AudioFlinger.cpp revision 34542acfa25c6413c87a94b6f7cc315a0c496277
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#include <cutils/compiler.h> 41 42#include <system/audio.h> 43#include <hardware/audio.h> 44 45#include "AudioMixer.h" 46#include "AudioFlinger.h" 47#include "ServiceUtilities.h" 48 49#include <media/EffectsFactoryApi.h> 50#include <audio_effects/effect_visualizer.h> 51#include <audio_effects/effect_ns.h> 52#include <audio_effects/effect_aec.h> 53 54#include <audio_utils/primitives.h> 55 56#include <powermanager/PowerManager.h> 57 58#include <common_time/cc_helper.h> 59 60#include <media/IMediaLogService.h> 61 62#include <media/nbaio/Pipe.h> 63#include <media/nbaio/PipeReader.h> 64#include <media/AudioParameter.h> 65#include <private/android_filesystem_config.h> 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 85static const char kHardwareLockedString[] = "Hardware 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// ---------------------------------------------------------------------------- 103 104static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 105{ 106 const hw_module_t *mod; 107 int rc; 108 109 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 110 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 111 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 112 if (rc) { 113 goto out; 114 } 115 rc = audio_hw_device_open(mod, dev); 116 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 117 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 118 if (rc) { 119 goto out; 120 } 121 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 122 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 123 rc = BAD_VALUE; 124 goto out; 125 } 126 return 0; 127 128out: 129 *dev = NULL; 130 return rc; 131} 132 133// ---------------------------------------------------------------------------- 134 135AudioFlinger::AudioFlinger() 136 : BnAudioFlinger(), 137 mPrimaryHardwareDev(NULL), 138 mHardwareStatus(AUDIO_HW_IDLE), 139 mMasterVolume(1.0f), 140 mMasterMute(false), 141 mNextUniqueId(1), 142 mMode(AUDIO_MODE_INVALID), 143 mBtNrecIsOff(false), 144 mIsLowRamDevice(true), 145 mIsDeviceTypeKnown(false) 146{ 147 getpid_cached = getpid(); 148 char value[PROPERTY_VALUE_MAX]; 149 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 150 if (doLog) { 151 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters"); 152 } 153#ifdef TEE_SINK 154 (void) property_get("ro.debuggable", value, "0"); 155 int debuggable = atoi(value); 156 int teeEnabled = 0; 157 if (debuggable) { 158 (void) property_get("af.tee", value, "0"); 159 teeEnabled = atoi(value); 160 } 161 if (teeEnabled & 1) 162 mTeeSinkInputEnabled = true; 163 if (teeEnabled & 2) 164 mTeeSinkOutputEnabled = true; 165 if (teeEnabled & 4) 166 mTeeSinkTrackEnabled = true; 167#endif 168} 169 170void AudioFlinger::onFirstRef() 171{ 172 int rc = 0; 173 174 Mutex::Autolock _l(mLock); 175 176 /* TODO: move all this work into an Init() function */ 177 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 178 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 179 uint32_t int_val; 180 if (1 == sscanf(val_str, "%u", &int_val)) { 181 mStandbyTimeInNsecs = milliseconds(int_val); 182 ALOGI("Using %u mSec as standby time.", int_val); 183 } else { 184 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 185 ALOGI("Using default %u mSec as standby time.", 186 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 187 } 188 } 189 190 mMode = AUDIO_MODE_NORMAL; 191} 192 193AudioFlinger::~AudioFlinger() 194{ 195 while (!mRecordThreads.isEmpty()) { 196 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 197 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 198 } 199 while (!mPlaybackThreads.isEmpty()) { 200 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 201 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 202 } 203 204 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 205 // no mHardwareLock needed, as there are no other references to this 206 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 207 delete mAudioHwDevs.valueAt(i); 208 } 209} 210 211static const char * const audio_interfaces[] = { 212 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 213 AUDIO_HARDWARE_MODULE_ID_A2DP, 214 AUDIO_HARDWARE_MODULE_ID_USB, 215}; 216#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 217 218AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 219 audio_module_handle_t module, 220 audio_devices_t devices) 221{ 222 // if module is 0, the request comes from an old policy manager and we should load 223 // well known modules 224 if (module == 0) { 225 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 226 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 227 loadHwModule_l(audio_interfaces[i]); 228 } 229 // then try to find a module supporting the requested device. 230 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 231 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 232 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 233 if ((dev->get_supported_devices != NULL) && 234 (dev->get_supported_devices(dev) & devices) == devices) 235 return audioHwDevice; 236 } 237 } else { 238 // check a match for the requested module handle 239 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 240 if (audioHwDevice != NULL) { 241 return audioHwDevice; 242 } 243 } 244 245 return NULL; 246} 247 248void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 249{ 250 const size_t SIZE = 256; 251 char buffer[SIZE]; 252 String8 result; 253 254 result.append("Clients:\n"); 255 for (size_t i = 0; i < mClients.size(); ++i) { 256 sp<Client> client = mClients.valueAt(i).promote(); 257 if (client != 0) { 258 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 259 result.append(buffer); 260 } 261 } 262 263 result.append("Global session refs:\n"); 264 result.append(" session pid count\n"); 265 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 266 AudioSessionRef *r = mAudioSessionRefs[i]; 267 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 268 result.append(buffer); 269 } 270 write(fd, result.string(), result.size()); 271} 272 273 274void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 275{ 276 const size_t SIZE = 256; 277 char buffer[SIZE]; 278 String8 result; 279 hardware_call_state hardwareStatus = mHardwareStatus; 280 281 snprintf(buffer, SIZE, "Hardware status: %d\n" 282 "Standby Time mSec: %u\n", 283 hardwareStatus, 284 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 285 result.append(buffer); 286 write(fd, result.string(), result.size()); 287} 288 289void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 290{ 291 const size_t SIZE = 256; 292 char buffer[SIZE]; 293 String8 result; 294 snprintf(buffer, SIZE, "Permission Denial: " 295 "can't dump AudioFlinger from pid=%d, uid=%d\n", 296 IPCThreadState::self()->getCallingPid(), 297 IPCThreadState::self()->getCallingUid()); 298 result.append(buffer); 299 write(fd, result.string(), result.size()); 300} 301 302bool AudioFlinger::dumpTryLock(Mutex& mutex) 303{ 304 bool locked = false; 305 for (int i = 0; i < kDumpLockRetries; ++i) { 306 if (mutex.tryLock() == NO_ERROR) { 307 locked = true; 308 break; 309 } 310 usleep(kDumpLockSleepUs); 311 } 312 return locked; 313} 314 315status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 316{ 317 if (!dumpAllowed()) { 318 dumpPermissionDenial(fd, args); 319 } else { 320 // get state of hardware lock 321 bool hardwareLocked = dumpTryLock(mHardwareLock); 322 if (!hardwareLocked) { 323 String8 result(kHardwareLockedString); 324 write(fd, result.string(), result.size()); 325 } else { 326 mHardwareLock.unlock(); 327 } 328 329 bool locked = dumpTryLock(mLock); 330 331 // failed to lock - AudioFlinger is probably deadlocked 332 if (!locked) { 333 String8 result(kDeadlockedString); 334 write(fd, result.string(), result.size()); 335 } 336 337 dumpClients(fd, args); 338 dumpInternals(fd, args); 339 340 // dump playback threads 341 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 342 mPlaybackThreads.valueAt(i)->dump(fd, args); 343 } 344 345 // dump record threads 346 for (size_t i = 0; i < mRecordThreads.size(); i++) { 347 mRecordThreads.valueAt(i)->dump(fd, args); 348 } 349 350 // dump all hardware devs 351 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 352 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 353 dev->dump(dev, fd); 354 } 355 356#ifdef TEE_SINK 357 // dump the serially shared record tee sink 358 if (mRecordTeeSource != 0) { 359 dumpTee(fd, mRecordTeeSource); 360 } 361#endif 362 363 if (locked) { 364 mLock.unlock(); 365 } 366 367 // append a copy of media.log here by forwarding fd to it, but don't attempt 368 // to lookup the service if it's not running, as it will block for a second 369 if (mLogMemoryDealer != 0) { 370 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 371 if (binder != 0) { 372 fdprintf(fd, "\nmedia.log:\n"); 373 Vector<String16> args; 374 binder->dump(fd, args); 375 } 376 } 377 } 378 return NO_ERROR; 379} 380 381sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 382{ 383 // If pid is already in the mClients wp<> map, then use that entry 384 // (for which promote() is always != 0), otherwise create a new entry and Client. 