AudioFlinger.cpp revision fd4c14883b268a0bc5514da135fe6b7d1ce2071b
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/EffectsFactoryApi.h> 49#include <audio_effects/effect_visualizer.h> 50#include <audio_effects/effect_ns.h> 51#include <audio_effects/effect_aec.h> 52 53#include <audio_utils/primitives.h> 54 55#include <powermanager/PowerManager.h> 56 57#include <common_time/cc_helper.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <private/android_filesystem_config.h> 65 66// ---------------------------------------------------------------------------- 67 68// Note: the following macro is used for extremely verbose logging message. In 69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70// 0; but one side effect of this is to turn all LOGV's as well. Some messages 71// are so verbose that we want to suppress them even when we have ALOG_ASSERT 72// turned on. Do not uncomment the #def below unless you really know what you 73// are doing and want to see all of the extremely verbose messages. 74//#define VERY_VERY_VERBOSE_LOGGING 75#ifdef VERY_VERY_VERBOSE_LOGGING 76#define ALOGVV ALOGV 77#else 78#define ALOGVV(a...) do { } while(0) 79#endif 80 81namespace android { 82 83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85static const char kClientLockedString[] = "Client lock is taken\n"; 86 87 88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 89 90uint32_t AudioFlinger::mScreenState; 91 92#ifdef TEE_SINK 93bool AudioFlinger::mTeeSinkInputEnabled = false; 94bool AudioFlinger::mTeeSinkOutputEnabled = false; 95bool AudioFlinger::mTeeSinkTrackEnabled = false; 96 97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 100#endif 101 102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 103// we define a minimum time during which a global effect is considered enabled. 104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 105 106// ---------------------------------------------------------------------------- 107 108const char *formatToString(audio_format_t format) { 109 switch (format & AUDIO_FORMAT_MAIN_MASK) { 110 case AUDIO_FORMAT_PCM: 111 switch (format) { 112 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 113 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 114 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 115 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 116 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 117 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 118 default: 119 break; 120 } 121 break; 122 case AUDIO_FORMAT_MP3: return "mp3"; 123 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 124 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 125 case AUDIO_FORMAT_AAC: return "aac"; 126 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 127 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 128 case AUDIO_FORMAT_VORBIS: return "vorbis"; 129 case AUDIO_FORMAT_OPUS: return "opus"; 130 case AUDIO_FORMAT_AC3: return "ac-3"; 131 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 132 default: 133 break; 134 } 135 return "unknown"; 136} 137 138static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 139{ 140 const hw_module_t *mod; 141 int rc; 142 143 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 144 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 145 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 146 if (rc) { 147 goto out; 148 } 149 rc = audio_hw_device_open(mod, dev); 150 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 156 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 157 rc = BAD_VALUE; 158 goto out; 159 } 160 return 0; 161 162out: 163 *dev = NULL; 164 return rc; 165} 166 167// ---------------------------------------------------------------------------- 168 169AudioFlinger::AudioFlinger() 170 : BnAudioFlinger(), 171 mPrimaryHardwareDev(NULL), 172 mAudioHwDevs(NULL), 173 mHardwareStatus(AUDIO_HW_IDLE), 174 mMasterVolume(1.0f), 175 mMasterMute(false), 176 mNextUniqueId(1), 177 mMode(AUDIO_MODE_INVALID), 178 mBtNrecIsOff(false), 179 mIsLowRamDevice(true), 180 mIsDeviceTypeKnown(false), 181 mGlobalEffectEnableTime(0), 182 mPrimaryOutputSampleRate(0) 183{ 184 getpid_cached = getpid(); 185 char value[PROPERTY_VALUE_MAX]; 186 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 187 if (doLog) { 188 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 189 } 190 191#ifdef TEE_SINK 192 (void) property_get("ro.debuggable", value, "0"); 193 int debuggable = atoi(value); 194 int teeEnabled = 0; 195 if (debuggable) { 196 (void) property_get("af.tee", value, "0"); 197 teeEnabled = atoi(value); 198 } 199 // FIXME symbolic constants here 200 if (teeEnabled & 1) { 201 mTeeSinkInputEnabled = true; 202 } 203 if (teeEnabled & 2) { 204 mTeeSinkOutputEnabled = true; 205 } 206 if (teeEnabled & 4) { 207 mTeeSinkTrackEnabled = true; 208 } 209#endif 210} 211 212void AudioFlinger::onFirstRef() 213{ 214 int rc = 0; 215 216 Mutex::Autolock _l(mLock); 217 218 /* TODO: move all this work into an Init() function */ 219 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 220 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 221 uint32_t int_val; 222 if (1 == sscanf(val_str, "%u", &int_val)) { 223 mStandbyTimeInNsecs = milliseconds(int_val); 224 ALOGI("Using %u mSec as standby time.", int_val); 225 } else { 226 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 227 ALOGI("Using default %u mSec as standby time.", 228 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 229 } 230 } 231 232 mPatchPanel = new PatchPanel(this); 233 234 mMode = AUDIO_MODE_NORMAL; 235} 236 237AudioFlinger::~AudioFlinger() 238{ 239 while (!mRecordThreads.isEmpty()) { 240 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 241 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 242 } 243 while (!mPlaybackThreads.isEmpty()) { 244 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 245 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 246 } 247 248 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 249 // no mHardwareLock needed, as there are no other references to this 250 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 251 delete mAudioHwDevs.valueAt(i); 252 } 253 254 // Tell media.log service about any old writers that still need to be unregistered 255 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 256 if (binder != 0) { 257 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 258 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 259 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 260 mUnregisteredWriters.pop(); 261 mediaLogService->unregisterWriter(iMemory); 262 } 263 } 264 265} 266 267static const char * const audio_interfaces[] = { 268 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 269 AUDIO_HARDWARE_MODULE_ID_A2DP, 270 AUDIO_HARDWARE_MODULE_ID_USB, 271}; 272#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 273 274AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 275 audio_module_handle_t module, 276 audio_devices_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 // then try to find a module supporting the requested device. 286 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 287 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 288 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 289 if ((dev->get_supported_devices != NULL) && 290 (dev->get_supported_devices(dev) & devices) == devices) 291 return audioHwDevice; 292 } 293 } else { 294 // check a match for the requested module handle 295 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 296 if (audioHwDevice != NULL) { 297 return audioHwDevice; 298 } 299 } 300 301 return NULL; 302} 303 304void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Notification Clients:\n"); 320 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 321 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 322 result.append(buffer); 323 } 324 325 result.append("Global session refs:\n"); 326 result.append(" session pid count\n"); 327 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 328 AudioSessionRef *r = mAudioSessionRefs[i]; 329 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 330 result.append(buffer); 331 } 332 write(fd, result.string(), result.size()); 333} 334 335 336void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 337{ 338 const size_t SIZE = 256; 339 char buffer[SIZE]; 340 String8 result; 341 hardware_call_state hardwareStatus = mHardwareStatus; 342 343 snprintf(buffer, SIZE, "Hardware status: %d\n" 344 "Standby Time mSec: %u\n", 345 hardwareStatus, 346 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 347 result.append(buffer); 348 write(fd, result.string(), result.size()); 349} 350 351void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 352{ 353 const size_t SIZE = 256; 354 char buffer[SIZE]; 355 String8 result; 356 snprintf(buffer, SIZE, "Permission Denial: " 357 "can't dump AudioFlinger from pid=%d, uid=%d\n", 358 IPCThreadState::self()->getCallingPid(), 359 IPCThreadState::self()->getCallingUid()); 360 result.append(buffer); 361 write(fd, result.string(), result.size()); 362} 363 364bool AudioFlinger::dumpTryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = dumpTryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = dumpTryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 bool clientLocked = dumpTryLock(mClientLock); 400 if (!clientLocked) { 401 String8 result(kClientLockedString); 402 write(fd, result.string(), result.size()); 403 } 404 dumpClients(fd, args); 405 if (clientLocked) { 406 mClientLock.unlock(); 407 } 408 409 dumpInternals(fd, args); 410 411 // dump playback threads 412 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 413 mPlaybackThreads.valueAt(i)->dump(fd, args); 414 } 415 416 // dump record threads 417 for (size_t i = 0; i < mRecordThreads.size(); i++) { 418 mRecordThreads.valueAt(i)->dump(fd, args); 419 } 420 421 // dump orphan effect chains 422 if (mOrphanEffectChains.size() != 0) { 423 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 424 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 425 mOrphanEffectChains.valueAt(i)->dump(fd, args); 426 } 427 } 428 // dump all hardware devs 429 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 430 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 431 dev->dump(dev, fd); 432 } 433 434#ifdef TEE_SINK 435 // dump the serially shared record tee sink 436 if (mRecordTeeSource != 0) { 437 dumpTee(fd, mRecordTeeSource); 438 } 439#endif 440 441 if (locked) { 442 mLock.unlock(); 443 } 444 445 // append a copy of media.log here by forwarding fd to it, but don't attempt 446 // to lookup the service if it's not running, as it will block for a second 447 if (mLogMemoryDealer != 0) { 448 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 449 if (binder != 0) { 450 dprintf(fd, "\nmedia.log:\n"); 451 Vector<String16> args; 452 binder->dump(fd, args); 453 } 454 } 455 } 456 return NO_ERROR; 457} 458 459sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 460{ 461 Mutex::Autolock _cl(mClientLock); 462 // If pid is already in the mClients wp<> map, then use that entry 463 // (for which promote() is always != 0), otherwise create a new entry and Client. 