1 /* //device/include/server/AudioFlinger/AudioFlinger.cpp 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 <binder/Parcel.h> 31 #include <binder/IPCThreadState.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 39 #include <media/AudioTrack.h> 40 #include <media/AudioRecord.h> 41 #include <media/IMediaPlayerService.h> 42 43 #include <private/media/AudioTrackShared.h> 44 #include <private/media/AudioEffectShared.h> 45 46 #include <system/audio.h> 47 #include <hardware/audio.h> 48 49 #include "AudioMixer.h" 50 #include "AudioFlinger.h" 51 52 #include <media/EffectsFactoryApi.h> 53 #include <audio_effects/effect_visualizer.h> 54 #include <audio_effects/effect_ns.h> 55 #include <audio_effects/effect_aec.h> 56 57 #include <cpustats/ThreadCpuUsage.h> 58 #include <powermanager/PowerManager.h> 59 // #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 60 61 // ---------------------------------------------------------------------------- 62 63 64 namespace android { 65 66 static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 67 static const char* kHardwareLockedString = "Hardware lock is taken\n"; 68 69 //static const nsecs_t kStandbyTimeInNsecs = seconds(3); 70 static const float MAX_GAIN = 4096.0f; 71 static const float MAX_GAIN_INT = 0x1000; 72 73 // retry counts for buffer fill timeout 74 // 50 * ~20msecs = 1 second 75 static const int8_t kMaxTrackRetries = 50; 76 static const int8_t kMaxTrackStartupRetries = 50; 77 // allow less retry attempts on direct output thread. 78 // direct outputs can be a scarce resource in audio hardware and should 79 // be released as quickly as possible. 80 static const int8_t kMaxTrackRetriesDirect = 2; 81 82 static const int kDumpLockRetries = 50; 83 static const int kDumpLockSleep = 20000; 84 85 static const nsecs_t kWarningThrottle = seconds(5); 86 87 // RecordThread loop sleep time upon application overrun or audio HAL read error 88 static const int kRecordThreadSleepUs = 5000; 89 90 static const nsecs_t kSetParametersTimeout = seconds(2); 91 92 // minimum sleep time for the mixer thread loop when tracks are active but in underrun 93 static const uint32_t kMinThreadSleepTimeUs = 5000; 94 // maximum divider applied to the active sleep time in the mixer thread loop 95 static const uint32_t kMaxThreadSleepTimeShift = 2; 96 97 98 // ---------------------------------------------------------------------------- 99 100 static bool recordingAllowed() { 101 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 102 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 103 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 104 return ok; 105 } 106 107 static bool settingsAllowed() { 108 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 109 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 110 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 111 return ok; 112 } 113 114 // To collect the amplifier usage 115 static void addBatteryData(uint32_t params) { 116 sp<IBinder> binder = 117 defaultServiceManager()->getService(String16("media.player")); 118 sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); 119 if (service.get() == NULL) { 120 LOGW("Cannot connect to the MediaPlayerService for battery tracking"); 121 return; 122 } 123 124 service->addBatteryData(params); 125 } 126 127 static int load_audio_interface(const char *if_name, const hw_module_t **mod, 128 audio_hw_device_t **dev) 129 { 130 int rc; 131 132 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 133 if (rc) 134 goto out; 135 136 rc = audio_hw_device_open(*mod, dev); 137 LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 138 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 139 if (rc) 140 goto out; 141 142 return 0; 143 144 out: 145 *mod = NULL; 146 *dev = NULL; 147 return rc; 148 } 149 150 static const char *audio_interfaces[] = { 151 "primary", 152 "a2dp", 153 "usb", 154 }; 155 #define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 156 157 // ---------------------------------------------------------------------------- 158 159 AudioFlinger::AudioFlinger() 160 : BnAudioFlinger(), 161 mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 162 mBtNrecIsOff(false) 163 { 164 } 165 166 void AudioFlinger::onFirstRef() 167 { 168 int rc = 0; 169 170 Mutex::Autolock _l(mLock); 171 172 /* TODO: move all this work into an Init() function */ 173 mHardwareStatus = AUDIO_HW_IDLE; 174 175 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 176 const hw_module_t *mod; 177 audio_hw_device_t *dev; 178 179 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 180 if (rc) 181 continue; 182 183 LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 184 mod->name, mod->id); 185 mAudioHwDevs.push(dev); 186 187 if (!mPrimaryHardwareDev) { 188 mPrimaryHardwareDev = dev; 189 LOGI("Using '%s' (%s.%s) as the primary audio interface", 190 mod->name, mod->id, audio_interfaces[i]); 191 } 192 } 193 194 mHardwareStatus = AUDIO_HW_INIT; 195 196 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 197 LOGE("Primary audio interface not found"); 198 return; 199 } 200 201 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 202 audio_hw_device_t *dev = mAudioHwDevs[i]; 203 204 mHardwareStatus = AUDIO_HW_INIT; 205 rc = dev->init_check(dev); 206 if (rc == 0) { 207 AutoMutex lock(mHardwareLock); 208 209 mMode = AUDIO_MODE_NORMAL; 210 mHardwareStatus = AUDIO_HW_SET_MODE; 211 dev->set_mode(dev, mMode); 212 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 213 dev->set_master_volume(dev, 1.0f); 214 mHardwareStatus = AUDIO_HW_IDLE; 215 } 216 } 217 } 218 219 status_t AudioFlinger::initCheck() const 220 { 221 Mutex::Autolock _l(mLock); 222 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 223 return NO_INIT; 224 return NO_ERROR; 225 } 226 227 AudioFlinger::~AudioFlinger() 228 { 229 int num_devs = mAudioHwDevs.size(); 230 231 while (!mRecordThreads.isEmpty()) { 232 // closeInput() will remove first entry from mRecordThreads 233 closeInput(mRecordThreads.keyAt(0)); 234 } 235 while (!mPlaybackThreads.isEmpty()) { 236 // closeOutput() will remove first entry from mPlaybackThreads 237 closeOutput(mPlaybackThreads.keyAt(0)); 238 } 239 240 for (int i = 0; i < num_devs; i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 audio_hw_device_close(dev); 243 } 244 mAudioHwDevs.clear(); 245 } 246 247 audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 248 { 249 /* first matching HW device is returned */ 250 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 251 audio_hw_device_t *dev = mAudioHwDevs[i]; 252 if ((dev->get_supported_devices(dev) & devices) == devices) 253 return dev; 254 } 255 return NULL; 256 } 257 258 status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 259 { 260 const size_t SIZE = 256; 261 char buffer[SIZE]; 262 String8 result; 263 264 result.append("Clients:\n"); 265 for (size_t i = 0; i < mClients.size(); ++i) { 266 wp<Client> wClient = mClients.valueAt(i); 267 if (wClient != 0) { 268 sp<Client> client = wClient.promote(); 269 if (client != 0) { 270 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 271 result.append(buffer); 272 } 273 } 274 } 275 276 result.append("Global session refs:\n"); 277 result.append(" session pid cnt\n"); 278 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 279 AudioSessionRef *r = mAudioSessionRefs[i]; 280 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 281 result.append(buffer); 282 } 283 write(fd, result.string(), result.size()); 284 return NO_ERROR; 285 } 286 287 288 status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 289 { 290 const size_t SIZE = 256; 291 char buffer[SIZE]; 292 String8 result; 293 int hardwareStatus = mHardwareStatus; 294 295 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 296 result.append(buffer); 297 write(fd, result.string(), result.size()); 298 return NO_ERROR; 299 } 300 301 status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 302 { 303 const size_t SIZE = 256; 304 char buffer[SIZE]; 305 String8 result; 306 snprintf(buffer, SIZE, "Permission Denial: " 307 "can't dump AudioFlinger from pid=%d, uid=%d\n", 308 IPCThreadState::self()->getCallingPid(), 309 IPCThreadState::self()->getCallingUid()); 310 result.append(buffer); 311 write(fd, result.string(), result.size()); 312 return NO_ERROR; 313 } 314 315 static bool tryLock(Mutex& mutex) 316 { 317 bool locked = false; 318 for (int i = 0; i < kDumpLockRetries; ++i) { 319 if (mutex.tryLock() == NO_ERROR) { 320 locked = true; 321 break; 322 } 323 usleep(kDumpLockSleep); 324 } 325 return locked; 326 } 327 328 status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 329 { 330 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 331 dumpPermissionDenial(fd, args); 332 } else { 333 // get state of hardware lock 334 bool hardwareLocked = tryLock(mHardwareLock); 335 if (!hardwareLocked) { 336 String8 result(kHardwareLockedString); 337 write(fd, result.string(), result.size()); 338 } else { 339 mHardwareLock.unlock(); 340 } 341 342 bool locked = tryLock(mLock); 343 344 // failed to lock - AudioFlinger is probably deadlocked 345 if (!locked) { 346 String8 result(kDeadlockedString); 347 write(fd, result.string(), result.size()); 348 } 349 350 dumpClients(fd, args); 351 dumpInternals(fd, args); 352 353 // dump playback threads 354 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 355 mPlaybackThreads.valueAt(i)->dump(fd, args); 356 } 357 358 // dump record threads 359 for (size_t i = 0; i < mRecordThreads.size(); i++) { 360 mRecordThreads.valueAt(i)->dump(fd, args); 361 } 362 363 // dump all hardware devs 364 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 365 audio_hw_device_t *dev = mAudioHwDevs[i]; 366 dev->dump(dev, fd); 367 } 368 if (locked) mLock.unlock(); 369 } 370 return NO_ERROR; 371 } 372 373 374 // IAudioFlinger interface 375 376 377 sp<IAudioTrack> AudioFlinger::createTrack( 378 pid_t pid, 379 int streamType, 380 uint32_t sampleRate, 381 uint32_t format, 382 uint32_t channelMask, 383 int frameCount, 384 uint32_t flags, 385 const sp<IMemory>& sharedBuffer, 386 int output, 387 int *sessionId, 388 status_t *status) 389 { 390 sp<PlaybackThread::Track> track; 391 sp<TrackHandle> trackHandle; 392 sp<Client> client; 393 wp<Client> wclient; 394 status_t lStatus; 395 int lSessionId; 396 397 if (streamType >= AUDIO_STREAM_CNT) { 398 LOGE("invalid stream type"); 399 lStatus = BAD_VALUE; 400 goto Exit; 401 } 402 403 { 404 Mutex::Autolock _l(mLock); 405 PlaybackThread *thread = checkPlaybackThread_l(output); 406 PlaybackThread *effectThread = NULL; 407 if (thread == NULL) { 408 LOGE("unknown output thread"); 409 lStatus = BAD_VALUE; 410 goto Exit; 411 } 412 413 wclient = mClients.valueFor(pid); 414 415 if (wclient != NULL) { 416 client = wclient.promote(); 417 } else { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 423 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 424 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 425 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 426 if (mPlaybackThreads.keyAt(i) != output) { 427 // prevent same audio session on different output threads 428 uint32_t sessions = t->hasAudioSession(*sessionId); 429 if (sessions & PlaybackThread::TRACK_SESSION) { 430 lStatus = BAD_VALUE; 431 goto Exit; 432 } 433 // check if an effect with same session ID is waiting for a track to be created 434 if (sessions & PlaybackThread::EFFECT_SESSION) { 435 effectThread = t.get(); 436 } 437 } 438 } 439 lSessionId = *sessionId; 440 } else { 441 // if no audio session id is provided, create one here 442 lSessionId = nextUniqueId(); 443 if (sessionId != NULL) { 444 *sessionId = lSessionId; 445 } 446 } 447 LOGV("createTrack() lSessionId: %d", lSessionId); 448 449 track = thread->createTrack_l(client, streamType, sampleRate, format, 450 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 451 452 // move effect chain to this output thread if an effect on same session was waiting 453 // for a track to be created 454 if (lStatus == NO_ERROR && effectThread != NULL) { 455 Mutex::Autolock _dl(thread->mLock); 456 Mutex::Autolock _sl(effectThread->mLock); 457 moveEffectChain_l(lSessionId, effectThread, thread, true); 458 } 459 } 460 if (lStatus == NO_ERROR) { 461 trackHandle = new TrackHandle(track); 462 } else { 463 // remove local strong reference to Client before deleting the Track so that the Client 464 // destructor is called by the TrackBase destructor with mLock held 465 client.clear(); 466 track.clear(); 467 } 468 469 Exit: 470 if(status) { 471 *status = lStatus; 472 } 473 return trackHandle; 474 } 475 476 uint32_t AudioFlinger::sampleRate(int output) const 477 { 478 Mutex::Autolock _l(mLock); 479 PlaybackThread *thread = checkPlaybackThread_l(output); 480 if (thread == NULL) { 481 LOGW("sampleRate() unknown thread %d", output); 482 return 0; 483 } 484 return thread->sampleRate(); 485 } 486 487 int AudioFlinger::channelCount(int output) const 488 { 489 Mutex::Autolock _l(mLock); 490 PlaybackThread *thread = checkPlaybackThread_l(output); 491 if (thread == NULL) { 492 LOGW("channelCount() unknown thread %d", output); 493 return 0; 494 } 495 return thread->channelCount(); 496 } 497 498 uint32_t AudioFlinger::format(int output) const 499 { 500 Mutex::Autolock _l(mLock); 501 PlaybackThread *thread = checkPlaybackThread_l(output); 502 if (thread == NULL) { 503 LOGW("format() unknown thread %d", output); 504 return 0; 505 } 506 return thread->format(); 507 } 508 509 size_t AudioFlinger::frameCount(int output) const 510 { 511 Mutex::Autolock _l(mLock); 512 PlaybackThread *thread = checkPlaybackThread_l(output); 513 if (thread == NULL) { 514 LOGW("frameCount() unknown thread %d", output); 515 return 0; 516 } 517 return thread->frameCount(); 518 } 519 520 uint32_t AudioFlinger::latency(int output) const 521 { 522 Mutex::Autolock _l(mLock); 523 PlaybackThread *thread = checkPlaybackThread_l(output); 524 if (thread == NULL) { 525 LOGW("latency() unknown thread %d", output); 526 return 0; 527 } 528 return thread->latency(); 529 } 530 531 status_t AudioFlinger::setMasterVolume(float value) 532 { 533 status_t ret = initCheck(); 534 if (ret != NO_ERROR) { 535 return ret; 536 } 537 538 // check calling permissions 539 if (!settingsAllowed()) { 540 return PERMISSION_DENIED; 541 } 542 543 // when hw supports master volume, don't scale in sw mixer 544 { // scope for the lock 545 AutoMutex lock(mHardwareLock); 546 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 547 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 548 value = 1.0f; 549 } 550 mHardwareStatus = AUDIO_HW_IDLE; 551 } 552 553 Mutex::Autolock _l(mLock); 554 mMasterVolume = value; 555 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 556 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 557 558 return NO_ERROR; 559 } 560 561 status_t AudioFlinger::setMode(int mode) 562 { 563 status_t ret = initCheck(); 564 if (ret != NO_ERROR) { 565 return ret; 566 } 567 568 // check calling permissions 569 if (!settingsAllowed()) { 570 return PERMISSION_DENIED; 571 } 572 if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { 573 LOGW("Illegal value: setMode(%d)", mode); 574 return BAD_VALUE; 575 } 576 577 { // scope for the lock 578 AutoMutex lock(mHardwareLock); 579 