ids.size(); ++j) { 1070 int sessionId = ids.keyAt(j); 1071 thread->setEffectSuspended(FX_IID_AEC, 1072 suspend, 1073 sessionId); 1074 thread->setEffectSuspended(FX_IID_NS, 1075 suspend, 1076 sessionId); 1077 } 1078 } 1079 mBtNrecIsOff = btNrecIsOff; 1080 } 1081 } 1082 String8 screenState; 1083 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1084 bool isOff = screenState == "off"; 1085 if (isOff != (AudioFlinger::mScreenState & 1)) { 1086 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1087 } 1088 } 1089 return final_result; 1090 } 1091 1092 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1093 // and the thread is exited once the lock is released 1094 sp<ThreadBase> thread; 1095 { 1096 Mutex::Autolock _l(mLock); 1097 thread = checkPlaybackThread_l(ioHandle); 1098 if (thread == 0) { 1099 thread = checkRecordThread_l(ioHandle); 1100 } else if (thread == primaryPlaybackThread_l()) { 1101 // indicate output device change to all input threads for pre processing 1102 AudioParameter param = AudioParameter(keyValuePairs); 1103 int value; 1104 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1105 (value != 0)) { 1106 broacastParametersToRecordThreads_l(keyValuePairs); 1107 } 1108 } 1109 } 1110 if (thread != 0) { 1111 return thread->setParameters(keyValuePairs); 1112 } 1113 return BAD_VALUE; 1114} 1115 1116String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1117{ 1118 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1119 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1120 1121 Mutex::Autolock _l(mLock); 1122 1123 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1124 String8 out_s8; 1125 1126 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1127 char *s; 1128 { 1129 AutoMutex lock(mHardwareLock); 1130 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1131 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1132 s = dev->get_parameters(dev, keys.string()); 1133 mHardwareStatus = AUDIO_HW_IDLE; 1134 } 1135 out_s8 += String8(s ? s : ""); 1136 free(s); 1137 } 1138 return out_s8; 1139 } 1140 1141 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1142 if (playbackThread != NULL) { 1143 return playbackThread->getParameters(keys); 1144 } 1145 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1146 if (recordThread != NULL) { 1147 return recordThread->getParameters(keys); 1148 } 1149 return String8(""); 1150} 1151 1152size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1153 audio_channel_mask_t channelMask) const 1154{ 1155 status_t ret = initCheck(); 1156 if (ret != NO_ERROR) { 1157 return 0; 1158 } 1159 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1160 return 0; 1161 } 1162 1163 AutoMutex lock(mHardwareLock); 1164 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1165 audio_config_t config, proposed; 1166 memset(&proposed, 0, sizeof(proposed)); 1167 proposed.sample_rate = sampleRate; 1168 proposed.channel_mask = channelMask; 1169 proposed.format = format; 1170 1171 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1172 size_t frames; 1173 for (;;) { 1174 // Note: config is currently a const parameter for get_input_buffer_size() 1175 // but we use a copy from proposed in case config changes from the call. 1176 config = proposed; 1177 frames = dev->get_input_buffer_size(dev, &config); 1178 if (frames != 0) { 1179 break; // hal success, config is the result 1180 } 1181 // change one parameter of the configuration each iteration to a more "common" value 1182 // to see if the device will support it. 1183 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1184 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1185 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1186 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1187 } else { 1188 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1189 "format %#x, channelMask 0x%X", 1190 sampleRate, format, channelMask); 1191 break; // retries failed, break out of loop with frames == 0. 1192 } 1193 } 1194 mHardwareStatus = AUDIO_HW_IDLE; 1195 if (frames > 0 && config.sample_rate != sampleRate) { 1196 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1197 } 1198 return frames; // may be converted to bytes at the Java level. 1199} 1200 1201uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1202{ 1203 Mutex::Autolock _l(mLock); 1204 1205 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1206 if (recordThread != NULL) { 1207 return recordThread->getInputFramesLost(); 1208 } 1209 return 0; 1210} 1211 1212status_t AudioFlinger::setVoiceVolume(float value) 1213{ 1214 status_t ret = initCheck(); 1215 if (ret != NO_ERROR) { 1216 return ret; 1217 } 1218 1219 // check calling permissions 1220 if (!settingsAllowed()) { 1221 return PERMISSION_DENIED; 1222 } 1223 1224 AutoMutex lock(mHardwareLock); 1225 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1226 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1227 ret = dev->set_voice_volume(dev, value); 1228 mHardwareStatus = AUDIO_HW_IDLE; 1229 1230 return ret; 1231} 1232 1233status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1234 audio_io_handle_t output) const 1235{ 1236 status_t status; 1237 1238 Mutex::Autolock _l(mLock); 1239 1240 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1241 if (playbackThread != NULL) { 1242 return playbackThread->getRenderPosition(halFrames, dspFrames); 1243 } 1244 1245 return BAD_VALUE; 1246} 1247 1248void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1249{ 1250 Mutex::Autolock _l(mLock); 1251 if (client == 0) { 1252 return; 1253 } 1254 pid_t pid = IPCThreadState::self()->getCallingPid(); 1255 { 1256 Mutex::Autolock _cl(mClientLock); 1257 if (mNotificationClients.indexOfKey(pid) < 0) { 1258 sp<NotificationClient> notificationClient = new NotificationClient(this, 1259 client, 1260 pid); 1261 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1262 1263 mNotificationClients.add(pid, notificationClient); 1264 1265 sp<IBinder> binder = IInterface::asBinder(client); 1266 binder->linkToDeath(notificationClient); 1267 } 1268 } 1269 1270 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1271 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1272 // the config change is always sent from playback or record threads to avoid deadlock 1273 // with AudioSystem::gLock 1274 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1275 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1276 } 1277 1278 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1279 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1280 } 1281} 1282 1283void AudioFlinger::removeNotificationClient(pid_t pid) 1284{ 1285 Mutex::Autolock _l(mLock); 1286 { 1287 Mutex::Autolock _cl(mClientLock); 1288 mNotificationClients.removeItem(pid); 1289 } 1290 1291 ALOGV("%d died, releasing its sessions", pid); 1292 size_t num = mAudioSessionRefs.size(); 1293 bool removed = false; 1294 for (size_t i = 0; i< num; ) { 1295 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1296 ALOGV(" pid %d @ %d", ref->mPid, i); 1297 if (ref->mPid == pid) { 1298 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1299 mAudioSessionRefs.removeAt(i); 1300 delete ref; 1301 removed = true; 1302 num--; 1303 } else { 1304 i++; 1305 } 1306 } 1307 if (removed) { 1308 purgeStaleEffects_l(); 1309 } 1310} 1311 1312void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1313 const sp<AudioIoDescriptor>& ioDesc, 1314 pid_t pid) 1315{ 1316 Mutex::Autolock _l(mClientLock); 1317 size_t size = mNotificationClients.size(); 1318 for (size_t i = 0; i < size; i++) { 1319 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1320 