AudioFlinger.cpp revision 030033342a6ea17003e6af38a56c7edc6d2ead01
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// ---------------------------------------------------------------------------- 102 103static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 104{ 105 const hw_module_t *mod; 106 int rc; 107 108 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 109 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 110 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 111 if (rc) { 112 goto out; 113 } 114 rc = audio_hw_device_open(mod, dev); 115 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 116 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 117 if (rc) { 118 goto out; 119 } 120 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 121 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 122 rc = BAD_VALUE; 123 goto out; 124 } 125 return 0; 126 127out: 128 *dev = NULL; 129 return rc; 130} 131 132// ---------------------------------------------------------------------------- 133 134AudioFlinger::AudioFlinger() 135 : BnAudioFlinger(), 136 mPrimaryHardwareDev(NULL), 137 mHardwareStatus(AUDIO_HW_IDLE), 138 mMasterVolume(1.0f), 139 mMasterMute(false), 140 mNextUniqueId(1), 141 mMode(AUDIO_MODE_INVALID), 142 mBtNrecIsOff(false), 143 mIsLowRamDevice(true), 144 mIsDeviceTypeKnown(false) 145{ 146 getpid_cached = getpid(); 147 char value[PROPERTY_VALUE_MAX]; 148 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 149 if (doLog) { 150 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters"); 151 } 152#ifdef TEE_SINK 153 (void) property_get("ro.debuggable", value, "0"); 154 int debuggable = atoi(value); 155 int teeEnabled = 0; 156 if (debuggable) { 157 (void) property_get("af.tee", value, "0"); 158 teeEnabled = atoi(value); 159 } 160 if (teeEnabled & 1) 161 mTeeSinkInputEnabled = true; 162 if (teeEnabled & 2) 163 mTeeSinkOutputEnabled = true; 164 if (teeEnabled & 4) 165 mTeeSinkTrackEnabled = true; 166#endif 167} 168 169void AudioFlinger::onFirstRef() 170{ 171 int rc = 0; 172 173 Mutex::Autolock _l(mLock); 174 175 /* TODO: move all this work into an Init() function */ 176 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 177 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 178 uint32_t int_val; 179 if (1 == sscanf(val_str, "%u", &int_val)) { 180 mStandbyTimeInNsecs = milliseconds(int_val); 181 ALOGI("Using %u mSec as standby time.", int_val); 182 } else { 183 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 184 ALOGI("Using default %u mSec as standby time.", 185 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 186 } 187 } 188 189 mMode = AUDIO_MODE_NORMAL; 190} 191 192AudioFlinger::~AudioFlinger() 193{ 194 while (!mRecordThreads.isEmpty()) { 195 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 196 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 197 } 198 while (!mPlaybackThreads.isEmpty()) { 199 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 200 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 201 } 202 203 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 204 // no mHardwareLock needed, as there are no other references to this 205 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 206 delete mAudioHwDevs.valueAt(i); 207 } 208} 209 210static const char * const audio_interfaces[] = { 211 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 212 AUDIO_HARDWARE_MODULE_ID_A2DP, 213 AUDIO_HARDWARE_MODULE_ID_USB, 214}; 215#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 216 217AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 218 audio_module_handle_t module, 219 audio_devices_t devices) 220{ 221 // if module is 0, the request comes from an old policy manager and we should load 222 // well known modules 223 if (module == 0) { 224 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 225 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 226 loadHwModule_l(audio_interfaces[i]); 227 } 228 // then try to find a module supporting the requested device. 229 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 230 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 231 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 232 if ((dev->get_supported_devices != NULL) && 233 (dev->get_supported_devices(dev) & devices) == devices) 234 return audioHwDevice; 235 } 236 } else { 237 // check a match for the requested module handle 238 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 239 if (audioHwDevice != NULL) { 240 return audioHwDevice; 241 } 242 } 243 244 return NULL; 245} 246 247void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 248{ 249 const size_t SIZE = 256; 250 char buffer[SIZE]; 251 String8 result; 252 253 result.append("Clients:\n"); 254 for (size_t i = 0; i < mClients.size(); ++i) { 255 sp<Client> client = mClients.valueAt(i).promote(); 256 if (client != 0) { 257 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 258 result.append(buffer); 259 } 260 } 261 262 result.append("Global session refs:\n"); 263 result.append(" session pid count\n"); 264 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 265 AudioSessionRef *r = mAudioSessionRefs[i]; 266 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 267 result.append(buffer); 268 } 269 write(fd, result.string(), result.size()); 270} 271 272 273void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 274{ 275 const size_t SIZE = 256; 276 char buffer[SIZE]; 277 String8 result; 278 hardware_call_state hardwareStatus = mHardwareStatus; 279 280 snprintf(buffer, SIZE, "Hardware status: %d\n" 281 "Standby Time mSec: %u\n", 282 hardwareStatus, 283 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 284 result.append(buffer); 285 write(fd, result.string(), result.size()); 286} 287 288void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 289{ 290 const size_t SIZE = 256; 291 char buffer[SIZE]; 292 String8 result; 293 snprintf(buffer, SIZE, "Permission Denial: " 294 "can't dump AudioFlinger from pid=%d, uid=%d\n", 295 IPCThreadState::self()->getCallingPid(), 296 IPCThreadState::self()->getCallingUid()); 297 result.append(buffer); 298 write(fd, result.string(), result.size()); 299} 300 301bool AudioFlinger::dumpTryLock(Mutex& mutex) 302{ 303 bool locked = false; 304 for (int i = 0; i < kDumpLockRetries; ++i) { 305 if (mutex.tryLock() == NO_ERROR) { 306 locked = true; 307 break; 308 } 309 usleep(kDumpLockSleepUs); 310 } 311 return locked; 312} 313 314status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 315{ 316 if (!dumpAllowed()) { 317 dumpPermissionDenial(fd, args); 318 } else { 319 // get state of hardware lock 320 bool hardwareLocked = dumpTryLock(mHardwareLock); 321 if (!hardwareLocked) { 322 String8 result(kHardwareLockedString); 323 write(fd, result.string(), result.size()); 324 } else { 325 mHardwareLock.unlock(); 326 } 327 328 bool locked = dumpTryLock(mLock); 329 330 // failed to lock - AudioFlinger is probably deadlocked 331 if (!locked) { 332 String8 result(kDeadlockedString); 333 write(fd, result.string(), result.size()); 334 } 335 336 dumpClients(fd, args); 337 dumpInternals(fd, args); 338 339 // dump playback threads 340 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 341 mPlaybackThreads.valueAt(i)->dump(fd, args); 342 } 343 344 // dump record threads 345 for (size_t i = 0; i < mRecordThreads.size(); i++) { 346 mRecordThreads.valueAt(i)->dump(fd, args); 347 } 348 349 // dump all hardware devs 350 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 351 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 352 dev->dump(dev, fd); 353 } 354 355#ifdef TEE_SINK 356 // dump the serially shared record tee sink 357 if (mRecordTeeSource != 0) { 358 dumpTee(fd, mRecordTeeSource); 359 } 360#endif 361 362 if (locked) { 363 mLock.unlock(); 364 } 365 366 // append a copy of media.log here by forwarding fd to it, but don't attempt 367 // to lookup the service if it's not running, as it will block for a second 368 if (mLogMemoryDealer != 0) { 369 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 370 if (binder != 0) { 371 fdprintf(fd, "\nmedia.log:\n"); 372 Vector<String16> args; 373 binder->dump(fd, args); 374 } 375 } 376 } 377 return NO_ERROR; 378} 379 380sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 381{ 382 // If pid is already in the mClients wp<> map, then use that entry 383 // (for which promote() is always != 0), otherwise create a new entry and Client. 