AudioFlinger.cpp revision 685ef09bcaf5de6abf2064d552296f70eaec6761
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 <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40//#include <private/media/AudioTrackShared.h> 41//#include <private/media/AudioEffectShared.h> 42 43#include <system/audio.h> 44#include <hardware/audio.h> 45 46#include "AudioMixer.h" 47#include "AudioFlinger.h" 48#include "ServiceUtilities.h" 49 50#include <media/EffectsFactoryApi.h> 51#include <audio_effects/effect_visualizer.h> 52#include <audio_effects/effect_ns.h> 53#include <audio_effects/effect_aec.h> 54 55#include <audio_utils/primitives.h> 56 57#include <powermanager/PowerManager.h> 58 59#include <common_time/cc_helper.h> 60//#include <common_time/local_clock.h> 61 62#include <media/IMediaLogService.h> 63 64// ---------------------------------------------------------------------------- 65 66// Note: the following macro is used for extremely verbose logging message. In 67// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 68// 0; but one side effect of this is to turn all LOGV's as well. Some messages 69// are so verbose that we want to suppress them even when we have ALOG_ASSERT 70// turned on. Do not uncomment the #def below unless you really know what you 71// are doing and want to see all of the extremely verbose messages. 72//#define VERY_VERY_VERBOSE_LOGGING 73#ifdef VERY_VERY_VERBOSE_LOGGING 74#define ALOGVV ALOGV 75#else 76#define ALOGVV(a...) do { } while(0) 77#endif 78 79namespace android { 80 81static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 82static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 83 84 85nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 86 87uint32_t AudioFlinger::mScreenState; 88 89// ---------------------------------------------------------------------------- 90 91static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 92{ 93 const hw_module_t *mod; 94 int rc; 95 96 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 97 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 98 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 99 if (rc) { 100 goto out; 101 } 102 rc = audio_hw_device_open(mod, dev); 103 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 104 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 105 if (rc) { 106 goto out; 107 } 108 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 109 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 110 rc = BAD_VALUE; 111 goto out; 112 } 113 return 0; 114 115out: 116 *dev = NULL; 117 return rc; 118} 119 120// ---------------------------------------------------------------------------- 121 122AudioFlinger::AudioFlinger() 123 : BnAudioFlinger(), 124 mPrimaryHardwareDev(NULL), 125 mHardwareStatus(AUDIO_HW_IDLE), 126 mMasterVolume(1.0f), 127 mMasterMute(false), 128 mNextUniqueId(1), 129 mMode(AUDIO_MODE_INVALID), 130 mBtNrecIsOff(false) 131{ 132 char value[PROPERTY_VALUE_MAX]; 133 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 134 if (doLog) { 135 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters"); 136 } 137} 138 139void AudioFlinger::onFirstRef() 140{ 141 int rc = 0; 142 143 Mutex::Autolock _l(mLock); 144 145 /* TODO: move all this work into an Init() function */ 146 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 147 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 148 uint32_t int_val; 149 if (1 == sscanf(val_str, "%u", &int_val)) { 150 mStandbyTimeInNsecs = milliseconds(int_val); 151 ALOGI("Using %u mSec as standby time.", int_val); 152 } else { 153 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 154 ALOGI("Using default %u mSec as standby time.", 155 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 156 } 157 } 158 159 mMode = AUDIO_MODE_NORMAL; 160} 161 162AudioFlinger::~AudioFlinger() 163{ 164 while (!mRecordThreads.isEmpty()) { 165 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 166 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 167 } 168 while (!mPlaybackThreads.isEmpty()) { 169 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 170 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 171 } 172 173 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 174 // no mHardwareLock needed, as there are no other references to this 175 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 176 delete mAudioHwDevs.valueAt(i); 177 } 178} 179 180static const char * const audio_interfaces[] = { 181 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 182 AUDIO_HARDWARE_MODULE_ID_A2DP, 183 AUDIO_HARDWARE_MODULE_ID_USB, 184}; 185#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 186 187AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 188 audio_module_handle_t module, 189 audio_devices_t devices) 190{ 191 // if module is 0, the request comes from an old policy manager and we should load 192 // well known modules 193 if (module == 0) { 194 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 195 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 196 loadHwModule_l(audio_interfaces[i]); 197 } 198 // then try to find a module supporting the requested device. 199 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 200 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 201 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 202 if ((dev->get_supported_devices != NULL) && 203 (dev->get_supported_devices(dev) & devices) == devices) 204 return audioHwDevice; 205 } 206 } else { 207 // check a match for the requested module handle 208 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 209 if (audioHwDevice != NULL) { 210 return audioHwDevice; 211 } 212 } 213 214 return NULL; 215} 216 217void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 218{ 219 const size_t SIZE = 256; 220 char buffer[SIZE]; 221 String8 result; 222 223 result.append("Clients:\n"); 224 for (size_t i = 0; i < mClients.size(); ++i) { 225 sp<Client> client = mClients.valueAt(i).promote(); 226 if (client != 0) { 227 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 228 result.append(buffer); 229 } 230 } 231 232 result.append("Global session refs:\n"); 233 result.append(" session pid count\n"); 234 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 235 AudioSessionRef *r = mAudioSessionRefs[i]; 236 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 237 result.append(buffer); 238 } 239 write(fd, result.string(), result.size()); 240} 241 242 243void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 244{ 245 const size_t SIZE = 256; 246 char buffer[SIZE]; 247 String8 result; 248 hardware_call_state hardwareStatus = mHardwareStatus; 249 250 snprintf(buffer, SIZE, "Hardware status: %d\n" 251 "Standby Time mSec: %u\n", 252 hardwareStatus, 253 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 254 result.append(buffer); 255 write(fd, result.string(), result.size()); 256} 257 258void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 259{ 260 const size_t SIZE = 256; 261 char buffer[SIZE]; 262 String8 result; 263 snprintf(buffer, SIZE, "Permission Denial: " 264 "can't dump AudioFlinger from pid=%d, uid=%d\n", 265 IPCThreadState::self()->getCallingPid(), 266 IPCThreadState::self()->getCallingUid()); 267 result.append(buffer); 268 write(fd, result.string(), result.size()); 269} 270 271bool AudioFlinger::dumpTryLock(Mutex& mutex) 272{ 273 bool locked = false; 274 for (int i = 0; i < kDumpLockRetries; ++i) { 275 if (mutex.tryLock() == NO_ERROR) { 276 locked = true; 277 break; 278 } 279 usleep(kDumpLockSleepUs); 280 } 281 return locked; 282} 283 284status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 285{ 286 if (!dumpAllowed()) { 287 dumpPermissionDenial(fd, args); 288 } else { 289 // get state of hardware lock 290 bool hardwareLocked = dumpTryLock(mHardwareLock); 291 if (!hardwareLocked) { 292 String8 result(kHardwareLockedString); 293 write(fd, result.string(), result.size()); 294 } else { 295 mHardwareLock.unlock(); 296 } 297 298 bool locked = dumpTryLock(mLock); 299 300 // failed to lock - AudioFlinger is probably deadlocked 301 if (!locked) { 302 String8 result(kDeadlockedString); 303 write(fd, result.string(), result.size()); 304 } 305 306 dumpClients(fd, args); 307 dumpInternals(fd, args); 308 309 // dump playback threads 310 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 311 mPlaybackThreads.valueAt(i)->dump(fd, args); 312 } 313 314 // dump record threads 315 for (size_t i = 0; i < mRecordThreads.size(); i++) { 316 mRecordThreads.valueAt(i)->dump(fd, args); 317 } 318 319 // dump all hardware devs 320 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 321 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 322 dev->dump(dev, fd); 323 } 324 325 // dump the serially shared record tee sink 326 if (mRecordTeeSource != 0) { 327 dumpTee(fd, mRecordTeeSource); 328 } 329 330 if (locked) { 331 mLock.unlock(); 332 } 333 334 // append a copy of media.log here by forwarding fd to it, but don't attempt 335 // to lookup the service if it's not running, as it will block for a second 336 if (mLogMemoryDealer != 0) { 337 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 338 if (binder != 0) { 339 fdprintf(fd, "\nmedia.log:\n"); 340 Vector<String16> args; 341 binder->dump(fd, args); 342 } 343 } 344 } 345 return NO_ERROR; 346} 347 348sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 349{ 350 // If pid is already in the mClients wp<> map, then use that entry 351 // (for which promote() is always != 0), otherwise create a new entry and Client. 