mHardwareStatus = AUDIO_HW_SET_MODE; 580 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 581 mHardwareStatus = AUDIO_HW_IDLE; 582 } 583 584 if (NO_ERROR == ret) { 585 Mutex::Autolock _l(mLock); 586 mMode = mode; 587 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 588 mPlaybackThreads.valueAt(i)->setMode(mode); 589 } 590 591 return ret; 592 } 593 594 status_t AudioFlinger::setMicMute(bool state) 595 { 596 status_t ret = initCheck(); 597 if (ret != NO_ERROR) { 598 return ret; 599 } 600 601 // check calling permissions 602 if (!settingsAllowed()) { 603 return PERMISSION_DENIED; 604 } 605 606 AutoMutex lock(mHardwareLock); 607 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 608 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 609 mHardwareStatus = AUDIO_HW_IDLE; 610 return ret; 611 } 612 613 bool AudioFlinger::getMicMute() const 614 { 615 status_t ret = initCheck(); 616 if (ret != NO_ERROR) { 617 return false; 618 } 619 620 bool state = AUDIO_MODE_INVALID; 621 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 622 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 623 mHardwareStatus = AUDIO_HW_IDLE; 624 return state; 625 } 626 627 status_t AudioFlinger::setMasterMute(bool muted) 628 { 629 // check calling permissions 630 if (!settingsAllowed()) { 631 return PERMISSION_DENIED; 632 } 633 634 Mutex::Autolock _l(mLock); 635 mMasterMute = muted; 636 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 637 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 638 639 return NO_ERROR; 640 } 641 642 float AudioFlinger::masterVolume() const 643 { 644 return mMasterVolume; 645 } 646 647 bool AudioFlinger::masterMute() const 648 { 649 return mMasterMute; 650 } 651 652 status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 653 { 654 // check calling permissions 655 if (!settingsAllowed()) { 656 return PERMISSION_DENIED; 657 } 658 659 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 660 return BAD_VALUE; 661 } 662 663 AutoMutex lock(mLock); 664 PlaybackThread *thread = NULL; 665 if (output) { 666 thread = checkPlaybackThread_l(output); 667 if (thread == NULL) { 668 return BAD_VALUE; 669 } 670 } 671 672 mStreamTypes[stream].volume = value; 673 674 if (thread == NULL) { 675 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 676 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 677 } 678 } else { 679 thread->setStreamVolume(stream, value); 680 } 681 682 return NO_ERROR; 683 } 684 685 status_t AudioFlinger::setStreamMute(int stream, bool muted) 686 { 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 693 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 694 return BAD_VALUE; 695 } 696 697 AutoMutex lock(mLock); 698 mStreamTypes[stream].mute = muted; 699 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 700 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 701 702 return NO_ERROR; 703 } 704 705 float AudioFlinger::streamVolume(int stream, int output) const 706 { 707 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 708 return 0.0f; 709 } 710 711 AutoMutex lock(mLock); 712 float volume; 713 if (output) { 714 PlaybackThread *thread = checkPlaybackThread_l(output); 715 if (thread == NULL) { 716 return 0.0f; 717 } 718 volume = thread->streamVolume(stream); 719 } else { 720 volume = mStreamTypes[stream].volume; 721 } 722 723 return volume; 724 } 725 726 bool AudioFlinger::streamMute(int stream) const 727 { 728 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 729 return true; 730 } 731 732 return mStreamTypes[stream].mute; 733 } 734 735 status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 736 { 737 status_t result; 738 739 LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 740 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 741 // check calling permissions 742 if (!settingsAllowed()) { 743 return PERMISSION_DENIED; 744 } 745 746 // ioHandle == 0 means the parameters are global to the audio hardware interface 747 if (ioHandle == 0) { 748 AutoMutex lock(mHardwareLock); 749 mHardwareStatus = AUDIO_SET_PARAMETER; 750 status_t final_result = NO_ERROR; 751 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 752 audio_hw_device_t *dev = mAudioHwDevs[i]; 753 result = dev->set_parameters(dev, keyValuePairs.string()); 754 final_result = result ?: final_result; 755 } 756 mHardwareStatus = AUDIO_HW_IDLE; 757 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 758 AudioParameter param = AudioParameter(keyValuePairs); 759 String8 value; 760 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 761 Mutex::Autolock _l(mLock); 762 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 763 if (mBtNrecIsOff != btNrecIsOff) { 764 for (size_t i = 0; i < mRecordThreads.size(); i++) { 765 sp<RecordThread> thread = mRecordThreads.valueAt(i); 766 RecordThread::RecordTrack *track = thread->track(); 767 if (track != NULL) { 768 audio_devices_t device = (audio_devices_t)( 769 thread->device() & AUDIO_DEVICE_IN_ALL); 770 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 771 thread->setEffectSuspended(FX_IID_AEC, 772 suspend, 773 track->sessionId()); 774 thread->setEffectSuspended(FX_IID_NS, 775 suspend, 776 track->sessionId()); 777 } 778 } 779 mBtNrecIsOff = btNrecIsOff; 780 } 781 } 782 return final_result; 783 } 784 785 // hold a strong ref on thread in case closeOutput() or closeInput() is called 786 // and the thread is exited once the lock is released 787 sp<ThreadBase> thread; 788 { 789 Mutex::Autolock _l(mLock); 790 thread = checkPlaybackThread_l(ioHandle); 791 if (thread == NULL) { 792 thread = checkRecordThread_l(ioHandle); 793 } else if (thread.get() == primaryPlaybackThread_l()) { 794 // indicate output device change to all input threads for pre processing 795 AudioParameter param = AudioParameter(keyValuePairs); 796 int value; 797 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 798 for (size_t i = 0; i < mRecordThreads.size(); i++) { 799 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 800 } 801 } 802 } 803 } 804 if (thread != NULL) { 805 result = thread->setParameters(keyValuePairs); 806 return result; 807 } 808 return BAD_VALUE; 809 } 810 811 String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 812 { 813 // LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 814 // ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 815 816 if (ioHandle == 0) { 817 String8 out_s8; 818 819 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 820 audio_hw_device_t *dev = mAudioHwDevs[i]; 821 char *s = dev->get_parameters(dev, keys.string()); 822 out_s8 += String8(s); 823 free(s); 824 } 825 return out_s8; 826 } 827 828 Mutex::Autolock _l(mLock); 829 830 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 831 if (playbackThread != NULL) { 832 return playbackThread->getParameters(keys); 833 } 834 RecordThread *recordThread = checkRecordThread_l(ioHandle); 835 if (recordThread != NULL) { 836 return recordThread->getParameters(keys); 837 } 838 return String8(""); 839 } 840 841 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 842 { 843 status_t ret = initCheck(); 844 if (ret != NO_ERROR) { 845 return 0; 846 } 847 848 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 849 } 850 851 unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 852 { 853 if (ioHandle == 0) { 854 return 0; 855 } 856 857 Mutex::Autolock _l(mLock); 858 859 RecordThread *recordThread = checkRecordThread_l(ioHandle); 860 if (recordThread != NULL) { 861 return recordThread->getInputFramesLost(); 862 } 863 return 0; 864 } 865 866 status_t AudioFlinger::setVoiceVolume(float value) 867 { 868 status_t ret = initCheck(); 869 if (ret != NO_ERROR) { 870 return ret; 871 } 872 873 // check calling permissions 874 if (!settingsAllowed()) { 875 return PERMISSION_DENIED; 876 } 877 878 AutoMutex lock(mHardwareLock); 879 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 880 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 881 mHardwareStatus = AUDIO_HW_IDLE; 882 883 return ret; 884 } 885 886 status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 887 { 888 status_t status; 889 890 Mutex::Autolock _l(mLock); 891 892 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 893 if (playbackThread != NULL) { 894 return playbackThread->getRenderPosition(halFrames, dspFrames); 895 } 896 897 return BAD_VALUE; 898 } 899 900 void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 901 { 902 903 Mutex::Autolock _l(mLock); 904 905 int pid = IPCThreadState::self()->getCallingPid(); 906 if (mNotificationClients.indexOfKey(pid) < 0) { 907 sp<NotificationClient> notificationClient = new NotificationClient(this, 908 client, 909 pid); 910 LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 911 912 mNotificationClients.add(pid, notificationClient); 913 914 sp<IBinder> binder = client->asBinder(); 915 binder->linkToDeath(notificationClient); 916 917 // the config change is always sent from playback or record threads to avoid deadlock 918 // with AudioSystem::gLock 919 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 920 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 921 } 922 923 for (size_t i = 0; i < mRecordThreads.size(); i++) { 924 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 925 } 926 } 927 } 928 929 void AudioFlinger::removeNotificationClient(pid_t pid) 930 { 931 Mutex::Autolock _l(mLock); 932 933 int index = mNotificationClients.indexOfKey(pid); 934 if (index >= 0) { 935 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 936 LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 937 mNotificationClients.removeItem(pid); 938 } 939 940 LOGV("%d died, releasing its sessions", pid); 941 int num = mAudioSessionRefs.size(); 942 bool removed = false; 943 for (int i = 0; i< num; i++) { 944 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 945 LOGV(" pid %d @ %d", ref->pid, i); 946 if (ref->pid == pid) { 947 LOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 948 mAudioSessionRefs.removeAt(i); 949 delete ref; 950 removed = true; 951 i--; 952 num--; 953 } 954 } 955 if (removed) { 956 purgeStaleEffects_l(); 957 } 958 } 959 960 // audioConfigChanged_l() must be called with AudioFlinger::mLock held 961 void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 962 { 963 size_t size = mNotificationClients.size(); 964 for (size_t i = 0; i < size; i++) { 965 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 966 } 967 } 968 969 // removeClient_l() must be called with AudioFlinger::mLock held 970 void AudioFlinger::removeClient_l(pid_t pid) 971 { 972 LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 973 mClients.removeItem(pid); 974 } 975 976 977 // ---------------------------------------------------------------------------- 978 979 AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 980 : Thread(false), 981 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 982 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), 983 mDevice(device) 984 { 985 mDeathRecipient = new PMDeathRecipient(this); 986 } 987 988 AudioFlinger::ThreadBase::~ThreadBase() 989 { 990 mParamCond.broadcast(); 991 mNewParameters.clear(); 992 // do not lock the mutex in destructor 993 releaseWakeLock_l(); 994 if (mPowerManager != 0) { 995 sp<IBinder> binder = mPowerManager->asBinder(); 996 binder->unlinkToDeath(mDeathRecipient); 997 } 998 } 999 1000 void AudioFlinger::ThreadBase::exit() 1001 { 1002 // keep a strong ref on ourself so that we wont get 1003 // destroyed in the middle of requestExitAndWait() 1004 sp <ThreadBase> strongMe = this; 1005 1006 LOGV("ThreadBase::exit"); 1007 { 1008 AutoMutex lock(&mLock); 1009 mExiting = true; 1010 requestExit(); 1011 mWaitWorkCV.signal(); 1012 } 1013 requestExitAndWait(); 1014 } 1015 1016 uint32_t AudioFlinger::ThreadBase::sampleRate() const 1017 { 1018 return mSampleRate; 1019 } 1020 1021 int AudioFlinger::ThreadBase::channelCount() const 1022 { 1023 return (int)mChannelCount; 1024 } 1025 1026 uint32_t AudioFlinger::ThreadBase::format() const 1027 { 1028 return mFormat; 1029 } 1030 1031 size_t AudioFlinger::ThreadBase::frameCount() const 1032 { 1033 return mFrameCount; 1034 } 1035 1036 status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1037 { 1038 status_t status; 1039 1040 LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1041 Mutex::Autolock _l(mLock); 1042 1043 mNewParameters.add(keyValuePairs); 1044 mWaitWorkCV.signal(); 1045 // wait condition with timeout in case the thread loop has exited 1046 // before the request could be processed 1047 if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) { 1048 status = mParamStatus; 1049 mWaitWorkCV.signal(); 1050 } else { 1051 status = TIMED_OUT; 1052 } 1053 return status; 1054 } 1055 1056 void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1057 { 1058 Mutex::Autolock _l(mLock); 1059 sendConfigEvent_l(event, param); 1060 } 1061 1062 // sendConfigEvent_l() must be called with ThreadBase::mLock held 1063 void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1064 { 1065 ConfigEvent *configEvent = new ConfigEvent(); 1066 configEvent->mEvent = event; 1067 configEvent->mParam = param; 1068 mConfigEvents.add(configEvent); 1069 LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1070 mWaitWorkCV.signal(); 1071 } 1072 1073 void AudioFlinger::ThreadBase::processConfigEvents() 1074 { 1075 mLock.lock(); 1076 while(!mConfigEvents.isEmpty()) { 1077 LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1078 ConfigEvent *configEvent = mConfigEvents[0]; 1079 mConfigEvents.removeAt(0); 1080 // release mLock before locking AudioFlinger mLock: lock order is always 1081 // AudioFlinger then ThreadBase to avoid cross deadlock 1082 mLock.unlock(); 1083 mAudioFlinger->mLock.lock(); 1084 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 1085 mAudioFlinger->mLock.unlock(); 1086 delete configEvent; 1087 mLock.lock(); 1088 } 1089 mLock.unlock(); 1090 } 1091 1092 status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1093 { 1094 const size_t SIZE = 256; 1095 char buffer[SIZE]; 1096 String8 result; 1097 1098 bool locked = tryLock(mLock); 1099 if (!locked) { 1100 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1101 write(fd, buffer, strlen(buffer)); 1102 } 1103 1104 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1105 result.append(buffer); 1106 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1107 result.append(buffer); 1108 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1109 result.append(buffer); 1110 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1111 result.append(buffer); 1112 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1113 result.append(buffer); 1114 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1115 result.append(buffer); 1116 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1117 result.append(buffer); 1118 1119 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1120 result.append(buffer); 1121 result.append(" Index Command"); 1122 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1123 snprintf(buffer, SIZE, "\n %02d ", i); 1124 result.append(buffer); 1125 result.append(mNewParameters[i]); 1126 } 1127 1128 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1129 result.append(buffer); 1130 snprintf(buffer, SIZE, " Index event param\n"); 1131 