385 sp<Client> client = mClients.valueFor(pid).promote(); 386 if (client == 0) { 387 client = new Client(this, pid); 388 mClients.add(pid, client); 389 } 390 391 return client; 392} 393 394sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 395{ 396 if (mLogMemoryDealer == 0) { 397 return new NBLog::Writer(); 398 } 399 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 400 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 401 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 402 if (binder != 0) { 403 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 404 } 405 return writer; 406} 407 408void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 409{ 410 if (writer == 0) { 411 return; 412 } 413 sp<IMemory> iMemory(writer->getIMemory()); 414 if (iMemory == 0) { 415 return; 416 } 417 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 418 if (binder != 0) { 419 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 420 // Now the media.log remote reference to IMemory is gone. 421 // When our last local reference to IMemory also drops to zero, 422 // the IMemory destructor will deallocate the region from mMemoryDealer. 423 } 424} 425 426// IAudioFlinger interface 427 428 429sp<IAudioTrack> AudioFlinger::createTrack( 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 audio_channel_mask_t channelMask, 434 size_t frameCount, 435 IAudioFlinger::track_flags_t *flags, 436 const sp<IMemory>& sharedBuffer, 437 audio_io_handle_t output, 438 pid_t tid, 439 int *sessionId, 440 status_t *status) 441{ 442 sp<PlaybackThread::Track> track; 443 sp<TrackHandle> trackHandle; 444 sp<Client> client; 445 status_t lStatus; 446 int lSessionId; 447 448 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 449 // but if someone uses binder directly they could bypass that and cause us to crash 450 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 451 ALOGE("createTrack() invalid stream type %d", streamType); 452 lStatus = BAD_VALUE; 453 goto Exit; 454 } 455 456 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 457 // and we don't yet support 8.24 or 32-bit PCM 458 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 459 ALOGE("createTrack() invalid format %d", format); 460 lStatus = BAD_VALUE; 461 goto Exit; 462 } 463 464 { 465 Mutex::Autolock _l(mLock); 466 PlaybackThread *thread = checkPlaybackThread_l(output); 467 PlaybackThread *effectThread = NULL; 468 if (thread == NULL) { 469 ALOGE("no playback thread found for output handle %d", output); 470 lStatus = BAD_VALUE; 471 goto Exit; 472 } 473 474 pid_t pid = IPCThreadState::self()->getCallingPid(); 475 client = registerPid_l(pid); 476 477 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 478 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 479 // check if an effect chain with the same session ID is present on another 480 // output thread and move it here. 481 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 482 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 483 if (mPlaybackThreads.keyAt(i) != output) { 484 uint32_t sessions = t->hasAudioSession(*sessionId); 485 if (sessions & PlaybackThread::EFFECT_SESSION) { 486 effectThread = t.get(); 487 break; 488 } 489 } 490 } 491 lSessionId = *sessionId; 492 } else { 493 // if no audio session id is provided, create one here 494 lSessionId = nextUniqueId(); 495 if (sessionId != NULL) { 496 *sessionId = lSessionId; 497 } 498 } 499 ALOGV("createTrack() lSessionId: %d", lSessionId); 500 501 track = thread->createTrack_l(client, streamType, sampleRate, format, 502 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 503 504 // move effect chain to this output thread if an effect on same session was waiting 505 // for a track to be created 506 if (lStatus == NO_ERROR && effectThread != NULL) { 507 Mutex::Autolock _dl(thread->mLock); 508 Mutex::Autolock _sl(effectThread->mLock); 509 moveEffectChain_l(lSessionId, effectThread, thread, true); 510 } 511 512 // Look for sync events awaiting for a session to be used. 513 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 514 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 515 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 516 if (lStatus == NO_ERROR) { 517 (void) track->setSyncEvent(mPendingSyncEvents[i]); 518 } else { 519 mPendingSyncEvents[i]->cancel(); 520 } 521 mPendingSyncEvents.removeAt(i); 522 i--; 523 } 524 } 525 } 526 } 527 if (lStatus == NO_ERROR) { 528 trackHandle = new TrackHandle(track); 529 } else { 530 // remove local strong reference to Client before deleting the Track so that the Client 531 // destructor is called by the TrackBase destructor with mLock held 532 client.clear(); 533 track.clear(); 534 } 535 536Exit: 537 if (status != NULL) { 538 *status = lStatus; 539 } 540 return trackHandle; 541} 542 543uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 544{ 545 Mutex::Autolock _l(mLock); 546 PlaybackThread *thread = checkPlaybackThread_l(output); 547 if (thread == NULL) { 548 ALOGW("sampleRate() unknown thread %d", output); 549 return 0; 550 } 551 return thread->sampleRate(); 552} 553 554int AudioFlinger::channelCount(audio_io_handle_t output) const 555{ 556 Mutex::Autolock _l(mLock); 557 PlaybackThread *thread = checkPlaybackThread_l(output); 558 if (thread == NULL) { 559 ALOGW("channelCount() unknown thread %d", output); 560 return 0; 561 } 562 return thread->channelCount(); 563} 564 565audio_format_t AudioFlinger::format(audio_io_handle_t output) const 566{ 567 Mutex::Autolock _l(mLock); 568 PlaybackThread *thread = checkPlaybackThread_l(output); 569 if (thread == NULL) { 570 ALOGW("format() unknown thread %d", output); 571 return AUDIO_FORMAT_INVALID; 572 } 573 return thread->format(); 574} 575 576size_t AudioFlinger::frameCount(audio_io_handle_t output) const 577{ 578 Mutex::Autolock _l(mLock); 579 PlaybackThread *thread = checkPlaybackThread_l(output); 580 if (thread == NULL) { 581 ALOGW("frameCount() unknown thread %d", output); 582 return 0; 583 } 584 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 585 // should examine all callers and fix them to handle smaller counts 586 return thread->frameCount(); 587} 588 589uint32_t AudioFlinger::latency(audio_io_handle_t output) const 590{ 591 Mutex::Autolock _l(mLock); 592 PlaybackThread *thread = checkPlaybackThread_l(output); 593 if (thread == NULL) { 594 ALOGW("latency(): no playback thread found for output handle %d", output); 595 return 0; 596 } 597 return thread->latency(); 598} 599 600status_t AudioFlinger::setMasterVolume(float value) 601{ 602 status_t ret = initCheck(); 603 if (ret != NO_ERROR) { 604 return ret; 605 } 606 607 // check calling permissions 608 if (!settingsAllowed()) { 609 return PERMISSION_DENIED; 610 } 611 612 Mutex::Autolock _l(mLock); 613 mMasterVolume = value; 614 615 // Set master volume in the HALs which support it. 616 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 617 AutoMutex lock(mHardwareLock); 618 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 619 620 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 621 if (dev->canSetMasterVolume()) { 622 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 623 } 624 mHardwareStatus = AUDIO_HW_IDLE; 625 } 626 627 // Now set the master volume in each playback thread. Playback threads 628 // assigned to HALs which do not have master volume support will apply 629 // master volume during the mix operation. Threads with HALs which do 630 // support master volume will simply ignore the setting. 