352 sp<Client> client = mClients.valueFor(pid).promote(); 353 if (client == 0) { 354 client = new Client(this, pid); 355 mClients.add(pid, client); 356 } 357 358 return client; 359} 360 361sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 362{ 363 if (mLogMemoryDealer == 0) { 364 return new NBLog::Writer(); 365 } 366 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 367 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 368 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 369 if (binder != 0) { 370 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 371 } 372 return writer; 373} 374 375void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 376{ 377 if (writer == 0) { 378 return; 379 } 380 sp<IMemory> iMemory(writer->getIMemory()); 381 if (iMemory == 0) { 382 return; 383 } 384 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 385 if (binder != 0) { 386 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 387 // Now the media.log remote reference to IMemory is gone. 388 // When our last local reference to IMemory also drops to zero, 389 // the IMemory destructor will deallocate the region from mMemoryDealer. 390 } 391} 392 393// IAudioFlinger interface 394 395 396sp<IAudioTrack> AudioFlinger::createTrack( 397 audio_stream_type_t streamType, 398 uint32_t sampleRate, 399 audio_format_t format, 400 audio_channel_mask_t channelMask, 401 size_t frameCount, 402 IAudioFlinger::track_flags_t *flags, 403 const sp<IMemory>& sharedBuffer, 404 audio_io_handle_t output, 405 pid_t tid, 406 int *sessionId, 407 status_t *status) 408{ 409 sp<PlaybackThread::Track> track; 410 sp<TrackHandle> trackHandle; 411 sp<Client> client; 412 status_t lStatus; 413 int lSessionId; 414 415 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 416 // but if someone uses binder directly they could bypass that and cause us to crash 417 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 418 ALOGE("createTrack() invalid stream type %d", streamType); 419 lStatus = BAD_VALUE; 420 goto Exit; 421 } 422 423 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 424 // and we don't yet support 8.24 or 32-bit PCM 425 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 426 ALOGE("createTrack() invalid format %d", format); 427 lStatus = BAD_VALUE; 428 goto Exit; 429 } 430 431 { 432 Mutex::Autolock _l(mLock); 433 PlaybackThread *thread = checkPlaybackThread_l(output); 434 PlaybackThread *effectThread = NULL; 435 if (thread == NULL) { 436 ALOGE("unknown output thread"); 437 lStatus = BAD_VALUE; 438 goto Exit; 439 } 440 441 pid_t pid = IPCThreadState::self()->getCallingPid(); 442 client = registerPid_l(pid); 443 444 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 445 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 446 // check if an effect chain with the same session ID is present on another 447 // output thread and move it here. 448 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 449 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 450 if (mPlaybackThreads.keyAt(i) != output) { 451 uint32_t sessions = t->hasAudioSession(*sessionId); 452 if (sessions & PlaybackThread::EFFECT_SESSION) { 453 effectThread = t.get(); 454 break; 455 } 456 } 457 } 458 lSessionId = *sessionId; 459 } else { 460 // if no audio session id is provided, create one here 461 lSessionId = nextUniqueId(); 462 if (sessionId != NULL) { 463 *sessionId = lSessionId; 464 } 465 } 466 ALOGV("createTrack() lSessionId: %d", lSessionId); 467 468 track = thread->createTrack_l(client, streamType, sampleRate, format, 469 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 470 471 // move effect chain to this output thread if an effect on same session was waiting 472 // for a track to be created 473 if (lStatus == NO_ERROR && effectThread != NULL) { 474 Mutex::Autolock _dl(thread->mLock); 475 Mutex::Autolock _sl(effectThread->mLock); 476 moveEffectChain_l(lSessionId, effectThread, thread, true); 477 } 478 479 // Look for sync events awaiting for a session to be used. 480 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 481 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 482 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 483 if (lStatus == NO_ERROR) { 484 (void) track->setSyncEvent(mPendingSyncEvents[i]); 485 } else { 486 mPendingSyncEvents[i]->cancel(); 487 } 488 mPendingSyncEvents.removeAt(i); 489 i--; 490 } 491 } 492 } 493 } 494 if (lStatus == NO_ERROR) { 495 trackHandle = new TrackHandle(track); 496 } else { 497 // remove local strong reference to Client before deleting the Track so that the Client 498 // destructor is called by the TrackBase destructor with mLock held 499 client.clear(); 500 track.clear(); 501 } 502 503Exit: 504 if (status != NULL) { 505 *status = lStatus; 506 } 507 return trackHandle; 508} 509 510uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 511{ 512 Mutex::Autolock _l(mLock); 513 PlaybackThread *thread = checkPlaybackThread_l(output); 514 if (thread == NULL) { 515 ALOGW("sampleRate() unknown thread %d", output); 516 return 0; 517 } 518 return thread->sampleRate(); 519} 520 521int AudioFlinger::channelCount(audio_io_handle_t output) const 522{ 523 Mutex::Autolock _l(mLock); 524 PlaybackThread *thread = checkPlaybackThread_l(output); 525 if (thread == NULL) { 526 ALOGW("channelCount() unknown thread %d", output); 527 return 0; 528 } 529 return thread->channelCount(); 530} 531 532audio_format_t AudioFlinger::format(audio_io_handle_t output) const 533{ 534 Mutex::Autolock _l(mLock); 535 PlaybackThread *thread = checkPlaybackThread_l(output); 536 if (thread == NULL) { 537 ALOGW("format() unknown thread %d", output); 538 return AUDIO_FORMAT_INVALID; 539 } 540 return thread->format(); 541} 542 543size_t AudioFlinger::frameCount(audio_io_handle_t output) const 544{ 545 Mutex::Autolock _l(mLock); 546 PlaybackThread *thread = checkPlaybackThread_l(output); 547 if (thread == NULL) { 548 ALOGW("frameCount() unknown thread %d", output); 549 return 0; 550 } 551 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 552 // should examine all callers and fix them to handle smaller counts 553 return thread->frameCount(); 554} 555 556uint32_t AudioFlinger::latency(audio_io_handle_t output) const 557{ 558 Mutex::Autolock _l(mLock); 559 PlaybackThread *thread = checkPlaybackThread_l(output); 560 if (thread == NULL) { 561 ALOGW("latency() unknown thread %d", output); 562 return 0; 563 } 564 return thread->latency(); 565} 566 567status_t AudioFlinger::setMasterVolume(float value) 568{ 569 status_t ret = initCheck(); 570 if (ret != NO_ERROR) { 571 return ret; 572 } 573 574 // check calling permissions 575 if (!settingsAllowed()) { 576 return PERMISSION_DENIED; 577 } 578 579 Mutex::Autolock _l(mLock); 580 mMasterVolume = value; 581 582 // Set master volume in the HALs which support it. 583 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 584 AutoMutex lock(mHardwareLock); 585 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 586 587 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 588 if (dev->canSetMasterVolume()) { 589 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 590 } 591 mHardwareStatus = AUDIO_HW_IDLE; 592 } 593 594 // Now set the master volume in each playback thread. Playback threads 595 // assigned to HALs which do not have master volume support will apply 596 // master volume during the mix operation. Threads with HALs which do 597 // support master volume will simply ignore the setting. 