464 sp<Client> client = mClients.valueFor(pid).promote(); 465 if (client == 0) { 466 client = new Client(this, pid); 467 mClients.add(pid, client); 468 } 469 470 return client; 471} 472 473sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 474{ 475 // If there is no memory allocated for logs, return a dummy writer that does nothing 476 if (mLogMemoryDealer == 0) { 477 return new NBLog::Writer(); 478 } 479 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 480 // Similarly if we can't contact the media.log service, also return a dummy writer 481 if (binder == 0) { 482 return new NBLog::Writer(); 483 } 484 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 485 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 486 // If allocation fails, consult the vector of previously unregistered writers 487 // and garbage-collect one or more them until an allocation succeeds 488 if (shared == 0) { 489 Mutex::Autolock _l(mUnregisteredWritersLock); 490 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 491 { 492 // Pick the oldest stale writer to garbage-collect 493 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 494 mUnregisteredWriters.removeAt(0); 495 mediaLogService->unregisterWriter(iMemory); 496 // Now the media.log remote reference to IMemory is gone. When our last local 497 // reference to IMemory also drops to zero at end of this block, 498 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 499 } 500 // Re-attempt the allocation 501 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 502 if (shared != 0) { 503 goto success; 504 } 505 } 506 // Even after garbage-collecting all old writers, there is still not enough memory, 507 // so return a dummy writer 508 return new NBLog::Writer(); 509 } 510success: 511 mediaLogService->registerWriter(shared, size, name); 512 return new NBLog::Writer(size, shared); 513} 514 515void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 516{ 517 if (writer == 0) { 518 return; 519 } 520 sp<IMemory> iMemory(writer->getIMemory()); 521 if (iMemory == 0) { 522 return; 523 } 524 // Rather than removing the writer immediately, append it to a queue of old writers to 525 // be garbage-collected later. This allows us to continue to view old logs for a while. 526 Mutex::Autolock _l(mUnregisteredWritersLock); 527 mUnregisteredWriters.push(writer); 528} 529 530// IAudioFlinger interface 531 532 533sp<IAudioTrack> AudioFlinger::createTrack( 534 audio_stream_type_t streamType, 535 uint32_t sampleRate, 536 audio_format_t format, 537 audio_channel_mask_t channelMask, 538 size_t *frameCount, 539 IAudioFlinger::track_flags_t *flags, 540 const sp<IMemory>& sharedBuffer, 541 audio_io_handle_t output, 542 pid_t tid, 543 int *sessionId, 544 int clientUid, 545 status_t *status) 546{ 547 sp<PlaybackThread::Track> track; 548 sp<TrackHandle> trackHandle; 549 sp<Client> client; 550 status_t lStatus; 551 int lSessionId; 552 553 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 554 // but if someone uses binder directly they could bypass that and cause us to crash 555 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 556 ALOGE("createTrack() invalid stream type %d", streamType); 557 lStatus = BAD_VALUE; 558 goto Exit; 559 } 560 561 // further sample rate checks are performed by createTrack_l() depending on the thread type 562 if (sampleRate == 0) { 563 ALOGE("createTrack() invalid sample rate %u", sampleRate); 564 lStatus = BAD_VALUE; 565 goto Exit; 566 } 567 568 // further channel mask checks are performed by createTrack_l() depending on the thread type 569 if (!audio_is_output_channel(channelMask)) { 570 ALOGE("createTrack() invalid channel mask %#x", channelMask); 571 lStatus = BAD_VALUE; 572 goto Exit; 573 } 574 575 // further format checks are performed by createTrack_l() depending on the thread type 576 if (!audio_is_valid_format(format)) { 577 ALOGE("createTrack() invalid format %#x", format); 578 lStatus = BAD_VALUE; 579 goto Exit; 580 } 581 582 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 583 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 584 lStatus = BAD_VALUE; 585 goto Exit; 586 } 587 588 { 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGE("no playback thread found for output handle %d", output); 593 lStatus = BAD_VALUE; 594 goto Exit; 595 } 596 597 pid_t pid = IPCThreadState::self()->getCallingPid(); 598 client = registerPid(pid); 599 600 PlaybackThread *effectThread = NULL; 601 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 602 lSessionId = *sessionId; 603 // check if an effect chain with the same session ID is present on another 604 // output thread and move it here. 605 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 606 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 607 if (mPlaybackThreads.keyAt(i) != output) { 608 uint32_t sessions = t->hasAudioSession(lSessionId); 609 if (sessions & PlaybackThread::EFFECT_SESSION) { 610 effectThread = t.get(); 611 break; 612 } 613 } 614 } 615 } else { 616 // if no audio session id is provided, create one here 617 lSessionId = nextUniqueId(); 618 if (sessionId != NULL) { 619 *sessionId = lSessionId; 620 } 621 } 622 ALOGV("createTrack() lSessionId: %d", lSessionId); 623 624 track = thread->createTrack_l(client, streamType, sampleRate, format, 625 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 626 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 627 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 628 629 // move effect chain to this output thread if an effect on same session was waiting 630 // for a track to be created 631 if (lStatus == NO_ERROR && effectThread != NULL) { 632 // no risk of deadlock because AudioFlinger::mLock is held 633 Mutex::Autolock _dl(thread->mLock); 634 Mutex::Autolock _sl(effectThread->mLock); 635 moveEffectChain_l(lSessionId, effectThread, thread, true); 636 } 637 638 // Look for sync events awaiting for a session to be used. 639 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 640 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 641 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 642 if (lStatus == NO_ERROR) { 643 (void) track->setSyncEvent(mPendingSyncEvents[i]); 644 } else { 645 mPendingSyncEvents[i]->cancel(); 646 } 647 mPendingSyncEvents.removeAt(i); 648 i--; 649 } 650 } 651 } 652 653 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 654 } 655 656 if (lStatus != NO_ERROR) { 657 // remove local strong reference to Client before deleting the Track so that the 658 // Client destructor is called by the TrackBase destructor with mClientLock held 659 // Don't hold mClientLock when releasing the reference on the track as the 660 // destructor will acquire it. 661 { 662 Mutex::Autolock _cl(mClientLock); 663 client.clear(); 664 } 665 track.clear(); 666 goto Exit; 667 } 668 669 // return handle to client 670 trackHandle = new TrackHandle(track); 671 672Exit: 673 *status = lStatus; 674 return trackHandle; 675} 676 677uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 678{ 679 Mutex::Autolock _l(mLock); 680 PlaybackThread *thread = checkPlaybackThread_l(output); 681 if (thread == NULL) { 682 ALOGW("sampleRate() unknown thread %d", output); 683 return 0; 684 } 685 return thread->sampleRate(); 686} 687 688audio_format_t AudioFlinger::format(audio_io_handle_t output) const 689{ 690 Mutex::Autolock _l(mLock); 691 PlaybackThread *thread = checkPlaybackThread_l(output); 692 if (thread == NULL) { 693 ALOGW("format() unknown thread %d", output); 694 return AUDIO_FORMAT_INVALID; 695 } 696 return thread->format(); 697} 698 699size_t AudioFlinger::frameCount(audio_io_handle_t output) const 700{ 701 Mutex::Autolock _l(mLock); 702 PlaybackThread *thread = checkPlaybackThread_l(output); 703 if (thread == NULL) { 704 ALOGW("frameCount() unknown thread %d", output); 705 return 0; 706 } 707 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 708 // should examine all callers and fix them to handle smaller counts 709 return thread->frameCount(); 710} 711 712uint32_t AudioFlinger::latency(audio_io_handle_t output) const 713{ 714 Mutex::Autolock _l(mLock); 715 PlaybackThread *thread = checkPlaybackThread_l(output); 716 if (thread == NULL) { 717 ALOGW("latency(): no playback thread found for output handle %d", output); 718 return 0; 719 } 720 return thread->latency(); 721} 722 723status_t AudioFlinger::setMasterVolume(float value) 724{ 725 status_t ret = initCheck(); 726 if (ret != NO_ERROR) { 727 return ret; 728 } 729 730 // check calling permissions 731 if (!settingsAllowed()) { 732 return PERMISSION_DENIED; 733 } 734 735 Mutex::Autolock _l(mLock); 736 mMasterVolume = value; 737 738 // Set master volume in the HALs which support it. 739 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 740 AutoMutex lock(mHardwareLock); 741 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 742 743 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 744 if (dev->canSetMasterVolume()) { 745 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 746 } 747 mHardwareStatus = AUDIO_HW_IDLE; 748 } 749 750 // Now set the master volume in each playback thread. Playback threads 751 // assigned to HALs which do not have master volume support will apply 752 // master volume during the mix operation. Threads with HALs which do 753 // support master volume will simply ignore the setting. 