AudioFlinger.cpp revision 6146c08f0c3dd8b9e5788063aa433f304a810602
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) { 110 case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8"; 111 case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16"; 112 case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32"; 113 case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24"; 114 case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24"; 115 case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat"; 116 case AUDIO_FORMAT_MP3: return "mp3"; 117 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 118 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 119 case AUDIO_FORMAT_AAC: return "aac"; 120 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 121 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 122 case AUDIO_FORMAT_VORBIS: return "vorbis"; 123 default: 124 break; 125 } 126 return "unknown"; 127} 128 129static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 130{ 131 const hw_module_t *mod; 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 135 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 136 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 137 if (rc) { 138 goto out; 139 } 140 rc = audio_hw_device_open(mod, dev); 141 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 142 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 143 if (rc) { 144 goto out; 145 } 146 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 147 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 148 rc = BAD_VALUE; 149 goto out; 150 } 151 return 0; 152 153out: 154 *dev = NULL; 155 return rc; 156} 157 158// ---------------------------------------------------------------------------- 159 160AudioFlinger::AudioFlinger() 161 : BnAudioFlinger(), 162 mPrimaryHardwareDev(NULL), 163 mAudioHwDevs(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), 165 mMasterVolume(1.0f), 166 mMasterMute(false), 167 mNextUniqueId(1), 168 mMode(AUDIO_MODE_INVALID), 169 mBtNrecIsOff(false), 170 mIsLowRamDevice(true), 171 mIsDeviceTypeKnown(false), 172 mGlobalEffectEnableTime(0), 173 mPrimaryOutputSampleRate(0) 174{ 175 getpid_cached = getpid(); 176 char value[PROPERTY_VALUE_MAX]; 177 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 178 if (doLog) { 179 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 180 } 181 182#ifdef TEE_SINK 183 (void) property_get("ro.debuggable", value, "0"); 184 int debuggable = atoi(value); 185 int teeEnabled = 0; 186 if (debuggable) { 187 (void) property_get("af.tee", value, "0"); 188 teeEnabled = atoi(value); 189 } 190 // FIXME symbolic constants here 191 if (teeEnabled & 1) { 192 mTeeSinkInputEnabled = true; 193 } 194 if (teeEnabled & 2) { 195 mTeeSinkOutputEnabled = true; 196 } 197 if (teeEnabled & 4) { 198 mTeeSinkTrackEnabled = true; 199 } 200#endif 201} 202 203void AudioFlinger::onFirstRef() 204{ 205 int rc = 0; 206 207 Mutex::Autolock _l(mLock); 208 209 /* TODO: move all this work into an Init() function */ 210 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 211 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 212 uint32_t int_val; 213 if (1 == sscanf(val_str, "%u", &int_val)) { 214 mStandbyTimeInNsecs = milliseconds(int_val); 215 ALOGI("Using %u mSec as standby time.", int_val); 216 } else { 217 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 218 ALOGI("Using default %u mSec as standby time.", 219 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 220 } 221 } 222 223 mPatchPanel = new PatchPanel(this); 224 225 mMode = AUDIO_MODE_NORMAL; 226} 227 228AudioFlinger::~AudioFlinger() 229{ 230 while (!mRecordThreads.isEmpty()) { 231 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 232 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 233 } 234 while (!mPlaybackThreads.isEmpty()) { 235 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 236 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 237 } 238 239 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 240 // no mHardwareLock needed, as there are no other references to this 241 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 242 delete mAudioHwDevs.valueAt(i); 243 } 244 245 // Tell media.log service about any old writers that still need to be unregistered 246 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 247 if (binder != 0) { 248 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 249 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 250 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 251 mUnregisteredWriters.pop(); 252 mediaLogService->unregisterWriter(iMemory); 253 } 254 } 255 256} 257 258static const char * const audio_interfaces[] = { 259 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 260 AUDIO_HARDWARE_MODULE_ID_A2DP, 261 AUDIO_HARDWARE_MODULE_ID_USB, 262}; 263#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 264 265AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 266 audio_module_handle_t module, 267 audio_devices_t devices) 268{ 269 // if module is 0, the request comes from an old policy manager and we should load 270 // well known modules 271 if (module == 0) { 272 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 273 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 274 loadHwModule_l(audio_interfaces[i]); 275 } 276 // then try to find a module supporting the requested device. 277 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 278 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 279 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 280 if ((dev->get_supported_devices != NULL) && 281 (dev->get_supported_devices(dev) & devices) == devices) 282 return audioHwDevice; 283 } 284 } else { 285 // check a match for the requested module handle 286 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 287 if (audioHwDevice != NULL) { 288 return audioHwDevice; 289 } 290 } 291 292 return NULL; 293} 294 295void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 296{ 297 const size_t SIZE = 256; 298 char buffer[SIZE]; 299 String8 result; 300 301 result.append("Clients:\n"); 302 for (size_t i = 0; i < mClients.size(); ++i) { 303 sp<Client> client = mClients.valueAt(i).promote(); 304 if (client != 0) { 305 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 306 result.append(buffer); 307 } 308 } 309 310 result.append("Notification Clients:\n"); 311 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 312 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 313 result.append(buffer); 314 } 315 316 result.append("Global session refs:\n"); 317 result.append(" session pid count\n"); 318 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 319 AudioSessionRef *r = mAudioSessionRefs[i]; 320 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 321 result.append(buffer); 322 } 323 write(fd, result.string(), result.size()); 324} 325 326 327void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 328{ 329 const size_t SIZE = 256; 330 char buffer[SIZE]; 331 String8 result; 332 hardware_call_state hardwareStatus = mHardwareStatus; 333 334 snprintf(buffer, SIZE, "Hardware status: %d\n" 335 "Standby Time mSec: %u\n", 336 hardwareStatus, 337 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 338 result.append(buffer); 339 write(fd, result.string(), result.size()); 340} 341 342void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 343{ 344 const size_t SIZE = 256; 345 char buffer[SIZE]; 346 String8 result; 347 snprintf(buffer, SIZE, "Permission Denial: " 348 "can't dump AudioFlinger from pid=%d, uid=%d\n", 349 IPCThreadState::self()->getCallingPid(), 350 IPCThreadState::self()->getCallingUid()); 351 result.append(buffer); 352 write(fd, result.string(), result.size()); 353} 354 355bool AudioFlinger::dumpTryLock(Mutex& mutex) 356{ 357 bool locked = false; 358 for (int i = 0; i < kDumpLockRetries; ++i) { 359 if (mutex.tryLock() == NO_ERROR) { 360 locked = true; 361 break; 362 } 363 usleep(kDumpLockSleepUs); 364 } 365 return locked; 366} 367 368status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 369{ 370 if (!dumpAllowed()) { 371 dumpPermissionDenial(fd, args); 372 } else { 373 // get state of hardware lock 374 bool hardwareLocked = dumpTryLock(mHardwareLock); 375 if (!hardwareLocked) { 376 String8 result(kHardwareLockedString); 377 write(fd, result.string(), result.size()); 378 } else { 379 mHardwareLock.unlock(); 380 } 381 382 bool locked = dumpTryLock(mLock); 383 384 // failed to lock - AudioFlinger is probably deadlocked 385 if (!locked) { 386 String8 result(kDeadlockedString); 387 write(fd, result.string(), result.size()); 388 } 389 390 bool clientLocked = dumpTryLock(mClientLock); 391 if (!clientLocked) { 392 String8 result(kClientLockedString); 393 write(fd, result.string(), result.size()); 394 } 395 dumpClients(fd, args); 396 if (clientLocked) { 397 mClientLock.unlock(); 398 } 399 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 415 dev->dump(dev, fd); 416 } 417 418#ifdef TEE_SINK 419 // dump the serially shared record tee sink 420 if (mRecordTeeSource != 0) { 421 dumpTee(fd, mRecordTeeSource); 422 } 423#endif 424 425 if (locked) { 426 mLock.unlock(); 427 } 428 429 // append a copy of media.log here by forwarding fd to it, but don't attempt 430 // to lookup the service if it's not running, as it will block for a second 431 if (mLogMemoryDealer != 0) { 432 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 433 if (binder != 0) { 434 dprintf(fd, "\nmedia.log:\n"); 435 Vector<String16> args; 436 binder->dump(fd, args); 437 } 438 } 439 } 440 return NO_ERROR; 441} 442 443sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 444{ 445 Mutex::Autolock _cl(mClientLock); 446 // If pid is already in the mClients wp<> map, then use that entry 447 // (for which promote() is always != 0), otherwise create a new entry and Client. 