384 sp<Client> client = mClients.valueFor(pid).promote(); 385 if (client == 0) { 386 client = new Client(this, pid); 387 mClients.add(pid, client); 388 } 389 390 return client; 391} 392 393sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 394{ 395 if (mLogMemoryDealer == 0) { 396 return new NBLog::Writer(); 397 } 398 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 399 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 400 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 401 if (binder != 0) { 402 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 403 } 404 return writer; 405} 406 407void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 408{ 409 if (writer == 0) { 410 return; 411 } 412 sp<IMemory> iMemory(writer->getIMemory()); 413 if (iMemory == 0) { 414 return; 415 } 416 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 417 if (binder != 0) { 418 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 419 // Now the media.log remote reference to IMemory is gone. 420 // When our last local reference to IMemory also drops to zero, 421 // the IMemory destructor will deallocate the region from mMemoryDealer. 422 } 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 audio_stream_type_t streamType, 430 uint32_t sampleRate, 431 audio_format_t format, 432 audio_channel_mask_t channelMask, 433 size_t frameCount, 434 IAudioFlinger::track_flags_t *flags, 435 const sp<IMemory>& sharedBuffer, 436 audio_io_handle_t output, 437 pid_t tid, 438 int *sessionId, 439 String8& name, 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 // s for server's pid, n for normal mixer name, f for fast index 529 name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, 530 track->fastIndex()); 531 trackHandle = new TrackHandle(track); 532 } else { 533 // remove local strong reference to Client before deleting the Track so that the Client 534 // destructor is called by the TrackBase destructor with mLock held 535 client.clear(); 536 track.clear(); 537 } 538 539Exit: 540 *status = lStatus; 541 return trackHandle; 542} 543 544uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 545{ 546 Mutex::Autolock _l(mLock); 547 PlaybackThread *thread = checkPlaybackThread_l(output); 548 if (thread == NULL) { 549 ALOGW("sampleRate() unknown thread %d", output); 550 return 0; 551 } 552 return thread->sampleRate(); 553} 554 555int AudioFlinger::channelCount(audio_io_handle_t output) const 556{ 557 Mutex::Autolock _l(mLock); 558 PlaybackThread *thread = checkPlaybackThread_l(output); 559 if (thread == NULL) { 560 ALOGW("channelCount() unknown thread %d", output); 561 return 0; 562 } 563 return thread->channelCount(); 564} 565 566audio_format_t AudioFlinger::format(audio_io_handle_t output) const 567{ 568 Mutex::Autolock _l(mLock); 569 PlaybackThread *thread = checkPlaybackThread_l(output); 570 if (thread == NULL) { 571 ALOGW("format() unknown thread %d", output); 572 return AUDIO_FORMAT_INVALID; 573 } 574 return thread->format(); 575} 576 577size_t AudioFlinger::frameCount(audio_io_handle_t output) const 578{ 579 Mutex::Autolock _l(mLock); 580 PlaybackThread *thread = checkPlaybackThread_l(output); 581 if (thread == NULL) { 582 ALOGW("frameCount() unknown thread %d", output); 583 return 0; 584 } 585 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 586 // should examine all callers and fix them to handle smaller counts 587 return thread->frameCount(); 588} 589 590uint32_t AudioFlinger::latency(audio_io_handle_t output) const 591{ 592 Mutex::Autolock _l(mLock); 593 PlaybackThread *thread = checkPlaybackThread_l(output); 594 if (thread == NULL) { 595 ALOGW("latency(): no playback thread found for output handle %d", output); 596 return 0; 597 } 598 return thread->latency(); 599} 600 601status_t AudioFlinger::setMasterVolume(float value) 602{ 603 status_t ret = initCheck(); 604 if (ret != NO_ERROR) { 605 return ret; 606 } 607 608 // check calling permissions 609 if (!settingsAllowed()) { 610 return PERMISSION_DENIED; 611 } 612 613 Mutex::Autolock _l(mLock); 614 mMasterVolume = value; 615 616 // Set master volume in the HALs which support it. 617 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 618 AutoMutex lock(mHardwareLock); 619 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 620 621 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 622 if (dev->canSetMasterVolume()) { 623 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 624 } 625 mHardwareStatus = AUDIO_HW_IDLE; 626 } 627 628 // Now set the master volume in each playback thread. Playback threads 629 // assigned to HALs which do not have master volume support will apply 630 // master volume during the mix operation. Threads with HALs which do 631 // support master volume will simply ignore the setting. 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 657 mHardwareStatus = AUDIO_HW_SET_MODE; 658 ret = dev->set_mode(dev, mode); 659 mHardwareStatus = AUDIO_HW_IDLE; 660 } 661 662 if (NO_ERROR == ret) { 663 Mutex::Autolock _l(mLock); 664 mMode = mode; 665 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 666 mPlaybackThreads.valueAt(i)->setMode(mode); 667 } 668 669 return ret; 670} 671 672status_t AudioFlinger::setMicMute(bool state) 673{ 674 status_t ret = initCheck(); 675 if (ret != NO_ERROR) { 676 return ret; 677 } 678 679 // check calling permissions 680 if (!settingsAllowed()) { 681 return PERMISSION_DENIED; 682 } 683 684 AutoMutex lock(mHardwareLock); 685 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 686 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 687 ret = dev->set_mic_mute(dev, state); 688 mHardwareStatus = AUDIO_HW_IDLE; 689 return ret; 690} 691 692bool AudioFlinger::getMicMute() const 693{ 694 status_t ret = initCheck(); 695 if (ret != NO_ERROR) { 696 return false; 697 } 698 699 bool state = AUDIO_MODE_INVALID; 700 AutoMutex lock(mHardwareLock); 701 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 702 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 703 dev->get_mic_mute(dev, &state); 704 mHardwareStatus = AUDIO_HW_IDLE; 705 return state; 706} 707 708status_t AudioFlinger::setMasterMute(bool muted) 709{ 710 status_t ret = initCheck(); 711 if (ret != NO_ERROR) { 712 return ret; 713 } 714 715 // check calling permissions 716 if (!settingsAllowed()) { 717 return PERMISSION_DENIED; 718 } 719 720 Mutex::Autolock _l(mLock); 721 mMasterMute = muted; 722 723 // Set master mute in the HALs which support it. 724 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 725 AutoMutex lock(mHardwareLock); 726 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 727 728 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 729 if (dev->canSetMasterMute()) { 730 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 731 } 732 mHardwareStatus = AUDIO_HW_IDLE; 733 } 734 735 // Now set the master mute in each playback thread. Playback threads 736 // assigned to HALs which do not have master mute support will apply master 737 // mute during the mix operation. Threads with HALs which do support master 738 // mute will simply ignore the setting. 