AudioFlinger.cpp revision 813e2a74853bde19e37d878c596a044b3f299efc
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/EffectsFactoryApi.h> 49#include <audio_effects/effect_visualizer.h> 50#include <audio_effects/effect_ns.h> 51#include <audio_effects/effect_aec.h> 52 53#include <audio_utils/primitives.h> 54 55#include <powermanager/PowerManager.h> 56 57#include <common_time/cc_helper.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <private/android_filesystem_config.h> 65 66// ---------------------------------------------------------------------------- 67 68// Note: the following macro is used for extremely verbose logging message. In 69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70// 0; but one side effect of this is to turn all LOGV's as well. Some messages 71// are so verbose that we want to suppress them even when we have ALOG_ASSERT 72// turned on. Do not uncomment the #def below unless you really know what you 73// are doing and want to see all of the extremely verbose messages. 74//#define VERY_VERY_VERBOSE_LOGGING 75#ifdef VERY_VERY_VERBOSE_LOGGING 76#define ALOGVV ALOGV 77#else 78#define ALOGVV(a...) do { } while(0) 79#endif 80 81namespace android { 82 83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85 86 87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 88 89uint32_t AudioFlinger::mScreenState; 90 91#ifdef TEE_SINK 92bool AudioFlinger::mTeeSinkInputEnabled = false; 93bool AudioFlinger::mTeeSinkOutputEnabled = false; 94bool AudioFlinger::mTeeSinkTrackEnabled = false; 95 96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 99#endif 100 101//TODO: remove when effect offload is implemented 102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 103// we define a minimum time during which a global effect is considered enabled. 104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 105 106// ---------------------------------------------------------------------------- 107 108static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 109{ 110 const hw_module_t *mod; 111 int rc; 112 113 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 114 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 115 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 116 if (rc) { 117 goto out; 118 } 119 rc = audio_hw_device_open(mod, dev); 120 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 121 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 122 if (rc) { 123 goto out; 124 } 125 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 126 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 127 rc = BAD_VALUE; 128 goto out; 129 } 130 return 0; 131 132out: 133 *dev = NULL; 134 return rc; 135} 136 137// ---------------------------------------------------------------------------- 138 139AudioFlinger::AudioFlinger() 140 : BnAudioFlinger(), 141 mPrimaryHardwareDev(NULL), 142 mHardwareStatus(AUDIO_HW_IDLE), 143 mMasterVolume(1.0f), 144 mMasterMute(false), 145 mNextUniqueId(1), 146 mMode(AUDIO_MODE_INVALID), 147 mBtNrecIsOff(false), 148 mIsLowRamDevice(true), 149 mIsDeviceTypeKnown(false), 150 mGlobalEffectEnableTime(0) 151{ 152 getpid_cached = getpid(); 153 char value[PROPERTY_VALUE_MAX]; 154 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 155 if (doLog) { 156 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters"); 157 } 158#ifdef TEE_SINK 159 (void) property_get("ro.debuggable", value, "0"); 160 int debuggable = atoi(value); 161 int teeEnabled = 0; 162 if (debuggable) { 163 (void) property_get("af.tee", value, "0"); 164 teeEnabled = atoi(value); 165 } 166 if (teeEnabled & 1) 167 mTeeSinkInputEnabled = true; 168 if (teeEnabled & 2) 169 mTeeSinkOutputEnabled = true; 170 if (teeEnabled & 4) 171 mTeeSinkTrackEnabled = true; 172#endif 173} 174 175void AudioFlinger::onFirstRef() 176{ 177 int rc = 0; 178 179 Mutex::Autolock _l(mLock); 180 181 /* TODO: move all this work into an Init() function */ 182 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 183 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 184 uint32_t int_val; 185 if (1 == sscanf(val_str, "%u", &int_val)) { 186 mStandbyTimeInNsecs = milliseconds(int_val); 187 ALOGI("Using %u mSec as standby time.", int_val); 188 } else { 189 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 190 ALOGI("Using default %u mSec as standby time.", 191 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 192 } 193 } 194 195 mMode = AUDIO_MODE_NORMAL; 196} 197 198AudioFlinger::~AudioFlinger() 199{ 200 while (!mRecordThreads.isEmpty()) { 201 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 202 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 203 } 204 while (!mPlaybackThreads.isEmpty()) { 205 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 206 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 207 } 208 209 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 210 // no mHardwareLock needed, as there are no other references to this 211 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 212 delete mAudioHwDevs.valueAt(i); 213 } 214} 215 216static const char * const audio_interfaces[] = { 217 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 218 AUDIO_HARDWARE_MODULE_ID_A2DP, 219 AUDIO_HARDWARE_MODULE_ID_USB, 220}; 221#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 222 223AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 224 audio_module_handle_t module, 225 audio_devices_t devices) 226{ 227 // if module is 0, the request comes from an old policy manager and we should load 228 // well known modules 229 if (module == 0) { 230 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 231 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 232 loadHwModule_l(audio_interfaces[i]); 233 } 234 // then try to find a module supporting the requested device. 235 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 236 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 237 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 238 if ((dev->get_supported_devices != NULL) && 239 (dev->get_supported_devices(dev) & devices) == devices) 240 return audioHwDevice; 241 } 242 } else { 243 // check a match for the requested module handle 244 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 245 if (audioHwDevice != NULL) { 246 return audioHwDevice; 247 } 248 } 249 250 return NULL; 251} 252 253void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 254{ 255 const size_t SIZE = 256; 256 char buffer[SIZE]; 257 String8 result; 258 259 result.append("Clients:\n"); 260 for (size_t i = 0; i < mClients.size(); ++i) { 261 sp<Client> client = mClients.valueAt(i).promote(); 262 if (client != 0) { 263 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 264 result.append(buffer); 265 } 266 } 267 268 result.append("Global session refs:\n"); 269 result.append(" session pid count\n"); 270 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 271 AudioSessionRef *r = mAudioSessionRefs[i]; 272 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 273 result.append(buffer); 274 } 275 write(fd, result.string(), result.size()); 276} 277 278 279void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 280{ 281 const size_t SIZE = 256; 282 char buffer[SIZE]; 283 String8 result; 284 hardware_call_state hardwareStatus = mHardwareStatus; 285 286 snprintf(buffer, SIZE, "Hardware status: %d\n" 287 "Standby Time mSec: %u\n", 288 hardwareStatus, 289 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 290 result.append(buffer); 291 write(fd, result.string(), result.size()); 292} 293 294void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 295{ 296 const size_t SIZE = 256; 297 char buffer[SIZE]; 298 String8 result; 299 snprintf(buffer, SIZE, "Permission Denial: " 300 "can't dump AudioFlinger from pid=%d, uid=%d\n", 301 IPCThreadState::self()->getCallingPid(), 302 IPCThreadState::self()->getCallingUid()); 303 result.append(buffer); 304 write(fd, result.string(), result.size()); 305} 306 307bool AudioFlinger::dumpTryLock(Mutex& mutex) 308{ 309 bool locked = false; 310 for (int i = 0; i < kDumpLockRetries; ++i) { 311 if (mutex.tryLock() == NO_ERROR) { 312 locked = true; 313 break; 314 } 315 usleep(kDumpLockSleepUs); 316 } 317 return locked; 318} 319 320status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 321{ 322 if (!dumpAllowed()) { 323 dumpPermissionDenial(fd, args); 324 } else { 325 // get state of hardware lock 326 bool hardwareLocked = dumpTryLock(mHardwareLock); 327 if (!hardwareLocked) { 328 String8 result(kHardwareLockedString); 329 write(fd, result.string(), result.size()); 330 } else { 331 mHardwareLock.unlock(); 332 } 333 334 bool locked = dumpTryLock(mLock); 335 336 // failed to lock - AudioFlinger is probably deadlocked 337 if (!locked) { 338 String8 result(kDeadlockedString); 339 write(fd, result.string(), result.size()); 340 } 341 342 dumpClients(fd, args); 343 dumpInternals(fd, args); 344 345 // dump playback threads 346 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 347 mPlaybackThreads.valueAt(i)->dump(fd, args); 348 } 349 350 // dump record threads 351 for (size_t i = 0; i < mRecordThreads.size(); i++) { 352 mRecordThreads.valueAt(i)->dump(fd, args); 353 } 354 355 // dump all hardware devs 356 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 357 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 358 dev->dump(dev, fd); 359 } 360 361#ifdef TEE_SINK 362 // dump the serially shared record tee sink 363 if (mRecordTeeSource != 0) { 364 dumpTee(fd, mRecordTeeSource); 365 } 366#endif 367 368 if (locked) { 369 mLock.unlock(); 370 } 371 372 // append a copy of media.log here by forwarding fd to it, but don't attempt 373 // to lookup the service if it's not running, as it will block for a second 374 if (mLogMemoryDealer != 0) { 375 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 376 if (binder != 0) { 377 fdprintf(fd, "\nmedia.log:\n"); 378 Vector<String16> args; 379 binder->dump(fd, args); 380 } 381 } 382 } 383 return NO_ERROR; 384} 385 386sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 387{ 388 // If pid is already in the mClients wp<> map, then use that entry 389 // (for which promote() is always != 0), otherwise create a new entry and Client. 