448 sp<Client> client = mClients.valueFor(pid).promote(); 449 if (client == 0) { 450 client = new Client(this, pid); 451 mClients.add(pid, client); 452 } 453 454 return client; 455} 456 457sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 458{ 459 // If there is no memory allocated for logs, return a dummy writer that does nothing 460 if (mLogMemoryDealer == 0) { 461 return new NBLog::Writer(); 462 } 463 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 464 // Similarly if we can't contact the media.log service, also return a dummy writer 465 if (binder == 0) { 466 return new NBLog::Writer(); 467 } 468 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 469 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 470 // If allocation fails, consult the vector of previously unregistered writers 471 // and garbage-collect one or more them until an allocation succeeds 472 if (shared == 0) { 473 Mutex::Autolock _l(mUnregisteredWritersLock); 474 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 475 { 476 // Pick the oldest stale writer to garbage-collect 477 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 478 mUnregisteredWriters.removeAt(0); 479 mediaLogService->unregisterWriter(iMemory); 480 // Now the media.log remote reference to IMemory is gone. When our last local 481 // reference to IMemory also drops to zero at end of this block, 482 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 483 } 484 // Re-attempt the allocation 485 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 486 if (shared != 0) { 487 goto success; 488 } 489 } 490 // Even after garbage-collecting all old writers, there is still not enough memory, 491 // so return a dummy writer 492 return new NBLog::Writer(); 493 } 494success: 495 mediaLogService->registerWriter(shared, size, name); 496 return new NBLog::Writer(size, shared); 497} 498 499void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 500{ 501 if (writer == 0) { 502 return; 503 } 504 sp<IMemory> iMemory(writer->getIMemory()); 505 if (iMemory == 0) { 506 return; 507 } 508 // Rather than removing the writer immediately, append it to a queue of old writers to 509 // be garbage-collected later. This allows us to continue to view old logs for a while. 510 Mutex::Autolock _l(mUnregisteredWritersLock); 511 mUnregisteredWriters.push(writer); 512} 513 514// IAudioFlinger interface 515 516 517sp<IAudioTrack> AudioFlinger::createTrack( 518 audio_stream_type_t streamType, 519 uint32_t sampleRate, 520 audio_format_t format, 521 audio_channel_mask_t channelMask, 522 size_t *frameCount, 523 IAudioFlinger::track_flags_t *flags, 524 const sp<IMemory>& sharedBuffer, 525 audio_io_handle_t output, 526 pid_t tid, 527 int *sessionId, 528 int clientUid, 529 status_t *status) 530{ 531 sp<PlaybackThread::Track> track; 532 sp<TrackHandle> trackHandle; 533 sp<Client> client; 534 status_t lStatus; 535 int lSessionId; 536 537 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 538 // but if someone uses binder directly they could bypass that and cause us to crash 539 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 540 ALOGE("createTrack() invalid stream type %d", streamType); 541 lStatus = BAD_VALUE; 542 goto Exit; 543 } 544 545 // further sample rate checks are performed by createTrack_l() depending on the thread type 546 if (sampleRate == 0) { 547 ALOGE("createTrack() invalid sample rate %u", sampleRate); 548 lStatus = BAD_VALUE; 549 goto Exit; 550 } 551 552 // further channel mask checks are performed by createTrack_l() depending on the thread type 553 if (!audio_is_output_channel(channelMask)) { 554 ALOGE("createTrack() invalid channel mask %#x", channelMask); 555 lStatus = BAD_VALUE; 556 goto Exit; 557 } 558 559 // further format checks are performed by createTrack_l() depending on the thread type 560 if (!audio_is_valid_format(format)) { 561 ALOGE("createTrack() invalid format %#x", format); 562 lStatus = BAD_VALUE; 563 goto Exit; 564 } 565 566 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 567 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 568 lStatus = BAD_VALUE; 569 goto Exit; 570 } 571 572 { 573 Mutex::Autolock _l(mLock); 574 PlaybackThread *thread = checkPlaybackThread_l(output); 575 if (thread == NULL) { 576 ALOGE("no playback thread found for output handle %d", output); 577 lStatus = BAD_VALUE; 578 goto Exit; 579 } 580 581 pid_t pid = IPCThreadState::self()->getCallingPid(); 582 client = registerPid(pid); 583 584 PlaybackThread *effectThread = NULL; 585 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 586 lSessionId = *sessionId; 587 // check if an effect chain with the same session ID is present on another 588 // output thread and move it here. 589 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 590 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 591 if (mPlaybackThreads.keyAt(i) != output) { 592 uint32_t sessions = t->hasAudioSession(lSessionId); 593 if (sessions & PlaybackThread::EFFECT_SESSION) { 594 effectThread = t.get(); 595 break; 596 } 597 } 598 } 599 } else { 600 // if no audio session id is provided, create one here 601 lSessionId = nextUniqueId(); 602 if (sessionId != NULL) { 603 *sessionId = lSessionId; 604 } 605 } 606 ALOGV("createTrack() lSessionId: %d", lSessionId); 607 608 track = thread->createTrack_l(client, streamType, sampleRate, format, 609 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 610 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 611 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 612 613 // move effect chain to this output thread if an effect on same session was waiting 614 // for a track to be created 615 if (lStatus == NO_ERROR && effectThread != NULL) { 616 // no risk of deadlock because AudioFlinger::mLock is held 617 Mutex::Autolock _dl(thread->mLock); 618 Mutex::Autolock _sl(effectThread->mLock); 619 moveEffectChain_l(lSessionId, effectThread, thread, true); 620 } 621 622 // Look for sync events awaiting for a session to be used. 623 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 624 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 625 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 626 if (lStatus == NO_ERROR) { 627 (void) track->setSyncEvent(mPendingSyncEvents[i]); 628 } else { 629 mPendingSyncEvents[i]->cancel(); 630 } 631 mPendingSyncEvents.removeAt(i); 632 i--; 633 } 634 } 635 } 636 637 } 638 639 if (lStatus != NO_ERROR) { 640 // remove local strong reference to Client before deleting the Track so that the 641 // Client destructor is called by the TrackBase destructor with mClientLock held 642 // Don't hold mClientLock when releasing the reference on the track as the 643 // destructor will acquire it. 644 { 645 Mutex::Autolock _cl(mClientLock); 646 client.clear(); 647 } 648 track.clear(); 649 goto Exit; 650 } 651 652 // return handle to client 653 trackHandle = new TrackHandle(track); 654 655Exit: 656 *status = lStatus; 657 return trackHandle; 658} 659 660uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 661{ 662 Mutex::Autolock _l(mLock); 663 PlaybackThread *thread = checkPlaybackThread_l(output); 664 if (thread == NULL) { 665 ALOGW("sampleRate() unknown thread %d", output); 666 return 0; 667 } 668 return thread->sampleRate(); 669} 670 671int AudioFlinger::channelCount(audio_io_handle_t output) const 672{ 673 Mutex::Autolock _l(mLock); 674 PlaybackThread *thread = checkPlaybackThread_l(output); 675 if (thread == NULL) { 676 ALOGW("channelCount() unknown thread %d", output); 677 return 0; 678 } 679 return thread->channelCount(); 680} 681 682audio_format_t AudioFlinger::format(audio_io_handle_t output) const 683{ 684 Mutex::Autolock _l(mLock); 685 PlaybackThread *thread = checkPlaybackThread_l(output); 686 if (thread == NULL) { 687 ALOGW("format() unknown thread %d", output); 688 return AUDIO_FORMAT_INVALID; 689 } 690 return thread->format(); 691} 692 693size_t AudioFlinger::frameCount(audio_io_handle_t output) const 694{ 695 Mutex::Autolock _l(mLock); 696 PlaybackThread *thread = checkPlaybackThread_l(output); 697 if (thread == NULL) { 698 ALOGW("frameCount() unknown thread %d", output); 699 return 0; 700 } 701 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 702 // should examine all callers and fix them to handle smaller counts 703 return thread->frameCount(); 704} 705 706uint32_t AudioFlinger::latency(audio_io_handle_t output) const 707{ 708 Mutex::Autolock _l(mLock); 709 PlaybackThread *thread = checkPlaybackThread_l(output); 710 if (thread == NULL) { 711 ALOGW("latency(): no playback thread found for output handle %d", output); 712 return 0; 713 } 714 return thread->latency(); 715} 716 717status_t AudioFlinger::setMasterVolume(float value) 718{ 719 status_t ret = initCheck(); 720 if (ret != NO_ERROR) { 721 return ret; 722 } 723 724 // check calling permissions 725 if (!settingsAllowed()) { 726 return PERMISSION_DENIED; 727 } 728 729 Mutex::Autolock _l(mLock); 730 mMasterVolume = value; 731 732 // Set master volume in the HALs which support it. 733 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 734 AutoMutex lock(mHardwareLock); 735 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 736 737 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 738 if (dev->canSetMasterVolume()) { 739 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 740 } 741 mHardwareStatus = AUDIO_HW_IDLE; 742 } 743 744 // Now set the master volume in each playback thread. Playback threads 745 // assigned to HALs which do not have master volume support will apply 746 // master volume during the mix operation. Threads with HALs which do 747 // support master volume will simply ignore the setting. 