739 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 740 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 741 742 return NO_ERROR; 743} 744 745float AudioFlinger::masterVolume() const 746{ 747 Mutex::Autolock _l(mLock); 748 return masterVolume_l(); 749} 750 751bool AudioFlinger::masterMute() const 752{ 753 Mutex::Autolock _l(mLock); 754 return masterMute_l(); 755} 756 757float AudioFlinger::masterVolume_l() const 758{ 759 return mMasterVolume; 760} 761 762bool AudioFlinger::masterMute_l() const 763{ 764 return mMasterMute; 765} 766 767status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 768 audio_io_handle_t output) 769{ 770 // check calling permissions 771 if (!settingsAllowed()) { 772 return PERMISSION_DENIED; 773 } 774 775 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 776 ALOGE("setStreamVolume() invalid stream %d", stream); 777 return BAD_VALUE; 778 } 779 780 AutoMutex lock(mLock); 781 PlaybackThread *thread = NULL; 782 if (output) { 783 thread = checkPlaybackThread_l(output); 784 if (thread == NULL) { 785 return BAD_VALUE; 786 } 787 } 788 789 mStreamTypes[stream].volume = value; 790 791 if (thread == NULL) { 792 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 793 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 794 } 795 } else { 796 thread->setStreamVolume(stream, value); 797 } 798 799 return NO_ERROR; 800} 801 802status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 803{ 804 // check calling permissions 805 if (!settingsAllowed()) { 806 return PERMISSION_DENIED; 807 } 808 809 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 810 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 811 ALOGE("setStreamMute() invalid stream %d", stream); 812 return BAD_VALUE; 813 } 814 815 AutoMutex lock(mLock); 816 mStreamTypes[stream].mute = muted; 817 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 818 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 819 820 return NO_ERROR; 821} 822 823float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 824{ 825 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 826 return 0.0f; 827 } 828 829 AutoMutex lock(mLock); 830 float volume; 831 if (output) { 832 PlaybackThread *thread = checkPlaybackThread_l(output); 833 if (thread == NULL) { 834 return 0.0f; 835 } 836 volume = thread->streamVolume(stream); 837 } else { 838 volume = streamVolume_l(stream); 839 } 840 841 return volume; 842} 843 844bool AudioFlinger::streamMute(audio_stream_type_t stream) const 845{ 846 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 847 return true; 848 } 849 850 AutoMutex lock(mLock); 851 return streamMute_l(stream); 852} 853 854status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 855{ 856 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 857 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 858 859 // check calling permissions 860 if (!settingsAllowed()) { 861 return PERMISSION_DENIED; 862 } 863 864 // ioHandle == 0 means the parameters are global to the audio hardware interface 865 if (ioHandle == 0) { 866 Mutex::Autolock _l(mLock); 867 status_t final_result = NO_ERROR; 868 { 869 AutoMutex lock(mHardwareLock); 870 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 871 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 872 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 873 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 874 final_result = result ?: final_result; 875 } 876 mHardwareStatus = AUDIO_HW_IDLE; 877 } 878 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 879 AudioParameter param = AudioParameter(keyValuePairs); 880 String8 value; 881 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 882 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 883 if (mBtNrecIsOff != btNrecIsOff) { 884 for (size_t i = 0; i < mRecordThreads.size(); i++) { 885 sp<RecordThread> thread = mRecordThreads.valueAt(i); 886 audio_devices_t device = thread->inDevice(); 887 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 888 // collect all of the thread's session IDs 889 KeyedVector<int, bool> ids = thread->sessionIds(); 890 // suspend effects associated with those session IDs 891 for (size_t j = 0; j < ids.size(); ++j) { 892 int sessionId = ids.keyAt(j); 893 thread->setEffectSuspended(FX_IID_AEC, 894 suspend, 895 sessionId); 896 thread->setEffectSuspended(FX_IID_NS, 897 suspend, 898 sessionId); 899 } 900 } 901 mBtNrecIsOff = btNrecIsOff; 902 } 903 } 904 String8 screenState; 905 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 906 bool isOff = screenState == "off"; 907 if (isOff != (AudioFlinger::mScreenState & 1)) { 908 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 909 } 910 } 911 return final_result; 912 } 913 914 // hold a strong ref on thread in case closeOutput() or closeInput() is called 915 // and the thread is exited once the lock is released 916 sp<ThreadBase> thread; 917 { 918 Mutex::Autolock _l(mLock); 919 thread = checkPlaybackThread_l(ioHandle); 920 if (thread == 0) { 921 thread = checkRecordThread_l(ioHandle); 922 } else if (thread == primaryPlaybackThread_l()) { 923 // indicate output device change to all input threads for pre processing 924 AudioParameter param = AudioParameter(keyValuePairs); 925 int value; 926 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 927 (value != 0)) { 928 for (size_t i = 0; i < mRecordThreads.size(); i++) { 929 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 930 } 931 } 932 } 933 } 934 if (thread != 0) { 935 return thread->setParameters(keyValuePairs); 936 } 937 return BAD_VALUE; 938} 939 940String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 941{ 942 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 943 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 944 945 Mutex::Autolock _l(mLock); 946 947 if (ioHandle == 0) { 948 String8 out_s8; 949 950 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 951 char *s; 952 { 953 AutoMutex lock(mHardwareLock); 954 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 955 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 956 s = dev->get_parameters(dev, keys.string()); 957 mHardwareStatus = AUDIO_HW_IDLE; 958 } 959 out_s8 += String8(s ? s : ""); 960 free(s); 961 } 962 return out_s8; 963 } 964 965 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 966 if (playbackThread != NULL) { 967 return playbackThread->getParameters(keys); 968 } 969 RecordThread *recordThread = checkRecordThread_l(ioHandle); 970 if (recordThread != NULL) { 971 return recordThread->getParameters(keys); 972 } 973 return String8(""); 974} 975 976size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 977 audio_channel_mask_t channelMask) const 978{ 979 status_t ret = initCheck(); 980 if (ret != NO_ERROR) { 981 return 0; 982 } 983 984 AutoMutex lock(mHardwareLock); 985 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 986 struct audio_config config; 987 memset(&config, 0, sizeof(config)); 988 config.sample_rate = sampleRate; 989 config.channel_mask = channelMask; 990 config.format = format; 991 992 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 993 size_t size = dev->get_input_buffer_size(dev, &config); 994 mHardwareStatus = AUDIO_HW_IDLE; 995 return size; 996} 997 998unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 999{ 1000 Mutex::Autolock _l(mLock); 1001 1002 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1003 if (recordThread != NULL) { 1004 return recordThread->getInputFramesLost(); 1005 } 1006 return 0; 1007} 1008 1009status_t AudioFlinger::setVoiceVolume(float value) 1010{ 1011 status_t ret = initCheck(); 1012 if (ret != NO_ERROR) { 1013 return ret; 1014 } 1015 1016 // check calling permissions 1017 if (!settingsAllowed()) { 1018 return PERMISSION_DENIED; 1019 } 1020 1021 AutoMutex lock(mHardwareLock); 1022 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1023 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1024 ret = dev->set_voice_volume(dev, value); 1025 mHardwareStatus = AUDIO_HW_IDLE; 1026 1027 return ret; 1028} 1029 1030status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1031 audio_io_handle_t output) const 1032{ 1033 status_t status; 1034 1035 Mutex::Autolock _l(mLock); 1036 1037 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1038 if (playbackThread != NULL) { 1039 return playbackThread->getRenderPosition(halFrames, dspFrames); 1040 } 1041 1042 return BAD_VALUE; 1043} 1044 1045void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1046{ 1047 1048 Mutex::Autolock _l(mLock); 1049 1050 pid_t pid = IPCThreadState::self()->getCallingPid(); 1051 if (mNotificationClients.indexOfKey(pid) < 0) { 1052 sp<NotificationClient> notificationClient = new NotificationClient(this, 1053 client, 1054 pid); 1055 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1056 1057 mNotificationClients.add(pid, notificationClient); 1058 1059 sp<IBinder> binder = client->asBinder(); 1060 binder->linkToDeath(notificationClient); 1061 1062 // the config change is always sent from playback or record threads to avoid deadlock 1063 // with AudioSystem::gLock 1064 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1065 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1066 } 1067 1068 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1069 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1070 } 1071 } 1072} 1073 1074void AudioFlinger::removeNotificationClient(pid_t pid) 1075{ 1076 Mutex::Autolock _l(mLock); 1077 1078 mNotificationClients.removeItem(pid); 1079 1080 ALOGV("%d died, releasing its sessions", pid); 1081 size_t num = mAudioSessionRefs.size(); 1082 bool removed = false; 1083 for (size_t i = 0; i< num; ) { 1084 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1085 ALOGV(" pid %d @ %d", ref->mPid, i); 1086 if (ref->mPid == pid) { 1087 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1088 mAudioSessionRefs.removeAt(i); 1089 delete ref; 1090 removed = true; 1091 num--; 1092 } else { 1093 i++; 1094 } 1095 } 1096 if (removed) { 1097 purgeStaleEffects_l(); 1098 } 1099} 1100 1101// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1102void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1103{ 1104 size_t size = mNotificationClients.size(); 1105 for (size_t i = 0; i < size; i++) { 1106 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1107 param2); 1108 } 1109} 1110 1111// removeClient_l() must be called with AudioFlinger::mLock held 1112void AudioFlinger::removeClient_l(pid_t pid) 1113{ 1114 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1115 IPCThreadState::self()->getCallingPid()); 1116 mClients.removeItem(pid); 1117} 1118 1119// getEffectThread_l() must be called with AudioFlinger::mLock held 1120sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1121{ 1122 sp<PlaybackThread> thread; 1123 1124 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1125 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1126 ALOG_ASSERT(thread == 0); 1127 thread = mPlaybackThreads.valueAt(i); 1128 } 1129 } 1130 1131 return thread; 1132} 1133 1134 1135 1136// ---------------------------------------------------------------------------- 1137 1138AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1139 : RefBase(), 1140 mAudioFlinger(audioFlinger), 1141 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1142 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1143 mPid(pid), 1144 mTimedTrackCount(0) 1145{ 1146 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1147} 1148 1149// Client destructor must be called with AudioFlinger::mLock held 1150AudioFlinger::Client::~Client() 1151{ 1152 mAudioFlinger->removeClient_l(mPid); 1153} 1154 1155sp<MemoryDealer> AudioFlinger::Client::heap() const 1156{ 1157 return mMemoryDealer; 1158} 1159 1160// Reserve one of the limited slots for a timed audio track associated 1161// with this client 1162bool AudioFlinger::Client::reserveTimedTrack() 1163{ 1164 const int kMaxTimedTracksPerClient = 4; 1165 1166 Mutex::Autolock _l(mTimedTrackLock); 1167 1168 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1169 ALOGW("can not create timed track - pid %d has exceeded the limit", 1170 mPid); 1171 return false; 1172 } 1173 1174 mTimedTrackCount++; 1175 return true; 1176} 1177 1178// Release a slot for a timed audio track 1179void AudioFlinger::Client::releaseTimedTrack() 1180{ 1181 Mutex::Autolock _l(mTimedTrackLock); 1182 mTimedTrackCount--; 1183} 1184 1185// ---------------------------------------------------------------------------- 1186 1187AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1188 const sp<IAudioFlingerClient>& client, 1189 pid_t pid) 1190 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1191{ 1192} 1193 1194AudioFlinger::NotificationClient::~NotificationClient() 1195{ 1196} 1197 1198void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1199{ 1200 sp<NotificationClient> keep(this); 1201 mAudioFlinger->removeNotificationClient(mPid); 1202} 1203 1204 1205// ---------------------------------------------------------------------------- 1206 1207sp<IAudioRecord> AudioFlinger::openRecord( 1208 audio_io_handle_t input, 1209 uint32_t sampleRate, 1210 audio_format_t format, 1211 audio_channel_mask_t channelMask, 1212 size_t frameCount, 1213 IAudioFlinger::track_flags_t *flags, 1214 pid_t tid, 1215 int *sessionId, 1216 status_t *status) 1217{ 1218 sp<RecordThread::RecordTrack> recordTrack; 1219 sp<RecordHandle> recordHandle; 1220 sp<Client> client; 1221 status_t lStatus; 1222 RecordThread *thread; 1223 size_t inFrameCount; 1224 int lSessionId; 1225 1226 // check calling permissions 1227 if (!recordingAllowed()) { 1228 lStatus = PERMISSION_DENIED; 1229 goto Exit; 1230 } 1231 1232 if (format != AUDIO_FORMAT_PCM_16_BIT) { 1233 ALOGE("openRecord() invalid format %d", format); 1234 lStatus = BAD_VALUE; 1235 goto Exit; 1236 } 1237 1238 // add client to list 1239 { // scope for mLock 1240 Mutex::Autolock _l(mLock); 1241 thread = checkRecordThread_l(input); 1242 if (thread == NULL) { 1243 lStatus = BAD_VALUE; 1244 goto Exit; 1245 } 1246 1247 pid_t pid = IPCThreadState::self()->getCallingPid(); 1248 client = registerPid_l(pid); 1249 1250 // If no audio session id is provided, create one here 1251 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1252 lSessionId = *sessionId; 1253 } else { 1254 lSessionId = nextUniqueId(); 1255 if (sessionId != NULL) { 1256 *sessionId = lSessionId; 1257 } 1258 } 1259 // create new record track. 1260 // The record track uses one track in mHardwareMixerThread by convention. 1261 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1262 frameCount, lSessionId, flags, tid, &lStatus); 1263 } 1264 if (lStatus != NO_ERROR) { 1265 // remove local strong reference to Client before deleting the RecordTrack so that the 1266 // Client destructor is called by the TrackBase destructor with mLock held 1267 client.clear(); 1268 recordTrack.clear(); 1269 goto Exit; 1270 } 1271 1272 // return to handle to client 1273 recordHandle = new RecordHandle(recordTrack); 1274 1275Exit: 1276 *status = lStatus; 1277 return recordHandle; 1278} 1279 1280 1281 1282// ---------------------------------------------------------------------------- 1283 1284audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1285{ 1286 if (!settingsAllowed()) { 1287 return 0; 1288 } 1289 Mutex::Autolock _l(mLock); 1290 return loadHwModule_l(name); 1291} 1292 1293// loadHwModule_l() must be called with AudioFlinger::mLock held 1294audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1295{ 1296 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1297 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1298 ALOGW("loadHwModule() module %s already loaded", name); 1299 return mAudioHwDevs.keyAt(i); 1300 } 1301 } 1302 1303 audio_hw_device_t *dev; 1304 1305 int rc = load_audio_interface(name, &dev); 1306 if (rc) { 1307 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1308 return 0; 1309 } 1310 1311 mHardwareStatus = AUDIO_HW_INIT; 1312 rc = dev->init_check(dev); 1313 mHardwareStatus = AUDIO_HW_IDLE; 1314 if (rc) { 1315 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1316 return 0; 1317 } 1318 1319 // Check and cache this HAL's level of support for master mute and master 1320 // volume. If this is the first HAL opened, and it supports the get 1321 // methods, use the initial values provided by the HAL as the current 1322 // master mute and volume settings. 1323 1324 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1325 { // scope for auto-lock pattern 1326 AutoMutex lock(mHardwareLock); 1327 1328 if (0 == mAudioHwDevs.size()) { 1329 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1330 if (NULL != dev->get_master_volume) { 1331 float mv; 1332 if (OK == dev->get_master_volume(dev, &mv)) { 1333 mMasterVolume = mv; 1334 } 1335 } 1336 1337 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1338 if (NULL != dev->get_master_mute) { 1339 bool mm; 1340 if (OK == dev->get_master_mute(dev, &mm)) { 1341 mMasterMute = mm; 1342 } 1343 } 1344 } 1345 1346 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1347 if ((NULL != dev->set_master_volume) && 1348 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1349 flags = static_cast<AudioHwDevice::Flags>(flags | 1350 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1351 } 1352 1353 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1354 if ((NULL != dev->set_master_mute) && 1355 (OK == dev->set_master_mute(dev, mMasterMute))) { 1356 flags = static_cast<AudioHwDevice::Flags>(flags | 1357 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1358 } 1359 1360 mHardwareStatus = AUDIO_HW_IDLE; 1361 } 1362 1363 audio_module_handle_t handle = nextUniqueId(); 1364 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1365 1366 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1367 name, dev->common.module->name, dev->common.module->id, handle); 1368 1369 return handle; 1370 1371} 1372 