result.append(buffer); 1132 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1133 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 1134 result.append(buffer); 1135 } 1136 result.append("\n"); 1137 1138 write(fd, result.string(), result.size()); 1139 1140 if (locked) { 1141 mLock.unlock(); 1142 } 1143 return NO_ERROR; 1144 } 1145 1146 status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1147 { 1148 const size_t SIZE = 256; 1149 char buffer[SIZE]; 1150 String8 result; 1151 1152 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1153 write(fd, buffer, strlen(buffer)); 1154 1155 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1156 sp<EffectChain> chain = mEffectChains[i]; 1157 if (chain != 0) { 1158 chain->dump(fd, args); 1159 } 1160 } 1161 return NO_ERROR; 1162 } 1163 1164 void AudioFlinger::ThreadBase::acquireWakeLock() 1165 { 1166 Mutex::Autolock _l(mLock); 1167 acquireWakeLock_l(); 1168 } 1169 1170 void AudioFlinger::ThreadBase::acquireWakeLock_l() 1171 { 1172 if (mPowerManager == 0) { 1173 // use checkService() to avoid blocking if power service is not up yet 1174 sp<IBinder> binder = 1175 defaultServiceManager()->checkService(String16("power")); 1176 if (binder == 0) { 1177 LOGW("Thread %s cannot connect to the power manager service", mName); 1178 } else { 1179 mPowerManager = interface_cast<IPowerManager>(binder); 1180 binder->linkToDeath(mDeathRecipient); 1181 } 1182 } 1183 if (mPowerManager != 0) { 1184 sp<IBinder> binder = new BBinder(); 1185 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1186 binder, 1187 String16(mName)); 1188 if (status == NO_ERROR) { 1189 mWakeLockToken = binder; 1190 } 1191 LOGV("acquireWakeLock_l() %s status %d", mName, status); 1192 } 1193 } 1194 1195 void AudioFlinger::ThreadBase::releaseWakeLock() 1196 { 1197 Mutex::Autolock _l(mLock); 1198 releaseWakeLock_l(); 1199 } 1200 1201 void AudioFlinger::ThreadBase::releaseWakeLock_l() 1202 { 1203 if (mWakeLockToken != 0) { 1204 LOGV("releaseWakeLock_l() %s", mName); 1205 if (mPowerManager != 0) { 1206 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1207 } 1208 mWakeLockToken.clear(); 1209 } 1210 } 1211 1212 void AudioFlinger::ThreadBase::clearPowerManager() 1213 { 1214 Mutex::Autolock _l(mLock); 1215 releaseWakeLock_l(); 1216 mPowerManager.clear(); 1217 } 1218 1219 void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1220 { 1221 sp<ThreadBase> thread = mThread.promote(); 1222 if (thread != 0) { 1223 thread->clearPowerManager(); 1224 } 1225 LOGW("power manager service died !!!"); 1226 } 1227 1228 void AudioFlinger::ThreadBase::setEffectSuspended( 1229 const effect_uuid_t *type, bool suspend, int sessionId) 1230 { 1231 Mutex::Autolock _l(mLock); 1232 setEffectSuspended_l(type, suspend, sessionId); 1233 } 1234 1235 void AudioFlinger::ThreadBase::setEffectSuspended_l( 1236 const effect_uuid_t *type, bool suspend, int sessionId) 1237 { 1238 sp<EffectChain> chain; 1239 chain = getEffectChain_l(sessionId); 1240 if (chain != 0) { 1241 if (type != NULL) { 1242 chain->setEffectSuspended_l(type, suspend); 1243 } else { 1244 chain->setEffectSuspendedAll_l(suspend); 1245 } 1246 } 1247 1248 updateSuspendedSessions_l(type, suspend, sessionId); 1249 } 1250 1251 void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1252 { 1253 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1254 if (index < 0) { 1255 return; 1256 } 1257 1258 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1259 mSuspendedSessions.editValueAt(index); 1260 1261 for (size_t i = 0; i < sessionEffects.size(); i++) { 1262 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1263 for (int j = 0; j < desc->mRefCount; j++) { 1264 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1265 chain->setEffectSuspendedAll_l(true); 1266 } else { 1267 LOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1268 desc->mType.timeLow); 1269 chain->setEffectSuspended_l(&desc->mType, true); 1270 } 1271 } 1272 } 1273 } 1274 1275 void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1276 bool suspend, 1277 int sessionId) 1278 { 1279 int index = mSuspendedSessions.indexOfKey(sessionId); 1280 1281 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1282 1283 if (suspend) { 1284 if (index >= 0) { 1285 sessionEffects = mSuspendedSessions.editValueAt(index); 1286 } else { 1287 mSuspendedSessions.add(sessionId, sessionEffects); 1288 } 1289 } else { 1290 if (index < 0) { 1291 return; 1292 } 1293 sessionEffects = mSuspendedSessions.editValueAt(index); 1294 } 1295 1296 1297 int key = EffectChain::kKeyForSuspendAll; 1298 if (type != NULL) { 1299 key = type->timeLow; 1300 } 1301 index = sessionEffects.indexOfKey(key); 1302 1303 sp <SuspendedSessionDesc> desc; 1304 if (suspend) { 1305 if (index >= 0) { 1306 desc = sessionEffects.valueAt(index); 1307 } else { 1308 desc = new SuspendedSessionDesc(); 1309 if (type != NULL) { 1310 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1311 } 1312 sessionEffects.add(key, desc); 1313 LOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1314 } 1315 desc->mRefCount++; 1316 } else { 1317 if (index < 0) { 1318 return; 1319 } 1320 desc = sessionEffects.valueAt(index); 1321 if (--desc->mRefCount == 0) { 1322 LOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1323 sessionEffects.removeItemsAt(index); 1324 if (sessionEffects.isEmpty()) { 1325 LOGV("updateSuspendedSessions_l() restore removing session %d", 1326 sessionId); 1327 mSuspendedSessions.removeItem(sessionId); 1328 } 1329 } 1330 } 1331 if (!sessionEffects.isEmpty()) { 1332 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1333 } 1334 } 1335 1336 void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1337 bool enabled, 1338 int sessionId) 1339 { 1340 Mutex::Autolock _l(mLock); 1341 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1342 } 1343 1344 void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1345 bool enabled, 1346 int sessionId) 1347 { 1348 if (mType != RECORD) { 1349 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1350 // another session. This gives the priority to well behaved effect control panels 1351 // and applications not using global effects. 1352 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1353 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1354 } 1355 } 1356 1357 sp<EffectChain> chain = getEffectChain_l(sessionId); 1358 if (chain != 0) { 1359 chain->checkSuspendOnEffectEnabled(effect, enabled); 1360 } 1361 } 1362 1363 // ---------------------------------------------------------------------------- 1364 1365 AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1366 AudioStreamOut* output, 1367 int id, 1368 uint32_t device) 1369 : ThreadBase(audioFlinger, id, device), 1370 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1371 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1372 { 1373 snprintf(mName, kNameLength, "AudioOut_%d", id); 1374 1375 readOutputParameters(); 1376 1377 mMasterVolume = mAudioFlinger->masterVolume(); 1378 mMasterMute = mAudioFlinger->masterMute(); 1379 1380 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1381 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1382 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1383 mStreamTypes[stream].valid = true; 1384 } 1385 } 1386 1387 AudioFlinger::PlaybackThread::~PlaybackThread() 1388 { 1389 delete [] mMixBuffer; 1390 } 1391 1392 status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1393 { 1394 dumpInternals(fd, args); 1395 dumpTracks(fd, args); 1396 dumpEffectChains(fd, args); 1397 return NO_ERROR; 1398 } 1399 1400 status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1401 { 1402 const size_t SIZE = 256; 1403 char buffer[SIZE]; 1404 String8 result; 1405 1406 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1407 result.append(buffer); 1408 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1409 for (size_t i = 0; i < mTracks.size(); ++i) { 1410 sp<Track> track = mTracks[i]; 1411 if (track != 0) { 1412 track->dump(buffer, SIZE); 1413 result.append(buffer); 1414 } 1415 } 1416 1417 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1418 result.append(buffer); 1419 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1420 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1421 wp<Track> wTrack = mActiveTracks[i]; 1422 if (wTrack != 0) { 1423 sp<Track> track = wTrack.promote(); 1424 if (track != 0) { 1425 track->dump(buffer, SIZE); 1426 result.append(buffer); 1427 } 1428 } 1429 } 1430 write(fd, result.string(), result.size()); 1431 return NO_ERROR; 1432 } 1433 1434 status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1435 { 1436 const size_t SIZE = 256; 1437 char buffer[SIZE]; 1438 String8 result; 1439 1440 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1441 result.append(buffer); 1442 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1443 result.append(buffer); 1444 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1445 result.append(buffer); 1446 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1447 result.append(buffer); 1448 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1449 result.append(buffer); 1450 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1451 result.append(buffer); 1452 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1453 result.append(buffer); 1454 write(fd, result.string(), result.size()); 1455 1456 dumpBase(fd, args); 1457 1458 return NO_ERROR; 1459 } 1460 1461 // Thread virtuals 1462 status_t AudioFlinger::PlaybackThread::readyToRun() 1463 { 1464 status_t status = initCheck(); 1465 if (status == NO_ERROR) { 1466 LOGI("AudioFlinger's thread %p ready to run", this); 1467 } else { 1468 LOGE("No working audio driver found."); 1469 } 1470 return status; 1471 } 1472 1473 void AudioFlinger::PlaybackThread::onFirstRef() 1474 { 1475 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1476 } 1477 1478 // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1479 sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1480 const sp<AudioFlinger::Client>& client, 1481 int streamType, 1482 uint32_t sampleRate, 1483 uint32_t format, 1484 uint32_t channelMask, 1485 int frameCount, 1486 const sp<IMemory>& sharedBuffer, 1487 int sessionId, 1488 status_t *status) 1489 { 1490 sp<Track> track; 1491 status_t lStatus; 1492 1493 if (mType == DIRECT) { 1494 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1495 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1496 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1497 "for output %p with format %d", 1498 sampleRate, format, channelMask, mOutput, mFormat); 1499 lStatus = BAD_VALUE; 1500 goto Exit; 1501 } 1502 } 1503 } else { 1504 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1505 if (sampleRate > mSampleRate*2) { 1506 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1507 lStatus = BAD_VALUE; 1508 goto Exit; 1509 } 1510 } 1511 1512 lStatus = initCheck(); 1513 if (lStatus != NO_ERROR) { 1514 LOGE("Audio driver not initialized."); 1515 goto Exit; 1516 } 1517 1518 { // scope for mLock 1519 Mutex::Autolock _l(mLock); 1520 1521 // all tracks in same audio session must share the same routing strategy otherwise 1522 // conflicts will happen when tracks are moved from one output to another by audio policy 1523 // manager 1524 uint32_t strategy = 1525 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1526 for (size_t i = 0; i < mTracks.size(); ++i) { 1527 sp<Track> t = mTracks[i]; 1528 if (t != 0) { 1529 if (sessionId == t->sessionId() && 1530 strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) { 1531 lStatus = BAD_VALUE; 1532 goto Exit; 1533 } 1534 } 1535 } 1536 1537 track = new Track(this, client, streamType, sampleRate, format, 1538 channelMask, frameCount, sharedBuffer, sessionId); 1539 if (track->getCblk() == NULL || track->name() < 0) { 1540 lStatus = NO_MEMORY; 1541 goto Exit; 1542 } 1543 mTracks.add(track); 1544 1545 sp<EffectChain> chain = getEffectChain_l(sessionId); 1546 if (chain != 0) { 1547 LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1548 track->setMainBuffer(chain->inBuffer()); 1549 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1550 chain->incTrackCnt(); 1551 } 1552 1553 // invalidate track immediately if the stream type was moved to another thread since 1554 // createTrack() was called by the client process. 1555 if (!mStreamTypes[streamType].valid) { 1556 LOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1557 this, streamType); 1558 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1559 } 1560 } 1561 lStatus = NO_ERROR; 1562 1563 Exit: 1564 if(status) { 1565 *status = lStatus; 1566 } 1567 return track; 1568 } 1569 1570 uint32_t AudioFlinger::PlaybackThread::latency() const 1571 { 1572 Mutex::Autolock _l(mLock); 1573 if (initCheck() == NO_ERROR) { 1574 return mOutput->stream->get_latency(mOutput->stream); 1575 } else { 1576 return 0; 1577 } 1578 } 1579 1580 status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1581 { 1582 mMasterVolume = value; 1583 return NO_ERROR; 1584 } 1585 1586 status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1587 { 1588 mMasterMute = muted; 1589 return NO_ERROR; 1590 } 1591 1592 float AudioFlinger::PlaybackThread::masterVolume() const 1593 { 1594 return mMasterVolume; 1595 } 1596 1597 bool AudioFlinger::PlaybackThread::masterMute() const 1598 { 1599 return mMasterMute; 1600 } 1601 1602 status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1603 { 1604 mStreamTypes[stream].volume = value; 1605 return NO_ERROR; 1606 } 1607 1608 status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1609 { 1610 mStreamTypes[stream].mute = muted; 1611 return NO_ERROR; 1612 } 1613 1614 float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1615 { 1616 return mStreamTypes[stream].volume; 1617 } 1618 1619 bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1620 { 1621 return mStreamTypes[stream].mute; 1622 } 1623 1624 // addTrack_l() must be called with ThreadBase::mLock held 1625 status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1626 { 1627 status_t status = ALREADY_EXISTS; 1628 1629 // set retry count for buffer fill 1630 track->mRetryCount = kMaxTrackStartupRetries; 1631 if (mActiveTracks.indexOf(track) < 0) { 1632 // the track is newly added, make sure it fills up all its 1633 // buffers before playing. This is to ensure the client will 1634 // effectively get the latency it requested. 