748 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 749 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 750 751 return NO_ERROR; 752} 753 754status_t AudioFlinger::setMode(audio_mode_t mode) 755{ 756 status_t ret = initCheck(); 757 if (ret != NO_ERROR) { 758 return ret; 759 } 760 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 766 ALOGW("Illegal value: setMode(%d)", mode); 767 return BAD_VALUE; 768 } 769 770 { // scope for the lock 771 AutoMutex lock(mHardwareLock); 772 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 773 mHardwareStatus = AUDIO_HW_SET_MODE; 774 ret = dev->set_mode(dev, mode); 775 mHardwareStatus = AUDIO_HW_IDLE; 776 } 777 778 if (NO_ERROR == ret) { 779 Mutex::Autolock _l(mLock); 780 mMode = mode; 781 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 782 mPlaybackThreads.valueAt(i)->setMode(mode); 783 } 784 785 return ret; 786} 787 788status_t AudioFlinger::setMicMute(bool state) 789{ 790 status_t ret = initCheck(); 791 if (ret != NO_ERROR) { 792 return ret; 793 } 794 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 AutoMutex lock(mHardwareLock); 801 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 802 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 803 ret = dev->set_mic_mute(dev, state); 804 mHardwareStatus = AUDIO_HW_IDLE; 805 return ret; 806} 807 808bool AudioFlinger::getMicMute() const 809{ 810 status_t ret = initCheck(); 811 if (ret != NO_ERROR) { 812 return false; 813 } 814 815 bool state = AUDIO_MODE_INVALID; 816 AutoMutex lock(mHardwareLock); 817 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 818 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 819 dev->get_mic_mute(dev, &state); 820 mHardwareStatus = AUDIO_HW_IDLE; 821 return state; 822} 823 824status_t AudioFlinger::setMasterMute(bool muted) 825{ 826 status_t ret = initCheck(); 827 if (ret != NO_ERROR) { 828 return ret; 829 } 830 831 // check calling permissions 832 if (!settingsAllowed()) { 833 return PERMISSION_DENIED; 834 } 835 836 Mutex::Autolock _l(mLock); 837 mMasterMute = muted; 838 839 // Set master mute in the HALs which support it. 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 AutoMutex lock(mHardwareLock); 842 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 843 844 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 845 if (dev->canSetMasterMute()) { 846 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 847 } 848 mHardwareStatus = AUDIO_HW_IDLE; 849 } 850 851 // Now set the master mute in each playback thread. Playback threads 852 // assigned to HALs which do not have master mute support will apply master 853 // mute during the mix operation. Threads with HALs which do support master 854 // mute will simply ignore the setting. 855 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 856 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 857 858 return NO_ERROR; 859} 860 861float AudioFlinger::masterVolume() const 862{ 863 Mutex::Autolock _l(mLock); 864 return masterVolume_l(); 865} 866 867bool AudioFlinger::masterMute() const 868{ 869 Mutex::Autolock _l(mLock); 870 return masterMute_l(); 871} 872 873float AudioFlinger::masterVolume_l() const 874{ 875 return mMasterVolume; 876} 877 878bool AudioFlinger::masterMute_l() const 879{ 880 return mMasterMute; 881} 882 883status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 884 audio_io_handle_t output) 885{ 886 // check calling permissions 887 if (!settingsAllowed()) { 888 return PERMISSION_DENIED; 889 } 890 891 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 892 ALOGE("setStreamVolume() invalid stream %d", stream); 893 return BAD_VALUE; 894 } 895 896 AutoMutex lock(mLock); 897 PlaybackThread *thread = NULL; 898 if (output != AUDIO_IO_HANDLE_NONE) { 899 thread = checkPlaybackThread_l(output); 900 if (thread == NULL) { 901 return BAD_VALUE; 902 } 903 } 904 905 mStreamTypes[stream].volume = value; 906 907 if (thread == NULL) { 908 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 909 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 910 } 911 } else { 912 thread->setStreamVolume(stream, value); 913 } 914 915 return NO_ERROR; 916} 917 918status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 919{ 920 // check calling permissions 921 if (!settingsAllowed()) { 922 return PERMISSION_DENIED; 923 } 924 925 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 926 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 927 ALOGE("setStreamMute() invalid stream %d", stream); 928 return BAD_VALUE; 929 } 930 931 AutoMutex lock(mLock); 932 mStreamTypes[stream].mute = muted; 933 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 934 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 935 936 return NO_ERROR; 937} 938 939float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 940{ 941 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 942 return 0.0f; 943 } 944 945 AutoMutex lock(mLock); 946 float volume; 947 if (output != AUDIO_IO_HANDLE_NONE) { 948 PlaybackThread *thread = checkPlaybackThread_l(output); 949 if (thread == NULL) { 950 return 0.0f; 951 } 952 volume = thread->streamVolume(stream); 953 } else { 954 volume = streamVolume_l(stream); 955 } 956 957 return volume; 958} 959 960bool AudioFlinger::streamMute(audio_stream_type_t stream) const 961{ 962 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 963 return true; 964 } 965 966 AutoMutex lock(mLock); 967 return streamMute_l(stream); 968} 969 970status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 971{ 972 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 973 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 974 975 // check calling permissions 976 if (!settingsAllowed()) { 977 return PERMISSION_DENIED; 978 } 979 980 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 981 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 982 Mutex::Autolock _l(mLock); 983 status_t final_result = NO_ERROR; 984 { 985 AutoMutex lock(mHardwareLock); 986 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 987 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 988 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 989 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 990 final_result = result ?: final_result; 991 } 992 mHardwareStatus = AUDIO_HW_IDLE; 993 } 994 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 995 AudioParameter param = AudioParameter(keyValuePairs); 996 String8 value; 997 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 998 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 999 if (mBtNrecIsOff != btNrecIsOff) { 1000 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1001 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1002 audio_devices_t device = thread->inDevice(); 1003 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1004 // collect all of the thread's session IDs 1005 KeyedVector<int, bool> ids = thread->sessionIds(); 1006 // suspend effects associated with those session IDs 1007 for (size_t j = 0; j < ids.size(); ++j) { 1008 int sessionId = ids.keyAt(j); 1009 thread->setEffectSuspended(FX_IID_AEC, 1010 suspend, 1011 sessionId); 1012 thread->setEffectSuspended(FX_IID_NS, 1013 suspend, 1014 sessionId); 1015 } 1016 } 1017 mBtNrecIsOff = btNrecIsOff; 1018 } 1019 } 1020 String8 screenState; 1021 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1022 bool isOff = screenState == "off"; 1023 if (isOff != (AudioFlinger::mScreenState & 1)) { 1024 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1025 } 1026 } 1027 return final_result; 1028 } 1029 1030 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1031 // and the thread is exited once the lock is released 1032 sp<ThreadBase> thread; 1033 { 1034 Mutex::Autolock _l(mLock); 1035 thread = checkPlaybackThread_l(ioHandle); 1036 if (thread == 0) { 1037 thread = checkRecordThread_l(ioHandle); 1038 } else if (thread == primaryPlaybackThread_l()) { 1039 // indicate output device change to all input threads for pre processing 1040 AudioParameter param = AudioParameter(keyValuePairs); 1041 int value; 1042 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1043 (value != 0)) { 1044 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1045 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1046 } 1047 } 1048 } 1049 } 1050 if (thread != 0) { 1051 return thread->setParameters(keyValuePairs); 1052 } 1053 return BAD_VALUE; 1054} 1055 1056String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1057{ 1058 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1059 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1060 1061 Mutex::Autolock _l(mLock); 1062 1063 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1064 String8 out_s8; 1065 1066 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1067 char *s; 1068 { 1069 AutoMutex lock(mHardwareLock); 1070 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1071 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1072 s = dev->get_parameters(dev, keys.string()); 1073 mHardwareStatus = AUDIO_HW_IDLE; 1074 } 1075 out_s8 += String8(s ? s : ""); 1076 free(s); 1077 } 1078 return out_s8; 1079 } 1080 1081 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1082 if (playbackThread != NULL) { 1083 return playbackThread->getParameters(keys); 1084 } 1085 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1086 if (recordThread != NULL) { 1087 return recordThread->getParameters(keys); 1088 } 1089 return String8(""); 1090} 1091 1092size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1093 audio_channel_mask_t channelMask) const 1094{ 1095 status_t ret = initCheck(); 1096 if (ret != NO_ERROR) { 1097 return 0; 1098 } 1099 1100 AutoMutex lock(mHardwareLock); 1101 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1102 struct audio_config config; 1103 memset(&config, 0, sizeof(config)); 1104 config.sample_rate = sampleRate; 1105 config.channel_mask = channelMask; 1106 config.format = format; 1107 1108 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1109 size_t size = dev->get_input_buffer_size(dev, &config); 