AudioFlinger.cpp revision bfb1b832079bbb9426f72f3863199a54aefd02da
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 status_t status; 1414 PlaybackThread *thread = NULL; 1415 struct audio_config config; 1416 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1417 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1418 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1419 if (offloadInfo) { 1420 config.offload_info = *offloadInfo; 1421 } 1422 1423 audio_stream_out_t *outStream = NULL; 1424 AudioHwDevice *outHwDev; 1425 1426 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1427 module, 1428 (pDevices != NULL) ? *pDevices : 0, 1429 config.sample_rate, 1430 config.format, 1431 config.channel_mask, 1432 flags); 1433 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1434 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); 1435 1436 if (pDevices == NULL || *pDevices == 0) { 1437 return 0; 1438 } 1439 1440 Mutex::Autolock _l(mLock); 1441 1442 outHwDev = findSuitableHwDev_l(module, *pDevices); 1443 if (outHwDev == NULL) 1444 return 0; 1445 1446 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1447 audio_io_handle_t id = nextUniqueId(); 1448 1449 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1450 1451 status = hwDevHal->open_output_stream(hwDevHal, 1452 id, 1453 *pDevices, 1454 (audio_output_flags_t)flags, 1455 &config, 1456 &outStream); 1457 1458 mHardwareStatus = AUDIO_HW_IDLE; 1459 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1460 "Channels %x, status %d", 1461 outStream, 1462 config.sample_rate, 1463 config.format, 1464 config.channel_mask, 1465 status); 1466 1467 if (status == NO_ERROR && outStream != NULL) { 1468 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1469 1470 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1471 thread = new OffloadThread(this, output, id, *pDevices); 1472 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1473 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1474 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1475 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1476 thread = new DirectOutputThread(this, output, id, *pDevices); 1477 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1478 } else { 1479 thread = new MixerThread(this, output, id, *pDevices); 1480 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1481 } 1482 mPlaybackThreads.add(id, thread); 1483 1484 if (pSamplingRate != NULL) { 1485 *pSamplingRate = config.sample_rate; 1486 } 1487 if (pFormat != NULL) { 1488 *pFormat = config.format; 1489 } 1490 if (pChannelMask != NULL) { 1491 *pChannelMask = config.channel_mask; 1492 } 1493 if (pLatencyMs != NULL) { 1494 *pLatencyMs = thread->latency(); 1495 } 1496 1497 // notify client processes of the new output creation 1498 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1499 1500 // the first primary output opened designates the primary hw device 1501 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1502 ALOGI("Using module %d has the primary audio interface", module); 1503 mPrimaryHardwareDev = outHwDev; 1504 1505 AutoMutex lock(mHardwareLock); 1506 mHardwareStatus = AUDIO_HW_SET_MODE; 1507 hwDevHal->set_mode(hwDevHal, mMode); 1508 mHardwareStatus = AUDIO_HW_IDLE; 1509 } 1510 return id; 1511 } 1512 1513 return 0; 1514} 1515 1516audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1517 audio_io_handle_t output2) 1518{ 1519 Mutex::Autolock _l(mLock); 1520 MixerThread *thread1 = checkMixerThread_l(output1); 1521 MixerThread *thread2 = checkMixerThread_l(output2); 1522 1523 if (thread1 == NULL || thread2 == NULL) { 1524 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1525 output2); 1526 return 0; 1527 } 1528 1529 audio_io_handle_t id = nextUniqueId(); 1530 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1531 thread->addOutputTrack(thread2); 1532 mPlaybackThreads.add(id, thread); 1533 // notify client processes of the new output creation 1534 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1535 return id; 1536} 1537 1538status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1539{ 1540 return closeOutput_nonvirtual(output); 1541} 1542 1543status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1544{ 1545 // keep strong reference on the playback thread so that 1546 // it is not destroyed while exit() is executed 1547 sp<PlaybackThread> thread; 1548 { 1549 Mutex::Autolock _l(mLock); 1550 thread = checkPlaybackThread_l(output); 1551 if (thread == NULL) { 1552 return BAD_VALUE; 1553 } 1554 1555 ALOGV("closeOutput() %d", output); 1556 1557 if (thread->type() == ThreadBase::MIXER) { 