390 sp<Client> client = mClients.valueFor(pid).promote(); 391 if (client == 0) { 392 client = new Client(this, pid); 393 mClients.add(pid, client); 394 } 395 396 return client; 397} 398 399sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 400{ 401 if (mLogMemoryDealer == 0) { 402 return new NBLog::Writer(); 403 } 404 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 405 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 406 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 407 if (binder != 0) { 408 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 409 } 410 return writer; 411} 412 413void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 414{ 415 if (writer == 0) { 416 return; 417 } 418 sp<IMemory> iMemory(writer->getIMemory()); 419 if (iMemory == 0) { 420 return; 421 } 422 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 423 if (binder != 0) { 424 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 425 // Now the media.log remote reference to IMemory is gone. 426 // When our last local reference to IMemory also drops to zero, 427 // the IMemory destructor will deallocate the region from mMemoryDealer. 428 } 429} 430 431// IAudioFlinger interface 432 433 434sp<IAudioTrack> AudioFlinger::createTrack( 435 audio_stream_type_t streamType, 436 uint32_t sampleRate, 437 audio_format_t format, 438 audio_channel_mask_t channelMask, 439 size_t frameCount, 440 IAudioFlinger::track_flags_t *flags, 441 const sp<IMemory>& sharedBuffer, 442 audio_io_handle_t output, 443 pid_t tid, 444 int *sessionId, 445 String8& name, 446 status_t *status) 447{ 448 sp<PlaybackThread::Track> track; 449 sp<TrackHandle> trackHandle; 450 sp<Client> client; 451 status_t lStatus; 452 int lSessionId; 453 454 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 455 // but if someone uses binder directly they could bypass that and cause us to crash 456 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 457 ALOGE("createTrack() invalid stream type %d", streamType); 458 lStatus = BAD_VALUE; 459 goto Exit; 460 } 461 462 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 463 // and we don't yet support 8.24 or 32-bit PCM 464 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 465 ALOGE("createTrack() invalid format %d", format); 466 lStatus = BAD_VALUE; 467 goto Exit; 468 } 469 470 { 471 Mutex::Autolock _l(mLock); 472 PlaybackThread *thread = checkPlaybackThread_l(output); 473 PlaybackThread *effectThread = NULL; 474 if (thread == NULL) { 475 ALOGE("no playback thread found for output handle %d", output); 476 lStatus = BAD_VALUE; 477 goto Exit; 478 } 479 480 pid_t pid = IPCThreadState::self()->getCallingPid(); 481 client = registerPid_l(pid); 482 483 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 484 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 485 // check if an effect chain with the same session ID is present on another 486 // output thread and move it here. 487 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 488 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 489 if (mPlaybackThreads.keyAt(i) != output) { 490 uint32_t sessions = t->hasAudioSession(*sessionId); 491 if (sessions & PlaybackThread::EFFECT_SESSION) { 492 effectThread = t.get(); 493 break; 494 } 495 } 496 } 497 lSessionId = *sessionId; 498 } else { 499 // if no audio session id is provided, create one here 500 lSessionId = nextUniqueId(); 501 if (sessionId != NULL) { 502 *sessionId = lSessionId; 503 } 504 } 505 ALOGV("createTrack() lSessionId: %d", lSessionId); 506 507 track = thread->createTrack_l(client, streamType, sampleRate, format, 508 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 509 510 // move effect chain to this output thread if an effect on same session was waiting 511 // for a track to be created 512 if (lStatus == NO_ERROR && effectThread != NULL) { 513 Mutex::Autolock _dl(thread->mLock); 514 Mutex::Autolock _sl(effectThread->mLock); 515 moveEffectChain_l(lSessionId, effectThread, thread, true); 516 } 517 518 // Look for sync events awaiting for a session to be used. 519 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 520 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 521 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 522 if (lStatus == NO_ERROR) { 523 (void) track->setSyncEvent(mPendingSyncEvents[i]); 524 } else { 525 mPendingSyncEvents[i]->cancel(); 526 } 527 mPendingSyncEvents.removeAt(i); 528 i--; 529 } 530 } 531 } 532 } 533 if (lStatus == NO_ERROR) { 534 // s for server's pid, n for normal mixer name, f for fast index 535 name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, 536 track->fastIndex()); 537 trackHandle = new TrackHandle(track); 538 } else { 539 // remove local strong reference to Client before deleting the Track so that the Client 540 // destructor is called by the TrackBase destructor with mLock held 541 client.clear(); 542 track.clear(); 543 } 544 545Exit: 546 if (status != NULL) { 547 *status = lStatus; 548 } 549 return trackHandle; 550} 551 552uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 553{ 554 Mutex::Autolock _l(mLock); 555 PlaybackThread *thread = checkPlaybackThread_l(output); 556 if (thread == NULL) { 557 ALOGW("sampleRate() unknown thread %d", output); 558 return 0; 559 } 560 return thread->sampleRate(); 561} 562 563int AudioFlinger::channelCount(audio_io_handle_t output) const 564{ 565 Mutex::Autolock _l(mLock); 566 PlaybackThread *thread = checkPlaybackThread_l(output); 567 if (thread == NULL) { 568 ALOGW("channelCount() unknown thread %d", output); 569 return 0; 570 } 571 return thread->channelCount(); 572} 573 574audio_format_t AudioFlinger::format(audio_io_handle_t output) const 575{ 576 Mutex::Autolock _l(mLock); 577 PlaybackThread *thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 ALOGW("format() unknown thread %d", output); 580 return AUDIO_FORMAT_INVALID; 581 } 582 return thread->format(); 583} 584 585size_t AudioFlinger::frameCount(audio_io_handle_t output) const 586{ 587 Mutex::Autolock _l(mLock); 588 PlaybackThread *thread = checkPlaybackThread_l(output); 589 if (thread == NULL) { 590 ALOGW("frameCount() unknown thread %d", output); 591 return 0; 592 } 593 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 594 // should examine all callers and fix them to handle smaller counts 595 return thread->frameCount(); 596} 597 598uint32_t AudioFlinger::latency(audio_io_handle_t output) const 599{ 600 Mutex::Autolock _l(mLock); 601 PlaybackThread *thread = checkPlaybackThread_l(output); 602 if (thread == NULL) { 603 ALOGW("latency(): no playback thread found for output handle %d", output); 604 return 0; 605 } 606 return thread->latency(); 607} 608 609status_t AudioFlinger::setMasterVolume(float value) 610{ 611 status_t ret = initCheck(); 612 if (ret != NO_ERROR) { 613 return ret; 614 } 615 616 // check calling permissions 617 if (!settingsAllowed()) { 618 return PERMISSION_DENIED; 619 } 620 621 Mutex::Autolock _l(mLock); 622 mMasterVolume = value; 623 624 // Set master volume in the HALs which support it. 625 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 626 AutoMutex lock(mHardwareLock); 627 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 628 629 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 630 if (dev->canSetMasterVolume()) { 631 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 632 } 633 mHardwareStatus = AUDIO_HW_IDLE; 634 } 635 636 // Now set the master volume in each playback thread. Playback threads 637 // assigned to HALs which do not have master volume support will apply 638 // master volume during the mix operation. Threads with HALs which do 639 // support master volume will simply ignore the setting. 