AudioFlinger.cpp revision cf7863ea8d9137aadf6bfd9756eb07ebd1c81b5c
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/EffectsFactoryApi.h> 49#include <audio_effects/effect_visualizer.h> 50#include <audio_effects/effect_ns.h> 51#include <audio_effects/effect_aec.h> 52 53#include <audio_utils/primitives.h> 54 55#include <powermanager/PowerManager.h> 56 57#include <common_time/cc_helper.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <private/android_filesystem_config.h> 65 66// ---------------------------------------------------------------------------- 67 68// Note: the following macro is used for extremely verbose logging message. In 69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70// 0; but one side effect of this is to turn all LOGV's as well. Some messages 71// are so verbose that we want to suppress them even when we have ALOG_ASSERT 72// turned on. Do not uncomment the #def below unless you really know what you 73// are doing and want to see all of the extremely verbose messages. 74//#define VERY_VERY_VERBOSE_LOGGING 75#ifdef VERY_VERY_VERBOSE_LOGGING 76#define ALOGVV ALOGV 77#else 78#define ALOGVV(a...) do { } while(0) 79#endif 80 81namespace android { 82 83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85static const char kClientLockedString[] = "Client lock is taken\n"; 86 87 88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 89 90uint32_t AudioFlinger::mScreenState; 91 92#ifdef TEE_SINK 93bool AudioFlinger::mTeeSinkInputEnabled = false; 94bool AudioFlinger::mTeeSinkOutputEnabled = false; 95bool AudioFlinger::mTeeSinkTrackEnabled = false; 96 97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 100#endif 101 102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 103// we define a minimum time during which a global effect is considered enabled. 104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 105 106// ---------------------------------------------------------------------------- 107 108const char *formatToString(audio_format_t format) { 109 switch (format & AUDIO_FORMAT_MAIN_MASK) { 110 case AUDIO_FORMAT_PCM: 111 switch (format) { 112 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 113 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 114 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 115 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 116 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 117 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 118 default: 119 break; 120 } 121 break; 122 case AUDIO_FORMAT_MP3: return "mp3"; 123 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 124 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 125 case AUDIO_FORMAT_AAC: return "aac"; 126 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 127 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 128 case AUDIO_FORMAT_VORBIS: return "vorbis"; 129 case AUDIO_FORMAT_OPUS: return "opus"; 130 case AUDIO_FORMAT_AC3: return "ac-3"; 131 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 132 default: 133 break; 134 } 135 return "unknown"; 136} 137 138static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 139{ 140 const hw_module_t *mod; 141 int rc; 142 143 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 144 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 145 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 146 if (rc) { 147 goto out; 148 } 149 rc = audio_hw_device_open(mod, dev); 150 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 156 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 157 rc = BAD_VALUE; 158 goto out; 159 } 160 return 0; 161 162out: 163 *dev = NULL; 164 return rc; 165} 166 167// ---------------------------------------------------------------------------- 168 169AudioFlinger::AudioFlinger() 170 : BnAudioFlinger(), 171 mPrimaryHardwareDev(NULL), 172 mAudioHwDevs(NULL), 173 mHardwareStatus(AUDIO_HW_IDLE), 174 mMasterVolume(1.0f), 175 mMasterMute(false), 176 mNextUniqueId(1), 177 mMode(AUDIO_MODE_INVALID), 178 mBtNrecIsOff(false), 179 mIsLowRamDevice(true), 180 mIsDeviceTypeKnown(false), 181 mGlobalEffectEnableTime(0), 182 mPrimaryOutputSampleRate(0) 183{ 184 getpid_cached = getpid(); 185 char value[PROPERTY_VALUE_MAX]; 186 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 187 if (doLog) { 188 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 189 } 190 191#ifdef TEE_SINK 192 (void) property_get("ro.debuggable", value, "0"); 193 int debuggable = atoi(value); 194 int teeEnabled = 0; 195 if (debuggable) { 196 (void) property_get("af.tee", value, "0"); 197 teeEnabled = atoi(value); 198 } 199 // FIXME symbolic constants here 200 if (teeEnabled & 1) { 201 mTeeSinkInputEnabled = true; 202 } 203 if (teeEnabled & 2) { 204 mTeeSinkOutputEnabled = true; 205 } 206 if (teeEnabled & 4) { 207 mTeeSinkTrackEnabled = true; 208 } 209#endif 210} 211 212void AudioFlinger::onFirstRef() 213{ 214 int rc = 0; 215 216 Mutex::Autolock _l(mLock); 217 218 /* TODO: move all this work into an Init() function */ 219 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 220 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 221 uint32_t int_val; 222 if (1 == sscanf(val_str, "%u", &int_val)) { 223 mStandbyTimeInNsecs = milliseconds(int_val); 224 ALOGI("Using %u mSec as standby time.", int_val); 225 } else { 226 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 227 ALOGI("Using default %u mSec as standby time.", 228 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 229 } 230 } 231 232 mPatchPanel = new PatchPanel(this); 233 234 mMode = AUDIO_MODE_NORMAL; 235} 236 237AudioFlinger::~AudioFlinger() 238{ 239 while (!mRecordThreads.isEmpty()) { 240 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 241 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 242 } 243 while (!mPlaybackThreads.isEmpty()) { 244 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 245 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 246 } 247 248 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 249 // no mHardwareLock needed, as there are no other references to this 250 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 251 delete mAudioHwDevs.valueAt(i); 252 } 253 254 // Tell media.log service about any old writers that still need to be unregistered 255 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 256 if (binder != 0) { 257 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 258 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 259 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 260 mUnregisteredWriters.pop(); 261 mediaLogService->unregisterWriter(iMemory); 262 } 263 } 264 265} 266 267static const char * const audio_interfaces[] = { 268 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 269 AUDIO_HARDWARE_MODULE_ID_A2DP, 270 AUDIO_HARDWARE_MODULE_ID_USB, 271}; 272#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 273 274AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 275 audio_module_handle_t module, 276 audio_devices_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 // then try to find a module supporting the requested device. 286 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 287 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 288 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 289 if ((dev->get_supported_devices != NULL) && 290 (dev->get_supported_devices(dev) & devices) == devices) 291 return audioHwDevice; 292 } 293 } else { 294 // check a match for the requested module handle 295 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 296 if (audioHwDevice != NULL) { 297 return audioHwDevice; 298 } 299 } 300 301 return NULL; 302} 303 304void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Notification Clients:\n"); 320 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 321 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 322 result.append(buffer); 323 } 324 325 result.append("Global session refs:\n"); 326 result.append(" session pid count\n"); 327 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 328 AudioSessionRef *r = mAudioSessionRefs[i]; 329 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 330 result.append(buffer); 331 } 332 write(fd, result.string(), result.size()); 333} 334 335 336void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 337{ 338 const size_t SIZE = 256; 339 char buffer[SIZE]; 340 String8 result; 341 hardware_call_state hardwareStatus = mHardwareStatus; 342 343 snprintf(buffer, SIZE, "Hardware status: %d\n" 344 "Standby Time mSec: %u\n", 345 hardwareStatus, 346 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 347 result.append(buffer); 348 write(fd, result.string(), result.size()); 349} 350 351void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 352{ 353 const size_t SIZE = 256; 354 char buffer[SIZE]; 355 String8 result; 356 snprintf(buffer, SIZE, "Permission Denial: " 357 "can't dump AudioFlinger from pid=%d, uid=%d\n", 358 IPCThreadState::self()->getCallingPid(), 359 IPCThreadState::self()->getCallingUid()); 360 result.append(buffer); 361 write(fd, result.string(), result.size()); 362} 363 364bool AudioFlinger::dumpTryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = dumpTryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = dumpTryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 bool clientLocked = dumpTryLock(mClientLock); 400 if (!clientLocked) { 401 String8 result(kClientLockedString); 402 write(fd, result.string(), result.size()); 403 } 404 dumpClients(fd, args); 405 if (clientLocked) { 406 mClientLock.unlock(); 407 } 408 409 dumpInternals(fd, args); 410 411 // dump playback threads 412 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 413 mPlaybackThreads.valueAt(i)->dump(fd, args); 414 } 415 416 // dump record threads 417 for (size_t i = 0; i < mRecordThreads.size(); i++) { 418 mRecordThreads.valueAt(i)->dump(fd, args); 419 } 420 421 // dump orphan effect chains 422 if (mOrphanEffectChains.size() != 0) { 423 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 424 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 425 mOrphanEffectChains.valueAt(i)->dump(fd, args); 426 } 427 } 428 // dump all hardware devs 429 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 430 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 431 dev->dump(dev, fd); 432 } 433 434#ifdef TEE_SINK 435 // dump the serially shared record tee sink 436 if (mRecordTeeSource != 0) { 437 dumpTee(fd, mRecordTeeSource); 438 } 439#endif 440 441 if (locked) { 442 mLock.unlock(); 443 } 444 445 // append a copy of media.log here by forwarding fd to it, but don't attempt 446 // to lookup the service if it's not running, as it will block for a second 447 if (mLogMemoryDealer != 0) { 448 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 449 if (binder != 0) { 450 dprintf(fd, "\nmedia.log:\n"); 451 Vector<String16> args; 452 binder->dump(fd, args); 453 } 454 } 455 } 456 return NO_ERROR; 457} 458 459sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 460{ 461 Mutex::Autolock _cl(mClientLock); 462 // If pid is already in the mClients wp<> map, then use that entry 463 // (for which promote() is always != 0), otherwise create a new entry and Client. 