1558 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1559 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1560 DuplicatingThread *dupThread = 1561 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1562 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1563 1564 } 1565 } 1566 } 1567 1568 1569 mPlaybackThreads.removeItem(output); 1570 // save all effects to the default thread 1571 if (mPlaybackThreads.size()) { 1572 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1573 if (dstThread != NULL) { 1574 // audioflinger lock is held here so the acquisition order of thread locks does not 1575 // matter 1576 Mutex::Autolock _dl(dstThread->mLock); 1577 Mutex::Autolock _sl(thread->mLock); 1578 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1579 for (size_t i = 0; i < effectChains.size(); i ++) { 1580 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1581 } 1582 } 1583 } 1584 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1585 } 1586 thread->exit(); 1587 // The thread entity (active unit of execution) is no longer running here, 1588 // but the ThreadBase container still exists. 1589 1590 if (thread->type() != ThreadBase::DUPLICATING) { 1591 AudioStreamOut *out = thread->clearOutput(); 1592 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1593 // from now on thread->mOutput is NULL 1594 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1595 delete out; 1596 } 1597 return NO_ERROR; 1598} 1599 1600status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1601{ 1602 Mutex::Autolock _l(mLock); 1603 PlaybackThread *thread = checkPlaybackThread_l(output); 1604 1605 if (thread == NULL) { 1606 return BAD_VALUE; 1607 } 1608 1609 ALOGV("suspendOutput() %d", output); 1610 thread->suspend(); 1611 1612 return NO_ERROR; 1613} 1614 1615status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1616{ 1617 Mutex::Autolock _l(mLock); 1618 PlaybackThread *thread = checkPlaybackThread_l(output); 1619 1620 if (thread == NULL) { 1621 return BAD_VALUE; 1622 } 1623 1624 ALOGV("restoreOutput() %d", output); 1625 1626 thread->restore(); 1627 1628 return NO_ERROR; 1629} 1630 1631audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1632 audio_devices_t *pDevices, 1633 uint32_t *pSamplingRate, 1634 audio_format_t *pFormat, 1635 audio_channel_mask_t *pChannelMask) 1636{ 1637 status_t status; 1638 RecordThread *thread = NULL; 1639 struct audio_config config; 1640 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1641 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1642 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1643 1644 uint32_t reqSamplingRate = config.sample_rate; 1645 audio_format_t reqFormat = config.format; 1646 audio_channel_mask_t reqChannels = config.channel_mask; 1647 audio_stream_in_t *inStream = NULL; 1648 AudioHwDevice *inHwDev; 1649 1650 if (pDevices == NULL || *pDevices == 0) { 1651 return 0; 1652 } 1653 1654 Mutex::Autolock _l(mLock); 1655 1656 inHwDev = findSuitableHwDev_l(module, *pDevices); 1657 if (inHwDev == NULL) 1658 return 0; 1659 1660 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1661 audio_io_handle_t id = nextUniqueId(); 1662 1663 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1664 &inStream); 1665 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1666 "status %d", 1667 inStream, 1668 config.sample_rate, 1669 config.format, 1670 config.channel_mask, 1671 status); 1672 1673 // If the input could not be opened with the requested parameters and we can handle the 1674 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1675 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1676 if (status == BAD_VALUE && 1677 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1678 (config.sample_rate <= 2 * reqSamplingRate) && 1679 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 1680 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1681 inStream = NULL; 1682 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1683 } 1684 1685 if (status == NO_ERROR && inStream != NULL) { 1686 1687#ifdef TEE_SINK 1688 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1689 // or (re-)create if current Pipe is idle and does not match the new format 1690 sp<NBAIO_Sink> teeSink; 1691 enum { 1692 TEE_SINK_NO, // don't copy input 1693 TEE_SINK_NEW, // copy input using a new pipe 1694 TEE_SINK_OLD, // copy input using an existing pipe 1695 } kind; 1696 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1697 popcount(inStream->common.get_channels(&inStream->common))); 1698 if (!mTeeSinkInputEnabled) { 1699 kind = TEE_SINK_NO; 1700 } else