631 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 632 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 633 634 return NO_ERROR; 635} 636 637status_t AudioFlinger::setMode(audio_mode_t mode) 638{ 639 status_t ret = initCheck(); 640 if (ret != NO_ERROR) { 641 return ret; 642 } 643 644 // check calling permissions 645 if (!settingsAllowed()) { 646 return PERMISSION_DENIED; 647 } 648 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 649 ALOGW("Illegal value: setMode(%d)", mode); 650 return BAD_VALUE; 651 } 652 653 { // scope for the lock 654 AutoMutex lock(mHardwareLock); 655 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = dev->set_mode(dev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 685 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 686 ret = dev->set_mic_mute(dev, state); 687 mHardwareStatus = AUDIO_HW_IDLE; 688 return ret; 689} 690 691bool AudioFlinger::getMicMute() const 692{ 693 status_t ret = initCheck(); 694 if (ret != NO_ERROR) { 695 return false; 696 } 697 698 bool state = AUDIO_MODE_INVALID; 699 AutoMutex lock(mHardwareLock); 700 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 701 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 702 dev->get_mic_mute(dev, &state); 703 mHardwareStatus = AUDIO_HW_IDLE; 704 return state; 705} 706 707status_t AudioFlinger::setMasterMute(bool muted) 708{ 709 status_t ret = initCheck(); 710 if (ret != NO_ERROR) { 711 return ret; 712 } 713 714 // check calling permissions 715 if (!settingsAllowed()) { 716 return PERMISSION_DENIED; 717 } 718 719 Mutex::Autolock _l(mLock); 720 mMasterMute = muted; 721 722 // Set master mute in the HALs which support it. 723 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 724 AutoMutex lock(mHardwareLock); 725 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 726 727 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 728 if (dev->canSetMasterMute()) { 729 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 730 } 731 mHardwareStatus = AUDIO_HW_IDLE; 732 } 733 734 // Now set the master mute in each playback thread. Playback threads 735 // assigned to HALs which do not have master mute support will apply master 736 // mute during the mix operation. Threads with HALs which do support master 737 // mute will simply ignore the setting. 738 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 739 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 740 741 return NO_ERROR; 742} 743 744float AudioFlinger::masterVolume() const 745{ 746 Mutex::Autolock _l(mLock); 747 return masterVolume_l(); 748} 749 750bool AudioFlinger::masterMute() const 751{ 752 Mutex::Autolock _l(mLock); 753 return masterMute_l(); 754} 755 756float AudioFlinger::masterVolume_l() const 757{ 758 return mMasterVolume; 759} 760 761bool AudioFlinger::masterMute_l() const 762{ 763 return mMasterMute; 764} 765 766status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 767 audio_io_handle_t output) 768{ 769 // check calling permissions 770 if (!settingsAllowed()) { 771 return PERMISSION_DENIED; 772 } 773 774 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 775 ALOGE("setStreamVolume() invalid stream %d", stream); 776 return BAD_VALUE; 777 } 778 779 AutoMutex lock(mLock); 780 PlaybackThread *thread = NULL; 781 if (output) { 782 thread = checkPlaybackThread_l(output); 783 if (thread == NULL) { 784 return BAD_VALUE; 785 } 786 } 787 788 mStreamTypes[stream].volume = value; 789 790 if (thread == NULL) { 791 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 792 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 793 } 794 } else { 795 thread->setStreamVolume(stream, value); 796 } 797 798 return NO_ERROR; 799} 800 801status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 802{ 803 // check calling permissions 804 if (!settingsAllowed()) { 805 return PERMISSION_DENIED; 806 } 807 808 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 809 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 810 ALOGE("setStreamMute() invalid stream %d", stream); 811 return BAD_VALUE; 812 } 813 814 AutoMutex lock(mLock); 815 mStreamTypes[stream].mute = muted; 816 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 817 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 818 819 return NO_ERROR; 820} 821 822float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 823{ 824 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 825 return 0.0f; 826 } 827 828 AutoMutex lock(mLock); 829 float volume; 830 if (output) { 831 PlaybackThread *thread = checkPlaybackThread_l(output); 832 if (thread == NULL) { 833 return 0.0f; 834 } 835 volume = thread->streamVolume(stream); 836 } else { 837 volume = streamVolume_l(stream); 838 } 839 840 return volume; 841} 842 843bool AudioFlinger::streamMute(audio_stream_type_t stream) const 844{ 845 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 846 return true; 847 } 848 849 AutoMutex lock(mLock); 850 return streamMute_l(stream); 851} 852 853status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 854{ 855 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 856 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 857 858 // check calling permissions 859 if (!settingsAllowed()) { 860 return PERMISSION_DENIED; 861 } 862 863 // ioHandle == 0 means the parameters are global to the audio hardware interface 864 if (ioHandle == 0) { 865 Mutex::Autolock _l(mLock); 866 status_t final_result = NO_ERROR; 867 { 868 AutoMutex lock(mHardwareLock); 869 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 870 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 871 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 872 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 873 final_result = result ?: final_result; 874 } 875 mHardwareStatus = AUDIO_HW_IDLE; 876 } 877 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 878 AudioParameter param = AudioParameter(keyValuePairs); 879 String8 value; 880 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 881 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 882 if (mBtNrecIsOff != btNrecIsOff) { 883 for (size_t i = 0; i < mRecordThreads.size(); i++) { 884 sp<RecordThread> thread = mRecordThreads.valueAt(i); 885 audio_devices_t device = thread->inDevice(); 886 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 887 // collect all of the thread's session IDs 888 KeyedVector<int, bool> ids = thread->sessionIds(); 889 // suspend effects associated with those session IDs 890 for (size_t j = 0; j < ids.size(); ++j) { 891 int sessionId = ids.keyAt(j); 892 thread->setEffectSuspended(FX_IID_AEC, 893 suspend, 894 sessionId); 895 thread->setEffectSuspended(FX_IID_NS, 896 suspend, 897 sessionId); 898 } 899 } 900 mBtNrecIsOff = btNrecIsOff; 901 } 902 } 903 String8 screenState; 904 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 905 bool isOff = screenState == "off"; 906 if (isOff != (AudioFlinger::mScreenState & 1)) { 907 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 908 } 909 } 910 return final_result; 911 } 912 913 // hold a strong ref on thread in case closeOutput() or closeInput() is called 914 // and the thread is exited once the lock is released 915 sp<ThreadBase> thread; 916 { 917 Mutex::Autolock _l(mLock); 918 thread = checkPlaybackThread_l(ioHandle); 919 if (thread == 0) { 920 thread = checkRecordThread_l(ioHandle); 921 } else if (thread == primaryPlaybackThread_l()) { 922 // indicate output device change to all input threads for pre processing 923 AudioParameter param = AudioParameter(keyValuePairs); 924 int value; 925 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 926 (value != 0)) { 927 for (size_t i = 0; i < mRecordThreads.size(); i++) { 928 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 929 } 930 } 931 } 932 } 933 if (thread != 0) { 934 return thread->setParameters(keyValuePairs); 935 } 936 return BAD_VALUE; 937} 938 939String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 940{ 941 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 942 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 943 944 Mutex::Autolock _l(mLock); 945 946 if (ioHandle == 0) { 947 String8 out_s8; 948 949 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 950 char *s; 951 { 952 AutoMutex lock(mHardwareLock); 953 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 954 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 955 s = dev->get_parameters(dev, keys.string()); 956 mHardwareStatus = AUDIO_HW_IDLE; 957 } 958 out_s8 += String8(s ? s : ""); 959 free(s); 960 } 961 return out_s8; 962 } 963 964 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 965 if (playbackThread != NULL) { 966 return playbackThread->getParameters(keys); 967 } 968 RecordThread *recordThread = checkRecordThread_l(ioHandle); 969 if (recordThread != NULL) { 970 return recordThread->getParameters(keys); 971 } 972 return String8(""); 973} 974 975size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 976 audio_channel_mask_t channelMask) const 977{ 978 status_t ret = initCheck(); 979 if (ret != NO_ERROR) { 980 return 0; 981 } 982 983 AutoMutex lock(mHardwareLock); 984 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 985 struct audio_config config; 986 memset(&config, 0, sizeof(config)); 987 config.sample_rate = sampleRate; 988 config.channel_mask = channelMask; 989 config.format = format; 990 991 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 992 size_t size = dev->get_input_buffer_size(dev, &config); 993 mHardwareStatus = AUDIO_HW_IDLE; 994 return size; 995} 996 997unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 998{ 999 Mutex::Autolock _l(mLock); 1000 1001 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1002 if (recordThread != NULL) { 1003 return recordThread->getInputFramesLost(); 1004 } 1005 return 0; 1006} 1007 1008status_t AudioFlinger::setVoiceVolume(float value) 1009{ 1010 status_t ret = initCheck(); 1011 if (ret != NO_ERROR) { 1012 return ret; 1013 } 1014 1015 // check calling permissions 1016 if (!settingsAllowed()) { 1017 return PERMISSION_DENIED; 1018 } 1019 1020 AutoMutex lock(mHardwareLock); 1021 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1022 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1023 ret = dev->set_voice_volume(dev, value); 1024 mHardwareStatus = AUDIO_HW_IDLE; 1025 1026 return ret; 1027} 1028 1029status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1030 audio_io_handle_t output) const 1031{ 1032 status_t status; 1033 1034 Mutex::Autolock _l(mLock); 1035 1036 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1037 if (playbackThread != NULL) { 1038 return playbackThread->getRenderPosition(halFrames, dspFrames); 1039 } 1040 1041 return BAD_VALUE; 1042} 1043 1044void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1045{ 1046 1047 Mutex::Autolock _l(mLock); 1048 1049 pid_t pid = IPCThreadState::self()->getCallingPid(); 1050 if (mNotificationClients.indexOfKey(pid) < 0) { 1051 sp<NotificationClient> notificationClient = new NotificationClient(this, 1052 client, 1053 pid); 1054 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1055 1056 mNotificationClients.add(pid, notificationClient); 1057 1058 sp<IBinder> binder = client->asBinder(); 1059 binder->linkToDeath(notificationClient); 1060 1061 // the config change is always sent from playback or record threads to avoid deadlock 1062 // with AudioSystem::gLock 1063 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1064 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1065 } 1066 1067 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1068 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1069 } 1070 } 1071} 1072 1073void AudioFlinger::removeNotificationClient(pid_t pid) 1074{ 1075 Mutex::Autolock _l(mLock); 1076 1077 mNotificationClients.removeItem(pid); 1078 1079 ALOGV("%d died, releasing its sessions", pid); 1080 size_t num = mAudioSessionRefs.size(); 1081 bool removed = false; 1082 for (size_t i = 0; i< num; ) { 1083 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1084 ALOGV(" pid %d @ %d", ref->mPid, i); 1085 if (ref->mPid == pid) { 1086 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1087 mAudioSessionRefs.removeAt(i); 1088 delete ref; 1089 removed = true; 1090 num--; 1091 } else { 1092 i++; 1093 } 1094 } 1095 if (removed) { 1096 purgeStaleEffects_l(); 1097 } 1098} 1099 1100// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1101void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1102{ 1103 size_t size = mNotificationClients.size(); 1104 for (size_t i = 0; i < size; i++) { 1105 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1106 param2); 1107 } 1108} 1109 1110// removeClient_l() must be called with AudioFlinger::mLock held 1111void AudioFlinger::removeClient_l(pid_t pid) 1112{ 1113 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1114 IPCThreadState::self()->getCallingPid()); 1115 mClients.removeItem(pid); 1116} 1117 1118// getEffectThread_l() must be called with AudioFlinger::mLock held 1119sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1120{ 1121 sp<PlaybackThread> thread; 1122 1123 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1124 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1125 ALOG_ASSERT(thread == 0); 1126 thread = mPlaybackThreads.valueAt(i); 1127 } 1128 } 1129 1130 return thread; 1131} 1132 1133 1134 1135// ---------------------------------------------------------------------------- 1136 1137AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1138 : RefBase(), 1139 mAudioFlinger(audioFlinger), 1140 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1141 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1142 mPid(pid), 1143 mTimedTrackCount(0) 1144{ 1145 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1146} 1147 1148// Client destructor must be called with AudioFlinger::mLock held 1149AudioFlinger::Client::~Client() 1150{ 1151 mAudioFlinger->removeClient_l(mPid); 1152} 1153 1154sp<MemoryDealer> AudioFlinger::Client::heap() const 1155{ 1156 return mMemoryDealer; 1157} 1158 1159// Reserve one of the limited slots for a timed audio track associated 1160// with this client 1161bool AudioFlinger::Client::reserveTimedTrack() 1162{ 1163 const int kMaxTimedTracksPerClient = 4; 1164 1165 Mutex::Autolock _l(mTimedTrackLock); 1166 1167 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1168 ALOGW("can not create timed track - pid %d has exceeded the limit", 1169 mPid); 1170 return false; 1171 } 1172 1173 mTimedTrackCount++; 1174 return true; 1175} 1176 1177// Release a slot for a timed audio track 1178void AudioFlinger::Client::releaseTimedTrack() 1179{ 1180 Mutex::Autolock _l(mTimedTrackLock); 1181 mTimedTrackCount--; 1182} 1183 1184// ---------------------------------------------------------------------------- 1185 1186AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1187 const sp<IAudioFlingerClient>& client, 1188 pid_t pid) 1189 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1190{ 1191} 1192 1193AudioFlinger::NotificationClient::~NotificationClient() 1194{ 1195} 1196 1197void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1198{ 1199 sp<NotificationClient> keep(this); 1200 mAudioFlinger->removeNotificationClient(mPid); 1201} 1202 1203 1204// ---------------------------------------------------------------------------- 1205 1206sp<IAudioRecord> AudioFlinger::openRecord( 1207 audio_io_handle_t input, 1208 uint32_t sampleRate, 1209 audio_format_t format, 1210 audio_channel_mask_t channelMask, 1211 size_t frameCount, 1212 IAudioFlinger::track_flags_t flags, 1213 pid_t tid, 1214 int *sessionId, 1215 status_t *status) 1216{ 1217 sp<RecordThread::RecordTrack> recordTrack; 1218 sp<RecordHandle> recordHandle; 1219 sp<Client> client; 1220 status_t lStatus; 1221 RecordThread *thread; 1222 size_t inFrameCount; 1223 int lSessionId; 1224 1225 // check calling permissions 1226 if (!recordingAllowed()) { 1227 lStatus = PERMISSION_DENIED; 1228 goto Exit; 1229 } 1230 1231 // add client to list 1232 { // scope for mLock 1233 Mutex::Autolock _l(mLock); 1234 thread = checkRecordThread_l(input); 1235 if (thread == NULL) { 1236 lStatus = BAD_VALUE; 1237 goto Exit; 1238 } 1239 1240 pid_t pid = IPCThreadState::self()->getCallingPid(); 1241 client = registerPid_l(pid); 1242 1243 // If no audio session id is provided, create one here 1244 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1245 lSessionId = *sessionId; 1246 } else { 1247 lSessionId = nextUniqueId(); 1248 if (sessionId != NULL) { 1249 *sessionId = lSessionId; 1250 } 1251 } 1252 // create new record track. 1253 // The record track uses one track in mHardwareMixerThread by convention. 1254 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1255 frameCount, lSessionId, flags, tid, &lStatus); 1256 } 1257 if (lStatus != NO_ERROR) { 1258 // remove local strong reference to Client before deleting the RecordTrack so that the 1259 // Client destructor is called by the TrackBase destructor with mLock held 1260 client.clear(); 1261 recordTrack.clear(); 1262 goto Exit; 1263 } 1264 1265 // return to handle to client 1266 recordHandle = new RecordHandle(recordTrack); 1267 lStatus = NO_ERROR; 1268 1269Exit: 1270 if (status) { 1271 *status = lStatus; 1272 } 1273 return recordHandle; 1274} 1275 1276 1277 1278// ---------------------------------------------------------------------------- 1279 1280audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1281{ 1282 if (!settingsAllowed()) { 1283 return 0; 1284 } 1285 Mutex::Autolock _l(mLock); 1286 return loadHwModule_l(name); 1287} 1288 1289// loadHwModule_l() must be called with AudioFlinger::mLock held 1290audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1291{ 1292 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1293 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1294 ALOGW("loadHwModule() module %s already loaded", name); 1295 return mAudioHwDevs.keyAt(i); 1296 } 1297 } 1298 1299 audio_hw_device_t *dev; 1300 1301 int rc = load_audio_interface(name, &dev); 1302 if (rc) { 1303 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1304 return 0; 1305 } 1306 1307 mHardwareStatus = AUDIO_HW_INIT; 1308 rc = dev->init_check(dev); 1309 mHardwareStatus = AUDIO_HW_IDLE; 1310 if (rc) { 1311 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1312 return 0; 1313 } 1314 1315 // Check and cache this HAL's level of support for master mute and master 1316 // volume. If this is the first HAL opened, and it supports the get 1317 // methods, use the initial values provided by the HAL as the current 1318 // master mute and volume settings. 