AudioFlinger.cpp revision 72215491c60fbcdb9a2f0be782e24e39cca249c5
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/EffectsFactoryApi.h> 49#include <audio_effects/effect_visualizer.h> 50#include <audio_effects/effect_ns.h> 51#include <audio_effects/effect_aec.h> 52 53#include <audio_utils/primitives.h> 54 55#include <powermanager/PowerManager.h> 56 57#include <common_time/cc_helper.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <private/android_filesystem_config.h> 65 66// ---------------------------------------------------------------------------- 67 68// Note: the following macro is used for extremely verbose logging message. In 69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70// 0; but one side effect of this is to turn all LOGV's as well. Some messages 71// are so verbose that we want to suppress them even when we have ALOG_ASSERT 72// turned on. Do not uncomment the #def below unless you really know what you 73// are doing and want to see all of the extremely verbose messages. 74//#define VERY_VERY_VERBOSE_LOGGING 75#ifdef VERY_VERY_VERBOSE_LOGGING 76#define ALOGVV ALOGV 77#else 78#define ALOGVV(a...) do { } while(0) 79#endif 80 81namespace android { 82 83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85static const char kClientLockedString[] = "Client lock is taken\n"; 86 87 88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 89 90uint32_t AudioFlinger::mScreenState; 91 92#ifdef TEE_SINK 93bool AudioFlinger::mTeeSinkInputEnabled = false; 94bool AudioFlinger::mTeeSinkOutputEnabled = false; 95bool AudioFlinger::mTeeSinkTrackEnabled = false; 96 97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 100#endif 101 102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 103// we define a minimum time during which a global effect is considered enabled. 104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 105 106// ---------------------------------------------------------------------------- 107 108const char *formatToString(audio_format_t format) { 109 switch (format & AUDIO_FORMAT_MAIN_MASK) { 110 case AUDIO_FORMAT_PCM: 111 switch (format) { 112 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 113 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 114 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 115 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 116 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 117 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 118 default: 119 break; 120 } 121 break; 122 case AUDIO_FORMAT_MP3: return "mp3"; 123 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 124 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 125 case AUDIO_FORMAT_AAC: return "aac"; 126 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 127 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 128 case AUDIO_FORMAT_VORBIS: return "vorbis"; 129 case AUDIO_FORMAT_OPUS: return "opus"; 130 case AUDIO_FORMAT_AC3: return "ac-3"; 131 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 132 default: 133 break; 134 } 135 return "unknown"; 136} 137 138static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 139{ 140 const hw_module_t *mod; 141 int rc; 142 143 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 144 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 145 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 146 if (rc) { 147 goto out; 148 } 149 rc = audio_hw_device_open(mod, dev); 150 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 156 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 157 rc = BAD_VALUE; 158 goto out; 159 } 160 return 0; 161 162out: 163 *dev = NULL; 164 return rc; 165} 166 167// ---------------------------------------------------------------------------- 168 169AudioFlinger::AudioFlinger() 170 : BnAudioFlinger(), 171 mPrimaryHardwareDev(NULL), 172 mAudioHwDevs(NULL), 173 mHardwareStatus(AUDIO_HW_IDLE), 174 mMasterVolume(1.0f), 175 mMasterMute(false), 176 mNextUniqueId(1), 177 mMode(AUDIO_MODE_INVALID), 178 mBtNrecIsOff(false), 179 mIsLowRamDevice(true), 180 mIsDeviceTypeKnown(false), 181 mGlobalEffectEnableTime(0), 182 mPrimaryOutputSampleRate(0) 183{ 184 getpid_cached = getpid(); 185 char value[PROPERTY_VALUE_MAX]; 186 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 187 if (doLog) { 188 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 189 } 190 191#ifdef TEE_SINK 192 (void) property_get("ro.debuggable", value, "0"); 193 int debuggable = atoi(value); 194 int teeEnabled = 0; 195 if (debuggable) { 196 (void) property_get("af.tee", value, "0"); 197 teeEnabled = atoi(value); 198 } 199 // FIXME symbolic constants here 200 if (teeEnabled & 1) { 201 mTeeSinkInputEnabled = true; 202 } 203 if (teeEnabled & 2) { 204 mTeeSinkOutputEnabled = true; 205 } 206 if (teeEnabled & 4) { 207 mTeeSinkTrackEnabled = true; 208 } 209#endif 210} 211 212void AudioFlinger::onFirstRef() 213{ 214 int rc = 0; 215 216 Mutex::Autolock _l(mLock); 217 218 /* TODO: move all this work into an Init() function */ 219 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 220 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 221 uint32_t int_val; 222 if (1 == sscanf(val_str, "%u", &int_val)) { 223 mStandbyTimeInNsecs = milliseconds(int_val); 224 ALOGI("Using %u mSec as standby time.", int_val); 225 } else { 226 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 227 ALOGI("Using default %u mSec as standby time.", 228 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 229 } 230 } 231 232 mPatchPanel = new PatchPanel(this); 233 234 mMode = AUDIO_MODE_NORMAL; 235} 236 237AudioFlinger::~AudioFlinger() 238{ 239 while (!mRecordThreads.isEmpty()) { 240 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 241 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 242 } 243 while (!mPlaybackThreads.isEmpty()) { 244 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 245 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 246 } 247 248 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 249 // no mHardwareLock needed, as there are no other references to this 250 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 251 delete mAudioHwDevs.valueAt(i); 252 } 253 254 // Tell media.log service about any old writers that still need to be unregistered 255 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 256 if (binder != 0) { 257 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 258 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 259 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 260 mUnregisteredWriters.pop(); 261 mediaLogService->unregisterWriter(iMemory); 262 } 263 } 264 265} 266 267static const char * const audio_interfaces[] = { 268 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 269 AUDIO_HARDWARE_MODULE_ID_A2DP, 270 AUDIO_HARDWARE_MODULE_ID_USB, 271}; 272#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 273 274AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 275 audio_module_handle_t module, 276 audio_devices_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 // then try to find a module supporting the requested device. 286 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 287 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 288 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 289 if ((dev->get_supported_devices != NULL) && 290 (dev->get_supported_devices(dev) & devices) == devices) 291 return audioHwDevice; 292 } 293 } else { 294 // check a match for the requested module handle 295 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 296 if (audioHwDevice != NULL) { 297 return audioHwDevice; 298 } 299 } 300 301 return NULL; 302} 303 304void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Notification Clients:\n"); 320 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 321 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 322 result.append(buffer); 323 } 324 325 result.append("Global session refs:\n"); 326 result.append(" session pid count\n"); 327 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 328 AudioSessionRef *r = mAudioSessionRefs[i]; 329 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 330 result.append(buffer); 331 } 332 write(fd, result.string(), result.size()); 333} 334 335 336void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 337{ 338 const size_t SIZE = 256; 339 char buffer[SIZE]; 340 String8 result; 341 hardware_call_state hardwareStatus = mHardwareStatus; 342 343 snprintf(buffer, SIZE, "Hardware status: %d\n" 344 "Standby Time mSec: %u\n", 345 hardwareStatus, 346 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 347 result.append(buffer); 348 write(fd, result.string(), result.size()); 349} 350 351void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 352{ 353 const size_t SIZE = 256; 354 char buffer[SIZE]; 355 String8 result; 356 snprintf(buffer, SIZE, "Permission Denial: " 357 "can't dump AudioFlinger from pid=%d, uid=%d\n", 358 IPCThreadState::self()->getCallingPid(), 359 IPCThreadState::self()->getCallingUid()); 360 result.append(buffer); 361 write(fd, result.string(), result.size()); 362} 363 364bool AudioFlinger::dumpTryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = dumpTryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = dumpTryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 bool clientLocked = dumpTryLock(mClientLock); 400 if (!clientLocked) { 401 String8 result(kClientLockedString); 402 write(fd, result.string(), result.size()); 403 } 404 dumpClients(fd, args); 405 if (clientLocked) { 406 mClientLock.unlock(); 407 } 408 409 dumpInternals(fd, args); 410 411 // dump playback threads 412 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 413 mPlaybackThreads.valueAt(i)->dump(fd, args); 414 } 415 416 // dump record threads 417 for (size_t i = 0; i < mRecordThreads.size(); i++) { 418 mRecordThreads.valueAt(i)->dump(fd, args); 419 } 420 421 // dump orphan effect chains 422 if (mOrphanEffectChains.size() != 0) { 423 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 424 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 425 mOrphanEffectChains.valueAt(i)->dump(fd, args); 426 } 427 } 428 // dump all hardware devs 429 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 430 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 431 dev->dump(dev, fd); 432 } 433 434#ifdef TEE_SINK 435 // dump the serially shared record tee sink 436 if (mRecordTeeSource != 0) { 437 dumpTee(fd, mRecordTeeSource); 438 } 439#endif 440 441 if (locked) { 442 mLock.unlock(); 443 } 444 445 // append a copy of media.log here by forwarding fd to it, but don't attempt 446 // to lookup the service if it's not running, as it will block for a second 447 if (mLogMemoryDealer != 0) { 448 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 449 if (binder != 0) { 450 dprintf(fd, "\nmedia.log:\n"); 451 Vector<String16> args; 452 binder->dump(fd, args); 453 } 454 } 455 } 456 return NO_ERROR; 457} 458 459sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 460{ 461 Mutex::Autolock _cl(mClientLock); 462 // If pid is already in the mClients wp<> map, then use that entry 463 // (for which promote() is always != 0), otherwise create a new entry and Client. 