AudioFlinger.cpp revision 298c4dc7e90cae4873d89098b777d1068a4e35ea
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 } 404 405 EffectDumpEffects(fd); 406 407 dumpClients(fd, args); 408 if (clientLocked) { 409 mClientLock.unlock(); 410 } 411 412 dumpInternals(fd, args); 413 414 // dump playback threads 415 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 416 mPlaybackThreads.valueAt(i)->dump(fd, args); 417 } 418 419 // dump record threads 420 for (size_t i = 0; i < mRecordThreads.size(); i++) { 421 mRecordThreads.valueAt(i)->dump(fd, args); 422 } 423 424 // dump orphan effect chains 425 if (mOrphanEffectChains.size() != 0) { 426 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 427 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 428 mOrphanEffectChains.valueAt(i)->dump(fd, args); 429 } 430 } 431 // dump all hardware devs 432 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 433 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 434 dev->dump(dev, fd); 435 } 436 437#ifdef TEE_SINK 438 // dump the serially shared record tee sink 439 if (mRecordTeeSource != 0) { 440 dumpTee(fd, mRecordTeeSource); 441 } 442#endif 443 444 if (locked) { 445 mLock.unlock(); 446 } 447 448 // append a copy of media.log here by forwarding fd to it, but don't attempt 449 // to lookup the service if it's not running, as it will block for a second 450 if (mLogMemoryDealer != 0) { 451 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 452 if (binder != 0) { 453 dprintf(fd, "\nmedia.log:\n"); 454 Vector<String16> args; 455 binder->dump(fd, args); 456 } 457 } 458 } 459 return NO_ERROR; 460} 461 462sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 463{ 464 Mutex::Autolock _cl(mClientLock); 465 // If pid is already in the mClients wp<> map, then use that entry 466 // (for which promote() is always != 0), otherwise create a new entry and Client. 467 sp<Client> client = mClients.valueFor(pid).promote(); 468 if (client == 0) { 469 client = new Client(this, pid); 470 mClients.add(pid, client); 471 } 472 473 return client; 474} 475 476sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 477{ 478 // If there is no memory allocated for logs, return a dummy writer that does nothing 479 if (mLogMemoryDealer == 0) { 480 return new NBLog::Writer(); 481 } 482 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 483 // Similarly if we can't contact the media.log service, also return a dummy writer 484 if (binder == 0) { 485 return new NBLog::Writer(); 486 } 487 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 488 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 489 // If allocation fails, consult the vector of previously unregistered writers 490 // and garbage-collect one or more them until an allocation succeeds 491 if (shared == 0) { 492 Mutex::Autolock _l(mUnregisteredWritersLock); 493 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 494 { 495 // Pick the oldest stale writer to garbage-collect 496 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 497 mUnregisteredWriters.removeAt(0); 498 mediaLogService->unregisterWriter(iMemory); 499 // Now the media.log remote reference to IMemory is gone. When our last local 500 // reference to IMemory also drops to zero at end of this block, 501 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 502 } 503 // Re-attempt the allocation 504 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 505 if (shared != 0) { 506 goto success; 507 } 508 } 509 // Even after garbage-collecting all old writers, there is still not enough memory, 510 // so return a dummy writer 511 return new NBLog::Writer(); 512 } 513success: 514 mediaLogService->registerWriter(shared, size, name); 515 return new NBLog::Writer(size, shared); 516} 517 518void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 519{ 520 if (writer == 0) { 521 return; 522 } 523 sp<IMemory> iMemory(writer->getIMemory()); 524 if (iMemory == 0) { 525 return; 526 } 527 // Rather than removing the writer immediately, append it to a queue of old writers to 528 // be garbage-collected later. This allows us to continue to view old logs for a while. 529 Mutex::Autolock _l(mUnregisteredWritersLock); 530 mUnregisteredWriters.push(writer); 531} 532 533// IAudioFlinger interface 534 535 536sp<IAudioTrack> AudioFlinger::createTrack( 537 audio_stream_type_t streamType, 538 uint32_t sampleRate, 539 audio_format_t format, 540 audio_channel_mask_t channelMask, 541 size_t *frameCount, 542 IAudioFlinger::track_flags_t *flags, 543 const sp<IMemory>& sharedBuffer, 544 audio_io_handle_t output, 545 pid_t tid, 546 int *sessionId, 547 int clientUid, 548 status_t *status) 549{ 550 sp<PlaybackThread::Track> track; 551 sp<TrackHandle> trackHandle; 552 sp<Client> client; 553 status_t lStatus; 554 int lSessionId; 555 556 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 557 // but if someone uses binder directly they could bypass that and cause us to crash 558 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 559 ALOGE("createTrack() invalid stream type %d", streamType); 560 lStatus = BAD_VALUE; 561 goto Exit; 562 } 563 564 // further sample rate checks are performed by createTrack_l() depending on the thread type 565 if (sampleRate == 0) { 566 ALOGE("createTrack() invalid sample rate %u", sampleRate); 567 lStatus = BAD_VALUE; 568 goto Exit; 569 } 570 571 // further channel mask checks are performed by createTrack_l() depending on the thread type 572 if (!audio_is_output_channel(channelMask)) { 573 ALOGE("createTrack() invalid channel mask %#x", channelMask); 574 lStatus = BAD_VALUE; 575 goto Exit; 576 } 577 578 // further format checks are performed by createTrack_l() depending on the thread type 579 if (!audio_is_valid_format(format)) { 580 ALOGE("createTrack() invalid format %#x", format); 581 lStatus = BAD_VALUE; 582 goto Exit; 583 } 584 585 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 586 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 587 lStatus = BAD_VALUE; 588 goto Exit; 589 } 590 591 { 592 Mutex::Autolock _l(mLock); 593 PlaybackThread *thread = checkPlaybackThread_l(output); 594 if (thread == NULL) { 595 ALOGE("no playback thread found for output handle %d", output); 596 lStatus = BAD_VALUE; 597 goto Exit; 598 } 599 600 pid_t pid = IPCThreadState::self()->getCallingPid(); 601 client = registerPid(pid); 602 603 PlaybackThread *effectThread = NULL; 604 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 605 lSessionId = *sessionId; 606 // check if an effect chain with the same session ID is present on another 607 // output thread and move it here. 608 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 609 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 610 if (mPlaybackThreads.keyAt(i) != output) { 611 uint32_t sessions = t->hasAudioSession(lSessionId); 612 if (sessions & PlaybackThread::EFFECT_SESSION) { 613 effectThread = t.get(); 614 break; 615 } 616 } 617 } 618 } else { 619 // if no audio session id is provided, create one here 620 lSessionId = nextUniqueId(); 621 if (sessionId != NULL) { 622 *sessionId = lSessionId; 623 } 624 } 625 ALOGV("createTrack() lSessionId: %d", lSessionId); 626 627 track = thread->createTrack_l(client, streamType, sampleRate, format, 628 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 629 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 630 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 631 632 // move effect chain to this output thread if an effect on same session was waiting 633 // for a track to be created 634 if (lStatus == NO_ERROR && effectThread != NULL) { 635 // no risk of deadlock because AudioFlinger::mLock is held 636 Mutex::Autolock _dl(thread->mLock); 637 Mutex::Autolock _sl(effectThread->mLock); 638 moveEffectChain_l(lSessionId, effectThread, thread, true); 639 } 640 641 // Look for sync events awaiting for a session to be used. 642 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 643 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 644 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 645 if (lStatus == NO_ERROR) { 646 (void) track->setSyncEvent(mPendingSyncEvents[i]); 647 } else { 648 mPendingSyncEvents[i]->cancel(); 649 } 650 mPendingSyncEvents.removeAt(i); 651 i--; 652 } 653 } 654 } 655 656 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 657 } 658 659 if (lStatus != NO_ERROR) { 660 // remove local strong reference to Client before deleting the Track so that the 661 // Client destructor is called by the TrackBase destructor with mClientLock held 662 // Don't hold mClientLock when releasing the reference on the track as the 663 // destructor will acquire it. 664 { 665 Mutex::Autolock _cl(mClientLock); 666 client.clear(); 667 } 668 track.clear(); 669 goto Exit; 670 } 671 672 // return handle to client 673 trackHandle = new TrackHandle(track); 674 675Exit: 676 *status = lStatus; 677 return trackHandle; 678} 679 680uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 681{ 682 Mutex::Autolock _l(mLock); 683 PlaybackThread *thread = checkPlaybackThread_l(output); 684 if (thread == NULL) { 685 ALOGW("sampleRate() unknown thread %d", output); 686 return 0; 687 } 688 return thread->sampleRate(); 689} 690 691audio_format_t AudioFlinger::format(audio_io_handle_t output) const 692{ 693 Mutex::Autolock _l(mLock); 694 PlaybackThread *thread = checkPlaybackThread_l(output); 695 if (thread == NULL) { 696 ALOGW("format() unknown thread %d", output); 697 return AUDIO_FORMAT_INVALID; 698 } 699 return thread->format(); 700} 701 702size_t AudioFlinger::frameCount(audio_io_handle_t output) const 703{ 704 Mutex::Autolock _l(mLock); 705 PlaybackThread *thread = checkPlaybackThread_l(output); 706 if (thread == NULL) { 707 ALOGW("frameCount() unknown thread %d", output); 708 