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1321 } 1322 } 1323} 1324 1325// removeClient_l() must be called with AudioFlinger::mClientLock held 1326void AudioFlinger::removeClient_l(pid_t pid) 1327{ 1328 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1329 IPCThreadState::self()->getCallingPid()); 1330 mClients.removeItem(pid); 1331} 1332 1333// getEffectThread_l() must be called with AudioFlinger::mLock held 1334sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1335{ 1336 sp<PlaybackThread> thread; 1337 1338 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1339 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1340 ALOG_ASSERT(thread == 0); 1341 thread = mPlaybackThreads.valueAt(i); 1342 } 1343 } 1344 1345 return thread; 1346} 1347 1348 1349 1350// ---------------------------------------------------------------------------- 1351 1352AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1353 : RefBase(), 1354 mAudioFlinger(audioFlinger), 1355 mPid(pid), 1356 mTimedTrackCount(0) 1357{ 1358 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1359 heapSize *= 1024; 1360 if (!heapSize) { 1361 heapSize = kClientSharedHeapSizeBytes; 1362 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1363 // invalidated tracks 1364 if (!audioFlinger->isLowRamDevice()) { 1365 heapSize *= kClientSharedHeapSizeMultiplier; 1366 } 1367 } 1368 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1369} 1370 1371// Client destructor must be called with AudioFlinger::mClientLock held 1372AudioFlinger::Client::~Client() 1373{ 1374 mAudioFlinger->removeClient_l(mPid); 1375} 1376 1377sp<MemoryDealer> AudioFlinger::Client::heap() const 1378{ 1379 return mMemoryDealer; 1380} 1381 1382// Reserve one of the limited slots for a timed audio track associated 1383// with this client 1384bool AudioFlinger::Client::reserveTimedTrack() 1385{ 1386 const int kMaxTimedTracksPerClient = 4; 1387 1388 Mutex::Autolock _l(mTimedTrackLock); 1389 1390 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1391 ALOGW("can not create timed track - pid %d has exceeded the limit", 1392 mPid); 1393 return false; 1394 } 1395 1396 mTimedTrackCount++; 1397 return true; 1398} 1399 1400// Release a slot for a timed audio track 1401void AudioFlinger::Client::releaseTimedTrack() 1402{ 1403 Mutex::Autolock _l(mTimedTrackLock); 1404 mTimedTrackCount--; 1405} 1406 1407// ---------------------------------------------------------------------------- 1408 1409AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1410 const sp<IAudioFlingerClient>& client, 1411 pid_t pid) 1412 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1413{ 1414} 1415 1416AudioFlinger::NotificationClient::~NotificationClient() 1417{ 1418} 1419 1420void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1421{ 1422 sp<NotificationClient> keep(this); 1423 mAudioFlinger->removeNotificationClient(mPid); 1424} 1425 1426 1427// ---------------------------------------------------------------------------- 1428 1429static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1430 return audio_is_remote_submix_device(inDevice); 1431} 1432 1433sp<IAudioRecord> AudioFlinger::openRecord( 1434 audio_io_handle_t input, 1435 uint32_t sampleRate, 1436 audio_format_t format, 1437 audio_channel_mask_t channelMask, 1438 const String16& opPackageName, 1439 size_t *frameCount, 1440 IAudioFlinger::track_flags_t *flags, 1441 pid_t tid, 1442 int clientUid, 1443 int *sessionId, 1444 size_t *notificationFrames, 1445 sp<IMemory>& cblk, 1446 sp<IMemory>& buffers, 1447 status_t *status) 1448{ 1449 sp<RecordThread::RecordTrack> recordTrack; 1450 sp<RecordHandle> recordHandle; 1451 sp<Client> client; 1452 status_t lStatus; 1453 int lSessionId; 1454 1455 cblk.clear(); 1456 buffers.clear(); 1457 1458 // check calling permissions 1459 if (!recordingAllowed(opPackageName)) { 1460 ALOGE("openRecord() permission denied: recording not allowed"); 1461 lStatus = PERMISSION_DENIED; 1462 goto Exit; 1463 } 1464 1465 // further sample rate checks are performed by createRecordTrack_l() 1466 if (sampleRate == 0) { 1467 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1468 lStatus = BAD_VALUE; 1469 goto Exit; 1470 } 1471 1472 // we don't yet support anything other than linear PCM 1473 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1474 ALOGE("openRecord() invalid format %#x", format); 1475 lStatus = BAD_VALUE; 1476 goto Exit; 1477 } 1478 1479 // further channel mask checks are performed by createRecordTrack_l() 1480 if (!audio_is_input_channel(channelMask)) { 1481 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1482 lStatus = BAD_VALUE; 1483 goto Exit; 1484 } 1485 1486 { 1487 Mutex::Autolock _l(mLock); 1488 RecordThread *thread = checkRecordThread_l(input); 1489 if (thread == NULL) { 1490 ALOGE("openRecord() checkRecordThread_l failed"); 1491 lStatus = BAD_VALUE; 1492 goto Exit; 1493 } 1494 1495 pid_t pid = IPCThreadState::self()->getCallingPid(); 1496 client = registerPid(pid); 1497 1498 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1499 lSessionId = *sessionId; 1500 } else { 1501 // if no audio session id is provided, create one here 1502 lSessionId = nextUniqueId(); 1503 if (sessionId != NULL) { 1504 *sessionId = lSessionId; 1505 } 1506 } 1507 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1508 1509 // TODO: the uid should be passed in as a parameter to openRecord 1510 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1511 frameCount, lSessionId, notificationFrames, 1512 clientUid, flags, tid, &lStatus); 1513 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1514 1515 if (lStatus == NO_ERROR) { 1516 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1517 // session and move it to this thread. 1518 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1519 if (chain != 0) { 1520 Mutex::Autolock _l(thread->mLock); 1521 thread->addEffectChain_l(chain); 1522 } 1523 } 1524 } 1525 1526 if (lStatus != NO_ERROR) { 1527 // remove local strong reference to Client before deleting the RecordTrack so that the 1528 // Client destructor is called by the TrackBase destructor with mClientLock held 1529 // Don't hold mClientLock when releasing the reference on the track as the 1530 // destructor will acquire it. 1531 { 1532 Mutex::Autolock _cl(mClientLock); 1533 client.clear(); 1534 } 1535 recordTrack.clear(); 1536 goto Exit; 1537 } 1538 1539 cblk = recordTrack->getCblk(); 1540 buffers = recordTrack->getBuffers(); 1541 1542 // return handle to client 1543 recordHandle = new RecordHandle(recordTrack); 1544 1545Exit: 1546 *status = lStatus; 1547 return recordHandle; 1548} 1549 1550 1551 1552// ---------------------------------------------------------------------------- 1553 1554audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1555{ 1556 if (name == NULL) { 1557 return 0; 1558 } 1559 if (!settingsAllowed()) { 1560 return 0; 1561 } 1562 Mutex::Autolock _l(mLock); 1563 return loadHwModule_l(name); 1564} 1565 1566// loadHwModule_l() must be called with AudioFlinger::mLock held 1567audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1568{ 1569 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1570 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1571 ALOGW("loadHwModule() module %s already loaded", name); 1572 return mAudioHwDevs.keyAt(i); 1573 } 1574 } 1575 1576 audio_hw_device_t *dev; 1577 1578 int rc = load_audio_interface(name, &dev); 1579 if (rc) { 1580 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1581 return 0; 1582 } 1583 1584 mHardwareStatus = AUDIO_HW_INIT; 1585 rc = dev->init_check(dev); 1586 mHardwareStatus = AUDIO_HW_IDLE; 1587 if (rc) { 1588 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1589 return 0; 1590 } 1591 1592 // Check and cache this HAL's level of support for master mute and master 1593 // volume. If this is the first HAL opened, and it supports the get 1594 // methods, use the initial values provided by the HAL as the current 1595 // master mute and volume settings. 