464 sp<Client> client = mClients.valueFor(pid).promote(); 465 if (client == 0) { 466 client = new Client(this, pid); 467 mClients.add(pid, client); 468 } 469 470 return client; 471} 472 473sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 474{ 475 // If there is no memory allocated for logs, return a dummy writer that does nothing 476 if (mLogMemoryDealer == 0) { 477 return new NBLog::Writer(); 478 } 479 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 480 // Similarly if we can't contact the media.log service, also return a dummy writer 481 if (binder == 0) { 482 return new NBLog::Writer(); 483 } 484 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 485 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 486 // If allocation fails, consult the vector of previously unregistered writers 487 // and garbage-collect one or more them until an allocation succeeds 488 if (shared == 0) { 489 Mutex::Autolock _l(mUnregisteredWritersLock); 490 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 491 { 492 // Pick the oldest stale writer to garbage-collect 493 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 494 mUnregisteredWriters.removeAt(0); 495 mediaLogService->unregisterWriter(iMemory); 496 // Now the media.log remote reference to IMemory is gone. When our last local 497 // reference to IMemory also drops to zero at end of this block, 498 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 499 } 500 // Re-attempt the allocation 501 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 502 if (shared != 0) { 503 goto success; 504 } 505 } 506 // Even after garbage-collecting all old writers, there is still not enough memory, 507 // so return a dummy writer 508 return new NBLog::Writer(); 509 } 510success: 511 mediaLogService->registerWriter(shared, size, name); 512 return new NBLog::Writer(size, shared); 513} 514 515void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 516{ 517 if (writer == 0) { 518 return; 519 } 520 sp<IMemory> iMemory(writer->getIMemory()); 521 if (iMemory == 0) { 522 return; 523 } 524 // Rather than removing the writer immediately, append it to a queue of old writers to 525 // be garbage-collected later. This allows us to continue to view old logs for a while. 526 Mutex::Autolock _l(mUnregisteredWritersLock); 527 mUnregisteredWriters.push(writer); 528} 529 530// IAudioFlinger interface 531 532 533sp<IAudioTrack> AudioFlinger::createTrack( 534 audio_stream_type_t streamType, 535 uint32_t sampleRate, 536 audio_format_t format, 537 audio_channel_mask_t channelMask, 538 size_t *frameCount, 539 IAudioFlinger::track_flags_t *flags, 540 const sp<IMemory>& sharedBuffer, 541 audio_io_handle_t output, 542 pid_t tid, 543 int *sessionId, 544 int clientUid, 545 status_t *status) 546{ 547 sp<PlaybackThread::Track> track; 548 sp<TrackHandle> trackHandle; 549 sp<Client> client; 550 status_t lStatus; 551 int lSessionId; 552 553 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 554 // but if someone uses binder directly they could bypass that and cause us to crash 555 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 556 ALOGE("createTrack() invalid stream type %d", streamType); 557 lStatus = BAD_VALUE; 558 goto Exit; 559 } 560 561 // further sample rate checks are performed by createTrack_l() depending on the thread type 562 if (sampleRate == 0) { 563 ALOGE("createTrack() invalid sample rate %u", sampleRate); 564 lStatus = BAD_VALUE; 565 goto Exit; 566 } 567 568 // further channel mask checks are performed by createTrack_l() depending on the thread type 569 if (!audio_is_output_channel(channelMask)) { 570 ALOGE("createTrack() invalid channel mask %#x", channelMask); 571 lStatus = BAD_VALUE; 572 goto Exit; 573 } 574 575 // further format checks are performed by createTrack_l() depending on the thread type 576 if (!audio_is_valid_format(format)) { 577 ALOGE("createTrack() invalid format %#x", format); 578 lStatus = BAD_VALUE; 579 goto Exit; 580 } 581 582 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 583 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 584 lStatus = BAD_VALUE; 585 goto Exit; 586 } 587 588 { 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGE("no playback thread found for output handle %d", output); 593 lStatus = BAD_VALUE; 594 goto Exit; 595 } 596 597 pid_t pid = IPCThreadState::self()->getCallingPid(); 598 client = registerPid(pid); 599 600 PlaybackThread *effectThread = NULL; 601 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 602 lSessionId = *sessionId; 603 // check if an effect chain with the same session ID is present on another 604 // output thread and move it here. 605 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 606 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 607 if (mPlaybackThreads.keyAt(i) != output) { 608 uint32_t sessions = t->hasAudioSession(lSessionId); 609 if (sessions & PlaybackThread::EFFECT_SESSION) { 610 effectThread = t.get(); 611 break; 612 } 613 } 614 } 615 } else { 616 // if no audio session id is provided, create one here 617 lSessionId = nextUniqueId(); 618 if (sessionId != NULL) { 619 *sessionId = lSessionId; 620 } 621 } 622 ALOGV("createTrack() lSessionId: %d", lSessionId); 623 624 track = thread->createTrack_l(client, streamType, sampleRate, format, 625 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 626 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 627 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 628 629 // move effect chain to this output thread if an effect on same session was waiting 630 // for a track to be created 631 if (lStatus == NO_ERROR && effectThread != NULL) { 632 // no risk of deadlock because AudioFlinger::mLock is held 633 Mutex::Autolock _dl(thread->mLock); 634 Mutex::Autolock _sl(effectThread->mLock); 635 moveEffectChain_l(lSessionId, effectThread, thread, true); 636 } 637 638 // Look for sync events awaiting for a session to be used. 639 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 640 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 641 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 642 if (lStatus == NO_ERROR) { 643 (void) track->setSyncEvent(mPendingSyncEvents[i]); 644 } else { 645 mPendingSyncEvents[i]->cancel(); 646 } 647 mPendingSyncEvents.removeAt(i); 648 i--; 649 } 650 } 651 } 652 653 } 654 655 if (lStatus != NO_ERROR) { 656 // remove local strong reference to Client before deleting the Track so that the 657 // Client destructor is called by the TrackBase destructor with mClientLock held 658 // Don't hold mClientLock when releasing the reference on the track as the 659 // destructor will acquire it. 660 { 661 Mutex::Autolock _cl(mClientLock); 662 client.clear(); 663 } 664 track.clear(); 665 goto Exit; 666 } 667 668 // return handle to client 669 trackHandle = new TrackHandle(track); 670 671Exit: 672 *status = lStatus; 673 return trackHandle; 674} 675 676uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 677{ 678 Mutex::Autolock _l(mLock); 679 PlaybackThread *thread = checkPlaybackThread_l(output); 680 if (thread == NULL) { 681 ALOGW("sampleRate() unknown thread %d", output); 682 return 0; 683 } 684 return thread->sampleRate(); 685} 686 687audio_format_t AudioFlinger::format(audio_io_handle_t output) const 688{ 689 Mutex::Autolock _l(mLock); 690 PlaybackThread *thread = checkPlaybackThread_l(output); 691 if (thread == NULL) { 692 ALOGW("format() unknown thread %d", output); 693 return AUDIO_FORMAT_INVALID; 694 } 695 return thread->format(); 696} 697 698size_t AudioFlinger::frameCount(audio_io_handle_t output) const 699{ 700 Mutex::Autolock _l(mLock); 701 PlaybackThread *thread = checkPlaybackThread_l(output); 702 if (thread == NULL) { 703 ALOGW("frameCount() unknown thread %d", output); 704 return 0; 705 } 706 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 707 // should examine all callers and fix them to handle smaller counts 708 return thread->frameCount(); 709} 710 711uint32_t AudioFlinger::latency(audio_io_handle_t output) const 712{ 713 Mutex::Autolock _l(mLock); 714 PlaybackThread *thread = checkPlaybackThread_l(output); 715 if (thread == NULL) { 716 ALOGW("latency(): no playback thread found for output handle %d", output); 717 return 0; 718 } 719 return thread->latency(); 720} 721 722status_t AudioFlinger::setMasterVolume(float value) 723{ 724 status_t ret = initCheck(); 725 if (ret != NO_ERROR) { 726 return ret; 727 } 728 729 // check calling permissions 730 if (!settingsAllowed()) { 731 return PERMISSION_DENIED; 732 } 733 734 Mutex::Autolock _l(mLock); 735 mMasterVolume = value; 736 737 // Set master volume in the HALs which support it. 738 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 739 AutoMutex lock(mHardwareLock); 740 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 741 742 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 743 if (dev->canSetMasterVolume()) { 744 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 745 } 746 mHardwareStatus = AUDIO_HW_IDLE; 747 } 748 749 // Now set the master volume in each playback thread. Playback threads 750 // assigned to HALs which do not have master volume support will apply 751 // master volume during the mix operation. Threads with HALs which do 752 // support master volume will simply ignore the setting. 