return 0; 709 } 710 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 711 // should examine all callers and fix them to handle smaller counts 712 return thread->frameCount(); 713} 714 715uint32_t AudioFlinger::latency(audio_io_handle_t output) const 716{ 717 Mutex::Autolock _l(mLock); 718 PlaybackThread *thread = checkPlaybackThread_l(output); 719 if (thread == NULL) { 720 ALOGW("latency(): no playback thread found for output handle %d", output); 721 return 0; 722 } 723 return thread->latency(); 724} 725 726status_t AudioFlinger::setMasterVolume(float value) 727{ 728 status_t ret = initCheck(); 729 if (ret != NO_ERROR) { 730 return ret; 731 } 732 733 // check calling permissions 734 if (!settingsAllowed()) { 735 return PERMISSION_DENIED; 736 } 737 738 Mutex::Autolock _l(mLock); 739 mMasterVolume = value; 740 741 // Set master volume in the HALs which support it. 742 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 743 AutoMutex lock(mHardwareLock); 744 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 745 746 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 747 if (dev->canSetMasterVolume()) { 748 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 749 } 750 mHardwareStatus = AUDIO_HW_IDLE; 751 } 752 753 // Now set the master volume in each playback thread. Playback threads 754 // assigned to HALs which do not have master volume support will apply 755 // master volume during the mix operation. Threads with HALs which do 756 // support master volume will simply ignore the setting. 757 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 758 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 759 760 return NO_ERROR; 761} 762 763status_t AudioFlinger::setMode(audio_mode_t mode) 764{ 765 status_t ret = initCheck(); 766 if (ret != NO_ERROR) { 767 return ret; 768 } 769 770 // check calling permissions 771 if (!settingsAllowed()) { 772 return PERMISSION_DENIED; 773 } 774 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 775 ALOGW("Illegal value: setMode(%d)", mode); 776 return BAD_VALUE; 777 } 778 779 { // scope for the lock 780 AutoMutex lock(mHardwareLock); 781 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 782 mHardwareStatus = AUDIO_HW_SET_MODE; 783 ret = dev->set_mode(dev, mode); 784 mHardwareStatus = AUDIO_HW_IDLE; 785 } 786 787 if (NO_ERROR == ret) { 788 Mutex::Autolock _l(mLock); 789 mMode = mode; 790 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 791 mPlaybackThreads.valueAt(i)->setMode(mode); 792 } 793 794 return ret; 795} 796 797status_t AudioFlinger::setMicMute(bool state) 798{ 799 status_t ret = initCheck(); 800 if (ret != NO_ERROR) { 801 return ret; 802 } 803 804 // check calling permissions 805 if (!settingsAllowed()) { 806 return PERMISSION_DENIED; 807 } 808 809 AutoMutex lock(mHardwareLock); 810 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 811 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 812 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 813 status_t result = dev->set_mic_mute(dev, state); 814 if (result != NO_ERROR) { 815 ret = result; 816 } 817 } 818 mHardwareStatus = AUDIO_HW_IDLE; 819 return ret; 820} 821 822bool AudioFlinger::getMicMute() const 823{ 824 status_t ret = initCheck(); 825 if (ret != NO_ERROR) { 826 return false; 827 } 828 829 bool state = AUDIO_MODE_INVALID; 830 AutoMutex lock(mHardwareLock); 831 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 832 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 833 dev->get_mic_mute(dev, &state); 834 mHardwareStatus = AUDIO_HW_IDLE; 835 return state; 836} 837 838status_t AudioFlinger::setMasterMute(bool muted) 839{ 840 status_t ret = initCheck(); 841 if (ret != NO_ERROR) { 842 return ret; 843 } 844 845 // check calling permissions 846 if (!settingsAllowed()) { 847 return PERMISSION_DENIED; 848 } 849 850 Mutex::Autolock _l(mLock); 851 mMasterMute = muted; 852 853 // Set master mute in the HALs which support it. 854 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 855 AutoMutex lock(mHardwareLock); 856 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 857 858 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 859 if (dev->canSetMasterMute()) { 860 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 861 } 862 mHardwareStatus = AUDIO_HW_IDLE; 863 } 864 865 // Now set the master mute in each playback thread. Playback threads 866 // assigned to HALs which do not have master mute support will apply master 867 // mute during the mix operation. Threads with HALs which do support master 868 // mute will simply ignore the setting. 869 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 870 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 871 872 return NO_ERROR; 873} 874 875float AudioFlinger::masterVolume() const 876{ 877 Mutex::Autolock _l(mLock); 878 return masterVolume_l(); 879} 880 881bool AudioFlinger::masterMute() const 882{ 883 Mutex::Autolock _l(mLock); 884 return masterMute_l(); 885} 886 887float AudioFlinger::masterVolume_l() const 888{ 889 return mMasterVolume; 890} 891 892bool AudioFlinger::masterMute_l() const 893{ 894 return mMasterMute; 895} 896 897status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 898 audio_io_handle_t output) 899{ 900 // check calling permissions 901 if (!settingsAllowed()) { 902 return PERMISSION_DENIED; 903 } 904 905 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 906 ALOGE("setStreamVolume() invalid stream %d", stream); 907 return BAD_VALUE; 908 } 909 910 AutoMutex lock(mLock); 911 PlaybackThread *thread = NULL; 912 if (output != AUDIO_IO_HANDLE_NONE) { 913 thread = checkPlaybackThread_l(output); 914 if (thread == NULL) { 915 return BAD_VALUE; 916 } 917 } 918 919 mStreamTypes[stream].volume = value; 920 921 if (thread == NULL) { 922 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 923 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 924 } 925 } else { 926 thread->setStreamVolume(stream, value); 927 } 928 929 return NO_ERROR; 930} 931 932status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 933{ 934 // check calling permissions 935 if (!settingsAllowed()) { 936 return PERMISSION_DENIED; 937 } 938 939 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 940 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 941 ALOGE("setStreamMute() invalid stream %d", stream); 942 return BAD_VALUE; 943 } 944 945 AutoMutex lock(mLock); 946 mStreamTypes[stream].mute = muted; 947 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 948 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 949 950 return NO_ERROR; 951} 952 953float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 954{ 955 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 956 return 0.0f; 957 } 958 959 AutoMutex lock(mLock); 960 float volume; 961 if (output != AUDIO_IO_HANDLE_NONE) { 962 PlaybackThread *thread = checkPlaybackThread_l(output); 963 if (thread == NULL) { 964 return 0.0f; 965 } 966 volume = thread->streamVolume(stream); 967 } else { 968 volume = streamVolume_l(stream); 969 } 970 971 return volume; 972} 973 974bool AudioFlinger::streamMute(audio_stream_type_t stream) const 975{ 976 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 977 return true; 978 } 979 980 AutoMutex lock(mLock); 981 return streamMute_l(stream); 982} 983 984status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 985{ 986 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 987 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 988 989 // check calling permissions 990 if (!settingsAllowed()) { 991 return PERMISSION_DENIED; 992 } 993 994 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 995 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 996 Mutex::Autolock _l(mLock); 997 status_t final_result = NO_ERROR; 998 { 999 AutoMutex lock(mHardwareLock); 1000 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1001 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1002 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1003 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1004 final_result = result ?: final_result; 1005 } 1006 mHardwareStatus = AUDIO_HW_IDLE; 1007 } 1008 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1009 AudioParameter param = AudioParameter(keyValuePairs); 1010 String8 value; 1011 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1012 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1013 if (mBtNrecIsOff != btNrecIsOff) { 1014 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1015 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1016 audio_devices_t device = thread->inDevice(); 1017 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1018 // collect all of the thread's session IDs 1019 KeyedVector<int, bool> ids = thread->sessionIds(); 1020 // suspend effects associated with those session IDs 1021 for (size_t j = 0; j < ids.size(); ++j) { 1022 int sessionId = ids.keyAt(j); 1023 thread->setEffectSuspended(FX_IID_AEC, 1024 suspend, 1025 sessionId); 1026 thread->setEffectSuspended(FX_IID_NS, 1027 suspend, 1028 sessionId); 1029 } 1030 } 1031 mBtNrecIsOff = btNrecIsOff; 1032 } 1033 } 1034 String8 screenState; 1035 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1036 bool isOff = screenState == "off"; 1037 if (isOff != (AudioFlinger::mScreenState & 1)) { 1038 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1039 } 1040 } 1041 return final_result; 1042 } 1043 1044 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1045 // and the thread is exited once the lock is released 1046 sp<ThreadBase> thread; 1047 { 1048 Mutex::Autolock _l(mLock); 1049 thread = checkPlaybackThread_l(ioHandle); 1050 if (thread == 0) { 1051 thread = checkRecordThread_l(ioHandle); 1052 } else if (thread == primaryPlaybackThread_l()) { 1053 // indicate