598 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 599 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 600 601 return NO_ERROR; 602} 603 604status_t AudioFlinger::setMode(audio_mode_t mode) 605{ 606 status_t ret = initCheck(); 607 if (ret != NO_ERROR) { 608 return ret; 609 } 610 611 // check calling permissions 612 if (!settingsAllowed()) { 613 return PERMISSION_DENIED; 614 } 615 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 616 ALOGW("Illegal value: setMode(%d)", mode); 617 return BAD_VALUE; 618 } 619 620 { // scope for the lock 621 AutoMutex lock(mHardwareLock); 622 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 623 mHardwareStatus = AUDIO_HW_SET_MODE; 624 ret = dev->set_mode(dev, mode); 625 mHardwareStatus = AUDIO_HW_IDLE; 626 } 627 628 if (NO_ERROR == ret) { 629 Mutex::Autolock _l(mLock); 630 mMode = mode; 631 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 632 mPlaybackThreads.valueAt(i)->setMode(mode); 633 } 634 635 return ret; 636} 637 638status_t AudioFlinger::setMicMute(bool state) 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 } 649 650 AutoMutex lock(mHardwareLock); 651 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 652 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 653 ret = dev->set_mic_mute(dev, state); 654 mHardwareStatus = AUDIO_HW_IDLE; 655 return ret; 656} 657 658bool AudioFlinger::getMicMute() const 659{ 660 status_t ret = initCheck(); 661 if (ret != NO_ERROR) { 662 return false; 663 } 664 665 bool state = AUDIO_MODE_INVALID; 666 AutoMutex lock(mHardwareLock); 667 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 668 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 669 dev->get_mic_mute(dev, &state); 670 mHardwareStatus = AUDIO_HW_IDLE; 671 return state; 672} 673 674status_t AudioFlinger::setMasterMute(bool muted) 675{ 676 status_t ret = initCheck(); 677 if (ret != NO_ERROR) { 678 return ret; 679 } 680 681 // check calling permissions 682 if (!settingsAllowed()) { 683 return PERMISSION_DENIED; 684 } 685 686 Mutex::Autolock _l(mLock); 687 mMasterMute = muted; 688 689 // Set master mute in the HALs which support it. 690 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 691 AutoMutex lock(mHardwareLock); 692 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 693 694 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 695 if (dev->canSetMasterMute()) { 696 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 697 } 698 mHardwareStatus = AUDIO_HW_IDLE; 699 } 700 701 // Now set the master mute in each playback thread. Playback threads 702 // assigned to HALs which do not have master mute support will apply master 703 // mute during the mix operation. Threads with HALs which do support master 704 // mute will simply ignore the setting. 705 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 706 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 707 708 return NO_ERROR; 709} 710 711float AudioFlinger::masterVolume() const 712{ 713 Mutex::Autolock _l(mLock); 714 return masterVolume_l(); 715} 716 717bool AudioFlinger::masterMute() const 718{ 719 Mutex::Autolock _l(mLock); 720 return masterMute_l(); 721} 722 723float AudioFlinger::masterVolume_l() const 724{ 725 return mMasterVolume; 726} 727 728bool AudioFlinger::masterMute_l() const 729{ 730 return mMasterMute; 731} 732 733status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 734 audio_io_handle_t output) 735{ 736 // check calling permissions 737 if (!settingsAllowed()) { 738 return PERMISSION_DENIED; 739 } 740 741 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 742 ALOGE("setStreamVolume() invalid stream %d", stream); 743 return BAD_VALUE; 744 } 745 746 AutoMutex lock(mLock); 747 PlaybackThread *thread = NULL; 748 if (output) { 749 thread = checkPlaybackThread_l(output); 750 if (thread == NULL) { 751 return BAD_VALUE; 752 } 753 } 754 755 mStreamTypes[stream].volume = value; 756 757 if (thread == NULL) { 758 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 759 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 760 } 761 } else { 762 thread->setStreamVolume(stream, value); 763 } 764 765 return NO_ERROR; 766} 767 768status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 769{ 770 // check calling permissions 771 if (!settingsAllowed()) { 772 return PERMISSION_DENIED; 773 } 774 775 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 776 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 777 ALOGE("setStreamMute() invalid stream %d", stream); 778 return BAD_VALUE; 779 } 780 781 AutoMutex lock(mLock); 782 mStreamTypes[stream].mute = muted; 783 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 784 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 785 786 return NO_ERROR; 787} 788 789float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 790{ 791 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 792 return 0.0f; 793 } 794 795 AutoMutex lock(mLock); 796 float volume; 797 if (output) { 798 PlaybackThread *thread = checkPlaybackThread_l(output); 799 if (thread == NULL) { 800 return 0.0f; 801 } 802 volume = thread->streamVolume(stream); 803 } else { 804 volume = streamVolume_l(stream); 805 } 806 807 return volume; 808} 809 810bool AudioFlinger::streamMute(audio_stream_type_t stream) const 811{ 812 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 813 return true; 814 } 815 816 AutoMutex lock(mLock); 817 return streamMute_l(stream); 818} 819 820status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 821{ 822 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 823 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 824 825 // check calling permissions 826 if (!settingsAllowed()) { 827 return PERMISSION_DENIED; 828 } 829 830 // ioHandle == 0 means the parameters are global to the audio hardware interface 831 if (ioHandle == 0) { 832 Mutex::Autolock _l(mLock); 833 status_t final_result = NO_ERROR; 834 { 835 AutoMutex lock(mHardwareLock); 836 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 837 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 838 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 839 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 840 final_result = result ?: final_result; 841 } 842 mHardwareStatus = AUDIO_HW_IDLE; 843 } 844 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 845 AudioParameter param = AudioParameter(keyValuePairs); 846 String8 value; 847 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 848 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 849 if (mBtNrecIsOff != btNrecIsOff) { 850 for (size_t i = 0; i < mRecordThreads.size(); i++) { 851 sp<RecordThread> thread = mRecordThreads.valueAt(i); 852 audio_devices_t device = thread->inDevice(); 853 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 854 // collect all of the thread's session IDs 855 KeyedVector<int, bool> ids = thread->sessionIds(); 856 // suspend effects associated with those session IDs 857 for (size_t j = 0; j < ids.size(); ++j) { 858 int sessionId = ids.keyAt(j); 859 thread->setEffectSuspended(FX_IID_AEC, 860 suspend, 861 sessionId); 862 thread->setEffectSuspended(FX_IID_NS, 863 suspend, 864 sessionId); 865 } 866 } 867 mBtNrecIsOff = btNrecIsOff; 868 } 869 } 870 String8 screenState; 871 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 872 bool isOff = screenState == "off"; 873 if (isOff != (AudioFlinger::mScreenState & 1)) { 874 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 875 } 876 } 877 return final_result; 878 } 879 880 // hold a strong ref on thread in case closeOutput() or closeInput() is called 881 // and the thread is exited once the lock is released 882 sp<ThreadBase> thread; 883 { 884 Mutex::Autolock _l(mLock); 885 thread = checkPlaybackThread_l(ioHandle); 886 if (thread == 0) { 887 thread = checkRecordThread_l(ioHandle); 888 } else if (thread == primaryPlaybackThread_l()) { 889 // indicate output device change to all input threads for pre processing 890 AudioParameter param = AudioParameter(keyValuePairs); 891 int value; 892 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 893 (value != 0)) { 894 for (size_t i = 0; i < mRecordThreads.size(); i++) { 895 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 