464 sp<Client> client = mClients.valueFor(pid).promote(); 465 if (client == 0) { 466 client = new Client(this, pid); 467 mClients.add(pid, client); 468 } 469 470 return client; 471} 472 473sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 474{ 475 // If there is no memory allocated for logs, return a dummy writer that does nothing 476 if (mLogMemoryDealer == 0) { 477 return new NBLog::Writer(); 478 } 479 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 480 // Similarly if we can't contact the media.log service, also return a dummy writer 481 if (binder == 0) { 482 return new NBLog::Writer(); 483 } 484 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 485 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 486 // If allocation fails, consult the vector of previously unregistered writers 487 // and garbage-collect one or more them until an allocation succeeds 488 if (shared == 0) { 489 Mutex::Autolock _l(mUnregisteredWritersLock); 490 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 491 { 492 // Pick the oldest stale writer to garbage-collect 493 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 494 mUnregisteredWriters.removeAt(0); 495 mediaLogService->unregisterWriter(iMemory); 496 // Now the media.log remote reference to IMemory is gone. When our last local 497 // reference to IMemory also drops to zero at end of this block, 498 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 499 } 500 // Re-attempt the allocation 501 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 502 if (shared != 0) { 503 goto success; 504 } 505 } 506 // Even after garbage-collecting all old writers, there is still not enough memory, 507 // so return a dummy writer 508 return new NBLog::Writer(); 509 } 510success: 511 mediaLogService->registerWriter(shared, size, name); 512 return new NBLog::Writer(size, shared); 513} 514 515void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 516{ 517 if (writer == 0) { 518 return; 519 } 520 sp<IMemory> iMemory(writer->getIMemory()); 521 if (iMemory == 0) { 522 return; 523 } 524 // Rather than removing the writer immediately, append it to a queue of old writers to 525 // be garbage-collected later. This allows us to continue to view old logs for a while. 526 Mutex::Autolock _l(mUnregisteredWritersLock); 527 mUnregisteredWriters.push(writer); 528} 529 530// IAudioFlinger interface 531 532 533sp<IAudioTrack> AudioFlinger::createTrack( 534 audio_stream_type_t streamType, 535 uint32_t sampleRate, 536 audio_format_t format, 537 audio_channel_mask_t channelMask, 538 size_t *frameCount, 539 IAudioFlinger::track_flags_t *flags, 540 const sp<IMemory>& sharedBuffer, 541 audio_io_handle_t output, 542 pid_t tid, 543 int *sessionId, 544 int clientUid, 545 status_t *status) 546{ 547 sp<PlaybackThread::Track> track; 548 sp<TrackHandle> trackHandle; 549 sp<Client> client; 550 status_t lStatus; 551 int lSessionId; 552 553 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 554 // but if someone uses binder directly they could bypass that and cause us to crash 555 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 556 ALOGE("createTrack() invalid stream type %d", streamType); 557 lStatus = BAD_VALUE; 558 goto Exit; 559 } 560 561 // further sample rate checks are performed by createTrack_l() depending on the thread type 562 if (sampleRate == 0) { 563 ALOGE("createTrack() invalid sample rate %u", sampleRate); 564 lStatus = BAD_VALUE; 565 goto Exit; 566 } 567 568 // further channel mask checks are performed by createTrack_l() depending on the thread type 569 if (!audio_is_output_channel(channelMask)) { 570 ALOGE("createTrack() invalid channel mask %#x", channelMask); 571 lStatus = BAD_VALUE; 572 goto Exit; 573 } 574 575 // further format checks are performed by createTrack_l() depending on the thread type 576 if (!audio_is_valid_format(format)) { 577 ALOGE("createTrack() invalid format %#x", format); 578 lStatus = BAD_VALUE; 579 goto Exit; 580 } 581 582 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 583 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 584 lStatus = BAD_VALUE; 585 goto Exit; 586 } 587 588 { 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGE("no playback thread found for output handle %d", output); 593 lStatus = BAD_VALUE; 594 goto Exit; 595 } 596 597 pid_t pid = IPCThreadState::self()->getCallingPid(); 598 client = registerPid(pid); 599 600 PlaybackThread *effectThread = NULL; 601 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 602 lSessionId = *sessionId; 603 // check if an effect chain with the same session ID is present on another 604 // output thread and move it here. 605 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 606 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 607 if (mPlaybackThreads.keyAt(i) != output) { 608 uint32_t sessions = t->hasAudioSession(lSessionId); 609 if (sessions & PlaybackThread::EFFECT_SESSION) { 610 effectThread = t.get(); 611 break; 612 } 613 } 614 } 615 } else { 616 // if no audio session id is provided, create one here 617 lSessionId = nextUniqueId(); 618 if (sessionId != NULL) { 619 *sessionId = lSessionId; 620 } 621 } 622 ALOGV("createTrack() lSessionId: %d", lSessionId); 623 624 track = thread->createTrack_l(client, streamType, sampleRate, format, 625 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 626 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 627 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 628 629 // move effect chain to this output thread if an effect on same session was waiting 630 // for a track to be created 631 if (lStatus == NO_ERROR && effectThread != NULL) { 632 // no risk of deadlock because AudioFlinger::mLock is held 633 Mutex::Autolock _dl(thread->mLock); 634 Mutex::Autolock _sl(effectThread->mLock); 635 moveEffectChain_l(lSessionId, effectThread, thread, true); 636 } 637 638 // Look for sync events awaiting for a session to be used. 639 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 640 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 641 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 642 if (lStatus == NO_ERROR) { 643 (void) track->setSyncEvent(mPendingSyncEvents[i]); 644 } else { 645 mPendingSyncEvents[i]->cancel(); 646 } 647 mPendingSyncEvents.removeAt(i); 648 i--; 649 } 650 } 651 } 652 653 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 654 } 655 656 if (lStatus != NO_ERROR) { 657 // remove local strong reference to Client before deleting the Track so that the 658 // Client destructor is called by the TrackBase destructor with mClientLock held 659 // Don't hold mClientLock when releasing the reference on the track as the 660 // destructor will acquire it. 661 { 662 Mutex::Autolock _cl(mClientLock); 663 client.clear(); 664 } 665 track.clear(); 666 goto Exit; 667 } 668 669 // return handle to client 670 trackHandle = new TrackHandle(track); 671 672Exit: 673 *status = lStatus; 674 return trackHandle; 675} 676 677uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 678{ 679 Mutex::Autolock _l(mLock); 680 PlaybackThread *thread = checkPlaybackThread_l(output); 681 if (thread == NULL) { 682 ALOGW("sampleRate() unknown thread %d", output); 683 return 0; 684 } 685 return thread->sampleRate(); 686} 687 688audio_format_t AudioFlinger::format(audio_io_handle_t output) const 689{ 690 Mutex::Autolock _l(mLock); 691 PlaybackThread *thread = checkPlaybackThread_l(output); 692 if (thread == NULL) { 693 ALOGW("format() unknown thread %d", output); 694 return AUDIO_FORMAT_INVALID; 695 } 696 return thread->format(); 697} 698 699size_t AudioFlinger::frameCount(audio_io_handle_t output) const 700{ 701 Mutex::Autolock _l(mLock); 702 PlaybackThread *thread = checkPlaybackThread_l(output); 703 if (thread == NULL) { 704 ALOGW("frameCount() unknown thread %d", output); 705 return 0; 706 } 707 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 708 // should examine all callers and fix them to handle smaller counts 709 return thread->frameCount(); 710} 711 712uint32_t AudioFlinger::latency(audio_io_handle_t output) const 713{ 714 Mutex::Autolock _l(mLock); 715 PlaybackThread *thread = checkPlaybackThread_l(output); 716 if (thread == NULL) { 717 ALOGW("latency(): no playback thread found for output handle %d", output); 718 return 0; 719 } 720 return thread->latency(); 721} 722 723status_t AudioFlinger::setMasterVolume(float value) 724{ 725 status_t ret = initCheck(); 726 if (ret != NO_ERROR) { 727 return ret; 728 } 729 730 // check calling permissions 731 if (!settingsAllowed()) { 732 return PERMISSION_DENIED; 733 } 734 735 Mutex::Autolock _l(mLock); 736 mMasterVolume = value; 737 738 // Set master volume in the HALs which support it. 739 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 740 AutoMutex lock(mHardwareLock); 741 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 742 743 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 744 if (dev->canSetMasterVolume()) { 745 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 746 } 747 mHardwareStatus = AUDIO_HW_IDLE; 748 } 749 750 // Now set the master volume in each playback thread. Playback threads 751 // assigned to HALs which do not have master volume support will apply 752 // master volume during the mix operation. Threads with HALs which do 753 // support master volume will simply ignore the setting. 