1110 mHardwareStatus = AUDIO_HW_IDLE; 1111 return size; 1112} 1113 1114uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1115{ 1116 Mutex::Autolock _l(mLock); 1117 1118 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1119 if (recordThread != NULL) { 1120 return recordThread->getInputFramesLost(); 1121 } 1122 return 0; 1123} 1124 1125status_t AudioFlinger::setVoiceVolume(float value) 1126{ 1127 status_t ret = initCheck(); 1128 if (ret != NO_ERROR) { 1129 return ret; 1130 } 1131 1132 // check calling permissions 1133 if (!settingsAllowed()) { 1134 return PERMISSION_DENIED; 1135 } 1136 1137 AutoMutex lock(mHardwareLock); 1138 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1139 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1140 ret = dev->set_voice_volume(dev, value); 1141 mHardwareStatus = AUDIO_HW_IDLE; 1142 1143 return ret; 1144} 1145 1146status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1147 audio_io_handle_t output) const 1148{ 1149 status_t status; 1150 1151 Mutex::Autolock _l(mLock); 1152 1153 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1154 if (playbackThread != NULL) { 1155 return playbackThread->getRenderPosition(halFrames, dspFrames); 1156 } 1157 1158 return BAD_VALUE; 1159} 1160 1161void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1162{ 1163 Mutex::Autolock _l(mLock); 1164 bool clientAdded = false; 1165 { 1166 Mutex::Autolock _cl(mClientLock); 1167 1168 pid_t pid = IPCThreadState::self()->getCallingPid(); 1169 if (mNotificationClients.indexOfKey(pid) < 0) { 1170 sp<NotificationClient> notificationClient = new NotificationClient(this, 1171 client, 1172 pid); 1173 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1174 1175 mNotificationClients.add(pid, notificationClient); 1176 1177 sp<IBinder> binder = client->asBinder(); 1178 binder->linkToDeath(notificationClient); 1179 clientAdded = true; 1180 } 1181 } 1182 1183 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1184 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1185 if (clientAdded) { 1186 // the config change is always sent from playback or record threads to avoid deadlock 1187 // with AudioSystem::gLock 1188 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1189 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1190 } 1191 1192 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1193 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1194 } 1195 } 1196} 1197 1198void AudioFlinger::removeNotificationClient(pid_t pid) 1199{ 1200 Mutex::Autolock _l(mLock); 1201 { 1202 Mutex::Autolock _cl(mClientLock); 1203 mNotificationClients.removeItem(pid); 1204 } 1205 1206 ALOGV("%d died, releasing its sessions", pid); 1207 size_t num = mAudioSessionRefs.size(); 1208 bool removed = false; 1209 for (size_t i = 0; i< num; ) { 1210 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1211 ALOGV(" pid %d @ %d", ref->mPid, i); 1212 if (ref->mPid == pid) { 1213 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1214 mAudioSessionRefs.removeAt(i); 1215 delete ref; 1216 removed = true; 1217 num--; 1218 } else { 1219 i++; 1220 } 1221 } 1222 if (removed) { 1223 purgeStaleEffects_l(); 1224 } 1225} 1226 1227void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1228{ 1229 Mutex::Autolock _l(mClientLock); 1230 size_t size = mNotificationClients.size(); 1231 for (size_t i = 0; i < size; i++) { 1232 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1233 ioHandle, 1234 param2); 1235 } 1236} 1237 1238// removeClient_l() must be called with AudioFlinger::mClientLock held 1239void AudioFlinger::removeClient_l(pid_t pid) 1240{ 1241 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1242 IPCThreadState::self()->getCallingPid()); 1243 mClients.removeItem(pid); 1244} 1245 1246// getEffectThread_l() must be called with AudioFlinger::mLock held 1247sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1248{ 1249 sp<PlaybackThread> thread; 1250 1251 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1252 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1253 ALOG_ASSERT(thread == 0); 1254 thread = mPlaybackThreads.valueAt(i); 1255 } 1256 } 1257 1258 return thread; 1259} 1260 1261 1262 1263// ---------------------------------------------------------------------------- 1264 1265AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1266 : RefBase(), 1267 mAudioFlinger(audioFlinger), 1268 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1269 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1270 mPid(pid), 1271 mTimedTrackCount(0) 1272{ 1273 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1274} 1275 1276// Client destructor must be called with AudioFlinger::mClientLock held 1277AudioFlinger::Client::~Client() 1278{ 1279 mAudioFlinger->removeClient_l(mPid); 1280} 1281 1282sp<MemoryDealer> AudioFlinger::Client::heap() const 1283{ 1284 return mMemoryDealer; 1285} 1286 1287// Reserve one of the limited slots for a timed audio track associated 1288// with this client 1289bool AudioFlinger::Client::reserveTimedTrack() 1290{ 1291 const int kMaxTimedTracksPerClient = 4; 1292 1293 Mutex::Autolock _l(mTimedTrackLock); 1294 1295 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1296 ALOGW("can not create timed track - pid %d has exceeded the limit", 1297 mPid); 1298 return false; 1299 } 1300 1301 mTimedTrackCount++; 1302 return true; 1303} 1304 1305// Release a slot for a timed audio track 1306void AudioFlinger::Client::releaseTimedTrack() 1307{ 1308 Mutex::Autolock _l(mTimedTrackLock); 1309 mTimedTrackCount--; 1310} 1311 1312// ---------------------------------------------------------------------------- 1313 1314AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1315 const sp<IAudioFlingerClient>& client, 1316 pid_t pid) 1317 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1318{ 1319} 1320 1321AudioFlinger::NotificationClient::~NotificationClient() 1322{ 1323} 1324 1325void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1326{ 1327 sp<NotificationClient> keep(this); 1328 mAudioFlinger->removeNotificationClient(mPid); 1329} 1330 1331 1332// ---------------------------------------------------------------------------- 1333 1334static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1335 return audio_is_remote_submix_device(inDevice); 1336} 1337 1338sp<IAudioRecord> AudioFlinger::openRecord( 1339 audio_io_handle_t input, 1340 uint32_t sampleRate, 1341 audio_format_t format, 1342 audio_channel_mask_t channelMask, 1343 size_t *frameCount, 1344 IAudioFlinger::track_flags_t *flags, 1345 pid_t tid, 1346 int *sessionId, 1347 sp<IMemory>& cblk, 1348 sp<IMemory>& buffers, 1349 status_t *status) 1350{ 1351 sp<RecordThread::RecordTrack> recordTrack; 1352 sp<RecordHandle> recordHandle; 1353 sp<Client> client; 1354 status_t lStatus; 1355 int lSessionId; 1356 1357 cblk.clear(); 1358 buffers.clear(); 1359 1360 // check calling permissions 1361 if (!recordingAllowed()) { 1362 ALOGE("openRecord() permission denied: recording not allowed"); 1363 lStatus = PERMISSION_DENIED; 1364 goto Exit; 1365 } 1366 1367 // further sample rate checks are performed by createRecordTrack_l() 1368 if (sampleRate == 0) { 1369 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1370 lStatus = BAD_VALUE; 1371 goto Exit; 1372 } 1373 1374 // we don't yet support anything other than 16-bit PCM 1375 if (!(audio_is_valid_format(format) && 1376 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1377 ALOGE("openRecord() invalid format %#x", format); 1378 lStatus = BAD_VALUE; 1379 goto Exit; 1380 } 1381 1382 // further channel mask checks are performed by createRecordTrack_l() 1383 if (!audio_is_input_channel(channelMask)) { 1384 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1385 lStatus = BAD_VALUE; 1386 goto Exit; 1387 } 1388 1389 { 1390 Mutex::Autolock _l(mLock); 1391 RecordThread *thread = checkRecordThread_l(input); 1392 if (thread == NULL) { 1393 ALOGE("openRecord() checkRecordThread_l failed"); 1394 lStatus = BAD_VALUE; 1395 goto Exit; 1396 } 1397 1398 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1399 && !captureAudioOutputAllowed()) { 1400 ALOGE("openRecord() permission denied: capture not allowed"); 1401 lStatus = PERMISSION_DENIED; 1402 goto Exit; 1403 } 1404 1405 pid_t pid = IPCThreadState::self()->getCallingPid(); 1406 client = registerPid(pid); 1407 1408 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1409 lSessionId = *sessionId; 1410 } else { 1411 // if no audio session id is provided, create one here 1412 lSessionId = nextUniqueId(); 1413 if (sessionId != NULL) { 1414 *sessionId = lSessionId; 1415 } 1416 } 1417 ALOGV("openRecord() lSessionId: %d", lSessionId); 1418 1419 // TODO: the uid should be passed in as a parameter to openRecord 1420 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1421 frameCount, lSessionId, 1422 IPCThreadState::self()->getCallingUid(), 1423 flags, tid, &lStatus); 1424 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1425 } 1426 1427 if (lStatus != NO_ERROR) { 1428 // remove local strong reference to Client before deleting the RecordTrack so that the 1429 // Client destructor is called by the TrackBase destructor with mClientLock held 1430 // Don't hold mClientLock when releasing the reference on the track as the 1431 // destructor will acquire it. 1432 { 1433 Mutex::Autolock _cl(mClientLock); 1434 client.clear(); 1435 } 1436 recordTrack.clear(); 1437 goto Exit; 1438 } 1439 1440 cblk = recordTrack->getCblk(); 1441 buffers = recordTrack->getBuffers(); 1442 1443 // return handle to client 1444 recordHandle = new RecordHandle(recordTrack); 1445 1446Exit: 1447 *status = lStatus; 1448 return recordHandle; 1449} 1450 1451 1452 1453// ---------------------------------------------------------------------------- 1454 1455audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1456{ 1457 if (!settingsAllowed()) { 1458 return 0; 1459 } 1460 Mutex::Autolock _l(mLock); 1461 return loadHwModule_l(name); 1462} 1463 1464// loadHwModule_l() must be called with AudioFlinger::mLock held 1465audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1466{ 1467 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1468 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1469 ALOGW("loadHwModule() module %s already loaded", name); 1470 return mAudioHwDevs.keyAt(i); 1471 } 1472 } 1473 1474 audio_hw_device_t *dev; 1475 1476 int rc = load_audio_interface(name, &dev); 1477 if (rc) { 1478 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1479 return 0; 1480 } 1481 1482 mHardwareStatus = AUDIO_HW_INIT; 1483 rc = dev->init_check(dev); 1484 mHardwareStatus = AUDIO_HW_IDLE; 1485 if (rc) { 1486 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1487 return 0; 1488 } 1489 1490 // Check and cache this HAL's level of support for master mute and master 1491 // volume. If this is the first HAL opened, and it supports the get 1492 // methods, use the initial values provided by the HAL as the current 1493 // master mute and volume settings. 