754 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 755 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 756 757 return NO_ERROR; 758} 759 760status_t AudioFlinger::setMode(audio_mode_t mode) 761{ 762 status_t ret = initCheck(); 763 if (ret != NO_ERROR) { 764 return ret; 765 } 766 767 // check calling permissions 768 if (!settingsAllowed()) { 769 return PERMISSION_DENIED; 770 } 771 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 772 ALOGW("Illegal value: setMode(%d)", mode); 773 return BAD_VALUE; 774 } 775 776 { // scope for the lock 777 AutoMutex lock(mHardwareLock); 778 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 779 mHardwareStatus = AUDIO_HW_SET_MODE; 780 ret = dev->set_mode(dev, mode); 781 mHardwareStatus = AUDIO_HW_IDLE; 782 } 783 784 if (NO_ERROR == ret) { 785 Mutex::Autolock _l(mLock); 786 mMode = mode; 787 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setMode(mode); 789 } 790 791 return ret; 792} 793 794status_t AudioFlinger::setMicMute(bool state) 795{ 796 status_t ret = initCheck(); 797 if (ret != NO_ERROR) { 798 return ret; 799 } 800 801 // check calling permissions 802 if (!settingsAllowed()) { 803 return PERMISSION_DENIED; 804 } 805 806 AutoMutex lock(mHardwareLock); 807 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 808 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 809 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 810 status_t result = dev->set_mic_mute(dev, state); 811 if (result != NO_ERROR) { 812 ret = result; 813 } 814 } 815 mHardwareStatus = AUDIO_HW_IDLE; 816 return ret; 817} 818 819bool AudioFlinger::getMicMute() const 820{ 821 status_t ret = initCheck(); 822 if (ret != NO_ERROR) { 823 return false; 824 } 825 826 bool state = AUDIO_MODE_INVALID; 827 AutoMutex lock(mHardwareLock); 828 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 829 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 830 dev->get_mic_mute(dev, &state); 831 mHardwareStatus = AUDIO_HW_IDLE; 832 return state; 833} 834 835status_t AudioFlinger::setMasterMute(bool muted) 836{ 837 status_t ret = initCheck(); 838 if (ret != NO_ERROR) { 839 return ret; 840 } 841 842 // check calling permissions 843 if (!settingsAllowed()) { 844 return PERMISSION_DENIED; 845 } 846 847 Mutex::Autolock _l(mLock); 848 mMasterMute = muted; 849 850 // Set master mute in the HALs which support it. 851 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 852 AutoMutex lock(mHardwareLock); 853 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 854 855 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 856 if (dev->canSetMasterMute()) { 857 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 858 } 859 mHardwareStatus = AUDIO_HW_IDLE; 860 } 861 862 // Now set the master mute in each playback thread. Playback threads 863 // assigned to HALs which do not have master mute support will apply master 864 // mute during the mix operation. Threads with HALs which do support master 865 // mute will simply ignore the setting. 866 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 867 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 868 869 return NO_ERROR; 870} 871 872float AudioFlinger::masterVolume() const 873{ 874 Mutex::Autolock _l(mLock); 875 return masterVolume_l(); 876} 877 878bool AudioFlinger::masterMute() const 879{ 880 Mutex::Autolock _l(mLock); 881 return masterMute_l(); 882} 883 884float AudioFlinger::masterVolume_l() const 885{ 886 return mMasterVolume; 887} 888 889bool AudioFlinger::masterMute_l() const 890{ 891 return mMasterMute; 892} 893 894status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 895 audio_io_handle_t output) 896{ 897 // check calling permissions 898 if (!settingsAllowed()) { 899 return PERMISSION_DENIED; 900 } 901 902 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 903 ALOGE("setStreamVolume() invalid stream %d", stream); 904 return BAD_VALUE; 905 } 906 907 AutoMutex lock(mLock); 908 PlaybackThread *thread = NULL; 909 if (output != AUDIO_IO_HANDLE_NONE) { 910 thread = checkPlaybackThread_l(output); 911 if (thread == NULL) { 912 return BAD_VALUE; 913 } 914 } 915 916 mStreamTypes[stream].volume = value; 917 918 if (thread == NULL) { 919 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 920 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 921 } 922 } else { 923 thread->setStreamVolume(stream, value); 924 } 925 926 return NO_ERROR; 927} 928 929status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 930{ 931 // check calling permissions 932 if (!settingsAllowed()) { 933 return PERMISSION_DENIED; 934 } 935 936 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 937 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 938 ALOGE("setStreamMute() invalid stream %d", stream); 939 return BAD_VALUE; 940 } 941 942 AutoMutex lock(mLock); 943 mStreamTypes[stream].mute = muted; 944 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 945 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 946 947 return NO_ERROR; 948} 949 950float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 951{ 952 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 953 return 0.0f; 954 } 955 956 AutoMutex lock(mLock); 957 float volume; 958 if (output != AUDIO_IO_HANDLE_NONE) { 959 PlaybackThread *thread = checkPlaybackThread_l(output); 960 if (thread == NULL) { 961 return 0.0f; 962 } 963 volume = thread->streamVolume(stream); 964 } else { 965 volume = streamVolume_l(stream); 966 } 967 968 return volume; 969} 970 971bool AudioFlinger::streamMute(audio_stream_type_t stream) const 972{ 973 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 974 return true; 975 } 976 977 AutoMutex lock(mLock); 978 return streamMute_l(stream); 979} 980 981status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 982{ 983 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 984 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 985 986 // check calling permissions 987 if (!settingsAllowed()) { 988 return PERMISSION_DENIED; 989 } 990 991 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 992 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 993 Mutex::Autolock _l(mLock); 994 status_t final_result = NO_ERROR; 995 { 996 AutoMutex lock(mHardwareLock); 997 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 998 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 999 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1000 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1001 final_result = result ?: final_result; 1002 } 1003 mHardwareStatus = AUDIO_HW_IDLE; 1004 } 1005 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1006 AudioParameter param = AudioParameter(keyValuePairs); 1007 String8 value; 1008 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1009 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1010 if (mBtNrecIsOff != btNrecIsOff) { 1011 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1012 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1013 audio_devices_t device = thread->inDevice(); 1014 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1015 // collect all of the thread's session IDs 1016 KeyedVector<int, bool> ids = thread->sessionIds(); 1017 // suspend effects associated with those session IDs 1018 for (size_t j = 0; j < ids.size(); ++j) { 1019 int sessionId = ids.keyAt(j); 1020 thread->setEffectSuspended(FX_IID_AEC, 1021 suspend, 1022 sessionId); 1023 thread->setEffectSuspended(FX_IID_NS, 1024 suspend, 1025 sessionId); 1026 } 1027 } 1028 mBtNrecIsOff = btNrecIsOff; 1029 } 1030 } 1031 String8 screenState; 1032 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1033 bool isOff = screenState == "off"; 1034 if (isOff != (AudioFlinger::mScreenState & 1)) { 1035 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1036 } 1037 } 1038 return final_result; 1039 } 1040 1041 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1042 // and the thread is exited once the lock is released 1043 sp<ThreadBase> thread; 1044 { 1045 Mutex::Autolock _l(mLock); 1046 thread = checkPlaybackThread_l(ioHandle); 1047 if (thread == 0) { 1048 thread = checkRecordThread_l(ioHandle); 1049 } else if (thread == primaryPlaybackThread_l()) { 1050 // indicate output device change to all input threads for pre processing 1051 AudioParameter param = AudioParameter(keyValuePairs); 1052 int value; 1053 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1054 (value != 0)) { 1055 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1056 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1057 } 1058 } 1059 } 1060 } 1061 if (thread != 0) { 1062 return thread->setParameters(keyValuePairs); 1063 } 1064 return BAD_VALUE; 1065} 1066 1067String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1068{ 1069 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1070 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1071 1072 Mutex::Autolock _l(mLock); 1073 1074 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1075 String8 out_s8; 1076 1077 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1078 char *s; 1079 { 1080 AutoMutex lock(mHardwareLock); 1081 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1082 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1083 s = dev->get_parameters(dev, keys.string()); 1084 mHardwareStatus = AUDIO_HW_IDLE; 1085 } 1086 out_s8 += String8(s ? s : ""); 1087 free(s); 1088 } 1089 return out_s8; 1090 } 1091 1092 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1093 if (playbackThread != NULL) { 1094 return playbackThread->getParameters(keys); 1095 } 1096 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1097 if (recordThread != NULL) { 1098 return recordThread->getParameters(keys); 1099 } 1100 return String8(""); 1101} 1102 1103size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1104 audio_channel_mask_t channelMask) const 1105{ 1106 status_t ret = initCheck(); 1107 if (ret != NO_ERROR) { 1108 return 0; 1109 } 1110 1111 AutoMutex lock(mHardwareLock); 1112 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1113 audio_config_t config; 1114 memset(&config, 0, sizeof(config)); 1115 config.sample_rate = sampleRate; 1116 config.channel_mask = channelMask; 1117 config.format = format; 1118 1119 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1120 size_t size = dev->get_input_buffer_size(dev, &config); 1121 mHardwareStatus = AUDIO_HW_IDLE; 1122 return size; 1123} 1124 1125uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1126{ 1127 Mutex::Autolock _l(mLock); 1128 1129 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1130 if (recordThread != NULL) { 1131 return recordThread->getInputFramesLost(); 1132 } 1133 return 0; 1134} 1135 1136status_t AudioFlinger::setVoiceVolume(float value) 1137{ 1138 status_t ret = initCheck(); 1139 if (ret != NO_ERROR) { 1140 return ret; 1141 } 1142 1143 // check calling permissions 1144 if (!settingsAllowed()) { 1145 return PERMISSION_DENIED; 1146 } 1147 1148 AutoMutex lock(mHardwareLock); 1149 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1150 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1151 ret = dev->set_voice_volume(dev, value); 1152 mHardwareStatus = AUDIO_HW_IDLE; 1153 1154 return ret; 1155} 1156 1157status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1158 audio_io_handle_t output) const 1159{ 1160 status_t status; 1161 1162 Mutex::Autolock _l(mLock); 1163 1164 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1165 if (playbackThread != NULL) { 1166 return playbackThread->getRenderPosition(halFrames, dspFrames); 1167 } 1168 1169 return BAD_VALUE; 1170} 1171 1172void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1173{ 1174 Mutex::Autolock _l(mLock); 1175 if (client == 0) { 1176 return; 1177 } 1178 bool clientAdded = false; 1179 { 1180 Mutex::Autolock _cl(mClientLock); 1181 1182 pid_t pid = IPCThreadState::self()->getCallingPid(); 1183 if (mNotificationClients.indexOfKey(pid) < 0) { 1184 sp<NotificationClient> notificationClient = new NotificationClient(this, 1185 client, 1186 pid); 1187 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1188 1189 mNotificationClients.add(pid, notificationClient); 1190 1191 sp<IBinder> binder = client->asBinder(); 1192 binder->linkToDeath(notificationClient); 1193 clientAdded = true; 1194 } 1195 } 1196 1197 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1198 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1199 if (clientAdded) { 1200 // the config change is always sent from playback or record threads to avoid deadlock 1201 // with AudioSystem::gLock 1202 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1203 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1204 } 1205 1206 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1207 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1208 } 1209 } 1210} 1211 1212void AudioFlinger::removeNotificationClient(pid_t pid) 1213{ 1214 Mutex::Autolock _l(mLock); 1215 { 1216 Mutex::Autolock _cl(mClientLock); 1217 mNotificationClients.removeItem(pid); 1218 } 1219 1220 ALOGV("%d died, releasing its sessions", pid); 1221 size_t num = mAudioSessionRefs.size(); 1222 bool removed = false; 1223 for (size_t i = 0; i< num; ) { 1224 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1225 ALOGV(" pid %d @ %d", ref->mPid, i); 1226 if (ref->mPid == pid) { 1227 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1228 mAudioSessionRefs.removeAt(i); 1229 delete ref; 1230 removed = true; 1231 num--; 1232 } else { 1233 i++; 1234 } 1235 } 1236 if (removed) { 1237 purgeStaleEffects_l(); 1238 } 1239} 1240 1241void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1242{ 1243 Mutex::Autolock _l(mClientLock); 1244 size_t size = mNotificationClients.size(); 1245 for (size_t i = 0; i < size; i++) { 1246 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1247 ioHandle, 1248 param2); 1249 } 1250} 1251 1252// removeClient_l() must be called with AudioFlinger::mClientLock held 1253void AudioFlinger::removeClient_l(pid_t pid) 1254{ 1255 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1256 IPCThreadState::self()->getCallingPid()); 1257 mClients.removeItem(pid); 1258} 1259 1260// getEffectThread_l() must be called with AudioFlinger::mLock held 1261sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1262{ 1263 sp<PlaybackThread> thread; 1264 1265 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1266 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1267 ALOG_ASSERT(thread == 0); 1268 thread = mPlaybackThreads.valueAt(i); 1269 } 1270 } 1271 1272 return thread; 1273} 1274 1275 1276 1277// ---------------------------------------------------------------------------- 1278 1279AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1280 : RefBase(), 1281 mAudioFlinger(audioFlinger), 1282 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1283 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1284 mPid(pid), 1285 mTimedTrackCount(0) 1286{ 1287 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1288} 1289 1290// Client destructor must be called with AudioFlinger::mClientLock held 1291AudioFlinger::Client::~Client() 1292{ 1293 mAudioFlinger->removeClient_l(mPid); 1294} 1295 1296sp<MemoryDealer> AudioFlinger::Client::heap() const 1297{ 1298 return mMemoryDealer; 1299} 1300 1301// Reserve one of the limited slots for a timed audio track associated 1302// with this client 1303bool AudioFlinger::Client::reserveTimedTrack() 1304{ 1305 const int kMaxTimedTracksPerClient = 4; 1306 1307 Mutex::Autolock _l(mTimedTrackLock); 1308 1309 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1310 ALOGW("can not create timed track - pid %d has exceeded the limit", 1311 mPid); 1312 return false; 1313 } 1314 1315 mTimedTrackCount++; 1316 return true; 1317} 1318 1319// Release a slot for a timed audio track 1320void AudioFlinger::Client::releaseTimedTrack() 1321{ 1322 Mutex::Autolock _l(mTimedTrackLock); 1323 mTimedTrackCount--; 1324} 1325 1326// ---------------------------------------------------------------------------- 1327 1328AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1329 const sp<IAudioFlingerClient>& client, 1330 pid_t pid) 1331 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1332{ 1333} 1334 1335AudioFlinger::NotificationClient::~NotificationClient() 1336{ 1337} 1338 1339void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1340{ 1341 sp<NotificationClient> keep(this); 1342 mAudioFlinger->removeNotificationClient(mPid); 1343} 1344 1345 1346// ---------------------------------------------------------------------------- 1347 1348static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1349 return audio_is_remote_submix_device(inDevice); 1350} 1351 1352sp<IAudioRecord> AudioFlinger::openRecord( 1353 audio_io_handle_t input, 1354 uint32_t sampleRate, 1355 audio_format_t format, 1356 audio_channel_mask_t channelMask, 1357 size_t *frameCount, 1358 IAudioFlinger::track_flags_t *flags, 1359 pid_t tid, 1360 int *sessionId, 1361 size_t *notificationFrames, 1362 sp<IMemory>& cblk, 1363 sp<IMemory>& buffers, 1364 status_t *status) 1365{ 1366 sp<RecordThread::RecordTrack> recordTrack; 1367 sp<RecordHandle> recordHandle; 1368 sp<Client> client; 1369 status_t lStatus; 1370 int lSessionId; 1371 1372 cblk.clear(); 1373 buffers.clear(); 1374 1375 // check calling permissions 1376 if (!recordingAllowed()) { 1377 ALOGE("openRecord() permission denied: recording not allowed"); 1378 lStatus = PERMISSION_DENIED; 1379 goto Exit; 1380 } 1381 1382 // further sample rate checks are performed by createRecordTrack_l() 1383 if (sampleRate == 0) { 1384 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1385 lStatus = BAD_VALUE; 1386 goto Exit; 1387 } 1388 1389 // we don't yet support anything other than 16-bit PCM 1390 if (!(audio_is_valid_format(format) && 1391 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1392 ALOGE("openRecord() invalid format %#x", format); 1393 lStatus = BAD_VALUE; 1394 goto Exit; 1395 } 1396 1397 // further channel mask checks are performed by createRecordTrack_l() 1398 if (!audio_is_input_channel(channelMask)) { 1399 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1400 lStatus = BAD_VALUE; 1401 goto Exit; 1402 } 1403 1404 { 1405 Mutex::Autolock _l(mLock); 1406 RecordThread *thread = checkRecordThread_l(input); 1407 if (thread == NULL) { 1408 ALOGE("openRecord() checkRecordThread_l failed"); 1409 lStatus = BAD_VALUE; 1410 goto Exit; 1411 } 1412 1413 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1414 && !captureAudioOutputAllowed()) { 1415 ALOGE("openRecord() permission denied: capture not allowed"); 1416 lStatus = PERMISSION_DENIED; 1417 goto Exit; 1418 } 1419 1420 pid_t pid = IPCThreadState::self()->getCallingPid(); 1421 client = registerPid(pid); 1422 1423 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1424 lSessionId = *sessionId; 1425 } else { 1426 // if no audio session id is provided, create one here 1427 lSessionId = nextUniqueId(); 1428 if (sessionId != NULL) { 1429 *sessionId = lSessionId; 1430 } 1431 } 1432 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1433 1434 // TODO: the uid should be passed in as a parameter to openRecord 1435 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1436 frameCount, lSessionId, notificationFrames, 1437 IPCThreadState::self()->getCallingUid(), 1438 flags, tid, &lStatus); 1439 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1440 1441 if (lStatus == NO_ERROR) { 1442 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1443 // session and move it to this thread. 1444 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1445 if (chain != 0) { 1446 Mutex::Autolock _l(thread->mLock); 1447 thread->addEffectChain_l(chain); 1448 } 1449 } 1450 } 1451 1452 if (lStatus != NO_ERROR) { 1453 // remove local strong reference to Client before deleting the RecordTrack so that the 1454 // Client destructor is called by the TrackBase destructor with mClientLock held 1455 // Don't hold mClientLock when releasing the reference on the track as the 1456 // destructor will acquire it. 1457 { 1458 Mutex::Autolock _cl(mClientLock); 1459 client.clear(); 1460 } 1461 recordTrack.clear(); 1462 goto Exit; 1463 } 1464 1465 cblk = recordTrack->getCblk(); 1466 buffers = recordTrack->getBuffers(); 1467 1468 // return handle to client 1469 recordHandle = new RecordHandle(recordTrack); 1470 1471Exit: 1472 *status = lStatus; 1473 return recordHandle; 1474} 1475 1476 1477 1478// ---------------------------------------------------------------------------- 1479 1480audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1481{ 1482 if (name == NULL) { 1483 return 0; 1484 } 1485 if (!settingsAllowed()) { 1486 return 0; 1487 } 1488 Mutex::Autolock _l(mLock); 1489 return loadHwModule_l(name); 1490} 1491 1492// loadHwModule_l() must be called with AudioFlinger::mLock held 1493audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1494{ 1495 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1496 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1497 ALOGW("loadHwModule() module %s already loaded", name); 1498 return mAudioHwDevs.keyAt(i); 1499 } 1500 } 1501 1502 audio_hw_device_t *dev; 1503 1504 int rc = load_audio_interface(name, &dev); 1505 if (rc) { 1506 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1507 return 0; 1508 } 1509 1510 mHardwareStatus = AUDIO_HW_INIT; 1511 rc = dev->init_check(dev); 1512 mHardwareStatus = AUDIO_HW_IDLE; 1513 if (rc) { 1514 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1515 return 0; 1516 } 1517 1518 // Check and cache this HAL's level of support for master mute and master 1519 // volume. If this is the first HAL opened, and it supports the get 1520 // methods, use the initial values provided by the HAL as the current 1521 // master mute and volume settings. 