754 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 755 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 756 757 return NO_ERROR; 758} 759 760status_t AudioFlinger::setMode(audio_mode_t mode) 761{ 762 status_t ret = initCheck(); 763 if (ret != NO_ERROR) { 764 return ret; 765 } 766 767 // check calling permissions 768 if (!settingsAllowed()) { 769 return PERMISSION_DENIED; 770 } 771 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 772 ALOGW("Illegal value: setMode(%d)", mode); 773 return BAD_VALUE; 774 } 775 776 { // scope for the lock 777 AutoMutex lock(mHardwareLock); 778 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 779 mHardwareStatus = AUDIO_HW_SET_MODE; 780 ret = dev->set_mode(dev, mode); 781 mHardwareStatus = AUDIO_HW_IDLE; 782 } 783 784 if (NO_ERROR == ret) { 785 Mutex::Autolock _l(mLock); 786 mMode = mode; 787 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setMode(mode); 789 } 790 791 return ret; 792} 793 794status_t AudioFlinger::setMicMute(bool state) 795{ 796 status_t ret = initCheck(); 797 if (ret != NO_ERROR) { 798 return ret; 799 } 800 801 // check calling permissions 802 if (!settingsAllowed()) { 803 return PERMISSION_DENIED; 804 } 805 806 AutoMutex lock(mHardwareLock); 807 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 808 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 809 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 810 status_t result = dev->set_mic_mute(dev, state); 811 if (result != NO_ERROR) { 812 ret = result; 813 } 814 } 815 mHardwareStatus = AUDIO_HW_IDLE; 816 return ret; 817} 818 819bool AudioFlinger::getMicMute() const 820{ 821 status_t ret = initCheck(); 822 if (ret != NO_ERROR) { 823 return false; 824 } 825 826 bool state = AUDIO_MODE_INVALID; 827 AutoMutex lock(mHardwareLock); 828 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 829 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 830 dev->get_mic_mute(dev, &state); 831 mHardwareStatus = AUDIO_HW_IDLE; 832 return state; 833} 834 835status_t AudioFlinger::setMasterMute(bool muted) 836{ 837 status_t ret = initCheck(); 838 if (ret != NO_ERROR) { 839 return ret; 840 } 841 842 // check calling permissions 843 if (!settingsAllowed()) { 844 return PERMISSION_DENIED; 845 } 846 847 Mutex::Autolock _l(mLock); 848 mMasterMute = muted; 849 850 // Set master mute in the HALs which support it. 851 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 852 AutoMutex lock(mHardwareLock); 853 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 854 855 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 856 if (dev->canSetMasterMute()) { 857 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 858 } 859 mHardwareStatus = AUDIO_HW_IDLE; 860 } 861 862 // Now set the master mute in each playback thread. Playback threads 863 // assigned to HALs which do not have master mute support will apply master 864 // mute during the mix operation. Threads with HALs which do support master 865 // mute will simply ignore the setting. 866 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 867 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 868 869 return NO_ERROR; 870} 871 872float AudioFlinger::masterVolume() const 873{ 874 Mutex::Autolock _l(mLock); 875 return masterVolume_l(); 876} 877 878bool AudioFlinger::masterMute() const 879{ 880 Mutex::Autolock _l(mLock); 881 return masterMute_l(); 882} 883 884float AudioFlinger::masterVolume_l() const 885{ 886 return mMasterVolume; 887} 888 889bool AudioFlinger::masterMute_l() const 890{ 891 return mMasterMute; 892} 893 894status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 895{ 896 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 897 ALOGW("setStreamVolume() invalid stream %d", stream); 898 return BAD_VALUE; 899 } 900 pid_t caller = IPCThreadState::self()->getCallingPid(); 901 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 902 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 903 return PERMISSION_DENIED; 904 } 905 906 return NO_ERROR; 907} 908 909status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 910 audio_io_handle_t output) 911{ 912 // check calling permissions 913 if (!settingsAllowed()) { 914 return PERMISSION_DENIED; 915 } 916 917 status_t status = checkStreamType(stream); 918 if (status != NO_ERROR) { 919 return status; 920 } 921 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 922 923 AutoMutex lock(mLock); 924 PlaybackThread *thread = NULL; 925 if (output != AUDIO_IO_HANDLE_NONE) { 926 thread = checkPlaybackThread_l(output); 927 if (thread == NULL) { 928 return BAD_VALUE; 929 } 930 } 931 932 mStreamTypes[stream].volume = value; 933 934 if (thread == NULL) { 935 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 936 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 937 } 938 } else { 939 thread->setStreamVolume(stream, value); 940 } 941 942 return NO_ERROR; 943} 944 945status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 946{ 947 // check calling permissions 948 if (!settingsAllowed()) { 949 return PERMISSION_DENIED; 950 } 951 952 status_t status = checkStreamType(stream); 953 if (status != NO_ERROR) { 954 return status; 955 } 956 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 957 958 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 959 ALOGE("setStreamMute() invalid stream %d", stream); 960 return BAD_VALUE; 961 } 962 963 AutoMutex lock(mLock); 964 mStreamTypes[stream].mute = muted; 965 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 966 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 967 968 return NO_ERROR; 969} 970 971float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 972{ 973 status_t status = checkStreamType(stream); 974 if (status != NO_ERROR) { 975 return 0.0f; 976 } 977 978 AutoMutex lock(mLock); 979 float volume; 980 if (output != AUDIO_IO_HANDLE_NONE) { 981 PlaybackThread *thread = checkPlaybackThread_l(output); 982 if (thread == NULL) { 983 return 0.0f; 984 } 985 volume = thread->streamVolume(stream); 986 } else { 987 volume = streamVolume_l(stream); 988 } 989 990 return volume; 991} 992 993bool AudioFlinger::streamMute(audio_stream_type_t stream) const 994{ 995 status_t status = checkStreamType(stream); 996 if (status != NO_ERROR) { 997 return true; 998 } 999 1000 AutoMutex lock(mLock); 1001 return streamMute_l(stream); 1002} 1003 1004status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1005{ 1006 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1007 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1008 1009 // check calling permissions 1010 if (!settingsAllowed()) { 1011 return PERMISSION_DENIED; 1012 } 1013 1014 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1015 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1016 Mutex::Autolock _l(mLock); 1017 status_t final_result = NO_ERROR; 1018 { 1019 AutoMutex lock(mHardwareLock); 1020 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1021 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1022 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1023 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1024 final_result = result ?: final_result; 1025 } 1026 mHardwareStatus = AUDIO_HW_IDLE; 1027 } 1028 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1029 AudioParameter param = AudioParameter(keyValuePairs); 1030 String8 value; 1031 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1032 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1033 if (mBtNrecIsOff != btNrecIsOff) { 1034 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1035 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1036 audio_devices_t device = thread->inDevice(); 1037 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1038 // collect all of the thread's session IDs 1039 KeyedVector<int, bool> ids = thread->sessionIds(); 1040 // suspend effects associated with those session IDs 1041 for (size_t j = 0; j < ids.size(); ++j) { 1042 int sessionId = ids.keyAt(j); 1043 thread->setEffectSuspended(FX_IID_AEC, 1044 suspend, 1045 sessionId); 1046 thread->setEffectSuspended(FX_IID_NS, 1047 suspend, 1048 sessionId); 1049 } 1050 } 1051 mBtNrecIsOff = btNrecIsOff; 1052 } 1053 } 1054 String8 screenState; 1055 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1056 bool isOff = screenState == "off"; 1057 if (isOff != (AudioFlinger::mScreenState & 1)) { 1058 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1059 } 1060 } 1061 return final_result; 1062 } 1063 1064 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1065 // and the thread is exited once the lock is released 1066 sp<ThreadBase> thread; 1067 { 1068 Mutex::Autolock _l(mLock); 1069 thread = checkPlaybackThread_l(ioHandle); 1070 if (thread == 0) { 1071 thread = checkRecordThread_l(ioHandle); 1072 } else if (thread == primaryPlaybackThread_l()) { 1073 // indicate output device change to all input threads for pre processing 1074 AudioParameter param = AudioParameter(keyValuePairs); 1075 int value; 1076 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1077 (value != 0)) { 1078 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1079 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1080 } 1081 } 1082 } 1083 } 1084 if (thread != 0) { 1085 return thread->setParameters(keyValuePairs); 1086 } 1087 return BAD_VALUE; 1088} 1089 1090String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1091{ 1092 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1093 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1094 1095 Mutex::Autolock _l(mLock); 1096 1097 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1098 String8 out_s8; 1099 1100 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1101 char *s; 1102 { 1103 AutoMutex lock(mHardwareLock); 1104 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1105 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1106 s = dev->get_parameters(dev, keys.string()); 1107 mHardwareStatus = AUDIO_HW_IDLE; 1108 } 1109 out_s8 += String8(s ? s : ""); 1110 free(s); 1111 } 1112 return out_s8; 1113 } 1114 1115 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1116 if (playbackThread != NULL) { 1117 return playbackThread->getParameters(keys); 1118 } 1119 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1120 if (recordThread != NULL) { 1121 return recordThread->getParameters(keys); 1122 } 1123 return String8(""); 1124} 1125 1126size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1127 audio_channel_mask_t channelMask) const 1128{ 1129 status_t ret = initCheck(); 1130 if (ret != NO_ERROR) { 1131 return 0; 1132 } 1133 1134 AutoMutex lock(mHardwareLock); 1135 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1136 audio_config_t config; 1137 memset(&config, 0, sizeof(config)); 1138 config.sample_rate = sampleRate; 1139 config.channel_mask = channelMask; 1140 config.format = format; 1141 1142 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1143 size_t size = dev->get_input_buffer_size(dev, &config); 1144 mHardwareStatus = AUDIO_HW_IDLE; 1145 return size; 1146} 1147 1148uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1149{ 1150 Mutex::Autolock _l(mLock); 1151 1152 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1153 if (recordThread != NULL) { 1154 return recordThread->getInputFramesLost(); 1155 } 1156 return 0; 1157} 1158 1159status_t AudioFlinger::setVoiceVolume(float value) 1160{ 1161 status_t ret = initCheck(); 1162 if (ret != NO_ERROR) { 1163 return ret; 1164 } 1165 1166 // check calling permissions 1167 if (!settingsAllowed()) { 1168 return PERMISSION_DENIED; 1169 } 1170 1171 AutoMutex lock(mHardwareLock); 1172 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1173 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1174 ret = dev->set_voice_volume(dev, value); 1175 mHardwareStatus = AUDIO_HW_IDLE; 1176 1177 return ret; 1178} 1179 1180status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1181 audio_io_handle_t output) const 1182{ 1183 status_t status; 1184 1185 Mutex::Autolock _l(mLock); 1186 1187 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1188 if (playbackThread != NULL) { 1189 return playbackThread->getRenderPosition(halFrames, dspFrames); 1190 } 1191 1192 return BAD_VALUE; 1193} 1194 1195void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1196{ 1197 Mutex::Autolock _l(mLock); 1198 if (client == 0) { 1199 return; 1200 } 1201 bool clientAdded = false; 1202 { 1203 Mutex::Autolock _cl(mClientLock); 1204 1205 pid_t