1596 1597 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1598 { // scope for auto-lock pattern 1599 AutoMutex lock(mHardwareLock); 1600 1601 if (0 == mAudioHwDevs.size()) { 1602 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1603 if (NULL != dev->get_master_volume) { 1604 float mv; 1605 if (OK == dev->get_master_volume(dev, &mv)) { 1606 mMasterVolume = mv; 1607 } 1608 } 1609 1610 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1611 if (NULL != dev->get_master_mute) { 1612 bool mm; 1613 if (OK == dev->get_master_mute(dev, &mm)) { 1614 mMasterMute = mm; 1615 } 1616 } 1617 } 1618 1619 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1620 if ((NULL != dev->set_master_volume) && 1621 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1622 flags = static_cast<AudioHwDevice::Flags>(flags | 1623 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1624 } 1625 1626 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1627 if ((NULL != dev->set_master_mute) && 1628 (OK == dev->set_master_mute(dev, mMasterMute))) { 1629 flags = static_cast<AudioHwDevice::Flags>(flags | 1630 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1631 } 1632 1633 mHardwareStatus = AUDIO_HW_IDLE; 1634 } 1635 1636 audio_module_handle_t handle = nextUniqueId(); 1637 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1638 1639 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1640 name, dev->common.module->name, dev->common.module->id, handle); 1641 1642 return handle; 1643 1644} 1645 1646// ---------------------------------------------------------------------------- 1647 1648uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1649{ 1650 Mutex::Autolock _l(mLock); 1651 PlaybackThread *thread = primaryPlaybackThread_l(); 1652 return thread != NULL ? thread->sampleRate() : 0; 1653} 1654 1655size_t AudioFlinger::getPrimaryOutputFrameCount() 1656{ 1657 Mutex::Autolock _l(mLock); 1658 PlaybackThread *thread = primaryPlaybackThread_l(); 1659 return thread != NULL ? thread->frameCountHAL() : 0; 1660} 1661 1662// ---------------------------------------------------------------------------- 1663 1664status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1665{ 1666 uid_t uid = IPCThreadState::self()->getCallingUid(); 1667 if (uid != AID_SYSTEM) { 1668 return PERMISSION_DENIED; 1669 } 1670 Mutex::Autolock _l(mLock); 1671 if (mIsDeviceTypeKnown) { 1672 return INVALID_OPERATION; 1673 } 1674 mIsLowRamDevice = isLowRamDevice; 1675 mIsDeviceTypeKnown = true; 1676 return NO_ERROR; 1677} 1678 1679audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1680{ 1681 Mutex::Autolock _l(mLock); 1682 1683 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1684 if (index >= 0) { 1685 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1686 mHwAvSyncIds.valueAt(index), sessionId); 1687 return mHwAvSyncIds.valueAt(index); 1688 } 1689 1690 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1691 if (dev == NULL) { 1692 return AUDIO_HW_SYNC_INVALID; 1693 } 1694 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1695 AudioParameter param = AudioParameter(String8(reply)); 1696 free(reply); 1697 1698 int value; 1699 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1700 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1701 return AUDIO_HW_SYNC_INVALID; 1702 } 1703 1704 // allow only one session for a given HW A/V sync ID. 1705 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1706 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1707 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1708 value, mHwAvSyncIds.keyAt(i)); 1709 mHwAvSyncIds.removeItemsAt(i); 1710 break; 1711 } 1712 } 1713 1714 mHwAvSyncIds.add(sessionId, value); 1715 1716 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1717 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1718 uint32_t sessions = thread->hasAudioSession(sessionId); 1719 if (sessions & PlaybackThread::TRACK_SESSION) { 1720 AudioParameter param = AudioParameter(); 1721 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1722 thread->setParameters(param.toString()); 1723 break; 1724 } 1725 } 1726 1727 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1728 return (audio_hw_sync_t)value; 1729} 1730 1731status_t AudioFlinger::systemReady() 1732{ 1733 Mutex::Autolock _l(mLock); 1734 ALOGI("%s", __FUNCTION__); 1735 if (mSystemReady) { 1736 ALOGW("%s called twice", __FUNCTION__); 1737 return NO_ERROR; 1738 } 1739 mSystemReady = true; 1740 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1741 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1742 thread->systemReady(); 1743 } 1744 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1745 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1746 thread->systemReady(); 1747 } 1748 return NO_ERROR; 1749} 1750 1751// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1752void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1753{ 1754 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1755 if (index >= 0) { 1756 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1757 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1758 AudioParameter param = AudioParameter(); 1759 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1760 thread->setParameters(param.toString()); 1761 } 1762} 1763 1764 1765// ---------------------------------------------------------------------------- 1766 1767 1768sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1769 audio_io_handle_t *output, 1770 audio_config_t *config, 1771 audio_devices_t devices, 1772 const String8& address, 1773 audio_output_flags_t flags) 1774{ 1775 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1776 if (outHwDev == NULL) { 1777 return 0; 1778 } 1779 1780 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1781 if (*output == AUDIO_IO_HANDLE_NONE) { 1782 *output = nextUniqueId(); 1783 } 1784 1785 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1786 1787 // FOR TESTING ONLY: 1788 // This if statement allows overriding the audio policy settings 1789 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1790 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1791 // Check only for Normal Mixing mode 1792 if (kEnableExtendedPrecision) { 1793 // Specify format (uncomment one below to choose) 1794 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1795 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1796 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1797 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1798 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1799 } 1800 if (kEnableExtendedChannels) { 1801 // Specify channel mask (uncomment one below to choose) 1802 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1803 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1804 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1805 } 1806 } 1807 1808 AudioStreamOut *outputStream = NULL; 1809 status_t status = outHwDev->openOutputStream( 1810 &outputStream, 1811 *output, 1812 devices, 1813 flags, 1814 config, 1815 address.string()); 1816 1817 mHardwareStatus = AUDIO_HW_IDLE; 1818 1819 if (status == NO_ERROR) { 1820 1821 PlaybackThread *thread; 1822 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1823 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1824 