1494 1495 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1496 { // scope for auto-lock pattern 1497 AutoMutex lock(mHardwareLock); 1498 1499 if (0 == mAudioHwDevs.size()) { 1500 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1501 if (NULL != dev->get_master_volume) { 1502 float mv; 1503 if (OK == dev->get_master_volume(dev, &mv)) { 1504 mMasterVolume = mv; 1505 } 1506 } 1507 1508 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1509 if (NULL != dev->get_master_mute) { 1510 bool mm; 1511 if (OK == dev->get_master_mute(dev, &mm)) { 1512 mMasterMute = mm; 1513 } 1514 } 1515 } 1516 1517 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1518 if ((NULL != dev->set_master_volume) && 1519 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1520 flags = static_cast<AudioHwDevice::Flags>(flags | 1521 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1522 } 1523 1524 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1525 if ((NULL != dev->set_master_mute) && 1526 (OK == dev->set_master_mute(dev, mMasterMute))) { 1527 flags = static_cast<AudioHwDevice::Flags>(flags | 1528 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1529 } 1530 1531 mHardwareStatus = AUDIO_HW_IDLE; 1532 } 1533 1534 audio_module_handle_t handle = nextUniqueId(); 1535 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1536 1537 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1538 name, dev->common.module->name, dev->common.module->id, handle); 1539 1540 return handle; 1541 1542} 1543 1544// ---------------------------------------------------------------------------- 1545 1546uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1547{ 1548 Mutex::Autolock _l(mLock); 1549 PlaybackThread *thread = primaryPlaybackThread_l(); 1550 return thread != NULL ? thread->sampleRate() : 0; 1551} 1552 1553size_t AudioFlinger::getPrimaryOutputFrameCount() 1554{ 1555 Mutex::Autolock _l(mLock); 1556 PlaybackThread *thread = primaryPlaybackThread_l(); 1557 return thread != NULL ? thread->frameCountHAL() : 0; 1558} 1559 1560// ---------------------------------------------------------------------------- 1561 1562status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1563{ 1564 uid_t uid = IPCThreadState::self()->getCallingUid(); 1565 if (uid != AID_SYSTEM) { 1566 return PERMISSION_DENIED; 1567 } 1568 Mutex::Autolock _l(mLock); 1569 if (mIsDeviceTypeKnown) { 1570 return INVALID_OPERATION; 1571 } 1572 mIsLowRamDevice = isLowRamDevice; 1573 mIsDeviceTypeKnown = true; 1574 return NO_ERROR; 1575} 1576 1577// ---------------------------------------------------------------------------- 1578 1579audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1580 audio_devices_t *pDevices, 1581 uint32_t *pSamplingRate, 1582 audio_format_t *pFormat, 1583 audio_channel_mask_t *pChannelMask, 1584 uint32_t *pLatencyMs, 1585 audio_output_flags_t flags, 1586 const audio_offload_info_t *offloadInfo) 1587{ 1588 struct audio_config config; 1589 memset(&config, 0, sizeof(config)); 1590 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1591 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1592 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1593 if (offloadInfo != NULL) { 1594 config.offload_info = *offloadInfo; 1595 } 1596 1597 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1598 module, 1599 (pDevices != NULL) ? *pDevices : 0, 1600 config.sample_rate, 1601 config.format, 1602 config.channel_mask, 1603 flags); 1604 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1605 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version); 1606 1607 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1608 return AUDIO_IO_HANDLE_NONE; 1609 } 1610 1611 Mutex::Autolock _l(mLock); 1612 1613 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices); 1614 if (outHwDev == NULL) { 1615 return AUDIO_IO_HANDLE_NONE; 1616 } 1617 1618 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1619 audio_io_handle_t id = nextUniqueId(); 1620 1621 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1622 1623 audio_stream_out_t *outStream = NULL; 1624 1625 // FOR TESTING ONLY: 1626 // Enable increased sink precision for mixing mode if kEnableExtendedPrecision is true. 1627 if (kEnableExtendedPrecision && // Check only for Normal Mixing mode 1628 !(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1629 // Update format 1630 //config.format = AUDIO_FORMAT_PCM_FLOAT; 1631 //config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1632 //config.format = AUDIO_FORMAT_PCM_32_BIT; 1633 //config.format = AUDIO_FORMAT_PCM_8_24_BIT; 1634 // ALOGV("openOutput() upgrading format to %#08x", config.format); 1635 } 1636 1637 status_t status = hwDevHal->open_output_stream(hwDevHal, 1638 id, 1639 *pDevices, 1640 (audio_output_flags_t)flags, 1641 &config, 1642 &outStream); 1643 1644 mHardwareStatus = AUDIO_HW_IDLE; 1645 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1646 "Channels %x, status %d", 1647 outStream, 1648 config.sample_rate, 1649 config.format, 1650 config.channel_mask, 1651 status); 1652 1653 if (status == NO_ERROR && outStream != NULL) { 1654 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1655 1656 PlaybackThread *thread; 1657 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1658 thread = new OffloadThread(this, output, id, *pDevices); 1659 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1660 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1661 || !isValidPcmSinkFormat(config.format) 1662 || (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1663 thread = new DirectOutputThread(this, output, id, *pDevices); 1664 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1665 } else { 1666 thread = new MixerThread(this, output, id, *pDevices); 1667 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1668 } 1669 mPlaybackThreads.add(id, thread); 1670 1671 if (pSamplingRate != NULL) { 1672 *pSamplingRate = config.sample_rate; 1673 } 1674 if (pFormat != NULL) { 1675 *pFormat = config.format; 1676 } 1677 if (pChannelMask != NULL) { 1678 *pChannelMask = config.channel_mask; 1679 } 1680 if (pLatencyMs != NULL) { 1681 *pLatencyMs = thread->latency(); 1682 } 1683 1684 // notify client processes of the new output creation 1685 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1686 1687 // the first primary output opened designates the primary hw device 1688 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1689 ALOGI("Using module %d has the primary audio interface", module); 1690 mPrimaryHardwareDev = outHwDev; 1691 1692 AutoMutex lock(mHardwareLock); 1693 mHardwareStatus = AUDIO_HW_SET_MODE; 1694 hwDevHal->set_mode(hwDevHal, mMode); 1695 mHardwareStatus = AUDIO_HW_IDLE; 1696 1697 mPrimaryOutputSampleRate = config.sample_rate; 1698 } 1699 return id; 1700 } 1701 1702 return AUDIO_IO_HANDLE_NONE; 1703} 1704 1705audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1706 audio_io_handle_t output2) 1707{ 1708 Mutex::Autolock _l(mLock); 1709 MixerThread *thread1 = checkMixerThread_l(output1); 1710 MixerThread *thread2 = checkMixerThread_l(output2); 1711 1712 if (thread1 == NULL || thread2 == NULL) { 1713 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1714 output2); 1715 return AUDIO_IO_HANDLE_NONE; 1716 } 1717 1718 audio_io_handle_t id = nextUniqueId(); 1719 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1720 thread->addOutputTrack(thread2); 1721 mPlaybackThreads.add(id, thread); 1722 // notify client processes of the new output creation 1723 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1724 return id; 1725} 1726 1727status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1728{ 1729 return closeOutput_nonvirtual(output); 1730} 1731 1732status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1733{ 1734 // keep strong reference on the playback thread so that 1735 // it is not destroyed while exit() is executed 1736 sp<PlaybackThread> thread; 1737 { 1738 Mutex::Autolock _l(mLock); 1739 thread = checkPlaybackThread_l(output); 1740 if (thread == NULL) { 1741 return BAD_VALUE; 1742 } 1743 1744 ALOGV("closeOutput() %d", output); 1745 1746 if (thread->type() == ThreadBase::MIXER) { 1747 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1748 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1749 DuplicatingThread *dupThread = 1750 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1751 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1752 1753 } 1754 } 1755 } 1756 1757 1758 mPlaybackThreads.removeItem(output); 1759 // save all effects to the default thread 1760 if (mPlaybackThreads.size()) { 1761 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1762 if (dstThread != NULL) { 1763 // audioflinger lock is held here so the acquisition order of thread locks does not 1764 // matter 1765 Mutex::Autolock _dl(dstThread->mLock); 1766 Mutex::Autolock _sl(thread->mLock); 1767 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1768 for (size_t i = 0; i < effectChains.size(); i ++) { 1769 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1770 } 1771 } 1772 } 1773 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1774 } 1775 thread->exit(); 1776 // The thread entity (active unit of execution) is no longer running here, 1777 // but the ThreadBase container still exists. 