pid = IPCThreadState::self()->getCallingPid(); 1206 if (mNotificationClients.indexOfKey(pid) < 0) { 1207 sp<NotificationClient> notificationClient = new NotificationClient(this, 1208 client, 1209 pid); 1210 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1211 1212 mNotificationClients.add(pid, notificationClient); 1213 1214 sp<IBinder> binder = IInterface::asBinder(client); 1215 binder->linkToDeath(notificationClient); 1216 clientAdded = true; 1217 } 1218 } 1219 1220 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1221 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1222 if (clientAdded) { 1223 // the config change is always sent from playback or record threads to avoid deadlock 1224 // with AudioSystem::gLock 1225 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1226 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1227 } 1228 1229 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1230 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1231 } 1232 } 1233} 1234 1235void AudioFlinger::removeNotificationClient(pid_t pid) 1236{ 1237 Mutex::Autolock _l(mLock); 1238 { 1239 Mutex::Autolock _cl(mClientLock); 1240 mNotificationClients.removeItem(pid); 1241 } 1242 1243 ALOGV("%d died, releasing its sessions", pid); 1244 size_t num = mAudioSessionRefs.size(); 1245 bool removed = false; 1246 for (size_t i = 0; i< num; ) { 1247 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1248 ALOGV(" pid %d @ %d", ref->mPid, i); 1249 if (ref->mPid == pid) { 1250 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1251 mAudioSessionRefs.removeAt(i); 1252 delete ref; 1253 removed = true; 1254 num--; 1255 } else { 1256 i++; 1257 } 1258 } 1259 if (removed) { 1260 purgeStaleEffects_l(); 1261 } 1262} 1263 1264void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1265{ 1266 Mutex::Autolock _l(mClientLock); 1267 size_t size = mNotificationClients.size(); 1268 for (size_t i = 0; i < size; i++) { 1269 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1270 ioHandle, 1271 param2); 1272 } 1273} 1274 1275// removeClient_l() must be called with AudioFlinger::mClientLock held 1276void AudioFlinger::removeClient_l(pid_t pid) 1277{ 1278 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1279 IPCThreadState::self()->getCallingPid()); 1280 mClients.removeItem(pid); 1281} 1282 1283// getEffectThread_l() must be called with AudioFlinger::mLock held 1284sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1285{ 1286 sp<PlaybackThread> thread; 1287 1288 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1289 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1290 ALOG_ASSERT(thread == 0); 1291 thread = mPlaybackThreads.valueAt(i); 1292 } 1293 } 1294 1295 return thread; 1296} 1297 1298 1299 1300// ---------------------------------------------------------------------------- 1301 1302AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1303 : RefBase(), 1304 mAudioFlinger(audioFlinger), 1305 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1306 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1307 mPid(pid), 1308 mTimedTrackCount(0) 1309{ 1310 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1311} 1312 1313// Client destructor must be called with AudioFlinger::mClientLock held 1314AudioFlinger::Client::~Client() 1315{ 1316 mAudioFlinger->removeClient_l(mPid); 1317} 1318 1319sp<MemoryDealer> AudioFlinger::Client::heap() const 1320{ 1321 return mMemoryDealer; 1322} 1323 1324// Reserve one of the limited slots for a timed audio track associated 1325// with this client 1326bool AudioFlinger::Client::reserveTimedTrack() 1327{ 1328 const int kMaxTimedTracksPerClient = 4; 1329 1330 Mutex::Autolock _l(mTimedTrackLock); 1331 1332 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1333 ALOGW("can not create timed track - pid %d has exceeded the limit", 1334 mPid); 1335 return false; 1336 } 1337 1338 mTimedTrackCount++; 1339 return true; 1340} 1341 1342// Release a slot for a timed audio track 1343void AudioFlinger::Client::releaseTimedTrack() 1344{ 1345 Mutex::Autolock _l(mTimedTrackLock); 1346 mTimedTrackCount--; 1347} 1348 1349// ---------------------------------------------------------------------------- 1350 1351AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1352 const sp<IAudioFlingerClient>& client, 1353 pid_t pid) 1354 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1355{ 1356} 1357 1358AudioFlinger::NotificationClient::~NotificationClient() 1359{ 1360} 1361 1362void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1363{ 1364 sp<NotificationClient> keep(this); 1365 mAudioFlinger->removeNotificationClient(mPid); 1366} 1367 1368 1369// ---------------------------------------------------------------------------- 1370 1371static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1372 return audio_is_remote_submix_device(inDevice); 1373} 1374 1375sp<IAudioRecord> AudioFlinger::openRecord( 1376 audio_io_handle_t input, 1377 uint32_t sampleRate, 1378 audio_format_t format, 1379 audio_channel_mask_t channelMask, 1380 size_t *frameCount, 1381 IAudioFlinger::track_flags_t *flags, 1382 pid_t tid, 1383 int *sessionId, 1384 size_t *notificationFrames, 1385 sp<IMemory>& cblk, 1386 sp<IMemory>& buffers, 1387 status_t *status) 1388{ 1389 sp<RecordThread::RecordTrack> recordTrack; 1390 sp<RecordHandle> recordHandle; 1391 sp<Client> client; 1392 status_t lStatus; 1393 int lSessionId; 1394 1395 cblk.clear(); 1396 buffers.clear(); 1397 1398 // check calling permissions 1399 if (!recordingAllowed()) { 1400 ALOGE("openRecord() permission denied: recording not allowed"); 1401 lStatus = PERMISSION_DENIED; 1402 goto Exit; 1403 } 1404 1405 // further sample rate checks are performed by createRecordTrack_l() 1406 if (sampleRate == 0) { 1407 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1408 lStatus = BAD_VALUE; 1409 goto Exit; 1410 } 1411 1412 // we don't yet support anything other than 16-bit PCM 1413 if (!(audio_is_valid_format(format) && 1414 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1415 ALOGE("openRecord() invalid format %#x", format); 1416 lStatus = BAD_VALUE; 1417 goto Exit; 1418 } 1419 1420 // further channel mask checks are performed by createRecordTrack_l() 1421 if (!audio_is_input_channel(channelMask)) { 1422 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1423 lStatus = BAD_VALUE; 1424 goto Exit; 1425 } 1426 1427 { 1428 Mutex::Autolock _l(mLock); 1429 RecordThread *thread = checkRecordThread_l(input); 1430 if (thread == NULL) { 1431 ALOGE("openRecord() checkRecordThread_l failed"); 1432 lStatus = BAD_VALUE; 1433 goto Exit; 1434 } 1435 1436 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1437 && !captureAudioOutputAllowed()) { 1438 ALOGE("openRecord() permission denied: capture not allowed"); 1439 lStatus = PERMISSION_DENIED; 1440 goto Exit; 1441 } 1442 1443 pid_t pid = IPCThreadState::self()->getCallingPid(); 1444 client = registerPid(pid); 1445 1446 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1447 lSessionId = *sessionId; 1448 } else { 1449 // if no audio session id is provided, create one here 1450 lSessionId = nextUniqueId(); 1451 if (sessionId != NULL) { 1452 *sessionId = lSessionId; 1453 } 1454 } 1455 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1456 1457 // TODO: the uid should be passed in as a parameter to openRecord 1458 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1459 frameCount, lSessionId, notificationFrames, 1460 IPCThreadState::self()->getCallingUid(), 1461 flags, tid, &lStatus); 1462 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1463 1464 if (lStatus == NO_ERROR) { 1465 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1466 // session and move it to this thread. 1467 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1468 if (chain != 0) { 1469 Mutex::Autolock _l(thread->mLock); 1470 thread->addEffectChain_l(chain); 1471 } 1472 } 1473 } 1474 1475 if (lStatus != NO_ERROR) { 1476 // remove local strong reference to Client before deleting the RecordTrack so that the 1477 // Client destructor is called by the TrackBase destructor with mClientLock held 1478 // Don't hold mClientLock when releasing the reference on the track as the 1479 // destructor will acquire it. 1480 { 1481 Mutex::Autolock _cl(mClientLock); 1482 client.clear(); 1483 } 1484 recordTrack.clear(); 1485 goto Exit; 1486 } 1487 1488 cblk = recordTrack->getCblk(); 1489 buffers = recordTrack->getBuffers(); 1490 1491 // return handle to client 1492 recordHandle = new RecordHandle(recordTrack); 1493 1494Exit: 1495 *status = lStatus; 1496 return recordHandle; 1497} 1498 1499 1500 1501// ---------------------------------------------------------------------------- 1502 1503audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1504{ 1505 if (name == NULL) { 1506 return 0; 1507 } 1508 if (!settingsAllowed()) { 1509 return 0; 1510 } 1511 Mutex::Autolock _l(mLock); 1512 return loadHwModule_l(name); 1513} 1514 1515// loadHwModule_l() must be called with AudioFlinger::mLock held 1516audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1517{ 1518 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1519 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1520 ALOGW("loadHwModule() module %s already loaded", name); 1521 return mAudioHwDevs.keyAt(i); 1522 } 1523 } 1524 1525 audio_hw_device_t *dev; 1526 1527 int rc = load_audio_interface(name, &dev); 1528 if (rc) { 1529 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1530 return 0; 1531 } 1532 1533 mHardwareStatus = AUDIO_HW_INIT; 1534 rc = dev->init_check(dev); 1535 mHardwareStatus = AUDIO_HW_IDLE; 1536 if (rc) { 1537 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1538 return 0; 1539 } 1540 1541 // Check and cache this HAL's level of support for master mute and master 1542 // volume. If this is the first HAL opened, and it supports the get 1543 // methods, use the initial values provided by the HAL as the current 1544 // master mute and volume settings. 