if (format == Format_Invalid) { 1701 kind = TEE_SINK_NO; 1702 } else if (mRecordTeeSink == 0) { 1703 kind = TEE_SINK_NEW; 1704 } else if (mRecordTeeSink->getStrongCount() != 1) { 1705 kind = TEE_SINK_NO; 1706 } else if (format == mRecordTeeSink->format()) { 1707 kind = TEE_SINK_OLD; 1708 } else { 1709 kind = TEE_SINK_NEW; 1710 } 1711 switch (kind) { 1712 case TEE_SINK_NEW: { 1713 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1714 size_t numCounterOffers = 0; 1715 const NBAIO_Format offers[1] = {format}; 1716 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1717 ALOG_ASSERT(index == 0); 1718 PipeReader *pipeReader = new PipeReader(*pipe); 1719 numCounterOffers = 0; 1720 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1721 ALOG_ASSERT(index == 0); 1722 mRecordTeeSink = pipe; 1723 mRecordTeeSource = pipeReader; 1724 teeSink = pipe; 1725 } 1726 break; 1727 case TEE_SINK_OLD: 1728 teeSink = mRecordTeeSink; 1729 break; 1730 case TEE_SINK_NO: 1731 default: 1732 break; 1733 } 1734#endif 1735 1736 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1737 1738 // Start record thread 1739 // RecorThread require both input and output device indication to forward to audio 1740 // pre processing modules 1741 thread = new RecordThread(this, 1742 input, 1743 reqSamplingRate, 1744 reqChannels, 1745 id, 1746 primaryOutputDevice_l(), 1747 *pDevices 1748#ifdef TEE_SINK 1749 , teeSink 1750#endif 1751 ); 1752 mRecordThreads.add(id, thread); 1753 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1754 if (pSamplingRate != NULL) { 1755 *pSamplingRate = reqSamplingRate; 1756 } 1757 if (pFormat != NULL) { 1758 *pFormat = config.format; 1759 } 1760 if (pChannelMask != NULL) { 1761 *pChannelMask = reqChannels; 1762 } 1763 1764 // notify client processes of the new input creation 1765 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1766 return id; 1767 } 1768 1769 return 0; 1770} 1771 1772status_t AudioFlinger::closeInput(audio_io_handle_t input) 1773{ 1774 return closeInput_nonvirtual(input); 1775} 1776 1777status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1778{ 1779 // keep strong reference on the record thread so that 1780 // it is not destroyed while exit() is executed 1781 sp<RecordThread> thread; 1782 { 1783 Mutex::Autolock _l(mLock); 1784 thread = checkRecordThread_l(input); 1785 if (thread == 0) { 1786 return BAD_VALUE; 1787 } 1788 1789 ALOGV("closeInput() %d", input); 1790 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1791 mRecordThreads.removeItem(input); 1792 } 1793 thread->exit(); 1794 // The thread entity (active unit of execution) is no longer running here, 1795 // but the ThreadBase container still exists. 1796 1797 AudioStreamIn *in = thread->clearInput(); 1798 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1799 // from now on thread->mInput is NULL 1800 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1801 delete in; 1802 1803 return NO_ERROR; 1804} 1805 1806status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1807{ 1808 Mutex::Autolock _l(mLock); 1809 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1810 1811 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1812 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1813 thread->invalidateTracks(stream); 1814 } 1815 1816 return NO_ERROR; 1817} 1818 1819 1820int AudioFlinger::newAudioSessionId() 1821{ 1822 return nextUniqueId(); 1823} 1824 1825void AudioFlinger::acquireAudioSessionId(int audioSession) 1826{ 1827 Mutex::Autolock _l(mLock); 1828 pid_t caller = IPCThreadState::self()->getCallingPid(); 1829 ALOGV("acquiring %d from %d", audioSession, caller); 1830 size_t num = mAudioSessionRefs.size(); 1831 for (size_t i = 0; i< num; i++) { 1832 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1833 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1834 ref->mCnt++; 1835 ALOGV(" incremented refcount to %d", ref->mCnt); 1836 return; 1837 } 1838 } 1839 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1840 ALOGV(" added new entry for %d", audioSession); 1841} 1842 1843void AudioFlinger::releaseAudioSessionId(int audioSession) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 pid_t caller = IPCThreadState::self()->getCallingPid(); 1847 ALOGV("releasing %d from %d", audioSession, caller); 1848 size_t num = mAudioSessionRefs.size(); 1849 for (size_t i = 0; i< num; i++) { 1850 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1851 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1852 ref->mCnt--; 1853 ALOGV(" decremented refcount to %d", ref->mCnt); 1854 if (ref->mCnt == 0) { 1855 