896 } 897 } 898 } 899 } 900 if (thread != 0) { 901 return thread->setParameters(keyValuePairs); 902 } 903 return BAD_VALUE; 904} 905 906String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 907{ 908 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 909 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 910 911 Mutex::Autolock _l(mLock); 912 913 if (ioHandle == 0) { 914 String8 out_s8; 915 916 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 917 char *s; 918 { 919 AutoMutex lock(mHardwareLock); 920 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 921 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 922 s = dev->get_parameters(dev, keys.string()); 923 mHardwareStatus = AUDIO_HW_IDLE; 924 } 925 out_s8 += String8(s ? s : ""); 926 free(s); 927 } 928 return out_s8; 929 } 930 931 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 932 if (playbackThread != NULL) { 933 return playbackThread->getParameters(keys); 934 } 935 RecordThread *recordThread = checkRecordThread_l(ioHandle); 936 if (recordThread != NULL) { 937 return recordThread->getParameters(keys); 938 } 939 return String8(""); 940} 941 942size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 943 audio_channel_mask_t channelMask) const 944{ 945 status_t ret = initCheck(); 946 if (ret != NO_ERROR) { 947 return 0; 948 } 949 950 AutoMutex lock(mHardwareLock); 951 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 952 struct audio_config config = { 953 sample_rate: sampleRate, 954 channel_mask: channelMask, 955 format: format, 956 }; 957 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 958 size_t size = dev->get_input_buffer_size(dev, &config); 959 mHardwareStatus = AUDIO_HW_IDLE; 960 return size; 961} 962 963unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 964{ 965 Mutex::Autolock _l(mLock); 966 967 RecordThread *recordThread = checkRecordThread_l(ioHandle); 968 if (recordThread != NULL) { 969 return recordThread->getInputFramesLost(); 970 } 971 return 0; 972} 973 974status_t AudioFlinger::setVoiceVolume(float value) 975{ 976 status_t ret = initCheck(); 977 if (ret != NO_ERROR) { 978 return ret; 979 } 980 981 // check calling permissions 982 if (!settingsAllowed()) { 983 return PERMISSION_DENIED; 984 } 985 986 AutoMutex lock(mHardwareLock); 987 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 988 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 989 ret = dev->set_voice_volume(dev, value); 990 mHardwareStatus = AUDIO_HW_IDLE; 991 992 return ret; 993} 994 995status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 996 audio_io_handle_t output) const 997{ 998 status_t status; 999 1000 Mutex::Autolock _l(mLock); 1001 1002 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1003 if (playbackThread != NULL) { 1004 return playbackThread->getRenderPosition(halFrames, dspFrames); 1005 } 1006 1007 return BAD_VALUE; 1008} 1009 1010void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1011{ 1012 1013 Mutex::Autolock _l(mLock); 1014 1015 pid_t pid = IPCThreadState::self()->getCallingPid(); 1016 if (mNotificationClients.indexOfKey(pid) < 0) { 1017 sp<NotificationClient> notificationClient = new NotificationClient(this, 1018 client, 1019 pid); 1020 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1021 1022 mNotificationClients.add(pid, notificationClient); 1023 1024 sp<IBinder> binder = client->asBinder(); 1025 binder->linkToDeath(notificationClient); 1026 1027 // the config change is always sent from playback or record threads to avoid deadlock 1028 // with AudioSystem::gLock 1029 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1030 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1031 } 1032 1033 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1034 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1035 } 1036 } 1037} 1038 1039void AudioFlinger::removeNotificationClient(pid_t pid) 1040{ 1041 Mutex::Autolock _l(mLock); 1042 1043 mNotificationClients.removeItem(pid); 1044 1045 ALOGV("%d died, releasing its sessions", pid); 1046 size_t num = mAudioSessionRefs.size(); 1047 bool removed = false; 1048 for (size_t i = 0; i< num; ) { 1049 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1050 ALOGV(" pid %d @ %d", ref->mPid, i); 1051 if (ref->mPid == pid) { 1052 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1053 mAudioSessionRefs.removeAt(i); 1054 delete ref; 1055 removed = true; 1056 num--; 1057 } else { 1058 i++; 1059 } 1060 } 1061 if (removed) { 1062 purgeStaleEffects_l(); 1063 } 1064} 1065 1066// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1067void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1068{ 1069 size_t size = mNotificationClients.size(); 1070 for (size_t i = 0; i < size; i++) { 1071 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1072 param2); 1073 } 1074} 1075 1076// removeClient_l() must be called with AudioFlinger::mLock held 1077void AudioFlinger::removeClient_l(pid_t pid) 1078{ 1079 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1080 IPCThreadState::self()->getCallingPid()); 1081 mClients.removeItem(pid); 1082} 1083 1084// getEffectThread_l() must be called with AudioFlinger::mLock held 1085sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1086{ 1087 sp<PlaybackThread> thread; 1088 1089 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1090 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1091 ALOG_ASSERT(thread == 0); 1092 thread = mPlaybackThreads.valueAt(i); 1093 } 1094 } 1095 1096 return thread; 1097} 1098 1099 1100 1101// ---------------------------------------------------------------------------- 1102 1103AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1104 : RefBase(), 1105 mAudioFlinger(audioFlinger), 1106 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1107 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1108 mPid(pid), 1109 mTimedTrackCount(0) 1110{ 1111 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1112} 1113 1114// Client destructor must be called with AudioFlinger::mLock held 1115AudioFlinger::Client::~Client() 1116{ 1117 mAudioFlinger->removeClient_l(mPid); 1118} 1119 1120sp<MemoryDealer> AudioFlinger::Client::heap() const 1121{ 1122 return mMemoryDealer; 1123} 1124 1125// Reserve one of the limited slots for a timed audio track associated 1126// with this client 1127bool AudioFlinger::Client::reserveTimedTrack() 1128{ 1129 const int kMaxTimedTracksPerClient = 4; 1130 1131 Mutex::Autolock _l(mTimedTrackLock); 1132 1133 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1134 ALOGW("can not create timed track - pid %d has exceeded the limit", 1135 mPid); 1136 return false; 1137 } 1138 1139 mTimedTrackCount++; 1140 return true; 1141} 1142 1143// Release a slot for a timed audio track 1144void AudioFlinger::Client::releaseTimedTrack() 1145{ 1146 Mutex::Autolock _l(mTimedTrackLock); 1147 mTimedTrackCount--; 1148} 1149 1150// ---------------------------------------------------------------------------- 1151 1152AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1153 const sp<IAudioFlingerClient>& client, 1154 pid_t pid) 1155 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1156{ 1157} 1158 1159AudioFlinger::NotificationClient::~NotificationClient() 1160{ 1161} 1162 1163void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1164{ 1165 sp<NotificationClient> keep(this); 1166 mAudioFlinger->removeNotificationClient(mPid); 1167} 1168 1169 1170// ---------------------------------------------------------------------------- 1171 1172sp<IAudioRecord> AudioFlinger::openRecord( 1173 audio_io_handle_t input, 1174 uint32_t sampleRate, 1175 audio_format_t format, 1176 audio_channel_mask_t channelMask, 1177 size_t frameCount, 1178 IAudioFlinger::track_flags_t flags, 1179 pid_t tid, 1180 int *sessionId, 1181 status_t *status) 1182{ 1183 sp<RecordThread::RecordTrack> recordTrack; 1184 sp<RecordHandle> recordHandle; 1185 sp<Client> client; 1186 status_t lStatus; 1187 RecordThread *thread; 1188 size_t inFrameCount; 1189 int lSessionId; 1190 1191 // check calling permissions 1192 if (!recordingAllowed()) { 1193 lStatus = PERMISSION_DENIED; 1194 goto Exit; 1195 } 1196 1197 // add client to list 1198 { // scope for mLock 1199 Mutex::Autolock _l(mLock); 1200 thread = checkRecordThread_l(input); 1201 if (thread == NULL) { 1202 lStatus = BAD_VALUE; 1203 goto Exit; 1204 } 1205 1206 pid_t pid = IPCThreadState::self()->getCallingPid(); 1207 client = registerPid_l(pid); 1208 1209 // If no audio session id is provided, create one here 1210 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1211 lSessionId = *sessionId; 1212 } else { 1213 lSessionId = nextUniqueId(); 1214 if (sessionId != NULL) { 1215 *sessionId = lSessionId; 1216 } 1217 } 1218 // create new record track. 