output device change to all input threads for pre processing 1054 AudioParameter param = AudioParameter(keyValuePairs); 1055 int value; 1056 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1057 (value != 0)) { 1058 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1059 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1060 } 1061 } 1062 } 1063 } 1064 if (thread != 0) { 1065 return thread->setParameters(keyValuePairs); 1066 } 1067 return BAD_VALUE; 1068} 1069 1070String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1071{ 1072 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1073 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1074 1075 Mutex::Autolock _l(mLock); 1076 1077 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1078 String8 out_s8; 1079 1080 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1081 char *s; 1082 { 1083 AutoMutex lock(mHardwareLock); 1084 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1085 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1086 s = dev->get_parameters(dev, keys.string()); 1087 mHardwareStatus = AUDIO_HW_IDLE; 1088 } 1089 out_s8 += String8(s ? s : ""); 1090 free(s); 1091 } 1092 return out_s8; 1093 } 1094 1095 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1096 if (playbackThread != NULL) { 1097 return playbackThread->getParameters(keys); 1098 } 1099 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1100 if (recordThread != NULL) { 1101 return recordThread->getParameters(keys); 1102 } 1103 return String8(""); 1104} 1105 1106size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1107 audio_channel_mask_t channelMask) const 1108{ 1109 status_t ret = initCheck(); 1110 if (ret != NO_ERROR) { 1111 return 0; 1112 } 1113 1114 AutoMutex lock(mHardwareLock); 1115 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1116 audio_config_t config; 1117 memset(&config, 0, sizeof(config)); 1118 config.sample_rate = sampleRate; 1119 config.channel_mask = channelMask; 1120 config.format = format; 1121 1122 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1123 size_t size = dev->get_input_buffer_size(dev, &config); 1124 mHardwareStatus = AUDIO_HW_IDLE; 1125 return size; 1126} 1127 1128uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1129{ 1130 Mutex::Autolock _l(mLock); 1131 1132 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1133 if (recordThread != NULL) { 1134 return recordThread->getInputFramesLost(); 1135 } 1136 return 0; 1137} 1138 1139status_t AudioFlinger::setVoiceVolume(float value) 1140{ 1141 status_t ret = initCheck(); 1142 if (ret != NO_ERROR) { 1143 return ret; 1144 } 1145 1146 // check calling permissions 1147 if (!settingsAllowed()) { 1148 return PERMISSION_DENIED; 1149 } 1150 1151 AutoMutex lock(mHardwareLock); 1152 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1153 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1154 ret = dev->set_voice_volume(dev, value); 1155 mHardwareStatus = AUDIO_HW_IDLE; 1156 1157 return ret; 1158} 1159 1160status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1161 audio_io_handle_t output) const 1162{ 1163 status_t status; 1164 1165 Mutex::Autolock _l(mLock); 1166 1167 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1168 if (playbackThread != NULL) { 1169 return playbackThread->getRenderPosition(halFrames, dspFrames); 1170 } 1171 1172 return BAD_VALUE; 1173} 1174 1175void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1176{ 1177 Mutex::Autolock _l(mLock); 1178 if (client == 0) { 1179 return; 1180 } 1181 bool clientAdded = false; 1182 { 1183 Mutex::Autolock _cl(mClientLock); 1184 1185 pid_t pid = IPCThreadState::self()->getCallingPid(); 1186 if (mNotificationClients.indexOfKey(pid) < 0) { 1187 sp<NotificationClient> notificationClient = new NotificationClient(this, 1188 client, 1189 pid); 1190 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1191 1192 mNotificationClients.add(pid, notificationClient); 1193 1194 sp<IBinder> binder = client->asBinder(); 1195 binder->linkToDeath(notificationClient); 1196 clientAdded = true; 1197 } 1198 } 1199 1200 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1201 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1202 if (clientAdded) { 1203 // the config change is always sent from playback or record threads to avoid deadlock 1204 // with AudioSystem::gLock 1205 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1206 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1207 } 1208 1209 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1210 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1211 } 1212 } 1213} 1214 1215void AudioFlinger::removeNotificationClient(pid_t pid) 1216{ 1217 Mutex::Autolock _l(mLock); 1218 { 1219 Mutex::Autolock _cl(mClientLock); 1220 mNotificationClients.removeItem(pid); 1221 } 1222 1223 ALOGV("%d died, releasing its sessions", pid); 1224 size_t num = mAudioSessionRefs.size(); 1225 bool removed = false; 1226 for (size_t i = 0; i< num; ) { 1227 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1228 ALOGV(" pid %d @ %d", ref->mPid, i); 1229 if (ref->mPid == pid) { 1230 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1231 mAudioSessionRefs.removeAt(i); 1232 delete ref; 1233 removed = true; 1234 num--; 1235 } else { 1236 i++; 1237 } 1238 } 1239 if (removed) { 1240 purgeStaleEffects_l(); 1241 } 1242} 1243 1244void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1245{ 1246 Mutex::Autolock _l(mClientLock); 1247 size_t size = mNotificationClients.size(); 1248 for (size_t i = 0; i < size; i++) { 1249 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1250 ioHandle, 1251 param2); 1252 } 1253} 1254 1255// removeClient_l() must be called with AudioFlinger::mClientLock held 1256void AudioFlinger::removeClient_l(pid_t pid) 1257{ 1258 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1259 IPCThreadState::self()->getCallingPid()); 1260 mClients.removeItem(pid); 1261} 1262 1263// getEffectThread_l() must be called with AudioFlinger::mLock held 1264sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1265{ 1266 sp<PlaybackThread> thread; 1267 1268 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1269 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1270 ALOG_ASSERT(thread == 0); 1271 thread = mPlaybackThreads.valueAt(i); 1272 } 1273 } 1274 1275 return thread; 1276} 1277 1278 1279 1280// ---------------------------------------------------------------------------- 1281 1282AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1283 : RefBase(), 1284 mAudioFlinger(audioFlinger), 1285 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1286 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1287 mPid(pid), 1288 mTimedTrackCount(0) 1289{ 1290 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1291} 1292 1293// Client destructor must be called with AudioFlinger::mClientLock held 1294AudioFlinger::Client::~Client() 1295{ 1296 mAudioFlinger->removeClient_l(mPid); 1297} 1298 1299sp<MemoryDealer> AudioFlinger::Client::heap() const 1300{ 1301 return mMemoryDealer; 1302} 1303 1304// Reserve one of the limited slots for a timed audio track associated 1305// with this client 1306bool AudioFlinger::Client::reserveTimedTrack() 1307{ 1308 const int kMaxTimedTracksPerClient = 4; 1309 1310 Mutex::Autolock _l(mTimedTrackLock); 1311 1312 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1313 ALOGW("can not create timed track - pid %d has exceeded the limit", 1314 mPid); 1315 return false; 1316 } 1317 1318 mTimedTrackCount++; 1319 return true; 1320} 1321 1322// Release a slot for a timed audio track 1323void AudioFlinger::Client::releaseTimedTrack() 1324{ 1325 Mutex::Autolock _l(mTimedTrackLock); 1326 mTimedTrackCount--; 1327} 1328 1329// ---------------------------------------------------------------------------- 1330 1331AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1332 const sp<IAudioFlingerClient>& client, 1333 pid_t pid) 1334 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1335{ 1336} 1337 1338AudioFlinger::NotificationClient::~NotificationClient() 1339{ 1340} 1341 1342void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1343{ 1344 sp<NotificationClient> keep(this); 1345 mAudioFlinger->removeNotificationClient(mPid); 1346} 1347 1348 1349// ---------------------------------------------------------------------------- 1350 1351static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1352 return audio_is_remote_submix_device(inDevice); 1353} 1354 1355sp<IAudioRecord> AudioFlinger::openRecord( 1356 audio_io_handle_t input, 1357 uint32_t sampleRate, 1358 audio_format_t format, 1359 audio_channel_mask_t channelMask, 1360 size_t *frameCount, 1361 IAudioFlinger::track_flags_t *flags, 1362 pid_t tid, 1363 int *sessionId, 1364 size_t *notificationFrames, 1365 sp<IMemory>& cblk, 1366 sp<IMemory>& buffers, 1367 status_t *status) 1368{ 1369 sp<RecordThread::RecordTrack> recordTrack; 1370 sp<RecordHandle> recordHandle; 1371 sp<Client> client; 1372 status_t lStatus; 1373 int lSessionId; 1374 1375 cblk.clear(); 1376 buffers.clear(); 1377 1378 // check calling permissions 1379 if (!recordingAllowed()) { 1380 ALOGE("openRecord() permission denied: recording not allowed"); 1381 lStatus = PERMISSION_DENIED; 1382 goto Exit; 1383 } 1384 1385 // further sample rate checks are performed by createRecordTrack_l() 1386 if (sampleRate == 0) { 1387 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1388 lStatus = BAD_VALUE; 1389 goto Exit; 1390 } 1391 1392 // we don't yet support anything other than 16-bit PCM 1393 if (!