mAudioSessionRefs.removeAt(i); 1856 delete ref; 1857 purgeStaleEffects_l(); 1858 } 1859 return; 1860 } 1861 } 1862 ALOGW("session id %d not found for pid %d", audioSession, caller); 1863} 1864 1865void AudioFlinger::purgeStaleEffects_l() { 1866 1867 ALOGV("purging stale effects"); 1868 1869 Vector< sp<EffectChain> > chains; 1870 1871 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1872 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1873 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1874 sp<EffectChain> ec = t->mEffectChains[j]; 1875 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1876 chains.push(ec); 1877 } 1878 } 1879 } 1880 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1881 sp<RecordThread> t = mRecordThreads.valueAt(i); 1882 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1883 sp<EffectChain> ec = t->mEffectChains[j]; 1884 chains.push(ec); 1885 } 1886 } 1887 1888 for (size_t i = 0; i < chains.size(); i++) { 1889 sp<EffectChain> ec = chains[i]; 1890 int sessionid = ec->sessionId(); 1891 sp<ThreadBase> t = ec->mThread.promote(); 1892 if (t == 0) { 1893 continue; 1894 } 1895 size_t numsessionrefs = mAudioSessionRefs.size(); 1896 bool found = false; 1897 for (size_t k = 0; k < numsessionrefs; k++) { 1898 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1899 if (ref->mSessionid == sessionid) { 1900 ALOGV(" session %d still exists for %d with %d refs", 1901 sessionid, ref->mPid, ref->mCnt); 1902 found = true; 1903 break; 1904 } 1905 } 1906 if (!found) { 1907 Mutex::Autolock _l (t->mLock); 1908 // remove all effects from the chain 1909 while (ec->mEffects.size()) { 1910 sp<EffectModule> effect = ec->mEffects[0]; 1911 effect->unPin(); 1912 t->removeEffect_l(effect); 1913 if (effect->purgeHandles()) { 1914 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1915 } 1916 AudioSystem::unregisterEffect(effect->id()); 1917 } 1918 } 1919 } 1920 return; 1921} 1922 1923// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1924AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1925{ 1926 return mPlaybackThreads.valueFor(output).get(); 1927} 1928 1929// checkMixerThread_l() must be called with AudioFlinger::mLock held 1930AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1931{ 1932 PlaybackThread *thread = checkPlaybackThread_l(output); 1933 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1934} 1935 1936// checkRecordThread_l() must be called with AudioFlinger::mLock held 1937AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1938{ 1939 return mRecordThreads.valueFor(input).get(); 1940} 1941 1942uint32_t AudioFlinger::nextUniqueId() 1943{ 1944 return android_atomic_inc(&mNextUniqueId); 1945} 1946 1947AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1948{ 1949 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1950 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1951 AudioStreamOut *output = thread->getOutput(); 1952 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1953 return thread; 1954 } 1955 } 1956 return NULL; 1957} 1958 1959audio_devices_t AudioFlinger::primaryOutputDevice_l() const 1960{ 1961 PlaybackThread *thread = primaryPlaybackThread_l(); 1962 1963 if (thread == NULL) { 1964 return 0; 1965 } 1966 1967 return thread->outDevice(); 1968} 1969 1970sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 1971 int triggerSession, 1972 int listenerSession, 1973 sync_event_callback_t callBack, 1974 void *cookie) 1975{ 1976 Mutex::Autolock _l(mLock); 1977 1978 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 1979 status_t playStatus = NAME_NOT_FOUND; 1980 status_t recStatus = NAME_NOT_FOUND; 1981 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1982 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 1983 if (playStatus == NO_ERROR) { 1984 return event; 1985 } 1986 } 1987 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1988 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 1989 if (recStatus == NO_ERROR) { 1990 return event; 1991 } 1992 } 1993 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 1994 mPendingSyncEvents.add(event); 1995 } else { 1996 ALOGV("createSyncEvent() invalid event %d", event->type()); 1997 event.clear(); 1998 } 1999 return event; 2000} 2001 2002// ---------------------------------------------------------------------------- 2003// Effect management 2004// ---------------------------------------------------------------------------- 2005 2006 2007status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2008{ 2009 Mutex::Autolock _l(mLock); 2010 return EffectQueryNumberEffects(numEffects); 