1522 1523 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1524 { // scope for auto-lock pattern 1525 AutoMutex lock(mHardwareLock); 1526 1527 if (0 == mAudioHwDevs.size()) { 1528 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1529 if (NULL != dev->get_master_volume) { 1530 float mv; 1531 if (OK == dev->get_master_volume(dev, &mv)) { 1532 mMasterVolume = mv; 1533 } 1534 } 1535 1536 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1537 if (NULL != dev->get_master_mute) { 1538 bool mm; 1539 if (OK == dev->get_master_mute(dev, &mm)) { 1540 mMasterMute = mm; 1541 } 1542 } 1543 } 1544 1545 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1546 if ((NULL != dev->set_master_volume) && 1547 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1548 flags = static_cast<AudioHwDevice::Flags>(flags | 1549 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1550 } 1551 1552 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1553 if ((NULL != dev->set_master_mute) && 1554 (OK == dev->set_master_mute(dev, mMasterMute))) { 1555 flags = static_cast<AudioHwDevice::Flags>(flags | 1556 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1557 } 1558 1559 mHardwareStatus = AUDIO_HW_IDLE; 1560 } 1561 1562 audio_module_handle_t handle = nextUniqueId(); 1563 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1564 1565 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1566 name, dev->common.module->name, dev->common.module->id, handle); 1567 1568 return handle; 1569 1570} 1571 1572// ---------------------------------------------------------------------------- 1573 1574uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1575{ 1576 Mutex::Autolock _l(mLock); 1577 PlaybackThread *thread = primaryPlaybackThread_l(); 1578 return thread != NULL ? thread->sampleRate() : 0; 1579} 1580 1581size_t AudioFlinger::getPrimaryOutputFrameCount() 1582{ 1583 Mutex::Autolock _l(mLock); 1584 PlaybackThread *thread = primaryPlaybackThread_l(); 1585 return thread != NULL ? thread->frameCountHAL() : 0; 1586} 1587 1588// ---------------------------------------------------------------------------- 1589 1590status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1591{ 1592 uid_t uid = IPCThreadState::self()->getCallingUid(); 1593 if (uid != AID_SYSTEM) { 1594 return PERMISSION_DENIED; 1595 } 1596 Mutex::Autolock _l(mLock); 1597 if (mIsDeviceTypeKnown) { 1598 return INVALID_OPERATION; 1599 } 1600 mIsLowRamDevice = isLowRamDevice; 1601 mIsDeviceTypeKnown = true; 1602 return NO_ERROR; 1603} 1604 1605audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1606{ 1607 Mutex::Autolock _l(mLock); 1608 1609 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1610 if (index >= 0) { 1611 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1612 mHwAvSyncIds.valueAt(index), sessionId); 1613 return mHwAvSyncIds.valueAt(index); 1614 } 1615 1616 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1617 if (dev == NULL) { 1618 return AUDIO_HW_SYNC_INVALID; 1619 } 1620 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1621 AudioParameter param = AudioParameter(String8(reply)); 1622 free(reply); 1623 1624 int value; 1625 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1626 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1627 return AUDIO_HW_SYNC_INVALID; 1628 } 1629 1630 // allow only one session for a given HW A/V sync ID. 1631 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1632 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1633 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1634 value, mHwAvSyncIds.keyAt(i)); 1635 mHwAvSyncIds.removeItemsAt(i); 1636 break; 1637 } 1638 } 1639 1640 mHwAvSyncIds.add(sessionId, value); 1641 1642 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1643 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1644 uint32_t sessions = thread->hasAudioSession(sessionId); 1645 if (sessions & PlaybackThread::TRACK_SESSION) { 1646 AudioParameter param = AudioParameter(); 1647 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1648 thread->setParameters(param.toString()); 1649 break; 1650 } 1651 } 1652 1653 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1654 return (audio_hw_sync_t)value; 1655} 1656 1657// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1658void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1659{ 1660 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1661 if (index >= 0) { 1662 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1663 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1664 AudioParameter param = AudioParameter(); 1665 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1666 thread->setParameters(param.toString()); 1667 } 1668} 1669 1670 1671// ---------------------------------------------------------------------------- 1672 1673 1674sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1675 audio_io_handle_t *output, 1676 audio_config_t *config, 1677 audio_devices_t devices, 1678 const String8& address, 1679 audio_output_flags_t flags) 1680{ 1681 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1682 if (outHwDev == NULL) { 1683 return 0; 1684 } 1685 1686 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1687 if (*output == AUDIO_IO_HANDLE_NONE) { 1688 *output = nextUniqueId(); 1689 } 1690 1691 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1692 1693 audio_stream_out_t *outStream = NULL; 1694 1695 // FOR TESTING ONLY: 1696 // This if statement allows overriding the audio policy settings 1697 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1698 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1699 // Check only for Normal Mixing mode 1700 if (kEnableExtendedPrecision) { 1701 // Specify format (uncomment one below to choose) 1702 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1703 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1704 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1705 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1706 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1707 } 1708 if (kEnableExtendedChannels) { 1709 // Specify channel mask (uncomment one below to choose) 1710 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1711 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1712 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1713 } 1714 } 1715 1716 status_t status = hwDevHal->open_output_stream(hwDevHal, 1717 *output, 1718 devices, 1719 flags, 1720 config, 1721 &outStream, 1722 address.string()); 1723 1724 mHardwareStatus = AUDIO_HW_IDLE; 1725 ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, " 1726 "channelMask %#x, status %d", 1727 outStream, 1728 config->sample_rate, 1729 config->format, 1730 config->channel_mask, 1731 status); 1732 1733 if (status == NO_ERROR && outStream != NULL) { 1734 AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags); 1735 1736 PlaybackThread *thread; 1737 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1738 thread = new OffloadThread(this, outputStream, *output, devices); 1739 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1740 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1741 || !isValidPcmSinkFormat(config->format) 1742 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1743 thread = new DirectOutputThread(this, outputStream, *output, devices); 1744 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1745 } else { 1746 thread = new MixerThread(this, outputStream, *output, devices); 1747 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1748 } 1749 mPlaybackThreads.add(*output, thread); 1750 return thread; 1751 } 1752 1753 return 0; 1754} 1755 1756status_t AudioFlinger::openOutput(audio_module_handle_t module, 1757 audio_io_handle_t *output, 1758 audio_config_t *config, 1759 audio_devices_t *devices, 1760 const String8& address, 1761 uint32_t *latencyMs, 1762 audio_output_flags_t flags) 1763{ 1764 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1765 module, 1766 (devices != NULL) ? *devices : 0, 1767 config->sample_rate, 1768 config->format, 1769 config->channel_mask, 1770 flags); 1771 1772 if (*devices == AUDIO_DEVICE_NONE) { 1773 return BAD_VALUE; 1774 } 1775 1776 Mutex::Autolock _l(mLock); 1777 1778 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1779 if (thread != 0) { 1780 *latencyMs = thread->latency(); 1781 1782 // notify client processes of the new output creation 1783 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1784 1785 // the first primary output opened designates the primary hw device 1786 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1787 ALOGI("Using module %d has the primary audio interface", module); 1788 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1789 1790 AutoMutex lock(mHardwareLock); 1791 mHardwareStatus = AUDIO_HW_SET_MODE; 1792 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1793 mHardwareStatus = AUDIO_HW_IDLE; 1794 1795 mPrimaryOutputSampleRate = config->sample_rate; 1796 } 1797 return NO_ERROR; 1798 } 1799 1800 return NO_INIT; 1801} 1802 1803audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1804 audio_io_handle_t output2) 1805{ 1806 Mutex::Autolock _l(mLock); 1807 MixerThread *thread1 = checkMixerThread_l(output1); 1808 MixerThread *thread2 = checkMixerThread_l(output2); 1809 1810 if (thread1 == NULL || thread2 == NULL) { 1811 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1812 output2); 1813 return AUDIO_IO_HANDLE_NONE; 1814 } 1815 1816 audio_io_handle_t id = nextUniqueId(); 1817 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1818 thread->addOutputTrack(thread2); 1819 mPlaybackThreads.add(id, thread); 1820 // notify client processes of the new output creation 1821 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1822 return id; 1823} 1824 1825status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1826{ 1827 return closeOutput_nonvirtual(output); 1828} 1829 1830status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1831{ 1832 // keep strong reference on the playback thread so that 1833 // it is not destroyed while exit() is executed 1834 sp<PlaybackThread> thread; 1835 { 1836 Mutex::Autolock _l(mLock); 1837 thread = checkPlaybackThread_l(output); 1838 if (thread == NULL) { 1839 return BAD_VALUE; 1840 } 1841 1842 ALOGV("closeOutput() %d", output); 1843 1844 if (thread->type() == ThreadBase::MIXER) { 1845 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1846 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1847 DuplicatingThread *dupThread = 1848 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1849 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1850 1851 } 1852 } 1853 } 1854 1855 1856 mPlaybackThreads.removeItem(output); 1857 // save all effects to the default thread 1858 if (mPlaybackThreads.size()) { 1859 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1860 if (dstThread != NULL) { 1861 // audioflinger lock is held here so the acquisition order of thread locks does not 1862 // matter 1863 Mutex::Autolock _dl(dstThread->mLock); 1864 Mutex::Autolock _sl(thread->mLock); 1865 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1866 for (size_t i = 0; i < effectChains.size(); i ++) { 1867 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1868 } 1869 } 1870 } 1871 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1872 } 1873 thread->exit(); 1874 // The thread entity (active unit of execution) is no longer running here, 1875 // but the ThreadBase container still exists. 