(audio_is_valid_format(format) && 1394 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1395 ALOGE("openRecord() invalid format %#x", format); 1396 lStatus = BAD_VALUE; 1397 goto Exit; 1398 } 1399 1400 // further channel mask checks are performed by createRecordTrack_l() 1401 if (!audio_is_input_channel(channelMask)) { 1402 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1403 lStatus = BAD_VALUE; 1404 goto Exit; 1405 } 1406 1407 { 1408 Mutex::Autolock _l(mLock); 1409 RecordThread *thread = checkRecordThread_l(input); 1410 if (thread == NULL) { 1411 ALOGE("openRecord() checkRecordThread_l failed"); 1412 lStatus = BAD_VALUE; 1413 goto Exit; 1414 } 1415 1416 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1417 && !captureAudioOutputAllowed()) { 1418 ALOGE("openRecord() permission denied: capture not allowed"); 1419 lStatus = PERMISSION_DENIED; 1420 goto Exit; 1421 } 1422 1423 pid_t pid = IPCThreadState::self()->getCallingPid(); 1424 client = registerPid(pid); 1425 1426 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1427 lSessionId = *sessionId; 1428 } else { 1429 // if no audio session id is provided, create one here 1430 lSessionId = nextUniqueId(); 1431 if (sessionId != NULL) { 1432 *sessionId = lSessionId; 1433 } 1434 } 1435 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1436 1437 // TODO: the uid should be passed in as a parameter to openRecord 1438 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1439 frameCount, lSessionId, notificationFrames, 1440 IPCThreadState::self()->getCallingUid(), 1441 flags, tid, &lStatus); 1442 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1443 1444 if (lStatus == NO_ERROR) { 1445 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1446 // session and move it to this thread. 1447 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1448 if (chain != 0) { 1449 Mutex::Autolock _l(thread->mLock); 1450 thread->addEffectChain_l(chain); 1451 } 1452 } 1453 } 1454 1455 if (lStatus != NO_ERROR) { 1456 // remove local strong reference to Client before deleting the RecordTrack so that the 1457 // Client destructor is called by the TrackBase destructor with mClientLock held 1458 // Don't hold mClientLock when releasing the reference on the track as the 1459 // destructor will acquire it. 1460 { 1461 Mutex::Autolock _cl(mClientLock); 1462 client.clear(); 1463 } 1464 recordTrack.clear(); 1465 goto Exit; 1466 } 1467 1468 cblk = recordTrack->getCblk(); 1469 buffers = recordTrack->getBuffers(); 1470 1471 // return handle to client 1472 recordHandle = new RecordHandle(recordTrack); 1473 1474Exit: 1475 *status = lStatus; 1476 return recordHandle; 1477} 1478 1479 1480 1481// ---------------------------------------------------------------------------- 1482 1483audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1484{ 1485 if (name == NULL) { 1486 return 0; 1487 } 1488 if (!settingsAllowed()) { 1489 return 0; 1490 } 1491 Mutex::Autolock _l(mLock); 1492 return loadHwModule_l(name); 1493} 1494 1495// loadHwModule_l() must be called with AudioFlinger::mLock held 1496audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1497{ 1498 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1499 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1500 ALOGW("loadHwModule() module %s already loaded", name); 1501 return mAudioHwDevs.keyAt(i); 1502 } 1503 } 1504 1505 audio_hw_device_t *dev; 1506 1507 int rc = load_audio_interface(name, &dev); 1508 if (rc) { 1509 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1510 return 0; 1511 } 1512 1513 mHardwareStatus = AUDIO_HW_INIT; 1514 rc = dev->init_check(dev); 1515 mHardwareStatus = AUDIO_HW_IDLE; 1516 if (rc) { 1517 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1518 return 0; 1519 } 1520 1521 // Check and cache this HAL's level of support for master mute and master 1522 // volume. If this is the first HAL opened, and it supports the get 1523 // methods, use the initial values provided by the HAL as the current 1524 // master mute and volume settings. 1525 1526 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1527 { // scope for auto-lock pattern 1528 AutoMutex lock(mHardwareLock); 1529 1530 if (0 == mAudioHwDevs.size()) { 1531 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1532 if (NULL != dev->get_master_volume) { 1533 float mv; 1534 if (OK == dev->get_master_volume(dev, &mv)) { 1535 mMasterVolume = mv; 1536 } 1537 } 1538 1539 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1540 if (NULL != dev->get_master_mute) { 1541 bool mm; 1542 if (OK == dev->get_master_mute(dev, &mm)) { 1543 mMasterMute = mm; 1544 } 1545 } 1546 } 1547 1548 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1549 if ((NULL != dev->set_master_volume) && 1550 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1551 flags = static_cast<AudioHwDevice::Flags>(flags | 1552 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1553 } 1554 1555 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1556 if ((NULL != dev->set_master_mute) && 1557 (OK == dev->set_master_mute(dev, mMasterMute))) { 1558 flags = static_cast<AudioHwDevice::Flags>(flags | 1559 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1560 } 1561 1562 mHardwareStatus = AUDIO_HW_IDLE; 1563 } 1564 1565 audio_module_handle_t handle = nextUniqueId(); 1566 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1567 1568 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1569 name, dev->common.module->name, dev->common.module->id, handle); 1570 1571 return handle; 1572 1573} 1574 1575// ---------------------------------------------------------------------------- 1576 1577uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1578{ 1579 Mutex::Autolock _l(mLock); 1580 PlaybackThread *thread = primaryPlaybackThread_l(); 1581 return thread != NULL ? thread->sampleRate() : 0; 1582} 1583 1584size_t AudioFlinger::getPrimaryOutputFrameCount() 1585{ 1586 Mutex::Autolock _l(mLock); 1587 PlaybackThread *thread = primaryPlaybackThread_l(); 1588 return thread != NULL ? thread->frameCountHAL() : 0; 1589} 1590 1591// ---------------------------------------------------------------------------- 1592 1593status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1594{ 1595 uid_t uid = IPCThreadState::self()->getCallingUid(); 1596 if (uid != AID_SYSTEM) { 1597 return PERMISSION_DENIED; 1598 } 1599 Mutex::Autolock _l(mLock); 1600 if (mIsDeviceTypeKnown) { 1601 return INVALID_OPERATION; 1602 } 1603 mIsLowRamDevice = isLowRamDevice; 1604 mIsDeviceTypeKnown = true; 1605 return NO_ERROR; 1606} 1607 1608audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1609{ 1610 Mutex::Autolock _l(mLock); 1611 1612 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1613 if (index >= 0) { 1614 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1615 mHwAvSyncIds.valueAt(index), sessionId); 1616 return mHwAvSyncIds.valueAt(index); 1617 } 1618 1619 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1620 if (dev == NULL) { 1621 return AUDIO_HW_SYNC_INVALID; 1622 } 1623 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1624 AudioParameter param = AudioParameter(String8(reply)); 1625 free(reply); 1626 1627 int value; 1628 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1629 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1630 return AUDIO_HW_SYNC_INVALID; 1631 } 1632 1633 // allow only one session for a given HW A/V sync ID. 1634 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1635 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1636 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1637 value, mHwAvSyncIds.keyAt(i)); 1638 mHwAvSyncIds.removeItemsAt(i); 1639 break; 1640 } 1641 } 1642 1643 mHwAvSyncIds.add(sessionId, value); 1644 1645 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1646 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1647 uint32_t sessions = thread->hasAudioSession(sessionId); 1648 if (sessions & PlaybackThread::TRACK_SESSION) { 1649 AudioParameter param = AudioParameter(); 1650 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1651 thread->setParameters(param.toString()); 1652 break; 1653 } 1654 } 1655 1656 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1657 return (audio_hw_sync_t)value; 1658} 1659 1660// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1661void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1662{ 1663 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1664 if (index >= 0) { 1665 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1666 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1667 AudioParameter param = AudioParameter(); 1668 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1669 thread->setParameters(param.toString()); 1670 } 1671} 1672 1673 1674// ---------------------------------------------------------------------------- 1675 1676 1677sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1678 audio_io_handle_t *output, 1679 audio_config_t *config, 1680 audio_devices_t devices, 1681 const String8& address, 1682 audio_output_flags_t flags) 1683{ 1684 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1685 if (outHwDev == NULL) { 1686 return 0; 1687 } 1688 1689 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1690 if (*output == AUDIO_IO_HANDLE_NONE) { 1691 *output = nextUniqueId(); 1692 } 1693 1694 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1695 1696 audio_stream_out_t *outStream = NULL; 1697 1698 // FOR TESTING ONLY: 1699 // This if statement allows overriding the audio policy settings 1700 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1701 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1702 // Check only for Normal Mixing mode 1703 if (kEnableExtendedPrecision) { 1704 // Specify format (uncomment one below to choose) 1705 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1706 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1707 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1708 