640 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 641 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 642 643 return NO_ERROR; 644} 645 646status_t AudioFlinger::setMode(audio_mode_t mode) 647{ 648 status_t ret = initCheck(); 649 if (ret != NO_ERROR) { 650 return ret; 651 } 652 653 // check calling permissions 654 if (!settingsAllowed()) { 655 return PERMISSION_DENIED; 656 } 657 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 658 ALOGW("Illegal value: setMode(%d)", mode); 659 return BAD_VALUE; 660 } 661 662 { // scope for the lock 663 AutoMutex lock(mHardwareLock); 664 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 665 mHardwareStatus = AUDIO_HW_SET_MODE; 666 ret = dev->set_mode(dev, mode); 667 mHardwareStatus = AUDIO_HW_IDLE; 668 } 669 670 if (NO_ERROR == ret) { 671 Mutex::Autolock _l(mLock); 672 mMode = mode; 673 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 674 mPlaybackThreads.valueAt(i)->setMode(mode); 675 } 676 677 return ret; 678} 679 680status_t AudioFlinger::setMicMute(bool state) 681{ 682 status_t ret = initCheck(); 683 if (ret != NO_ERROR) { 684 return ret; 685 } 686 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 AutoMutex lock(mHardwareLock); 693 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 694 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 695 ret = dev->set_mic_mute(dev, state); 696 mHardwareStatus = AUDIO_HW_IDLE; 697 return ret; 698} 699 700bool AudioFlinger::getMicMute() const 701{ 702 status_t ret = initCheck(); 703 if (ret != NO_ERROR) { 704 return false; 705 } 706 707 bool state = AUDIO_MODE_INVALID; 708 AutoMutex lock(mHardwareLock); 709 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 710 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 711 dev->get_mic_mute(dev, &state); 712 mHardwareStatus = AUDIO_HW_IDLE; 713 return state; 714} 715 716status_t AudioFlinger::setMasterMute(bool muted) 717{ 718 status_t ret = initCheck(); 719 if (ret != NO_ERROR) { 720 return ret; 721 } 722 723 // check calling permissions 724 if (!settingsAllowed()) { 725 return PERMISSION_DENIED; 726 } 727 728 Mutex::Autolock _l(mLock); 729 mMasterMute = muted; 730 731 // Set master mute in the HALs which support it. 732 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 733 AutoMutex lock(mHardwareLock); 734 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 735 736 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 737 if (dev->canSetMasterMute()) { 738 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 739 } 740 mHardwareStatus = AUDIO_HW_IDLE; 741 } 742 743 // Now set the master mute in each playback thread. Playback threads 744 // assigned to HALs which do not have master mute support will apply master 745 // mute during the mix operation. Threads with HALs which do support master 746 // mute will simply ignore the setting. 747 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 748 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 749 750 return NO_ERROR; 751} 752 753float AudioFlinger::masterVolume() const 754{ 755 Mutex::Autolock _l(mLock); 756 return masterVolume_l(); 757} 758 759bool AudioFlinger::masterMute() const 760{ 761 Mutex::Autolock _l(mLock); 762 return masterMute_l(); 763} 764 765float AudioFlinger::masterVolume_l() const 766{ 767 return mMasterVolume; 768} 769 770bool AudioFlinger::masterMute_l() const 771{ 772 return mMasterMute; 773} 774 775status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 776 audio_io_handle_t output) 777{ 778 // check calling permissions 779 if (!settingsAllowed()) { 780 return PERMISSION_DENIED; 781 } 782 783 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 784 ALOGE("setStreamVolume() invalid stream %d", stream); 785 return BAD_VALUE; 786 } 787 788 AutoMutex lock(mLock); 789 PlaybackThread *thread = NULL; 790 if (output) { 791 thread = checkPlaybackThread_l(output); 792 if (thread == NULL) { 793 return BAD_VALUE; 794 } 795 } 796 797 mStreamTypes[stream].volume = value; 798 799 if (thread == NULL) { 800 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 801 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 802 } 803 } else { 804 thread->setStreamVolume(stream, value); 805 } 806 807 return NO_ERROR; 808} 809 810status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 811{ 812 // check calling permissions 813 if (!settingsAllowed()) { 814 return PERMISSION_DENIED; 815 } 816 817 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 818 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 819 ALOGE("setStreamMute() invalid stream %d", stream); 820 return BAD_VALUE; 821 } 822 823 AutoMutex lock(mLock); 824 mStreamTypes[stream].mute = muted; 825 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 826 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 827 828 return NO_ERROR; 829} 830 831float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 832{ 833 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 834 return 0.0f; 835 } 836 837 AutoMutex lock(mLock); 838 float volume; 839 if (output) { 840 PlaybackThread *thread = checkPlaybackThread_l(output); 841 if (thread == NULL) { 842 return 0.0f; 843 } 844 volume = thread->streamVolume(stream); 845 } else { 846 volume = streamVolume_l(stream); 847 } 848 849 return volume; 850} 851 852bool AudioFlinger::streamMute(audio_stream_type_t stream) const 853{ 854 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 855 return true; 856 } 857 858 AutoMutex lock(mLock); 859 return streamMute_l(stream); 860} 861 862status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 863{ 864 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 865 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 866 867 // check calling permissions 868 if (!settingsAllowed()) { 869 return PERMISSION_DENIED; 870 } 871 872 // ioHandle == 0 means the parameters are global to the audio hardware interface 873 if (ioHandle == 0) { 874 Mutex::Autolock _l(mLock); 875 status_t final_result = NO_ERROR; 876 { 877 AutoMutex lock(mHardwareLock); 878 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 879 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 880 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 881 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 882 final_result = result ?: final_result; 883 } 884 mHardwareStatus = AUDIO_HW_IDLE; 885 } 886 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 887 AudioParameter param = AudioParameter(keyValuePairs); 888 String8 value; 889 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 890 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 891 if (mBtNrecIsOff != btNrecIsOff) { 892 for (size_t i = 0; i < mRecordThreads.size(); i++) { 893 sp<RecordThread> thread = mRecordThreads.valueAt(i); 894 audio_devices_t device = thread->inDevice(); 895 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 896 // collect all of the thread's session IDs 897 KeyedVector<int, bool> ids = thread->sessionIds(); 898 // suspend effects associated with those session IDs 899 for (size_t j = 0; j < ids.size(); ++j) { 900 int sessionId = ids.keyAt(j); 901 thread->setEffectSuspended(FX_IID_AEC, 902 suspend, 903 sessionId); 904 thread->setEffectSuspended(FX_IID_NS, 905 suspend, 906 sessionId); 907 } 908 } 909 mBtNrecIsOff = btNrecIsOff; 910 } 911 } 912 String8 screenState; 913 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 914 bool isOff = screenState == "off"; 915 if (isOff != (AudioFlinger::mScreenState & 1)) { 916 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 917 } 918 } 919 return final_result; 920 } 921 922 // hold a strong ref on thread in case closeOutput() or closeInput() is called 923 // and the thread is exited once the lock is released 924 sp<ThreadBase> thread; 925 { 926 Mutex::Autolock _l(mLock); 927 thread = checkPlaybackThread_l(ioHandle); 928 if (thread == 0) { 929 thread = checkRecordThread_l(ioHandle); 930 } else if (thread == primaryPlaybackThread_l()) { 931 // indicate output device change to all input threads for pre processing 932 AudioParameter param = AudioParameter(keyValuePairs); 933 int value; 934 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 935 (value != 0)) { 936 for (size_t i = 0; i < mRecordThreads.size(); i++) { 937 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 938 } 939 } 940 } 941 } 942 if (thread != 0) { 943 return thread->setParameters(keyValuePairs); 944 } 945 return BAD_VALUE; 946} 947 948String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 949{ 950 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 951 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 952 953 Mutex::Autolock _l(mLock); 954 955 if (ioHandle == 0) { 956 String8 out_s8; 957 958 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 959 char *s; 960 { 961 AutoMutex lock(mHardwareLock); 962 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 963 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 964 s = dev->get_parameters(dev, keys.string()); 965 mHardwareStatus = AUDIO_HW_IDLE; 966 } 967 out_s8 += String8(s ? s : ""); 968 free(s); 969 } 970 return out_s8; 971 } 972 973 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 974 if (playbackThread != NULL) { 975 return playbackThread->getParameters(keys); 976 } 977 RecordThread *recordThread = checkRecordThread_l(ioHandle); 978 if (recordThread != NULL) { 979 return recordThread->getParameters(keys); 980 } 981 return String8(""); 982} 983 984size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 985 audio_channel_mask_t channelMask) const 986{ 987 status_t ret = initCheck(); 988 if (ret != NO_ERROR) { 989 return 0; 990 } 991 992 AutoMutex lock(mHardwareLock); 993 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 994 struct audio_config config; 995 memset(&config, 0, sizeof(config)); 996 config.sample_rate = sampleRate; 997 config.channel_mask = channelMask; 998 config.format = format; 999 1000 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1001 size_t size = dev->get_input_buffer_size(dev, &config); 1002 mHardwareStatus = AUDIO_HW_IDLE; 1003 return size; 1004} 1005 1006unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1007{ 1008 Mutex::Autolock _l(mLock); 1009 1010 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1011 if (recordThread != NULL) { 1012 return recordThread->getInputFramesLost(); 1013 } 1014 return 0; 1015} 1016 1017status_t AudioFlinger::setVoiceVolume(float value) 1018{ 1019 status_t ret = initCheck(); 1020 if (ret != NO_ERROR) { 1021 return ret; 1022 } 1023 1024 // check calling permissions 1025 if (!settingsAllowed()) { 1026 return PERMISSION_DENIED; 1027 } 1028 1029 AutoMutex lock(mHardwareLock); 1030 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1031 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1032 ret = dev->set_voice_volume(dev, value); 1033 mHardwareStatus = AUDIO_HW_IDLE; 1034 1035 return ret; 1036} 1037 1038status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1039 audio_io_handle_t output) const 1040{ 1041 status_t status; 1042 1043 Mutex::Autolock _l(mLock); 1044 1045 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1046 if (playbackThread != NULL) { 1047 return playbackThread->getRenderPosition(halFrames, dspFrames); 1048 } 1049 1050 return BAD_VALUE; 1051} 1052 1053void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1054{ 1055 1056 Mutex::Autolock _l(mLock); 1057 1058 pid_t pid = IPCThreadState::self()->getCallingPid(); 1059 if (mNotificationClients.indexOfKey(pid) < 0) { 1060 sp<NotificationClient> notificationClient = new NotificationClient(this, 1061 client, 1062 pid); 1063 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1064 1065 mNotificationClients.add(pid, notificationClient); 1066 1067 sp<IBinder> binder = client->asBinder(); 1068 binder->linkToDeath(notificationClient); 1069 1070 // the config change is always sent from playback or record threads to avoid deadlock 1071 // with AudioSystem::gLock 1072 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1073 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1074 } 1075 1076 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1077 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1078 } 1079 } 1080} 1081 1082void AudioFlinger::removeNotificationClient(pid_t pid) 1083{ 1084 Mutex::Autolock _l(mLock); 1085 1086 mNotificationClients.removeItem(pid); 1087 1088 ALOGV("%d died, releasing its sessions", pid); 1089 size_t num = mAudioSessionRefs.size(); 1090 bool removed = false; 1091 for (size_t i = 0; i< num; ) { 1092 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1093 ALOGV(" pid %d @ %d", ref->mPid, i); 1094 if (ref->mPid == pid) { 1095 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1096 mAudioSessionRefs.removeAt(i); 1097 delete ref; 1098 removed = true; 1099 num--; 1100 } else { 1101 i++; 1102 } 1103 } 1104 if (removed) { 1105 purgeStaleEffects_l(); 1106 } 1107} 1108 1109// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1110void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1111{ 1112 size_t size = mNotificationClients.size(); 1113 for (size_t i = 0; i < size; i++) { 1114 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1115 param2); 1116 } 1117} 1118 1119// removeClient_l() must be called with AudioFlinger::mLock held 1120void AudioFlinger::removeClient_l(pid_t pid) 1121{ 1122 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1123 IPCThreadState::self()->getCallingPid()); 1124 mClients.removeItem(pid); 1125} 1126 1127// getEffectThread_l() must be called with AudioFlinger::mLock held 1128sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1129{ 1130 sp<PlaybackThread> thread; 1131 1132 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1133 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1134 ALOG_ASSERT(thread == 0); 1135 thread = mPlaybackThreads.valueAt(i); 1136 } 1137 } 1138 1139 return thread; 1140} 1141 1142 1143 1144// ---------------------------------------------------------------------------- 1145 1146AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1147 : RefBase(), 1148 mAudioFlinger(audioFlinger), 1149 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1150 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1151 mPid(pid), 1152 mTimedTrackCount(0) 1153{ 1154 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1155} 1156 1157// Client destructor must be called with AudioFlinger::mLock held 1158AudioFlinger::Client::~Client() 1159{ 1160 mAudioFlinger->removeClient_l(mPid); 1161} 1162 1163sp<MemoryDealer> AudioFlinger::Client::heap() const 1164{ 1165 return mMemoryDealer; 1166} 1167 1168// Reserve one of the limited slots for a timed audio track associated 1169// with this client 1170bool AudioFlinger::Client::reserveTimedTrack() 1171{ 1172 const int kMaxTimedTracksPerClient = 4; 1173 1174 Mutex::Autolock _l(mTimedTrackLock); 1175 1176 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1177 ALOGW("can not create timed track - pid %d has exceeded the limit", 1178 mPid); 1179 return false; 1180 } 1181 1182 mTimedTrackCount++; 1183 return true; 1184} 1185 1186// Release a slot for a timed audio track 1187void AudioFlinger::Client::releaseTimedTrack() 1188{ 1189 Mutex::Autolock _l(mTimedTrackLock); 1190 mTimedTrackCount--; 1191} 1192 1193// ---------------------------------------------------------------------------- 1194 1195AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1196 const sp<IAudioFlingerClient>& client, 1197 pid_t pid) 1198 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1199{ 1200} 1201 1202AudioFlinger::NotificationClient::~NotificationClient() 1203{ 1204} 1205 1206void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1207{ 1208 sp<NotificationClient> keep(this); 1209 mAudioFlinger->removeNotificationClient(mPid); 1210} 1211 1212 1213// ---------------------------------------------------------------------------- 1214 1215static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1216 return audio_is_remote_submix_device(inDevice); 1217} 1218 1219sp<IAudioRecord> AudioFlinger::openRecord( 1220 audio_io_handle_t input, 1221 uint32_t sampleRate, 1222 audio_format_t format, 1223 audio_channel_mask_t channelMask, 1224 size_t frameCount, 1225 IAudioFlinger::track_flags_t *flags, 1226 pid_t tid, 1227 int *sessionId, 1228 status_t *status) 1229{ 1230 sp<RecordThread::RecordTrack> recordTrack; 1231 sp<RecordHandle> recordHandle; 1232 sp<Client> client; 1233 status_t lStatus; 1234 RecordThread *thread; 1235 size_t inFrameCount; 1236 int lSessionId; 1237 1238 // check calling permissions 1239 if (!recordingAllowed()) { 1240 lStatus = PERMISSION_DENIED; 1241 goto Exit; 1242 } 1243 1244 if (format != AUDIO_FORMAT_PCM_16_BIT) { 1245 ALOGE("openRecord() invalid format %d", format); 1246 lStatus = BAD_VALUE; 1247 goto Exit; 1248 } 1249 1250 // add client to list 1251 { // scope for mLock 1252 Mutex::Autolock _l(mLock); 1253 thread = checkRecordThread_l(input); 1254 if (thread == NULL) { 1255 lStatus = BAD_VALUE; 1256 goto Exit; 1257 } 1258 1259 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1260 && !captureAudioOutputAllowed()) { 1261 lStatus = PERMISSION_DENIED; 1262 goto Exit; 1263 } 1264 1265 pid_t pid = IPCThreadState::self()->getCallingPid(); 1266 client = registerPid_l(pid); 1267 1268 // If no audio session id is provided, create one here 1269 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1270 lSessionId = *sessionId; 1271 } else { 1272 lSessionId = nextUniqueId(); 1273 if (sessionId != NULL) { 1274 *sessionId = lSessionId; 1275 } 1276 } 1277 // create new record track. 1278 // The record track uses one track in mHardwareMixerThread by convention. 1279 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1280 frameCount, lSessionId, flags, tid, &lStatus); 1281 } 1282 if (lStatus != NO_ERROR) { 1283 // remove local strong reference to Client before deleting the RecordTrack so that the 1284 // Client destructor is called by the TrackBase destructor with mLock held 1285 client.clear(); 1286 recordTrack.clear(); 1287 goto Exit; 1288 } 1289 1290 // return to handle to client 1291 recordHandle = new RecordHandle(recordTrack); 1292 lStatus = NO_ERROR; 1293 1294Exit: 1295 if (status) { 1296 *status = lStatus; 1297 } 1298 return recordHandle; 1299} 1300 1301 1302 1303// ---------------------------------------------------------------------------- 1304 1305audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1306{ 1307 if (!settingsAllowed()) { 1308 return 0; 1309 } 1310 Mutex::Autolock _l(mLock); 1311 return loadHwModule_l(name); 1312} 1313 1314// loadHwModule_l() must be called with AudioFlinger::mLock held 1315audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1316{ 1317 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1318 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1319 ALOGW("loadHwModule() module %s already loaded", name); 1320 return mAudioHwDevs.keyAt(i); 1321 } 1322 } 1323 1324 audio_hw_device_t *dev; 1325 1326 int rc = load_audio_interface(name, &dev); 1327 if (rc) { 1328 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1329 return 0; 1330 } 1331 1332 mHardwareStatus = AUDIO_HW_INIT; 1333 rc = dev->init_check(dev); 1334 mHardwareStatus = AUDIO_HW_IDLE; 1335 if (rc) { 1336 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1337 return 0; 1338 } 1339 1340 // Check and cache this HAL's level of support for master mute and master 1341 // volume. If this is the first HAL opened, and it supports the get 1342 // methods, use the initial values provided by the HAL as the current 1343 // master mute and volume settings. 