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1825 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1826 || !isValidPcmSinkFormat(config->format) 1827 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1828 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1829 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1830 } else { 1831 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1832 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1833 } 1834 mPlaybackThreads.add(*output, thread); 1835 return thread; 1836 } 1837 1838 return 0; 1839} 1840 1841status_t AudioFlinger::openOutput(audio_module_handle_t module, 1842 audio_io_handle_t *output, 1843 audio_config_t *config, 1844 audio_devices_t *devices, 1845 const String8& address, 1846 uint32_t *latencyMs, 1847 audio_output_flags_t flags) 1848{ 1849 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1850 module, 1851 (devices != NULL) ? *devices : 0, 1852 config->sample_rate, 1853 config->format, 1854 config->channel_mask, 1855 flags); 1856 1857 if (*devices == AUDIO_DEVICE_NONE) { 1858 return BAD_VALUE; 1859 } 1860 1861 Mutex::Autolock _l(mLock); 1862 1863 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1864 if (thread != 0) { 1865 *latencyMs = thread->latency(); 1866 1867 // notify client processes of the new output creation 1868 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1869 1870 // the first primary output opened designates the primary hw device 1871 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1872 ALOGI("Using module %d has the primary audio interface", module); 1873 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1874 1875 AutoMutex lock(mHardwareLock); 1876 mHardwareStatus = AUDIO_HW_SET_MODE; 1877 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1878 mHardwareStatus = AUDIO_HW_IDLE; 1879 } 1880 return NO_ERROR; 1881 } 1882 1883 return NO_INIT; 1884} 1885 1886audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1887 audio_io_handle_t output2) 1888{ 1889 Mutex::Autolock _l(mLock); 1890 MixerThread *thread1 = checkMixerThread_l(output1); 1891 MixerThread *thread2 = checkMixerThread_l(output2); 1892 1893 if (thread1 == NULL || thread2 == NULL) { 1894 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1895 output2); 1896 return AUDIO_IO_HANDLE_NONE; 1897 } 1898 1899 audio_io_handle_t id = nextUniqueId(); 1900 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1901 thread->addOutputTrack(thread2); 1902 mPlaybackThreads.add(id, thread); 1903 // notify client processes of the new output creation 1904 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1905 return id; 1906} 1907 1908status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1909{ 1910 return closeOutput_nonvirtual(output); 1911} 1912 1913status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1914{ 1915 // keep strong reference on the playback thread so that 1916 // it is not destroyed while exit() is executed 1917 sp<PlaybackThread> thread; 1918 { 1919 Mutex::Autolock _l(mLock); 1920 thread = checkPlaybackThread_l(output); 1921 if (thread == NULL) { 1922 return BAD_VALUE; 1923 } 1924 1925 ALOGV("closeOutput() %d", output); 1926 1927 if (thread->type() == ThreadBase::MIXER) { 1928 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1929 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1930 DuplicatingThread *dupThread = 1931 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1932 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1933 } 1934 } 1935 } 1936 1937 1938 mPlaybackThreads.removeItem(output); 1939 // save all effects to the default thread 1940 if (mPlaybackThreads.size()) { 1941 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1942 if (dstThread != NULL) { 1943 // audioflinger lock is held here so the acquisition order of thread locks does not 1944 // matter 1945 Mutex::Autolock _dl(dstThread->mLock); 1946 Mutex::Autolock _sl(thread->mLock); 1947 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1948 for (size_t i = 0; i < effectChains.size(); i ++) { 1949 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1950 } 1951 } 1952 } 1953 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1954 ioDesc->mIoHandle = output; 1955 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1956 } 1957 thread->exit(); 1958 // The thread entity (active unit of execution) is no longer running here, 1959 // but the ThreadBase container still exists. 1960 1961 if (!thread->isDuplicating()) { 1962 closeOutputFinish(thread); 1963 } 1964 1965 return NO_ERROR; 1966} 1967 1968void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread) 1969{ 1970 AudioStreamOut *out = thread->clearOutput(); 1971 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1972 // from now on thread->mOutput is NULL 1973 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1974 delete out; 1975} 1976 1977void AudioFlinger::closeOutputInternal_l(const sp<PlaybackThread>& thread) 1978{ 1979 mPlaybackThreads.removeItem(thread->mId); 1980 thread->exit(); 1981 closeOutputFinish(thread); 1982} 1983 1984status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1985{ 1986 Mutex::Autolock _l(mLock); 1987 PlaybackThread *thread = checkPlaybackThread_l(output); 1988 1989 if (thread == NULL) { 1990 return BAD_VALUE; 1991 } 1992 1993 ALOGV("suspendOutput() %d", output); 1994 thread->suspend(); 1995 1996 return NO_ERROR; 1997} 1998 1999status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2000{ 2001 Mutex::Autolock _l(mLock); 2002 PlaybackThread *thread = checkPlaybackThread_l(output); 2003 2004 if (thread == NULL) { 2005 return BAD_VALUE; 2006 } 2007 2008 ALOGV("restoreOutput() %d", output); 2009 2010 thread->restore(); 2011 2012 return NO_ERROR; 2013} 2014 2015status_t AudioFlinger::openInput(audio_module_handle_t module, 2016 audio_io_handle_t *input, 2017 audio_config_t *config, 2018 audio_devices_t *devices, 2019 const String8& address, 2020 audio_source_t source, 2021 audio_input_flags_t flags) 2022{ 2023 Mutex::Autolock _l(mLock); 2024 2025 if (*devices == AUDIO_DEVICE_NONE) { 2026 return BAD_VALUE; 2027 } 2028 2029 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2030 2031 if (thread != 0) { 2032 // notify client processes of the new input creation 2033 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2034 return NO_ERROR; 2035 } 2036 return NO_INIT; 2037} 2038 2039sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2040 audio_io_handle_t *input, 2041 audio_config_t *config, 2042 audio_devices_t devices, 2043 const String8& address, 2044 audio_source_t source, 2045 audio_input_flags_t flags) 2046{ 2047 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2048 if (inHwDev == NULL) { 2049 *input = AUDIO_IO_HANDLE_NONE; 2050 return 0; 2051 } 2052 2053 if (*input == AUDIO_IO_HANDLE_NONE) { 2054 *input = nextUniqueId(); 2055 } 2056 2057 audio_config_t halconfig = *config; 2058 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2059 audio_stream_in_t *inStream = NULL; 2060 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2061 &inStream, flags, address.string(), source); 2062 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2063 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2064 inStream, 2065 halconfig.sample_rate, 2066 halconfig.format, 2067 halconfig.channel_mask, 2068 flags, 2069 status, address.string()); 2070 2071 // If the input could not be opened with the requested parameters and we can handle the 2072 // conversion internally, try to open again with the proposed parameters. 