1545 1546 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1547 { // scope for auto-lock pattern 1548 AutoMutex lock(mHardwareLock); 1549 1550 if (0 == mAudioHwDevs.size()) { 1551 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1552 if (NULL != dev->get_master_volume) { 1553 float mv; 1554 if (OK == dev->get_master_volume(dev, &mv)) { 1555 mMasterVolume = mv; 1556 } 1557 } 1558 1559 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1560 if (NULL != dev->get_master_mute) { 1561 bool mm; 1562 if (OK == dev->get_master_mute(dev, &mm)) { 1563 mMasterMute = mm; 1564 } 1565 } 1566 } 1567 1568 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1569 if ((NULL != dev->set_master_volume) && 1570 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1571 flags = static_cast<AudioHwDevice::Flags>(flags | 1572 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1573 } 1574 1575 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1576 if ((NULL != dev->set_master_mute) && 1577 (OK == dev->set_master_mute(dev, mMasterMute))) { 1578 flags = static_cast<AudioHwDevice::Flags>(flags | 1579 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1580 } 1581 1582 mHardwareStatus = AUDIO_HW_IDLE; 1583 } 1584 1585 audio_module_handle_t handle = nextUniqueId(); 1586 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1587 1588 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1589 name, dev->common.module->name, dev->common.module->id, handle); 1590 1591 return handle; 1592 1593} 1594 1595// ---------------------------------------------------------------------------- 1596 1597uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1598{ 1599 Mutex::Autolock _l(mLock); 1600 PlaybackThread *thread = primaryPlaybackThread_l(); 1601 return thread != NULL ? thread->sampleRate() : 0; 1602} 1603 1604size_t AudioFlinger::getPrimaryOutputFrameCount() 1605{ 1606 Mutex::Autolock _l(mLock); 1607 PlaybackThread *thread = primaryPlaybackThread_l(); 1608 return thread != NULL ? thread->frameCountHAL() : 0; 1609} 1610 1611// ---------------------------------------------------------------------------- 1612 1613status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1614{ 1615 uid_t uid = IPCThreadState::self()->getCallingUid(); 1616 if (uid != AID_SYSTEM) { 1617 return PERMISSION_DENIED; 1618 } 1619 Mutex::Autolock _l(mLock); 1620 if (mIsDeviceTypeKnown) { 1621 return INVALID_OPERATION; 1622 } 1623 mIsLowRamDevice = isLowRamDevice; 1624 mIsDeviceTypeKnown = true; 1625 return NO_ERROR; 1626} 1627 1628audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1629{ 1630 Mutex::Autolock _l(mLock); 1631 1632 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1633 if (index >= 0) { 1634 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1635 mHwAvSyncIds.valueAt(index), sessionId); 1636 return mHwAvSyncIds.valueAt(index); 1637 } 1638 1639 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1640 if (dev == NULL) { 1641 return AUDIO_HW_SYNC_INVALID; 1642 } 1643 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1644 AudioParameter param = AudioParameter(String8(reply)); 1645 free(reply); 1646 1647 int value; 1648 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1649 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1650 return AUDIO_HW_SYNC_INVALID; 1651 } 1652 1653 // allow only one session for a given HW A/V sync ID. 1654 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1655 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1656 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1657 value, mHwAvSyncIds.keyAt(i)); 1658 mHwAvSyncIds.removeItemsAt(i); 1659 break; 1660 } 1661 } 1662 1663 mHwAvSyncIds.add(sessionId, value); 1664 1665 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1666 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1667 uint32_t sessions = thread->hasAudioSession(sessionId); 1668 if (sessions & PlaybackThread::TRACK_SESSION) { 1669 AudioParameter param = AudioParameter(); 1670 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1671 thread->setParameters(param.toString()); 1672 break; 1673 } 1674 } 1675 1676 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1677 return (audio_hw_sync_t)value; 1678} 1679 1680// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1681void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1682{ 1683 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1684 if (index >= 0) { 1685 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1686 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1687 AudioParameter param = AudioParameter(); 1688 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1689 thread->setParameters(param.toString()); 1690 } 1691} 1692 1693 1694// ---------------------------------------------------------------------------- 1695 1696 1697sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1698 audio_io_handle_t *output, 1699 audio_config_t *config, 1700 audio_devices_t devices, 1701 const String8& address, 1702 audio_output_flags_t flags) 1703{ 1704 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1705 if (outHwDev == NULL) { 1706 return 0; 1707 } 1708 1709 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1710 if (*output == AUDIO_IO_HANDLE_NONE) { 1711 *output = nextUniqueId(); 1712 } 1713 1714 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1715 1716 audio_stream_out_t *outStream = NULL; 1717 1718 // FOR TESTING ONLY: 1719 // This if statement allows overriding the audio policy settings 1720 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1721 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1722 // Check only for Normal Mixing mode 1723 if (kEnableExtendedPrecision) { 1724 // Specify format (uncomment one below to choose) 1725 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1726 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1727 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1728 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1729 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1730 } 1731 if (kEnableExtendedChannels) { 1732 // Specify channel mask (uncomment one below to choose) 1733 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1734 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1735 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1736 } 1737 } 1738 1739 status_t status = hwDevHal->open_output_stream(hwDevHal, 1740 *output, 1741 devices, 1742 flags, 1743 config, 1744 &outStream, 1745 address.string()); 1746 1747 mHardwareStatus = AUDIO_HW_IDLE; 1748 ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, " 1749 "channelMask %#x, status %d", 1750 outStream, 1751 config->sample_rate, 1752 config->format, 1753 config->channel_mask, 1754 status); 1755 1756 if (status == NO_ERROR && outStream != NULL) { 1757 AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags); 1758 1759 PlaybackThread *thread; 1760 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1761 thread = new OffloadThread(this, outputStream, *output, devices); 1762 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1763 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1764 || !isValidPcmSinkFormat(config->format) 1765 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1766 thread = new DirectOutputThread(this, outputStream, *output, devices); 1767 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1768 } else { 1769 thread = new MixerThread(this, outputStream, *output, devices); 1770 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1771 } 1772 mPlaybackThreads.add(*output, thread); 1773 return thread; 1774 } 1775 1776 return 0; 1777} 1778 1779status_t AudioFlinger::openOutput(audio_module_handle_t module, 1780 audio_io_handle_t *output, 1781 audio_config_t *config, 1782 audio_devices_t *devices, 1783 const String8& address, 1784 uint32_t *latencyMs, 1785 audio_output_flags_t flags) 1786{ 1787 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1788 module, 1789 (devices != NULL) ? *devices : 0, 1790 config->sample_rate, 1791 config->format, 1792 config->channel_mask, 1793 flags); 1794 1795 if (*devices == AUDIO_DEVICE_NONE) { 1796 return BAD_VALUE; 1797 } 1798 1799 Mutex::Autolock _l(mLock); 1800 1801 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1802 if (thread != 0) { 1803 *latencyMs = thread->latency(); 1804 1805 // notify client processes of the new output creation 1806 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1807 1808 // the first primary output opened designates the primary hw device 1809 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1810 ALOGI("Using module %d has the primary audio interface", module); 1811 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1812 1813 AutoMutex lock(mHardwareLock); 1814 mHardwareStatus = AUDIO_HW_SET_MODE; 1815 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1816 mHardwareStatus = AUDIO_HW_IDLE; 1817 1818 mPrimaryOutputSampleRate = config->sample_rate; 1819 } 1820 return NO_ERROR; 1821 } 1822 1823 return NO_INIT; 1824} 1825 1826audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1827 audio_io_handle_t output2) 1828{ 1829 Mutex::Autolock _l(mLock); 1830 MixerThread *thread1 = checkMixerThread_l(output1); 1831 MixerThread *thread2 = checkMixerThread_l(output2); 1832 1833 if (thread1 == NULL || thread2 == NULL) { 1834 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1835 output2); 1836 return AUDIO_IO_HANDLE_NONE; 1837 } 1838 1839 audio_io_handle_t id = nextUniqueId(); 1840 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1841 thread->addOutputTrack(thread2); 1842 mPlaybackThreads.add(id, thread); 1843 // notify client processes of the new output creation 1844 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1845 return id; 1846} 1847 1848status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1849{ 1850 return closeOutput_nonvirtual(output); 1851} 1852 1853status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1854{ 1855 // keep strong reference on the playback thread so that 1856 // it is not destroyed while exit() is executed 1857 sp<PlaybackThread> thread; 1858 { 1859 Mutex::Autolock _l(mLock); 1860 thread = checkPlaybackThread_l(output); 1861 if (thread == NULL) { 1862 return BAD_VALUE; 1863 } 1864 1865 ALOGV("closeOutput() %d", output); 1866 1867 if (thread->type() == ThreadBase::MIXER) { 1868 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1869 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1870 DuplicatingThread *dupThread = 1871 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1872 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1873 1874 } 1875 } 1876 } 1877 1878 1879 mPlaybackThreads.removeItem(output); 1880 // save all effects to the default thread 1881 if (mPlaybackThreads.size()) { 1882 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1883 if (dstThread != NULL) { 1884 // audioflinger lock is held here so the acquisition order of thread locks does not 1885 // matter 1886 Mutex::Autolock _dl(dstThread->mLock); 1887 Mutex::Autolock _sl(thread->mLock); 1888 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1889 for (size_t i = 0; i < effectChains.size(); i ++) { 1890 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1891 } 1892 } 1893 } 1894 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1895 } 1896 thread->exit(); 1897 // The thread entity (active unit of execution) is no longer running here, 1898 // but the ThreadBase container still exists. 