1344 1345 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1346 { // scope for auto-lock pattern 1347 AutoMutex lock(mHardwareLock); 1348 1349 if (0 == mAudioHwDevs.size()) { 1350 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1351 if (NULL != dev->get_master_volume) { 1352 float mv; 1353 if (OK == dev->get_master_volume(dev, &mv)) { 1354 mMasterVolume = mv; 1355 } 1356 } 1357 1358 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1359 if (NULL != dev->get_master_mute) { 1360 bool mm; 1361 if (OK == dev->get_master_mute(dev, &mm)) { 1362 mMasterMute = mm; 1363 } 1364 } 1365 } 1366 1367 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1368 if ((NULL != dev->set_master_volume) && 1369 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1370 flags = static_cast<AudioHwDevice::Flags>(flags | 1371 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1372 } 1373 1374 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1375 if ((NULL != dev->set_master_mute) && 1376 (OK == dev->set_master_mute(dev, mMasterMute))) { 1377 flags = static_cast<AudioHwDevice::Flags>(flags | 1378 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1379 } 1380 1381 mHardwareStatus = AUDIO_HW_IDLE; 1382 } 1383 1384 audio_module_handle_t handle = nextUniqueId(); 1385 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1386 1387 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1388 name, dev->common.module->name, dev->common.module->id, handle); 1389 1390 return handle; 1391 1392} 1393 1394// ---------------------------------------------------------------------------- 1395 1396uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1397{ 1398 Mutex::Autolock _l(mLock); 1399 PlaybackThread *thread = primaryPlaybackThread_l(); 1400 return thread != NULL ? thread->sampleRate() : 0; 1401} 1402 1403size_t AudioFlinger::getPrimaryOutputFrameCount() 1404{ 1405 Mutex::Autolock _l(mLock); 1406 PlaybackThread *thread = primaryPlaybackThread_l(); 1407 return thread != NULL ? thread->frameCountHAL() : 0; 1408} 1409 1410// ---------------------------------------------------------------------------- 1411 1412status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1413{ 1414 uid_t uid = IPCThreadState::self()->getCallingUid(); 1415 if (uid != AID_SYSTEM) { 1416 return PERMISSION_DENIED; 1417 } 1418 Mutex::Autolock _l(mLock); 1419 if (mIsDeviceTypeKnown) { 1420 return INVALID_OPERATION; 1421 } 1422 mIsLowRamDevice = isLowRamDevice; 1423 mIsDeviceTypeKnown = true; 1424 return NO_ERROR; 1425} 1426 1427// ---------------------------------------------------------------------------- 1428 1429audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1430 audio_devices_t *pDevices, 1431 uint32_t *pSamplingRate, 1432 audio_format_t *pFormat, 1433 audio_channel_mask_t *pChannelMask, 1434 uint32_t *pLatencyMs, 1435 audio_output_flags_t flags, 1436 const audio_offload_info_t *offloadInfo) 1437{ 1438 PlaybackThread *thread = NULL; 1439 struct audio_config config; 1440 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1441 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1442 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1443 if (offloadInfo) { 1444 config.offload_info = *offloadInfo; 1445 } 1446 1447 audio_stream_out_t *outStream = NULL; 1448 AudioHwDevice *outHwDev; 1449 1450 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1451 module, 1452 (pDevices != NULL) ? *pDevices : 0, 1453 config.sample_rate, 1454 config.format, 1455 config.channel_mask, 1456 flags); 1457 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1458 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); 1459 1460 if (pDevices == NULL || *pDevices == 0) { 1461 return 0; 1462 } 1463 1464 Mutex::Autolock _l(mLock); 1465 1466 outHwDev = findSuitableHwDev_l(module, *pDevices); 1467 if (outHwDev == NULL) 1468 return 0; 1469 1470 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1471 audio_io_handle_t id = nextUniqueId(); 1472 1473 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1474 1475 status_t status = hwDevHal->open_output_stream(hwDevHal, 1476 id, 1477 *pDevices, 1478 (audio_output_flags_t)flags, 1479 &config, 1480 &outStream); 1481 1482 mHardwareStatus = AUDIO_HW_IDLE; 1483 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1484 "Channels %x, status %d", 1485 outStream, 1486 config.sample_rate, 1487 config.format, 1488 config.channel_mask, 1489 status); 1490 1491 if (status == NO_ERROR && outStream != NULL) { 1492 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1493 1494 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1495 thread = new OffloadThread(this, output, id, *pDevices); 1496 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1497 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1498 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1499 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1500 thread = new DirectOutputThread(this, output, id, *pDevices); 1501 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1502 } else { 1503 thread = new MixerThread(this, output, id, *pDevices); 1504 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1505 } 1506 mPlaybackThreads.add(id, thread); 1507 1508 if (pSamplingRate != NULL) { 1509 *pSamplingRate = config.sample_rate; 1510 } 1511 if (pFormat != NULL) { 1512 *pFormat = config.format; 1513 } 1514 if (pChannelMask != NULL) { 1515 *pChannelMask = config.channel_mask; 1516 } 1517 if (pLatencyMs != NULL) { 1518 *pLatencyMs = thread->latency(); 1519 } 1520 1521 // notify client processes of the new output creation 1522 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1523 1524 // the first primary output opened designates the primary hw device 1525 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1526 ALOGI("Using module %d has the primary audio interface", module); 1527 mPrimaryHardwareDev = outHwDev; 1528 1529 AutoMutex lock(mHardwareLock); 1530 mHardwareStatus = AUDIO_HW_SET_MODE; 1531 hwDevHal->set_mode(hwDevHal, mMode); 1532 mHardwareStatus = AUDIO_HW_IDLE; 1533 } 1534 return id; 1535 } 1536 1537 return 0; 1538} 1539 1540audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1541 audio_io_handle_t output2) 1542{ 1543 Mutex::Autolock _l(mLock); 1544 MixerThread *thread1 = checkMixerThread_l(output1); 1545 MixerThread *thread2 = checkMixerThread_l(output2); 1546 1547 if (thread1 == NULL || thread2 == NULL) { 1548 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1549 output2); 1550 return 0; 1551 } 1552 1553 audio_io_handle_t id = nextUniqueId(); 1554 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1555 thread->addOutputTrack(thread2); 1556 mPlaybackThreads.add(id, thread); 1557 // notify client processes of the new output creation 1558 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1559 return id; 1560} 1561 1562status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1563{ 1564 return closeOutput_nonvirtual(output); 1565} 1566 1567status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1568{ 1569 // keep strong reference on the playback thread so that 1570 // it is not destroyed while exit() is executed 1571 sp<PlaybackThread> thread; 1572 { 1573 Mutex::Autolock _l(mLock); 1574 thread = checkPlaybackThread_l(output); 1575 if (thread == NULL) { 1576 return BAD_VALUE; 1577 } 1578 1579 ALOGV("closeOutput() %d", output); 1580 1581 if (thread->type() == ThreadBase::MIXER) { 1582 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1583 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1584 DuplicatingThread *dupThread = 1585 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1586 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1587 1588 } 1589 } 1590 } 1591 1592 1593 mPlaybackThreads.removeItem(output); 1594 // save all effects to the default thread 1595 if (mPlaybackThreads.size()) { 1596 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1597 if (dstThread != NULL) { 1598 // audioflinger lock is held here so the acquisition order of thread locks does not 1599 // matter 1600 Mutex::Autolock _dl(dstThread->mLock); 1601 Mutex::Autolock _sl(thread->mLock); 1602 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1603 for (size_t i = 0; i < effectChains.size(); i ++) { 1604 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1605 } 1606 } 1607 } 1608 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1609 } 1610 thread->exit(); 1611 // The thread entity (active unit of execution) is no longer running here, 1612 // but the ThreadBase container still exists. 