2073 if (status == BAD_VALUE && 2074 audio_is_linear_pcm(config->format) && 2075 audio_is_linear_pcm(halconfig.format) && 2076 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2077 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2078 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2079 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2080 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2081 inStream = NULL; 2082 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2083 &inStream, flags, address.string(), source); 2084 // FIXME log this new status; HAL should not propose any further changes 2085 } 2086 2087 if (status == NO_ERROR && inStream != NULL) { 2088 2089#ifdef TEE_SINK 2090 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2091 // or (re-)create if current Pipe is idle and does not match the new format 2092 sp<NBAIO_Sink> teeSink; 2093 enum { 2094 TEE_SINK_NO, // don't copy input 2095 TEE_SINK_NEW, // copy input using a new pipe 2096 TEE_SINK_OLD, // copy input using an existing pipe 2097 } kind; 2098 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2099 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2100 if (!mTeeSinkInputEnabled) { 2101 kind = TEE_SINK_NO; 2102 } else if (!Format_isValid(format)) { 2103 kind = TEE_SINK_NO; 2104 } else if (mRecordTeeSink == 0) { 2105 kind = TEE_SINK_NEW; 2106 } else if (mRecordTeeSink->getStrongCount() != 1) { 2107 kind = TEE_SINK_NO; 2108 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2109 kind = TEE_SINK_OLD; 2110 } else { 2111 kind = TEE_SINK_NEW; 2112 } 2113 switch (kind) { 2114 case TEE_SINK_NEW: { 2115 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2116 size_t numCounterOffers = 0; 2117 const NBAIO_Format offers[1] = {format}; 2118 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2119 ALOG_ASSERT(index == 0); 2120 PipeReader *pipeReader = new PipeReader(*pipe); 2121 numCounterOffers = 0; 2122 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2123 ALOG_ASSERT(index == 0); 2124 mRecordTeeSink = pipe; 2125 mRecordTeeSource = pipeReader; 2126 teeSink = pipe; 2127 } 2128 break; 2129 case TEE_SINK_OLD: 2130 teeSink = mRecordTeeSink; 2131 break; 2132 case TEE_SINK_NO: 2133 default: 2134 break; 2135 } 2136#endif 2137 2138 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2139 2140 // Start record thread 2141 // RecordThread requires both input and output device indication to forward to audio 2142 // pre processing modules 2143 sp<RecordThread> thread = new RecordThread(this, 2144 inputStream, 2145 *input, 2146 primaryOutputDevice_l(), 2147 devices, 2148 mSystemReady 2149#ifdef TEE_SINK 2150 , teeSink 2151#endif 2152 ); 2153 mRecordThreads.add(*input, thread); 2154 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2155 return thread; 2156 } 2157 2158 *input = AUDIO_IO_HANDLE_NONE; 2159 return 0; 2160} 2161 2162status_t AudioFlinger::closeInput(audio_io_handle_t input) 2163{ 2164 return closeInput_nonvirtual(input); 2165} 2166 2167status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2168{ 2169 // keep strong reference on the record thread so that 2170 // it is not destroyed while exit() is executed 2171 sp<RecordThread> thread; 2172 { 2173 Mutex::Autolock _l(mLock); 2174 thread = checkRecordThread_l(input); 2175 if (thread == 0) { 2176 return BAD_VALUE; 2177 } 2178 2179 ALOGV("closeInput() %d", input); 2180 2181 // If we still have effect chains, it means that a client still holds a handle 2182 // on at least one effect. We must either move the chain to an existing thread with the 2183 // same session ID or put it aside in case a new record thread is opened for a 2184 // new capture on the same session 2185 sp<EffectChain> chain; 2186 { 2187 Mutex::Autolock _sl(thread->mLock); 2188 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2189 // Note: maximum one chain per record thread 2190 if (effectChains.size() != 0) { 2191 chain = effectChains[0]; 2192 } 2193 } 2194 if (chain != 0) { 2195 // first check if a record thread is already opened with a client on the same session. 2196 // This should only happen in case of overlap between one thread tear down and the 2197 // creation of its replacement 2198 size_t i; 2199 for (i = 0; i < mRecordThreads.size(); i++) { 2200 sp<RecordThread> t = mRecordThreads.valueAt(i); 2201 if (t == thread) { 2202 continue; 2203 } 2204 if (t->hasAudioSession(chain->sessionId()) != 0) { 2205 Mutex::Autolock _l(t->mLock); 2206 ALOGV("closeInput() found thread %d for effect session %d", 2207 t->id(), chain->sessionId()); 2208 t->addEffectChain_l(chain); 2209 break; 2210 } 2211 } 2212 // put the chain aside if we could not find a record thread with the same session id. 2213 if (i == mRecordThreads.size()) { 2214 putOrphanEffectChain_l(chain); 2215 } 2216 } 2217 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2218 ioDesc->mIoHandle = input; 2219 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2220 mRecordThreads.removeItem(input); 2221 } 2222 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2223 // we have a different lock for notification client 2224 closeInputFinish(thread); 2225 return NO_ERROR; 2226} 2227 2228void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread) 2229{ 2230 thread->exit(); 2231 AudioStreamIn *in = thread->clearInput(); 2232 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2233 // from now on thread->mInput is NULL 2234 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2235 delete in; 2236} 2237 2238void AudioFlinger::closeInputInternal_l(const sp<RecordThread>& thread) 2239{ 2240 mRecordThreads.removeItem(thread->mId); 2241 closeInputFinish(thread); 2242} 2243 2244status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2245{ 2246 Mutex::Autolock _l(mLock); 2247 ALOGV("invalidateStream() stream %d", stream); 2248 2249 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2250 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2251 thread->invalidateTracks(stream); 2252 } 2253 2254 return NO_ERROR; 2255} 2256 2257 2258audio_unique_id_t AudioFlinger::newAudioUniqueId() 2259{ 2260 return nextUniqueId(); 2261} 2262 2263void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2264{ 2265 Mutex::Autolock _l(mLock); 2266 pid_t caller = IPCThreadState::self()->getCallingPid(); 2267 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2268 if (pid != -1 && (caller == getpid_cached)) { 2269 caller = pid; 2270 } 2271 2272 { 2273 Mutex::Autolock _cl(mClientLock); 2274 // Ignore requests received from processes not known as notification client. The request 2275 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2276 // called from a different pid leaving a stale session reference. Also we don't know how 2277 // to clear this reference if the client process dies. 2278 if (mNotificationClients.indexOfKey(caller) < 0) { 2279 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2280 return; 2281 } 2282 } 2283 2284 size_t num = mAudioSessionRefs.size(); 2285 for (size_t i = 0; i< num; i++) { 2286 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2287 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2288 ref->mCnt++; 2289 ALOGV(" incremented refcount to %d", ref->mCnt); 2290 return; 2291 } 2292 } 2293 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2294 ALOGV(" added new entry for %d", audioSession); 2295} 2296 2297void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2298{ 2299 Mutex::Autolock _l(mLock); 2300 pid_t caller = IPCThreadState::self()->getCallingPid(); 2301 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2302 if (pid != -1 && (caller == getpid_cached)) { 2303 caller = pid; 2304 } 2305 size_t num = mAudioSessionRefs.size(); 2306 for (size_t i = 0; i< num; i++) { 2307 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2308 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2309 ref->mCnt--; 2310 ALOGV(" decremented refcount to %d", ref->mCnt); 2311 if (ref->mCnt == 0) { 2312 mAudioSessionRefs.removeAt(i); 2313 delete ref; 2314 purgeStaleEffects_l(); 2315 } 2316 return; 2317 } 2318 } 2319 // If the caller is mediaserver it is likely that the session being released was acquired 2320 // on behalf of a process not in notification clients and we ignore the warning. 