753 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 754 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 755 756 return NO_ERROR; 757} 758 759status_t AudioFlinger::setMode(audio_mode_t mode) 760{ 761 status_t ret = initCheck(); 762 if (ret != NO_ERROR) { 763 return ret; 764 } 765 766 // check calling permissions 767 if (!settingsAllowed()) { 768 return PERMISSION_DENIED; 769 } 770 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 771 ALOGW("Illegal value: setMode(%d)", mode); 772 return BAD_VALUE; 773 } 774 775 { // scope for the lock 776 AutoMutex lock(mHardwareLock); 777 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 778 mHardwareStatus = AUDIO_HW_SET_MODE; 779 ret = dev->set_mode(dev, mode); 780 mHardwareStatus = AUDIO_HW_IDLE; 781 } 782 783 if (NO_ERROR == ret) { 784 Mutex::Autolock _l(mLock); 785 mMode = mode; 786 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 787 mPlaybackThreads.valueAt(i)->setMode(mode); 788 } 789 790 return ret; 791} 792 793status_t AudioFlinger::setMicMute(bool state) 794{ 795 status_t ret = initCheck(); 796 if (ret != NO_ERROR) { 797 return ret; 798 } 799 800 // check calling permissions 801 if (!settingsAllowed()) { 802 return PERMISSION_DENIED; 803 } 804 805 AutoMutex lock(mHardwareLock); 806 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 807 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 808 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 809 status_t result = dev->set_mic_mute(dev, state); 810 if (result != NO_ERROR) { 811 ret = result; 812 } 813 } 814 mHardwareStatus = AUDIO_HW_IDLE; 815 return ret; 816} 817 818bool AudioFlinger::getMicMute() const 819{ 820 status_t ret = initCheck(); 821 if (ret != NO_ERROR) { 822 return false; 823 } 824 825 bool state = AUDIO_MODE_INVALID; 826 AutoMutex lock(mHardwareLock); 827 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 828 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 829 dev->get_mic_mute(dev, &state); 830 mHardwareStatus = AUDIO_HW_IDLE; 831 return state; 832} 833 834status_t AudioFlinger::setMasterMute(bool muted) 835{ 836 status_t ret = initCheck(); 837 if (ret != NO_ERROR) { 838 return ret; 839 } 840 841 // check calling permissions 842 if (!settingsAllowed()) { 843 return PERMISSION_DENIED; 844 } 845 846 Mutex::Autolock _l(mLock); 847 mMasterMute = muted; 848 849 // Set master mute in the HALs which support it. 850 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 851 AutoMutex lock(mHardwareLock); 852 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 853 854 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 855 if (dev->canSetMasterMute()) { 856 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 857 } 858 mHardwareStatus = AUDIO_HW_IDLE; 859 } 860 861 // Now set the master mute in each playback thread. Playback threads 862 // assigned to HALs which do not have master mute support will apply master 863 // mute during the mix operation. Threads with HALs which do support master 864 // mute will simply ignore the setting. 865 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 866 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 867 868 return NO_ERROR; 869} 870 871float AudioFlinger::masterVolume() const 872{ 873 Mutex::Autolock _l(mLock); 874 return masterVolume_l(); 875} 876 877bool AudioFlinger::masterMute() const 878{ 879 Mutex::Autolock _l(mLock); 880 return masterMute_l(); 881} 882 883float AudioFlinger::masterVolume_l() const 884{ 885 return mMasterVolume; 886} 887 888bool AudioFlinger::masterMute_l() const 889{ 890 return mMasterMute; 891} 892 893status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 894 audio_io_handle_t output) 895{ 896 // check calling permissions 897 if (!settingsAllowed()) { 898 return PERMISSION_DENIED; 899 } 900 901 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 902 ALOGE("setStreamVolume() invalid stream %d", stream); 903 return BAD_VALUE; 904 } 905 906 AutoMutex lock(mLock); 907 PlaybackThread *thread = NULL; 908 if (output != AUDIO_IO_HANDLE_NONE) { 909 thread = checkPlaybackThread_l(output); 910 if (thread == NULL) { 911 return BAD_VALUE; 912 } 913 } 914 915 mStreamTypes[stream].volume = value; 916 917 if (thread == NULL) { 918 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 919 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 920 } 921 } else { 922 thread->setStreamVolume(stream, value); 923 } 924 925 return NO_ERROR; 926} 927 928status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 929{ 930 // check calling permissions 931 if (!settingsAllowed()) { 932 return PERMISSION_DENIED; 933 } 934 935 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 936 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 937 ALOGE("setStreamMute() invalid stream %d", stream); 938 return BAD_VALUE; 939 } 940 941 AutoMutex lock(mLock); 942 mStreamTypes[stream].mute = muted; 943 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 944 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 945 946 return NO_ERROR; 947} 948 949float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 950{ 951 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 952 return 0.0f; 953 } 954 955 AutoMutex lock(mLock); 956 float volume; 957 if (output != AUDIO_IO_HANDLE_NONE) { 958 PlaybackThread *thread = checkPlaybackThread_l(output); 959 if (thread == NULL) { 960 return 0.0f; 961 } 962 volume = thread->streamVolume(stream); 963 } else { 964 volume = streamVolume_l(stream); 965 } 966 967 return volume; 968} 969 970bool AudioFlinger::streamMute(audio_stream_type_t stream) const 971{ 972 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 973 return true; 974 } 975 976 AutoMutex lock(mLock); 977 return streamMute_l(stream); 978} 979 980status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 981{ 982 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 983 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 984 985 // check calling permissions 986 if (!settingsAllowed()) { 987 return PERMISSION_DENIED; 988 } 989 990 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 991 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 992 Mutex::Autolock _l(mLock); 993 status_t final_result = NO_ERROR; 994 { 995 AutoMutex lock(mHardwareLock); 996 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 997 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 998 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 999 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1000 final_result = result ?: final_result; 1001 } 1002 mHardwareStatus = AUDIO_HW_IDLE; 1003 } 1004 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1005 AudioParameter param = AudioParameter(keyValuePairs); 1006 String8 value; 1007 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1008 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1009 if (mBtNrecIsOff != btNrecIsOff) { 1010 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1011 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1012 audio_devices_t device = thread->inDevice(); 1013 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1014 // collect all of the thread's session IDs 1015 KeyedVector<int, bool> ids = thread->sessionIds(); 1016 // suspend effects associated with those session IDs 1017 for (size_t j = 0; j < ids.size(); ++j) { 1018 int sessionId = ids.keyAt(j); 1019 thread->setEffectSuspended(FX_IID_AEC, 1020 suspend, 1021 sessionId); 1022 thread->setEffectSuspended(FX_IID_NS, 1023 suspend, 1024 sessionId); 1025 } 1026 } 1027 mBtNrecIsOff = btNrecIsOff; 1028 } 1029 } 1030 String8 screenState; 1031 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1032 bool isOff = screenState == "off"; 1033 if (isOff != (AudioFlinger::mScreenState & 1)) { 1034 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1035 } 1036 } 1037 return final_result; 1038 } 1039 1040 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1041 // and the thread is exited once the lock is released 1042 sp<ThreadBase> thread; 1043 { 1044 Mutex::Autolock _l(mLock); 1045 thread = checkPlaybackThread_l(ioHandle); 1046 if (thread == 0) { 1047 thread = checkRecordThread_l(ioHandle); 1048 } else if (thread == primaryPlaybackThread_l()) { 1049 // indicate output device change to all input threads for pre processing 1050 AudioParameter param = AudioParameter(keyValuePairs); 1051 int value; 1052 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1053 (value != 0)) { 1054 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1055 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1056 } 1057 } 1058 } 1059 } 1060 if (thread != 0) { 1061 return thread->setParameters(keyValuePairs); 1062 } 1063 return BAD_VALUE; 1064} 1065 1066String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1067{ 1068 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1069 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1070 1071 Mutex::Autolock _l(mLock); 1072 1073 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1074 String8 out_s8; 1075 1076 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1077 char *s; 1078 { 1079 AutoMutex lock(mHardwareLock); 1080 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1081 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1082 s = dev->get_parameters(dev, keys.string()); 1083 mHardwareStatus = AUDIO_HW_IDLE; 1084 } 1085 out_s8 += String8(s ? s : ""); 1086 free(s); 1087 } 1088 return out_s8; 1089 } 1090 1091 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1092 if (playbackThread != NULL) { 1093 return playbackThread->getParameters(keys); 1094 } 1095 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1096 if (recordThread != NULL) { 1097 return recordThread->getParameters(keys); 1098 } 1099 return String8(""); 1100} 1101 1102size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1103 audio_channel_mask_t channelMask) const 1104{ 1105 status_t ret = initCheck(); 1106 if (ret != NO_ERROR) { 1107 return 0; 1108 } 1109 1110 AutoMutex lock(mHardwareLock); 1111 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1112 audio_config_t config; 1113 memset(&config, 0, sizeof(config)); 1114 config.sample_rate = sampleRate; 1115 config.channel_mask = channelMask; 1116 config.format = format; 1117 1118 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1119 size_t size = dev->get_input_buffer_size(dev, &config); 1120 mHardwareStatus = AUDIO_HW_IDLE; 1121 return size; 1122} 1123 1124uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1125{ 1126 Mutex::Autolock _l(mLock); 1127 1128 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1129 if (recordThread != NULL) { 1130 return recordThread->getInputFramesLost(); 1131 } 1132 return 0; 1133} 1134 1135status_t AudioFlinger::setVoiceVolume(float value) 1136{ 1137 status_t ret = initCheck(); 1138 if (ret != NO_ERROR) { 1139 return ret; 1140 } 1141 1142 // check calling permissions 1143 if (!settingsAllowed()) { 1144 return PERMISSION_DENIED; 1145 } 1146 1147 AutoMutex lock(mHardwareLock); 1148 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1149 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1150 ret = dev->set_voice_volume(dev, value); 1151 mHardwareStatus = AUDIO_HW_IDLE; 1152 1153 return ret; 1154} 1155 1156status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1157 audio_io_handle_t output) const 1158{ 1159 status_t status; 1160 1161 Mutex::Autolock _l(mLock); 1162 1163 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1164 if (playbackThread != NULL) { 1165 return playbackThread->getRenderPosition(halFrames, dspFrames); 1166 } 1167 1168 return BAD_VALUE; 1169} 1170 1171void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1172{ 1173 Mutex::Autolock _l(mLock); 1174 if (client == 0) { 1175 return; 1176 } 1177 bool clientAdded = false; 1178 { 1179 Mutex::Autolock _cl(mClientLock); 1180 1181 pid_t pid = IPCThreadState::self()->getCallingPid(); 1182 if (mNotificationClients.indexOfKey(pid) < 0) { 1183 sp<NotificationClient> notificationClient = new NotificationClient(this, 1184 client, 1185 pid); 1186 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1187 1188 mNotificationClients.add(pid, notificationClient); 1189 1190 sp<IBinder> binder = client->asBinder(); 1191 binder->linkToDeath(notificationClient); 1192 clientAdded = true; 1193 } 1194 } 1195 1196 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1197 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1198 if (clientAdded) { 1199 // the config change is always sent from playback or record threads to avoid deadlock 1200 // with AudioSystem::gLock 1201 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1202 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1203 } 1204 1205 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1206 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1207 } 1208 } 1209} 1210 1211void AudioFlinger::removeNotificationClient(pid_t pid) 1212{ 1213 Mutex::Autolock _l(mLock); 1214 { 1215 Mutex::Autolock _cl(mClientLock); 1216 mNotificationClients.removeItem(pid); 1217 } 1218 1219 ALOGV("%d died, releasing its sessions", pid); 1220 size_t num = mAudioSessionRefs.size(); 1221 bool removed = false; 1222 for (size_t i = 0; i< num; ) { 1223 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1224 ALOGV(" pid %d @ %d", ref->mPid, i); 1225 if (ref->mPid == pid) { 1226 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1227 mAudioSessionRefs.removeAt(i); 1228 delete ref; 1229 removed = true; 1230 num--; 1231 } else { 1232 i++; 1233 } 1234 } 1235 if (removed) { 1236 purgeStaleEffects_l(); 1237 } 1238} 1239 1240void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1241{ 1242 Mutex::Autolock _l(mClientLock); 1243 size_t size = mNotificationClients.size(); 1244 for (size_t i = 0; i < size; i++) { 1245 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1246 ioHandle, 1247 param2); 1248 } 1249} 1250 1251// removeClient_l() must be called with AudioFlinger::mClientLock held 1252void AudioFlinger::removeClient_l(pid_t pid) 1253{ 1254 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1255 IPCThreadState::self()->getCallingPid()); 1256 mClients.removeItem(pid); 1257} 1258 1259// getEffectThread_l() must be called with AudioFlinger::mLock held 1260sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1261{ 1262 sp<PlaybackThread> thread; 1263 1264 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1265 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1266 ALOG_ASSERT(thread == 0); 1267 thread = mPlaybackThreads.valueAt(i); 1268 } 1269 } 1270 1271 return thread; 1272} 1273 1274 1275 1276// ---------------------------------------------------------------------------- 1277 1278AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1279 : RefBase(), 1280 mAudioFlinger(audioFlinger), 1281 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1282 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1283 mPid(pid), 1284 mTimedTrackCount(0) 1285{ 1286 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1287} 1288 1289// Client destructor must be called with AudioFlinger::mClientLock held 1290AudioFlinger::Client::~Client() 1291{ 1292 mAudioFlinger->removeClient_l(mPid); 1293} 1294 1295sp<MemoryDealer> AudioFlinger::Client::heap() const 1296{ 1297 return mMemoryDealer; 1298} 1299 1300// Reserve one of the limited slots for a timed audio track associated 1301// with this client 1302bool AudioFlinger::Client::reserveTimedTrack() 1303{ 1304 const int kMaxTimedTracksPerClient = 4; 1305 1306 Mutex::Autolock _l(mTimedTrackLock); 1307 1308 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1309 ALOGW("can not create timed track - pid %d has exceeded the limit", 1310 mPid); 1311 return false; 1312 } 1313 1314 mTimedTrackCount++; 1315 return true; 1316} 1317 1318// Release a slot for a timed audio track 1319void AudioFlinger::Client::releaseTimedTrack() 1320{ 1321 Mutex::Autolock _l(mTimedTrackLock); 1322 mTimedTrackCount--; 1323} 1324 1325// ---------------------------------------------------------------------------- 1326 1327AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1328 const sp<IAudioFlingerClient>& client, 1329 pid_t pid) 1330 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1331{ 1332} 1333 1334AudioFlinger::NotificationClient::~NotificationClient() 1335{ 1336} 1337 1338void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1339{ 1340 sp<NotificationClient> keep(this); 1341 mAudioFlinger->removeNotificationClient(mPid); 1342} 1343 1344 1345// ---------------------------------------------------------------------------- 1346 1347static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1348 return audio_is_remote_submix_device(inDevice); 1349} 1350 1351sp<IAudioRecord> AudioFlinger::openRecord( 1352 audio_io_handle_t input, 1353 uint32_t sampleRate, 1354 audio_format_t format, 1355 audio_channel_mask_t channelMask, 1356 size_t *frameCount, 1357 IAudioFlinger::track_flags_t *flags, 1358 pid_t tid, 1359 int *sessionId, 1360 size_t *notificationFrames, 1361 sp<IMemory>& cblk, 1362 sp<IMemory>& buffers, 1363 status_t *status) 1364{ 1365 sp<RecordThread::RecordTrack> recordTrack; 1366 sp<RecordHandle> recordHandle; 1367 sp<Client> client; 1368 status_t lStatus; 1369 int lSessionId; 1370 1371 cblk.clear(); 1372 buffers.clear(); 1373 1374 // check calling permissions 1375 if (!recordingAllowed()) { 1376 ALOGE("openRecord() permission denied: recording not allowed"); 1377 lStatus = PERMISSION_DENIED; 1378 goto Exit; 1379 } 1380 1381 // further sample rate checks are performed by createRecordTrack_l() 1382 if (sampleRate == 0) { 1383 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1384 lStatus = BAD_VALUE; 1385 goto Exit; 1386 } 1387 1388 // we don't yet support anything other than 16-bit PCM 1389 if (!(audio_is_valid_format(format) && 1390 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1391 ALOGE("openRecord() invalid format %#x", format); 1392 lStatus = BAD_VALUE; 1393 goto Exit; 1394 } 1395 1396 // further channel mask checks are performed by createRecordTrack_l() 1397 if (!audio_is_input_channel(channelMask)) { 1398 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1399 lStatus = BAD_VALUE; 1400 goto Exit; 1401 } 1402 1403 { 1404 Mutex::Autolock _l(mLock); 1405 RecordThread *thread = checkRecordThread_l(input); 1406 if (thread == NULL) { 1407 ALOGE("openRecord() checkRecordThread_l failed"); 1408 lStatus = BAD_VALUE; 1409 goto Exit; 1410 } 1411 1412 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1413 && !captureAudioOutputAllowed()) { 1414 ALOGE("openRecord() permission denied: capture not allowed"); 1415 lStatus = PERMISSION_DENIED; 1416 goto Exit; 1417 } 1418 1419 pid_t pid = IPCThreadState::self()->getCallingPid(); 1420 client = registerPid(pid); 1421 1422 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1423 lSessionId = *sessionId; 1424 } else { 1425 // if no audio session id is provided, create one here 1426 lSessionId = nextUniqueId(); 1427 if (sessionId != NULL) { 1428 *sessionId = lSessionId; 1429 } 1430 } 1431 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1432 1433 // TODO: the uid should be passed in as a parameter to openRecord 1434 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1435 frameCount, lSessionId, notificationFrames, 1436 IPCThreadState::self()->getCallingUid(), 1437 flags, tid, &lStatus); 1438 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1439 } 1440 1441 if (lStatus != NO_ERROR) { 1442 // remove local strong reference to Client before deleting the RecordTrack so that the 1443 // Client destructor is called by the TrackBase destructor with mClientLock held 1444 // Don't hold mClientLock when releasing the reference on the track as the 1445 // destructor will acquire it. 1446 { 1447 Mutex::Autolock _cl(mClientLock); 1448 client.clear(); 1449 } 1450 recordTrack.clear(); 1451 goto Exit; 1452 } 1453 1454 cblk = recordTrack->getCblk(); 1455 buffers = recordTrack->getBuffers(); 1456 1457 // return handle to client 1458 recordHandle = new RecordHandle(recordTrack); 1459 1460Exit: 1461 *status = lStatus; 1462 return recordHandle; 1463} 1464 1465 1466 1467// ---------------------------------------------------------------------------- 1468 1469audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1470{ 1471 if (name == NULL) { 1472 return 0; 1473 } 1474 if (!settingsAllowed()) { 1475 return 0; 1476 } 1477 Mutex::Autolock _l(mLock); 1478 return loadHwModule_l(name); 1479} 1480 1481// loadHwModule_l() must be called with AudioFlinger::mLock held 1482audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1483{ 1484 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1485 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1486 ALOGW("loadHwModule() module %s already loaded", name); 1487 return mAudioHwDevs.keyAt(i); 1488 } 1489 } 1490 1491 audio_hw_device_t *dev; 1492 1493 int rc = load_audio_interface(name, &dev); 1494 if (rc) { 1495 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1496 return 0; 1497 } 1498 1499 mHardwareStatus = AUDIO_HW_INIT; 1500 rc = dev->init_check(dev); 1501 mHardwareStatus = AUDIO_HW_IDLE; 1502 if (rc) { 1503 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1504 return 0; 1505 } 1506 1507 // Check and cache this HAL's level of support for master mute and master 1508 // volume. If this is the first HAL opened, and it supports the get 1509 // methods, use the initial values provided by the HAL as the current 1510 // master mute and volume settings. 