1899 1900 if (thread->type() != ThreadBase::DUPLICATING) { 1901 closeOutputFinish(thread); 1902 } 1903 1904 return NO_ERROR; 1905} 1906 1907void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1908{ 1909 AudioStreamOut *out = thread->clearOutput(); 1910 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1911 // from now on thread->mOutput is NULL 1912 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1913 delete out; 1914} 1915 1916void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1917{ 1918 mPlaybackThreads.removeItem(thread->mId); 1919 thread->exit(); 1920 closeOutputFinish(thread); 1921} 1922 1923status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1924{ 1925 Mutex::Autolock _l(mLock); 1926 PlaybackThread *thread = checkPlaybackThread_l(output); 1927 1928 if (thread == NULL) { 1929 return BAD_VALUE; 1930 } 1931 1932 ALOGV("suspendOutput() %d", output); 1933 thread->suspend(); 1934 1935 return NO_ERROR; 1936} 1937 1938status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1939{ 1940 Mutex::Autolock _l(mLock); 1941 PlaybackThread *thread = checkPlaybackThread_l(output); 1942 1943 if (thread == NULL) { 1944 return BAD_VALUE; 1945 } 1946 1947 ALOGV("restoreOutput() %d", output); 1948 1949 thread->restore(); 1950 1951 return NO_ERROR; 1952} 1953 1954status_t AudioFlinger::openInput(audio_module_handle_t module, 1955 audio_io_handle_t *input, 1956 audio_config_t *config, 1957 audio_devices_t *device, 1958 const String8& address, 1959 audio_source_t source, 1960 audio_input_flags_t flags) 1961{ 1962 Mutex::Autolock _l(mLock); 1963 1964 if (*device == AUDIO_DEVICE_NONE) { 1965 return BAD_VALUE; 1966 } 1967 1968 sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags); 1969 1970 if (thread != 0) { 1971 // notify client processes of the new input creation 1972 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1973 return NO_ERROR; 1974 } 1975 return NO_INIT; 1976} 1977 1978sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 1979 audio_io_handle_t *input, 1980 audio_config_t *config, 1981 audio_devices_t device, 1982 const String8& address, 1983 audio_source_t source, 1984 audio_input_flags_t flags) 1985{ 1986 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device); 1987 if (inHwDev == NULL) { 1988 *input = AUDIO_IO_HANDLE_NONE; 1989 return 0; 1990 } 1991 1992 if (*input == AUDIO_IO_HANDLE_NONE) { 1993 *input = nextUniqueId(); 1994 } 1995 1996 audio_config_t halconfig = *config; 1997 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1998 audio_stream_in_t *inStream = NULL; 1999 status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 2000 &inStream, flags, address.string(), source); 2001 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2002 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2003 inStream, 2004 halconfig.sample_rate, 2005 halconfig.format, 2006 halconfig.channel_mask, 2007 flags, 2008 status, address.string()); 2009 2010 // If the input could not be opened with the requested parameters and we can handle the 2011 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 2012 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 2013 if (status == BAD_VALUE && 2014 config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT && 2015 (halconfig.sample_rate <= 2 * config->sample_rate) && 2016 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2017 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2018 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2019 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2020 inStream = NULL; 2021 status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 2022 &inStream, flags, address.string(), source); 2023 // FIXME log this new status; HAL should not propose any further changes 2024 } 2025 2026 if (status == NO_ERROR && inStream != NULL) { 2027 2028#ifdef TEE_SINK 2029 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2030 // or (re-)create if current Pipe is idle and does not match the new format 2031 sp<NBAIO_Sink> teeSink; 2032 enum { 2033 TEE_SINK_NO, // don't copy input 2034 TEE_SINK_NEW, // copy input using a new pipe 2035 TEE_SINK_OLD, // copy input using an existing pipe 2036 } kind; 2037 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2038 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2039 if (!mTeeSinkInputEnabled) { 2040 kind = TEE_SINK_NO; 2041 } else if (!Format_isValid(format)) { 2042 kind = TEE_SINK_NO; 2043 } else if (mRecordTeeSink == 0) { 2044 kind = TEE_SINK_NEW; 2045 } else if (mRecordTeeSink->getStrongCount() != 1) { 2046 kind = TEE_SINK_NO; 2047 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2048 kind = TEE_SINK_OLD; 2049 } else { 2050 kind = TEE_SINK_NEW; 2051 } 2052 switch (kind) { 2053 case TEE_SINK_NEW: { 2054 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2055 size_t numCounterOffers = 0; 2056 const NBAIO_Format offers[1] = {format}; 2057 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2058 ALOG_ASSERT(index == 0); 2059 PipeReader *pipeReader = new PipeReader(*pipe); 2060 numCounterOffers = 0; 2061 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2062 ALOG_ASSERT(index == 0); 2063 mRecordTeeSink = pipe; 2064 mRecordTeeSource = pipeReader; 2065 teeSink = pipe; 2066 } 2067 break; 2068 case TEE_SINK_OLD: 2069 teeSink = mRecordTeeSink; 2070 break; 2071 case TEE_SINK_NO: 2072 default: 2073 break; 2074 } 2075#endif 2076 2077 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2078 2079 // Start record thread 2080 // RecordThread requires both input and output device indication to forward to audio 2081 // pre processing modules 2082 sp<RecordThread> thread = new RecordThread(this, 2083 inputStream, 2084 *input, 2085 primaryOutputDevice_l(), 2086 device 2087#ifdef TEE_SINK 2088 , teeSink 2089#endif 2090 ); 2091 mRecordThreads.add(*input, thread); 2092 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2093 return thread; 2094 } 2095 2096 *input = AUDIO_IO_HANDLE_NONE; 2097 return 0; 2098} 2099 2100status_t AudioFlinger::closeInput(audio_io_handle_t input) 2101{ 2102 return closeInput_nonvirtual(input); 2103} 2104 2105status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2106{ 2107 // keep strong reference on the record thread so that 2108 // it is not destroyed while exit() is executed 2109 sp<RecordThread> thread; 2110 { 2111 Mutex::Autolock _l(mLock); 2112 thread = checkRecordThread_l(input); 2113 if (thread == 0) { 2114 return BAD_VALUE; 2115 } 2116 2117 ALOGV("closeInput() %d", input); 2118 2119 // If we still have effect chains, it means that a client still holds a handle 2120 // on at least one effect. We must either move the chain to an existing thread with the 2121 // same session ID or put it aside in case a new record thread is opened for a 2122 // new capture on the same session 2123 sp<EffectChain> chain; 2124 { 2125 Mutex::Autolock _sl(thread->mLock); 2126 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2127 // Note: maximum one chain per record thread 2128 if (effectChains.size() != 0) { 2129 chain = effectChains[0]; 2130 } 2131 } 2132 if (chain != 0) { 2133 // first check if a record thread is already opened with a client on the same session. 2134 // This should only happen in case of overlap between one thread tear down and the 2135 // creation of its replacement 2136 size_t i; 2137 for (i = 0; i < mRecordThreads.size(); i++) { 2138 sp<RecordThread> t = mRecordThreads.valueAt(i); 2139 if (t == thread) { 2140 continue; 2141 } 2142 if (t->hasAudioSession(chain->sessionId()) != 0) { 2143 Mutex::Autolock _l(t->mLock); 2144 ALOGV("closeInput() found thread %d for effect session %d", 2145 t->id(), chain->sessionId()); 2146 t->addEffectChain_l(chain); 2147 break; 2148 } 2149 } 2150 // put the chain aside if we could not find a record thread with the same session id. 2151 if (i == mRecordThreads.size()) { 2152 putOrphanEffectChain_l(chain); 2153 } 2154 } 2155 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 2156 mRecordThreads.removeItem(input); 2157 } 2158 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2159 // we have a different lock for notification client 2160 closeInputFinish(thread); 2161 return NO_ERROR; 2162} 2163 2164void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2165{ 2166 thread->exit(); 2167 AudioStreamIn *in = thread->clearInput(); 2168 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2169 // from now on thread->mInput is NULL 2170 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2171 delete in; 2172} 2173 2174void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2175{ 2176 mRecordThreads.removeItem(thread->mId); 2177 closeInputFinish(thread); 2178} 2179 2180status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2181{ 2182 Mutex::Autolock _l(mLock); 2183 ALOGV("invalidateStream() stream %d", stream); 2184 2185 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2186 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2187 thread->invalidateTracks(stream); 2188 } 2189 2190 return NO_ERROR; 2191} 2192 2193 2194audio_unique_id_t AudioFlinger::newAudioUniqueId() 2195{ 2196 return nextUniqueId(); 2197} 2198 2199void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2200{ 2201 Mutex::Autolock _l(mLock); 2202 pid_t caller = IPCThreadState::self()->getCallingPid(); 2203 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2204 if (pid != -1 && (caller == getpid_cached)) { 2205 caller = pid; 2206 } 2207 2208 { 2209 Mutex::Autolock _cl(mClientLock); 2210 // Ignore requests received from processes not known as notification client. The request 2211 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2212 // called from a different pid leaving a stale session reference. Also we don't know how 2213 // to clear this reference if the client process dies. 2214 if (mNotificationClients.indexOfKey(caller) < 0) { 2215 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2216 return; 2217 } 2218 } 2219 2220 size_t num = mAudioSessionRefs.size(); 2221 for (size_t i = 0; i< num; i++) { 2222 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2223 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2224 ref->mCnt++; 2225 ALOGV(" incremented refcount to %d", ref->mCnt); 2226 return; 2227 } 2228 } 2229 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2230 ALOGV(" added new entry for %d", audioSession); 2231} 2232 2233void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2234{ 2235 Mutex::Autolock _l(mLock); 2236 pid_t caller = IPCThreadState::self()->getCallingPid(); 2237 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2238 if (pid != -1 && (caller == getpid_cached)) { 2239 caller = pid; 2240 } 2241 size_t num = mAudioSessionRefs.size(); 2242 for (size_t i = 0; i< num; i++) { 2243 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2244 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2245 ref->mCnt--; 2246 ALOGV(" decremented refcount to %d", ref->mCnt); 2247 if (ref->mCnt == 0) { 2248 mAudioSessionRefs.removeAt(i); 2249 delete ref; 2250 purgeStaleEffects_l(); 2251 } 2252 return; 2253 } 2254 } 2255 // If the caller is mediaserver it is likely that the session being released was acquired 2256 // on behalf of a process not in notification clients and we ignore the warning. 