1219 // The record track uses one track in mHardwareMixerThread by convention. 1220 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1221 frameCount, lSessionId, flags, tid, &lStatus); 1222 } 1223 if (lStatus != NO_ERROR) { 1224 // remove local strong reference to Client before deleting the RecordTrack so that the 1225 // Client destructor is called by the TrackBase destructor with mLock held 1226 client.clear(); 1227 recordTrack.clear(); 1228 goto Exit; 1229 } 1230 1231 // return to handle to client 1232 recordHandle = new RecordHandle(recordTrack); 1233 lStatus = NO_ERROR; 1234 1235Exit: 1236 if (status) { 1237 *status = lStatus; 1238 } 1239 return recordHandle; 1240} 1241 1242 1243 1244// ---------------------------------------------------------------------------- 1245 1246audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1247{ 1248 if (!settingsAllowed()) { 1249 return 0; 1250 } 1251 Mutex::Autolock _l(mLock); 1252 return loadHwModule_l(name); 1253} 1254 1255// loadHwModule_l() must be called with AudioFlinger::mLock held 1256audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1257{ 1258 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1259 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1260 ALOGW("loadHwModule() module %s already loaded", name); 1261 return mAudioHwDevs.keyAt(i); 1262 } 1263 } 1264 1265 audio_hw_device_t *dev; 1266 1267 int rc = load_audio_interface(name, &dev); 1268 if (rc) { 1269 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1270 return 0; 1271 } 1272 1273 mHardwareStatus = AUDIO_HW_INIT; 1274 rc = dev->init_check(dev); 1275 mHardwareStatus = AUDIO_HW_IDLE; 1276 if (rc) { 1277 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1278 return 0; 1279 } 1280 1281 // Check and cache this HAL's level of support for master mute and master 1282 // volume. If this is the first HAL opened, and it supports the get 1283 // methods, use the initial values provided by the HAL as the current 1284 // master mute and volume settings. 1285 1286 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1287 { // scope for auto-lock pattern 1288 AutoMutex lock(mHardwareLock); 1289 1290 if (0 == mAudioHwDevs.size()) { 1291 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1292 if (NULL != dev->get_master_volume) { 1293 float mv; 1294 if (OK == dev->get_master_volume(dev, &mv)) { 1295 mMasterVolume = mv; 1296 } 1297 } 1298 1299 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1300 if (NULL != dev->get_master_mute) { 1301 bool mm; 1302 if (OK == dev->get_master_mute(dev, &mm)) { 1303 mMasterMute = mm; 1304 } 1305 } 1306 } 1307 1308 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1309 if ((NULL != dev->set_master_volume) && 1310 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1311 flags = static_cast<AudioHwDevice::Flags>(flags | 1312 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1313 } 1314 1315 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1316 if ((NULL != dev->set_master_mute) && 1317 (OK == dev->set_master_mute(dev, mMasterMute))) { 1318 flags = static_cast<AudioHwDevice::Flags>(flags | 1319 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1320 } 1321 1322 mHardwareStatus = AUDIO_HW_IDLE; 1323 } 1324 1325 audio_module_handle_t handle = nextUniqueId(); 1326 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1327 1328 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1329 name, dev->common.module->name, dev->common.module->id, handle); 1330 1331 return handle; 1332 1333} 1334 1335// ---------------------------------------------------------------------------- 1336 1337uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1338{ 1339 Mutex::Autolock _l(mLock); 1340 PlaybackThread *thread = primaryPlaybackThread_l(); 1341 return thread != NULL ? thread->sampleRate() : 0; 1342} 1343 1344size_t AudioFlinger::getPrimaryOutputFrameCount() 1345{ 1346 Mutex::Autolock _l(mLock); 1347 PlaybackThread *thread = primaryPlaybackThread_l(); 1348 return thread != NULL ? thread->frameCountHAL() : 0; 1349} 1350 1351// ---------------------------------------------------------------------------- 1352 1353audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1354 audio_devices_t *pDevices, 1355 uint32_t *pSamplingRate, 1356 audio_format_t *pFormat, 1357 audio_channel_mask_t *pChannelMask, 1358 uint32_t *pLatencyMs, 1359 audio_output_flags_t flags) 1360{ 1361 status_t status; 1362 PlaybackThread *thread = NULL; 1363 struct audio_config config = { 1364 sample_rate: pSamplingRate ? *pSamplingRate : 0, 1365 channel_mask: pChannelMask ? *pChannelMask : 0, 1366 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 1367 }; 1368 audio_stream_out_t *outStream = NULL; 1369 AudioHwDevice *outHwDev; 1370 1371 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 1372 module, 1373 (pDevices != NULL) ? *pDevices : 0, 1374 config.sample_rate, 1375 config.format, 1376 config.channel_mask, 1377 flags); 1378 1379 if (pDevices == NULL || *pDevices == 0) { 1380 return 0; 1381 } 1382 1383 Mutex::Autolock _l(mLock); 1384 1385 outHwDev = findSuitableHwDev_l(module, *pDevices); 1386 if (outHwDev == NULL) 1387 return 0; 1388 1389 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1390 audio_io_handle_t id = nextUniqueId(); 1391 1392 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1393 1394 status = hwDevHal->open_output_stream(hwDevHal, 1395 id, 1396 *pDevices, 1397 (audio_output_flags_t)flags, 1398 &config, 1399 &outStream); 1400 1401 mHardwareStatus = AUDIO_HW_IDLE; 1402 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 1403 "Channels %x, status %d", 1404 outStream, 1405 config.sample_rate, 1406 config.format, 1407 config.channel_mask, 1408 status); 1409 1410 if (status == NO_ERROR && outStream != NULL) { 1411 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 1412 1413 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1414 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1415 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1416 thread = new DirectOutputThread(this, output, id, *pDevices); 1417 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1418 } else { 1419 thread = new MixerThread(this, output, id, *pDevices); 1420 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1421 } 1422 mPlaybackThreads.add(id, thread); 1423 1424 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 1425 if (pFormat != NULL) *pFormat = config.format; 1426 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 1427 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 1428 1429 // notify client processes of the new output creation 1430 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1431 1432 // the first primary output opened designates the primary hw device 1433 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1434 ALOGI("Using module %d has the primary audio interface", module); 1435 mPrimaryHardwareDev = outHwDev; 1436 1437 AutoMutex lock(mHardwareLock); 1438 mHardwareStatus = AUDIO_HW_SET_MODE; 1439 hwDevHal->set_mode(hwDevHal, mMode); 1440 mHardwareStatus = AUDIO_HW_IDLE; 1441 } 1442 return id; 1443 } 1444 1445 return 0; 1446} 1447 1448audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1449 audio_io_handle_t output2) 1450{ 1451 Mutex::Autolock _l(mLock); 1452 MixerThread *thread1 = checkMixerThread_l(output1); 1453 MixerThread *thread2 = checkMixerThread_l(output2); 1454 1455 if (thread1 == NULL || thread2 == NULL) { 1456 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1457 output2); 1458 return 0; 1459 } 1460 1461 audio_io_handle_t id = nextUniqueId(); 1462 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1463 thread->addOutputTrack(thread2); 1464 mPlaybackThreads.add(id, thread); 1465 // notify client processes of the new output creation 1466 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1467 return id; 1468} 1469 1470status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1471{ 1472 return closeOutput_nonvirtual(output); 1473} 1474 1475status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1476{ 1477 // keep strong reference on the playback thread so that 1478 // it is not destroyed while exit() is executed 1479 sp<PlaybackThread> thread; 1480 { 1481 Mutex::Autolock _l(mLock); 1482 thread = checkPlaybackThread_l(output); 1483 if (thread == NULL) { 1484 return BAD_VALUE; 1485 } 1486 1487 ALOGV("closeOutput() %d", output); 1488 1489 if (thread->type() == ThreadBase::MIXER) { 1490 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1491 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1492 DuplicatingThread *dupThread = 1493 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1494 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1495 } 1496 } 1497 } 1498 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1499 mPlaybackThreads.removeItem(output); 1500 } 1501 thread->exit(); 1502 // The thread entity (active unit of execution) is no longer running here, 1503 // but the ThreadBase container still exists. 