1778 1779 if (thread->type() != ThreadBase::DUPLICATING) { 1780 AudioStreamOut *out = thread->clearOutput(); 1781 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1782 // from now on thread->mOutput is NULL 1783 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1784 delete out; 1785 } 1786 return NO_ERROR; 1787} 1788 1789status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1790{ 1791 Mutex::Autolock _l(mLock); 1792 PlaybackThread *thread = checkPlaybackThread_l(output); 1793 1794 if (thread == NULL) { 1795 return BAD_VALUE; 1796 } 1797 1798 ALOGV("suspendOutput() %d", output); 1799 thread->suspend(); 1800 1801 return NO_ERROR; 1802} 1803 1804status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1805{ 1806 Mutex::Autolock _l(mLock); 1807 PlaybackThread *thread = checkPlaybackThread_l(output); 1808 1809 if (thread == NULL) { 1810 return BAD_VALUE; 1811 } 1812 1813 ALOGV("restoreOutput() %d", output); 1814 1815 thread->restore(); 1816 1817 return NO_ERROR; 1818} 1819 1820audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1821 audio_devices_t *pDevices, 1822 uint32_t *pSamplingRate, 1823 audio_format_t *pFormat, 1824 audio_channel_mask_t *pChannelMask) 1825{ 1826 struct audio_config config; 1827 memset(&config, 0, sizeof(config)); 1828 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1829 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1830 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1831 1832 uint32_t reqSamplingRate = config.sample_rate; 1833 audio_format_t reqFormat = config.format; 1834 audio_channel_mask_t reqChannelMask = config.channel_mask; 1835 1836 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1837 return 0; 1838 } 1839 1840 Mutex::Autolock _l(mLock); 1841 1842 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices); 1843 if (inHwDev == NULL) { 1844 return 0; 1845 } 1846 1847 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1848 audio_io_handle_t id = nextUniqueId(); 1849 1850 audio_stream_in_t *inStream = NULL; 1851 status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1852 &inStream); 1853 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, " 1854 "status %d", 1855 inStream, 1856 config.sample_rate, 1857 config.format, 1858 config.channel_mask, 1859 status); 1860 1861 // If the input could not be opened with the requested parameters and we can handle the 1862 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1863 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1864 if (status == BAD_VALUE && 1865 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1866 (config.sample_rate <= 2 * reqSamplingRate) && 1867 (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) && 1868 (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) { 1869 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1870 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1871 inStream = NULL; 1872 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1873 // FIXME log this new status; HAL should not propose any further changes 1874 } 1875 1876 if (status == NO_ERROR && inStream != NULL) { 1877 1878#ifdef TEE_SINK 1879 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1880 // or (re-)create if current Pipe is idle and does not match the new format 1881 sp<NBAIO_Sink> teeSink; 1882 enum { 1883 TEE_SINK_NO, // don't copy input 1884 TEE_SINK_NEW, // copy input using a new pipe 1885 TEE_SINK_OLD, // copy input using an existing pipe 1886 } kind; 1887 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1888 audio_channel_count_from_in_mask( 1889 inStream->common.get_channels(&inStream->common))); 1890 if (!mTeeSinkInputEnabled) { 1891 kind = TEE_SINK_NO; 1892 } else if (!Format_isValid(format)) { 1893 kind = TEE_SINK_NO; 1894 } else if (mRecordTeeSink == 0) { 1895 kind = TEE_SINK_NEW; 1896 } else if (mRecordTeeSink->getStrongCount() != 1) { 1897 kind = TEE_SINK_NO; 1898 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1899 kind = TEE_SINK_OLD; 1900 } else { 1901 kind = TEE_SINK_NEW; 1902 } 1903 switch (kind) { 1904 case TEE_SINK_NEW: { 1905 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1906 size_t numCounterOffers = 0; 1907 const NBAIO_Format offers[1] = {format}; 1908 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1909 ALOG_ASSERT(index == 0); 1910 PipeReader *pipeReader = new PipeReader(*pipe); 1911 numCounterOffers = 0; 1912 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1913 ALOG_ASSERT(index == 0); 1914 mRecordTeeSink = pipe; 1915 mRecordTeeSource = pipeReader; 1916 teeSink = pipe; 1917 } 1918 break; 1919 case TEE_SINK_OLD: 1920 teeSink = mRecordTeeSink; 1921 break; 1922 case TEE_SINK_NO: 1923 default: 1924 break; 1925 } 1926#endif 1927 1928 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1929 1930 // Start record thread 1931 // RecordThread requires both input and output device indication to forward to audio 1932 // pre processing modules 1933 RecordThread *thread = new RecordThread(this, 1934 input, 1935 id, 1936 primaryOutputDevice_l(), 1937 *pDevices 1938#ifdef TEE_SINK 1939 , teeSink 1940#endif 1941 ); 1942 mRecordThreads.add(id, thread); 1943 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1944 if (pSamplingRate != NULL) { 1945 *pSamplingRate = reqSamplingRate; 1946 } 1947 if (pFormat != NULL) { 1948 *pFormat = config.format; 1949 } 1950 if (pChannelMask != NULL) { 1951 *pChannelMask = reqChannelMask; 1952 } 1953 1954 // notify client processes of the new input creation 1955 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1956 return id; 1957 } 1958 1959 return 0; 1960} 1961 1962status_t AudioFlinger::closeInput(audio_io_handle_t input) 1963{ 1964 return closeInput_nonvirtual(input); 1965} 1966 1967status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1968{ 1969 // keep strong reference on the record thread so that 1970 // it is not destroyed while exit() is executed 1971 sp<RecordThread> thread; 1972 { 1973 Mutex::Autolock _l(mLock); 1974 thread = checkRecordThread_l(input); 1975 if (thread == 0) { 1976 return BAD_VALUE; 1977 } 1978 1979 ALOGV("closeInput() %d", input); 1980 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 1981 mRecordThreads.removeItem(input); 1982 } 1983 thread->exit(); 1984 // The thread entity (active unit of execution) is no longer running here, 1985 // but the ThreadBase container still exists. 1986 1987 AudioStreamIn *in = thread->clearInput(); 1988 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1989 // from now on thread->mInput is NULL 1990 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1991 delete in; 1992 1993 return NO_ERROR; 1994} 1995 1996status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 1997{ 1998 Mutex::Autolock _l(mLock); 1999 ALOGV("invalidateStream() stream %d", stream); 2000 2001 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2002 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2003 thread->invalidateTracks(stream); 2004 } 2005 2006 return NO_ERROR; 2007} 2008 2009 2010int AudioFlinger::newAudioSessionId() 2011{ 2012 return nextUniqueId(); 2013} 2014 2015void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2016{ 2017 Mutex::Autolock _l(mLock); 2018 pid_t caller = IPCThreadState::self()->getCallingPid(); 2019 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2020 if (pid != -1 && (caller == getpid_cached)) { 2021 caller = pid; 2022 } 2023 2024 { 2025 Mutex::Autolock _cl(mClientLock); 2026 // Ignore requests received from processes not known as notification client. The request 2027 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2028 // called from a different pid leaving a stale session reference. Also we don't know how 2029 // to clear this reference if the client process dies. 2030 if (mNotificationClients.indexOfKey(caller) < 0) { 2031 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2032 return; 2033 } 2034 } 2035 2036 size_t num = mAudioSessionRefs.size(); 2037 for (size_t i = 0; i< num; i++) { 2038 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2039 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2040 ref->mCnt++; 2041 ALOGV(" incremented refcount to %d", ref->mCnt); 2042 return; 2043 } 2044 } 2045 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2046 ALOGV(" added new entry for %d", audioSession); 2047} 2048 2049void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2050{ 2051 Mutex::Autolock _l(mLock); 2052 pid_t caller = IPCThreadState::self()->getCallingPid(); 2053 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2054 if (pid != -1 && (caller == getpid_cached)) { 2055 caller = pid; 2056 } 2057 size_t num = mAudioSessionRefs.size(); 2058 for (size_t i = 0; i< num; i++) { 2059 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2060 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2061 ref->mCnt--; 2062 ALOGV(" decremented refcount to %d", ref->mCnt); 2063 if (ref->mCnt == 0) { 2064 mAudioSessionRefs.removeAt(i); 2065 delete ref; 2066 purgeStaleEffects_l(); 2067 } 2068 return; 2069 } 2070 } 2071 // If the caller is mediaserver it is likely that the session being released was acquired 2072 // on behalf of a process not in notification clients and we ignore the warning. 