1613 1614 if (thread->type() != ThreadBase::DUPLICATING) { 1615 AudioStreamOut *out = thread->clearOutput(); 1616 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1617 // from now on thread->mOutput is NULL 1618 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1619 delete out; 1620 } 1621 return NO_ERROR; 1622} 1623 1624status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1625{ 1626 Mutex::Autolock _l(mLock); 1627 PlaybackThread *thread = checkPlaybackThread_l(output); 1628 1629 if (thread == NULL) { 1630 return BAD_VALUE; 1631 } 1632 1633 ALOGV("suspendOutput() %d", output); 1634 thread->suspend(); 1635 1636 return NO_ERROR; 1637} 1638 1639status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1640{ 1641 Mutex::Autolock _l(mLock); 1642 PlaybackThread *thread = checkPlaybackThread_l(output); 1643 1644 if (thread == NULL) { 1645 return BAD_VALUE; 1646 } 1647 1648 ALOGV("restoreOutput() %d", output); 1649 1650 thread->restore(); 1651 1652 return NO_ERROR; 1653} 1654 1655audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1656 audio_devices_t *pDevices, 1657 uint32_t *pSamplingRate, 1658 audio_format_t *pFormat, 1659 audio_channel_mask_t *pChannelMask) 1660{ 1661 status_t status; 1662 RecordThread *thread = NULL; 1663 struct audio_config config; 1664 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1665 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1666 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1667 1668 uint32_t reqSamplingRate = config.sample_rate; 1669 audio_format_t reqFormat = config.format; 1670 audio_channel_mask_t reqChannels = config.channel_mask; 1671 audio_stream_in_t *inStream = NULL; 1672 AudioHwDevice *inHwDev; 1673 1674 if (pDevices == NULL || *pDevices == 0) { 1675 return 0; 1676 } 1677 1678 Mutex::Autolock _l(mLock); 1679 1680 inHwDev = findSuitableHwDev_l(module, *pDevices); 1681 if (inHwDev == NULL) 1682 return 0; 1683 1684 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1685 audio_io_handle_t id = nextUniqueId(); 1686 1687 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1688 &inStream); 1689 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1690 "status %d", 1691 inStream, 1692 config.sample_rate, 1693 config.format, 1694 config.channel_mask, 1695 status); 1696 1697 // If the input could not be opened with the requested parameters and we can handle the 1698 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1699 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1700 if (status == BAD_VALUE && 1701 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1702 (config.sample_rate <= 2 * reqSamplingRate) && 1703 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 1704 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1705 inStream = NULL; 1706 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1707 } 1708 1709 if (status == NO_ERROR && inStream != NULL) { 1710 1711#ifdef TEE_SINK 1712 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1713 // or (re-)create if current Pipe is idle and does not match the new format 1714 sp<NBAIO_Sink> teeSink; 1715 enum { 1716 TEE_SINK_NO, // don't copy input 1717 TEE_SINK_NEW, // copy input using a new pipe 1718 TEE_SINK_OLD, // copy input using an existing pipe 1719 } kind; 1720 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1721 popcount(inStream->common.get_channels(&inStream->common))); 1722 if (!mTeeSinkInputEnabled) { 1723 kind = TEE_SINK_NO; 1724 } else if (format == Format_Invalid) { 1725 kind = TEE_SINK_NO; 1726 } else if (mRecordTeeSink == 0) { 1727 kind = TEE_SINK_NEW; 1728 } else if (mRecordTeeSink->getStrongCount() != 1) { 1729 kind = TEE_SINK_NO; 1730 } else if (format == mRecordTeeSink->format()) { 1731 kind = TEE_SINK_OLD; 1732 } else { 1733 kind = TEE_SINK_NEW; 1734 } 1735 switch (kind) { 1736 case TEE_SINK_NEW: { 1737 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1738 size_t numCounterOffers = 0; 1739 const NBAIO_Format offers[1] = {format}; 1740 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1741 ALOG_ASSERT(index == 0); 1742 PipeReader *pipeReader = new PipeReader(*pipe); 1743 numCounterOffers = 0; 1744 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1745 ALOG_ASSERT(index == 0); 1746 mRecordTeeSink = pipe; 1747 mRecordTeeSource = pipeReader; 1748 teeSink = pipe; 1749 } 1750 break; 1751 case TEE_SINK_OLD: 1752 teeSink = mRecordTeeSink; 1753 break; 1754 case TEE_SINK_NO: 1755 default: 1756 break; 1757 } 1758#endif 1759 1760 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1761 1762 // Start record thread 1763 // RecordThread requires both input and output device indication to forward to audio 1764 // pre processing modules 1765 thread = new RecordThread(this, 1766 input, 1767 reqSamplingRate, 1768 reqChannels, 1769 id, 1770 primaryOutputDevice_l(), 1771 *pDevices 1772#ifdef TEE_SINK 1773 , teeSink 1774#endif 1775 ); 1776 mRecordThreads.add(id, thread); 1777 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1778 if (pSamplingRate != NULL) { 1779 *pSamplingRate = reqSamplingRate; 1780 } 1781 if (pFormat != NULL) { 1782 *pFormat = config.format; 1783 } 1784 if (pChannelMask != NULL) { 1785 *pChannelMask = reqChannels; 1786 } 1787 1788 // notify client processes of the new input creation 1789 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1790 return id; 1791 } 1792 1793 return 0; 1794} 1795 1796status_t AudioFlinger::closeInput(audio_io_handle_t input) 1797{ 1798 return closeInput_nonvirtual(input); 1799} 1800 1801status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1802{ 1803 // keep strong reference on the record thread so that 1804 // it is not destroyed while exit() is executed 1805 sp<RecordThread> thread; 1806 { 1807 Mutex::Autolock _l(mLock); 1808 thread = checkRecordThread_l(input); 1809 if (thread == 0) { 1810 return BAD_VALUE; 1811 } 1812 1813 ALOGV("closeInput() %d", input); 1814 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1815 mRecordThreads.removeItem(input); 1816 } 1817 thread->exit(); 1818 // The thread entity (active unit of execution) is no longer running here, 1819 // but the ThreadBase container still exists. 1820 1821 AudioStreamIn *in = thread->clearInput(); 1822 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1823 // from now on thread->mInput is NULL 1824 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1825 delete in; 1826 1827 return NO_ERROR; 1828} 1829 1830status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1831{ 1832 Mutex::Autolock _l(mLock); 1833 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1834 1835 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1836 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1837 thread->invalidateTracks(stream); 1838 } 1839 1840 return NO_ERROR; 1841} 1842 1843 1844int AudioFlinger::newAudioSessionId() 1845{ 1846 return nextUniqueId(); 1847} 1848 1849void AudioFlinger::acquireAudioSessionId(int audioSession) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 pid_t caller = IPCThreadState::self()->getCallingPid(); 1853 ALOGV("acquiring %d from %d", audioSession, caller); 1854 size_t num = mAudioSessionRefs.size(); 1855 for (size_t i = 0; i< num; i++) { 1856 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1857 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1858 ref->mCnt++; 1859 ALOGV(" incremented refcount to %d", ref->mCnt); 1860 return; 1861 } 1862 } 1863 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1864 ALOGV(" added new entry for %d", audioSession); 1865} 1866 1867void AudioFlinger::releaseAudioSessionId(int audioSession) 1868{ 1869 Mutex::Autolock _l(mLock); 1870 pid_t caller = IPCThreadState::self()->getCallingPid(); 1871 ALOGV("releasing %d from %d", audioSession, caller); 1872 size_t num = mAudioSessionRefs.size(); 1873 for (size_t i = 0; i< num; i++) { 1874 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1875 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1876 ref->mCnt--; 1877 ALOGV(" decremented refcount to %d", ref->mCnt); 1878 if (ref->mCnt == 0) { 1879 mAudioSessionRefs.removeAt(i); 1880 delete ref; 1881 purgeStaleEffects_l(); 1882 } 1883 return; 1884 } 1885 } 1886 ALOGW("session id %d not found for pid %d", audioSession, caller); 1887} 1888 1889void AudioFlinger::purgeStaleEffects_l() { 1890 1891 ALOGV("purging stale effects"); 1892 1893 Vector< sp<EffectChain> > chains; 1894 1895 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1896 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1897 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1898 sp<EffectChain> ec = t->mEffectChains[j]; 1899 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1900 chains.push(ec); 1901 } 1902 } 1903 } 1904 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1905 sp<RecordThread> t = mRecordThreads.valueAt(i); 1906 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1907 sp<EffectChain> ec = t->mEffectChains[j]; 