1504 1505 if (thread->type() != ThreadBase::DUPLICATING) { 1506 AudioStreamOut *out = thread->clearOutput(); 1507 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1508 // from now on thread->mOutput is NULL 1509 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1510 delete out; 1511 } 1512 return NO_ERROR; 1513} 1514 1515status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1516{ 1517 Mutex::Autolock _l(mLock); 1518 PlaybackThread *thread = checkPlaybackThread_l(output); 1519 1520 if (thread == NULL) { 1521 return BAD_VALUE; 1522 } 1523 1524 ALOGV("suspendOutput() %d", output); 1525 thread->suspend(); 1526 1527 return NO_ERROR; 1528} 1529 1530status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1531{ 1532 Mutex::Autolock _l(mLock); 1533 PlaybackThread *thread = checkPlaybackThread_l(output); 1534 1535 if (thread == NULL) { 1536 return BAD_VALUE; 1537 } 1538 1539 ALOGV("restoreOutput() %d", output); 1540 1541 thread->restore(); 1542 1543 return NO_ERROR; 1544} 1545 1546audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1547 audio_devices_t *pDevices, 1548 uint32_t *pSamplingRate, 1549 audio_format_t *pFormat, 1550 audio_channel_mask_t *pChannelMask) 1551{ 1552 status_t status; 1553 RecordThread *thread = NULL; 1554 struct audio_config config = { 1555 sample_rate: pSamplingRate ? *pSamplingRate : 0, 1556 channel_mask: pChannelMask ? *pChannelMask : 0, 1557 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 1558 }; 1559 uint32_t reqSamplingRate = config.sample_rate; 1560 audio_format_t reqFormat = config.format; 1561 audio_channel_mask_t reqChannels = config.channel_mask; 1562 audio_stream_in_t *inStream = NULL; 1563 AudioHwDevice *inHwDev; 1564 1565 if (pDevices == NULL || *pDevices == 0) { 1566 return 0; 1567 } 1568 1569 Mutex::Autolock _l(mLock); 1570 1571 inHwDev = findSuitableHwDev_l(module, *pDevices); 1572 if (inHwDev == NULL) 1573 return 0; 1574 1575 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1576 audio_io_handle_t id = nextUniqueId(); 1577 1578 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1579 &inStream); 1580 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1581 "status %d", 1582 inStream, 1583 config.sample_rate, 1584 config.format, 1585 config.channel_mask, 1586 status); 1587 1588 // If the input could not be opened with the requested parameters and we can handle the 1589 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1590 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1591 if (status == BAD_VALUE && 1592 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1593 (config.sample_rate <= 2 * reqSamplingRate) && 1594 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 1595 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1596 inStream = NULL; 1597 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1598 } 1599 1600 if (status == NO_ERROR && inStream != NULL) { 1601 1602 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1603 // or (re-)create if current Pipe is idle and does not match the new format 1604 sp<NBAIO_Sink> teeSink; 1605#ifdef TEE_SINK_INPUT_FRAMES 1606 enum { 1607 TEE_SINK_NO, // don't copy input 1608 TEE_SINK_NEW, // copy input using a new pipe 1609 TEE_SINK_OLD, // copy input using an existing pipe 1610 } kind; 1611 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1612 popcount(inStream->common.get_channels(&inStream->common))); 1613 if (format == Format_Invalid) { 1614 kind = TEE_SINK_NO; 1615 } else if (mRecordTeeSink == 0) { 1616 kind = TEE_SINK_NEW; 1617 } else if (mRecordTeeSink->getStrongCount() != 1) { 1618 kind = TEE_SINK_NO; 1619 } else if (format == mRecordTeeSink->format()) { 1620 kind = TEE_SINK_OLD; 1621 } else { 1622 kind = TEE_SINK_NEW; 1623 } 1624 switch (kind) { 1625 case TEE_SINK_NEW: { 1626 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 1627 size_t numCounterOffers = 0; 1628 const NBAIO_Format offers[1] = {format}; 1629 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1630 ALOG_ASSERT(index == 0); 1631 PipeReader *pipeReader = new PipeReader(*pipe); 1632 numCounterOffers = 0; 1633 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1634 ALOG_ASSERT(index == 0); 1635 mRecordTeeSink = pipe; 1636 mRecordTeeSource = pipeReader; 1637 teeSink = pipe; 1638 } 1639 break; 1640 case TEE_SINK_OLD: 1641 teeSink = mRecordTeeSink; 1642 break; 1643 case TEE_SINK_NO: 1644 default: 1645 break; 1646 } 1647#endif 1648 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1649 1650 // Start record thread 1651 // RecorThread require both input and output device indication to forward to audio 1652 // pre processing modules 1653 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 1654 1655 thread = new RecordThread(this, 1656 input, 1657 reqSamplingRate, 1658 reqChannels, 1659 id, 1660 device, teeSink); 1661 mRecordThreads.add(id, thread); 1662 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1663 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 1664 if (pFormat != NULL) *pFormat = config.format; 1665 if (pChannelMask != NULL) *pChannelMask = reqChannels; 1666 1667 // notify client processes of the new input creation 1668 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1669 return id; 1670 } 1671 1672 return 0; 1673} 1674 1675status_t AudioFlinger::closeInput(audio_io_handle_t input) 1676{ 1677 return closeInput_nonvirtual(input); 1678} 1679 1680status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1681{ 1682 // keep strong reference on the record thread so that 1683 // it is not destroyed while exit() is executed 1684 sp<RecordThread> thread; 1685 { 1686 Mutex::Autolock _l(mLock); 1687 thread = checkRecordThread_l(input); 1688 if (thread == 0) { 1689 return BAD_VALUE; 1690 } 1691 1692 ALOGV("closeInput() %d", input); 1693 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1694 mRecordThreads.removeItem(input); 1695 } 1696 thread->exit(); 1697 // The thread entity (active unit of execution) is no longer running here, 1698 // but the ThreadBase container still exists. 1699 1700 AudioStreamIn *in = thread->clearInput(); 1701 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1702 // from now on thread->mInput is NULL 1703 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1704 delete in; 1705 1706 return NO_ERROR; 1707} 1708 1709status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1710{ 1711 Mutex::Autolock _l(mLock); 1712 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1713 1714 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1715 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1716 thread->invalidateTracks(stream); 1717 } 1718 1719 return NO_ERROR; 1720} 1721 1722 1723int AudioFlinger::newAudioSessionId() 1724{ 1725 return nextUniqueId(); 1726} 1727 1728void AudioFlinger::acquireAudioSessionId(int audioSession) 1729{ 1730 Mutex::Autolock _l(mLock); 1731 pid_t caller = IPCThreadState::self()->getCallingPid(); 1732 ALOGV("acquiring %d from %d", audioSession, caller); 1733 size_t num = mAudioSessionRefs.size(); 1734 for (size_t i = 0; i< num; i++) { 1735 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1736 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1737 ref->mCnt++; 1738 ALOGV(" incremented refcount to %d", ref->mCnt); 1739 return; 1740 } 1741 } 1742 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1743 ALOGV(" added new entry for %d", audioSession); 1744} 1745 1746void