2011} 2012 2013status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2014{ 2015 Mutex::Autolock _l(mLock); 2016 return EffectQueryEffect(index, descriptor); 2017} 2018 2019status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2020 effect_descriptor_t *descriptor) const 2021{ 2022 Mutex::Autolock _l(mLock); 2023 return EffectGetDescriptor(pUuid, descriptor); 2024} 2025 2026 2027sp<IEffect> AudioFlinger::createEffect( 2028 effect_descriptor_t *pDesc, 2029 const sp<IEffectClient>& effectClient, 2030 int32_t priority, 2031 audio_io_handle_t io, 2032 int sessionId, 2033 status_t *status, 2034 int *id, 2035 int *enabled) 2036{ 2037 status_t lStatus = NO_ERROR; 2038 sp<EffectHandle> handle; 2039 effect_descriptor_t desc; 2040 2041 pid_t pid = IPCThreadState::self()->getCallingPid(); 2042 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2043 pid, effectClient.get(), priority, sessionId, io); 2044 2045 if (pDesc == NULL) { 2046 lStatus = BAD_VALUE; 2047 goto Exit; 2048 } 2049 2050 // check audio settings permission for global effects 2051 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2052 lStatus = PERMISSION_DENIED; 2053 goto Exit; 2054 } 2055 2056 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2057 // that can only be created by audio policy manager (running in same process) 2058 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2059 lStatus = PERMISSION_DENIED; 2060 goto Exit; 2061 } 2062 2063 if (io == 0) { 2064 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2065 // output must be specified by AudioPolicyManager when using session 2066 // AUDIO_SESSION_OUTPUT_STAGE 2067 lStatus = BAD_VALUE; 2068 goto Exit; 2069 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2070 // if the output returned by getOutputForEffect() is removed before we lock the 2071 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2072 // and we will exit safely 2073 io = AudioSystem::getOutputForEffect(&desc); 2074 } 2075 } 2076 2077 { 2078 Mutex::Autolock _l(mLock); 2079 2080 2081 if (!EffectIsNullUuid(&pDesc->uuid)) { 2082 // if uuid is specified, request effect descriptor 2083 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2084 if (lStatus < 0) { 2085 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2086 goto Exit; 2087 } 2088 } else { 2089 // if uuid is not specified, look for an available implementation 2090 // of the required type in effect factory 2091 if (EffectIsNullUuid(&pDesc->type)) { 2092 ALOGW("createEffect() no effect type"); 2093 lStatus = BAD_VALUE; 2094 goto Exit; 2095 } 2096 uint32_t numEffects = 0; 2097 effect_descriptor_t d; 2098 d.flags = 0; // prevent compiler warning 2099 bool found = false; 2100 2101 lStatus = EffectQueryNumberEffects(&numEffects); 2102 if (lStatus < 0) { 2103 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2104 goto Exit; 2105 } 2106 for (uint32_t i = 0; i < numEffects; i++) { 2107 lStatus = EffectQueryEffect(i, &desc); 2108 if (lStatus < 0) { 2109 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2110 continue; 2111 } 2112 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2113 // If matching type found save effect descriptor. If the session is 2114 // 0 and the effect is not auxiliary, continue enumeration in case 2115 // an auxiliary version of this effect type is available 2116 found = true; 2117 d = desc; 2118 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2119 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2120 break; 2121 } 2122 } 2123 } 2124 if (!found) { 2125 lStatus = BAD_VALUE; 2126 ALOGW("createEffect() effect not found"); 2127 goto Exit; 2128 } 2129 // For same effect type, chose auxiliary version over insert version if 2130 // connect to output mix (Compliance to OpenSL ES) 2131 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2132 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2133 desc = d; 2134 } 2135 } 2136 2137 // Do not allow auxiliary effects on a session different from 0 (output mix) 2138 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2139 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2140 lStatus = INVALID_OPERATION; 2141 goto Exit; 2142 } 2143 2144 // check recording permission for visualizer 2145 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2146 !recordingAllowed()) { 2147 lStatus = PERMISSION_DENIED; 2148 goto Exit; 2149 } 2150 2151 // return effect descriptor 2152 *pDesc = desc; 2153 2154 // If output is not specified try to find a matching audio session ID in one of the 2155 // output threads. 