1511 1512 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1513 { // scope for auto-lock pattern 1514 AutoMutex lock(mHardwareLock); 1515 1516 if (0 == mAudioHwDevs.size()) { 1517 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1518 if (NULL != dev->get_master_volume) { 1519 float mv; 1520 if (OK == dev->get_master_volume(dev, &mv)) { 1521 mMasterVolume = mv; 1522 } 1523 } 1524 1525 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1526 if (NULL != dev->get_master_mute) { 1527 bool mm; 1528 if (OK == dev->get_master_mute(dev, &mm)) { 1529 mMasterMute = mm; 1530 } 1531 } 1532 } 1533 1534 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1535 if ((NULL != dev->set_master_volume) && 1536 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1537 flags = static_cast<AudioHwDevice::Flags>(flags | 1538 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1539 } 1540 1541 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1542 if ((NULL != dev->set_master_mute) && 1543 (OK == dev->set_master_mute(dev, mMasterMute))) { 1544 flags = static_cast<AudioHwDevice::Flags>(flags | 1545 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1546 } 1547 1548 mHardwareStatus = AUDIO_HW_IDLE; 1549 } 1550 1551 audio_module_handle_t handle = nextUniqueId(); 1552 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1553 1554 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1555 name, dev->common.module->name, dev->common.module->id, handle); 1556 1557 return handle; 1558 1559} 1560 1561// ---------------------------------------------------------------------------- 1562 1563uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1564{ 1565 Mutex::Autolock _l(mLock); 1566 PlaybackThread *thread = primaryPlaybackThread_l(); 1567 return thread != NULL ? thread->sampleRate() : 0; 1568} 1569 1570size_t AudioFlinger::getPrimaryOutputFrameCount() 1571{ 1572 Mutex::Autolock _l(mLock); 1573 PlaybackThread *thread = primaryPlaybackThread_l(); 1574 return thread != NULL ? thread->frameCountHAL() : 0; 1575} 1576 1577// ---------------------------------------------------------------------------- 1578 1579status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1580{ 1581 uid_t uid = IPCThreadState::self()->getCallingUid(); 1582 if (uid != AID_SYSTEM) { 1583 return PERMISSION_DENIED; 1584 } 1585 Mutex::Autolock _l(mLock); 1586 if (mIsDeviceTypeKnown) { 1587 return INVALID_OPERATION; 1588 } 1589 mIsLowRamDevice = isLowRamDevice; 1590 mIsDeviceTypeKnown = true; 1591 return NO_ERROR; 1592} 1593 1594audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1595{ 1596 Mutex::Autolock _l(mLock); 1597 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1598 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1599 if ((thread->hasAudioSession(sessionId) & ThreadBase::TRACK_SESSION) != 0) { 1600 // A session can only be on one thread, so exit after first match 1601 String8 reply = thread->getParameters(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC)); 1602 AudioParameter param = AudioParameter(reply); 1603 int value; 1604 if (param.getInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value) == NO_ERROR) { 1605 return value; 1606 } 1607 break; 1608 } 1609 } 1610 return AUDIO_HW_SYNC_INVALID; 1611} 1612 1613// ---------------------------------------------------------------------------- 1614 1615 1616sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1617 audio_io_handle_t *output, 1618 audio_config_t *config, 1619 audio_devices_t devices, 1620 const String8& address, 1621 audio_output_flags_t flags) 1622{ 1623 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1624 if (outHwDev == NULL) { 1625 return 0; 1626 } 1627 1628 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1629 if (*output == AUDIO_IO_HANDLE_NONE) { 1630 *output = nextUniqueId(); 1631 } 1632 1633 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1634 1635 audio_stream_out_t *outStream = NULL; 1636 1637 // FOR TESTING ONLY: 1638 // This if statement allows overriding the audio policy settings 1639 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1640 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1641 // Check only for Normal Mixing mode 1642 if (kEnableExtendedPrecision) { 1643 // Specify format (uncomment one below to choose) 1644 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1645 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1646 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1647 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1648 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1649 } 1650 if (kEnableExtendedChannels) { 1651 // Specify channel mask (uncomment one below to choose) 1652 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1653 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1654 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1655 } 1656 } 1657 1658 status_t status = hwDevHal->open_output_stream(hwDevHal, 1659 *output, 1660 devices, 1661 flags, 1662 config, 1663 &outStream, 1664 address.string()); 1665 1666 mHardwareStatus = AUDIO_HW_IDLE; 1667 ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, " 1668 "channelMask %#x, status %d", 1669 outStream, 1670 config->sample_rate, 1671 config->format, 1672 config->channel_mask, 1673 status); 1674 1675 if (status == NO_ERROR && outStream != NULL) { 1676 AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags); 1677 1678 PlaybackThread *thread; 1679 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1680 thread = new OffloadThread(this, outputStream, *output, devices); 1681 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1682 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1683 || !isValidPcmSinkFormat(config->format) 1684 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1685 thread = new DirectOutputThread(this, outputStream, *output, devices); 1686 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1687 } else { 1688 thread = new MixerThread(this, outputStream, *output, devices); 1689 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1690 } 1691 mPlaybackThreads.add(*output, thread); 1692 return thread; 1693 } 1694 1695 return 0; 1696} 1697 1698status_t AudioFlinger::openOutput(audio_module_handle_t module, 1699 audio_io_handle_t *output, 1700 audio_config_t *config, 1701 audio_devices_t *devices, 1702 const String8& address, 1703 uint32_t *latencyMs, 1704 audio_output_flags_t flags) 1705{ 1706 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1707 module, 1708 (devices != NULL) ? *devices : 0, 1709 config->sample_rate, 1710 config->format, 1711 config->channel_mask, 1712 flags); 1713 1714 if (*devices == AUDIO_DEVICE_NONE) { 1715 return BAD_VALUE; 1716 } 1717 1718 Mutex::Autolock _l(mLock); 1719 1720 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1721 if (thread != 0) { 1722 *latencyMs = thread->latency(); 1723 1724 // notify client processes of the new output creation 1725 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1726 1727 // the first primary output opened designates the primary hw device 1728 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1729 ALOGI("Using module %d has the primary audio interface", module); 1730 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1731 1732 AutoMutex lock(mHardwareLock); 1733 mHardwareStatus = AUDIO_HW_SET_MODE; 1734 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1735 mHardwareStatus = AUDIO_HW_IDLE; 1736 1737 mPrimaryOutputSampleRate = config->sample_rate; 1738 } 1739 return NO_ERROR; 1740 } 1741 1742 return NO_INIT; 1743} 1744 1745audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1746 audio_io_handle_t output2) 1747{ 1748 Mutex::Autolock _l(mLock); 1749 MixerThread *thread1 = checkMixerThread_l(output1); 1750 MixerThread *thread2 = checkMixerThread_l(output2); 1751 1752 if (thread1 == NULL || thread2 == NULL) { 1753 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1754 output2); 1755 return AUDIO_IO_HANDLE_NONE; 1756 } 1757 1758 audio_io_handle_t id = nextUniqueId(); 1759 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1760 thread->addOutputTrack(thread2); 1761 mPlaybackThreads.add(id, thread); 1762 // notify client processes of the new output creation 1763 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1764 return id; 1765} 1766 1767status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1768{ 1769 return closeOutput_nonvirtual(output); 1770} 1771 1772status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1773{ 1774 // keep strong reference on the playback thread so that 1775 // it is not destroyed while exit() is executed 1776 sp<PlaybackThread> thread; 1777 { 1778 Mutex::Autolock _l(mLock); 1779 thread = checkPlaybackThread_l(output); 1780 if (thread == NULL) { 1781 return BAD_VALUE; 1782 } 1783 1784 ALOGV("closeOutput() %d", output); 1785 1786 if (thread->type() == ThreadBase::MIXER) { 1787 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1788 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1789 DuplicatingThread *dupThread = 1790 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1791 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1792 1793 } 1794 } 1795 } 1796 1797 1798 mPlaybackThreads.removeItem(output); 1799 // save all effects to the default thread 1800 if (mPlaybackThreads.size()) { 1801 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1802 if (dstThread != NULL) { 1803 // audioflinger lock is held here so the acquisition order of thread locks does not 1804 // matter 1805 Mutex::Autolock _dl(dstThread->mLock); 1806 Mutex::Autolock _sl(thread->mLock); 1807 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1808 for (size_t i = 0; i < effectChains.size(); i ++) { 1809 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1810 } 1811 } 1812 } 1813 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1814 } 1815 thread->exit(); 1816 // The thread entity (active unit of execution) is no longer running here, 1817 // but the ThreadBase container still exists. 