AudioFlinger::releaseAudioSessionId(int audioSession) 1747{ 1748 Mutex::Autolock _l(mLock); 1749 pid_t caller = IPCThreadState::self()->getCallingPid(); 1750 ALOGV("releasing %d from %d", audioSession, caller); 1751 size_t num = mAudioSessionRefs.size(); 1752 for (size_t i = 0; i< num; i++) { 1753 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1754 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1755 ref->mCnt--; 1756 ALOGV(" decremented refcount to %d", ref->mCnt); 1757 if (ref->mCnt == 0) { 1758 mAudioSessionRefs.removeAt(i); 1759 delete ref; 1760 purgeStaleEffects_l(); 1761 } 1762 return; 1763 } 1764 } 1765 ALOGW("session id %d not found for pid %d", audioSession, caller); 1766} 1767 1768void AudioFlinger::purgeStaleEffects_l() { 1769 1770 ALOGV("purging stale effects"); 1771 1772 Vector< sp<EffectChain> > chains; 1773 1774 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1775 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1776 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1777 sp<EffectChain> ec = t->mEffectChains[j]; 1778 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1779 chains.push(ec); 1780 } 1781 } 1782 } 1783 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1784 sp<RecordThread> t = mRecordThreads.valueAt(i); 1785 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1786 sp<EffectChain> ec = t->mEffectChains[j]; 1787 chains.push(ec); 1788 } 1789 } 1790 1791 for (size_t i = 0; i < chains.size(); i++) { 1792 sp<EffectChain> ec = chains[i]; 1793 int sessionid = ec->sessionId(); 1794 sp<ThreadBase> t = ec->mThread.promote(); 1795 if (t == 0) { 1796 continue; 1797 } 1798 size_t numsessionrefs = mAudioSessionRefs.size(); 1799 bool found = false; 1800 for (size_t k = 0; k < numsessionrefs; k++) { 1801 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1802 if (ref->mSessionid == sessionid) { 1803 ALOGV(" session %d still exists for %d with %d refs", 1804 sessionid, ref->mPid, ref->mCnt); 1805 found = true; 1806 break; 1807 } 1808 } 1809 if (!found) { 1810 Mutex::Autolock _l (t->mLock); 1811 // remove all effects from the chain 1812 while (ec->mEffects.size()) { 1813 sp<EffectModule> effect = ec->mEffects[0]; 1814 effect->unPin(); 1815 t->removeEffect_l(effect); 1816 if (effect->purgeHandles()) { 1817 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1818 } 1819 AudioSystem::unregisterEffect(effect->id()); 1820 } 1821 } 1822 } 1823 return; 1824} 1825 1826// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1827AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1828{ 1829 return mPlaybackThreads.valueFor(output).get(); 1830} 1831 1832// checkMixerThread_l() must be called with AudioFlinger::mLock held 1833AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1834{ 1835 PlaybackThread *thread = checkPlaybackThread_l(output); 1836 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1837} 1838 1839// checkRecordThread_l() must be called with AudioFlinger::mLock held 1840AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1841{ 1842 return mRecordThreads.valueFor(input).get(); 1843} 1844 1845uint32_t AudioFlinger::nextUniqueId() 1846{ 1847 return android_atomic_inc(&mNextUniqueId); 1848} 1849 1850AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1851{ 1852 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1853 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1854 AudioStreamOut *output = thread->getOutput(); 1855 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1856 return thread; 1857 } 1858 } 1859 return NULL; 1860} 1861 1862audio_devices_t AudioFlinger::primaryOutputDevice_l() const 1863{ 1864 PlaybackThread *thread = primaryPlaybackThread_l(); 1865 1866 if (thread == NULL) { 1867 return 0; 1868 } 1869 1870 return thread->outDevice(); 1871} 1872 1873sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 1874 int triggerSession, 1875 int listenerSession, 1876 sync_event_callback_t callBack, 1877 void *cookie) 1878{ 1879 Mutex::Autolock _l(mLock); 1880 1881 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 1882 status_t playStatus = NAME_NOT_FOUND; 1883 status_t recStatus = NAME_NOT_FOUND; 1884 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1885 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 1886 if (playStatus == NO_ERROR) { 1887 return event; 1888 } 1889 } 1890 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1891 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 1892 if (recStatus == NO_ERROR) { 1893 return event; 1894 } 1895 } 1896 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 1897 mPendingSyncEvents.add(event); 1898 } else { 1899 ALOGV("createSyncEvent() invalid event %d", event->type()); 1900 event.clear(); 1901 } 1902 return event; 1903} 1904 1905// ---------------------------------------------------------------------------- 1906// Effect management 1907// ---------------------------------------------------------------------------- 1908 1909 1910status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 1911{ 1912 Mutex::Autolock _l(mLock); 1913 return EffectQueryNumberEffects(numEffects); 1914} 1915 1916status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 1917{ 1918 Mutex::Autolock _l(mLock); 1919 return EffectQueryEffect(index, descriptor); 1920} 1921 1922status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 1923 effect_descriptor_t *descriptor) const 1924{ 1925 Mutex::Autolock _l(mLock); 1926 return EffectGetDescriptor(pUuid, descriptor); 1927} 1928 1929 1930sp<IEffect> AudioFlinger::createEffect( 1931 effect_descriptor_t *pDesc, 1932 const sp<IEffectClient>& effectClient, 1933 int32_t priority, 1934 audio_io_handle_t io, 1935 int sessionId, 1936 status_t *status, 1937 int *id, 1938 int *enabled) 1939{ 1940 status_t lStatus = NO_ERROR; 1941 sp<EffectHandle> handle; 1942 effect_descriptor_t desc; 1943 1944 pid_t pid = IPCThreadState::self()->getCallingPid(); 1945 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 1946 pid, effectClient.get(), priority, sessionId, io); 1947 1948 if (pDesc == NULL) { 1949 lStatus = BAD_VALUE; 1950 goto Exit; 1951 } 1952 1953 // check audio settings permission for global effects 1954 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 1955 lStatus = PERMISSION_DENIED; 1956 goto Exit; 1957 } 1958 1959 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 1960 // that can only be created by audio policy manager (running in same process) 1961 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 1962 lStatus = PERMISSION_DENIED; 1963 goto Exit; 1964 } 1965 1966 if (io == 0) { 1967 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1968 // output must be specified by AudioPolicyManager when using session 1969 // AUDIO_SESSION_OUTPUT_STAGE 1970 lStatus = BAD_VALUE; 1971 goto Exit; 1972 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1973 // if the output returned by getOutputForEffect() is removed before we lock the 1974 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 1975 // and we will exit safely 1976 io = AudioSystem::getOutputForEffect(&desc); 1977 } 1978 } 1979 1980 { 1981 Mutex::Autolock _l(mLock); 1982 1983 1984 if (!EffectIsNullUuid(&pDesc->uuid)) { 1985 // if uuid is specified, request effect descriptor 1986 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 1987 if (lStatus < 0) { 1988 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 1989 goto Exit; 1990 } 1991 } else { 1992 // if uuid is not specified, look for an available implementation 1993 // of the required type in effect factory 1994 if (EffectIsNullUuid(&pDesc->type)) { 1995 ALOGW("createEffect() no effect type"); 1996 lStatus = BAD_VALUE; 1997 goto Exit; 1998 } 1999 uint32_t numEffects = 0; 2000 effect_descriptor_t d; 2001 d.flags = 0; // prevent compiler warning 2002 bool found = false; 2003 2004 lStatus = EffectQueryNumberEffects(&numEffects); 2005 if (lStatus < 0) { 2006 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2007 goto Exit; 2008 } 2009 for (uint32_t i = 0; i < numEffects; i++) { 2010 lStatus = EffectQueryEffect(i, &desc); 2011 if (lStatus < 0) { 2012 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2013 continue; 2014 } 2015 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2016 // If matching type found save effect descriptor. If the session is 2017 // 0 and the effect is not auxiliary, continue enumeration in case 2018 // an auxiliary version of this effect type is available 2019 found = true; 2020 d = desc; 2021 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2022 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2023 break; 2024 } 2025 } 2026 } 2027 if (!found) { 2028 lStatus = BAD_VALUE; 2029 ALOGW("createEffect() effect not found"); 2030 goto Exit; 2031 } 2032 // For same effect type, chose auxiliary version over insert version if 2033 // connect to output mix (Compliance to OpenSL ES) 2034 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2035 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2036 desc = d; 2037 } 2038 } 2039 2040 // Do not allow auxiliary effects on a session different from 0 (output mix) 2041 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2042 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2043 lStatus = INVALID_OPERATION; 2044 goto Exit; 2045 } 2046 2047 // check recording permission for visualizer 2048 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2049 !recordingAllowed()) { 2050 lStatus = PERMISSION_DENIED; 2051 goto Exit; 2052 } 2053 2054 // return effect descriptor 2055 *pDesc = desc; 2056 2057 // If output is not specified try to find a matching audio session ID in one of the 2058 // output threads. 