2073 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2074} 2075 2076void AudioFlinger::purgeStaleEffects_l() { 2077 2078 ALOGV("purging stale effects"); 2079 2080 Vector< sp<EffectChain> > chains; 2081 2082 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2083 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2084 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2085 sp<EffectChain> ec = t->mEffectChains[j]; 2086 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2087 chains.push(ec); 2088 } 2089 } 2090 } 2091 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2092 sp<RecordThread> t = mRecordThreads.valueAt(i); 2093 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2094 sp<EffectChain> ec = t->mEffectChains[j]; 2095 chains.push(ec); 2096 } 2097 } 2098 2099 for (size_t i = 0; i < chains.size(); i++) { 2100 sp<EffectChain> ec = chains[i]; 2101 int sessionid = ec->sessionId(); 2102 sp<ThreadBase> t = ec->mThread.promote(); 2103 if (t == 0) { 2104 continue; 2105 } 2106 size_t numsessionrefs = mAudioSessionRefs.size(); 2107 bool found = false; 2108 for (size_t k = 0; k < numsessionrefs; k++) { 2109 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2110 if (ref->mSessionid == sessionid) { 2111 ALOGV(" session %d still exists for %d with %d refs", 2112 sessionid, ref->mPid, ref->mCnt); 2113 found = true; 2114 break; 2115 } 2116 } 2117 if (!found) { 2118 Mutex::Autolock _l(t->mLock); 2119 // remove all effects from the chain 2120 while (ec->mEffects.size()) { 2121 sp<EffectModule> effect = ec->mEffects[0]; 2122 effect->unPin(); 2123 t->removeEffect_l(effect); 2124 if (effect->purgeHandles()) { 2125 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2126 } 2127 AudioSystem::unregisterEffect(effect->id()); 2128 } 2129 } 2130 } 2131 return; 2132} 2133 2134// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2135AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2136{ 2137 return mPlaybackThreads.valueFor(output).get(); 2138} 2139 2140// checkMixerThread_l() must be called with AudioFlinger::mLock held 2141AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2142{ 2143 PlaybackThread *thread = checkPlaybackThread_l(output); 2144 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2145} 2146 2147// checkRecordThread_l() must be called with AudioFlinger::mLock held 2148AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2149{ 2150 return mRecordThreads.valueFor(input).get(); 2151} 2152 2153uint32_t AudioFlinger::nextUniqueId() 2154{ 2155 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2156} 2157 2158AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2159{ 2160 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2161 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2162 AudioStreamOut *output = thread->getOutput(); 2163 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2164 return thread; 2165 } 2166 } 2167 return NULL; 2168} 2169 2170audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2171{ 2172 PlaybackThread *thread = primaryPlaybackThread_l(); 2173 2174 if (thread == NULL) { 2175 return 0; 2176 } 2177 2178 return thread->outDevice(); 2179} 2180 2181sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2182 int triggerSession, 2183 int listenerSession, 2184 sync_event_callback_t callBack, 2185 wp<RefBase> cookie) 2186{ 2187 Mutex::Autolock _l(mLock); 2188 2189 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2190 status_t playStatus = NAME_NOT_FOUND; 2191 status_t recStatus = NAME_NOT_FOUND; 2192 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2193 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2194 if (playStatus == NO_ERROR) { 2195 return event; 2196 } 2197 } 2198 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2199 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2200 if (recStatus == NO_ERROR) { 2201 return event; 2202 } 2203 } 2204 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2205 mPendingSyncEvents.add(event); 2206 } else { 2207 ALOGV("createSyncEvent() invalid event %d", event->type()); 2208 event.clear(); 2209 } 2210 return event; 2211} 2212 2213// ---------------------------------------------------------------------------- 2214// Effect management 2215// ---------------------------------------------------------------------------- 2216 2217 2218status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2219{ 2220 Mutex::Autolock _l(mLock); 2221 return EffectQueryNumberEffects(numEffects); 2222} 2223 2224status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2225{ 2226 Mutex::Autolock _l(mLock); 2227 return EffectQueryEffect(index, descriptor); 2228} 2229 2230status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2231 effect_descriptor_t *descriptor) const 2232{ 2233 Mutex::Autolock _l(mLock); 2234 return EffectGetDescriptor(pUuid, descriptor); 2235} 2236 2237 2238sp<IEffect> AudioFlinger::createEffect( 2239 effect_descriptor_t *pDesc, 2240 const sp<IEffectClient>& effectClient, 2241 int32_t priority, 2242 audio_io_handle_t io, 2243 int sessionId, 2244 status_t *status, 2245 int *id, 2246 int *enabled) 2247{ 2248 status_t lStatus = NO_ERROR; 2249 sp<EffectHandle> handle; 2250 effect_descriptor_t desc; 2251 2252 pid_t pid = IPCThreadState::self()->getCallingPid(); 2253 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2254 pid, effectClient.get(), priority, sessionId, io); 2255 2256 if (pDesc == NULL) { 2257 lStatus = BAD_VALUE; 2258 goto Exit; 2259 } 2260 2261 // check audio settings permission for global effects 2262 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2263 lStatus = PERMISSION_DENIED; 2264 goto Exit; 2265 } 2266 2267 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2268 // that can only be created by audio policy manager (running in same process) 2269 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2270 lStatus = PERMISSION_DENIED; 2271 goto Exit; 2272 } 2273 2274 { 2275 if (!EffectIsNullUuid(&pDesc->uuid)) { 2276 // if uuid is specified, request effect descriptor 2277 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2278 if (lStatus < 0) { 2279 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2280 goto Exit; 2281 } 2282 } else { 2283 // if uuid is not specified, look for an available implementation 2284 // of the required type in effect factory 2285 if (EffectIsNullUuid(&pDesc->type)) { 2286 ALOGW("createEffect() no effect type"); 2287 lStatus = BAD_VALUE; 2288 goto Exit; 2289 } 2290 uint32_t numEffects = 0; 2291 effect_descriptor_t d; 2292 d.flags = 0; // prevent compiler warning 2293 bool found = false; 2294 2295 lStatus = EffectQueryNumberEffects(&numEffects); 2296 if (lStatus < 0) { 2297 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2298 goto Exit; 2299 } 2300 for (uint32_t i = 0; i < numEffects; i++) { 2301 lStatus = EffectQueryEffect(i, &desc); 2302 if (lStatus < 0) { 2303 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2304 continue; 2305 } 2306 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2307 // If matching type found save effect descriptor. If the session is 2308 // 0 and the effect is not auxiliary, continue enumeration in case 2309 // an auxiliary version of this effect type is available 2310 found = true; 2311 d = desc; 2312 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2313 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2314 break; 2315 } 2316 } 2317 } 2318 if (!found) { 2319 lStatus = BAD_VALUE; 2320 ALOGW("createEffect() effect not found"); 2321 goto Exit; 2322 } 2323 // For same effect type, chose auxiliary version over insert version if 2324 // connect to output mix (Compliance to OpenSL ES) 2325 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2326 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2327 desc = d; 2328 } 2329 } 2330 2331 // Do not allow auxiliary effects on a session different from 0 (output mix) 2332 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2333 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2334 lStatus = INVALID_OPERATION; 2335 goto Exit; 2336 } 2337 2338 // check recording permission for visualizer 2339 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2340 !recordingAllowed()) { 2341 lStatus = PERMISSION_DENIED; 2342 goto Exit; 2343 } 2344 2345 // return effect descriptor 2346 *pDesc = desc; 2347 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2348 // if the output returned by getOutputForEffect() is removed before we lock the 2349 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2350 // and we will exit safely 2351 io = AudioSystem::getOutputForEffect(&desc); 2352 ALOGV("createEffect got output %d", io); 2353 } 2354 2355 Mutex::Autolock _l(mLock); 2356 2357 // If output is not specified try to find a matching audio session ID in one of the 2358 // output threads. 2359 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2360 // because of code checking output when entering the function. 