1319 1320 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1321 { // scope for auto-lock pattern 1322 AutoMutex lock(mHardwareLock); 1323 1324 if (0 == mAudioHwDevs.size()) { 1325 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1326 if (NULL != dev->get_master_volume) { 1327 float mv; 1328 if (OK == dev->get_master_volume(dev, &mv)) { 1329 mMasterVolume = mv; 1330 } 1331 } 1332 1333 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1334 if (NULL != dev->get_master_mute) { 1335 bool mm; 1336 if (OK == dev->get_master_mute(dev, &mm)) { 1337 mMasterMute = mm; 1338 } 1339 } 1340 } 1341 1342 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1343 if ((NULL != dev->set_master_volume) && 1344 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1345 flags = static_cast<AudioHwDevice::Flags>(flags | 1346 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1347 } 1348 1349 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1350 if ((NULL != dev->set_master_mute) && 1351 (OK == dev->set_master_mute(dev, mMasterMute))) { 1352 flags = static_cast<AudioHwDevice::Flags>(flags | 1353 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1354 } 1355 1356 mHardwareStatus = AUDIO_HW_IDLE; 1357 } 1358 1359 audio_module_handle_t handle = nextUniqueId(); 1360 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1361 1362 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1363 name, dev->common.module->name, dev->common.module->id, handle); 1364 1365 return handle; 1366 1367} 1368 1369// ---------------------------------------------------------------------------- 1370 1371uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1372{ 1373 Mutex::Autolock _l(mLock); 1374 PlaybackThread *thread = primaryPlaybackThread_l(); 1375 return thread != NULL ? thread->sampleRate() : 0; 1376} 1377 1378size_t AudioFlinger::getPrimaryOutputFrameCount() 1379{ 1380 Mutex::Autolock _l(mLock); 1381 PlaybackThread *thread = primaryPlaybackThread_l(); 1382 return thread != NULL ? thread->frameCountHAL() : 0; 1383} 1384 1385// ---------------------------------------------------------------------------- 1386 1387status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1388{ 1389 uid_t uid = IPCThreadState::self()->getCallingUid(); 1390 if (uid != AID_SYSTEM) { 1391 return PERMISSION_DENIED; 1392 } 1393 Mutex::Autolock _l(mLock); 1394 if (mIsDeviceTypeKnown) { 1395 return INVALID_OPERATION; 1396 } 1397 mIsLowRamDevice = isLowRamDevice; 1398 mIsDeviceTypeKnown = true; 1399 return NO_ERROR; 1400} 1401 1402// ---------------------------------------------------------------------------- 1403 1404audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1405 audio_devices_t *pDevices, 1406 uint32_t *pSamplingRate, 1407 audio_format_t *pFormat, 1408 audio_channel_mask_t *pChannelMask, 1409 uint32_t *pLatencyMs, 1410 audio_output_flags_t flags, 1411 const audio_offload_info_t *offloadInfo) 1412{ 1413 PlaybackThread *thread = NULL; 1414 struct audio_config config; 1415 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1416 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1417 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1418 if (offloadInfo) { 1419 config.offload_info = *offloadInfo; 1420 } 1421 1422 audio_stream_out_t *outStream = NULL; 1423 AudioHwDevice *outHwDev; 1424 1425 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 1426 module, 1427 (pDevices != NULL) ? *pDevices : 0, 1428 config.sample_rate, 1429 config.format, 1430 config.channel_mask, 1431 flags); 1432 1433 if (pDevices == NULL || *pDevices == 0) { 1434 return 0; 1435 } 1436 1437 Mutex::Autolock _l(mLock); 1438 1439 outHwDev = findSuitableHwDev_l(module, *pDevices); 1440 if (outHwDev == NULL) 1441 return 0; 1442 1443 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1444 audio_io_handle_t id = nextUniqueId(); 1445 1446 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1447 1448 status_t status = hwDevHal->open_output_stream(hwDevHal, 1449 id, 1450 *pDevices, 1451 (audio_output_flags_t)flags, 1452 &config, 1453 &outStream); 1454 1455 mHardwareStatus = AUDIO_HW_IDLE; 1456 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 1457 "Channels %x, status %d", 1458 outStream, 1459 config.sample_rate, 1460 config.format, 1461 config.channel_mask, 1462 status); 1463 1464 if (status == NO_ERROR && outStream != NULL) { 1465 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 1466 1467 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1468 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1469 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1470 thread = new DirectOutputThread(this, output, id, *pDevices); 1471 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1472 } else { 1473 thread = new MixerThread(this, output, id, *pDevices); 1474 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1475 } 1476 mPlaybackThreads.add(id, thread); 1477 1478 if (pSamplingRate != NULL) { 1479 *pSamplingRate = config.sample_rate; 1480 } 1481 if (pFormat != NULL) { 1482 *pFormat = config.format; 1483 } 1484 if (pChannelMask != NULL) { 1485 *pChannelMask = config.channel_mask; 1486 } 1487 if (pLatencyMs != NULL) { 1488 *pLatencyMs = thread->latency(); 1489 } 1490 1491 // notify client processes of the new output creation 1492 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1493 1494 // the first primary output opened designates the primary hw device 1495 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1496 ALOGI("Using module %d has the primary audio interface", module); 1497 mPrimaryHardwareDev = outHwDev; 1498 1499 AutoMutex lock(mHardwareLock); 1500 mHardwareStatus = AUDIO_HW_SET_MODE; 1501 hwDevHal->set_mode(hwDevHal, mMode); 1502 mHardwareStatus = AUDIO_HW_IDLE; 1503 } 1504 return id; 1505 } 1506 1507 return 0; 1508} 1509 1510audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1511 audio_io_handle_t output2) 1512{ 1513 Mutex::Autolock _l(mLock); 1514 MixerThread *thread1 = checkMixerThread_l(output1); 1515 MixerThread *thread2 = checkMixerThread_l(output2); 1516 1517 if (thread1 == NULL || thread2 == NULL) { 1518 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1519 output2); 1520 return 0; 1521 } 1522 1523 audio_io_handle_t id = nextUniqueId(); 1524 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1525 thread->addOutputTrack(thread2); 1526 mPlaybackThreads.add(id, thread); 1527 // notify client processes of the new output creation 1528 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1529 return id; 1530} 1531 1532status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1533{ 1534 return closeOutput_nonvirtual(output); 1535} 1536 1537status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1538{ 1539 // keep strong reference on the playback thread so that 1540 // it is not destroyed while exit() is executed 1541 sp<PlaybackThread> thread; 1542 { 1543 Mutex::Autolock _l(mLock); 1544 thread = checkPlaybackThread_l(output); 1545 if (thread == NULL) { 1546 return BAD_VALUE; 1547 } 1548 1549 ALOGV("closeOutput() %d", output); 1550 1551 if (thread->type() == ThreadBase::MIXER) { 1552 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1553 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1554 DuplicatingThread *dupThread = 1555 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1556 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1557 } 1558 } 1559 } 1560 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1561 mPlaybackThreads.removeItem(output); 1562 } 1563 thread->exit(); 1564 // The thread entity (active unit of execution) is no longer running here, 1565 // but the ThreadBase container still exists. 