1908 chains.push(ec); 1909 } 1910 } 1911 1912 for (size_t i = 0; i < chains.size(); i++) { 1913 sp<EffectChain> ec = chains[i]; 1914 int sessionid = ec->sessionId(); 1915 sp<ThreadBase> t = ec->mThread.promote(); 1916 if (t == 0) { 1917 continue; 1918 } 1919 size_t numsessionrefs = mAudioSessionRefs.size(); 1920 bool found = false; 1921 for (size_t k = 0; k < numsessionrefs; k++) { 1922 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1923 if (ref->mSessionid == sessionid) { 1924 ALOGV(" session %d still exists for %d with %d refs", 1925 sessionid, ref->mPid, ref->mCnt); 1926 found = true; 1927 break; 1928 } 1929 } 1930 if (!found) { 1931 Mutex::Autolock _l (t->mLock); 1932 // remove all effects from the chain 1933 while (ec->mEffects.size()) { 1934 sp<EffectModule> effect = ec->mEffects[0]; 1935 effect->unPin(); 1936 t->removeEffect_l(effect); 1937 if (effect->purgeHandles()) { 1938 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1939 } 1940 AudioSystem::unregisterEffect(effect->id()); 1941 } 1942 } 1943 } 1944 return; 1945} 1946 1947// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1948AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1949{ 1950 return mPlaybackThreads.valueFor(output).get(); 1951} 1952 1953// checkMixerThread_l() must be called with AudioFlinger::mLock held 1954AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1955{ 1956 PlaybackThread *thread = checkPlaybackThread_l(output); 1957 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1958} 1959 1960// checkRecordThread_l() must be called with AudioFlinger::mLock held 1961AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1962{ 1963 return mRecordThreads.valueFor(input).get(); 1964} 1965 1966uint32_t AudioFlinger::nextUniqueId() 1967{ 1968 return android_atomic_inc(&mNextUniqueId); 1969} 1970 1971AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1972{ 1973 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1974 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1975 AudioStreamOut *output = thread->getOutput(); 1976 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1977 return thread; 1978 } 1979 } 1980 return NULL; 1981} 1982 1983audio_devices_t AudioFlinger::primaryOutputDevice_l() const 1984{ 1985 PlaybackThread *thread = primaryPlaybackThread_l(); 1986 1987 if (thread == NULL) { 1988 return 0; 1989 } 1990 1991 return thread->outDevice(); 1992} 1993 1994sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 1995 int triggerSession, 1996 int listenerSession, 1997 sync_event_callback_t callBack, 1998 void *cookie) 1999{ 2000 Mutex::Autolock _l(mLock); 2001 2002 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2003 status_t playStatus = NAME_NOT_FOUND; 2004 status_t recStatus = NAME_NOT_FOUND; 2005 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2006 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2007 if (playStatus == NO_ERROR) { 2008 return event; 2009 } 2010 } 2011 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2012 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2013 if (recStatus == NO_ERROR) { 2014 return event; 2015 } 2016 } 2017 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2018 mPendingSyncEvents.add(event); 2019 } else { 2020 ALOGV("createSyncEvent() invalid event %d", event->type()); 2021 event.clear(); 2022 } 2023 return event; 2024} 2025 2026// ---------------------------------------------------------------------------- 2027// Effect management 2028// ---------------------------------------------------------------------------- 2029 2030 2031status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2032{ 2033 Mutex::Autolock _l(mLock); 2034 return EffectQueryNumberEffects(numEffects); 2035} 2036 2037status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2038{ 2039 Mutex::Autolock _l(mLock); 2040 return EffectQueryEffect(index, descriptor); 2041} 2042 2043status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2044 effect_descriptor_t *descriptor) const 2045{ 2046 Mutex::Autolock _l(mLock); 2047 return EffectGetDescriptor(pUuid, descriptor); 2048} 2049 2050 2051sp<IEffect> AudioFlinger::createEffect( 2052 effect_descriptor_t *pDesc, 2053 const sp<IEffectClient>& effectClient, 2054 int32_t priority, 2055 audio_io_handle_t io, 2056 int sessionId, 2057 status_t *status, 2058 int *id, 2059 int *enabled) 2060{ 2061 status_t lStatus = NO_ERROR; 2062 sp<EffectHandle> handle; 2063 effect_descriptor_t desc; 2064 2065 pid_t pid = IPCThreadState::self()->getCallingPid(); 2066 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2067 pid, effectClient.get(), priority, sessionId, io); 2068 2069 if (pDesc == NULL) { 2070 lStatus = BAD_VALUE; 2071 goto Exit; 2072 } 2073 2074 // check audio settings permission for global effects 2075 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2076 lStatus = PERMISSION_DENIED; 2077 goto Exit; 2078 } 2079 2080 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2081 // that can only be created by audio policy manager (running in same process) 2082 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2083 lStatus = PERMISSION_DENIED; 2084 goto Exit; 2085 } 2086 2087 if (io == 0) { 2088 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2089 // output must be specified by AudioPolicyManager when using session 2090 // AUDIO_SESSION_OUTPUT_STAGE 2091 lStatus = BAD_VALUE; 2092 goto Exit; 2093 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2094 // if the output returned by getOutputForEffect() is removed before we lock the 2095 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2096 // and we will exit safely 2097 io = AudioSystem::getOutputForEffect(&desc); 2098 } 2099 } 2100 2101 { 2102 Mutex::Autolock _l(mLock); 2103 2104 2105 if (!EffectIsNullUuid(&pDesc->uuid)) { 2106 // if uuid is specified, request effect descriptor 2107 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2108 if (lStatus < 0) { 2109 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2110 goto Exit; 2111 } 2112 } else { 2113 // if uuid is not specified, look for an available implementation 2114 // of the required type in effect factory 2115 if (EffectIsNullUuid(&pDesc->type)) { 2116 ALOGW("createEffect() no effect type"); 2117 lStatus = BAD_VALUE; 2118 goto Exit; 2119 } 2120 uint32_t numEffects = 0; 2121 effect_descriptor_t d; 2122 d.flags = 0; // prevent compiler warning 2123 bool found = false; 2124 2125 lStatus = EffectQueryNumberEffects(&numEffects); 2126 if (lStatus < 0) { 2127 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2128 goto Exit; 2129 } 2130 for (uint32_t i = 0; i < numEffects; i++) { 2131 lStatus = EffectQueryEffect(i, &desc); 2132 if (lStatus < 0) { 2133 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2134 continue; 2135 } 2136 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2137 // If matching type found save effect descriptor. If the session is 2138 // 0 and the effect is not auxiliary, continue enumeration in case 2139 // an auxiliary version of this effect type is available 2140 found = true; 2141 d = desc; 2142 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2143 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2144 break; 2145 } 2146 } 2147 } 2148 if (!found) { 2149 lStatus = BAD_VALUE; 2150 ALOGW("createEffect() effect not found"); 2151 goto Exit; 2152 } 2153 // For same effect type, chose auxiliary version over insert version if 2154 // connect to output mix (Compliance to OpenSL ES) 2155 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2156 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2157 desc = d; 2158 } 2159 } 2160 2161 // Do not allow auxiliary effects on a session different from 0 (output mix) 2162 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2163 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2164 lStatus = INVALID_OPERATION; 2165 goto Exit; 2166 } 2167 2168 // check recording permission for visualizer 2169 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2170 !recordingAllowed()) { 2171 lStatus = PERMISSION_DENIED; 2172 goto Exit; 2173 } 2174 2175 // return effect descriptor 2176 *pDesc = desc; 2177 2178 // If output is not specified try to find a matching audio session ID in one of the 2179 // output threads. 2180 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2181 // because of code checking output when entering the function. 