1876 1877 if (thread->type() != ThreadBase::DUPLICATING) { 1878 closeOutputFinish(thread); 1879 } 1880 1881 return NO_ERROR; 1882} 1883 1884void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1885{ 1886 AudioStreamOut *out = thread->clearOutput(); 1887 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1888 // from now on thread->mOutput is NULL 1889 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1890 delete out; 1891} 1892 1893void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1894{ 1895 mPlaybackThreads.removeItem(thread->mId); 1896 thread->exit(); 1897 closeOutputFinish(thread); 1898} 1899 1900status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1901{ 1902 Mutex::Autolock _l(mLock); 1903 PlaybackThread *thread = checkPlaybackThread_l(output); 1904 1905 if (thread == NULL) { 1906 return BAD_VALUE; 1907 } 1908 1909 ALOGV("suspendOutput() %d", output); 1910 thread->suspend(); 1911 1912 return NO_ERROR; 1913} 1914 1915status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1916{ 1917 Mutex::Autolock _l(mLock); 1918 PlaybackThread *thread = checkPlaybackThread_l(output); 1919 1920 if (thread == NULL) { 1921 return BAD_VALUE; 1922 } 1923 1924 ALOGV("restoreOutput() %d", output); 1925 1926 thread->restore(); 1927 1928 return NO_ERROR; 1929} 1930 1931status_t AudioFlinger::openInput(audio_module_handle_t module, 1932 audio_io_handle_t *input, 1933 audio_config_t *config, 1934 audio_devices_t *device, 1935 const String8& address, 1936 audio_source_t source, 1937 audio_input_flags_t flags) 1938{ 1939 Mutex::Autolock _l(mLock); 1940 1941 if (*device == AUDIO_DEVICE_NONE) { 1942 return BAD_VALUE; 1943 } 1944 1945 sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags); 1946 1947 if (thread != 0) { 1948 // notify client processes of the new input creation 1949 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1950 return NO_ERROR; 1951 } 1952 return NO_INIT; 1953} 1954 1955sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 1956 audio_io_handle_t *input, 1957 audio_config_t *config, 1958 audio_devices_t device, 1959 const String8& address, 1960 audio_source_t source, 1961 audio_input_flags_t flags) 1962{ 1963 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device); 1964 if (inHwDev == NULL) { 1965 *input = AUDIO_IO_HANDLE_NONE; 1966 return 0; 1967 } 1968 1969 if (*input == AUDIO_IO_HANDLE_NONE) { 1970 *input = nextUniqueId(); 1971 } 1972 1973 audio_config_t halconfig = *config; 1974 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1975 audio_stream_in_t *inStream = NULL; 1976 status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1977 &inStream, flags, address.string(), source); 1978 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 1979 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 1980 inStream, 1981 halconfig.sample_rate, 1982 halconfig.format, 1983 halconfig.channel_mask, 1984 flags, 1985 status, address.string()); 1986 1987 // If the input could not be opened with the requested parameters and we can handle the 1988 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1989 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1990 if (status == BAD_VALUE && 1991 config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT && 1992 (halconfig.sample_rate <= 2 * config->sample_rate) && 1993 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 1994 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 1995 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1996 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 1997 inStream = NULL; 1998 status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1999 &inStream, flags, address.string(), source); 2000 // FIXME log this new status; HAL should not propose any further changes 2001 } 2002 2003 if (status == NO_ERROR && inStream != NULL) { 2004 2005#ifdef TEE_SINK 2006 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2007 // or (re-)create if current Pipe is idle and does not match the new format 2008 sp<NBAIO_Sink> teeSink; 2009 enum { 2010 TEE_SINK_NO, // don't copy input 2011 TEE_SINK_NEW, // copy input using a new pipe 2012 TEE_SINK_OLD, // copy input using an existing pipe 2013 } kind; 2014 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2015 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2016 if (!mTeeSinkInputEnabled) { 2017 kind = TEE_SINK_NO; 2018 } else if (!Format_isValid(format)) { 2019 kind = TEE_SINK_NO; 2020 } else if (mRecordTeeSink == 0) { 2021 kind = TEE_SINK_NEW; 2022 } else if (mRecordTeeSink->getStrongCount() != 1) { 2023 kind = TEE_SINK_NO; 2024 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2025 kind = TEE_SINK_OLD; 2026 } else { 2027 kind = TEE_SINK_NEW; 2028 } 2029 switch (kind) { 2030 case TEE_SINK_NEW: { 2031 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2032 size_t numCounterOffers = 0; 2033 const NBAIO_Format offers[1] = {format}; 2034 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2035 ALOG_ASSERT(index == 0); 2036 PipeReader *pipeReader = new PipeReader(*pipe); 2037 numCounterOffers = 0; 2038 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2039 ALOG_ASSERT(index == 0); 2040 mRecordTeeSink = pipe; 2041 mRecordTeeSource = pipeReader; 2042 teeSink = pipe; 2043 } 2044 break; 2045 case TEE_SINK_OLD: 2046 teeSink = mRecordTeeSink; 2047 break; 2048 case TEE_SINK_NO: 2049 default: 2050 break; 2051 } 2052#endif 2053 2054 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2055 2056 // Start record thread 2057 // RecordThread requires both input and output device indication to forward to audio 2058 // pre processing modules 2059 sp<RecordThread> thread = new RecordThread(this, 2060 inputStream, 2061 *input, 2062 primaryOutputDevice_l(), 2063 device 2064#ifdef TEE_SINK 2065 , teeSink 2066#endif 2067 ); 2068 mRecordThreads.add(*input, thread); 2069 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2070 return thread; 2071 } 2072 2073 *input = AUDIO_IO_HANDLE_NONE; 2074 return 0; 2075} 2076 2077status_t AudioFlinger::closeInput(audio_io_handle_t input) 2078{ 2079 return closeInput_nonvirtual(input); 2080} 2081 2082status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2083{ 2084 // keep strong reference on the record thread so that 2085 // it is not destroyed while exit() is executed 2086 sp<RecordThread> thread; 2087 { 2088 Mutex::Autolock _l(mLock); 2089 thread = checkRecordThread_l(input); 2090 if (thread == 0) { 2091 return BAD_VALUE; 2092 } 2093 2094 ALOGV("closeInput() %d", input); 2095 2096 // If we still have effect chains, it means that a client still holds a handle 2097 // on at least one effect. We must either move the chain to an existing thread with the 2098 // same session ID or put it aside in case a new record thread is opened for a 2099 // new capture on the same session 2100 sp<EffectChain> chain; 2101 { 2102 Mutex::Autolock _sl(thread->mLock); 2103 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2104 // Note: maximum one chain per record thread 2105 if (effectChains.size() != 0) { 2106 chain = effectChains[0]; 2107 } 2108 } 2109 if (chain != 0) { 2110 // first check if a record thread is already opened with a client on the same session. 2111 // This should only happen in case of overlap between one thread tear down and the 2112 // creation of its replacement 2113 size_t i; 2114 for (i = 0; i < mRecordThreads.size(); i++) { 2115 sp<RecordThread> t = mRecordThreads.valueAt(i); 2116 if (t == thread) { 2117 continue; 2118 } 2119 if (t->hasAudioSession(chain->sessionId()) != 0) { 2120 Mutex::Autolock _l(t->mLock); 2121 ALOGV("closeInput() found thread %d for effect session %d", 2122 t->id(), chain->sessionId()); 2123 t->addEffectChain_l(chain); 2124 break; 2125 } 2126 } 2127 // put the chain aside if we could not find a record thread with the same session id. 2128 if (i == mRecordThreads.size()) { 2129 putOrphanEffectChain_l(chain); 2130 } 2131 } 2132 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 2133 mRecordThreads.removeItem(input); 2134 } 2135 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2136 // we have a different lock for notification client 2137 closeInputFinish(thread); 2138 return NO_ERROR; 2139} 2140 2141void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2142{ 2143 thread->exit(); 2144 AudioStreamIn *in = thread->clearInput(); 2145 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2146 // from now on thread->mInput is NULL 2147 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2148 delete in; 2149} 2150 2151void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2152{ 2153 mRecordThreads.removeItem(thread->mId); 2154 closeInputFinish(thread); 2155} 2156 2157status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2158{ 2159 Mutex::Autolock _l(mLock); 2160 ALOGV("invalidateStream() stream %d", stream); 2161 2162 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2163 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2164 thread->invalidateTracks(stream); 2165 } 2166 2167 return NO_ERROR; 2168} 2169 2170 2171audio_unique_id_t AudioFlinger::newAudioUniqueId() 2172{ 2173 return nextUniqueId(); 2174} 2175 2176void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2177{ 2178 Mutex::Autolock _l(mLock); 2179 pid_t caller = IPCThreadState::self()->getCallingPid(); 2180 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2181 if (pid != -1 && (caller == getpid_cached)) { 2182 caller = pid; 2183 } 2184 2185 { 2186 Mutex::Autolock _cl(mClientLock); 2187 // Ignore requests received from processes not known as notification client. The request 2188 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2189 // called from a different pid leaving a stale session reference. Also we don't know how 2190 // to clear this reference if the client process dies. 2191 if (mNotificationClients.indexOfKey(caller) < 0) { 2192 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2193 return; 2194 } 2195 } 2196 2197 size_t num = mAudioSessionRefs.size(); 2198 for (size_t i = 0; i< num; i++) { 2199 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2200 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2201 ref->mCnt++; 2202 ALOGV(" incremented refcount to %d", ref->mCnt); 2203 return; 2204 } 2205 } 2206 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2207 ALOGV(" added new entry for %d", audioSession); 2208} 2209 2210void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2211{ 2212 Mutex::Autolock _l(mLock); 2213 pid_t caller = IPCThreadState::self()->getCallingPid(); 2214 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2215 if (pid != -1 && (caller == getpid_cached)) { 2216 caller = pid; 2217 } 2218 size_t num = mAudioSessionRefs.size(); 2219 for (size_t i = 0; i< num; i++) { 2220 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2221 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2222 ref->mCnt--; 2223 ALOGV(" decremented refcount to %d", ref->mCnt); 2224 if (ref->mCnt == 0) { 2225 mAudioSessionRefs.removeAt(i); 2226 delete ref; 2227 purgeStaleEffects_l(); 2228 } 2229 return; 2230 } 2231 } 2232 // If the caller is mediaserver it is likely that the session being released was acquired 2233 // on behalf of a process not in notification clients and we ignore the warning. 