1566 1567 if (thread->type() != ThreadBase::DUPLICATING) { 1568 AudioStreamOut *out = thread->clearOutput(); 1569 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1570 // from now on thread->mOutput is NULL 1571 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1572 delete out; 1573 } 1574 return NO_ERROR; 1575} 1576 1577status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1578{ 1579 Mutex::Autolock _l(mLock); 1580 PlaybackThread *thread = checkPlaybackThread_l(output); 1581 1582 if (thread == NULL) { 1583 return BAD_VALUE; 1584 } 1585 1586 ALOGV("suspendOutput() %d", output); 1587 thread->suspend(); 1588 1589 return NO_ERROR; 1590} 1591 1592status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1593{ 1594 Mutex::Autolock _l(mLock); 1595 PlaybackThread *thread = checkPlaybackThread_l(output); 1596 1597 if (thread == NULL) { 1598 return BAD_VALUE; 1599 } 1600 1601 ALOGV("restoreOutput() %d", output); 1602 1603 thread->restore(); 1604 1605 return NO_ERROR; 1606} 1607 1608audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1609 audio_devices_t *pDevices, 1610 uint32_t *pSamplingRate, 1611 audio_format_t *pFormat, 1612 audio_channel_mask_t *pChannelMask) 1613{ 1614 status_t status; 1615 RecordThread *thread = NULL; 1616 struct audio_config config; 1617 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1618 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1619 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1620 1621 uint32_t reqSamplingRate = config.sample_rate; 1622 audio_format_t reqFormat = config.format; 1623 audio_channel_mask_t reqChannels = config.channel_mask; 1624 audio_stream_in_t *inStream = NULL; 1625 AudioHwDevice *inHwDev; 1626 1627 if (pDevices == NULL || *pDevices == 0) { 1628 return 0; 1629 } 1630 1631 Mutex::Autolock _l(mLock); 1632 1633 inHwDev = findSuitableHwDev_l(module, *pDevices); 1634 if (inHwDev == NULL) 1635 return 0; 1636 1637 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1638 audio_io_handle_t id = nextUniqueId(); 1639 1640 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1641 &inStream); 1642 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1643 "status %d", 1644 inStream, 1645 config.sample_rate, 1646 config.format, 1647 config.channel_mask, 1648 status); 1649 1650 // If the input could not be opened with the requested parameters and we can handle the 1651 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1652 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1653 if (status == BAD_VALUE && 1654 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1655 (config.sample_rate <= 2 * reqSamplingRate) && 1656 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 1657 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1658 inStream = NULL; 1659 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1660 } 1661 1662 if (status == NO_ERROR && inStream != NULL) { 1663 1664#ifdef TEE_SINK 1665 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1666 // or (re-)create if current Pipe is idle and does not match the new format 1667 sp<NBAIO_Sink> teeSink; 1668 enum { 1669 TEE_SINK_NO, // don't copy input 1670 TEE_SINK_NEW, // copy input using a new pipe 1671 TEE_SINK_OLD, // copy input using an existing pipe 1672 } kind; 1673 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1674 popcount(inStream->common.get_channels(&inStream->common))); 1675 if (!mTeeSinkInputEnabled) { 1676 kind = TEE_SINK_NO; 1677 } else if (format == Format_Invalid) { 1678 kind = TEE_SINK_NO; 1679 } else if (mRecordTeeSink == 0) { 1680 kind = TEE_SINK_NEW; 1681 } else if (mRecordTeeSink->getStrongCount() != 1) { 1682 kind = TEE_SINK_NO; 1683 } else if (format == mRecordTeeSink->format()) { 1684 kind = TEE_SINK_OLD; 1685 } else { 1686 kind = TEE_SINK_NEW; 1687 } 1688 switch (kind) { 1689 case TEE_SINK_NEW: { 1690 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1691 size_t numCounterOffers = 0; 1692 const NBAIO_Format offers[1] = {format}; 1693 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1694 ALOG_ASSERT(index == 0); 1695 PipeReader *pipeReader = new PipeReader(*pipe); 1696 numCounterOffers = 0; 1697 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1698 ALOG_ASSERT(index == 0); 1699 mRecordTeeSink = pipe; 1700 mRecordTeeSource = pipeReader; 1701 teeSink = pipe; 1702 } 1703 break; 1704 case TEE_SINK_OLD: 1705 teeSink = mRecordTeeSink; 1706 break; 1707 case TEE_SINK_NO: 1708 default: 1709 break; 1710 } 1711#endif 1712 1713 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1714 1715 // Start record thread 1716 // RecorThread require both input and output device indication to forward to audio 1717 // pre processing modules 1718 thread = new RecordThread(this, 1719 input, 1720 reqSamplingRate, 1721 reqChannels, 1722 id, 1723 primaryOutputDevice_l(), 1724 *pDevices 1725#ifdef TEE_SINK 1726 , teeSink 1727#endif 1728 ); 1729 mRecordThreads.add(id, thread); 1730 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1731 if (pSamplingRate != NULL) { 1732 *pSamplingRate = reqSamplingRate; 1733 } 1734 if (pFormat != NULL) { 1735 *pFormat = config.format; 1736 } 1737 if (pChannelMask != NULL) { 1738 *pChannelMask = reqChannels; 1739 } 1740 1741 // notify client processes of the new input creation 1742 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1743 return id; 1744 } 1745 1746 return 0; 1747} 1748 1749status_t AudioFlinger::closeInput(audio_io_handle_t input) 1750{ 1751 return closeInput_nonvirtual(input); 1752} 1753 1754status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1755{ 1756 // keep strong reference on the record thread so that 1757 // it is not destroyed while exit() is executed 1758 sp<RecordThread> thread; 1759 { 1760 Mutex::Autolock _l(mLock); 1761 thread = checkRecordThread_l(input); 1762 if (thread == 0) { 1763 return BAD_VALUE; 1764 } 1765 1766 ALOGV("closeInput() %d", input); 1767 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1768 mRecordThreads.removeItem(input); 1769 } 1770 thread->exit(); 1771 // The thread entity (active unit of execution) is no longer running here, 1772 // but the ThreadBase container still exists. 1773 1774 AudioStreamIn *in = thread->clearInput(); 1775 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1776 // from now on thread->mInput is NULL 1777 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1778 delete in; 1779 1780 return NO_ERROR; 1781} 1782 1783status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1784{ 1785 Mutex::Autolock _l(mLock); 1786 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1787 1788 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1789 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1790 thread->invalidateTracks(stream); 1791 } 1792 1793 return NO_ERROR; 1794} 1795 1796 1797int AudioFlinger::newAudioSessionId() 1798{ 1799 return nextUniqueId(); 1800} 1801 1802void AudioFlinger::acquireAudioSessionId(int audioSession) 1803{ 1804 Mutex::Autolock _l(mLock); 1805 pid_t caller = IPCThreadState::self()->getCallingPid(); 1806 ALOGV("acquiring %d from %d", audioSession, caller); 1807 size_t num = mAudioSessionRefs.size(); 1808 for (size_t i = 0; i< num; i++) { 1809 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1810 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1811 ref->mCnt++; 1812 ALOGV(" incremented refcount to %d", ref->mCnt); 1813 return; 1814 } 1815 } 1816 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1817 ALOGV(" added new entry for %d", audioSession); 1818} 1819 1820void AudioFlinger::releaseAudioSessionId(int audioSession) 1821{ 1822 Mutex::Autolock _l(mLock); 1823 pid_t caller = IPCThreadState::self()->getCallingPid(); 1824 ALOGV("releasing %d from %d", audioSession, caller); 1825 size_t num = mAudioSessionRefs.size(); 1826 for (size_t i = 0; i< num; i++) { 1827 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1828 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1829 ref->mCnt--; 1830 ALOGV(" decremented refcount to %d", ref->mCnt); 1831 if (ref->mCnt == 0) { 1832 mAudioSessionRefs.removeAt(i); 1833 delete ref; 1834 purgeStaleEffects_l(); 1835 } 1836 return; 1837 } 1838 } 1839 ALOGW("session id %d not found for pid %d", audioSession, caller); 1840} 1841 1842void AudioFlinger::purgeStaleEffects_l() { 1843 1844 ALOGV("purging stale effects"); 1845 1846 Vector< sp<EffectChain> > chains; 1847 1848 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1849 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1850 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1851 sp<EffectChain> ec = t->mEffectChains[j]; 1852 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1853 chains.push(ec); 1854 } 1855 } 1856 } 1857 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1858 sp<RecordThread> t = mRecordThreads.valueAt(i); 1859 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1860 sp<EffectChain> ec = t->mEffectChains[j]; 1861 chains.push(ec); 1862 } 1863 } 1864 1865 for (size_t i = 0; i < chains.size(); i++) { 1866 sp<EffectChain> ec = chains[i]; 1867 int