2059 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2060 // because of code checking output when entering the function. 2061 // Note: io is never 0 when creating an effect on an input 2062 if (io == 0) { 2063 // look for the thread where the specified audio session is present 2064 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2065 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2066 io = mPlaybackThreads.keyAt(i); 2067 break; 2068 } 2069 } 2070 if (io == 0) { 2071 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2072 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2073 io = mRecordThreads.keyAt(i); 2074 break; 2075 } 2076 } 2077 } 2078 // If no output thread contains the requested session ID, default to 2079 // first output. The effect chain will be moved to the correct output 2080 // thread when a track with the same session ID is created 2081 if (io == 0 && mPlaybackThreads.size()) { 2082 io = mPlaybackThreads.keyAt(0); 2083 } 2084 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2085 } 2086 ThreadBase *thread = checkRecordThread_l(io); 2087 if (thread == NULL) { 2088 thread = checkPlaybackThread_l(io); 2089 if (thread == NULL) { 2090 ALOGE("createEffect() unknown output thread"); 2091 lStatus = BAD_VALUE; 2092 goto Exit; 2093 } 2094 } 2095 2096 sp<Client> client = registerPid_l(pid); 2097 2098 // create effect on selected output thread 2099 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2100 &desc, enabled, &lStatus); 2101 if (handle != 0 && id != NULL) { 2102 *id = handle->id(); 2103 } 2104 } 2105 2106Exit: 2107 if (status != NULL) { 2108 *status = lStatus; 2109 } 2110 return handle; 2111} 2112 2113status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2114 audio_io_handle_t dstOutput) 2115{ 2116 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2117 sessionId, srcOutput, dstOutput); 2118 Mutex::Autolock _l(mLock); 2119 if (srcOutput == dstOutput) { 2120 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2121 return NO_ERROR; 2122 } 2123 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2124 if (srcThread == NULL) { 2125 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2126 return BAD_VALUE; 2127 } 2128 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2129 if (dstThread == NULL) { 2130 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2131 return BAD_VALUE; 2132 } 2133 2134 Mutex::Autolock _dl(dstThread->mLock); 2135 Mutex::Autolock _sl(srcThread->mLock); 2136 moveEffectChain_l(sessionId, srcThread, dstThread, false); 2137 2138 return NO_ERROR; 2139} 2140 2141// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2142status_t AudioFlinger::moveEffectChain_l(int sessionId, 2143 AudioFlinger::PlaybackThread *srcThread, 2144 AudioFlinger::PlaybackThread *dstThread, 2145 bool reRegister) 2146{ 2147 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2148 sessionId, srcThread, dstThread); 2149 2150 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2151 if (chain == 0) { 2152 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2153 sessionId, srcThread); 2154 return INVALID_OPERATION; 2155 } 2156 2157 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2158 // so that a new chain is created with correct parameters when first effect is added. This is 2159 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2160 // removed. 2161 srcThread->removeEffectChain_l(chain); 2162 2163 // transfer all effects one by one so that new effect chain is created on new thread with 2164 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2165 audio_io_handle_t dstOutput = dstThread->id(); 2166 sp<EffectChain> dstChain; 2167 uint32_t strategy = 0; // prevent compiler warning 2168 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2169 while (effect != 0) { 2170 srcThread->removeEffect_l(effect); 2171 dstThread->addEffect_l(effect); 2172 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2173 if (effect->state() == EffectModule::ACTIVE || 2174 effect->state() == EffectModule::STOPPING) { 2175 effect->start(); 2176 } 2177 // if the move request is not received from audio policy manager, the effect must be 2178 // re-registered with the new strategy and output 2179 if (dstChain == 0) { 2180 dstChain = effect->chain().promote(); 2181 if (dstChain == 0) { 2182 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2183 srcThread->addEffect_l(effect); 2184 return NO_INIT; 2185 } 2186 strategy = dstChain->strategy(); 2187 } 2188 if (reRegister) { 2189 AudioSystem::unregisterEffect(effect->id()); 2190 AudioSystem::registerEffect(&effect->desc(), 2191 dstOutput, 2192 strategy, 2193 sessionId, 2194 effect->id()); 2195 } 2196 effect = chain->getEffectFromId_l(0); 2197 } 2198 2199 return NO_ERROR; 2200} 2201 2202void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2203{ 2204 NBAIO_Source *teeSource = source.get(); 2205 if (teeSource != NULL) { 2206 char teeTime[16]; 2207 struct timeval tv; 2208 gettimeofday(&tv, NULL); 2209 struct tm tm; 2210 localtime_r(&tv.tv_sec, &tm); 2211 strftime(teeTime, sizeof(teeTime), "%T", &tm); 2212 char teePath[64]; 2213 sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); 2214 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 2215 if (teeFd >= 0) { 2216 char wavHeader[44]; 2217 memcpy(wavHeader, 2218 "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", 2219 sizeof(wavHeader)); 2220 NBAIO_Format format = teeSource->format(); 2221 unsigned channelCount = Format_channelCount(format); 2222 ALOG_ASSERT(channelCount <= FCC_2); 2223 uint32_t sampleRate = Format_sampleRate(format); 2224 wavHeader[22] = channelCount; // number of channels 2225 wavHeader[24] = sampleRate; // sample rate 2226 wavHeader[25] = sampleRate >> 8; 2227 wavHeader[32] = channelCount * 2; // block alignment 2228 write(teeFd, wavHeader, sizeof(wavHeader)); 2229 size_t total = 0; 2230 bool firstRead = true; 2231 for (;;) { 2232#define TEE_SINK_READ 1024 2233 short buffer[TEE_SINK_READ * FCC_2]; 2234 size_t count = TEE_SINK_READ; 2235 ssize_t actual = teeSource->read(buffer, count, 2236 AudioBufferProvider::kInvalidPTS); 2237 bool wasFirstRead = firstRead; 2238 firstRead = false; 2239 if (actual <= 0) { 2240 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2241 continue; 2242 } 2243 break; 2244 } 2245 ALOG_ASSERT(actual <= (ssize_t)count); 2246 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2247 total += actual; 2248 } 2249 lseek(teeFd, (off_t) 4, SEEK_SET); 2250 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2251 write(teeFd, &temp, sizeof(temp)); 2252 lseek(teeFd, (off_t) 40, SEEK_SET); 2253 temp = total * channelCount * sizeof(short); 2254 write(teeFd, &temp, sizeof(temp)); 2255 close(teeFd); 2256 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 2257 } else { 2258 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 2259 } 2260 } 2261} 2262 2263// ---------------------------------------------------------------------------- 2264 2265status_t AudioFlinger::onTransact( 2266 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2267{ 2268 return BnAudioFlinger::onTransact(code, data, reply, flags); 2269} 2270 2271}; // namespace android 2272