2234 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2235} 2236 2237void AudioFlinger::purgeStaleEffects_l() { 2238 2239 ALOGV("purging stale effects"); 2240 2241 Vector< sp<EffectChain> > chains; 2242 2243 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2244 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2245 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2246 sp<EffectChain> ec = t->mEffectChains[j]; 2247 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2248 chains.push(ec); 2249 } 2250 } 2251 } 2252 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2253 sp<RecordThread> t = mRecordThreads.valueAt(i); 2254 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2255 sp<EffectChain> ec = t->mEffectChains[j]; 2256 chains.push(ec); 2257 } 2258 } 2259 2260 for (size_t i = 0; i < chains.size(); i++) { 2261 sp<EffectChain> ec = chains[i]; 2262 int sessionid = ec->sessionId(); 2263 sp<ThreadBase> t = ec->mThread.promote(); 2264 if (t == 0) { 2265 continue; 2266 } 2267 size_t numsessionrefs = mAudioSessionRefs.size(); 2268 bool found = false; 2269 for (size_t k = 0; k < numsessionrefs; k++) { 2270 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2271 if (ref->mSessionid == sessionid) { 2272 ALOGV(" session %d still exists for %d with %d refs", 2273 sessionid, ref->mPid, ref->mCnt); 2274 found = true; 2275 break; 2276 } 2277 } 2278 if (!found) { 2279 Mutex::Autolock _l(t->mLock); 2280 // remove all effects from the chain 2281 while (ec->mEffects.size()) { 2282 sp<EffectModule> effect = ec->mEffects[0]; 2283 effect->unPin(); 2284 t->removeEffect_l(effect); 2285 if (effect->purgeHandles()) { 2286 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2287 } 2288 AudioSystem::unregisterEffect(effect->id()); 2289 } 2290 } 2291 } 2292 return; 2293} 2294 2295// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2296AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2297{ 2298 return mPlaybackThreads.valueFor(output).get(); 2299} 2300 2301// checkMixerThread_l() must be called with AudioFlinger::mLock held 2302AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2303{ 2304 PlaybackThread *thread = checkPlaybackThread_l(output); 2305 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2306} 2307 2308// checkRecordThread_l() must be called with AudioFlinger::mLock held 2309AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2310{ 2311 return mRecordThreads.valueFor(input).get(); 2312} 2313 2314uint32_t AudioFlinger::nextUniqueId() 2315{ 2316 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2317} 2318 2319AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2320{ 2321 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2322 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2323 AudioStreamOut *output = thread->getOutput(); 2324 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2325 return thread; 2326 } 2327 } 2328 return NULL; 2329} 2330 2331audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2332{ 2333 PlaybackThread *thread = primaryPlaybackThread_l(); 2334 2335 if (thread == NULL) { 2336 return 0; 2337 } 2338 2339 return thread->outDevice(); 2340} 2341 2342sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2343 int triggerSession, 2344 int listenerSession, 2345 sync_event_callback_t callBack, 2346 wp<RefBase> cookie) 2347{ 2348 Mutex::Autolock _l(mLock); 2349 2350 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2351 status_t playStatus = NAME_NOT_FOUND; 2352 status_t recStatus = NAME_NOT_FOUND; 2353 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2354 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2355 if (playStatus == NO_ERROR) { 2356 return event; 2357 } 2358 } 2359 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2360 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2361 if (recStatus == NO_ERROR) { 2362 return event; 2363 } 2364 } 2365 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2366 mPendingSyncEvents.add(event); 2367 } else { 2368 ALOGV("createSyncEvent() invalid event %d", event->type()); 2369 event.clear(); 2370 } 2371 return event; 2372} 2373 2374// ---------------------------------------------------------------------------- 2375// Effect management 2376// ---------------------------------------------------------------------------- 2377 2378 2379status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2380{ 2381 Mutex::Autolock _l(mLock); 2382 return EffectQueryNumberEffects(numEffects); 2383} 2384 2385status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2386{ 2387 Mutex::Autolock _l(mLock); 2388 return EffectQueryEffect(index, descriptor); 2389} 2390 2391status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2392 effect_descriptor_t *descriptor) const 2393{ 2394 Mutex::Autolock _l(mLock); 2395 return EffectGetDescriptor(pUuid, descriptor); 2396} 2397 2398 2399sp<IEffect> AudioFlinger::createEffect( 2400 effect_descriptor_t *pDesc, 2401 const sp<IEffectClient>& effectClient, 2402 int32_t priority, 2403 audio_io_handle_t io, 2404 int sessionId, 2405 status_t *status, 2406 int *id, 2407 int *enabled) 2408{ 2409 status_t lStatus = NO_ERROR; 2410 sp<EffectHandle> handle; 2411 effect_descriptor_t desc; 2412 2413 pid_t pid = IPCThreadState::self()->getCallingPid(); 2414 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2415 pid, effectClient.get(), priority, sessionId, io); 2416 2417 if (pDesc == NULL) { 2418 lStatus = BAD_VALUE; 2419 goto Exit; 2420 } 2421 2422 // check audio settings permission for global effects 2423 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2424 lStatus = PERMISSION_DENIED; 2425 goto Exit; 2426 } 2427 2428 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2429 // that can only be created by audio policy manager (running in same process) 2430 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2431 lStatus = PERMISSION_DENIED; 2432 goto Exit; 2433 } 2434 2435 { 2436 if (!EffectIsNullUuid(&pDesc->uuid)) { 2437 // if uuid is specified, request effect descriptor 2438 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2439 if (lStatus < 0) { 2440 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2441 goto Exit; 2442 } 2443 } else { 2444 // if uuid is not specified, look for an available implementation 2445 // of the required type in effect factory 2446 if (EffectIsNullUuid(&pDesc->type)) { 2447 ALOGW("createEffect() no effect type"); 2448 lStatus = BAD_VALUE; 2449 goto Exit; 2450 } 2451 uint32_t numEffects = 0; 2452 effect_descriptor_t d; 2453 d.flags = 0; // prevent compiler warning 2454 bool found = false; 2455 2456 lStatus = EffectQueryNumberEffects(&numEffects); 2457 if (lStatus < 0) { 2458 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2459 goto Exit; 2460 } 2461 for (uint32_t i = 0; i < numEffects; i++) { 2462 lStatus = EffectQueryEffect(i, &desc); 2463 if (lStatus < 0) { 2464 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2465 continue; 2466 } 2467 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2468 // If matching type found save effect descriptor. If the session is 2469 // 0 and the effect is not auxiliary, continue enumeration in case 2470 // an auxiliary version of this effect type is available 2471 found = true; 2472 d = desc; 2473 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2474 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2475 break; 2476 } 2477 } 2478 } 2479 if (!found) { 2480 lStatus = BAD_VALUE; 2481 ALOGW("createEffect() effect not found"); 2482 goto Exit; 2483 } 2484 // For same effect type, chose auxiliary version over insert version if 2485 // connect to output mix (Compliance to OpenSL ES) 2486 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2487 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2488 desc = d; 2489 } 2490 } 2491 2492 // Do not allow auxiliary effects on a session different from 0 (output mix) 2493 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2494 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2495 lStatus = INVALID_OPERATION; 2496 goto Exit; 2497 } 2498 2499 // check recording permission for visualizer 2500 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2501 !recordingAllowed()) { 2502 lStatus = PERMISSION_DENIED; 2503 goto Exit; 2504 } 2505 2506 // return effect descriptor 2507 *pDesc = desc; 2508 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2509 // if the output returned by getOutputForEffect() is removed before we lock the 2510 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2511 // and we will exit safely 2512 io = AudioSystem::getOutputForEffect(&desc); 2513 ALOGV("createEffect got output %d", io); 2514 } 2515 2516 Mutex::Autolock _l(mLock); 2517 2518 // If output is not specified try to find a matching audio session ID in one of the 2519 // output threads. 2520 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2521 // because of code checking output when entering the function. 2522 // Note: io is never 0 when creating an effect on an input 2523 if (io == AUDIO_IO_HANDLE_NONE) { 2524 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2525 // output must be specified by AudioPolicyManager when using session 2526 // AUDIO_SESSION_OUTPUT_STAGE 2527 lStatus = BAD_VALUE; 2528 goto Exit; 2529 } 2530 // look for the thread where the specified audio session is present 2531 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2532 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2533 io = mPlaybackThreads.keyAt(i); 2534 break; 2535 } 2536 } 2537 if (io == 0) { 2538 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2539 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2540 io = mRecordThreads.keyAt(i); 2541 break; 2542 } 2543 } 2544 } 2545 // If no output thread contains the requested session ID, default to 2546 // first output. The effect chain will be moved to the correct output 2547 // thread when a track with the same session ID is created 2548 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2549 io = mPlaybackThreads.keyAt(0); 2550 } 2551 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2552 } 2553 ThreadBase *thread = checkRecordThread_l(io); 2554 if (thread == NULL) { 2555 thread = checkPlaybackThread_l(io); 2556 if (thread == NULL) { 2557 ALOGE("createEffect() unknown output thread"); 2558 lStatus = BAD_VALUE; 2559 goto Exit; 2560 } 2561 } else { 2562 // Check if one effect chain was awaiting for an effect to be created on this 2563 // session and used it instead of creating a new one. 2564 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2565 if (chain != 0) { 2566 Mutex::Autolock _l(thread->mLock); 2567 thread->addEffectChain_l(chain); 2568 } 2569 } 2570 2571 sp<Client> client = registerPid(pid); 2572 2573 // create effect on selected output thread 2574 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2575 &desc, enabled, &lStatus); 2576 if (handle != 0 && id != NULL) { 2577 *id = handle->id(); 2578 } 2579 if (handle == 0) { 2580 // remove local strong reference to Client with mClientLock held 2581 Mutex::Autolock _cl(mClientLock); 2582 client.clear(); 2583 } 2584 } 2585 2586Exit: 2587 *status = lStatus; 2588 return handle; 2589} 2590 2591status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2592 audio_io_handle_t dstOutput) 2593{ 2594 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2595 sessionId, srcOutput, dstOutput); 2596 Mutex::Autolock _l(mLock); 2597 if (srcOutput == dstOutput) { 2598 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2599 return NO_ERROR; 2600 } 2601 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2602 if (srcThread == NULL) { 2603 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2604 return BAD_VALUE; 2605 } 2606 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2607 if (dstThread == NULL) { 2608 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2609 return BAD_VALUE; 2610 } 2611 2612 Mutex::Autolock _dl(dstThread->mLock); 2613 Mutex::Autolock _sl(srcThread->mLock); 2614 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2615} 2616 2617// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2618status_t AudioFlinger::moveEffectChain_l(int sessionId, 2619 AudioFlinger::PlaybackThread *srcThread, 2620 AudioFlinger::PlaybackThread *dstThread, 2621 bool reRegister) 2622{ 2623 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2624 sessionId, srcThread, dstThread); 2625 2626 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2627 if (chain == 0) { 2628 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2629 sessionId, srcThread); 2630 return INVALID_OPERATION; 2631 } 2632 2633 // Check whether the destination thread has a channel count of FCC_2, which is 2634 // currently required for (most) effects. Prevent moving the effect chain here rather 2635 // than disabling the addEffect_l() call in dstThread below. 