1373// ---------------------------------------------------------------------------- 1374 1375uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1376{ 1377 Mutex::Autolock _l(mLock); 1378 PlaybackThread *thread = primaryPlaybackThread_l(); 1379 return thread != NULL ? thread->sampleRate() : 0; 1380} 1381 1382size_t AudioFlinger::getPrimaryOutputFrameCount() 1383{ 1384 Mutex::Autolock _l(mLock); 1385 PlaybackThread *thread = primaryPlaybackThread_l(); 1386 return thread != NULL ? thread->frameCountHAL() : 0; 1387} 1388 1389// ---------------------------------------------------------------------------- 1390 1391status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1392{ 1393 uid_t uid = IPCThreadState::self()->getCallingUid(); 1394 if (uid != AID_SYSTEM) { 1395 return PERMISSION_DENIED; 1396 } 1397 Mutex::Autolock _l(mLock); 1398 if (mIsDeviceTypeKnown) { 1399 return INVALID_OPERATION; 1400 } 1401 mIsLowRamDevice = isLowRamDevice; 1402 mIsDeviceTypeKnown = true; 1403 return NO_ERROR; 1404} 1405 1406// ---------------------------------------------------------------------------- 1407 1408audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1409 audio_devices_t *pDevices, 1410 uint32_t *pSamplingRate, 1411 audio_format_t *pFormat, 1412 audio_channel_mask_t *pChannelMask, 1413 uint32_t *pLatencyMs, 1414 audio_output_flags_t flags, 1415 const audio_offload_info_t *offloadInfo) 1416{ 1417 PlaybackThread *thread = NULL; 1418 struct audio_config config; 1419 memset(&config, 0, sizeof(config)); 1420 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1421 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1422 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1423 if (offloadInfo != NULL) { 1424 config.offload_info = *offloadInfo; 1425 } 1426 1427 audio_stream_out_t *outStream = NULL; 1428 AudioHwDevice *outHwDev; 1429 1430 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1431 module, 1432 (pDevices != NULL) ? *pDevices : 0, 1433 config.sample_rate, 1434 config.format, 1435 config.channel_mask, 1436 flags); 1437 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1438 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); 1439 1440 if (pDevices == NULL || *pDevices == 0) { 1441 return 0; 1442 } 1443 1444 Mutex::Autolock _l(mLock); 1445 1446 outHwDev = findSuitableHwDev_l(module, *pDevices); 1447 if (outHwDev == NULL) 1448 return 0; 1449 1450 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1451 audio_io_handle_t id = nextUniqueId(); 1452 1453 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1454 1455 status_t status = hwDevHal->open_output_stream(hwDevHal, 1456 id, 1457 *pDevices, 1458 (audio_output_flags_t)flags, 1459 &config, 1460 &outStream); 1461 1462 mHardwareStatus = AUDIO_HW_IDLE; 1463 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1464 "Channels %x, status %d", 1465 outStream, 1466 config.sample_rate, 1467 config.format, 1468 config.channel_mask, 1469 status); 1470 1471 if (status == NO_ERROR && outStream != NULL) { 1472 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1473 1474 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1475 thread = new OffloadThread(this, output, id, *pDevices); 1476 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1477 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1478 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1479 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1480 thread = new DirectOutputThread(this, output, id, *pDevices); 1481 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1482 } else { 1483 thread = new MixerThread(this, output, id, *pDevices); 1484 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1485 } 1486 mPlaybackThreads.add(id, thread); 1487 1488 if (pSamplingRate != NULL) { 1489 *pSamplingRate = config.sample_rate; 1490 } 1491 if (pFormat != NULL) { 1492 *pFormat = config.format; 1493 } 1494 if (pChannelMask != NULL) { 1495 *pChannelMask = config.channel_mask; 1496 } 1497 if (pLatencyMs != NULL) { 1498 *pLatencyMs = thread->latency(); 1499 } 1500 1501 // notify client processes of the new output creation 1502 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1503 1504 // the first primary output opened designates the primary hw device 1505 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1506 ALOGI("Using module %d has the primary audio interface", module); 1507 mPrimaryHardwareDev = outHwDev; 1508 1509 AutoMutex lock(mHardwareLock); 1510 mHardwareStatus = AUDIO_HW_SET_MODE; 1511 hwDevHal->set_mode(hwDevHal, mMode); 1512 mHardwareStatus = AUDIO_HW_IDLE; 1513 } 1514 return id; 1515 } 1516 1517 return 0; 1518} 1519 1520audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1521 audio_io_handle_t output2) 1522{ 1523 Mutex::Autolock _l(mLock); 1524 MixerThread *thread1 = checkMixerThread_l(output1); 1525 MixerThread *thread2 = checkMixerThread_l(output2); 1526 1527 if (thread1 == NULL || thread2 == NULL) { 1528 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1529 output2); 1530 return 0; 1531 } 1532 1533 audio_io_handle_t id = nextUniqueId(); 1534 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1535 thread->addOutputTrack(thread2); 1536 mPlaybackThreads.add(id, thread); 1537 // notify client processes of the new output creation 1538 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1539 return id; 1540} 1541 1542status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1543{ 1544 return closeOutput_nonvirtual(output); 1545} 1546 1547status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1548{ 1549 // keep strong reference on the playback thread so that 1550 // it is not destroyed while exit() is executed 1551 sp<PlaybackThread> thread; 1552 { 1553 Mutex::Autolock _l(mLock); 1554 thread = checkPlaybackThread_l(output); 1555 if (thread == NULL) { 1556 return BAD_VALUE; 1557 } 1558 1559 ALOGV("closeOutput() %d", output); 1560 1561 if (thread->type() == ThreadBase::MIXER) { 1562 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1563 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1564 DuplicatingThread *dupThread = 1565 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1566 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1567 1568 } 1569 } 1570 } 1571 1572 1573 mPlaybackThreads.removeItem(output); 1574 // save all effects to the default thread 1575 if (mPlaybackThreads.size()) { 1576 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1577 if (dstThread != NULL) { 1578 // audioflinger lock is held here so the acquisition order of thread locks does not 1579 // matter 1580 Mutex::Autolock _dl(dstThread->mLock); 1581 Mutex::Autolock _sl(thread->mLock); 1582 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1583 for (size_t i = 0; i < effectChains.size(); i ++) { 1584 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1585 } 1586 } 1587 } 1588 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1589 } 1590 thread->exit(); 1591 // The thread entity (active unit of execution) is no longer running here, 1592 // but the ThreadBase container still exists. 1593 1594 if (thread->type() != ThreadBase::DUPLICATING) { 1595 AudioStreamOut *out = thread->clearOutput(); 1596 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1597 // from now on thread->mOutput is NULL 1598 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1599 delete out; 1600 } 1601 return NO_ERROR; 1602} 1603 1604status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1605{ 1606 Mutex::Autolock _l(mLock); 1607 PlaybackThread *thread = checkPlaybackThread_l(output); 1608 1609 if (thread == NULL) { 1610 return BAD_VALUE; 1611 } 1612 1613 ALOGV("suspendOutput() %d", output); 1614 thread->suspend(); 1615 1616 return NO_ERROR; 1617} 1618 1619status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1620{ 1621 Mutex::Autolock _l(mLock); 1622 PlaybackThread *thread = checkPlaybackThread_l(output); 1623 1624 if (thread == NULL) { 1625 return BAD_VALUE; 1626 } 1627 1628 ALOGV("restoreOutput() %d", output); 1629 1630 thread->restore(); 1631 1632 return NO_ERROR; 1633} 1634 1635audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1636 audio_devices_t *pDevices, 1637 uint32_t *pSamplingRate, 1638 audio_format_t *pFormat, 1639 audio_channel_mask_t *pChannelMask) 1640{ 1641 status_t status; 1642 RecordThread *thread = NULL; 1643 struct audio_config config; 1644 memset(&config, 0, sizeof(config)); 