2321 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2322} 2323 2324void AudioFlinger::purgeStaleEffects_l() { 2325 2326 ALOGV("purging stale effects"); 2327 2328 Vector< sp<EffectChain> > chains; 2329 2330 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2331 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2332 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2333 sp<EffectChain> ec = t->mEffectChains[j]; 2334 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2335 chains.push(ec); 2336 } 2337 } 2338 } 2339 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2340 sp<RecordThread> t = mRecordThreads.valueAt(i); 2341 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2342 sp<EffectChain> ec = t->mEffectChains[j]; 2343 chains.push(ec); 2344 } 2345 } 2346 2347 for (size_t i = 0; i < chains.size(); i++) { 2348 sp<EffectChain> ec = chains[i]; 2349 int sessionid = ec->sessionId(); 2350 sp<ThreadBase> t = ec->mThread.promote(); 2351 if (t == 0) { 2352 continue; 2353 } 2354 size_t numsessionrefs = mAudioSessionRefs.size(); 2355 bool found = false; 2356 for (size_t k = 0; k < numsessionrefs; k++) { 2357 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2358 if (ref->mSessionid == sessionid) { 2359 ALOGV(" session %d still exists for %d with %d refs", 2360 sessionid, ref->mPid, ref->mCnt); 2361 found = true; 2362 break; 2363 } 2364 } 2365 if (!found) { 2366 Mutex::Autolock _l(t->mLock); 2367 // remove all effects from the chain 2368 while (ec->mEffects.size()) { 2369 sp<EffectModule> effect = ec->mEffects[0]; 2370 effect->unPin(); 2371 t->removeEffect_l(effect); 2372 if (effect->purgeHandles()) { 2373 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2374 } 2375 AudioSystem::unregisterEffect(effect->id()); 2376 } 2377 } 2378 } 2379 return; 2380} 2381 2382// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2383AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2384{ 2385 return mPlaybackThreads.valueFor(output).get(); 2386} 2387 2388// checkMixerThread_l() must be called with AudioFlinger::mLock held 2389AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2390{ 2391 PlaybackThread *thread = checkPlaybackThread_l(output); 2392 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2393} 2394 2395// checkRecordThread_l() must be called with AudioFlinger::mLock held 2396AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2397{ 2398 return mRecordThreads.valueFor(input).get(); 2399} 2400 2401uint32_t AudioFlinger::nextUniqueId() 2402{ 2403 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2404} 2405 2406AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2407{ 2408 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2409 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2410 if(thread->isDuplicating()) { 2411 continue; 2412 } 2413 AudioStreamOut *output = thread->getOutput(); 2414 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2415 return thread; 2416 } 2417 } 2418 return NULL; 2419} 2420 2421audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2422{ 2423 PlaybackThread *thread = primaryPlaybackThread_l(); 2424 2425 if (thread == NULL) { 2426 return 0; 2427 } 2428 2429 return thread->outDevice(); 2430} 2431 2432sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2433 int triggerSession, 2434 int listenerSession, 2435 sync_event_callback_t callBack, 2436 const wp<RefBase>& cookie) 2437{ 2438 Mutex::Autolock _l(mLock); 2439 2440 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2441 status_t playStatus = NAME_NOT_FOUND; 2442 status_t recStatus = NAME_NOT_FOUND; 2443 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2444 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2445 if (playStatus == NO_ERROR) { 2446 return event; 2447 } 2448 } 2449 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2450 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2451 if (recStatus == NO_ERROR) { 2452 return event; 2453 } 2454 } 2455 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2456 mPendingSyncEvents.add(event); 2457 } else { 2458 ALOGV("createSyncEvent() invalid event %d", event->type()); 2459 event.clear(); 2460 } 2461 return event; 2462} 2463 2464// ---------------------------------------------------------------------------- 2465// Effect management 2466// ---------------------------------------------------------------------------- 2467 2468 2469status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2470{ 2471 Mutex::Autolock _l(mLock); 2472 return EffectQueryNumberEffects(numEffects); 2473} 2474 2475status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2476{ 2477 Mutex::Autolock _l(mLock); 2478 return EffectQueryEffect(index, descriptor); 2479} 2480 2481status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2482 effect_descriptor_t *descriptor) const 2483{ 2484 Mutex::Autolock _l(mLock); 2485 return EffectGetDescriptor(pUuid, descriptor); 2486} 2487 2488 2489sp<IEffect> AudioFlinger::createEffect( 2490 effect_descriptor_t *pDesc, 2491 const sp<IEffectClient>& effectClient, 2492 int32_t priority, 2493 audio_io_handle_t io, 2494 int sessionId, 2495 const String16& opPackageName, 2496 status_t *status, 2497 int *id, 2498 int *enabled) 2499{ 2500 status_t lStatus = NO_ERROR; 2501 sp<EffectHandle> handle; 2502 effect_descriptor_t desc; 2503 2504 pid_t pid = IPCThreadState::self()->getCallingPid(); 2505 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2506 pid, effectClient.get(), priority, sessionId, io); 2507 2508 if (pDesc == NULL) { 2509 lStatus = BAD_VALUE; 2510 goto Exit; 2511 } 2512 2513 // check audio settings permission for global effects 2514 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2515 lStatus = PERMISSION_DENIED; 2516 goto Exit; 2517 } 2518 2519 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2520 // that can only be created by audio policy manager (running in same process) 2521 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2522 lStatus = PERMISSION_DENIED; 2523 goto Exit; 2524 } 2525 2526 { 2527 if (!EffectIsNullUuid(&pDesc->uuid)) { 2528 // if uuid is specified, request effect descriptor 2529 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2530 if (lStatus < 0) { 2531 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2532 goto Exit; 2533 } 2534 } else { 2535 // if uuid is not specified, look for an available implementation 2536 // of the required type in effect factory 2537 if (EffectIsNullUuid(&pDesc->type)) { 2538 ALOGW("createEffect() no effect type"); 2539 lStatus = BAD_VALUE; 2540 goto Exit; 2541 } 2542 uint32_t numEffects = 0; 2543 effect_descriptor_t d; 2544 d.flags = 0; // prevent compiler warning 2545 bool found = false; 2546 2547 lStatus = EffectQueryNumberEffects(&numEffects); 2548 if (lStatus < 0) { 2549 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2550 goto Exit; 2551 } 2552 for (uint32_t i = 0; i < numEffects; i++) { 2553 lStatus = EffectQueryEffect(i, &desc); 2554 if (lStatus < 0) { 2555 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2556 continue; 2557 } 2558 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2559 // If matching type found save