2636 if (dstThread->mChannelCount != FCC_2) { 2637 ALOGW("moveEffectChain_l() effect chain failed because" 2638 " destination thread %p channel count(%u) != %u", 2639 dstThread, dstThread->mChannelCount, FCC_2); 2640 return INVALID_OPERATION; 2641 } 2642 2643 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2644 // so that a new chain is created with correct parameters when first effect is added. This is 2645 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2646 // removed. 2647 srcThread->removeEffectChain_l(chain); 2648 2649 // transfer all effects one by one so that new effect chain is created on new thread with 2650 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2651 sp<EffectChain> dstChain; 2652 uint32_t strategy = 0; // prevent compiler warning 2653 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2654 Vector< sp<EffectModule> > removed; 2655 status_t status = NO_ERROR; 2656 while (effect != 0) { 2657 srcThread->removeEffect_l(effect); 2658 removed.add(effect); 2659 status = dstThread->addEffect_l(effect); 2660 if (status != NO_ERROR) { 2661 break; 2662 } 2663 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2664 if (effect->state() == EffectModule::ACTIVE || 2665 effect->state() == EffectModule::STOPPING) { 2666 effect->start(); 2667 } 2668 // if the move request is not received from audio policy manager, the effect must be 2669 // re-registered with the new strategy and output 2670 if (dstChain == 0) { 2671 dstChain = effect->chain().promote(); 2672 if (dstChain == 0) { 2673 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2674 status = NO_INIT; 2675 break; 2676 } 2677 strategy = dstChain->strategy(); 2678 } 2679 if (reRegister) { 2680 AudioSystem::unregisterEffect(effect->id()); 2681 AudioSystem::registerEffect(&effect->desc(), 2682 dstThread->id(), 2683 strategy, 2684 sessionId, 2685 effect->id()); 2686 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2687 } 2688 effect = chain->getEffectFromId_l(0); 2689 } 2690 2691 if (status != NO_ERROR) { 2692 for (size_t i = 0; i < removed.size(); i++) { 2693 srcThread->addEffect_l(removed[i]); 2694 if (dstChain != 0 && reRegister) { 2695 AudioSystem::unregisterEffect(removed[i]->id()); 2696 AudioSystem::registerEffect(&removed[i]->desc(), 2697 srcThread->id(), 2698 strategy, 2699 sessionId, 2700 removed[i]->id()); 2701 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2702 } 2703 } 2704 } 2705 2706 return status; 2707} 2708 2709bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2710{ 2711 if (mGlobalEffectEnableTime != 0 && 2712 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2713 return true; 2714 } 2715 2716 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2717 sp<EffectChain> ec = 2718 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2719 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2720 return true; 2721 } 2722 } 2723 return false; 2724} 2725 2726void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2727{ 2728 Mutex::Autolock _l(mLock); 2729 2730 mGlobalEffectEnableTime = systemTime(); 2731 2732 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2733 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2734 if (t->mType == ThreadBase::OFFLOAD) { 2735 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2736 } 2737 } 2738 2739} 2740 2741status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2742{ 2743 audio_session_t session = (audio_session_t)chain->sessionId(); 2744 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2745 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2746 if (index >= 0) { 2747 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2748 return ALREADY_EXISTS; 2749 } 2750 mOrphanEffectChains.add(session, chain); 2751 return NO_ERROR; 2752} 2753 2754sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2755{ 2756 sp<EffectChain> chain; 2757 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2758 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2759 if (index >= 0) { 2760 chain = mOrphanEffectChains.valueAt(index); 2761 mOrphanEffectChains.removeItemsAt(index); 2762 } 2763 return chain; 2764} 2765 2766bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2767{ 2768 Mutex::Autolock _l(mLock); 2769 audio_session_t session = (audio_session_t)effect->sessionId(); 2770 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2771 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2772 if (index >= 0) { 2773 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2774 if (chain->removeEffect_l(effect) == 0) { 2775 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2776 mOrphanEffectChains.removeItemsAt(index); 2777 } 2778 return true; 2779 } 2780 return false; 2781} 2782 2783 2784struct Entry { 2785#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2786 char mName[MAX_NAME]; 2787}; 2788 2789int comparEntry(const void *p1, const void *p2) 2790{ 2791 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2792} 2793 2794#ifdef TEE_SINK 2795void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2796{ 2797 NBAIO_Source *teeSource = source.get(); 2798 if (teeSource != NULL) { 2799 // .wav rotation 2800 // There is a benign race condition if 2 threads call this simultaneously. 2801 // They would both traverse the directory, but the result would simply be 2802 // failures at unlink() which are ignored. It's also unlikely since 2803 // normally dumpsys is only done by bugreport or from the command line. 2804 char teePath[32+256]; 2805 strcpy(teePath, "/data/misc/media"); 2806 size_t teePathLen = strlen(teePath); 2807 DIR *dir = opendir(teePath); 2808 teePath[teePathLen++] = '/'; 2809 if (dir != NULL) { 2810#define MAX_SORT 20 // number of entries to sort 2811#define MAX_KEEP 10 // number of entries to keep 2812 struct Entry entries[MAX_SORT]; 2813 size_t entryCount = 0; 2814 while (entryCount < MAX_SORT) { 2815 struct dirent de; 2816 struct dirent *result = NULL; 2817 int rc = readdir_r(dir, &de, &result); 2818 if (rc != 0) { 2819 ALOGW("readdir_r failed %d", rc); 2820 break; 2821 } 2822 if (result == NULL) { 2823 break; 2824 } 2825 if (result != &de) { 2826 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2827 break; 2828 } 2829 // ignore non .wav file entries 2830 size_t nameLen = strlen(de.d_name); 2831 if (nameLen <= 4 || nameLen >= MAX_NAME || 2832 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2833 continue; 2834 } 2835 strcpy(entries[entryCount++].mName, de.d_name); 2836 } 2837 (void) closedir(dir); 2838 if (entryCount > MAX_KEEP) { 2839 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2840 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2841 strcpy(&teePath[teePathLen], entries[i].mName); 2842 (void) unlink(teePath); 2843 } 2844 } 2845 } else { 2846 if (fd >= 0) { 2847 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2848 } 2849 } 2850 char teeTime[16]; 2851 struct timeval tv; 2852 gettimeofday(&tv, NULL); 2853 struct tm tm; 2854 localtime_r(&tv.tv_sec, &tm); 2855 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2856 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2857 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2858 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2859 if (teeFd >= 0) { 2860 // FIXME use libsndfile 2861 char wavHeader[44]; 2862 memcpy(wavHeader, 2863 "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", 2864 sizeof(wavHeader)); 2865 NBAIO_Format format = teeSource->format(); 2866 unsigned channelCount = Format_channelCount(format); 2867 uint32_t sampleRate = Format_sampleRate(format); 2868 size_t frameSize = Format_frameSize(format); 2869 wavHeader[22] = channelCount; // number of channels 2870 wavHeader[24] = sampleRate; // sample rate 2871 wavHeader[25] = sampleRate >> 8; 2872 wavHeader[32] = frameSize; // block alignment 2873 wavHeader[33] = frameSize >> 8; 2874 write(teeFd, wavHeader, sizeof(wavHeader)); 2875 size_t total = 0; 2876 bool firstRead = true; 2877#define TEE_SINK_READ 1024 // frames per I/O operation 2878 void *buffer = malloc(TEE_SINK_READ * frameSize); 2879 for (;;) { 2880 size_t count = TEE_SINK_READ; 2881 ssize_t actual = teeSource->read(buffer, count, 2882 AudioBufferProvider::kInvalidPTS); 2883 bool wasFirstRead = firstRead; 2884 firstRead = false; 2885 if (actual <= 0) { 2886 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2887 continue; 2888 } 2889 break; 2890 } 2891 ALOG_ASSERT(actual <= (ssize_t)count); 2892 write(teeFd, buffer, actual * frameSize); 2893 total += actual; 2894 } 2895 free(buffer); 2896 lseek(teeFd, (off_t) 4, SEEK_SET); 2897 uint32_t temp = 44 + total * frameSize - 8; 2898 // FIXME not big-endian safe 2899 write(teeFd, &temp, sizeof(temp)); 2900 lseek(teeFd, (off_t) 40, SEEK_SET); 2901 temp = total * frameSize; 2902 // FIXME not big-endian safe 2903 write(teeFd, &temp, sizeof(temp)); 2904 close(teeFd); 2905 if (fd >= 0) { 2906 dprintf(fd, "tee copied to %s\n", teePath); 2907 } 2908 } else { 2909 if (fd >= 0) { 2910 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2911 } 2912 } 2913 } 2914} 2915#endif 2916 2917// ---------------------------------------------------------------------------- 2918 2919status_t AudioFlinger::onTransact( 2920 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2921{ 2922 return BnAudioFlinger::onTransact(code, data, reply, flags); 2923} 2924 2925}; // namespace android 2926