1818 1819 if (thread->type() != ThreadBase::DUPLICATING) { 1820 closeOutputFinish(thread); 1821 } 1822 1823 return NO_ERROR; 1824} 1825 1826void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1827{ 1828 AudioStreamOut *out = thread->clearOutput(); 1829 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1830 // from now on thread->mOutput is NULL 1831 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1832 delete out; 1833} 1834 1835void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1836{ 1837 mPlaybackThreads.removeItem(thread->mId); 1838 thread->exit(); 1839 closeOutputFinish(thread); 1840} 1841 1842status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1843{ 1844 Mutex::Autolock _l(mLock); 1845 PlaybackThread *thread = checkPlaybackThread_l(output); 1846 1847 if (thread == NULL) { 1848 return BAD_VALUE; 1849 } 1850 1851 ALOGV("suspendOutput() %d", output); 1852 thread->suspend(); 1853 1854 return NO_ERROR; 1855} 1856 1857status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1858{ 1859 Mutex::Autolock _l(mLock); 1860 PlaybackThread *thread = checkPlaybackThread_l(output); 1861 1862 if (thread == NULL) { 1863 return BAD_VALUE; 1864 } 1865 1866 ALOGV("restoreOutput() %d", output); 1867 1868 thread->restore(); 1869 1870 return NO_ERROR; 1871} 1872 1873status_t AudioFlinger::openInput(audio_module_handle_t module, 1874 audio_io_handle_t *input, 1875 audio_config_t *config, 1876 audio_devices_t *device, 1877 const String8& address, 1878 audio_source_t source, 1879 audio_input_flags_t flags) 1880{ 1881 Mutex::Autolock _l(mLock); 1882 1883 if (*device == AUDIO_DEVICE_NONE) { 1884 return BAD_VALUE; 1885 } 1886 1887 sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags); 1888 1889 if (thread != 0) { 1890 // notify client processes of the new input creation 1891 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1892 return NO_ERROR; 1893 } 1894 return NO_INIT; 1895} 1896 1897sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 1898 audio_io_handle_t *input, 1899 audio_config_t *config, 1900 audio_devices_t device, 1901 const String8& address, 1902 audio_source_t source, 1903 audio_input_flags_t flags) 1904{ 1905 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device); 1906 if (inHwDev == NULL) { 1907 *input = AUDIO_IO_HANDLE_NONE; 1908 return 0; 1909 } 1910 1911 if (*input == AUDIO_IO_HANDLE_NONE) { 1912 *input = nextUniqueId(); 1913 } 1914 1915 audio_config_t halconfig = *config; 1916 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1917 audio_stream_in_t *inStream = NULL; 1918 status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1919 &inStream, flags, address.string(), source); 1920 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 1921 ", Format %#x, Channels %x, flags %#x, status %d", 1922 inStream, 1923 halconfig.sample_rate, 1924 halconfig.format, 1925 halconfig.channel_mask, 1926 flags, 1927 status); 1928 1929 // If the input could not be opened with the requested parameters and we can handle the 1930 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1931 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1932 if (status == BAD_VALUE && 1933 config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT && 1934 (halconfig.sample_rate <= 2 * config->sample_rate) && 1935 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 1936 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 1937 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1938 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 1939 inStream = NULL; 1940 status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1941 &inStream, flags, address.string(), source); 1942 // FIXME log this new status; HAL should not propose any further changes 1943 } 1944 1945 if (status == NO_ERROR && inStream != NULL) { 1946 1947#ifdef TEE_SINK 1948 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1949 // or (re-)create if current Pipe is idle and does not match the new format 1950 sp<NBAIO_Sink> teeSink; 1951 enum { 1952 TEE_SINK_NO, // don't copy input 1953 TEE_SINK_NEW, // copy input using a new pipe 1954 TEE_SINK_OLD, // copy input using an existing pipe 1955 } kind; 1956 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 1957 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 1958 if (!mTeeSinkInputEnabled) { 1959 kind = TEE_SINK_NO; 1960 } else if (!Format_isValid(format)) { 1961 kind = TEE_SINK_NO; 1962 } else if (mRecordTeeSink == 0) { 1963 kind = TEE_SINK_NEW; 1964 } else if (mRecordTeeSink->getStrongCount() != 1) { 1965 kind = TEE_SINK_NO; 1966 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1967 kind = TEE_SINK_OLD; 1968 } else { 1969 kind = TEE_SINK_NEW; 1970 } 1971 switch (kind) { 1972 case TEE_SINK_NEW: { 1973 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1974 size_t numCounterOffers = 0; 1975 const NBAIO_Format offers[1] = {format}; 1976 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1977 ALOG_ASSERT(index == 0); 1978 PipeReader *pipeReader = new PipeReader(*pipe); 1979 numCounterOffers = 0; 1980 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1981 ALOG_ASSERT(index == 0); 1982 mRecordTeeSink = pipe; 1983 mRecordTeeSource = pipeReader; 1984 teeSink = pipe; 1985 } 1986 break; 1987 case TEE_SINK_OLD: 1988 teeSink = mRecordTeeSink; 1989 break; 1990 case TEE_SINK_NO: 1991 default: 1992 break; 1993 } 1994#endif 1995 1996 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 1997 1998 // Start record thread 1999 // RecordThread requires both input and output device indication to forward to audio 2000 // pre processing modules 2001 sp<RecordThread> thread = new RecordThread(this, 2002 inputStream, 2003 *input, 2004 primaryOutputDevice_l(), 2005 device 2006#ifdef TEE_SINK 2007 , teeSink 2008#endif 2009 ); 2010 mRecordThreads.add(*input, thread); 2011 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2012 return thread; 2013 } 2014 2015 *input = AUDIO_IO_HANDLE_NONE; 2016 return 0; 2017} 2018 2019status_t AudioFlinger::closeInput(audio_io_handle_t input) 2020{ 2021 return closeInput_nonvirtual(input); 2022} 2023 2024status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2025{ 2026 // keep strong reference on the record thread so that 2027 // it is not destroyed while exit() is executed 2028 sp<RecordThread> thread; 2029 { 2030 Mutex::Autolock _l(mLock); 2031 thread = checkRecordThread_l(input); 2032 if (thread == 0) { 2033 return BAD_VALUE; 2034 } 2035 2036 ALOGV("closeInput() %d", input); 2037 { 2038 // If we still have effect chains, it means that a client still holds a handle 2039 // on at least one effect. We must keep the chain alive in case a new record 2040 // thread is opened for a new capture on the same session 2041 Mutex::Autolock _sl(thread->mLock); 2042 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2043 for (size_t i = 0; i < effectChains.size(); i++) { 2044 putOrphanEffectChain_l(effectChains[i]); 2045 } 2046 } 2047 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 2048 mRecordThreads.removeItem(input); 2049 } 2050 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2051 // we have a different lock for notification client 2052 closeInputFinish(thread); 2053 return NO_ERROR; 2054} 2055 2056void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2057{ 2058 thread->exit(); 2059 AudioStreamIn *in = thread->clearInput(); 2060 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2061 // from now on thread->mInput is NULL 2062 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2063 delete in; 2064} 2065 2066void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2067{ 2068 mRecordThreads.removeItem(thread->mId); 2069 closeInputFinish(thread); 2070} 2071 2072status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2073{ 2074 Mutex::Autolock _l(mLock); 2075 ALOGV("invalidateStream() stream %d", stream); 2076 2077 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2078 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2079 thread->invalidateTracks(stream); 2080 } 2081 2082 return NO_ERROR; 2083} 2084 2085 2086audio_unique_id_t AudioFlinger::newAudioUniqueId() 2087{ 2088 return nextUniqueId(); 2089} 2090 2091void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2092{ 2093 Mutex::Autolock _l(mLock); 2094 pid_t caller = IPCThreadState::self()->getCallingPid(); 2095 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2096 if (pid != -1 && (caller == getpid_cached)) { 2097 caller = pid; 2098 } 2099 2100 { 2101 Mutex::Autolock _cl(mClientLock); 2102 // Ignore requests received from processes not known as notification client. The request 2103 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2104 // called from a different pid leaving a stale session reference. Also we don't know how 2105 // to clear this reference if the client process dies. 2106 if (mNotificationClients.indexOfKey(caller) < 0) { 2107 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2108 return; 2109 } 2110 } 2111 2112 size_t num = mAudioSessionRefs.size(); 2113 for (size_t i = 0; i< num; i++) { 2114 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2115 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2116 ref->mCnt++; 2117 ALOGV(" incremented refcount to %d", ref->mCnt); 2118 return; 2119 } 2120 } 2121 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2122 ALOGV(" added new entry for %d", audioSession); 2123} 2124 2125void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2126{ 2127 Mutex::Autolock _l(mLock); 2128 pid_t caller = IPCThreadState::self()->getCallingPid(); 2129 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2130 if (pid != -1 && (caller == getpid_cached)) { 2131 caller = pid; 2132 } 2133 size_t num = mAudioSessionRefs.size(); 2134 for (size_t i = 0; i< num; i++) { 2135 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2136 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2137 ref->mCnt--; 2138 ALOGV(" decremented refcount to %d", ref->mCnt); 2139 if (ref->mCnt == 0) { 2140 mAudioSessionRefs.removeAt(i); 2141 delete ref; 2142 purgeStaleEffects_l(); 2143 } 2144 return; 2145 } 2146 } 2147 // If the caller is mediaserver it is likely that the session being released was acquired 2148 // on behalf of a process not in notification clients and we ignore the warning. 