sessionid = ec->sessionId(); 1868 sp<ThreadBase> t = ec->mThread.promote(); 1869 if (t == 0) { 1870 continue; 1871 } 1872 size_t numsessionrefs = mAudioSessionRefs.size(); 1873 bool found = false; 1874 for (size_t k = 0; k < numsessionrefs; k++) { 1875 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1876 if (ref->mSessionid == sessionid) { 1877 ALOGV(" session %d still exists for %d with %d refs", 1878 sessionid, ref->mPid, ref->mCnt); 1879 found = true; 1880 break; 1881 } 1882 } 1883 if (!found) { 1884 Mutex::Autolock _l (t->mLock); 1885 // remove all effects from the chain 1886 while (ec->mEffects.size()) { 1887 sp<EffectModule> effect = ec->mEffects[0]; 1888 effect->unPin(); 1889 t->removeEffect_l(effect); 1890 if (effect->purgeHandles()) { 1891 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1892 } 1893 AudioSystem::unregisterEffect(effect->id()); 1894 } 1895 } 1896 } 1897 return; 1898} 1899 1900// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1901AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1902{ 1903 return mPlaybackThreads.valueFor(output).get(); 1904} 1905 1906// checkMixerThread_l() must be called with AudioFlinger::mLock held 1907AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1908{ 1909 PlaybackThread *thread = checkPlaybackThread_l(output); 1910 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1911} 1912 1913// checkRecordThread_l() must be called with AudioFlinger::mLock held 1914AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1915{ 1916 return mRecordThreads.valueFor(input).get(); 1917} 1918 1919uint32_t AudioFlinger::nextUniqueId() 1920{ 1921 return android_atomic_inc(&mNextUniqueId); 1922} 1923 1924AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1925{ 1926 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1927 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1928 AudioStreamOut *output = thread->getOutput(); 1929 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1930 return thread; 1931 } 1932 } 1933 return NULL; 1934} 1935 1936audio_devices_t AudioFlinger::primaryOutputDevice_l() const 1937{ 1938 PlaybackThread *thread = primaryPlaybackThread_l(); 1939 1940 if (thread == NULL) { 1941 return 0; 1942 } 1943 1944 return thread->outDevice(); 1945} 1946 1947sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 1948 int triggerSession, 1949 int listenerSession, 1950 sync_event_callback_t callBack, 1951 void *cookie) 1952{ 1953 Mutex::Autolock _l(mLock); 1954 1955 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 1956 status_t playStatus = NAME_NOT_FOUND; 1957 status_t recStatus = NAME_NOT_FOUND; 1958 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1959 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 1960 if (playStatus == NO_ERROR) { 1961 return event; 1962 } 1963 } 1964 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1965 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 1966 if (recStatus == NO_ERROR) { 1967 return event; 1968 } 1969 } 1970 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 1971 mPendingSyncEvents.add(event); 1972 } else { 1973 ALOGV("createSyncEvent() invalid event %d", event->type()); 1974 event.clear(); 1975 } 1976 return event; 1977} 1978 1979// ---------------------------------------------------------------------------- 1980// Effect management 1981// ---------------------------------------------------------------------------- 1982 1983 1984status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 1985{ 1986 Mutex::Autolock _l(mLock); 1987 return EffectQueryNumberEffects(numEffects); 1988} 1989 1990status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 1991{ 1992 Mutex::Autolock _l(mLock); 1993 return EffectQueryEffect(index, descriptor); 1994} 1995 1996status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 1997 effect_descriptor_t *descriptor) const 1998{ 1999 Mutex::Autolock _l(mLock); 2000 return EffectGetDescriptor(pUuid, descriptor); 2001} 2002 2003 2004sp<IEffect> AudioFlinger::createEffect( 2005 effect_descriptor_t *pDesc, 2006 const sp<IEffectClient>& effectClient, 2007 int32_t priority, 2008 audio_io_handle_t io, 2009 int sessionId, 2010 status_t *status, 2011 int *id, 2012 int *enabled) 2013{ 2014 status_t lStatus = NO_ERROR; 2015 sp<EffectHandle> handle; 2016 effect_descriptor_t desc; 2017 2018 pid_t pid = IPCThreadState::self()->getCallingPid(); 2019 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2020 pid, effectClient.get(), priority, sessionId, io); 2021 2022 if (pDesc == NULL) { 2023 lStatus = BAD_VALUE; 2024 goto Exit; 2025 } 2026 2027 // check audio settings permission for global effects 2028 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2029 lStatus = PERMISSION_DENIED; 2030 goto Exit; 2031 } 2032 2033 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2034 // that can only be created by audio policy manager (running in same process) 2035 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2036 lStatus = PERMISSION_DENIED; 2037 goto Exit; 2038 } 2039 2040 if (io == 0) { 2041 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2042 // output must be specified by AudioPolicyManager when using session 2043 // AUDIO_SESSION_OUTPUT_STAGE 2044 lStatus = BAD_VALUE; 2045 goto Exit; 2046 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2047 // if the output returned by getOutputForEffect() is removed before we lock the 2048 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2049 // and we will exit safely 2050 io = AudioSystem::getOutputForEffect(&desc); 2051 } 2052 } 2053 2054 { 2055 Mutex::Autolock _l(mLock); 2056 2057 2058 if (!EffectIsNullUuid(&pDesc->uuid)) { 2059 // if uuid is specified, request effect descriptor 2060 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2061 if (lStatus < 0) { 2062 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2063 goto Exit; 2064 } 2065 } else { 2066 // if uuid is not specified, look for an available implementation 2067 // of the required type in effect factory 2068 if (EffectIsNullUuid(&pDesc->type)) { 2069 ALOGW("createEffect() no effect type"); 2070 lStatus = BAD_VALUE; 2071 goto Exit; 2072 } 2073 uint32_t numEffects = 0; 2074 effect_descriptor_t d; 2075 d.flags = 0; // prevent compiler warning 2076 bool found = false; 2077 2078 lStatus = EffectQueryNumberEffects(&numEffects); 2079 if (lStatus < 0) { 2080 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2081 goto Exit; 2082 } 2083 for (uint32_t i = 0; i < numEffects; i++) { 2084 lStatus = EffectQueryEffect(i, &desc); 2085 if (lStatus < 0) { 2086 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2087 continue; 2088 } 2089 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2090 // If matching type found save effect descriptor. If the session is 2091 // 0 and the effect is not auxiliary, continue enumeration in case 2092 // an auxiliary version of this effect type is available 2093 found = true; 2094 d = desc; 2095 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2096 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2097 break; 2098 } 2099 } 2100 } 2101 if (!found) { 2102 lStatus = BAD_VALUE; 2103 ALOGW("createEffect() effect not found"); 2104 goto Exit; 2105 } 2106 // For same effect type, chose auxiliary version over insert version if 2107 // connect to output mix (Compliance to OpenSL ES) 2108 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2109 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2110 desc = d; 2111 } 2112 } 2113 2114 // Do not allow auxiliary effects on a session different from 0 (output mix) 2115 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2116 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2117 lStatus = INVALID_OPERATION; 2118 goto Exit; 2119 } 2120 2121 // check recording permission for visualizer 2122 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2123 !recordingAllowed()) { 2124 lStatus = PERMISSION_DENIED; 2125 goto Exit; 2126 } 2127 2128 // return effect descriptor 2129 *pDesc = desc; 2130 2131 // If output is not specified try to find a matching audio session ID in one of the 2132 // output threads. 2133 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2134 // because of code checking output when entering the function. 