1635 track->mFillingUpStatus = Track::FS_FILLING; 1636 track->mResetDone = false; 1637 mActiveTracks.add(track); 1638 if (track->mainBuffer() != mMixBuffer) { 1639 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1640 if (chain != 0) { 1641 LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1642 chain->incActiveTrackCnt(); 1643 } 1644 } 1645 1646 status = NO_ERROR; 1647 } 1648 1649 LOGV("mWaitWorkCV.broadcast"); 1650 mWaitWorkCV.broadcast(); 1651 1652 return status; 1653 } 1654 1655 // destroyTrack_l() must be called with ThreadBase::mLock held 1656 void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1657 { 1658 track->mState = TrackBase::TERMINATED; 1659 if (mActiveTracks.indexOf(track) < 0) { 1660 removeTrack_l(track); 1661 } 1662 } 1663 1664 void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1665 { 1666 mTracks.remove(track); 1667 deleteTrackName_l(track->name()); 1668 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1669 if (chain != 0) { 1670 chain->decTrackCnt(); 1671 } 1672 } 1673 1674 String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1675 { 1676 String8 out_s8 = String8(""); 1677 char *s; 1678 1679 Mutex::Autolock _l(mLock); 1680 if (initCheck() != NO_ERROR) { 1681 return out_s8; 1682 } 1683 1684 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1685 out_s8 = String8(s); 1686 free(s); 1687 return out_s8; 1688 } 1689 1690 // audioConfigChanged_l() must be called with AudioFlinger::mLock held 1691 void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1692 AudioSystem::OutputDescriptor desc; 1693 void *param2 = 0; 1694 1695 LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1696 1697 switch (event) { 1698 case AudioSystem::OUTPUT_OPENED: 1699 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1700 desc.channels = mChannelMask; 1701 desc.samplingRate = mSampleRate; 1702 desc.format = mFormat; 1703 desc.frameCount = mFrameCount; 1704 desc.latency = latency(); 1705 param2 = &desc; 1706 break; 1707 1708 case AudioSystem::STREAM_CONFIG_CHANGED: 1709 param2 = ¶m; 1710 case AudioSystem::OUTPUT_CLOSED: 1711 default: 1712 break; 1713 } 1714 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1715 } 1716 1717 void AudioFlinger::PlaybackThread::readOutputParameters() 1718 { 1719 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1720 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1721 mChannelCount = (uint16_t)popcount(mChannelMask); 1722 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1723 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1724 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1725 1726 // FIXME - Current mixer implementation only supports stereo output: Always 1727 // Allocate a stereo buffer even if HW output is mono. 1728 if (mMixBuffer != NULL) delete[] mMixBuffer; 1729 mMixBuffer = new int16_t[mFrameCount * 2]; 1730 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1731 1732 // force reconfiguration of effect chains and engines to take new buffer size and audio 1733 // parameters into account 1734 // Note that mLock is not held when readOutputParameters() is called from the constructor 1735 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1736 // matter. 1737 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1738 Vector< sp<EffectChain> > effectChains = mEffectChains; 1739 for (size_t i = 0; i < effectChains.size(); i ++) { 1740 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1741 } 1742 } 1743 1744 status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1745 { 1746 if (halFrames == 0 || dspFrames == 0) { 1747 return BAD_VALUE; 1748 } 1749 Mutex::Autolock _l(mLock); 1750 if (initCheck() != NO_ERROR) { 1751 return INVALID_OPERATION; 1752 } 1753 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1754 1755 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1756 } 1757 1758 uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1759 { 1760 Mutex::Autolock _l(mLock); 1761 uint32_t result = 0; 1762 if (getEffectChain_l(sessionId) != 0) { 1763 result = EFFECT_SESSION; 1764 } 1765 1766 for (size_t i = 0; i < mTracks.size(); ++i) { 1767 sp<Track> track = mTracks[i]; 1768 if (sessionId == track->sessionId() && 1769 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1770 result |= TRACK_SESSION; 1771 break; 1772 } 1773 } 1774 1775 return result; 1776 } 1777 1778 uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1779 { 1780 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1781 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1782 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1783 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1784 } 1785 for (size_t i = 0; i < mTracks.size(); i++) { 1786 sp<Track> track = mTracks[i]; 1787 if (sessionId == track->sessionId() && 1788 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1789 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1790 } 1791 } 1792 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1793 } 1794 1795 1796 AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1797 { 1798 Mutex::Autolock _l(mLock); 1799 return mOutput; 1800 } 1801 1802 AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1803 { 1804 Mutex::Autolock _l(mLock); 1805 AudioStreamOut *output = mOutput; 1806 mOutput = NULL; 1807 return output; 1808 } 1809 1810 // this method must always be called either with ThreadBase mLock held or inside the thread loop 1811 audio_stream_t* AudioFlinger::PlaybackThread::stream() 1812 { 1813 if (mOutput == NULL) { 1814 return NULL; 1815 } 1816 return &mOutput->stream->common; 1817 } 1818 1819 uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1820 { 1821 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1822 // decoding and transfer time. So sleeping for half of the latency would likely cause 1823 // underruns 1824 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1825 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1826 } else { 1827 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1828 } 1829 } 1830 1831 // ---------------------------------------------------------------------------- 1832 1833 AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1834 : PlaybackThread(audioFlinger, output, id, device), 1835 mAudioMixer(0) 1836 { 1837 mType = ThreadBase::MIXER; 1838 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1839 1840 // FIXME - Current mixer implementation only supports stereo output 1841 if (mChannelCount == 1) { 1842 LOGE("Invalid audio hardware channel count"); 1843 } 1844 } 1845 1846 AudioFlinger::MixerThread::~MixerThread() 1847 { 1848 delete mAudioMixer; 1849 } 1850 1851 bool AudioFlinger::MixerThread::threadLoop() 1852 { 1853 Vector< sp<Track> > tracksToRemove; 1854 uint32_t mixerStatus = MIXER_IDLE; 1855 nsecs_t standbyTime = systemTime(); 1856 size_t mixBufferSize = mFrameCount * mFrameSize; 1857 // FIXME: Relaxed timing because of a certain device that can't meet latency 1858 // Should be reduced to 2x after the vendor fixes the driver issue 1859 // increase threshold again due to low power audio mode. The way this warning threshold is 1860 // calculated and its usefulness should be reconsidered anyway. 1861 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1862 nsecs_t lastWarning = 0; 1863 bool longStandbyExit = false; 1864 uint32_t activeSleepTime = activeSleepTimeUs(); 1865 uint32_t idleSleepTime = idleSleepTimeUs(); 1866 uint32_t sleepTime = idleSleepTime; 1867 uint32_t sleepTimeShift = 0; 1868 Vector< sp<EffectChain> > effectChains; 1869 #ifdef DEBUG_CPU_USAGE 1870 ThreadCpuUsage cpu; 1871 const CentralTendencyStatistics& stats = cpu.statistics(); 1872 #endif 1873 1874 acquireWakeLock(); 1875 1876 while (!exitPending()) 1877 { 1878 #ifdef DEBUG_CPU_USAGE 1879 cpu.sampleAndEnable(); 1880 unsigned n = stats.n(); 1881 // cpu.elapsed() is expensive, so don't call it every loop 1882 if ((n & 127) == 1) { 1883 long long elapsed = cpu.elapsed(); 1884 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1885 double perLoop = elapsed / (double) n; 1886 double perLoop100 = perLoop * 0.01; 1887 double mean = stats.mean(); 1888 double stddev = stats.stddev(); 1889 double minimum = stats.minimum(); 1890 double maximum = stats.maximum(); 1891 cpu.resetStatistics(); 1892 LOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1893 elapsed * .000000001, n, perLoop * .000001, 1894 mean * .001, 1895 stddev * .001, 1896 minimum * .001, 1897 maximum * .001, 1898 mean / perLoop100, 1899 stddev / perLoop100, 1900 minimum / perLoop100, 1901 maximum / perLoop100); 1902 } 1903 } 1904 #endif 1905 processConfigEvents(); 1906 1907 mixerStatus = MIXER_IDLE; 1908 { // scope for mLock 1909 1910 Mutex::Autolock _l(mLock); 1911 1912 if (checkForNewParameters_l()) { 1913 mixBufferSize = mFrameCount * mFrameSize; 1914 // FIXME: Relaxed timing because of a certain device that can't meet latency 1915 // Should be reduced to 2x after the vendor fixes the driver issue 1916 // increase threshold again due to low power audio mode. The way this warning 1917 // threshold is calculated and its usefulness should be reconsidered anyway. 1918 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1919 activeSleepTime = activeSleepTimeUs(); 1920 idleSleepTime = idleSleepTimeUs(); 1921 } 1922 1923 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1924 1925 // put audio hardware into standby after short delay 1926 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1927 mSuspended) { 1928 if (!mStandby) { 1929 LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1930 mOutput->stream->common.standby(&mOutput->stream->common); 1931 mStandby = true; 1932 mBytesWritten = 0; 1933 } 1934 1935 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1936 // we're about to wait, flush the binder command buffer 1937 IPCThreadState::self()->flushCommands(); 1938 1939 if (exitPending()) break; 1940 1941 releaseWakeLock_l(); 1942 // wait until we have something to do... 1943 LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1944 mWaitWorkCV.wait(mLock); 1945 LOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1946 acquireWakeLock_l(); 1947 1948 if (mMasterMute == false) { 1949 char value[PROPERTY_VALUE_MAX]; 1950 property_get("ro.audio.silent", value, "0"); 1951 if (atoi(value)) { 1952 LOGD("Silence is golden"); 1953 setMasterMute(true); 1954 } 1955 } 1956 1957 standbyTime = systemTime() + kStandbyTimeInNsecs; 1958 sleepTime = idleSleepTime; 1959 sleepTimeShift = 0; 1960 continue; 1961 } 1962 } 1963 1964 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1965 1966 // prevent any changes in effect chain list and in each effect chain 1967 // during mixing and effect process as the audio buffers could be deleted 1968 // or modified if an effect is created or deleted 1969 lockEffectChains_l(effectChains); 1970 } 1971 1972 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1973 // mix buffers... 1974 mAudioMixer->process(); 1975 sleepTime = 0; 1976 // increase sleep time progressively when application underrun condition clears 1977 if (sleepTimeShift > 0) { 1978 sleepTimeShift--; 1979 } 1980 standbyTime = systemTime() + kStandbyTimeInNsecs; 1981 //TODO: delay standby when effects have a tail 1982 } else { 1983 // If no tracks are ready, sleep once for the duration of an output 1984 // buffer size, then write 0s to the output 1985 if (sleepTime == 0) { 1986 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1987 sleepTime = activeSleepTime >> sleepTimeShift; 1988 if (sleepTime < kMinThreadSleepTimeUs) { 1989 sleepTime = kMinThreadSleepTimeUs; 1990 } 1991 // reduce sleep time in case of consecutive application underruns to avoid 1992 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 1993 // duration we would end up writing less data than needed by the audio HAL if 1994 // the condition persists. 1995 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 1996 sleepTimeShift++; 1997 } 1998 } else { 1999 sleepTime = idleSleepTime; 2000 } 2001 } else if (mBytesWritten != 0 || 2002 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2003 memset (mMixBuffer, 0, mixBufferSize); 2004 sleepTime = 0; 2005 LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2006 } 2007 // TODO add standby time extension fct of effect tail 2008 } 2009 2010 if (mSuspended) { 2011 sleepTime = suspendSleepTimeUs(); 2012 } 2013 // sleepTime == 0 means we must write to audio hardware 2014 if (sleepTime == 0) { 2015 for (size_t i = 0; i < effectChains.size(); i ++) { 2016 effectChains[i]->process_l(); 2017 } 2018 // enable changes in effect chain 2019 unlockEffectChains(effectChains); 2020 mLastWriteTime = systemTime(); 2021 mInWrite = true; 2022 mBytesWritten += mixBufferSize; 2023 2024 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2025 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2026 mNumWrites++; 2027 mInWrite = false; 2028 nsecs_t now = systemTime(); 2029 nsecs_t delta = now - mLastWriteTime; 2030 if (!mStandby && delta > maxPeriod) { 2031 mNumDelayedWrites++; 2032 if ((now - lastWarning) > kWarningThrottle) { 2033 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2034 ns2ms(delta), mNumDelayedWrites, this); 2035 lastWarning = now; 2036 } 2037 if (mStandby) { 2038 longStandbyExit = true; 2039 } 2040 } 2041 mStandby = false; 2042 } else { 2043 // enable changes in effect chain 2044 unlockEffectChains(effectChains); 2045 usleep(sleepTime); 2046 } 2047 2048 // finally let go of all our tracks, without the lock held 2049 // since we can't guarantee the destructors won't acquire that 2050 // same lock. 2051 tracksToRemove.clear(); 2052 2053 // Effect chains will be actually deleted here if they were removed from 2054 // mEffectChains list during mixing or effects processing 2055 effectChains.clear(); 2056 } 2057 2058 if (!mStandby) { 2059 mOutput->stream->common.standby(&mOutput->stream->common); 2060 } 2061 2062 releaseWakeLock(); 2063 2064 LOGV("MixerThread %p exiting", this); 2065 return false; 2066 } 2067 2068 // prepareTracks_l() must be called with ThreadBase::mLock held 2069 uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2070 { 2071 2072 uint32_t mixerStatus = MIXER_IDLE; 2073 // find out which tracks need to be processed 2074 size_t count = activeTracks.size(); 2075 size_t mixedTracks = 0; 2076 size_t tracksWithEffect = 0; 2077 2078 float masterVolume = mMasterVolume; 2079 bool masterMute = mMasterMute; 2080 2081 if (masterMute) { 2082 masterVolume = 0; 2083 } 2084 // Delegate master volume control to effect in output mix effect chain if needed 2085 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2086 if (chain != 0) { 2087 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2088 chain->setVolume_l(&v, &v); 2089 masterVolume = (float)((v + (1 << 23)) >> 24); 2090 chain.clear(); 2091 } 2092 2093 for (size_t i=0 ; i<count ; i++) { 2094 sp<Track> t = activeTracks[i].promote(); 2095 if (t == 0) continue; 2096 2097 Track* const track = t.get(); 2098 audio_track_cblk_t* cblk = track->cblk(); 2099 2100 // The first time a track is added we wait 2101 // for all its buffers to be filled before processing it 2102 mAudioMixer->setActiveTrack(track->name()); 2103 // make sure that we have enough frames to mix one full buffer. 