2257 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2258} 2259 2260void AudioFlinger::purgeStaleEffects_l() { 2261 2262 ALOGV("purging stale effects"); 2263 2264 Vector< sp<EffectChain> > chains; 2265 2266 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2267 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2268 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2269 sp<EffectChain> ec = t->mEffectChains[j]; 2270 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2271 chains.push(ec); 2272 } 2273 } 2274 } 2275 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2276 sp<RecordThread> t = mRecordThreads.valueAt(i); 2277 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2278 sp<EffectChain> ec = t->mEffectChains[j]; 2279 chains.push(ec); 2280 } 2281 } 2282 2283 for (size_t i = 0; i < chains.size(); i++) { 2284 sp<EffectChain> ec = chains[i]; 2285 int sessionid = ec->sessionId(); 2286 sp<ThreadBase> t = ec->mThread.promote(); 2287 if (t == 0) { 2288 continue; 2289 } 2290 size_t numsessionrefs = mAudioSessionRefs.size(); 2291 bool found = false; 2292 for (size_t k = 0; k < numsessionrefs; k++) { 2293 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2294 if (ref->mSessionid == sessionid) { 2295 ALOGV(" session %d still exists for %d with %d refs", 2296 sessionid, ref->mPid, ref->mCnt); 2297 found = true; 2298 break; 2299 } 2300 } 2301 if (!found) { 2302 Mutex::Autolock _l(t->mLock); 2303 // remove all effects from the chain 2304 while (ec->mEffects.size()) { 2305 sp<EffectModule> effect = ec->mEffects[0]; 2306 effect->unPin(); 2307 t->removeEffect_l(effect); 2308 if (effect->purgeHandles()) { 2309 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2310 } 2311 AudioSystem::unregisterEffect(effect->id()); 2312 } 2313 } 2314 } 2315 return; 2316} 2317 2318// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2319AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2320{ 2321 return mPlaybackThreads.valueFor(output).get(); 2322} 2323 2324// checkMixerThread_l() must be called with AudioFlinger::mLock held 2325AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2326{ 2327 PlaybackThread *thread = checkPlaybackThread_l(output); 2328 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2329} 2330 2331// checkRecordThread_l() must be called with AudioFlinger::mLock held 2332AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2333{ 2334 return mRecordThreads.valueFor(input).get(); 2335} 2336 2337uint32_t AudioFlinger::nextUniqueId() 2338{ 2339 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2340} 2341 2342AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2343{ 2344 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2345 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2346 AudioStreamOut *output = thread->getOutput(); 2347 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2348 return thread; 2349 } 2350 } 2351 return NULL; 2352} 2353 2354audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2355{ 2356 PlaybackThread *thread = primaryPlaybackThread_l(); 2357 2358 if (thread == NULL) { 2359 return 0; 2360 } 2361 2362 return thread->outDevice(); 2363} 2364 2365sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2366 int triggerSession, 2367 int listenerSession, 2368 sync_event_callback_t callBack, 2369 wp<RefBase> cookie) 2370{ 2371 Mutex::Autolock _l(mLock); 2372 2373 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2374 status_t playStatus = NAME_NOT_FOUND; 2375 status_t recStatus = NAME_NOT_FOUND; 2376 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2377 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2378 if (playStatus == NO_ERROR) { 2379 return event; 2380 } 2381 } 2382 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2383 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2384 if (recStatus == NO_ERROR) { 2385 return event; 2386 } 2387 } 2388 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2389 mPendingSyncEvents.add(event); 2390 } else { 2391 ALOGV("createSyncEvent() invalid event %d", event->type()); 2392 event.clear(); 2393 } 2394 return event; 2395} 2396 2397// ---------------------------------------------------------------------------- 2398// Effect management 2399// ---------------------------------------------------------------------------- 2400 2401 2402status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2403{ 2404 Mutex::Autolock _l(mLock); 2405 return EffectQueryNumberEffects(numEffects); 2406} 2407 2408status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2409{ 2410 Mutex::Autolock _l(mLock); 2411 return EffectQueryEffect(index, descriptor); 2412} 2413 2414status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2415 effect_descriptor_t *descriptor) const 2416{ 2417 Mutex::Autolock _l(mLock); 2418 return EffectGetDescriptor(pUuid, descriptor); 2419} 2420 2421 2422sp<IEffect> AudioFlinger::createEffect( 2423 effect_descriptor_t *pDesc, 2424 const sp<IEffectClient>& effectClient, 2425 int32_t priority, 2426 audio_io_handle_t io, 2427 int sessionId, 2428 status_t *status, 2429 int *id, 2430 int *enabled) 2431{ 2432 status_t lStatus = NO_ERROR; 2433 sp<EffectHandle> handle; 2434 effect_descriptor_t desc; 2435 2436 pid_t pid = IPCThreadState::self()->getCallingPid(); 2437 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2438 pid, effectClient.get(), priority, sessionId, io); 2439 2440 if (pDesc == NULL) { 2441 lStatus = BAD_VALUE; 2442 goto Exit; 2443 } 2444 2445 // check audio settings permission for global effects 2446 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2447 lStatus = PERMISSION_DENIED; 2448 goto Exit; 2449 } 2450 2451 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2452 // that can only be created by audio policy manager (running in same process) 2453 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2454 lStatus = PERMISSION_DENIED; 2455 goto Exit; 2456 } 2457 2458 { 2459 if (!EffectIsNullUuid(&pDesc->uuid)) { 2460 // if uuid is specified, request effect descriptor 2461 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2462 if (lStatus < 0) { 2463 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2464 goto Exit; 2465 } 2466 } else { 2467 // if uuid is not specified, look for an available implementation 2468 // of the required type in effect factory 2469 if (EffectIsNullUuid(&pDesc->type)) { 2470 ALOGW("createEffect() no effect type"); 2471 lStatus = BAD_VALUE; 2472 goto Exit; 2473 } 2474 uint32_t numEffects = 0; 2475 effect_descriptor_t d; 2476 d.flags = 0; // prevent compiler warning 2477 bool found = false; 2478 2479 lStatus = EffectQueryNumberEffects(&numEffects); 2480 if (lStatus < 0) { 2481 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2482 goto Exit; 2483 } 2484 for (uint32_t i = 0; i < numEffects; i++) { 2485 lStatus = EffectQueryEffect(i, &desc); 2486 if (lStatus < 0) { 2487 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2488 continue; 2489 } 2490 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2491 // If matching type found save effect descriptor. If the session is 2492 // 0 and the effect is not auxiliary, continue enumeration in case 2493 // an auxiliary version of this effect type is available 2494 found = true; 2495 d = desc; 2496 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2497 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2498 break; 2499 } 2500 } 2501 } 2502 if (!found) { 2503 lStatus = BAD_VALUE; 2504 ALOGW("createEffect() effect not found"); 2505 goto Exit; 2506 } 2507 // For same effect type, chose auxiliary version over insert version if 2508 // connect to output mix (Compliance to OpenSL ES) 2509 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2510 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2511 desc = d; 2512 } 2513 } 2514 2515 // Do not allow auxiliary effects on a session different from 0 (output mix) 2516 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2517 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2518 lStatus = INVALID_OPERATION; 2519 goto Exit; 2520 } 2521 2522 // check recording permission for visualizer 2523 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2524 !recordingAllowed()) { 2525 lStatus = PERMISSION_DENIED; 2526 goto Exit; 2527 } 2528 2529 // return effect descriptor 2530 *pDesc = desc; 2531 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2532 // if the output returned by getOutputForEffect() is removed before we lock the 2533 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2534 // and we will exit safely 2535 io = AudioSystem::getOutputForEffect(&desc); 2536 ALOGV("createEffect got output %d", io); 2537 } 2538 2539 Mutex::Autolock _l(mLock); 2540 2541 // If output is not specified try to find a matching audio session ID in one of the 2542 // output threads. 2543 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2544 // because of code checking output when entering the function. 2545 // Note: io is never 0 when creating an effect on an input 2546 if (io == AUDIO_IO_HANDLE_NONE) { 2547 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2548 // output must be specified by AudioPolicyManager when using session 2549 // AUDIO_SESSION_OUTPUT_STAGE 2550 lStatus = BAD_VALUE; 2551 goto Exit; 2552 } 2553 // look for the thread where the specified audio session is present 2554 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2555 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2556 io = mPlaybackThreads.keyAt(i); 2557 break; 2558 } 2559 } 2560 if (io == 0) { 2561 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2562 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2563 io = mRecordThreads.keyAt(i); 2564 break; 2565 } 2566 } 2567 } 2568 // If no output thread contains the requested session ID, default to 2569 // first output. The effect chain will be moved to the correct output 2570 // thread when a track with the same session ID is created 2571 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2572 io = mPlaybackThreads.keyAt(0); 2573 } 2574 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2575 } 2576 ThreadBase *thread = checkRecordThread_l(io); 2577 if (thread == NULL) { 2578 thread = checkPlaybackThread_l(io); 2579 if (thread == NULL) { 2580 ALOGE("createEffect() unknown output thread"); 2581 lStatus = BAD_VALUE; 2582 goto Exit; 2583 } 2584 } else { 2585 // Check if one effect chain was awaiting for an effect to be created on this 2586 // session and used it instead of creating a new one. 2587 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2588 if (chain != 0) { 2589 Mutex::Autolock _l(thread->mLock); 2590 thread->addEffectChain_l(chain); 2591 } 2592 } 2593 2594 sp<Client> client = registerPid(pid); 2595 2596 // create effect on selected output thread 2597 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2598 &desc, enabled, &lStatus); 2599 if (handle != 0 && id != NULL) { 2600 *id = handle->id(); 2601 } 2602 if (handle == 0) { 2603 // remove local strong reference to Client with mClientLock held 2604 Mutex::Autolock _cl(mClientLock); 2605 client.clear(); 2606 } 2607 } 2608 2609Exit: 2610 *status = lStatus; 2611 return handle; 2612} 2613 2614status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2615 audio_io_handle_t dstOutput) 2616{ 2617 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2618 sessionId, srcOutput, dstOutput); 2619 Mutex::Autolock _l(mLock); 2620 if (srcOutput == dstOutput) { 2621 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2622 return NO_ERROR; 2623 } 2624 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2625 if (srcThread == NULL) { 2626 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2627 return BAD_VALUE; 2628 } 2629 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2630 if (dstThread == NULL) { 2631 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2632 return BAD_VALUE; 2633 } 2634 2635 Mutex::Autolock _dl(dstThread->mLock); 2636 Mutex::Autolock _sl(srcThread->mLock); 2637 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2638} 2639 2640// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2641status_t AudioFlinger::moveEffectChain_l(int sessionId, 2642 AudioFlinger::PlaybackThread *srcThread, 2643 AudioFlinger::PlaybackThread *dstThread, 2644 bool reRegister) 2645{ 2646 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2647 sessionId, srcThread, dstThread); 2648 2649 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2650 if (chain == 0) { 2651 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2652 sessionId, srcThread); 2653 return INVALID_OPERATION; 2654 } 2655 2656 // Check whether the destination thread has a channel count of FCC_2, which is 2657 // currently required for (most) effects. Prevent moving the effect chain here rather 2658 // than disabling the addEffect_l() call in dstThread below. 