2156 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2157 // because of code checking output when entering the function. 2158 // Note: io is never 0 when creating an effect on an input 2159 if (io == 0) { 2160 // look for the thread where the specified audio session is present 2161 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2162 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2163 io = mPlaybackThreads.keyAt(i); 2164 break; 2165 } 2166 } 2167 if (io == 0) { 2168 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2169 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2170 io = mRecordThreads.keyAt(i); 2171 break; 2172 } 2173 } 2174 } 2175 // If no output thread contains the requested session ID, default to 2176 // first output. The effect chain will be moved to the correct output 2177 // thread when a track with the same session ID is created 2178 if (io == 0 && mPlaybackThreads.size()) { 2179 io = mPlaybackThreads.keyAt(0); 2180 } 2181 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2182 } 2183 ThreadBase *thread = checkRecordThread_l(io); 2184 if (thread == NULL) { 2185 thread = checkPlaybackThread_l(io); 2186 if (thread == NULL) { 2187 ALOGE("createEffect() unknown output thread"); 2188 lStatus = BAD_VALUE; 2189 goto Exit; 2190 } 2191 } 2192 2193 sp<Client> client = registerPid_l(pid); 2194 2195 // create effect on selected output thread 2196 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2197 &desc, enabled, &lStatus); 2198 if (handle != 0 && id != NULL) { 2199 *id = handle->id(); 2200 } 2201 } 2202 2203Exit: 2204 if (status != NULL) { 2205 *status = lStatus; 2206 } 2207 return handle; 2208} 2209 2210status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2211 audio_io_handle_t dstOutput) 2212{ 2213 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2214 sessionId, srcOutput, dstOutput); 2215 Mutex::Autolock _l(mLock); 2216 if (srcOutput == dstOutput) { 2217 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2218 return NO_ERROR; 2219 } 2220 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2221 if (srcThread == NULL) { 2222 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2223 return BAD_VALUE; 2224 } 2225 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2226 if (dstThread == NULL) { 2227 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2228 return BAD_VALUE; 2229 } 2230 2231 Mutex::Autolock _dl(dstThread->mLock); 2232 Mutex::Autolock _sl(srcThread->mLock); 2233 moveEffectChain_l(sessionId, srcThread, dstThread, false); 2234 2235 return NO_ERROR; 2236} 2237 2238// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2239status_t AudioFlinger::moveEffectChain_l(int sessionId, 2240 AudioFlinger::PlaybackThread *srcThread, 2241 AudioFlinger::PlaybackThread *dstThread, 2242 bool reRegister) 2243{ 2244 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2245 sessionId, srcThread, dstThread); 2246 2247 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2248 if (chain == 0) { 2249 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2250 sessionId, srcThread); 2251 return INVALID_OPERATION; 2252 } 2253 2254 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2255 // so that a new chain is created with correct parameters when first effect is added. This is 2256 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2257 // removed. 2258 srcThread->removeEffectChain_l(chain); 2259 2260 // transfer all effects one by one so that new effect chain is created on new thread with 2261 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2262 audio_io_handle_t dstOutput = dstThread->id(); 2263 sp<EffectChain> dstChain; 2264 uint32_t strategy = 0; // prevent compiler warning 2265 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2266 while (effect != 0) { 2267 srcThread->removeEffect_l(effect); 2268 dstThread->addEffect_l(effect); 2269 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2270 if (effect->state() == EffectModule::ACTIVE || 2271 effect->state() == EffectModule::STOPPING) { 2272 effect->start(); 2273 } 2274 // if the move request is not received from audio policy manager, the effect must be 2275 // re-registered with the new strategy and output 2276 if (dstChain == 0) { 2277 dstChain = effect->chain().promote(); 2278 if (dstChain == 0) { 2279 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2280 srcThread->addEffect_l(effect); 2281 return NO_INIT; 2282 } 2283 strategy = dstChain->strategy(); 2284 } 2285 if (reRegister) { 2286 AudioSystem::unregisterEffect(effect->id()); 2287 AudioSystem::registerEffect(&effect->desc(), 2288 dstOutput, 2289 strategy, 2290 sessionId, 2291 effect->id()); 2292 } 2293 effect = chain->getEffectFromId_l(0); 2294 } 2295 2296 return NO_ERROR; 2297} 2298 2299struct Entry { 2300#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2301 char mName[MAX_NAME]; 2302}; 2303 2304int comparEntry(const void *p1, const void *p2) 2305{ 2306 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2307} 2308 2309#ifdef TEE_SINK 2310void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2311{ 2312 NBAIO_Source *teeSource = source.get(); 2313 if (teeSource != NULL) { 2314 // .wav rotation 2315 // There is a benign race condition if 2 threads call this simultaneously. 