2104 // enforce this condition only once to enable draining the buffer in case the client 2105 // app does not call stop() and relies on underrun to stop: 2106 // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed 2107 // during last round 2108 uint32_t minFrames = 1; 2109 if (!track->isStopped() && !track->isPausing() && 2110 (track->mRetryCount >= kMaxTrackRetries)) { 2111 if (t->sampleRate() == (int)mSampleRate) { 2112 minFrames = mFrameCount; 2113 } else { 2114 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1; 2115 } 2116 } 2117 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2118 !track->isPaused() && !track->isTerminated()) 2119 { 2120 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 2121 2122 mixedTracks++; 2123 2124 // track->mainBuffer() != mMixBuffer means there is an effect chain 2125 // connected to the track 2126 chain.clear(); 2127 if (track->mainBuffer() != mMixBuffer) { 2128 chain = getEffectChain_l(track->sessionId()); 2129 // Delegate volume control to effect in track effect chain if needed 2130 if (chain != 0) { 2131 tracksWithEffect++; 2132 } else { 2133 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 2134 track->name(), track->sessionId()); 2135 } 2136 } 2137 2138 2139 int param = AudioMixer::VOLUME; 2140 if (track->mFillingUpStatus == Track::FS_FILLED) { 2141 // no ramp for the first volume setting 2142 track->mFillingUpStatus = Track::FS_ACTIVE; 2143 if (track->mState == TrackBase::RESUMING) { 2144 track->mState = TrackBase::ACTIVE; 2145 param = AudioMixer::RAMP_VOLUME; 2146 } 2147 mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2148 } else if (cblk->server != 0) { 2149 // If the track is stopped before the first frame was mixed, 2150 // do not apply ramp 2151 param = AudioMixer::RAMP_VOLUME; 2152 } 2153 2154 // compute volume for this track 2155 uint32_t vl, vr, va; 2156 if (track->isMuted() || track->isPausing() || 2157 mStreamTypes[track->type()].mute) { 2158 vl = vr = va = 0; 2159 if (track->isPausing()) { 2160 track->setPaused(); 2161 } 2162 } else { 2163 2164 // read original volumes with volume control 2165 float typeVolume = mStreamTypes[track->type()].volume; 2166 float v = masterVolume * typeVolume; 2167 vl = (uint32_t)(v * cblk->volume[0]) << 12; 2168 vr = (uint32_t)(v * cblk->volume[1]) << 12; 2169 2170 va = (uint32_t)(v * cblk->sendLevel); 2171 } 2172 // Delegate volume control to effect in track effect chain if needed 2173 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2174 // Do not ramp volume if volume is controlled by effect 2175 param = AudioMixer::VOLUME; 2176 track->mHasVolumeController = true; 2177 } else { 2178 // force no volume ramp when volume controller was just disabled or removed 2179 // from effect chain to avoid volume spike 2180 if (track->mHasVolumeController) { 2181 param = AudioMixer::VOLUME; 2182 } 2183 track->mHasVolumeController = false; 2184 } 2185 2186 // Convert volumes from 8.24 to 4.12 format 2187 int16_t left, right, aux; 2188 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2189 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2190 left = int16_t(v_clamped); 2191 v_clamped = (vr + (1 << 11)) >> 12; 2192 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2193 right = int16_t(v_clamped); 2194 2195 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2196 aux = int16_t(va); 2197 2198 // XXX: these things DON'T need to be done each time 2199 mAudioMixer->setBufferProvider(track); 2200 mAudioMixer->enable(AudioMixer::MIXING); 2201 2202 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 2203 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 2204 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 2205 mAudioMixer->setParameter( 2206 AudioMixer::TRACK, 2207 AudioMixer::FORMAT, (void *)track->format()); 2208 mAudioMixer->setParameter( 2209 AudioMixer::TRACK, 2210 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2211 mAudioMixer->setParameter( 2212 AudioMixer::RESAMPLE, 2213 AudioMixer::SAMPLE_RATE, 2214 (void *)(cblk->sampleRate)); 2215 mAudioMixer->setParameter( 2216 AudioMixer::TRACK, 2217 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2218 mAudioMixer->setParameter( 2219 AudioMixer::TRACK, 2220 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2221 2222 // reset retry count 2223 track->mRetryCount = kMaxTrackRetries; 2224 mixerStatus = MIXER_TRACKS_READY; 2225 } else { 2226 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 2227 if (track->isStopped()) { 2228 track->reset(); 2229 } 2230 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2231 // We have consumed all the buffers of this track. 2232 // Remove it from the list of active tracks. 2233 tracksToRemove->add(track); 2234 } else { 2235 // No buffers for this track. Give it a few chances to 2236 // fill a buffer, then remove it from active list. 2237 if (--(track->mRetryCount) <= 0) { 2238 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 2239 tracksToRemove->add(track); 2240 // indicate to client process that the track was disabled because of underrun 2241 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2242 } else if (mixerStatus != MIXER_TRACKS_READY) { 2243 mixerStatus = MIXER_TRACKS_ENABLED; 2244 } 2245 } 2246 mAudioMixer->disable(AudioMixer::MIXING); 2247 } 2248 } 2249 2250 // remove all the tracks that need to be... 2251 count = tracksToRemove->size(); 2252 if (UNLIKELY(count)) { 2253 for (size_t i=0 ; i<count ; i++) { 2254 const sp<Track>& track = tracksToRemove->itemAt(i); 2255 mActiveTracks.remove(track); 2256 if (track->mainBuffer() != mMixBuffer) { 2257 chain = getEffectChain_l(track->sessionId()); 2258 if (chain != 0) { 2259 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2260 chain->decActiveTrackCnt(); 2261 } 2262 } 2263 if (track->isTerminated()) { 2264 removeTrack_l(track); 2265 } 2266 } 2267 } 2268 2269 // mix buffer must be cleared if all tracks are connected to an 2270 // effect chain as in this case the mixer will not write to 2271 // mix buffer and track effects will accumulate into it 2272 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2273 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2274 } 2275 2276 return mixerStatus; 2277 } 2278 2279 void AudioFlinger::MixerThread::invalidateTracks(int streamType) 2280 { 2281 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2282 this, streamType, mTracks.size()); 2283 Mutex::Autolock _l(mLock); 2284 2285 size_t size = mTracks.size(); 2286 for (size_t i = 0; i < size; i++) { 2287 sp<Track> t = mTracks[i]; 2288 if (t->type() == streamType) { 2289 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2290 t->mCblk->cv.signal(); 2291 } 2292 } 2293 } 2294 2295 void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid) 2296 { 2297 LOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2298 this, streamType, valid); 2299 Mutex::Autolock _l(mLock); 2300 2301 mStreamTypes[streamType].valid = valid; 2302 } 2303 2304 // getTrackName_l() must be called with ThreadBase::mLock held 2305 int AudioFlinger::MixerThread::getTrackName_l() 2306 { 2307 return mAudioMixer->getTrackName(); 2308 } 2309 2310 // deleteTrackName_l() must be called with ThreadBase::mLock held 2311 void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2312 { 2313 LOGV("remove track (%d) and delete from mixer", name); 2314 mAudioMixer->deleteTrackName(name); 2315 } 2316 2317 // checkForNewParameters_l() must be called with ThreadBase::mLock held 2318 bool AudioFlinger::MixerThread::checkForNewParameters_l() 2319 { 2320 bool reconfig = false; 2321 2322 while (!mNewParameters.isEmpty()) { 2323 status_t status = NO_ERROR; 2324 String8 keyValuePair = mNewParameters[0]; 2325 AudioParameter param = AudioParameter(keyValuePair); 2326 int value; 2327 2328 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2329 reconfig = true; 2330 } 2331 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2332 if (value != AUDIO_FORMAT_PCM_16_BIT) { 2333 status = BAD_VALUE; 2334 } else { 2335 reconfig = true; 2336 } 2337 } 2338 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2339 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2340 status = BAD_VALUE; 2341 } else { 2342 reconfig = true; 2343 } 2344 } 2345 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2346 // do not accept frame count changes if tracks are open as the track buffer 2347 // size depends on frame count and correct behavior would not be garantied 2348 // if frame count is changed after track creation 2349 if (!mTracks.isEmpty()) { 2350 status = INVALID_OPERATION; 2351 } else { 2352 reconfig = true; 2353 } 2354 } 2355 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2356 // when changing the audio output device, call addBatteryData to notify 2357 // the change 2358 if ((int)mDevice != value) { 2359 uint32_t params = 0; 2360 // check whether speaker is on 2361 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2362 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2363 } 2364 2365 int deviceWithoutSpeaker 2366 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2367 // check if any other device (except speaker) is on 2368 if (value & deviceWithoutSpeaker ) { 2369 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2370 } 2371 2372 if (params != 0) { 2373 addBatteryData(params); 2374 } 2375 } 2376 2377 // forward device change to effects that have requested to be 2378 // aware of attached audio device. 2379 mDevice = (uint32_t)value; 2380 for (size_t i = 0; i < mEffectChains.size(); i++) { 2381 mEffectChains[i]->setDevice_l(mDevice); 2382 } 2383 } 2384 2385 if (status == NO_ERROR) { 2386 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2387 keyValuePair.string()); 2388 if (!mStandby && status == INVALID_OPERATION) { 2389 mOutput->stream->common.standby(&mOutput->stream->common); 2390 mStandby = true; 2391 mBytesWritten = 0; 2392 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2393 keyValuePair.string()); 2394 } 2395 if (status == NO_ERROR && reconfig) { 2396 delete mAudioMixer; 2397 readOutputParameters(); 2398 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2399 for (size_t i = 0; i < mTracks.size() ; i++) { 2400 int name = getTrackName_l(); 2401 if (name < 0) break; 2402 mTracks[i]->mName = name; 2403 // limit track sample rate to 2 x new output sample rate 2404 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2405 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2406 } 2407 } 2408 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2409 } 2410 } 2411 2412 mNewParameters.removeAt(0); 2413 2414 mParamStatus = status; 2415 mParamCond.signal(); 2416 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2417 // already timed out waiting for the status and will never signal the condition. 2418 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 2419 } 2420 return reconfig; 2421 } 2422 2423 status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2424 { 2425 const size_t SIZE = 256; 2426 char buffer[SIZE]; 2427 String8 result; 2428 2429 PlaybackThread::dumpInternals(fd, args); 2430 2431 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2432 result.append(buffer); 2433 write(fd, result.string(), result.size()); 2434 return NO_ERROR; 2435 } 2436 2437 uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2438 { 2439 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2440 } 2441 2442 uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2443 { 2444 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2445 } 2446 2447 // ---------------------------------------------------------------------------- 2448 AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2449 : PlaybackThread(audioFlinger, output, id, device) 2450 { 2451 mType = ThreadBase::DIRECT; 2452 } 2453 2454 AudioFlinger::DirectOutputThread::~DirectOutputThread() 2455 { 2456 } 2457 2458 2459 static inline int16_t clamp16(int32_t sample) 2460 { 2461 if ((sample>>15) ^ (sample>>31)) 2462 sample = 0x7FFF ^ (sample>>31); 2463 return sample; 2464 } 2465 2466 static inline 2467 int32_t mul(int16_t in, int16_t v) 2468 { 2469 #if defined(__arm__) && !defined(__thumb__) 2470 int32_t out; 2471 asm( "smulbb %[out], %[in], %[v] \n" 2472 : [out]"=r"(out) 2473 : [in]"%r"(in), [v]"r"(v) 2474 : ); 2475 return out; 2476 #else 2477 return in * int32_t(v); 2478 #endif 2479 } 2480 2481 void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2482 { 2483 // Do not apply volume on compressed audio 2484 if (!audio_is_linear_pcm(mFormat)) { 2485 return; 2486 } 2487 2488 // convert to signed 16 bit before volume calculation 2489 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2490 size_t count = mFrameCount * mChannelCount; 2491 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2492 int16_t *dst = mMixBuffer + count-1; 2493 while(count--) { 2494 *dst-- = (int16_t)(*src--^0x80) << 8; 2495 } 2496 } 2497 2498 size_t frameCount = mFrameCount; 2499 int16_t *out = mMixBuffer; 2500 if (ramp) { 2501 if (mChannelCount == 1) { 2502 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2503 int32_t vlInc = d / (int32_t)frameCount; 2504 int32_t vl = ((int32_t)mLeftVolShort << 16); 2505 do { 2506 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2507 out++; 2508 vl += vlInc; 2509 } while (--frameCount); 2510 2511 } else { 2512 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2513 int32_t vlInc = d / (int32_t)frameCount; 2514 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2515 int32_t vrInc = d / (int32_t)frameCount; 2516 int32_t vl = ((int32_t)mLeftVolShort << 16); 2517 int32_t vr = ((int32_t)mRightVolShort << 16); 2518 do { 2519 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2520 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2521 out += 2; 2522 vl += vlInc; 2523 vr += vrInc; 2524 } while (--frameCount); 2525 } 2526 } else { 2527 if (mChannelCount == 1) { 2528 do { 2529 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2530 out++; 2531 } while (--frameCount); 2532 } else { 2533 do { 2534 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2535 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2536 out += 2; 2537 } while (--frameCount); 2538 } 2539 } 2540 2541 // convert back to unsigned 8 bit after volume calculation 2542 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2543 size_t count = mFrameCount * mChannelCount; 2544 int16_t *src = mMixBuffer; 2545 uint8_t *dst = (uint8_t *)mMixBuffer; 2546 while(count--) { 2547 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2548 } 2549 } 2550 2551 mLeftVolShort = leftVol; 2552 mRightVolShort = rightVol; 2553 } 2554 2555 bool AudioFlinger::DirectOutputThread::threadLoop() 2556 { 2557 uint32_t mixerStatus = MIXER_IDLE; 2558 sp<Track> trackToRemove; 2559 sp<Track> activeTrack; 2560 nsecs_t standbyTime = systemTime(); 2561 int8_t *curBuf; 2562 size_t mixBufferSize = mFrameCount*mFrameSize; 2563 uint32_t activeSleepTime = activeSleepTimeUs(); 2564 uint32_t idleSleepTime = idleSleepTimeUs(); 2565 uint32_t sleepTime = idleSleepTime; 2566 // use shorter standby delay as on normal output to release 2567 // hardware resources as soon as possible 2568 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2569 2570 acquireWakeLock(); 2571 2572 while (!exitPending()) 2573 { 2574 bool rampVolume; 2575 uint16_t leftVol; 2576 uint16_t rightVol; 2577 Vector< sp<EffectChain> > effectChains; 2578 2579 processConfigEvents(); 2580 2581 mixerStatus = MIXER_IDLE; 2582 2583 { // scope for the mLock 2584 2585 Mutex::Autolock _l(mLock); 2586 2587 if (checkForNewParameters_l()) { 2588 mixBufferSize = mFrameCount*mFrameSize; 2589 activeSleepTime = activeSleepTimeUs(); 2590 idleSleepTime = idleSleepTimeUs(); 2591 standbyDelay = microseconds(activeSleepTime*2); 2592 } 2593 2594 // put audio hardware into standby after short delay 2595 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2596 mSuspended) { 2597 // wait until we have something to do... 2598 if (!mStandby) { 2599 LOGV("Audio hardware entering standby, mixer %p\n", this); 2600 mOutput->stream->common.standby(&mOutput->stream->common); 2601 mStandby = true; 2602 mBytesWritten = 0; 2603 } 2604 