effect descriptor. If the session is 2560 // 0 and the effect is not auxiliary, continue enumeration in case 2561 // an auxiliary version of this effect type is available 2562 found = true; 2563 d = desc; 2564 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2565 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2566 break; 2567 } 2568 } 2569 } 2570 if (!found) { 2571 lStatus = BAD_VALUE; 2572 ALOGW("createEffect() effect not found"); 2573 goto Exit; 2574 } 2575 // For same effect type, chose auxiliary version over insert version if 2576 // connect to output mix (Compliance to OpenSL ES) 2577 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2578 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2579 desc = d; 2580 } 2581 } 2582 2583 // Do not allow auxiliary effects on a session different from 0 (output mix) 2584 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2585 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2586 lStatus = INVALID_OPERATION; 2587 goto Exit; 2588 } 2589 2590 // check recording permission for visualizer 2591 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2592 !recordingAllowed(opPackageName)) { 2593 lStatus = PERMISSION_DENIED; 2594 goto Exit; 2595 } 2596 2597 // return effect descriptor 2598 *pDesc = desc; 2599 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2600 // if the output returned by getOutputForEffect() is removed before we lock the 2601 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2602 // and we will exit safely 2603 io = AudioSystem::getOutputForEffect(&desc); 2604 ALOGV("createEffect got output %d", io); 2605 } 2606 2607 Mutex::Autolock _l(mLock); 2608 2609 // If output is not specified try to find a matching audio session ID in one of the 2610 // output threads. 2611 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2612 // because of code checking output when entering the function. 2613 // Note: io is never 0 when creating an effect on an input 2614 if (io == AUDIO_IO_HANDLE_NONE) { 2615 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2616 // output must be specified by AudioPolicyManager when using session 2617 // AUDIO_SESSION_OUTPUT_STAGE 2618 lStatus = BAD_VALUE; 2619 goto Exit; 2620 } 2621 // look for the thread where the specified audio session is present 2622 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2623 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2624 io = mPlaybackThreads.keyAt(i); 2625 break; 2626 } 2627 } 2628 if (io == 0) { 2629 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2630 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2631 io = mRecordThreads.keyAt(i); 2632 break; 2633 } 2634 } 2635 } 2636 // If no output thread contains the requested session ID, default to 2637 // first output. The effect chain will be moved to the correct output 2638 // thread when a track with the same session ID is created 2639 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2640 io = mPlaybackThreads.keyAt(0); 2641 } 2642 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2643 } 2644 ThreadBase *thread = checkRecordThread_l(io); 2645 if (thread == NULL) { 2646 thread = checkPlaybackThread_l(io); 2647 if (thread == NULL) { 2648 ALOGE("createEffect() unknown output thread"); 2649 lStatus = BAD_VALUE; 2650 goto Exit; 2651 } 2652 } else { 2653 // Check if one effect chain was awaiting for an effect to be created on this 2654 // session and used it instead of creating a new one. 2655 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2656 if (chain != 0) { 2657 Mutex::Autolock _l(thread->mLock); 2658 thread->addEffectChain_l(chain); 2659 } 2660 } 2661 2662 sp<Client> client = registerPid(pid); 2663 2664 // create effect on selected output thread 2665 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2666 &desc, enabled, &lStatus); 2667 if (handle != 0 && id != NULL) { 2668 *id = handle->id(); 2669 } 2670 if (handle == 0) { 2671 // remove local strong reference to Client with mClientLock held 2672 Mutex::Autolock _cl(mClientLock); 2673 client.clear(); 2674 } 2675 } 2676 2677Exit: 2678 *status = lStatus; 2679 return handle; 2680} 2681 2682status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2683 audio_io_handle_t dstOutput) 2684{ 2685 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2686 sessionId, srcOutput, dstOutput); 2687 Mutex::Autolock _l(mLock); 2688 if (srcOutput == dstOutput) { 2689 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2690 return NO_ERROR; 2691 } 2692 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2693 if (srcThread == NULL) { 2694 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2695 return BAD_VALUE; 2696 } 2697 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2698 if (dstThread == NULL) { 2699 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2700 return BAD_VALUE; 2701 } 2702 2703 Mutex::Autolock _dl(dstThread->mLock); 2704 Mutex::Autolock _sl(srcThread->mLock); 2705 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2706} 2707 2708// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2709status_t AudioFlinger::moveEffectChain_l(int sessionId, 2710 AudioFlinger::PlaybackThread *srcThread, 2711 AudioFlinger::PlaybackThread *dstThread, 2712 bool reRegister) 2713{ 2714 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2715 sessionId, srcThread, dstThread); 2716 2717 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2718 if (chain == 0) { 2719 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2720 sessionId, srcThread); 2721 return INVALID_OPERATION; 2722 } 2723 2724 // Check whether the destination thread has a channel count of FCC_2, which is 2725 // currently required for (most) effects. Prevent moving the effect chain here rather 2726 // than disabling the addEffect_l() call in dstThread below. 2727 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2728 dstThread->mChannelCount != FCC_2) { 2729 ALOGW("moveEffectChain_l() effect chain failed because" 2730 " destination thread %p channel count(%u) != %u", 2731 dstThread, dstThread->mChannelCount, FCC_2); 2732 return INVALID_OPERATION; 2733 } 2734 2735 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2736 // so that a new chain is created with correct parameters when first effect is added. This is 2737 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2738 // removed. 2739 srcThread->removeEffectChain_l(chain); 2740 2741 // transfer all effects one by one so that new effect chain is created on new thread with 2742 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2743 sp<EffectChain> dstChain; 2744 uint32_t strategy = 0; // prevent compiler warning 2745 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2746 Vector< sp<EffectModule> > removed; 2747 status_t status = NO_ERROR; 2748 while (effect != 0) { 2749 srcThread->removeEffect_l(effect); 2750 removed.add(effect); 2751 status = dstThread->addEffect_l(effect); 2752 if (status != NO_ERROR) { 2753 break; 2754 } 2755 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2756 if (effect->state() == EffectModule::ACTIVE || 2757 effect->state() == EffectModule::STOPPING) { 2758 effect->start(); 2759 } 2760 // if the move request is not received from audio policy manager, the effect must be 2761 // re-registered with the new strategy and output 2762 if (dstChain == 0) { 2763 dstChain = effect->chain().promote(); 2764 if (dstChain == 0) { 2765 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2766 status = NO_INIT; 2767 break; 2768 } 2769 strategy = dstChain->strategy(); 2770 } 2771 if (reRegister) { 2772 AudioSystem::unregisterEffect(effect->id()); 2773 AudioSystem::registerEffect(&effect->desc(), 