2316 // They would both traverse the directory, but the result would simply be 2317 // failures at unlink() which are ignored. It's also unlikely since 2318 // normally dumpsys is only done by bugreport or from the command line. 2319 char teePath[32+256]; 2320 strcpy(teePath, "/data/misc/media"); 2321 size_t teePathLen = strlen(teePath); 2322 DIR *dir = opendir(teePath); 2323 teePath[teePathLen++] = '/'; 2324 if (dir != NULL) { 2325#define MAX_SORT 20 // number of entries to sort 2326#define MAX_KEEP 10 // number of entries to keep 2327 struct Entry entries[MAX_SORT]; 2328 size_t entryCount = 0; 2329 while (entryCount < MAX_SORT) { 2330 struct dirent de; 2331 struct dirent *result = NULL; 2332 int rc = readdir_r(dir, &de, &result); 2333 if (rc != 0) { 2334 ALOGW("readdir_r failed %d", rc); 2335 break; 2336 } 2337 if (result == NULL) { 2338 break; 2339 } 2340 if (result != &de) { 2341 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2342 break; 2343 } 2344 // ignore non .wav file entries 2345 size_t nameLen = strlen(de.d_name); 2346 if (nameLen <= 4 || nameLen >= MAX_NAME || 2347 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2348 continue; 2349 } 2350 strcpy(entries[entryCount++].mName, de.d_name); 2351 } 2352 (void) closedir(dir); 2353 if (entryCount > MAX_KEEP) { 2354 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2355 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2356 strcpy(&teePath[teePathLen], entries[i].mName); 2357 (void) unlink(teePath); 2358 } 2359 } 2360 } else { 2361 if (fd >= 0) { 2362 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2363 } 2364 } 2365 char teeTime[16]; 2366 struct timeval tv; 2367 gettimeofday(&tv, NULL); 2368 struct tm tm; 2369 localtime_r(&tv.tv_sec, &tm); 2370 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2371 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2372 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2373 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2374 if (teeFd >= 0) { 2375 char wavHeader[44]; 2376 memcpy(wavHeader, 2377 "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", 2378 sizeof(wavHeader)); 2379 NBAIO_Format format = teeSource->format(); 2380 unsigned channelCount = Format_channelCount(format); 2381 ALOG_ASSERT(channelCount <= FCC_2); 2382 uint32_t sampleRate = Format_sampleRate(format); 2383 wavHeader[22] = channelCount; // number of channels 2384 wavHeader[24] = sampleRate; // sample rate 2385 wavHeader[25] = sampleRate >> 8; 2386 wavHeader[32] = channelCount * 2; // block alignment 2387 write(teeFd, wavHeader, sizeof(wavHeader)); 2388 size_t total = 0; 2389 bool firstRead = true; 2390 for (;;) { 2391#define TEE_SINK_READ 1024 2392 short buffer[TEE_SINK_READ * FCC_2]; 2393 size_t count = TEE_SINK_READ; 2394 ssize_t actual = teeSource->read(buffer, count, 2395 AudioBufferProvider::kInvalidPTS); 2396 bool wasFirstRead = firstRead; 2397 firstRead = false; 2398 if (actual <= 0) { 2399 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2400 continue; 2401 } 2402 break; 2403 } 2404 ALOG_ASSERT(actual <= (ssize_t)count); 2405 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2406 total += actual; 2407 } 2408 lseek(teeFd, (off_t) 4, SEEK_SET); 2409 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2410 write(teeFd, &temp, sizeof(temp)); 2411 lseek(teeFd, (off_t) 40, SEEK_SET); 2412 temp = total * channelCount * sizeof(short); 2413 write(teeFd, &temp, sizeof(temp)); 2414 close(teeFd); 2415 if (fd >= 0) { 2416 fdprintf(fd, "tee copied to %s\n", teePath); 2417 } 2418 } else { 2419 if (fd >= 0) { 2420 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2421 } 2422 } 2423 } 2424} 2425#endif 2426 2427// ---------------------------------------------------------------------------- 2428 2429status_t AudioFlinger::onTransact( 2430 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2431{ 2432 return BnAudioFlinger::onTransact(code, data, reply, flags); 2433} 2434 2435}; // namespace android 2436