2149 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2150} 2151 2152void AudioFlinger::purgeStaleEffects_l() { 2153 2154 ALOGV("purging stale effects"); 2155 2156 Vector< sp<EffectChain> > chains; 2157 2158 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2159 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2160 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2161 sp<EffectChain> ec = t->mEffectChains[j]; 2162 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2163 chains.push(ec); 2164 } 2165 } 2166 } 2167 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2168 sp<RecordThread> t = mRecordThreads.valueAt(i); 2169 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2170 sp<EffectChain> ec = t->mEffectChains[j]; 2171 chains.push(ec); 2172 } 2173 } 2174 2175 for (size_t i = 0; i < chains.size(); i++) { 2176 sp<EffectChain> ec = chains[i]; 2177 int sessionid = ec->sessionId(); 2178 sp<ThreadBase> t = ec->mThread.promote(); 2179 if (t == 0) { 2180 continue; 2181 } 2182 size_t numsessionrefs = mAudioSessionRefs.size(); 2183 bool found = false; 2184 for (size_t k = 0; k < numsessionrefs; k++) { 2185 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2186 if (ref->mSessionid == sessionid) { 2187 ALOGV(" session %d still exists for %d with %d refs", 2188 sessionid, ref->mPid, ref->mCnt); 2189 found = true; 2190 break; 2191 } 2192 } 2193 if (!found) { 2194 Mutex::Autolock _l(t->mLock); 2195 // remove all effects from the chain 2196 while (ec->mEffects.size()) { 2197 sp<EffectModule> effect = ec->mEffects[0]; 2198 effect->unPin(); 2199 t->removeEffect_l(effect); 2200 if (effect->purgeHandles()) { 2201 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2202 } 2203 AudioSystem::unregisterEffect(effect->id()); 2204 } 2205 } 2206 } 2207 return; 2208} 2209 2210// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2211AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2212{ 2213 return mPlaybackThreads.valueFor(output).get(); 2214} 2215 2216// checkMixerThread_l() must be called with AudioFlinger::mLock held 2217AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2218{ 2219 PlaybackThread *thread = checkPlaybackThread_l(output); 2220 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2221} 2222 2223// checkRecordThread_l() must be called with AudioFlinger::mLock held 2224AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2225{ 2226 return mRecordThreads.valueFor(input).get(); 2227} 2228 2229uint32_t AudioFlinger::nextUniqueId() 2230{ 2231 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2232} 2233 2234AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2235{ 2236 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2237 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2238 AudioStreamOut *output = thread->getOutput(); 2239 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2240 return thread; 2241 } 2242 } 2243 return NULL; 2244} 2245 2246audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2247{ 2248 PlaybackThread *thread = primaryPlaybackThread_l(); 2249 2250 if (thread == NULL) { 2251 return 0; 2252 } 2253 2254 return thread->outDevice(); 2255} 2256 2257sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2258 int triggerSession, 2259 int listenerSession, 2260 sync_event_callback_t callBack, 2261 wp<RefBase> cookie) 2262{ 2263 Mutex::Autolock _l(mLock); 2264 2265 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2266 status_t playStatus = NAME_NOT_FOUND; 2267 status_t recStatus = NAME_NOT_FOUND; 2268 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2269 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2270 if (playStatus == NO_ERROR) { 2271 return event; 2272 } 2273 } 2274 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2275 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2276 if (recStatus == NO_ERROR) { 2277 return event; 2278 } 2279 } 2280 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2281 mPendingSyncEvents.add(event); 2282 } else { 2283 ALOGV("createSyncEvent() invalid event %d", event->type()); 2284 event.clear(); 2285 } 2286 return event; 2287} 2288 2289// ---------------------------------------------------------------------------- 2290// Effect management 2291// ---------------------------------------------------------------------------- 2292 2293 2294status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2295{ 2296 Mutex::Autolock _l(mLock); 2297 return EffectQueryNumberEffects(numEffects); 2298} 2299 2300status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2301{ 2302 Mutex::Autolock _l(mLock); 2303 return EffectQueryEffect(index, descriptor); 2304} 2305 2306status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2307 effect_descriptor_t *descriptor) const 2308{ 2309 Mutex::Autolock _l(mLock); 2310 return EffectGetDescriptor(pUuid, descriptor); 2311} 2312 2313 2314sp<IEffect> AudioFlinger::createEffect( 2315 effect_descriptor_t *pDesc, 2316 const sp<IEffectClient>& effectClient, 2317 int32_t priority, 2318 audio_io_handle_t io, 2319 int sessionId, 2320 status_t *status, 2321 int *id, 2322 int *enabled) 2323{ 2324 status_t lStatus = NO_ERROR; 2325 sp<EffectHandle> handle; 2326 effect_descriptor_t desc; 2327 2328 pid_t pid = IPCThreadState::self()->getCallingPid(); 2329 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2330 pid, effectClient.get(), priority, sessionId, io); 2331 2332 if (pDesc == NULL) { 2333 lStatus = BAD_VALUE; 2334 goto Exit; 2335 } 2336 2337 // check audio settings permission for global effects 2338 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2339 lStatus = PERMISSION_DENIED; 2340 goto Exit; 2341 } 2342 2343 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2344 // that can only be created by audio policy manager (running in same process) 2345 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2346 lStatus = PERMISSION_DENIED; 2347 goto Exit; 2348 } 2349 2350 { 2351 if (!EffectIsNullUuid(&pDesc->uuid)) { 2352 // if uuid is specified, request effect descriptor 2353 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2354 if (lStatus < 0) { 2355 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2356 goto Exit; 2357 } 2358 } else { 2359 // if uuid is not specified, look for an available implementation 2360 // of the required type in effect factory 2361 if (EffectIsNullUuid(&pDesc->type)) { 2362 ALOGW("createEffect() no effect type"); 2363 lStatus = BAD_VALUE; 2364 goto Exit; 2365 } 2366 uint32_t numEffects = 0; 2367 effect_descriptor_t d; 2368 d.flags = 0; // prevent compiler warning 2369 bool found = false; 2370 2371 lStatus = EffectQueryNumberEffects(&numEffects); 2372 if (lStatus < 0) { 2373 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2374 goto Exit; 2375 } 2376 for (uint32_t i = 0; i < numEffects; i++) { 2377 lStatus = EffectQueryEffect(i, &desc); 2378 if (lStatus < 0) { 2379 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2380 continue; 2381 } 2382 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2383 // If matching type found save effect descriptor. If the session is 2384 // 0 and the effect is not auxiliary, continue enumeration in case 2385 // an auxiliary version of this effect type is available 2386 found = true; 2387 d = desc; 2388 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2389 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2390 break; 2391 } 2392 } 2393 } 2394 if (!found) { 2395 lStatus = BAD_VALUE; 2396 ALOGW("createEffect() effect not found"); 2397 goto Exit; 2398 } 2399 // For same effect type, chose auxiliary version over insert version if 2400 // connect to output mix (Compliance to OpenSL ES) 2401 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2402 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2403 desc = d; 2404 } 2405 } 2406 2407 // Do not allow auxiliary effects on a session different from 0 (output mix) 2408 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2409 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2410 lStatus = INVALID_OPERATION; 2411 goto Exit; 2412 } 2413 2414 // check recording permission for visualizer 2415 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2416 !recordingAllowed()) { 2417 lStatus = PERMISSION_DENIED; 2418 goto Exit; 2419 } 2420 2421 // return effect descriptor 2422 *pDesc = desc; 2423 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2424 // if the output returned by getOutputForEffect() is removed before we lock the 2425 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2426 // and we will exit safely 2427 io = AudioSystem::getOutputForEffect(&desc); 2428 ALOGV("createEffect got output %d", io); 2429 } 2430 2431 Mutex::Autolock _l(mLock); 2432 2433 // If output is not specified try to find a matching audio session ID in one of the 2434 // output threads. 2435 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2436 // because of code checking output when entering the function. 2437 // Note: io is never 0 when creating an effect on an input 2438 if (io == AUDIO_IO_HANDLE_NONE) { 2439 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2440 // output must be specified by AudioPolicyManager when using session 2441 // AUDIO_SESSION_OUTPUT_STAGE 2442 lStatus = BAD_VALUE; 2443 goto Exit; 2444 } 2445 // look for the thread where the specified audio session is present 2446 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2447 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2448 io = mPlaybackThreads.keyAt(i); 2449 break; 2450 } 2451 } 2452 if (io == 0) { 2453 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2454 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2455 io = mRecordThreads.keyAt(i); 2456 break; 2457 } 2458 } 2459 } 2460 // If no output thread contains the requested session ID, default to 2461 // first output. The effect chain will be moved to the correct output 2462 // thread when a track with the same session ID is created 2463 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2464 io = mPlaybackThreads.keyAt(0); 2465 } 2466 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2467 } 2468 ThreadBase *thread = checkRecordThread_l(io); 2469 if (thread == NULL) { 2470 thread = checkPlaybackThread_l(io); 2471 if (thread == NULL) { 2472 ALOGE("createEffect() unknown output thread"); 2473 lStatus = BAD_VALUE; 2474 goto Exit; 2475 } 2476 } else { 2477 // Check if one effect chain was awaiting for an effect to be created on this 2478 // session and used it instead of creating a new one. 2479 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2480 if (chain != 0) { 2481 thread->addEffectChain_l(chain); 2482 } 2483 } 2484 2485 sp<Client> client = registerPid(pid); 2486 2487 // create effect on selected output thread 2488 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2489 &desc, enabled, &lStatus); 2490 if (handle != 0 && id != NULL) { 2491 *id = handle->id(); 2492 } 2493 if (handle == 0) { 2494 // remove local strong reference to Client with mClientLock held 2495 Mutex::Autolock _cl(mClientLock); 2496 client.clear(); 2497 } 2498 } 2499 2500Exit: 2501 *status = lStatus; 2502 return handle; 2503} 2504 2505status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2506 audio_io_handle_t dstOutput) 2507{ 2508 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2509 sessionId, srcOutput, dstOutput); 2510 Mutex::Autolock _l(mLock); 2511 if (srcOutput == dstOutput) { 2512 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2513 return NO_ERROR; 2514 } 2515 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2516 if (srcThread == NULL) { 2517 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2518 return BAD_VALUE; 2519 } 2520 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2521 if (dstThread == NULL) { 2522 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2523 return BAD_VALUE; 2524 } 2525 2526 Mutex::Autolock _dl(dstThread->mLock); 2527 Mutex::Autolock _sl(srcThread->mLock); 2528 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2529} 2530 2531// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2532status_t AudioFlinger::moveEffectChain_l(int sessionId, 2533 AudioFlinger::PlaybackThread *srcThread, 2534 AudioFlinger::PlaybackThread *dstThread, 2535 bool reRegister) 2536{ 2537 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2538 sessionId, srcThread, dstThread); 2539 2540 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2541 if (chain == 0) { 2542 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2543 sessionId, srcThread); 2544 return INVALID_OPERATION; 2545 } 2546 2547 // Check whether the destination thread has a channel count of FCC_2, which is 2548 // currently required for (most) effects. Prevent moving the effect chain here rather 2549 // than disabling the addEffect_l() call in dstThread below. 