2774 dstThread->id(), 2775 strategy, 2776 sessionId, 2777 effect->id()); 2778 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2779 } 2780 effect = chain->getEffectFromId_l(0); 2781 } 2782 2783 if (status != NO_ERROR) { 2784 for (size_t i = 0; i < removed.size(); i++) { 2785 srcThread->addEffect_l(removed[i]); 2786 if (dstChain != 0 && reRegister) { 2787 AudioSystem::unregisterEffect(removed[i]->id()); 2788 AudioSystem::registerEffect(&removed[i]->desc(), 2789 srcThread->id(), 2790 strategy, 2791 sessionId, 2792 removed[i]->id()); 2793 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2794 } 2795 } 2796 } 2797 2798 return status; 2799} 2800 2801bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2802{ 2803 if (mGlobalEffectEnableTime != 0 && 2804 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2805 return true; 2806 } 2807 2808 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2809 sp<EffectChain> ec = 2810 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2811 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2812 return true; 2813 } 2814 } 2815 return false; 2816} 2817 2818void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2819{ 2820 Mutex::Autolock _l(mLock); 2821 2822 mGlobalEffectEnableTime = systemTime(); 2823 2824 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2825 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2826 if (t->mType == ThreadBase::OFFLOAD) { 2827 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2828 } 2829 } 2830 2831} 2832 2833status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2834{ 2835 audio_session_t session = (audio_session_t)chain->sessionId(); 2836 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2837 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2838 if (index >= 0) { 2839 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2840 return ALREADY_EXISTS; 2841 } 2842 mOrphanEffectChains.add(session, chain); 2843 return NO_ERROR; 2844} 2845 2846sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2847{ 2848 sp<EffectChain> chain; 2849 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2850 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2851 if (index >= 0) { 2852 chain = mOrphanEffectChains.valueAt(index); 2853 mOrphanEffectChains.removeItemsAt(index); 2854 } 2855 return chain; 2856} 2857 2858bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2859{ 2860 Mutex::Autolock _l(mLock); 2861 audio_session_t session = (audio_session_t)effect->sessionId(); 2862 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2863 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2864 if (index >= 0) { 2865 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2866 if (chain->removeEffect_l(effect) == 0) { 2867 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2868 mOrphanEffectChains.removeItemsAt(index); 2869 } 2870 return true; 2871 } 2872 return false; 2873} 2874 2875 2876struct Entry { 2877#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2878 char mFileName[TEE_MAX_FILENAME]; 2879}; 2880 2881int comparEntry(const void *p1, const void *p2) 2882{ 2883 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2884} 2885 2886#ifdef TEE_SINK 2887void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2888{ 2889 NBAIO_Source *teeSource = source.get(); 2890 if (teeSource != NULL) { 2891 // .wav rotation 2892 // There is a benign race condition if 2 threads call this simultaneously. 2893 // They would both traverse the directory, but the result would simply be 2894 // failures at unlink() which are ignored. It's also unlikely since 2895 // normally dumpsys is only done by bugreport or from the command line. 2896 char teePath[32+256]; 2897 strcpy(teePath, "/data/misc/media"); 2898 size_t teePathLen = strlen(teePath); 2899 DIR *dir = opendir(teePath); 2900 teePath[teePathLen++] = '/'; 2901 if (dir != NULL) { 2902#define TEE_MAX_SORT 20 // number of entries to sort 2903#define TEE_MAX_KEEP 10 // number of entries to keep 2904 struct Entry entries[TEE_MAX_SORT]; 2905 size_t entryCount = 0; 2906 while (entryCount < TEE_MAX_SORT) { 2907 struct dirent de; 2908 struct dirent *result = NULL; 2909 int rc = readdir_r(dir, &de, &result); 2910 if (rc != 0) { 2911 ALOGW("readdir_r failed %d", rc); 2912 break; 2913 } 2914 if (result == NULL) { 2915 break; 2916 } 2917 if (result != &de) { 2918 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2919 break; 2920 } 2921 // ignore non .wav file entries 2922 size_t nameLen = strlen(de.d_name); 2923 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2924 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2925 continue; 2926 } 2927 strcpy(entries[entryCount++].mFileName, de.d_name); 2928 } 2929 (void) closedir(dir); 2930 if (entryCount > TEE_MAX_KEEP) { 2931 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2932 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2933 strcpy(&teePath[teePathLen], entries[i].mFileName); 2934 (void) unlink(teePath); 2935 } 2936 } 2937 } else { 2938 if (fd >= 0) { 2939 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2940 } 2941 } 2942 char teeTime[16]; 2943 struct timeval tv; 2944 gettimeofday(&tv, NULL); 2945 struct tm tm; 2946 localtime_r(&tv.tv_sec, &tm); 2947 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2948 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2949 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2950 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2951 if (teeFd >= 0) { 2952 // FIXME use libsndfile 2953 char wavHeader[44]; 2954 memcpy(wavHeader, 2955 "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", 2956 sizeof(wavHeader)); 2957 NBAIO_Format format = teeSource->format(); 2958 unsigned channelCount = Format_channelCount(format); 2959 uint32_t sampleRate = Format_sampleRate(format); 2960 size_t frameSize = Format_frameSize(format); 2961 wavHeader[22] = channelCount; // number of channels 2962 wavHeader[24] = sampleRate; // sample rate 2963 wavHeader[25] = sampleRate >> 8; 2964 wavHeader[32] = frameSize; // block alignment 2965 wavHeader[33] = frameSize >> 8; 2966 write(teeFd, wavHeader, sizeof(wavHeader)); 2967 size_t total = 0; 2968 bool firstRead = true; 2969#define TEE_SINK_READ 1024 // frames per I/O operation 2970 void *buffer = malloc(TEE_SINK_READ * frameSize); 2971 for (;;) { 2972 size_t count = TEE_SINK_READ; 2973 ssize_t actual = teeSource->read(buffer, count, 2974 AudioBufferProvider::kInvalidPTS); 2975 bool wasFirstRead = firstRead; 2976 firstRead = false; 2977 if (actual <= 0) { 2978 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2979 continue; 2980 } 2981 break; 2982 } 2983 ALOG_ASSERT(actual <= (ssize_t)count); 2984 write(teeFd, buffer, actual * frameSize); 2985 total += actual; 2986 } 2987 free(buffer); 2988 lseek(teeFd, (off_t) 4, SEEK_SET); 2989 uint32_t temp = 44 + total * frameSize - 8; 2990 // FIXME not big-endian safe 2991 write(teeFd, &temp, sizeof(temp)); 2992 lseek(teeFd, (off_t) 40, SEEK_SET); 2993 temp = total * frameSize; 2994 // FIXME not big-endian safe 2995 write(teeFd, &temp, sizeof(temp)); 2996 close(teeFd); 2997 if (fd >= 0) { 2998 dprintf(fd, "tee copied to %s\n", teePath); 2999 } 3000 } else { 3001 if (fd >= 0) { 3002 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3003 } 3004 } 3005 } 3006} 3007#endif 3008 3009// ---------------------------------------------------------------------------- 3010 3011status_t AudioFlinger::onTransact( 3012 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3013{ 3014 return BnAudioFlinger::onTransact(code, data, reply, flags); 3015} 3016 3017} // namespace android 3018