1645 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1646 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1647 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1648 1649 uint32_t reqSamplingRate = config.sample_rate; 1650 audio_format_t reqFormat = config.format; 1651 audio_channel_mask_t reqChannelMask = config.channel_mask; 1652 audio_stream_in_t *inStream = NULL; 1653 AudioHwDevice *inHwDev; 1654 1655 if (pDevices == NULL || *pDevices == 0) { 1656 return 0; 1657 } 1658 1659 Mutex::Autolock _l(mLock); 1660 1661 inHwDev = findSuitableHwDev_l(module, *pDevices); 1662 if (inHwDev == NULL) 1663 return 0; 1664 1665 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1666 audio_io_handle_t id = nextUniqueId(); 1667 1668 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1669 &inStream); 1670 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1671 "status %d", 1672 inStream, 1673 config.sample_rate, 1674 config.format, 1675 config.channel_mask, 1676 status); 1677 1678 // If the input could not be opened with the requested parameters and we can handle the 1679 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1680 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1681 if (status == BAD_VALUE && 1682 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1683 (config.sample_rate <= 2 * reqSamplingRate) && 1684 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) { 1685 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1686 inStream = NULL; 1687 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1688 } 1689 1690 if (status == NO_ERROR && inStream != NULL) { 1691 1692#ifdef TEE_SINK 1693 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1694 // or (re-)create if current Pipe is idle and does not match the new format 1695 sp<NBAIO_Sink> teeSink; 1696 enum { 1697 TEE_SINK_NO, // don't copy input 1698 TEE_SINK_NEW, // copy input using a new pipe 1699 TEE_SINK_OLD, // copy input using an existing pipe 1700 } kind; 1701 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1702 popcount(inStream->common.get_channels(&inStream->common))); 1703 if (!mTeeSinkInputEnabled) { 1704 kind = TEE_SINK_NO; 1705 } else if (format == Format_Invalid) { 1706 kind = TEE_SINK_NO; 1707 } else if (mRecordTeeSink == 0) { 1708 kind = TEE_SINK_NEW; 1709 } else if (mRecordTeeSink->getStrongCount() != 1) { 1710 kind = TEE_SINK_NO; 1711 } else if (format == mRecordTeeSink->format()) { 1712 kind = TEE_SINK_OLD; 1713 } else { 1714 kind = TEE_SINK_NEW; 1715 } 1716 switch (kind) { 1717 case TEE_SINK_NEW: { 1718 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1719 size_t numCounterOffers = 0; 1720 const NBAIO_Format offers[1] = {format}; 1721 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1722 ALOG_ASSERT(index == 0); 1723 PipeReader *pipeReader = new PipeReader(*pipe); 1724 numCounterOffers = 0; 1725 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1726 ALOG_ASSERT(index == 0); 1727 mRecordTeeSink = pipe; 1728 mRecordTeeSource = pipeReader; 1729 teeSink = pipe; 1730 } 1731 break; 1732 case TEE_SINK_OLD: 1733 teeSink = mRecordTeeSink; 1734 break; 1735 case TEE_SINK_NO: 1736 default: 1737 break; 1738 } 1739#endif 1740 1741 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1742 1743 // Start record thread 1744 // RecordThread requires both input and output device indication to forward to audio 1745 // pre processing modules 1746 thread = new RecordThread(this, 1747 input, 1748 reqSamplingRate, 1749 reqChannelMask, 1750 id, 1751 primaryOutputDevice_l(), 1752 *pDevices 1753#ifdef TEE_SINK 1754 , teeSink 1755#endif 1756 ); 1757 mRecordThreads.add(id, thread); 1758 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1759 if (pSamplingRate != NULL) { 1760 *pSamplingRate = reqSamplingRate; 1761 } 1762 if (pFormat != NULL) { 1763 *pFormat = config.format; 1764 } 1765 if (pChannelMask != NULL) { 1766 *pChannelMask = reqChannelMask; 1767 } 1768 1769 // notify client processes of the new input creation 1770 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1771 return id; 1772 } 1773 1774 return 0; 1775} 1776 1777status_t AudioFlinger::closeInput(audio_io_handle_t input) 1778{ 1779 return closeInput_nonvirtual(input); 1780} 1781 1782status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1783{ 1784 // keep strong reference on the record thread so that 1785 // it is not destroyed while exit() is executed 1786 sp<RecordThread> thread; 1787 { 1788 Mutex::Autolock _l(mLock); 1789 thread = checkRecordThread_l(input); 1790 if (thread == 0) { 1791 return BAD_VALUE; 1792 } 1793 1794 ALOGV("closeInput() %d", input); 1795 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1796 mRecordThreads.removeItem(input); 1797 } 1798 thread->exit(); 1799 // The thread entity (active unit of execution) is no longer running here, 1800 // but the ThreadBase container still exists. 1801 1802 AudioStreamIn *in = thread->clearInput(); 1803 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1804 // from now on thread->mInput is NULL 1805 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1806 delete in; 1807 1808 return NO_ERROR; 1809} 1810 1811status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1812{ 1813 Mutex::Autolock _l(mLock); 1814 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1815 1816 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1817 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1818 thread->invalidateTracks(stream); 1819 } 1820 1821 return NO_ERROR; 1822} 1823 1824 1825int AudioFlinger::newAudioSessionId() 1826{ 1827 return nextUniqueId(); 1828} 1829 1830void AudioFlinger::acquireAudioSessionId(int audioSession) 1831{ 1832 Mutex::Autolock _l(mLock); 1833 pid_t caller = IPCThreadState::self()->getCallingPid(); 1834 ALOGV("acquiring %d from %d", audioSession, caller); 1835 size_t num = mAudioSessionRefs.size(); 1836 for (size_t i = 0; i< num; i++) { 1837 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1838 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1839 ref->mCnt++; 1840 ALOGV(" incremented refcount to %d", ref->mCnt); 1841 return; 1842 } 1843 } 1844 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1845 ALOGV(" added new entry for %d", audioSession); 1846} 1847 1848void AudioFlinger::releaseAudioSessionId(int audioSession) 1849{ 1850 Mutex::Autolock _l(mLock); 1851 pid_t caller = IPCThreadState::self()->getCallingPid(); 1852 ALOGV("releasing %d from %d", audioSession, caller); 1853 size_t num = mAudioSessionRefs.size(); 1854 for (size_t i = 0; i< num; i++) { 1855 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1856 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1857 ref->mCnt--; 1858 ALOGV(" decremented refcount to %d", ref->mCnt); 1859 if (ref->mCnt == 0) { 1860 mAudioSessionRefs.removeAt(i); 1861 delete ref; 1862 purgeStaleEffects_l(); 1863 } 1864 return; 1865 } 1866 } 1867 ALOGW("session id %d not found for pid %d", audioSession, caller); 1868} 1869 1870void AudioFlinger::purgeStaleEffects_l() { 1871 1872 ALOGV("purging stale effects"); 1873 1874 Vector< sp<EffectChain> > chains; 1875 1876 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1877 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1878 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1879 sp<EffectChain> ec = t->mEffectChains[j]; 1880 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1881 chains.push(ec); 1882 } 1883 } 1884 } 1885 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1886 sp<RecordThread> t = mRecordThreads.valueAt(i); 1887 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1888 sp<EffectChain> ec = t->mEffectChains[j]; 1889 chains.push(ec); 1890 } 1891 } 1892 1893 for (size_t i = 0; i < chains.size(); i++) { 1894 sp<EffectChain> ec = chains[i]; 1895 int sessionid = ec->sessionId(); 1896 sp<ThreadBase> t = ec->mThread.promote(); 1897 if (t == 0) { 1898 continue; 1899 } 1900 size_t numsessionrefs = mAudioSessionRefs.size(); 1901 bool found = false; 1902 for (size_t k = 0; k < numsessionrefs; k++) { 1903 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1904 if (ref->mSessionid == sessionid) { 1905 ALOGV(" session %d still exists for %d with %d refs", 1906 sessionid, ref->mPid, ref->mCnt); 1907 found = true; 1908 break; 1909 } 1910 } 1911 if (!found) { 1912 Mutex::Autolock _l (t->mLock); 1913 // remove all effects from the chain 1914 