2605 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2606 // we're about to wait, flush the binder command buffer 2607 IPCThreadState::self()->flushCommands(); 2608 2609 if (exitPending()) break; 2610 2611 releaseWakeLock_l(); 2612 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2613 mWaitWorkCV.wait(mLock); 2614 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2615 acquireWakeLock_l(); 2616 2617 if (mMasterMute == false) { 2618 char value[PROPERTY_VALUE_MAX]; 2619 property_get("ro.audio.silent", value, "0"); 2620 if (atoi(value)) { 2621 LOGD("Silence is golden"); 2622 setMasterMute(true); 2623 } 2624 } 2625 2626 standbyTime = systemTime() + standbyDelay; 2627 sleepTime = idleSleepTime; 2628 continue; 2629 } 2630 } 2631 2632 effectChains = mEffectChains; 2633 2634 // find out which tracks need to be processed 2635 if (mActiveTracks.size() != 0) { 2636 sp<Track> t = mActiveTracks[0].promote(); 2637 if (t == 0) continue; 2638 2639 Track* const track = t.get(); 2640 audio_track_cblk_t* cblk = track->cblk(); 2641 2642 // The first time a track is added we wait 2643 // for all its buffers to be filled before processing it 2644 if (cblk->framesReady() && track->isReady() && 2645 !track->isPaused() && !track->isTerminated()) 2646 { 2647 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2648 2649 if (track->mFillingUpStatus == Track::FS_FILLED) { 2650 track->mFillingUpStatus = Track::FS_ACTIVE; 2651 mLeftVolFloat = mRightVolFloat = 0; 2652 mLeftVolShort = mRightVolShort = 0; 2653 if (track->mState == TrackBase::RESUMING) { 2654 track->mState = TrackBase::ACTIVE; 2655 rampVolume = true; 2656 } 2657 } else if (cblk->server != 0) { 2658 // If the track is stopped before the first frame was mixed, 2659 // do not apply ramp 2660 rampVolume = true; 2661 } 2662 // compute volume for this track 2663 float left, right; 2664 if (track->isMuted() || mMasterMute || track->isPausing() || 2665 mStreamTypes[track->type()].mute) { 2666 left = right = 0; 2667 if (track->isPausing()) { 2668 track->setPaused(); 2669 } 2670 } else { 2671 float typeVolume = mStreamTypes[track->type()].volume; 2672 float v = mMasterVolume * typeVolume; 2673 float v_clamped = v * cblk->volume[0]; 2674 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2675 left = v_clamped/MAX_GAIN; 2676 v_clamped = v * cblk->volume[1]; 2677 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2678 right = v_clamped/MAX_GAIN; 2679 } 2680 2681 if (left != mLeftVolFloat || right != mRightVolFloat) { 2682 mLeftVolFloat = left; 2683 mRightVolFloat = right; 2684 2685 // If audio HAL implements volume control, 2686 // force software volume to nominal value 2687 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2688 left = 1.0f; 2689 right = 1.0f; 2690 } 2691 2692 // Convert volumes from float to 8.24 2693 uint32_t vl = (uint32_t)(left * (1 << 24)); 2694 uint32_t vr = (uint32_t)(right * (1 << 24)); 2695 2696 // Delegate volume control to effect in track effect chain if needed 2697 // only one effect chain can be present on DirectOutputThread, so if 2698 // there is one, the track is connected to it 2699 if (!effectChains.isEmpty()) { 2700 // Do not ramp volume if volume is controlled by effect 2701 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2702 rampVolume = false; 2703 } 2704 } 2705 2706 // Convert volumes from 8.24 to 4.12 format 2707 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2708 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2709 leftVol = (uint16_t)v_clamped; 2710 v_clamped = (vr + (1 << 11)) >> 12; 2711 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2712 rightVol = (uint16_t)v_clamped; 2713 } else { 2714 leftVol = mLeftVolShort; 2715 rightVol = mRightVolShort; 2716 rampVolume = false; 2717 } 2718 2719 // reset retry count 2720 track->mRetryCount = kMaxTrackRetriesDirect; 2721 activeTrack = t; 2722 mixerStatus = MIXER_TRACKS_READY; 2723 } else { 2724 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2725 if (track->isStopped()) { 2726 track->reset(); 2727 } 2728 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2729 // We have consumed all the buffers of this track. 2730 // Remove it from the list of active tracks. 2731 trackToRemove = track; 2732 } else { 2733 // No buffers for this track. Give it a few chances to 2734 // fill a buffer, then remove it from active list. 2735 if (--(track->mRetryCount) <= 0) { 2736 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2737 trackToRemove = track; 2738 } else { 2739 mixerStatus = MIXER_TRACKS_ENABLED; 2740 } 2741 } 2742 } 2743 } 2744 2745 // remove all the tracks that need to be... 2746 if (UNLIKELY(trackToRemove != 0)) { 2747 mActiveTracks.remove(trackToRemove); 2748 if (!effectChains.isEmpty()) { 2749 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2750 trackToRemove->sessionId()); 2751 effectChains[0]->decActiveTrackCnt(); 2752 } 2753 if (trackToRemove->isTerminated()) { 2754 removeTrack_l(trackToRemove); 2755 } 2756 } 2757 2758 lockEffectChains_l(effectChains); 2759 } 2760 2761 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2762 AudioBufferProvider::Buffer buffer; 2763 size_t frameCount = mFrameCount; 2764 curBuf = (int8_t *)mMixBuffer; 2765 // output audio to hardware 2766 while (frameCount) { 2767 buffer.frameCount = frameCount; 2768 activeTrack->getNextBuffer(&buffer); 2769 if (UNLIKELY(buffer.raw == 0)) { 2770 memset(curBuf, 0, frameCount * mFrameSize); 2771 break; 2772 } 2773 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2774 frameCount -= buffer.frameCount; 2775 curBuf += buffer.frameCount * mFrameSize; 2776 activeTrack->releaseBuffer(&buffer); 2777 } 2778 sleepTime = 0; 2779 standbyTime = systemTime() + standbyDelay; 2780 } else { 2781 if (sleepTime == 0) { 2782 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2783 sleepTime = activeSleepTime; 2784 } else { 2785 sleepTime = idleSleepTime; 2786 } 2787 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2788 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2789 sleepTime = 0; 2790 } 2791 } 2792 2793 if (mSuspended) { 2794 sleepTime = suspendSleepTimeUs(); 2795 } 2796 // sleepTime == 0 means we must write to audio hardware 2797 if (sleepTime == 0) { 2798 if (mixerStatus == MIXER_TRACKS_READY) { 2799 applyVolume(leftVol, rightVol, rampVolume); 2800 } 2801 for (size_t i = 0; i < effectChains.size(); i ++) { 2802 effectChains[i]->process_l(); 2803 } 2804 unlockEffectChains(effectChains); 2805 2806 mLastWriteTime = systemTime(); 2807 mInWrite = true; 2808 mBytesWritten += mixBufferSize; 2809 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2810 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2811 mNumWrites++; 2812 mInWrite = false; 2813 mStandby = false; 2814 } else { 2815 unlockEffectChains(effectChains); 2816 usleep(sleepTime); 2817 } 2818 2819 // finally let go of removed track, without the lock held 2820 // since we can't guarantee the destructors won't acquire that 2821 // same lock. 2822 trackToRemove.clear(); 2823 activeTrack.clear(); 2824 2825 // Effect chains will be actually deleted here if they were removed from 2826 // mEffectChains list during mixing or effects processing 2827 effectChains.clear(); 2828 } 2829 2830 if (!mStandby) { 2831 mOutput->stream->common.standby(&mOutput->stream->common); 2832 } 2833 2834 releaseWakeLock(); 2835 2836 LOGV("DirectOutputThread %p exiting", this); 2837 return false; 2838 } 2839 2840 // getTrackName_l() must be called with ThreadBase::mLock held 2841 int AudioFlinger::DirectOutputThread::getTrackName_l() 2842 { 2843 return 0; 2844 } 2845 2846 // deleteTrackName_l() must be called with ThreadBase::mLock held 2847 void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2848 { 2849 } 2850 2851 // checkForNewParameters_l() must be called with ThreadBase::mLock held 2852 bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2853 { 2854 bool reconfig = false; 2855 2856 while (!mNewParameters.isEmpty()) { 2857 status_t status = NO_ERROR; 2858 String8 keyValuePair = mNewParameters[0]; 2859 AudioParameter param = AudioParameter(keyValuePair); 2860 int value; 2861 2862 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2863 // do not accept frame count changes if tracks are open as the track buffer 2864 // size depends on frame count and correct behavior would not be garantied 2865 // if frame count is changed after track creation 2866 if (!mTracks.isEmpty()) { 2867 status = INVALID_OPERATION; 2868 } else { 2869 reconfig = true; 2870 } 2871 } 2872 if (status == NO_ERROR) { 2873 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2874 keyValuePair.string()); 2875 if (!mStandby && status == INVALID_OPERATION) { 2876 mOutput->stream->common.standby(&mOutput->stream->common); 2877 mStandby = true; 2878 mBytesWritten = 0; 2879 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2880 keyValuePair.string()); 2881 } 2882 if (status == NO_ERROR && reconfig) { 2883 readOutputParameters(); 2884 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2885 } 2886 } 2887 2888 mNewParameters.removeAt(0); 2889 2890 mParamStatus = status; 2891 mParamCond.signal(); 2892 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2893 // already timed out waiting for the status and will never signal the condition. 2894 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 2895 } 2896 return reconfig; 2897 } 2898 2899 uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2900 { 2901 uint32_t time; 2902 if (audio_is_linear_pcm(mFormat)) { 2903 time = PlaybackThread::activeSleepTimeUs(); 2904 } else { 2905 time = 10000; 2906 } 2907 return time; 2908 } 2909 2910 uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2911 { 2912 uint32_t time; 2913 if (audio_is_linear_pcm(mFormat)) { 2914 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2915 } else { 2916 time = 10000; 2917 } 2918 return time; 2919 } 2920 2921 uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2922 { 2923 uint32_t time; 2924 if (audio_is_linear_pcm(mFormat)) { 2925 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2926 } else { 2927 time = 10000; 2928 } 2929 return time; 2930 } 2931 2932 2933 // ---------------------------------------------------------------------------- 2934 2935 AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2936 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2937 { 2938 mType = ThreadBase::DUPLICATING; 2939 addOutputTrack(mainThread); 2940 } 2941 2942 AudioFlinger::DuplicatingThread::~DuplicatingThread() 2943 { 2944 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2945 mOutputTracks[i]->destroy(); 2946 } 2947 mOutputTracks.clear(); 2948 } 2949 2950 bool AudioFlinger::DuplicatingThread::threadLoop() 2951 { 2952 Vector< sp<Track> > tracksToRemove; 2953 uint32_t mixerStatus = MIXER_IDLE; 2954 nsecs_t standbyTime = systemTime(); 2955 size_t mixBufferSize = mFrameCount*mFrameSize; 2956 SortedVector< sp<OutputTrack> > outputTracks; 2957 uint32_t writeFrames = 0; 2958 uint32_t activeSleepTime = activeSleepTimeUs(); 2959 uint32_t idleSleepTime = idleSleepTimeUs(); 2960 uint32_t sleepTime = idleSleepTime; 2961 Vector< sp<EffectChain> > effectChains; 2962 2963 acquireWakeLock(); 2964 2965 while (!exitPending()) 2966 { 2967 processConfigEvents(); 2968 2969 mixerStatus = MIXER_IDLE; 2970 { // scope for the mLock 2971 2972 Mutex::Autolock _l(mLock); 2973 2974 if (checkForNewParameters_l()) { 2975 mixBufferSize = mFrameCount*mFrameSize; 2976 updateWaitTime(); 2977 activeSleepTime = activeSleepTimeUs(); 2978 idleSleepTime = idleSleepTimeUs(); 2979 } 2980 2981 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2982 2983 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2984 outputTracks.add(mOutputTracks[i]); 2985 } 2986 2987 // put audio hardware into standby after short delay 2988 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2989 mSuspended) { 2990 if (!mStandby) { 2991 for (size_t i = 0; i < outputTracks.size(); i++) { 2992 outputTracks[i]->stop(); 2993 } 2994 mStandby = true; 2995 mBytesWritten = 0; 2996 } 2997 2998 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2999 // we're about to wait, flush the binder command buffer 3000 IPCThreadState::self()->flushCommands(); 3001 outputTracks.clear(); 3002 3003 if (exitPending()) break; 3004 3005 releaseWakeLock_l(); 3006 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3007 mWaitWorkCV.wait(mLock); 3008 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3009 acquireWakeLock_l(); 3010 3011 if (mMasterMute == false) { 3012 char value[PROPERTY_VALUE_MAX]; 3013 property_get("ro.audio.silent", value, "0"); 3014 if (atoi(value)) { 3015 LOGD("Silence is golden"); 3016 setMasterMute(true); 3017 } 3018 } 3019 3020 standbyTime = systemTime() + kStandbyTimeInNsecs; 3021 sleepTime = idleSleepTime; 3022 continue; 3023 } 3024 } 3025 3026 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3027 3028 // prevent any changes in effect chain list and in each effect chain 3029 // during mixing and effect process as the audio buffers could be deleted 3030 // or modified if an effect is created or deleted 3031 lockEffectChains_l(effectChains); 3032 } 3033 3034 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3035 // mix buffers... 