2550 if (dstThread->mChannelCount != FCC_2) { 2551 ALOGW("moveEffectChain_l() effect chain failed because" 2552 " destination thread %p channel count(%u) != %u", 2553 dstThread, dstThread->mChannelCount, FCC_2); 2554 return INVALID_OPERATION; 2555 } 2556 2557 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2558 // so that a new chain is created with correct parameters when first effect is added. This is 2559 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2560 // removed. 2561 srcThread->removeEffectChain_l(chain); 2562 2563 // transfer all effects one by one so that new effect chain is created on new thread with 2564 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2565 sp<EffectChain> dstChain; 2566 uint32_t strategy = 0; // prevent compiler warning 2567 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2568 Vector< sp<EffectModule> > removed; 2569 status_t status = NO_ERROR; 2570 while (effect != 0) { 2571 srcThread->removeEffect_l(effect); 2572 removed.add(effect); 2573 status = dstThread->addEffect_l(effect); 2574 if (status != NO_ERROR) { 2575 break; 2576 } 2577 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2578 if (effect->state() == EffectModule::ACTIVE || 2579 effect->state() == EffectModule::STOPPING) { 2580 effect->start(); 2581 } 2582 // if the move request is not received from audio policy manager, the effect must be 2583 // re-registered with the new strategy and output 2584 if (dstChain == 0) { 2585 dstChain = effect->chain().promote(); 2586 if (dstChain == 0) { 2587 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2588 status = NO_INIT; 2589 break; 2590 } 2591 strategy = dstChain->strategy(); 2592 } 2593 if (reRegister) { 2594 AudioSystem::unregisterEffect(effect->id()); 2595 AudioSystem::registerEffect(&effect->desc(), 2596 dstThread->id(), 2597 strategy, 2598 sessionId, 2599 effect->id()); 2600 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2601 } 2602 effect = chain->getEffectFromId_l(0); 2603 } 2604 2605 if (status != NO_ERROR) { 2606 for (size_t i = 0; i < removed.size(); i++) { 2607 srcThread->addEffect_l(removed[i]); 2608 if (dstChain != 0 && reRegister) { 2609 AudioSystem::unregisterEffect(removed[i]->id()); 2610 AudioSystem::registerEffect(&removed[i]->desc(), 2611 srcThread->id(), 2612 strategy, 2613 sessionId, 2614 removed[i]->id()); 2615 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2616 } 2617 } 2618 } 2619 2620 return status; 2621} 2622 2623bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2624{ 2625 if (mGlobalEffectEnableTime != 0 && 2626 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2627 return true; 2628 } 2629 2630 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2631 sp<EffectChain> ec = 2632 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2633 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2634 return true; 2635 } 2636 } 2637 return false; 2638} 2639 2640void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2641{ 2642 Mutex::Autolock _l(mLock); 2643 2644 mGlobalEffectEnableTime = systemTime(); 2645 2646 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2647 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2648 if (t->mType == ThreadBase::OFFLOAD) { 2649 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2650 } 2651 } 2652 2653} 2654 2655status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2656{ 2657 audio_session_t session = (audio_session_t)chain->sessionId(); 2658 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2659 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2660 if (index >= 0) { 2661 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2662 return ALREADY_EXISTS; 2663 } 2664 mOrphanEffectChains.add(session, chain); 2665 return NO_ERROR; 2666} 2667 2668sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2669{ 2670 sp<EffectChain> chain; 2671 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2672 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2673 if (index >= 0) { 2674 chain = mOrphanEffectChains.valueAt(index); 2675 mOrphanEffectChains.removeItemsAt(index); 2676 } 2677 return chain; 2678} 2679 2680bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2681{ 2682 Mutex::Autolock _l(mLock); 2683 audio_session_t session = (audio_session_t)effect->sessionId(); 2684 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2685 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2686 if (index >= 0) { 2687 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2688 if (chain->removeEffect_l(effect) == 0) { 2689 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2690 mOrphanEffectChains.removeItemsAt(index); 2691 } 2692 return true; 2693 } 2694 return false; 2695} 2696 2697 2698struct Entry { 2699#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2700 char mName[MAX_NAME]; 2701}; 2702 2703int comparEntry(const void *p1, const void *p2) 2704{ 2705 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2706} 2707 2708#ifdef TEE_SINK 2709void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2710{ 2711 NBAIO_Source *teeSource = source.get(); 2712 if (teeSource != NULL) { 2713 // .wav rotation 2714 // There is a benign race condition if 2 threads call this simultaneously. 2715 // They would both traverse the directory, but the result would simply be 2716 // failures at unlink() which are ignored. It's also unlikely since 2717 // normally dumpsys is only done by bugreport or from the command line. 2718 char teePath[32+256]; 2719 strcpy(teePath, "/data/misc/media"); 2720 size_t teePathLen = strlen(teePath); 2721 DIR *dir = opendir(teePath); 2722 teePath[teePathLen++] = '/'; 2723 if (dir != NULL) { 2724#define MAX_SORT 20 // number of entries to sort 2725#define MAX_KEEP 10 // number of entries to keep 2726 struct Entry entries[MAX_SORT]; 2727 size_t entryCount = 0; 2728 while (entryCount < MAX_SORT) { 2729 struct dirent de; 2730 struct dirent *result = NULL; 2731 int rc = readdir_r(dir, &de, &result); 2732 if (rc != 0) { 2733 ALOGW("readdir_r failed %d", rc); 2734 break; 2735 } 2736 if (result == NULL) { 2737 break; 2738 } 2739 if (result != &de) { 2740 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2741 break; 2742 } 2743 // ignore non .wav file entries 2744 size_t nameLen = strlen(de.d_name); 2745 if (nameLen <= 4 || nameLen >= MAX_NAME || 2746 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2747 continue; 2748 } 2749 strcpy(entries[entryCount++].mName, de.d_name); 2750 } 2751 (void) closedir(dir); 2752 if (entryCount > MAX_KEEP) { 2753 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2754 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2755 strcpy(&teePath[teePathLen], entries[i].mName); 2756 (void) unlink(teePath); 2757 } 2758 } 2759 } else { 2760 if (fd >= 0) { 2761 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2762 } 2763 } 2764 char teeTime[16]; 2765 struct timeval tv; 2766 gettimeofday(&tv, NULL); 2767 struct tm tm; 2768 localtime_r(&tv.tv_sec, &tm); 2769 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2770 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2771 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2772 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2773 if (teeFd >= 0) { 2774 // FIXME use libsndfile 2775 char wavHeader[44]; 2776 memcpy(wavHeader, 2777 "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", 2778 sizeof(wavHeader)); 2779 NBAIO_Format format = teeSource->format(); 2780 unsigned channelCount = Format_channelCount(format); 2781 uint32_t sampleRate = Format_sampleRate(format); 2782 size_t frameSize = Format_frameSize(format); 2783 wavHeader[22] = channelCount; // number of channels 2784 wavHeader[24] = sampleRate; // sample rate 2785 wavHeader[25] = sampleRate >> 8; 2786 wavHeader[32] = frameSize; // block alignment 2787 wavHeader[33] = frameSize >> 8; 2788 write(teeFd, wavHeader, sizeof(wavHeader)); 2789 size_t total = 0; 2790 bool firstRead = true; 2791#define TEE_SINK_READ 1024 // frames per I/O operation 2792 void *buffer = malloc(TEE_SINK_READ * frameSize); 2793 for (;;) { 2794 size_t count = TEE_SINK_READ; 2795 ssize_t actual = teeSource->read(buffer, count, 2796 AudioBufferProvider::kInvalidPTS); 2797 bool wasFirstRead = firstRead; 2798 firstRead = false; 2799 if (actual <= 0) { 2800 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2801 continue; 2802 } 2803 break; 2804 } 2805 ALOG_ASSERT(actual <= (ssize_t)count); 2806 write(teeFd, buffer, actual * frameSize); 2807 total += actual; 2808 } 2809 free(buffer); 2810 lseek(teeFd, (off_t) 4, SEEK_SET); 2811 uint32_t temp = 44 + total * frameSize - 8; 2812 // FIXME not big-endian safe 2813 write(teeFd, &temp, sizeof(temp)); 2814 lseek(teeFd, (off_t) 40, SEEK_SET); 2815 temp = total * frameSize; 2816 // FIXME not big-endian safe 2817 write(teeFd, &temp, sizeof(temp)); 2818 close(teeFd); 2819 if (fd >= 0) { 2820 dprintf(fd, "tee copied to %s\n", teePath); 2821 } 2822 } else { 2823 if (fd >= 0) { 2824 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2825 } 2826 } 2827 } 2828} 2829#endif 2830 2831// ---------------------------------------------------------------------------- 2832 2833status_t AudioFlinger::onTransact( 2834 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2835{ 2836 return BnAudioFlinger::onTransact(code, data, reply, flags); 2837} 2838 2839}; // namespace android 2840