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1709 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1710 } 1711 if (kEnableExtendedChannels) { 1712 // Specify channel mask (uncomment one below to choose) 1713 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1714 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1715 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1716 } 1717 } 1718 1719 status_t status = hwDevHal->open_output_stream(hwDevHal, 1720 *output, 1721 devices, 1722 flags, 1723 config, 1724 &outStream, 1725 address.string()); 1726 1727 mHardwareStatus = AUDIO_HW_IDLE; 1728 ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, " 1729 "channelMask %#x, status %d", 1730 outStream, 1731 config->sample_rate, 1732 config->format, 1733 config->channel_mask, 1734 status); 1735 1736 if (status == NO_ERROR && outStream != NULL) { 1737 AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags); 1738 1739 PlaybackThread *thread; 1740 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1741 thread = new OffloadThread(this, outputStream, *output, devices); 1742 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1743 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1744 || !isValidPcmSinkFormat(config->format) 1745 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1746 thread = new DirectOutputThread(this, outputStream, *output, devices); 1747 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1748 } else { 1749 thread = new MixerThread(this, outputStream, *output, devices); 1750 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1751 } 1752 mPlaybackThreads.add(*output, thread); 1753 return thread; 1754 } 1755 1756 return 0; 1757} 1758 1759status_t AudioFlinger::openOutput(audio_module_handle_t module, 1760 audio_io_handle_t *output, 1761 audio_config_t *config, 1762 audio_devices_t *devices, 1763 const String8& address, 1764 uint32_t *latencyMs, 1765 audio_output_flags_t flags) 1766{ 1767 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1768 module, 1769 (devices != NULL) ? *devices : 0, 1770 config->sample_rate, 1771 config->format, 1772 config->channel_mask, 1773 flags); 1774 1775 if (*devices == AUDIO_DEVICE_NONE) { 1776 return BAD_VALUE; 1777 } 1778 1779 Mutex::Autolock _l(mLock); 1780 1781 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1782 if (thread != 0) { 1783 *latencyMs = thread->latency(); 1784 1785 // notify client processes of the new output creation 1786 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1787 1788 // the first primary output opened designates the primary hw device 1789 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1790 ALOGI("Using module %d has the primary audio interface", module); 1791 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1792 1793 AutoMutex lock(mHardwareLock); 1794 mHardwareStatus = AUDIO_HW_SET_MODE; 1795 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1796 mHardwareStatus = AUDIO_HW_IDLE; 1797 1798 mPrimaryOutputSampleRate = config->sample_rate; 1799 } 1800 return NO_ERROR; 1801 } 1802 1803 return NO_INIT; 1804} 1805 1806audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1807 audio_io_handle_t output2) 1808{ 1809 Mutex::Autolock _l(mLock); 1810 MixerThread *thread1 = checkMixerThread_l(output1); 1811 MixerThread *thread2 = checkMixerThread_l(output2); 1812 1813 if (thread1 == NULL || thread2 == NULL) { 1814 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1815 output2); 1816 return AUDIO_IO_HANDLE_NONE; 1817 } 1818 1819 audio_io_handle_t id = nextUniqueId(); 1820 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1821 thread->addOutputTrack(thread2); 1822 mPlaybackThreads.add(id, thread); 1823 // notify client processes of the new output creation 1824 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1825 return id; 1826} 1827 1828status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1829{ 1830 return closeOutput_nonvirtual(output); 1831} 1832 1833status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1834{ 1835 // keep strong reference on the playback thread so that 1836 // it is not destroyed while exit() is executed 1837 sp<PlaybackThread> thread; 1838 { 1839 Mutex::Autolock _l(mLock); 1840 thread = checkPlaybackThread_l(output); 1841 if (thread == NULL) { 1842 return BAD_VALUE; 1843 } 1844 1845 ALOGV("closeOutput() %d", output); 1846 1847 if (thread->type() == ThreadBase::MIXER) { 1848 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1849 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1850 DuplicatingThread *dupThread = 1851 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1852 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1853 1854 } 1855 } 1856 } 1857 1858 1859 mPlaybackThreads.removeItem(output); 1860 // save all effects to the default thread 1861 if (mPlaybackThreads.size()) { 1862 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1863 if (dstThread != NULL) { 1864 // audioflinger lock is held here so the acquisition order of thread locks does not 1865 // matter 1866 Mutex::Autolock _dl(dstThread->mLock); 1867 Mutex::Autolock _sl(thread->mLock); 1868 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1869 for (size_t i = 0; i < effectChains.size(); i ++) { 1870 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1871 } 1872 } 1873 } 1874 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1875 } 1876 thread->exit(); 1877 // The thread entity (active unit of execution) is no longer running here, 1878 // but the ThreadBase container still exists. 1879 1880 if (thread->type() != ThreadBase::DUPLICATING) { 1881 closeOutputFinish(thread); 1882 } 1883 1884 return NO_ERROR; 1885} 1886 1887void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1888{ 1889 AudioStreamOut *out = thread->clearOutput(); 1890 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1891 // from now on thread->mOutput is NULL 1892 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1893 delete out; 1894} 1895 1896void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1897{ 1898 mPlaybackThreads.removeItem(thread->mId); 1899 thread->exit(); 1900 closeOutputFinish(thread); 1901} 1902 1903status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1904{ 1905 Mutex::Autolock _l(mLock); 1906 PlaybackThread *thread = checkPlaybackThread_l(output); 1907 1908 if (thread == NULL) { 1909 return BAD_VALUE; 1910 } 1911 1912 ALOGV("suspendOutput() %d", output); 1913 thread->suspend(); 1914 1915 return NO_ERROR; 1916} 1917 1918status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1919{ 1920 Mutex::Autolock _l(mLock); 1921 PlaybackThread *thread = checkPlaybackThread_l(output); 1922 1923 if (thread == NULL) { 1924 return BAD_VALUE; 1925 } 1926 1927 ALOGV("restoreOutput() %d", output); 1928 1929 thread->restore(); 1930 1931 return NO_ERROR; 1932} 1933 1934status_t AudioFlinger::openInput(audio_module_handle_t module, 1935 audio_io_handle_t *input, 1936 audio_config_t *config, 1937 audio_devices_t *device, 1938 const String8& address, 1939 audio_source_t source, 1940 audio_input_flags_t flags) 1941{ 1942 Mutex::Autolock _l(mLock); 1943 1944 if (*device == AUDIO_DEVICE_NONE) { 1945 return BAD_VALUE; 1946 } 1947 1948 sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags); 1949 1950 if (thread != 0) { 1951 // notify client processes of the new input creation 1952 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1953 return NO_ERROR; 1954 } 1955 return NO_INIT; 1956} 1957 1958sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 1959 audio_io_handle_t *input, 1960 audio_config_t *config, 1961 audio_devices_t device, 1962 const String8& address, 1963 audio_source_t source, 1964 audio_input_flags_t flags) 1965{ 1966 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device); 1967 if (inHwDev == NULL) { 1968 *input = AUDIO_IO_HANDLE_NONE; 1969 return 0; 1970 } 1971 1972 if (*input == AUDIO_IO_HANDLE_NONE) { 1973 *input = nextUniqueId(); 1974 } 1975 1976 audio_config_t halconfig = *config; 1977 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1978 audio_stream_in_t *inStream = NULL; 1979 status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1980 &inStream, flags, address.string(), source); 1981 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 1982 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 1983 inStream, 1984 halconfig.sample_rate, 1985 halconfig.format, 1986 halconfig.channel_mask, 1987 flags, 1988 status, address.string()); 1989 1990 // If the input could not be opened with the requested parameters and we can handle the 1991 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1992 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1993 if (status == BAD_VALUE && 1994 config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT && 1995 (halconfig.sample_rate <= 2 * config->sample_rate) && 1996 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 1997 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 1998 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1999 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2000 inStream = NULL; 2001 status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 2002 &inStream, flags, address.string(), source); 2003 // FIXME log this new status; HAL should not propose any further changes 2004 } 2005 2006 if (status == NO_ERROR && inStream != NULL) { 2007 2008#ifdef TEE_SINK 2009 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2010 // or (re-)create if current Pipe is idle and does not match the new format 2011 sp<NBAIO_Sink> teeSink; 2012 enum { 2013 TEE_SINK_NO, // don't copy input 2014 TEE_SINK_NEW, // copy input using a new pipe 2015 TEE_SINK_OLD, // copy input using an existing pipe 2016 } kind; 2017 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2018 