2659 if ((dstThread->type() == ThreadBase::MIXER || dstThread->type() == ThreadBase::DUPLICATING) && 2660 dstThread->mChannelCount != FCC_2) { 2661 ALOGW("moveEffectChain_l() effect chain failed because" 2662 " destination thread %p channel count(%u) != %u", 2663 dstThread, dstThread->mChannelCount, FCC_2); 2664 return INVALID_OPERATION; 2665 } 2666 2667 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2668 // so that a new chain is created with correct parameters when first effect is added. This is 2669 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2670 // removed. 2671 srcThread->removeEffectChain_l(chain); 2672 2673 // transfer all effects one by one so that new effect chain is created on new thread with 2674 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2675 sp<EffectChain> dstChain; 2676 uint32_t strategy = 0; // prevent compiler warning 2677 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2678 Vector< sp<EffectModule> > removed; 2679 status_t status = NO_ERROR; 2680 while (effect != 0) { 2681 srcThread->removeEffect_l(effect); 2682 removed.add(effect); 2683 status = dstThread->addEffect_l(effect); 2684 if (status != NO_ERROR) { 2685 break; 2686 } 2687 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2688 if (effect->state() == EffectModule::ACTIVE || 2689 effect->state() == EffectModule::STOPPING) { 2690 effect->start(); 2691 } 2692 // if the move request is not received from audio policy manager, the effect must be 2693 // re-registered with the new strategy and output 2694 if (dstChain == 0) { 2695 dstChain = effect->chain().promote(); 2696 if (dstChain == 0) { 2697 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2698 status = NO_INIT; 2699 break; 2700 } 2701 strategy = dstChain->strategy(); 2702 } 2703 if (reRegister) { 2704 AudioSystem::unregisterEffect(effect->id()); 2705 AudioSystem::registerEffect(&effect->desc(), 2706 dstThread->id(), 2707 strategy, 2708 sessionId, 2709 effect->id()); 2710 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2711 } 2712 effect = chain->getEffectFromId_l(0); 2713 } 2714 2715 if (status != NO_ERROR) { 2716 for (size_t i = 0; i < removed.size(); i++) { 2717 srcThread->addEffect_l(removed[i]); 2718 if (dstChain != 0 && reRegister) { 2719 AudioSystem::unregisterEffect(removed[i]->id()); 2720 AudioSystem::registerEffect(&removed[i]->desc(), 2721 srcThread->id(), 2722 strategy, 2723 sessionId, 2724 removed[i]->id()); 2725 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2726 } 2727 } 2728 } 2729 2730 return status; 2731} 2732 2733bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2734{ 2735 if (mGlobalEffectEnableTime != 0 && 2736 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2737 return true; 2738 } 2739 2740 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2741 sp<EffectChain> ec = 2742 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2743 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2744 return true; 2745 } 2746 } 2747 return false; 2748} 2749 2750void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2751{ 2752 Mutex::Autolock _l(mLock); 2753 2754 mGlobalEffectEnableTime = systemTime(); 2755 2756 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2757 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2758 if (t->mType == ThreadBase::OFFLOAD) { 2759 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2760 } 2761 } 2762 2763} 2764 2765status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2766{ 2767 audio_session_t session = (audio_session_t)chain->sessionId(); 2768 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2769 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2770 if (index >= 0) { 2771 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2772 return ALREADY_EXISTS; 2773 } 2774 mOrphanEffectChains.add(session, chain); 2775 return NO_ERROR; 2776} 2777 2778sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2779{ 2780 sp<EffectChain> chain; 2781 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2782 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2783 if (index >= 0) { 2784 chain = mOrphanEffectChains.valueAt(index); 2785 mOrphanEffectChains.removeItemsAt(index); 2786 } 2787 return chain; 2788} 2789 2790bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2791{ 2792 Mutex::Autolock _l(mLock); 2793 audio_session_t session = (audio_session_t)effect->sessionId(); 2794 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2795 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2796 if (index >= 0) { 2797 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2798 if (chain->removeEffect_l(effect) == 0) { 2799 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2800 mOrphanEffectChains.removeItemsAt(index); 2801 } 2802 return true; 2803 } 2804 return false; 2805} 2806 2807 2808struct Entry { 2809#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2810 char mName[MAX_NAME]; 2811}; 2812 2813int comparEntry(const void *p1, const void *p2) 2814{ 2815 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2816} 2817 2818#ifdef TEE_SINK 2819void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2820{ 2821 NBAIO_Source *teeSource = source.get(); 2822 if (teeSource != NULL) { 2823 // .wav rotation 2824 // There is a benign race condition if 2 threads call this simultaneously. 2825 // They would both traverse the directory, but the result would simply be 2826 // failures at unlink() which are ignored. It's also unlikely since 2827 // normally dumpsys is only done by bugreport or from the command line. 2828 char teePath[32+256]; 2829 strcpy(teePath, "/data/misc/media"); 2830 size_t teePathLen = strlen(teePath); 2831 DIR *dir = opendir(teePath); 2832 teePath[teePathLen++] = '/'; 2833 if (dir != NULL) { 2834#define MAX_SORT 20 // number of entries to sort 2835#define MAX_KEEP 10 // number of entries to keep 2836 struct Entry entries[MAX_SORT]; 2837 size_t entryCount = 0; 2838 while (entryCount < MAX_SORT) { 2839 struct dirent de; 2840 struct dirent *result = NULL; 2841 int rc = readdir_r(dir, &de, &result); 2842 if (rc != 0) { 2843 ALOGW("readdir_r failed %d", rc); 2844 break; 2845 } 2846 if (result == NULL) { 2847 break; 2848 } 2849 if (result != &de) { 2850 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2851 break; 2852 } 2853 // ignore non .wav file entries 2854 size_t nameLen = strlen(de.d_name); 2855 if (nameLen <= 4 || nameLen >= MAX_NAME || 2856 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2857 continue; 2858 } 2859 strcpy(entries[entryCount++].mName, de.d_name); 2860 } 2861 (void) closedir(dir); 2862 if (entryCount > MAX_KEEP) { 2863 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2864 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2865 strcpy(&teePath[teePathLen], entries[i].mName); 2866 (void) unlink(teePath); 2867 } 2868 } 2869 } else { 2870 if (fd >= 0) { 2871 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2872 } 2873 } 2874 char teeTime[16]; 2875 struct timeval tv; 2876 gettimeofday(&tv, NULL); 2877 struct tm tm; 2878 localtime_r(&tv.tv_sec, &tm); 2879 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2880 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2881 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2882 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2883 if (teeFd >= 0) { 2884 // FIXME use libsndfile 2885 char wavHeader[44]; 2886 memcpy(wavHeader, 2887 "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", 2888 sizeof(wavHeader)); 2889 NBAIO_Format format = teeSource->format(); 2890 unsigned channelCount = Format_channelCount(format); 2891 uint32_t sampleRate = Format_sampleRate(format); 2892 size_t frameSize = Format_frameSize(format); 2893 wavHeader[22] = channelCount; // number of channels 2894 wavHeader[24] = sampleRate; // sample rate 2895 wavHeader[25] = sampleRate >> 8; 2896 wavHeader[32] = frameSize; // block alignment 2897 wavHeader[33] = frameSize >> 8; 2898 write(teeFd, wavHeader, sizeof(wavHeader)); 2899 size_t total = 0; 2900 bool firstRead = true; 2901#define TEE_SINK_READ 1024 // frames per I/O operation 2902 void *buffer = malloc(TEE_SINK_READ * frameSize); 2903 for (;;) { 2904 size_t count = TEE_SINK_READ; 2905 ssize_t actual = teeSource->read(buffer, count, 2906 AudioBufferProvider::kInvalidPTS); 2907 bool wasFirstRead = firstRead; 2908 firstRead = false; 2909 if (actual <= 0) { 2910 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2911 continue; 2912 } 2913 break; 2914 } 2915 ALOG_ASSERT(actual <= (ssize_t)count); 2916 write(teeFd, buffer, actual * frameSize); 2917 total += actual; 2918 } 2919 free(buffer); 2920 lseek(teeFd, (off_t) 4, SEEK_SET); 2921 uint32_t temp = 44 + total * frameSize - 8; 2922 // FIXME not big-endian safe 2923 write(teeFd, &temp, sizeof(temp)); 2924 lseek(teeFd, (off_t) 40, SEEK_SET); 2925 temp = total * frameSize; 2926 // FIXME not big-endian safe 2927 write(teeFd, &temp, sizeof(temp)); 2928 close(teeFd); 2929 if (fd >= 0) { 2930 dprintf(fd, "tee copied to %s\n", teePath); 2931 } 2932 } else { 2933 if (fd >= 0) { 2934 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2935 } 2936 } 2937 } 2938} 2939#endif 2940 2941// ---------------------------------------------------------------------------- 2942 2943status_t AudioFlinger::onTransact( 2944 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2945{ 2946 return BnAudioFlinger::onTransact(code, data, reply, flags); 2947} 2948 2949}; // namespace android 2950