3036 if (outputsReady(outputTracks)) { 3037 mAudioMixer->process(); 3038 } else { 3039 memset(mMixBuffer, 0, mixBufferSize); 3040 } 3041 sleepTime = 0; 3042 writeFrames = mFrameCount; 3043 } else { 3044 if (sleepTime == 0) { 3045 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3046 sleepTime = activeSleepTime; 3047 } else { 3048 sleepTime = idleSleepTime; 3049 } 3050 } else if (mBytesWritten != 0) { 3051 // flush remaining overflow buffers in output tracks 3052 for (size_t i = 0; i < outputTracks.size(); i++) { 3053 if (outputTracks[i]->isActive()) { 3054 sleepTime = 0; 3055 writeFrames = 0; 3056 memset(mMixBuffer, 0, mixBufferSize); 3057 break; 3058 } 3059 } 3060 } 3061 } 3062 3063 if (mSuspended) { 3064 sleepTime = suspendSleepTimeUs(); 3065 } 3066 // sleepTime == 0 means we must write to audio hardware 3067 if (sleepTime == 0) { 3068 for (size_t i = 0; i < effectChains.size(); i ++) { 3069 effectChains[i]->process_l(); 3070 } 3071 // enable changes in effect chain 3072 unlockEffectChains(effectChains); 3073 3074 standbyTime = systemTime() + kStandbyTimeInNsecs; 3075 for (size_t i = 0; i < outputTracks.size(); i++) { 3076 outputTracks[i]->write(mMixBuffer, writeFrames); 3077 } 3078 mStandby = false; 3079 mBytesWritten += mixBufferSize; 3080 } else { 3081 // enable changes in effect chain 3082 unlockEffectChains(effectChains); 3083 usleep(sleepTime); 3084 } 3085 3086 // finally let go of all our tracks, without the lock held 3087 // since we can't guarantee the destructors won't acquire that 3088 // same lock. 3089 tracksToRemove.clear(); 3090 outputTracks.clear(); 3091 3092 // Effect chains will be actually deleted here if they were removed from 3093 // mEffectChains list during mixing or effects processing 3094 effectChains.clear(); 3095 } 3096 3097 releaseWakeLock(); 3098 3099 return false; 3100 } 3101 3102 void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3103 { 3104 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3105 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3106 this, 3107 mSampleRate, 3108 mFormat, 3109 mChannelMask, 3110 frameCount); 3111 if (outputTrack->cblk() != NULL) { 3112 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3113 mOutputTracks.add(outputTrack); 3114 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3115 updateWaitTime(); 3116 } 3117 } 3118 3119 void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3120 { 3121 Mutex::Autolock _l(mLock); 3122 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3123 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3124 mOutputTracks[i]->destroy(); 3125 mOutputTracks.removeAt(i); 3126 updateWaitTime(); 3127 return; 3128 } 3129 } 3130 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3131 } 3132 3133 void AudioFlinger::DuplicatingThread::updateWaitTime() 3134 { 3135 mWaitTimeMs = UINT_MAX; 3136 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3137 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3138 if (strong != NULL) { 3139 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3140 if (waitTimeMs < mWaitTimeMs) { 3141 mWaitTimeMs = waitTimeMs; 3142 } 3143 } 3144 } 3145 } 3146 3147 3148 bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3149 { 3150 for (size_t i = 0; i < outputTracks.size(); i++) { 3151 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3152 if (thread == 0) { 3153 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3154 return false; 3155 } 3156 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3157 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3158 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3159 return false; 3160 } 3161 } 3162 return true; 3163 } 3164 3165 uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3166 { 3167 return (mWaitTimeMs * 1000) / 2; 3168 } 3169 3170 // ---------------------------------------------------------------------------- 3171 3172 // TrackBase constructor must be called with AudioFlinger::mLock held 3173 AudioFlinger::ThreadBase::TrackBase::TrackBase( 3174 const wp<ThreadBase>& thread, 3175 const sp<Client>& client, 3176 uint32_t sampleRate, 3177 uint32_t format, 3178 uint32_t channelMask, 3179 int frameCount, 3180 uint32_t flags, 3181 const sp<IMemory>& sharedBuffer, 3182 int sessionId) 3183 : RefBase(), 3184 mThread(thread), 3185 mClient(client), 3186 mCblk(0), 3187 mFrameCount(0), 3188 mState(IDLE), 3189 mClientTid(-1), 3190 mFormat(format), 3191 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3192 mSessionId(sessionId) 3193 { 3194 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3195 3196 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3197 size_t size = sizeof(audio_track_cblk_t); 3198 uint8_t channelCount = popcount(channelMask); 3199 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3200 if (sharedBuffer == 0) { 3201 size += bufferSize; 3202 } 3203 3204 if (client != NULL) { 3205 mCblkMemory = client->heap()->allocate(size); 3206 if (mCblkMemory != 0) { 3207 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3208 if (mCblk) { // construct the shared structure in-place. 3209 new(mCblk) audio_track_cblk_t(); 3210 // clear all buffers 3211 mCblk->frameCount = frameCount; 3212 mCblk->sampleRate = sampleRate; 3213 mChannelCount = channelCount; 3214 mChannelMask = channelMask; 3215 if (sharedBuffer == 0) { 3216 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3217 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3218 // Force underrun condition to avoid false underrun callback until first data is 3219 // written to buffer (other flags are cleared) 3220 mCblk->flags = CBLK_UNDERRUN_ON; 3221 } else { 3222 mBuffer = sharedBuffer->pointer(); 3223 } 3224 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3225 } 3226 } else { 3227 LOGE("not enough memory for AudioTrack size=%u", size); 3228 client->heap()->dump("AudioTrack"); 3229 return; 3230 } 3231 } else { 3232 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3233 if (mCblk) { // construct the shared structure in-place. 3234 new(mCblk) audio_track_cblk_t(); 3235 // clear all buffers 3236 mCblk->frameCount = frameCount; 3237 mCblk->sampleRate = sampleRate; 3238 mChannelCount = channelCount; 3239 mChannelMask = channelMask; 3240 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3241 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3242 // Force underrun condition to avoid false underrun callback until first data is 3243 // written to buffer (other flags are cleared) 3244 mCblk->flags = CBLK_UNDERRUN_ON; 3245 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3246 } 3247 } 3248 } 3249 3250 AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3251 { 3252 if (mCblk) { 3253 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3254 if (mClient == NULL) { 3255 delete mCblk; 3256 } 3257 } 3258 mCblkMemory.clear(); // and free the shared memory 3259 if (mClient != NULL) { 3260 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3261 mClient.clear(); 3262 } 3263 } 3264 3265 void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3266 { 3267 buffer->raw = 0; 3268 mFrameCount = buffer->frameCount; 3269 step(); 3270 buffer->frameCount = 0; 3271 } 3272 3273 bool AudioFlinger::ThreadBase::TrackBase::step() { 3274 bool result; 3275 audio_track_cblk_t* cblk = this->cblk(); 3276 3277 result = cblk->stepServer(mFrameCount); 3278 if (!result) { 3279 LOGV("stepServer failed acquiring cblk mutex"); 3280 mFlags |= STEPSERVER_FAILED; 3281 } 3282 return result; 3283 } 3284 3285 void AudioFlinger::ThreadBase::TrackBase::reset() { 3286 audio_track_cblk_t* cblk = this->cblk(); 3287 3288 cblk->user = 0; 3289 cblk->server = 0; 3290 cblk->userBase = 0; 3291 cblk->serverBase = 0; 3292 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3293 LOGV("TrackBase::reset"); 3294 } 3295 3296 sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3297 { 3298 return mCblkMemory; 3299 } 3300 3301 int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3302 return (int)mCblk->sampleRate; 3303 } 3304 3305 int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3306 return (const int)mChannelCount; 3307 } 3308 3309 uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3310 return mChannelMask; 3311 } 3312 3313 void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3314 audio_track_cblk_t* cblk = this->cblk(); 3315 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 3316 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 3317 3318 // Check validity of returned pointer in case the track control block would have been corrupted. 3319 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3320 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 3321 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3322 server %d, serverBase %d, user %d, userBase %d", 3323 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3324 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3325 return 0; 3326 } 3327 3328 return bufferStart; 3329 } 3330 3331 // ---------------------------------------------------------------------------- 3332 3333 // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3334 AudioFlinger::PlaybackThread::Track::Track( 3335 const wp<ThreadBase>& thread, 3336 const sp<Client>& client, 3337 int streamType, 3338 uint32_t sampleRate, 3339 uint32_t format, 3340 uint32_t channelMask, 3341 int frameCount, 3342 const sp<IMemory>& sharedBuffer, 3343 int sessionId) 3344 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3345 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3346 mAuxEffectId(0), mHasVolumeController(false) 3347 { 3348 if (mCblk != NULL) { 3349 sp<ThreadBase> baseThread = thread.promote(); 3350 if (baseThread != 0) { 3351 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3352 mName = playbackThread->getTrackName_l(); 3353 mMainBuffer = playbackThread->mixBuffer(); 3354 } 3355 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3356 if (mName < 0) { 3357 LOGE("no more track names available"); 3358 } 3359 mVolume[0] = 1.0f; 3360 mVolume[1] = 1.0f; 3361 mStreamType = streamType; 3362 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3363 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3364 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3365 } 3366 } 3367 3368 AudioFlinger::PlaybackThread::Track::~Track() 3369 { 3370 LOGV("PlaybackThread::Track destructor"); 3371 sp<ThreadBase> thread = mThread.promote(); 3372 if (thread != 0) { 3373 Mutex::Autolock _l(thread->mLock); 3374 mState = TERMINATED; 3375 } 3376 } 3377 3378 void AudioFlinger::PlaybackThread::Track::destroy() 3379 { 3380 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3381 // by removing it from mTracks vector, so there is a risk that this Tracks's 3382 // desctructor is called. As the destructor needs to lock mLock, 3383 // we must acquire a strong reference on this Track before locking mLock 3384 // here so that the destructor is called only when exiting this function. 3385 // On the other hand, as long as Track::destroy() is only called by 3386 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3387 // this Track with its member mTrack. 3388 sp<Track> keep(this); 3389 { // scope for mLock 3390 sp<ThreadBase> thread = mThread.promote(); 3391 if (thread != 0) { 3392 if (!isOutputTrack()) { 3393 if (mState == ACTIVE || mState == RESUMING) { 3394 AudioSystem::stopOutput(thread->id(), 3395 (audio_stream_type_t)mStreamType, 3396 mSessionId); 3397 3398 // to track the speaker usage 3399 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3400 } 3401 AudioSystem::releaseOutput(thread->id()); 3402 } 3403 Mutex::Autolock _l(thread->mLock); 3404 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3405 playbackThread->destroyTrack_l(this); 3406 } 3407 } 3408 } 3409 3410 void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3411 { 3412 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3413 mName - AudioMixer::TRACK0, 3414 (mClient == NULL) ? getpid() : mClient->pid(), 3415 mStreamType, 3416 mFormat, 3417 mChannelMask, 3418 mSessionId, 3419 mFrameCount, 3420 mState, 3421 mMute, 3422 mFillingUpStatus, 3423 mCblk->sampleRate, 3424 mCblk->volume[0], 3425 mCblk->volume[1], 3426 mCblk->server, 3427 mCblk->user, 3428 (int)mMainBuffer, 3429 (int)mAuxBuffer); 3430 } 3431 3432 status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3433 { 3434 audio_track_cblk_t* cblk = this->cblk(); 3435 uint32_t framesReady; 3436 uint32_t framesReq = buffer->frameCount; 3437 3438 // Check if last stepServer failed, try to step now 3439 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3440 if (!step()) goto getNextBuffer_exit; 3441 LOGV("stepServer recovered"); 3442 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3443 } 3444 3445 framesReady = cblk->framesReady(); 3446 3447 if (LIKELY(framesReady)) { 3448 uint32_t s = cblk->server; 3449 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3450 3451 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3452 if (framesReq > framesReady) { 3453 framesReq = framesReady; 3454 } 3455 if (s + framesReq > bufferEnd) { 3456 framesReq = bufferEnd - s; 3457 } 3458 3459 buffer->raw = getBuffer(s, framesReq); 3460 if (buffer->raw == 0) goto getNextBuffer_exit; 3461 3462 buffer->frameCount = framesReq; 3463 return NO_ERROR; 3464 } 3465 3466 getNextBuffer_exit: 3467 buffer->raw = 0; 3468 buffer->frameCount = 0; 3469 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3470 return NOT_ENOUGH_DATA; 3471 } 3472 3473 bool AudioFlinger::PlaybackThread::Track::isReady() const { 3474 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3475 3476 if (mCblk->framesReady() >= mCblk->frameCount || 3477 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3478 mFillingUpStatus = FS_FILLED; 3479 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3480 return true; 3481 } 3482 return false; 3483 } 3484 3485 status_t AudioFlinger::PlaybackThread::Track::start() 3486 { 3487 status_t status = NO_ERROR; 3488 LOGV("start(%d), calling thread %d session %d", 3489 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3490 sp<ThreadBase> thread = mThread.promote(); 3491 if (thread != 0) { 3492 Mutex::Autolock _l(thread->mLock); 3493 int state = mState; 3494 // here the track could be either new, or restarted 3495 // in both cases "unstop" the track 3496 if (mState == PAUSED) { 3497 mState = TrackBase::RESUMING; 3498 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3499 } else { 3500 mState = TrackBase::ACTIVE; 3501 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3502 } 3503 3504 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3505 thread->mLock.unlock(); 3506 status = AudioSystem::startOutput(thread->id(), 3507 (audio_stream_type_t)mStreamType, 3508 mSessionId); 3509 thread->mLock.lock(); 3510 3511 // to track the speaker usage 3512 if (status == NO_ERROR) { 3513 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3514 } 3515 } 3516 if (status == NO_ERROR) { 3517 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3518 playbackThread->addTrack_l(this); 3519 } else { 3520 mState = state; 3521 } 3522 } else { 3523 status = BAD_VALUE; 3524 } 3525 return status; 3526 } 3527 3528 void AudioFlinger::PlaybackThread::Track::stop() 3529 { 3530 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3531 sp<ThreadBase> thread = mThread.promote(); 3532 if (thread != 0) { 3533 Mutex::Autolock _l(thread->mLock); 3534 int state = mState; 3535 if (mState > STOPPED) { 3536 mState = STOPPED; 3537 // If the track is not active (PAUSED and buffers full), flush buffers 3538 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3539 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3540 reset(); 3541 } 3542 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3543 } 3544 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3545 thread->mLock.unlock(); 3546 AudioSystem::stopOutput(thread->id(), 3547 (audio_stream_type_t)mStreamType, 3548 mSessionId); 3549 thread->mLock.lock(); 3550 3551 // to track the speaker usage 3552 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3553 } 3554 } 3555 } 3556 3557 void AudioFlinger::PlaybackThread::Track::pause() 3558 { 3559 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3560 sp<ThreadBase> thread = mThread.promote(); 3561 if (thread != 0) { 3562 Mutex::Autolock _l(thread->mLock); 3563 if (mState == ACTIVE || mState == RESUMING) { 3564 mState = PAUSING; 3565 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3566 if (!isOutputTrack()) { 3567 thread->mLock.unlock(); 3568 AudioSystem::stopOutput(thread->id(), 3569 (audio_stream_type_t)mStreamType, 3570 mSessionId); 3571 thread->mLock.lock(); 3572 3573 // to track the speaker usage 3574 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3575 } 3576 } 3577 } 3578 } 3579 3580 void AudioFlinger::PlaybackThread::Track::flush() 3581 { 3582 LOGV("flush(%d)", mName); 3583 sp<ThreadBase> thread = mThread.promote(); 3584 if (thread != 0) { 3585 Mutex::Autolock _l(thread->mLock); 3586 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3587 return; 3588 } 3589 // No point remaining in PAUSED state after a flush => go to 3590 // STOPPED state 3591 mState = STOPPED; 3592 3593 // do not reset the track if it is still in the process of being stopped or paused. 