2135 // Note: io is never 0 when creating an effect on an input 2136 if (io == 0) { 2137 // look for the thread where the specified audio session is present 2138 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2139 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2140 io = mPlaybackThreads.keyAt(i); 2141 break; 2142 } 2143 } 2144 if (io == 0) { 2145 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2146 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2147 io = mRecordThreads.keyAt(i); 2148 break; 2149 } 2150 } 2151 } 2152 // If no output thread contains the requested session ID, default to 2153 // first output. The effect chain will be moved to the correct output 2154 // thread when a track with the same session ID is created 2155 if (io == 0 && mPlaybackThreads.size()) { 2156 io = mPlaybackThreads.keyAt(0); 2157 } 2158 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2159 } 2160 ThreadBase *thread = checkRecordThread_l(io); 2161 if (thread == NULL) { 2162 thread = checkPlaybackThread_l(io); 2163 if (thread == NULL) { 2164 ALOGE("createEffect() unknown output thread"); 2165 lStatus = BAD_VALUE; 2166 goto Exit; 2167 } 2168 } 2169 2170 sp<Client> client = registerPid_l(pid); 2171 2172 // create effect on selected output thread 2173 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2174 &desc, enabled, &lStatus); 2175 if (handle != 0 && id != NULL) { 2176 *id = handle->id(); 2177 } 2178 } 2179 2180Exit: 2181 if (status != NULL) { 2182 *status = lStatus; 2183 } 2184 return handle; 2185} 2186 2187status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2188 audio_io_handle_t dstOutput) 2189{ 2190 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2191 sessionId, srcOutput, dstOutput); 2192 Mutex::Autolock _l(mLock); 2193 if (srcOutput == dstOutput) { 2194 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2195 return NO_ERROR; 2196 } 2197 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2198 if (srcThread == NULL) { 2199 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2200 return BAD_VALUE; 2201 } 2202 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2203 if (dstThread == NULL) { 2204 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2205 return BAD_VALUE; 2206 } 2207 2208 Mutex::Autolock _dl(dstThread->mLock); 2209 Mutex::Autolock _sl(srcThread->mLock); 2210 moveEffectChain_l(sessionId, srcThread, dstThread, false); 2211 2212 return NO_ERROR; 2213} 2214 2215// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2216status_t AudioFlinger::moveEffectChain_l(int sessionId, 2217 AudioFlinger::PlaybackThread *srcThread, 2218 AudioFlinger::PlaybackThread *dstThread, 2219 bool reRegister) 2220{ 2221 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2222 sessionId, srcThread, dstThread); 2223 2224 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2225 if (chain == 0) { 2226 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2227 sessionId, srcThread); 2228 return INVALID_OPERATION; 2229 } 2230 2231 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2232 // so that a new chain is created with correct parameters when first effect is added. This is 2233 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2234 // removed. 2235 srcThread->removeEffectChain_l(chain); 2236 2237 // transfer all effects one by one so that new effect chain is created on new thread with 2238 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2239 audio_io_handle_t dstOutput = dstThread->id(); 2240 sp<EffectChain> dstChain; 2241 uint32_t strategy = 0; // prevent compiler warning 2242 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2243 while (effect != 0) { 2244 srcThread->removeEffect_l(effect); 2245 dstThread->addEffect_l(effect); 2246 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2247 if (effect->state() == EffectModule::ACTIVE || 2248 effect->state() == EffectModule::STOPPING) { 2249 effect->start(); 2250 } 2251 // if the move request is not received from audio policy manager, the effect must be 2252 // re-registered with the new strategy and output 2253 if (dstChain == 0) { 2254 dstChain = effect->chain().promote(); 2255 if (dstChain == 0) { 2256 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2257 srcThread->addEffect_l(effect); 2258 return NO_INIT; 2259 } 2260 strategy = dstChain->strategy(); 2261 } 2262 if (reRegister) { 2263 AudioSystem::unregisterEffect(effect->id()); 2264 AudioSystem::registerEffect(&effect->desc(), 2265 dstOutput, 2266 strategy, 2267 sessionId, 2268 effect->id()); 2269 } 2270 effect = chain->getEffectFromId_l(0); 2271 } 2272 2273 return NO_ERROR; 2274} 2275 2276struct Entry { 2277#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2278 char mName[MAX_NAME]; 2279}; 2280 2281int comparEntry(const void *p1, const void *p2) 2282{ 2283 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2284} 2285 2286#ifdef TEE_SINK 2287void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2288{ 2289 NBAIO_Source *teeSource = source.get(); 2290 if (teeSource != NULL) { 2291 // .wav rotation 2292 // There is a benign race condition if 2 threads call this simultaneously. 2293 // They would both traverse the directory, but the result would simply be 2294 // failures at unlink() which are ignored. It's also unlikely since 2295 // normally dumpsys is only done by bugreport or from the command line. 2296 char teePath[32+256]; 2297 strcpy(teePath, "/data/misc/media"); 2298 size_t teePathLen = strlen(teePath); 2299 DIR *dir = opendir(teePath); 2300 teePath[teePathLen++] = '/'; 2301 if (dir != NULL) { 2302#define MAX_SORT 20 // number of entries to sort 2303#define MAX_KEEP 10 // number of entries to keep 2304 struct Entry entries[MAX_SORT]; 2305 size_t entryCount = 0; 2306 while (entryCount < MAX_SORT) { 2307 struct dirent de; 2308 struct dirent *result = NULL; 2309 int rc = readdir_r(dir, &de, &result); 2310 if (rc != 0) { 2311 ALOGW("readdir_r failed %d", rc); 2312 break; 2313 } 2314 if (result == NULL) { 2315 break; 2316 } 2317 if (result != &de) { 2318 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2319 break; 2320 } 2321 // ignore non .wav file entries 2322 size_t nameLen = strlen(de.d_name); 2323 if (nameLen <= 4 || nameLen >= MAX_NAME || 2324 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2325 continue; 2326 } 2327 strcpy(entries[entryCount++].mName, de.d_name); 2328 } 2329 (void) closedir(dir); 2330 if (entryCount > MAX_KEEP) { 2331 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2332 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2333 strcpy(&teePath[teePathLen], entries[i].mName); 2334 (void) unlink(teePath); 2335 } 2336 } 2337 } else { 2338 if (fd >= 0) { 2339 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2340 } 2341 } 2342 char teeTime[16]; 2343 struct timeval tv; 2344 gettimeofday(&tv, NULL); 2345 struct tm tm; 2346 localtime_r(&tv.tv_sec, &tm); 2347 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2348 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2349 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2350 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2351 if (teeFd >= 0) { 2352 char wavHeader[44]; 2353 memcpy(wavHeader, 2354 "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", 2355 sizeof(wavHeader)); 2356 NBAIO_Format format = teeSource->format(); 2357 unsigned channelCount = Format_channelCount(format); 2358 ALOG_ASSERT(channelCount <= FCC_2); 2359 uint32_t sampleRate = Format_sampleRate(format); 2360 wavHeader[22] = channelCount; // number of channels 2361 wavHeader[24] = sampleRate; // sample rate 2362 wavHeader[25] = sampleRate >> 8; 2363 wavHeader[32] = channelCount * 2; // block alignment 2364 write(teeFd, wavHeader, sizeof(wavHeader)); 2365 size_t total = 0; 2366 bool firstRead = true; 2367 for (;;) { 2368#define TEE_SINK_READ 1024 2369 short buffer[TEE_SINK_READ * FCC_2]; 2370 size_t count = TEE_SINK_READ; 2371 ssize_t actual = teeSource->read(buffer, count, 2372 AudioBufferProvider::kInvalidPTS); 2373 bool wasFirstRead = firstRead; 2374 firstRead = false; 2375 if (actual <= 0) { 2376 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2377 continue; 2378 } 2379 break; 2380 } 2381 ALOG_ASSERT(actual <= (ssize_t)count); 2382 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2383 total += actual; 2384 } 2385 lseek(teeFd, (off_t) 4, SEEK_SET); 2386 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2387 write(teeFd, &temp, sizeof(temp)); 2388 lseek(teeFd, (off_t) 40, SEEK_SET); 2389 temp = total * channelCount * sizeof(short); 2390 write(teeFd, &temp, sizeof(temp)); 2391 close(teeFd); 2392 if (fd >= 0) { 2393 fdprintf(fd, "tee copied to %s\n", teePath); 2394 } 2395 } else { 2396 if (fd >= 0) { 2397 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2398 } 2399 } 2400 } 2401} 2402#endif 2403 2404// ---------------------------------------------------------------------------- 2405 2406status_t AudioFlinger::onTransact( 2407 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2408{ 2409 return BnAudioFlinger::onTransact(code, data, reply, flags); 2410} 2411 2412}; // namespace android 2413