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2019 if (!mTeeSinkInputEnabled) { 2020 kind = TEE_SINK_NO; 2021 } else if (!Format_isValid(format)) { 2022 kind = TEE_SINK_NO; 2023 } else if (mRecordTeeSink == 0) { 2024 kind = TEE_SINK_NEW; 2025 } else if (mRecordTeeSink->getStrongCount() != 1) { 2026 kind = TEE_SINK_NO; 2027 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2028 kind = TEE_SINK_OLD; 2029 } else { 2030 kind = TEE_SINK_NEW; 2031 } 2032 switch (kind) { 2033 case TEE_SINK_NEW: { 2034 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2035 size_t numCounterOffers = 0; 2036 const NBAIO_Format offers[1] = {format}; 2037 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2038 ALOG_ASSERT(index == 0); 2039 PipeReader *pipeReader = new PipeReader(*pipe); 2040 numCounterOffers = 0; 2041 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2042 ALOG_ASSERT(index == 0); 2043 mRecordTeeSink = pipe; 2044 mRecordTeeSource = pipeReader; 2045 teeSink = pipe; 2046 } 2047 break; 2048 case TEE_SINK_OLD: 2049 teeSink = mRecordTeeSink; 2050 break; 2051 case TEE_SINK_NO: 2052 default: 2053 break; 2054 } 2055#endif 2056 2057 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2058 2059 // Start record thread 2060 // RecordThread requires both input and output device indication to forward to audio 2061 // pre processing modules 2062 sp<RecordThread> thread = new RecordThread(this, 2063 inputStream, 2064 *input, 2065 primaryOutputDevice_l(), 2066 device 2067#ifdef TEE_SINK 2068 , teeSink 2069#endif 2070 ); 2071 mRecordThreads.add(*input, thread); 2072 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2073 return thread; 2074 } 2075 2076 *input = AUDIO_IO_HANDLE_NONE; 2077 return 0; 2078} 2079 2080status_t AudioFlinger::closeInput(audio_io_handle_t input) 2081{ 2082 return closeInput_nonvirtual(input); 2083} 2084 2085status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2086{ 2087 // keep strong reference on the record thread so that 2088 // it is not destroyed while exit() is executed 2089 sp<RecordThread> thread; 2090 { 2091 Mutex::Autolock _l(mLock); 2092 thread = checkRecordThread_l(input); 2093 if (thread == 0) { 2094 return BAD_VALUE; 2095 } 2096 2097 ALOGV("closeInput() %d", input); 2098 2099 // If we still have effect chains, it means that a client still holds a handle 2100 // on at least one effect. We must either move the chain to an existing thread with the 2101 // same session ID or put it aside in case a new record thread is opened for a 2102 // new capture on the same session 2103 sp<EffectChain> chain; 2104 { 2105 Mutex::Autolock _sl(thread->mLock); 2106 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2107 // Note: maximum one chain per record thread 2108 if (effectChains.size() != 0) { 2109 chain = effectChains[0]; 2110 } 2111 } 2112 if (chain != 0) { 2113 // first check if a record thread is already opened with a client on the same session. 2114 // This should only happen in case of overlap between one thread tear down and the 2115 // creation of its replacement 2116 size_t i; 2117 for (i = 0; i < mRecordThreads.size(); i++) { 2118 sp<RecordThread> t = mRecordThreads.valueAt(i); 2119 if (t == thread) { 2120 continue; 2121 } 2122 if (t->hasAudioSession(chain->sessionId()) != 0) { 2123 Mutex::Autolock _l(t->mLock); 2124 ALOGV("closeInput() found thread %d for effect session %d", 2125 t->id(), chain->sessionId()); 2126 t->addEffectChain_l(chain); 2127 break; 2128 } 2129 } 2130 // put the chain aside if we could not find a record thread with the same session id. 2131 if (i == mRecordThreads.size()) { 2132 putOrphanEffectChain_l(chain); 2133 } 2134 } 2135 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 2136 mRecordThreads.removeItem(input); 2137 } 2138 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2139 // we have a different lock for notification client 2140 closeInputFinish(thread); 2141 return NO_ERROR; 2142} 2143 2144void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2145{ 2146 thread->exit(); 2147 AudioStreamIn *in = thread->clearInput(); 2148 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2149 // from now on thread->mInput is NULL 2150 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2151 delete in; 2152} 2153 2154void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2155{ 2156 mRecordThreads.removeItem(thread->mId); 2157 closeInputFinish(thread); 2158} 2159 2160status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2161{ 2162 Mutex::Autolock _l(mLock); 2163 ALOGV("invalidateStream() stream %d", stream); 2164 2165 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2166 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2167 thread->invalidateTracks(stream); 2168 } 2169 2170 return NO_ERROR; 2171} 2172 2173 2174audio_unique_id_t AudioFlinger::newAudioUniqueId() 2175{ 2176 return nextUniqueId(); 2177} 2178 2179void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2180{ 2181 Mutex::Autolock _l(mLock); 2182 pid_t caller = IPCThreadState::self()->getCallingPid(); 2183 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2184 if (pid != -1 && (caller == getpid_cached)) { 2185 caller = pid; 2186 } 2187 2188 { 2189 Mutex::Autolock _cl(mClientLock); 2190 // Ignore requests received from processes not known as notification client. The request 2191 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2192 // called from a different pid leaving a stale session reference. Also we don't know how 2193 // to clear this reference if the client process dies. 2194 if (mNotificationClients.indexOfKey(caller) < 0) { 2195 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2196 return; 2197 } 2198 } 2199 2200 size_t num = mAudioSessionRefs.size(); 2201 for (size_t i = 0; i< num; i++) { 2202 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2203 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2204 ref->mCnt++; 2205 ALOGV(" incremented refcount to %d", ref->mCnt); 2206 return; 2207 } 2208 } 2209 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2210 ALOGV(" added new entry for %d", audioSession); 2211} 2212 2213void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2214{ 2215 Mutex::Autolock _l(mLock); 2216 pid_t caller = IPCThreadState::self()->getCallingPid(); 2217 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2218 if (pid != -1 && (caller == getpid_cached)) { 2219 caller = pid; 2220 } 2221 size_t num = mAudioSessionRefs.size(); 2222 for (size_t i = 0; i< num; i++) { 2223 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2224 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2225 ref->mCnt--; 2226 ALOGV(" decremented refcount to %d", ref->mCnt); 2227 if (ref->mCnt == 0) { 2228 mAudioSessionRefs.removeAt(i); 2229 delete ref; 2230 purgeStaleEffects_l(); 2231 } 2232 return; 2233 } 2234 } 2235 // If the caller is mediaserver it is likely that the session being released was acquired 2236 // on behalf of a process not in notification clients and we ignore the warning. 2237 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2238} 2239 2240void AudioFlinger::purgeStaleEffects_l() { 2241 2242 ALOGV("purging stale effects"); 2243 2244 Vector< sp<EffectChain> > chains; 2245 2246 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2247 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2248 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2249 sp<EffectChain> ec = t->mEffectChains[j]; 2250 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2251 chains.push(ec); 2252 } 2253 } 2254 } 2255 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2256 sp<RecordThread> t = mRecordThreads.valueAt(i); 2257 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2258 sp<EffectChain> ec = t->mEffectChains[j]; 2259 chains.push(ec); 2260 } 2261 } 2262 2263 for (size_t i = 0; i < chains.size(); i++) { 2264 sp<EffectChain> ec = chains[i]; 2265 int sessionid = ec->sessionId(); 2266 sp<ThreadBase> t = ec->mThread.promote(); 2267 if (t == 0) { 2268 continue; 2269 } 2270 size_t numsessionrefs = mAudioSessionRefs.size(); 2271 bool found = false; 2272 for (size_t k = 0; k < numsessionrefs; k++) { 2273 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2274 if (ref->mSessionid == sessionid) { 2275 ALOGV(" session %d still exists for %d with %d refs", 2276 sessionid, ref->mPid, ref->mCnt); 2277 found = true; 2278 break; 2279 } 2280 } 2281 if (!found) { 2282 Mutex::Autolock _l(t->mLock); 2283 // remove all effects from the chain 2284 while (ec->mEffects.size()) { 2285 sp<EffectModule> effect = ec->mEffects[0]; 2286 effect->unPin(); 2287 t->removeEffect_l(effect); 2288 if (effect->purgeHandles()) { 2289 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2290 } 2291 AudioSystem::unregisterEffect(effect->id()); 2292 } 2293 } 2294 } 2295 return; 2296} 2297 2298// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2299AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2300{ 2301 return mPlaybackThreads.valueFor(output).get(); 2302} 2303 2304// checkMixerThread_l() must be called with AudioFlinger::mLock held 2305AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2306{ 2307 PlaybackThread *thread = checkPlaybackThread_l(output); 2308 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2309} 2310 2311// checkRecordThread_l() must be called with AudioFlinger::mLock held 2312AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2313{ 2314 return mRecordThreads.valueFor(input).get(); 2315} 2316 2317uint32_t AudioFlinger::nextUniqueId() 2318{ 2319 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2320} 2321 2322AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2323{ 2324 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2325 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2326 