while (ec->mEffects.size()) { 1915 sp<EffectModule> effect = ec->mEffects[0]; 1916 effect->unPin(); 1917 t->removeEffect_l(effect); 1918 if (effect->purgeHandles()) { 1919 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1920 } 1921 AudioSystem::unregisterEffect(effect->id()); 1922 } 1923 } 1924 } 1925 return; 1926} 1927 1928// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1929AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1930{ 1931 return mPlaybackThreads.valueFor(output).get(); 1932} 1933 1934// checkMixerThread_l() must be called with AudioFlinger::mLock held 1935AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1936{ 1937 PlaybackThread *thread = checkPlaybackThread_l(output); 1938 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1939} 1940 1941// checkRecordThread_l() must be called with AudioFlinger::mLock held 1942AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1943{ 1944 return mRecordThreads.valueFor(input).get(); 1945} 1946 1947uint32_t AudioFlinger::nextUniqueId() 1948{ 1949 return android_atomic_inc(&mNextUniqueId); 1950} 1951 1952AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1953{ 1954 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1955 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1956 AudioStreamOut *output = thread->getOutput(); 1957 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1958 return thread; 1959 } 1960 } 1961 return NULL; 1962} 1963 1964audio_devices_t AudioFlinger::primaryOutputDevice_l() const 1965{ 1966 PlaybackThread *thread = primaryPlaybackThread_l(); 1967 1968 if (thread == NULL) { 1969 return 0; 1970 } 1971 1972 return thread->outDevice(); 1973} 1974 1975sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 1976 int triggerSession, 1977 int listenerSession, 1978 sync_event_callback_t callBack, 1979 void *cookie) 1980{ 1981 Mutex::Autolock _l(mLock); 1982 1983 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 1984 status_t playStatus = NAME_NOT_FOUND; 1985 status_t recStatus = NAME_NOT_FOUND; 1986 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1987 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 1988 if (playStatus == NO_ERROR) { 1989 return event; 1990 } 1991 } 1992 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1993 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 1994 if (recStatus == NO_ERROR) { 1995 return event; 1996 } 1997 } 1998 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 1999 mPendingSyncEvents.add(event); 2000 } else { 2001 ALOGV("createSyncEvent() invalid event %d", event->type()); 2002 event.clear(); 2003 } 2004 return event; 2005} 2006 2007// ---------------------------------------------------------------------------- 2008// Effect management 2009// ---------------------------------------------------------------------------- 2010 2011 2012status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2013{ 2014 Mutex::Autolock _l(mLock); 2015 return EffectQueryNumberEffects(numEffects); 2016} 2017 2018status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2019{ 2020 Mutex::Autolock _l(mLock); 2021 return EffectQueryEffect(index, descriptor); 2022} 2023 2024status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2025 effect_descriptor_t *descriptor) const 2026{ 2027 Mutex::Autolock _l(mLock); 2028 return EffectGetDescriptor(pUuid, descriptor); 2029} 2030 2031 2032sp<IEffect> AudioFlinger::createEffect( 2033 effect_descriptor_t *pDesc, 2034 const sp<IEffectClient>& effectClient, 2035 int32_t priority, 2036 audio_io_handle_t io, 2037 int sessionId, 2038 status_t *status, 2039 int *id, 2040 int *enabled) 2041{ 2042 status_t lStatus = NO_ERROR; 2043 sp<EffectHandle> handle; 2044 effect_descriptor_t desc; 2045 2046 pid_t pid = IPCThreadState::self()->getCallingPid(); 2047 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2048 pid, effectClient.get(), priority, sessionId, io); 2049 2050 if (pDesc == NULL) { 2051 lStatus = BAD_VALUE; 2052 goto Exit; 2053 } 2054 2055 // check audio settings permission for global effects 2056 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2057 lStatus = PERMISSION_DENIED; 2058 goto Exit; 2059 } 2060 2061 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2062 // that can only be created by audio policy manager (running in same process) 2063 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2064 lStatus = PERMISSION_DENIED; 2065 goto Exit; 2066 } 2067 2068 if (io == 0) { 2069 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2070 // output must be specified by AudioPolicyManager when using session 2071 // AUDIO_SESSION_OUTPUT_STAGE 2072 lStatus = BAD_VALUE; 2073 goto Exit; 2074 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2075 // if the output returned by getOutputForEffect() is removed before we lock the 2076 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2077 // and we will exit safely 2078 io = AudioSystem::getOutputForEffect(&desc); 2079 } 2080 } 2081 2082 { 2083 Mutex::Autolock _l(mLock); 2084 2085 2086 if (!EffectIsNullUuid(&pDesc->uuid)) { 2087 // if uuid is specified, request effect descriptor 2088 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2089 if (lStatus < 0) { 2090 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2091 goto Exit; 2092 } 2093 } else { 2094 // if uuid is not specified, look for an available implementation 2095 // of the required type in effect factory 2096 if (EffectIsNullUuid(&pDesc->type)) { 2097 ALOGW("createEffect() no effect type"); 2098 lStatus = BAD_VALUE; 2099 goto Exit; 2100 } 2101 uint32_t numEffects = 0; 2102 effect_descriptor_t d; 2103 d.flags = 0; // prevent compiler warning 2104 bool found = false; 2105 2106 lStatus = EffectQueryNumberEffects(&numEffects); 2107 if (lStatus < 0) { 2108 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2109 goto Exit; 2110 } 2111 for (uint32_t i = 0; i < numEffects; i++) { 2112 lStatus = EffectQueryEffect(i, &desc); 2113 if (lStatus < 0) { 2114 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2115 continue; 2116 } 2117 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2118 // If matching type found save effect descriptor. If the session is 2119 // 0 and the effect is not auxiliary, continue enumeration in case 2120 // an auxiliary version of this effect type is available 2121 found = true; 2122 d = desc; 2123 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2124 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2125 break; 2126 } 2127 } 2128 } 2129 if (!found) { 2130 lStatus = BAD_VALUE; 2131 ALOGW("createEffect() effect not found"); 2132 goto Exit; 2133 } 2134 // For same effect type, chose auxiliary version over insert version if 2135 // connect to output mix (Compliance to OpenSL ES) 2136 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2137 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2138 desc = d; 2139 } 2140 } 2141 2142 // Do not allow auxiliary effects on a session different from 0 (output mix) 2143 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2144 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2145 lStatus = INVALID_OPERATION; 2146 goto Exit; 2147 } 2148 2149 // check recording permission for visualizer 2150 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2151 !recordingAllowed()) { 2152 lStatus = PERMISSION_DENIED; 2153 goto Exit; 2154 } 2155 2156 // return effect descriptor 2157 *pDesc = desc; 2158 2159 // If output is not specified try to find a matching audio session ID in one of the 2160 // output threads. 2161 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2162 // because of code checking output when entering the function. 