2182 // Note: io is never 0 when creating an effect on an input 2183 if (io == 0) { 2184 // look for the thread where the specified audio session is present 2185 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2186 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2187 io = mPlaybackThreads.keyAt(i); 2188 break; 2189 } 2190 } 2191 if (io == 0) { 2192 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2193 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2194 io = mRecordThreads.keyAt(i); 2195 break; 2196 } 2197 } 2198 } 2199 // If no output thread contains the requested session ID, default to 2200 // first output. The effect chain will be moved to the correct output 2201 // thread when a track with the same session ID is created 2202 if (io == 0 && mPlaybackThreads.size()) { 2203 io = mPlaybackThreads.keyAt(0); 2204 } 2205 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2206 } 2207 ThreadBase *thread = checkRecordThread_l(io); 2208 if (thread == NULL) { 2209 thread = checkPlaybackThread_l(io); 2210 if (thread == NULL) { 2211 ALOGE("createEffect() unknown output thread"); 2212 lStatus = BAD_VALUE; 2213 goto Exit; 2214 } 2215 } 2216 2217 sp<Client> client = registerPid_l(pid); 2218 2219 // create effect on selected output thread 2220 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2221 &desc, enabled, &lStatus); 2222 if (handle != 0 && id != NULL) { 2223 *id = handle->id(); 2224 } 2225 } 2226 2227Exit: 2228 if (status != NULL) { 2229 *status = lStatus; 2230 } 2231 return handle; 2232} 2233 2234status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2235 audio_io_handle_t dstOutput) 2236{ 2237 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2238 sessionId, srcOutput, dstOutput); 2239 Mutex::Autolock _l(mLock); 2240 if (srcOutput == dstOutput) { 2241 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2242 return NO_ERROR; 2243 } 2244 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2245 if (srcThread == NULL) { 2246 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2247 return BAD_VALUE; 2248 } 2249 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2250 if (dstThread == NULL) { 2251 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2252 return BAD_VALUE; 2253 } 2254 2255 Mutex::Autolock _dl(dstThread->mLock); 2256 Mutex::Autolock _sl(srcThread->mLock); 2257 moveEffectChain_l(sessionId, srcThread, dstThread, false); 2258 2259 return NO_ERROR; 2260} 2261 2262// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2263status_t AudioFlinger::moveEffectChain_l(int sessionId, 2264 AudioFlinger::PlaybackThread *srcThread, 2265 AudioFlinger::PlaybackThread *dstThread, 2266 bool reRegister) 2267{ 2268 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2269 sessionId, srcThread, dstThread); 2270 2271 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2272 if (chain == 0) { 2273 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2274 sessionId, srcThread); 2275 return INVALID_OPERATION; 2276 } 2277 2278 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2279 // so that a new chain is created with correct parameters when first effect is added. This is 2280 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2281 // removed. 2282 srcThread->removeEffectChain_l(chain); 2283 2284 // transfer all effects one by one so that new effect chain is created on new thread with 2285 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2286 audio_io_handle_t dstOutput = dstThread->id(); 2287 sp<EffectChain> dstChain; 2288 uint32_t strategy = 0; // prevent compiler warning 2289 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2290 while (effect != 0) { 2291 srcThread->removeEffect_l(effect); 2292 dstThread->addEffect_l(effect); 2293 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2294 if (effect->state() == EffectModule::ACTIVE || 2295 effect->state() == EffectModule::STOPPING) { 2296 effect->start(); 2297 } 2298 // if the move request is not received from audio policy manager, the effect must be 2299 // re-registered with the new strategy and output 2300 if (dstChain == 0) { 2301 dstChain = effect->chain().promote(); 2302 if (dstChain == 0) { 2303 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2304 srcThread->addEffect_l(effect); 2305 return NO_INIT; 2306 } 2307 strategy = dstChain->strategy(); 2308 } 2309 if (reRegister) { 2310 AudioSystem::unregisterEffect(effect->id()); 2311 AudioSystem::registerEffect(&effect->desc(), 2312 dstOutput, 2313 strategy, 2314 sessionId, 2315 effect->id()); 2316 } 2317 effect = chain->getEffectFromId_l(0); 2318 } 2319 2320 return NO_ERROR; 2321} 2322 2323bool AudioFlinger::isGlobalEffectEnabled_l() 2324{ 2325 if (mGlobalEffectEnableTime != 0 && 2326 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2327 return true; 2328 } 2329 2330 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2331 sp<EffectChain> ec = 2332 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2333 if (ec != 0 && ec->isEnabled()) { 2334 return true; 2335 } 2336 } 2337 return false; 2338} 2339 2340void AudioFlinger::onGlobalEffectEnable() 2341{ 2342 Mutex::Autolock _l(mLock); 2343 2344 mGlobalEffectEnableTime = systemTime(); 2345 2346 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2347 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2348 if (t->mType == ThreadBase::OFFLOAD) { 2349 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2350 } 2351 } 2352 2353} 2354 2355struct Entry { 2356#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2357 char mName[MAX_NAME]; 2358}; 2359 2360int comparEntry(const void *p1, const void *p2) 2361{ 2362 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2363} 2364 2365#ifdef TEE_SINK 2366void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2367{ 2368 NBAIO_Source *teeSource = source.get(); 2369 if (teeSource != NULL) { 2370 // .wav rotation 2371 // There is a benign race condition if 2 threads call this simultaneously. 2372 // They would both traverse the directory, but the result would simply be 2373 // failures at unlink() which are ignored. It's also unlikely since 2374 // normally dumpsys is only done by bugreport or from the command line. 2375 char teePath[32+256]; 2376 strcpy(teePath, "/data/misc/media"); 2377 size_t teePathLen = strlen(teePath); 2378 DIR *dir = opendir(teePath); 2379 teePath[teePathLen++] = '/'; 2380 if (dir != NULL) { 2381#define MAX_SORT 20 // number of entries to sort 2382#define MAX_KEEP 10 // number of entries to keep 2383 struct Entry entries[MAX_SORT]; 2384 size_t entryCount = 0; 2385 while (entryCount < MAX_SORT) { 2386 struct dirent de; 2387 struct dirent *result = NULL; 2388 int rc = readdir_r(dir, &de, &result); 2389 if (rc != 0) { 2390 ALOGW("readdir_r failed %d", rc); 2391 break; 2392 } 2393 if (result == NULL) { 2394 break; 2395 } 2396 if (result != &de) { 2397 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2398 break; 2399 } 2400 // ignore non .wav file entries 2401 size_t nameLen = strlen(de.d_name); 2402 if (nameLen <= 4 || nameLen >= MAX_NAME || 2403 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2404 continue; 2405 } 2406 strcpy(entries[entryCount++].mName, de.d_name); 2407 } 2408 (void) closedir(dir); 2409 if (entryCount > MAX_KEEP) { 2410 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2411 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2412 strcpy(&teePath[teePathLen], entries[i].mName); 2413 (void) unlink(teePath); 2414 } 2415 } 2416 } else { 2417 if (fd >= 0) { 2418 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2419 } 2420 } 2421 char teeTime[16]; 2422 struct timeval tv; 2423 gettimeofday(&tv, NULL); 2424 struct tm tm; 2425 localtime_r(&tv.tv_sec, &tm); 2426 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2427 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2428 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2429 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2430 if (teeFd >= 0) { 2431 char wavHeader[44]; 2432 memcpy(wavHeader, 2433 "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", 2434 sizeof(wavHeader)); 2435 NBAIO_Format format = teeSource->format(); 2436 unsigned channelCount = Format_channelCount(format); 2437 ALOG_ASSERT(channelCount <= FCC_2); 2438 uint32_t sampleRate = Format_sampleRate(format); 2439 wavHeader[22] = channelCount; // number of channels 2440 wavHeader[24] = sampleRate; // sample rate 2441 wavHeader[25] = sampleRate >> 8; 2442 wavHeader[32] = channelCount * 2; // block alignment 2443 write(teeFd, wavHeader, sizeof(wavHeader)); 2444 size_t total = 0; 2445 bool firstRead = true; 2446 for (;;) { 2447#define TEE_SINK_READ 1024 2448 short buffer[TEE_SINK_READ * FCC_2]; 2449 size_t count = TEE_SINK_READ; 2450 ssize_t actual = teeSource->read(buffer, count, 2451 AudioBufferProvider::kInvalidPTS); 2452 bool wasFirstRead = firstRead; 2453 firstRead = false; 2454 if (actual <= 0) { 2455 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2456 continue; 2457 } 2458 break; 2459 } 2460 ALOG_ASSERT(actual <= (ssize_t)count); 2461 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2462 total += actual; 2463 } 2464 lseek(teeFd, (off_t) 4, SEEK_SET); 2465 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2466 write(teeFd, &temp, sizeof(temp)); 2467 lseek(teeFd, (off_t) 40, SEEK_SET); 2468 temp = total * channelCount * sizeof(short); 2469 write(teeFd, &temp, sizeof(temp)); 2470 close(teeFd); 2471 if (fd >= 0) { 2472 fdprintf(fd, "tee copied to %s\n", teePath); 2473 } 2474 } else { 2475 if (fd >= 0) { 2476 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2477 } 2478 } 2479 } 2480} 2481#endif 2482 2483// ---------------------------------------------------------------------------- 2484 2485status_t AudioFlinger::onTransact( 2486 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2487{ 2488 return BnAudioFlinger::onTransact(code, data, reply, flags); 2489} 2490 2491}; // namespace android 2492