AudioStreamOut *output = thread->getOutput(); 2327 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2328 return thread; 2329 } 2330 } 2331 return NULL; 2332} 2333 2334audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2335{ 2336 PlaybackThread *thread = primaryPlaybackThread_l(); 2337 2338 if (thread == NULL) { 2339 return 0; 2340 } 2341 2342 return thread->outDevice(); 2343} 2344 2345sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2346 int triggerSession, 2347 int listenerSession, 2348 sync_event_callback_t callBack, 2349 wp<RefBase> cookie) 2350{ 2351 Mutex::Autolock _l(mLock); 2352 2353 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2354 status_t playStatus = NAME_NOT_FOUND; 2355 status_t recStatus = NAME_NOT_FOUND; 2356 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2357 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2358 if (playStatus == NO_ERROR) { 2359 return event; 2360 } 2361 } 2362 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2363 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2364 if (recStatus == NO_ERROR) { 2365 return event; 2366 } 2367 } 2368 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2369 mPendingSyncEvents.add(event); 2370 } else { 2371 ALOGV("createSyncEvent() invalid event %d", event->type()); 2372 event.clear(); 2373 } 2374 return event; 2375} 2376 2377// ---------------------------------------------------------------------------- 2378// Effect management 2379// ---------------------------------------------------------------------------- 2380 2381 2382status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2383{ 2384 Mutex::Autolock _l(mLock); 2385 return EffectQueryNumberEffects(numEffects); 2386} 2387 2388status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2389{ 2390 Mutex::Autolock _l(mLock); 2391 return EffectQueryEffect(index, descriptor); 2392} 2393 2394status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2395 effect_descriptor_t *descriptor) const 2396{ 2397 Mutex::Autolock _l(mLock); 2398 return EffectGetDescriptor(pUuid, descriptor); 2399} 2400 2401 2402sp<IEffect> AudioFlinger::createEffect( 2403 effect_descriptor_t *pDesc, 2404 const sp<IEffectClient>& effectClient, 2405 int32_t priority, 2406 audio_io_handle_t io, 2407 int sessionId, 2408 status_t *status, 2409 int *id, 2410 int *enabled) 2411{ 2412 status_t lStatus = NO_ERROR; 2413 sp<EffectHandle> handle; 2414 effect_descriptor_t desc; 2415 2416 pid_t pid = IPCThreadState::self()->getCallingPid(); 2417 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2418 pid, effectClient.get(), priority, sessionId, io); 2419 2420 if (pDesc == NULL) { 2421 lStatus = BAD_VALUE; 2422 goto Exit; 2423 } 2424 2425 // check audio settings permission for global effects 2426 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2427 lStatus = PERMISSION_DENIED; 2428 goto Exit; 2429 } 2430 2431 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2432 // that can only be created by audio policy manager (running in same process) 2433 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2434 lStatus = PERMISSION_DENIED; 2435 goto Exit; 2436 } 2437 2438 { 2439 if (!EffectIsNullUuid(&pDesc->uuid)) { 2440 // if uuid is specified, request effect descriptor 2441 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2442 if (lStatus < 0) { 2443 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2444 goto Exit; 2445 } 2446 } else { 2447 // if uuid is not specified, look for an available implementation 2448 // of the required type in effect factory 2449 if (EffectIsNullUuid(&pDesc->type)) { 2450 ALOGW("createEffect() no effect type"); 2451 lStatus = BAD_VALUE; 2452 goto Exit; 2453 } 2454 uint32_t numEffects = 0; 2455 effect_descriptor_t d; 2456 d.flags = 0; // prevent compiler warning 2457 bool found = false; 2458 2459 lStatus = EffectQueryNumberEffects(&numEffects); 2460 if (lStatus < 0) { 2461 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2462 goto Exit; 2463 } 2464 for (uint32_t i = 0; i < numEffects; i++) { 2465 lStatus = EffectQueryEffect(i, &desc); 2466 if (lStatus < 0) { 2467 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2468 continue; 2469 } 2470 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2471 // If matching type found save effect descriptor. If the session is 2472 // 0 and the effect is not auxiliary, continue enumeration in case 2473 // an auxiliary version of this effect type is available 2474 found = true; 2475 d = desc; 2476 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2477 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2478 break; 2479 } 2480 } 2481 } 2482 if (!found) { 2483 lStatus = BAD_VALUE; 2484 ALOGW("createEffect() effect not found"); 2485 goto Exit; 2486 } 2487 // For same effect type, chose auxiliary version over insert version if 2488 // connect to output mix (Compliance to OpenSL ES) 2489 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2490 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2491 desc = d; 2492 } 2493 } 2494 2495 // Do not allow auxiliary effects on a session different from 0 (output mix) 2496 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2497 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2498 lStatus = INVALID_OPERATION; 2499 goto Exit; 2500 } 2501 2502 // check recording permission for visualizer 2503 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2504 !recordingAllowed()) { 2505 lStatus = PERMISSION_DENIED; 2506 goto Exit; 2507 } 2508 2509 // return effect descriptor 2510 *pDesc = desc; 2511 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2512 // if the output returned by getOutputForEffect() is removed before we lock the 2513 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2514 // and we will exit safely 2515 io = AudioSystem::getOutputForEffect(&desc); 2516 ALOGV("createEffect got output %d", io); 2517 } 2518 2519 Mutex::Autolock _l(mLock); 2520 2521 // If output is not specified try to find a matching audio session ID in one of the 2522 // output threads. 2523 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2524 // because of code checking output when entering the function. 2525 // Note: io is never 0 when creating an effect on an input 2526 if (io == AUDIO_IO_HANDLE_NONE) { 2527 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2528 // output must be specified by AudioPolicyManager when using session 2529 // AUDIO_SESSION_OUTPUT_STAGE 2530 lStatus = BAD_VALUE; 2531 goto Exit; 2532 } 2533 // look for the thread where the specified audio session is present 2534 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2535 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2536 io = mPlaybackThreads.keyAt(i); 2537 break; 2538 } 2539 } 2540 if (io == 0) { 2541 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2542 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2543 io = mRecordThreads.keyAt(i); 2544 break; 2545 } 2546 } 2547 } 2548 // If no output thread contains the requested session ID, default to 2549 // first output. The effect chain will be moved to the correct output 2550 // thread when a track with the same session ID is created 2551 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2552 io = mPlaybackThreads.keyAt(0); 2553 } 2554 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2555 } 2556 ThreadBase *thread = checkRecordThread_l(io); 2557 if (thread == NULL) { 2558 thread = checkPlaybackThread_l(io); 2559 if (thread == NULL) { 2560 ALOGE("createEffect() unknown output thread"); 2561 lStatus = BAD_VALUE; 2562 goto Exit; 2563 } 2564 } else { 2565 // Check if one effect chain was awaiting for an effect to be created on this 2566 // session and used it instead of creating a new one. 2567 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2568 if (chain != 0) { 2569 Mutex::Autolock _l(thread->mLock); 2570 thread->addEffectChain_l(chain); 2571 } 2572 } 2573 2574 sp<Client> client = registerPid(pid); 2575 2576 // create effect on selected output thread 2577 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2578 &desc, enabled, &lStatus); 2579 if (handle != 0 && id != NULL) { 2580 *id = handle->id(); 2581 } 2582 if (handle == 0) { 2583 // remove local strong reference to Client with mClientLock held 2584 Mutex::Autolock _cl(mClientLock); 2585 client.clear(); 2586 } 2587 } 2588 2589Exit: 2590 *status = lStatus; 2591 return handle; 2592} 2593 2594status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2595 audio_io_handle_t dstOutput) 2596{ 2597 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2598 sessionId, srcOutput, dstOutput); 2599 Mutex::Autolock _l(mLock); 2600 if (srcOutput == dstOutput) { 2601 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2602 return NO_ERROR; 2603 } 2604 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2605 if (srcThread == NULL) { 2606 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2607 return BAD_VALUE; 2608 } 2609 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2610 if (dstThread == NULL) { 2611 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2612 return BAD_VALUE; 2613 } 2614 2615 Mutex::Autolock _dl(dstThread->mLock); 2616 Mutex::Autolock _sl(srcThread->mLock); 2617 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2618} 2619 2620// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2621status_t AudioFlinger::moveEffectChain_l(int sessionId, 2622 AudioFlinger::PlaybackThread *srcThread, 2623 AudioFlinger::PlaybackThread *dstThread, 2624 bool reRegister) 2625{ 2626 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2627 sessionId, srcThread, dstThread); 2628 2629 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2630 if (chain == 0) { 2631 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2632 sessionId, srcThread); 2633 return INVALID_OPERATION; 2634 } 2635 2636 // Check whether the destination thread has a channel count of FCC_2, which is 2637 // currently required for (most) effects. Prevent moving the effect chain here rather 2638 // than disabling the addEffect_l() call in dstThread below. 