2163 // Note: io is never 0 when creating an effect on an input 2164 if (io == 0) { 2165 // look for the thread where the specified audio session is present 2166 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2167 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2168 io = mPlaybackThreads.keyAt(i); 2169 break; 2170 } 2171 } 2172 if (io == 0) { 2173 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2174 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2175 io = mRecordThreads.keyAt(i); 2176 break; 2177 } 2178 } 2179 } 2180 // If no output thread contains the requested session ID, default to 2181 // first output. The effect chain will be moved to the correct output 2182 // thread when a track with the same session ID is created 2183 if (io == 0 && mPlaybackThreads.size()) { 2184 io = mPlaybackThreads.keyAt(0); 2185 } 2186 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2187 } 2188 ThreadBase *thread = checkRecordThread_l(io); 2189 if (thread == NULL) { 2190 thread = checkPlaybackThread_l(io); 2191 if (thread == NULL) { 2192 ALOGE("createEffect() unknown output thread"); 2193 lStatus = BAD_VALUE; 2194 goto Exit; 2195 } 2196 } 2197 2198 sp<Client> client = registerPid_l(pid); 2199 2200 // create effect on selected output thread 2201 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2202 &desc, enabled, &lStatus); 2203 if (handle != 0 && id != NULL) { 2204 *id = handle->id(); 2205 } 2206 } 2207 2208Exit: 2209 *status = lStatus; 2210 return handle; 2211} 2212 2213status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2214 audio_io_handle_t dstOutput) 2215{ 2216 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2217 sessionId, srcOutput, dstOutput); 2218 Mutex::Autolock _l(mLock); 2219 if (srcOutput == dstOutput) { 2220 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2221 return NO_ERROR; 2222 } 2223 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2224 if (srcThread == NULL) { 2225 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2226 return BAD_VALUE; 2227 } 2228 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2229 if (dstThread == NULL) { 2230 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2231 return BAD_VALUE; 2232 } 2233 2234 Mutex::Autolock _dl(dstThread->mLock); 2235 Mutex::Autolock _sl(srcThread->mLock); 2236 moveEffectChain_l(sessionId, srcThread, dstThread, false); 2237 2238 return NO_ERROR; 2239} 2240 2241// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2242status_t AudioFlinger::moveEffectChain_l(int sessionId, 2243 AudioFlinger::PlaybackThread *srcThread, 2244 AudioFlinger::PlaybackThread *dstThread, 2245 bool reRegister) 2246{ 2247 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2248 sessionId, srcThread, dstThread); 2249 2250 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2251 if (chain == 0) { 2252 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2253 sessionId, srcThread); 2254 return INVALID_OPERATION; 2255 } 2256 2257 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2258 // so that a new chain is created with correct parameters when first effect is added. This is 2259 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2260 // removed. 2261 srcThread->removeEffectChain_l(chain); 2262 2263 // transfer all effects one by one so that new effect chain is created on new thread with 2264 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2265 audio_io_handle_t dstOutput = dstThread->id(); 2266 sp<EffectChain> dstChain; 2267 uint32_t strategy = 0; // prevent compiler warning 2268 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2269 while (effect != 0) { 2270 srcThread->removeEffect_l(effect); 2271 dstThread->addEffect_l(effect); 2272 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2273 if (effect->state() == EffectModule::ACTIVE || 2274 effect->state() == EffectModule::STOPPING) { 2275 effect->start(); 2276 } 2277 // if the move request is not received from audio policy manager, the effect must be 2278 // re-registered with the new strategy and output 2279 if (dstChain == 0) { 2280 dstChain = effect->chain().promote(); 2281 if (dstChain == 0) { 2282 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2283 srcThread->addEffect_l(effect); 2284 return NO_INIT; 2285 } 2286 strategy = dstChain->strategy(); 2287 } 2288 if (reRegister) { 2289 AudioSystem::unregisterEffect(effect->id()); 2290 AudioSystem::registerEffect(&effect->desc(), 2291 dstOutput, 2292 strategy, 2293 sessionId, 2294 effect->id()); 2295 } 2296 effect = chain->getEffectFromId_l(0); 2297 } 2298 2299 return NO_ERROR; 2300} 2301 2302struct Entry { 2303#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2304 char mName[MAX_NAME]; 2305}; 2306 2307int comparEntry(const void *p1, const void *p2) 2308{ 2309 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2310} 2311 2312#ifdef TEE_SINK 2313void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2314{ 2315 NBAIO_Source *teeSource = source.get(); 2316 if (teeSource != NULL) { 2317 // .wav rotation 2318 // There is a benign race condition if 2 threads call this simultaneously. 2319 // They would both traverse the directory, but the result would simply be 2320 // failures at unlink() which are ignored. It's also unlikely since 2321 // normally dumpsys is only done by bugreport or from the command line. 2322 char teePath[32+256]; 2323 strcpy(teePath, "/data/misc/media"); 2324 size_t teePathLen = strlen(teePath); 2325 DIR *dir = opendir(teePath); 2326 teePath[teePathLen++] = '/'; 2327 if (dir != NULL) { 2328#define MAX_SORT 20 // number of entries to sort 2329#define MAX_KEEP 10 // number of entries to keep 2330 struct Entry entries[MAX_SORT]; 2331 size_t entryCount = 0; 2332 while (entryCount < MAX_SORT) { 2333 struct dirent de; 2334 struct dirent *result = NULL; 2335 int rc = readdir_r(dir, &de, &result); 2336 if (rc != 0) { 2337 ALOGW("readdir_r failed %d", rc); 2338 break; 2339 } 2340 if (result == NULL) { 2341 break; 2342 } 2343 if (result != &de) { 2344 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2345 break; 2346 } 2347 // ignore non .wav file entries 2348 size_t nameLen = strlen(de.d_name); 2349 if (nameLen <= 4 || nameLen >= MAX_NAME || 2350 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2351 continue; 2352 } 2353 strcpy(entries[entryCount++].mName, de.d_name); 2354 } 2355 (void) closedir(dir); 2356 if (entryCount > MAX_KEEP) { 2357 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2358 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2359 strcpy(&teePath[teePathLen], entries[i].mName); 2360 (void) unlink(teePath); 2361 } 2362 } 2363 } else { 2364 if (fd >= 0) { 2365 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2366 } 2367 } 2368 char teeTime[16]; 2369 struct timeval tv; 2370 gettimeofday(&tv, NULL); 2371 struct tm tm; 2372 localtime_r(&tv.tv_sec, &tm); 2373 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2374 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2375 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2376 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2377 if (teeFd >= 0) { 2378 char wavHeader[44]; 2379 memcpy(wavHeader, 2380 "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", 2381 sizeof(wavHeader)); 2382 NBAIO_Format format = teeSource->format(); 2383 unsigned channelCount = Format_channelCount(format); 2384 ALOG_ASSERT(channelCount <= FCC_2); 2385 uint32_t sampleRate = Format_sampleRate(format); 2386 wavHeader[22] = channelCount; // number of channels 2387 wavHeader[24] = sampleRate; // sample rate 2388 wavHeader[25] = sampleRate >> 8; 2389 wavHeader[32] = channelCount * 2; // block alignment 2390 write(teeFd, wavHeader, sizeof(wavHeader)); 2391 size_t total = 0; 2392 bool firstRead = true; 2393 for (;;) { 2394#define TEE_SINK_READ 1024 2395 short buffer[TEE_SINK_READ * FCC_2]; 2396 size_t count = TEE_SINK_READ; 2397 ssize_t actual = teeSource->read(buffer, count, 2398 AudioBufferProvider::kInvalidPTS); 2399 bool wasFirstRead = firstRead; 2400 firstRead = false; 2401 if (actual <= 0) { 2402 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2403 continue; 2404 } 2405 break; 2406 } 2407 ALOG_ASSERT(actual <= (ssize_t)count); 2408 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2409 total += actual; 2410 } 2411 lseek(teeFd, (off_t) 4, SEEK_SET); 2412 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2413 write(teeFd, &temp, sizeof(temp)); 2414 lseek(teeFd, (off_t) 40, SEEK_SET); 2415 temp = total * channelCount * sizeof(short); 2416 write(teeFd, &temp, sizeof(temp)); 2417 close(teeFd); 2418 if (fd >= 0) { 2419 fdprintf(fd, "tee copied to %s\n", teePath); 2420 } 2421 } else { 2422 if (fd >= 0) { 2423 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2424 } 2425 } 2426 } 2427} 2428#endif 2429 2430// ---------------------------------------------------------------------------- 2431 2432status_t AudioFlinger::onTransact( 2433 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2434{ 2435 return BnAudioFlinger::onTransact(code, data, reply, flags); 2436} 2437 2438}; // namespace android 2439