2361 // Note: io is never 0 when creating an effect on an input 2362 if (io == AUDIO_IO_HANDLE_NONE) { 2363 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2364 // output must be specified by AudioPolicyManager when using session 2365 // AUDIO_SESSION_OUTPUT_STAGE 2366 lStatus = BAD_VALUE; 2367 goto Exit; 2368 } 2369 // look for the thread where the specified audio session is present 2370 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2371 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2372 io = mPlaybackThreads.keyAt(i); 2373 break; 2374 } 2375 } 2376 if (io == 0) { 2377 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2378 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2379 io = mRecordThreads.keyAt(i); 2380 break; 2381 } 2382 } 2383 } 2384 // If no output thread contains the requested session ID, default to 2385 // first output. The effect chain will be moved to the correct output 2386 // thread when a track with the same session ID is created 2387 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2388 io = mPlaybackThreads.keyAt(0); 2389 } 2390 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2391 } 2392 ThreadBase *thread = checkRecordThread_l(io); 2393 if (thread == NULL) { 2394 thread = checkPlaybackThread_l(io); 2395 if (thread == NULL) { 2396 ALOGE("createEffect() unknown output thread"); 2397 lStatus = BAD_VALUE; 2398 goto Exit; 2399 } 2400 } 2401 2402 sp<Client> client = registerPid(pid); 2403 2404 // create effect on selected output thread 2405 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2406 &desc, enabled, &lStatus); 2407 if (handle != 0 && id != NULL) { 2408 *id = handle->id(); 2409 } 2410 if (handle == 0) { 2411 // remove local strong reference to Client with mClientLock held 2412 Mutex::Autolock _cl(mClientLock); 2413 client.clear(); 2414 } 2415 } 2416 2417Exit: 2418 *status = lStatus; 2419 return handle; 2420} 2421 2422status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2423 audio_io_handle_t dstOutput) 2424{ 2425 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2426 sessionId, srcOutput, dstOutput); 2427 Mutex::Autolock _l(mLock); 2428 if (srcOutput == dstOutput) { 2429 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2430 return NO_ERROR; 2431 } 2432 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2433 if (srcThread == NULL) { 2434 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2435 return BAD_VALUE; 2436 } 2437 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2438 if (dstThread == NULL) { 2439 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2440 return BAD_VALUE; 2441 } 2442 2443 Mutex::Autolock _dl(dstThread->mLock); 2444 Mutex::Autolock _sl(srcThread->mLock); 2445 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2446} 2447 2448// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2449status_t AudioFlinger::moveEffectChain_l(int sessionId, 2450 AudioFlinger::PlaybackThread *srcThread, 2451 AudioFlinger::PlaybackThread *dstThread, 2452 bool reRegister) 2453{ 2454 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2455 sessionId, srcThread, dstThread); 2456 2457 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2458 if (chain == 0) { 2459 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2460 sessionId, srcThread); 2461 return INVALID_OPERATION; 2462 } 2463 2464 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2465 // so that a new chain is created with correct parameters when first effect is added. This is 2466 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2467 // removed. 2468 srcThread->removeEffectChain_l(chain); 2469 2470 // transfer all effects one by one so that new effect chain is created on new thread with 2471 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2472 sp<EffectChain> dstChain; 2473 uint32_t strategy = 0; // prevent compiler warning 2474 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2475 Vector< sp<EffectModule> > removed; 2476 status_t status = NO_ERROR; 2477 while (effect != 0) { 2478 srcThread->removeEffect_l(effect); 2479 removed.add(effect); 2480 status = dstThread->addEffect_l(effect); 2481 if (status != NO_ERROR) { 2482 break; 2483 } 2484 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2485 if (effect->state() == EffectModule::ACTIVE || 2486 effect->state() == EffectModule::STOPPING) { 2487 effect->start(); 2488 } 2489 // if the move request is not received from audio policy manager, the effect must be 2490 // re-registered with the new strategy and output 2491 if (dstChain == 0) { 2492 dstChain = effect->chain().promote(); 2493 if (dstChain == 0) { 2494 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2495 status = NO_INIT; 2496 break; 2497 } 2498 strategy = dstChain->strategy(); 2499 } 2500 if (reRegister) { 2501 AudioSystem::unregisterEffect(effect->id()); 2502 AudioSystem::registerEffect(&effect->desc(), 2503 dstThread->id(), 2504 strategy, 2505 sessionId, 2506 effect->id()); 2507 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2508 } 2509 effect = chain->getEffectFromId_l(0); 2510 } 2511 2512 if (status != NO_ERROR) { 2513 for (size_t i = 0; i < removed.size(); i++) { 2514 srcThread->addEffect_l(removed[i]); 2515 if (dstChain != 0 && reRegister) { 2516 AudioSystem::unregisterEffect(removed[i]->id()); 2517 AudioSystem::registerEffect(&removed[i]->desc(), 2518 srcThread->id(), 2519 strategy, 2520 sessionId, 2521 removed[i]->id()); 2522 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2523 } 2524 } 2525 } 2526 2527 return status; 2528} 2529 2530bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2531{ 2532 if (mGlobalEffectEnableTime != 0 && 2533 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2534 return true; 2535 } 2536 2537 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2538 sp<EffectChain> ec = 2539 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2540 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2541 return true; 2542 } 2543 } 2544 return false; 2545} 2546 2547void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2548{ 2549 Mutex::Autolock _l(mLock); 2550 2551 mGlobalEffectEnableTime = systemTime(); 2552 2553 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2554 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2555 if (t->mType == ThreadBase::OFFLOAD) { 2556 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2557 } 2558 } 2559 2560} 2561 2562struct Entry { 2563#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2564 char mName[MAX_NAME]; 2565}; 2566 2567int comparEntry(const void *p1, const void *p2) 2568{ 2569 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2570} 2571 2572#ifdef TEE_SINK 2573void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2574{ 2575 NBAIO_Source *teeSource = source.get(); 2576 if (teeSource != NULL) { 2577 // .wav rotation 2578 // There is a benign race condition if 2 threads call this simultaneously. 2579 // They would both traverse the directory, but the result would simply be 2580 // failures at unlink() which are ignored. It's also unlikely since 2581 // normally dumpsys is only done by bugreport or from the command line. 2582 char teePath[32+256]; 2583 strcpy(teePath, "/data/misc/media"); 2584 size_t teePathLen = strlen(teePath); 2585 DIR *dir = opendir(teePath); 2586 teePath[teePathLen++] = '/'; 2587 if (dir != NULL) { 2588#define MAX_SORT 20 // number of entries to sort 2589#define MAX_KEEP 10 // number of entries to keep 2590 struct Entry entries[MAX_SORT]; 2591 size_t entryCount = 0; 2592 while (entryCount < MAX_SORT) { 2593 struct dirent de; 2594 struct dirent *result = NULL; 2595 int rc = readdir_r(dir, &de, &result); 2596 if (rc != 0) { 2597 ALOGW("readdir_r failed %d", rc); 2598 break; 2599 } 2600 if (result == NULL) { 2601 break; 2602 } 2603 if (result != &de) { 2604 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2605 break; 2606 } 2607 // ignore non .wav file entries 2608 size_t nameLen = strlen(de.d_name); 2609 if (nameLen <= 4 || nameLen >= MAX_NAME || 2610 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2611 continue; 2612 } 2613 strcpy(entries[entryCount++].mName, de.d_name); 2614 } 2615 (void) closedir(dir); 2616 if (entryCount > MAX_KEEP) { 2617 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2618 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2619 strcpy(&teePath[teePathLen], entries[i].mName); 2620 (void) unlink(teePath); 2621 } 2622 } 2623 } else { 2624 if (fd >= 0) { 2625 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2626 } 2627 } 2628 char teeTime[16]; 2629 struct timeval tv; 2630 gettimeofday(&tv, NULL); 2631 struct tm tm; 2632 localtime_r(&tv.tv_sec, &tm); 2633 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2634 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2635 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2636 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2637 if (teeFd >= 0) { 2638 char wavHeader[44]; 2639 memcpy(wavHeader, 2640 "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", 2641 sizeof(wavHeader)); 2642 NBAIO_Format format = teeSource->format(); 2643 unsigned channelCount = Format_channelCount(format); 2644 ALOG_ASSERT(channelCount <= FCC_2); 2645 uint32_t sampleRate = Format_sampleRate(format); 2646 wavHeader[22] = channelCount; // number of channels 2647 wavHeader[24] = sampleRate; // sample rate 2648 wavHeader[25] = sampleRate >> 8; 2649 wavHeader[32] = channelCount * 2; // block alignment 2650 write(teeFd, wavHeader, sizeof(wavHeader)); 2651 size_t total = 0; 2652 bool firstRead = true; 2653 for (;;) { 2654#define TEE_SINK_READ 1024 2655 short buffer[TEE_SINK_READ * FCC_2]; 2656 size_t count = TEE_SINK_READ; 2657 ssize_t actual = teeSource->read(buffer, count, 2658 AudioBufferProvider::kInvalidPTS); 2659 bool wasFirstRead = firstRead; 2660 firstRead = false; 2661 if (actual <= 0) { 2662 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2663 continue; 2664 } 2665 break; 2666 } 2667 ALOG_ASSERT(actual <= (ssize_t)count); 2668 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2669 total += actual; 2670 } 2671 lseek(teeFd, (off_t) 4, SEEK_SET); 2672 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2673 write(teeFd, &temp, sizeof(temp)); 2674 lseek(teeFd, (off_t) 40, SEEK_SET); 2675 temp = total * channelCount * sizeof(short); 2676 write(teeFd, &temp, sizeof(temp)); 2677 close(teeFd); 2678 if (fd >= 0) { 2679 dprintf(fd, "tee copied to %s\n", teePath); 2680 } 2681 } else { 2682 if (fd >= 0) { 2683 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2684 } 2685 } 2686 } 2687} 2688#endif 2689 2690// ---------------------------------------------------------------------------- 2691 2692status_t AudioFlinger::onTransact( 2693 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2694{ 2695 return BnAudioFlinger::onTransact(code, data, reply, flags); 2696} 2697 2698}; // namespace android 2699