2639 if ((dstThread->type() == ThreadBase::MIXER || dstThread->type() == ThreadBase::DUPLICATING) && 2640 dstThread->mChannelCount != FCC_2) { 2641 ALOGW("moveEffectChain_l() effect chain failed because" 2642 " destination thread %p channel count(%u) != %u", 2643 dstThread, dstThread->mChannelCount, FCC_2); 2644 return INVALID_OPERATION; 2645 } 2646 2647 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2648 // so that a new chain is created with correct parameters when first effect is added. This is 2649 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2650 // removed. 2651 srcThread->removeEffectChain_l(chain); 2652 2653 // transfer all effects one by one so that new effect chain is created on new thread with 2654 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2655 sp<EffectChain> dstChain; 2656 uint32_t strategy = 0; // prevent compiler warning 2657 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2658 Vector< sp<EffectModule> > removed; 2659 status_t status = NO_ERROR; 2660 while (effect != 0) { 2661 srcThread->removeEffect_l(effect); 2662 removed.add(effect); 2663 status = dstThread->addEffect_l(effect); 2664 if (status != NO_ERROR) { 2665 break; 2666 } 2667 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2668 if (effect->state() == EffectModule::ACTIVE || 2669 effect->state() == EffectModule::STOPPING) { 2670 effect->start(); 2671 } 2672 // if the move request is not received from audio policy manager, the effect must be 2673 // re-registered with the new strategy and output 2674 if (dstChain == 0) { 2675 dstChain = effect->chain().promote(); 2676 if (dstChain == 0) { 2677 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2678 status = NO_INIT; 2679 break; 2680 } 2681 strategy = dstChain->strategy(); 2682 } 2683 if (reRegister) { 2684 AudioSystem::unregisterEffect(effect->id()); 2685 AudioSystem::registerEffect(&effect->desc(), 2686 dstThread->id(), 2687 strategy, 2688 sessionId, 2689 effect->id()); 2690 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2691 } 2692 effect = chain->getEffectFromId_l(0); 2693 } 2694 2695 if (status != NO_ERROR) { 2696 for (size_t i = 0; i < removed.size(); i++) { 2697 srcThread->addEffect_l(removed[i]); 2698 if (dstChain != 0 && reRegister) { 2699 AudioSystem::unregisterEffect(removed[i]->id()); 2700 AudioSystem::registerEffect(&removed[i]->desc(), 2701 srcThread->id(), 2702 strategy, 2703 sessionId, 2704 removed[i]->id()); 2705 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2706 } 2707 } 2708 } 2709 2710 return status; 2711} 2712 2713bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2714{ 2715 if (mGlobalEffectEnableTime != 0 && 2716 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2717 return true; 2718 } 2719 2720 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2721 sp<EffectChain> ec = 2722 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2723 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2724 return true; 2725 } 2726 } 2727 return false; 2728} 2729 2730void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2731{ 2732 Mutex::Autolock _l(mLock); 2733 2734 mGlobalEffectEnableTime = systemTime(); 2735 2736 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2737 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2738 if (t->mType == ThreadBase::OFFLOAD) { 2739 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2740 } 2741 } 2742 2743} 2744 2745status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2746{ 2747 audio_session_t session = (audio_session_t)chain->sessionId(); 2748 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2749 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2750 if (index >= 0) { 2751 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2752 return ALREADY_EXISTS; 2753 } 2754 mOrphanEffectChains.add(session, chain); 2755 return NO_ERROR; 2756} 2757 2758sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2759{ 2760 sp<EffectChain> chain; 2761 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2762 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2763 if (index >= 0) { 2764 chain = mOrphanEffectChains.valueAt(index); 2765 mOrphanEffectChains.removeItemsAt(index); 2766 } 2767 return chain; 2768} 2769 2770bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2771{ 2772 Mutex::Autolock _l(mLock); 2773 audio_session_t session = (audio_session_t)effect->sessionId(); 2774 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2775 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2776 if (index >= 0) { 2777 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2778 if (chain->removeEffect_l(effect) == 0) { 2779 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2780 mOrphanEffectChains.removeItemsAt(index); 2781 } 2782 return true; 2783 } 2784 return false; 2785} 2786 2787 2788struct Entry { 2789#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2790 char mName[MAX_NAME]; 2791}; 2792 2793int comparEntry(const void *p1, const void *p2) 2794{ 2795 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2796} 2797 2798#ifdef TEE_SINK 2799void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2800{ 2801 NBAIO_Source *teeSource = source.get(); 2802 if (teeSource != NULL) { 2803 // .wav rotation 2804 // There is a benign race condition if 2 threads call this simultaneously. 2805 // They would both traverse the directory, but the result would simply be 2806 // failures at unlink() which are ignored. It's also unlikely since 2807 // normally dumpsys is only done by bugreport or from the command line. 2808 char teePath[32+256]; 2809 strcpy(teePath, "/data/misc/media"); 2810 size_t teePathLen = strlen(teePath); 2811 DIR *dir = opendir(teePath); 2812 teePath[teePathLen++] = '/'; 2813 if (dir != NULL) { 2814#define MAX_SORT 20 // number of entries to sort 2815#define MAX_KEEP 10 // number of entries to keep 2816 struct Entry entries[MAX_SORT]; 2817 size_t entryCount = 0; 2818 while (entryCount < MAX_SORT) { 2819 struct dirent de; 2820 struct dirent *result = NULL; 2821 int rc = readdir_r(dir, &de, &result); 2822 if (rc != 0) { 2823 ALOGW("readdir_r failed %d", rc); 2824 break; 2825 } 2826 if (result == NULL) { 2827 break; 2828 } 2829 if (result != &de) { 2830 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2831 break; 2832 } 2833 // ignore non .wav file entries 2834 size_t nameLen = strlen(de.d_name); 2835 if (nameLen <= 4 || nameLen >= MAX_NAME || 2836 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2837 continue; 2838 } 2839 strcpy(entries[entryCount++].mName, de.d_name); 2840 } 2841 (void) closedir(dir); 2842 if (entryCount > MAX_KEEP) { 2843 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2844 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2845 strcpy(&teePath[teePathLen], entries[i].mName); 2846 (void) unlink(teePath); 2847 } 2848 } 2849 } else { 2850 if (fd >= 0) { 2851 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2852 } 2853 } 2854 char teeTime[16]; 2855 struct timeval tv; 2856 gettimeofday(&tv, NULL); 2857 struct tm tm; 2858 localtime_r(&tv.tv_sec, &tm); 2859 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2860 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2861 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2862 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2863 if (teeFd >= 0) { 2864 // FIXME use libsndfile 2865 char wavHeader[44]; 2866 memcpy(wavHeader, 2867 "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", 2868 sizeof(wavHeader)); 2869 NBAIO_Format format = teeSource->format(); 2870 unsigned channelCount = Format_channelCount(format); 2871 uint32_t sampleRate = Format_sampleRate(format); 2872 size_t frameSize = Format_frameSize(format); 2873 wavHeader[22] = channelCount; // number of channels 2874 wavHeader[24] = sampleRate; // sample rate 2875 wavHeader[25] = sampleRate >> 8; 2876 wavHeader[32] = frameSize; // block alignment 2877 wavHeader[33] = frameSize >> 8; 2878 write(teeFd, wavHeader, sizeof(wavHeader)); 2879 size_t total = 0; 2880 bool firstRead = true; 2881#define TEE_SINK_READ 1024 // frames per I/O operation 2882 void *buffer = malloc(TEE_SINK_READ * frameSize); 2883 for (;;) { 2884 size_t count = TEE_SINK_READ; 2885 ssize_t actual = teeSource->read(buffer, count, 2886 AudioBufferProvider::kInvalidPTS); 2887 bool wasFirstRead = firstRead; 2888 firstRead = false; 2889 if (actual <= 0) { 2890 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2891 continue; 2892 } 2893 break; 2894 } 2895 ALOG_ASSERT(actual <= (ssize_t)count); 2896 write(teeFd, buffer, actual * frameSize); 2897 total += actual; 2898 } 2899 free(buffer); 2900 lseek(teeFd, (off_t) 4, SEEK_SET); 2901 uint32_t temp = 44 + total * frameSize - 8; 2902 // FIXME not big-endian safe 2903 write(teeFd, &temp, sizeof(temp)); 2904 lseek(teeFd, (off_t) 40, SEEK_SET); 2905 temp = total * frameSize; 2906 // FIXME not big-endian safe 2907 write(teeFd, &temp, sizeof(temp)); 2908 close(teeFd); 2909 if (fd >= 0) { 2910 dprintf(fd, "tee copied to %s\n", teePath); 2911 } 2912 } else { 2913 if (fd >= 0) { 2914 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2915 } 2916 } 2917 } 2918} 2919#endif 2920 2921// ---------------------------------------------------------------------------- 2922 2923status_t AudioFlinger::onTransact( 2924 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2925{ 2926 return BnAudioFlinger::onTransact(code, data, reply, flags); 2927} 2928 2929}; // namespace android 2930