3594 // this will be done by prepareTracks_l() when the track is stopped. 3595 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3596 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3597 reset(); 3598 } 3599 } 3600 } 3601 3602 void AudioFlinger::PlaybackThread::Track::reset() 3603 { 3604 // Do not reset twice to avoid discarding data written just after a flush and before 3605 // the audioflinger thread detects the track is stopped. 3606 if (!mResetDone) { 3607 TrackBase::reset(); 3608 // Force underrun condition to avoid false underrun callback until first data is 3609 // written to buffer 3610 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3611 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3612 mFillingUpStatus = FS_FILLING; 3613 mResetDone = true; 3614 } 3615 } 3616 3617 void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3618 { 3619 mMute = muted; 3620 } 3621 3622 void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3623 { 3624 mVolume[0] = left; 3625 mVolume[1] = right; 3626 } 3627 3628 status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3629 { 3630 status_t status = DEAD_OBJECT; 3631 sp<ThreadBase> thread = mThread.promote(); 3632 if (thread != 0) { 3633 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3634 status = playbackThread->attachAuxEffect(this, EffectId); 3635 } 3636 return status; 3637 } 3638 3639 void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3640 { 3641 mAuxEffectId = EffectId; 3642 mAuxBuffer = buffer; 3643 } 3644 3645 // ---------------------------------------------------------------------------- 3646 3647 // RecordTrack constructor must be called with AudioFlinger::mLock held 3648 AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3649 const wp<ThreadBase>& thread, 3650 const sp<Client>& client, 3651 uint32_t sampleRate, 3652 uint32_t format, 3653 uint32_t channelMask, 3654 int frameCount, 3655 uint32_t flags, 3656 int sessionId) 3657 : TrackBase(thread, client, sampleRate, format, 3658 channelMask, frameCount, flags, 0, sessionId), 3659 mOverflow(false) 3660 { 3661 if (mCblk != NULL) { 3662 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3663 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3664 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3665 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3666 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3667 } else { 3668 mCblk->frameSize = sizeof(int8_t); 3669 } 3670 } 3671 } 3672 3673 AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3674 { 3675 sp<ThreadBase> thread = mThread.promote(); 3676 if (thread != 0) { 3677 AudioSystem::releaseInput(thread->id()); 3678 } 3679 } 3680 3681 status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3682 { 3683 audio_track_cblk_t* cblk = this->cblk(); 3684 uint32_t framesAvail; 3685 uint32_t framesReq = buffer->frameCount; 3686 3687 // Check if last stepServer failed, try to step now 3688 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3689 if (!step()) goto getNextBuffer_exit; 3690 LOGV("stepServer recovered"); 3691 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3692 } 3693 3694 framesAvail = cblk->framesAvailable_l(); 3695 3696 if (LIKELY(framesAvail)) { 3697 uint32_t s = cblk->server; 3698 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3699 3700 if (framesReq > framesAvail) { 3701 framesReq = framesAvail; 3702 } 3703 if (s + framesReq > bufferEnd) { 3704 framesReq = bufferEnd - s; 3705 } 3706 3707 buffer->raw = getBuffer(s, framesReq); 3708 if (buffer->raw == 0) goto getNextBuffer_exit; 3709 3710 buffer->frameCount = framesReq; 3711 return NO_ERROR; 3712 } 3713 3714 getNextBuffer_exit: 3715 buffer->raw = 0; 3716 buffer->frameCount = 0; 3717 return NOT_ENOUGH_DATA; 3718 } 3719 3720 status_t AudioFlinger::RecordThread::RecordTrack::start() 3721 { 3722 sp<ThreadBase> thread = mThread.promote(); 3723 if (thread != 0) { 3724 RecordThread *recordThread = (RecordThread *)thread.get(); 3725 return recordThread->start(this); 3726 } else { 3727 return BAD_VALUE; 3728 } 3729 } 3730 3731 void AudioFlinger::RecordThread::RecordTrack::stop() 3732 { 3733 sp<ThreadBase> thread = mThread.promote(); 3734 if (thread != 0) { 3735 RecordThread *recordThread = (RecordThread *)thread.get(); 3736 recordThread->stop(this); 3737 TrackBase::reset(); 3738 // Force overerrun condition to avoid false overrun callback until first data is 3739 // read from buffer 3740 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3741 } 3742 } 3743 3744 void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3745 { 3746 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3747 (mClient == NULL) ? getpid() : mClient->pid(), 3748 mFormat, 3749 mChannelMask, 3750 mSessionId, 3751 mFrameCount, 3752 mState, 3753 mCblk->sampleRate, 3754 mCblk->server, 3755 mCblk->user); 3756 } 3757 3758 3759 // ---------------------------------------------------------------------------- 3760 3761 AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3762 const wp<ThreadBase>& thread, 3763 DuplicatingThread *sourceThread, 3764 uint32_t sampleRate, 3765 uint32_t format, 3766 uint32_t channelMask, 3767 int frameCount) 3768 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3769 mActive(false), mSourceThread(sourceThread) 3770 { 3771 3772 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3773 if (mCblk != NULL) { 3774 mCblk->flags |= CBLK_DIRECTION_OUT; 3775 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3776 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3777 mOutBuffer.frameCount = 0; 3778 playbackThread->mTracks.add(this); 3779 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3780 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3781 mCblk, mBuffer, mCblk->buffers, 3782 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3783 } else { 3784 LOGW("Error creating output track on thread %p", playbackThread); 3785 } 3786 } 3787 3788 AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3789 { 3790 clearBufferQueue(); 3791 } 3792 3793 status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3794 { 3795 status_t status = Track::start(); 3796 if (status != NO_ERROR) { 3797 return status; 3798 } 3799 3800 mActive = true; 3801 mRetryCount = 127; 3802 return status; 3803 } 3804 3805 void AudioFlinger::PlaybackThread::OutputTrack::stop() 3806 { 3807 Track::stop(); 3808 clearBufferQueue(); 3809 mOutBuffer.frameCount = 0; 3810 mActive = false; 3811 } 3812 3813 bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3814 { 3815 Buffer *pInBuffer; 3816 Buffer inBuffer; 3817 uint32_t channelCount = mChannelCount; 3818 bool outputBufferFull = false; 3819 inBuffer.frameCount = frames; 3820 inBuffer.i16 = data; 3821 3822 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3823 3824 if (!mActive && frames != 0) { 3825 start(); 3826 sp<ThreadBase> thread = mThread.promote(); 3827 if (thread != 0) { 3828 MixerThread *mixerThread = (MixerThread *)thread.get(); 3829 if (mCblk->frameCount > frames){ 3830 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3831 uint32_t startFrames = (mCblk->frameCount - frames); 3832 pInBuffer = new Buffer; 3833 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3834 pInBuffer->frameCount = startFrames; 3835 pInBuffer->i16 = pInBuffer->mBuffer; 3836 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3837 mBufferQueue.add(pInBuffer); 3838 } else { 3839 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3840 } 3841 } 3842 } 3843 } 3844 3845 while (waitTimeLeftMs) { 3846 // First write pending buffers, then new data 3847 if (mBufferQueue.size()) { 3848 pInBuffer = mBufferQueue.itemAt(0); 3849 } else { 3850 pInBuffer = &inBuffer; 3851 } 3852 3853 if (pInBuffer->frameCount == 0) { 3854 break; 3855 } 3856 3857 if (mOutBuffer.frameCount == 0) { 3858 mOutBuffer.frameCount = pInBuffer->frameCount; 3859 nsecs_t startTime = systemTime(); 3860 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3861 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3862 outputBufferFull = true; 3863 break; 3864 } 3865 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3866 if (waitTimeLeftMs >= waitTimeMs) { 3867 waitTimeLeftMs -= waitTimeMs; 3868 } else { 3869 waitTimeLeftMs = 0; 3870 } 3871 } 3872 3873 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3874 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3875 mCblk->stepUser(outFrames); 3876 pInBuffer->frameCount -= outFrames; 3877 pInBuffer->i16 += outFrames * channelCount; 3878 mOutBuffer.frameCount -= outFrames; 3879 mOutBuffer.i16 += outFrames * channelCount; 3880 3881 if (pInBuffer->frameCount == 0) { 3882 if (mBufferQueue.size()) { 3883 mBufferQueue.removeAt(0); 3884 delete [] pInBuffer->mBuffer; 3885 delete pInBuffer; 3886 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3887 } else { 3888 break; 3889 } 3890 } 3891 } 3892 3893 // If we could not write all frames, allocate a buffer and queue it for next time. 3894 if (inBuffer.frameCount) { 3895 sp<ThreadBase> thread = mThread.promote(); 3896 if (thread != 0 && !thread->standby()) { 3897 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3898 pInBuffer = new Buffer; 3899 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3900 pInBuffer->frameCount = inBuffer.frameCount; 3901 pInBuffer->i16 = pInBuffer->mBuffer; 3902 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3903 mBufferQueue.add(pInBuffer); 3904 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3905 } else { 3906 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3907 } 3908 } 3909 } 3910 3911 // Calling write() with a 0 length buffer, means that no more data will be written: 3912 // If no more buffers are pending, fill output track buffer to make sure it is started 3913 // by output mixer. 3914 if (frames == 0 && mBufferQueue.size() == 0) { 3915 if (mCblk->user < mCblk->frameCount) { 3916 frames = mCblk->frameCount - mCblk->user; 3917 pInBuffer = new Buffer; 3918 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3919 pInBuffer->frameCount = frames; 3920 pInBuffer->i16 = pInBuffer->mBuffer; 3921 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3922 mBufferQueue.add(pInBuffer); 3923 } else if (mActive) { 3924 stop(); 3925 } 3926 } 3927 3928 return outputBufferFull; 3929 } 3930 3931 status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3932 { 3933 int active; 3934 status_t result; 3935 audio_track_cblk_t* cblk = mCblk; 3936 uint32_t framesReq = buffer->frameCount; 3937 3938 // LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3939 buffer->frameCount = 0; 3940 3941 uint32_t framesAvail = cblk->framesAvailable(); 3942 3943 3944 if (framesAvail == 0) { 3945 Mutex::Autolock _l(cblk->lock); 3946 goto start_loop_here; 3947 while (framesAvail == 0) { 3948 active = mActive; 3949 if (UNLIKELY(!active)) { 3950 LOGV("Not active and NO_MORE_BUFFERS"); 3951 return AudioTrack::NO_MORE_BUFFERS; 3952 } 3953 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3954 if (result != NO_ERROR) { 3955 return AudioTrack::NO_MORE_BUFFERS; 3956 } 3957 // read the server count again 3958 start_loop_here: 3959 framesAvail = cblk->framesAvailable_l(); 3960 } 3961 } 3962 3963 // if (framesAvail < framesReq) { 3964 // return AudioTrack::NO_MORE_BUFFERS; 3965 // } 3966 3967 if (framesReq > framesAvail) { 3968 framesReq = framesAvail; 3969 } 3970 3971 uint32_t u = cblk->user; 3972 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3973 3974 if (u + framesReq > bufferEnd) { 3975 framesReq = bufferEnd - u; 3976 } 3977 3978 buffer->frameCount = framesReq; 3979 buffer->raw = (void *)cblk->buffer(u); 3980 return NO_ERROR; 3981 } 3982 3983 3984 void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3985 { 3986 size_t size = mBufferQueue.size(); 3987 Buffer *pBuffer; 3988 3989 for (size_t i = 0; i < size; i++) { 3990 pBuffer = mBufferQueue.itemAt(i); 3991 delete [] pBuffer->mBuffer; 3992 delete pBuffer; 3993 } 3994 mBufferQueue.clear(); 3995 } 3996 3997 // ---------------------------------------------------------------------------- 3998 3999 AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4000 : RefBase(), 4001 mAudioFlinger(audioFlinger), 4002 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4003 mPid(pid) 4004 { 4005 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4006 } 4007 4008 // Client destructor must be called with AudioFlinger::mLock held 4009 AudioFlinger::Client::~Client() 4010 { 4011 mAudioFlinger->removeClient_l(mPid); 4012 } 4013 4014 const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4015 { 4016 return mMemoryDealer; 4017 } 4018 4019 // ---------------------------------------------------------------------------- 4020 4021 AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4022 const sp<IAudioFlingerClient>& client, 4023 pid_t pid) 4024 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4025 { 4026 } 4027 4028 AudioFlinger::NotificationClient::~NotificationClient() 4029 { 4030 mClient.clear(); 4031 } 4032 4033 void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4034 { 4035 sp<NotificationClient> keep(this); 4036 { 4037 mAudioFlinger->removeNotificationClient(mPid); 4038 } 4039 } 4040 4041 // ---------------------------------------------------------------------------- 4042 4043 AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4044 : BnAudioTrack(), 4045 mTrack(track) 4046 { 4047 } 4048 4049 AudioFlinger::TrackHandle::~TrackHandle() { 4050 // just stop the track on deletion, associated resources 4051 // will be freed from the main thread once all pending buffers have 4052 // been played. Unless it's not in the active track list, in which 4053 // case we free everything now... 4054 mTrack->destroy(); 4055 } 4056 4057 status_t AudioFlinger::TrackHandle::start() { 4058 return mTrack->start(); 4059 } 4060 4061 void AudioFlinger::TrackHandle::stop() { 4062 mTrack->stop(); 4063 } 4064 4065 void AudioFlinger::TrackHandle::flush() { 4066 mTrack->flush(); 4067 } 4068 4069 void AudioFlinger::TrackHandle::mute(bool e) { 4070 mTrack->mute(e); 4071 } 4072 4073 void AudioFlinger::TrackHandle::pause() { 4074 mTrack->pause(); 4075 } 4076 4077 void AudioFlinger::TrackHandle::setVolume(float left, float right) { 4078 mTrack->setVolume(left, right); 4079 } 4080 4081 sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4082 return mTrack->getCblk(); 4083 } 4084 4085 status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4086 { 4087 return mTrack->attachAuxEffect(EffectId); 4088 } 4089 4090 status_t AudioFlinger::TrackHandle::onTransact( 4091 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4092 { 4093 return BnAudioTrack::onTransact(code, data, reply, flags); 4094 } 4095 4096 // ---------------------------------------------------------------------------- 4097 4098 sp<IAudioRecord> AudioFlinger::openRecord( 4099 pid_t pid, 4100 int input, 4101 uint32_t sampleRate, 4102 uint32_t format, 4103 uint32_t channelMask, 4104 int frameCount, 4105 uint32_t flags, 4106 int *sessionId, 4107 status_t *status) 4108 { 4109 sp<RecordThread::RecordTrack> recordTrack; 4110 sp<RecordHandle> recordHandle; 4111 sp<Client> client; 4112 wp<Client> wclient; 4113 status_t lStatus; 4114 RecordThread *thread; 4115 size_t inFrameCount; 4116 int lSessionId; 4117 4118 // check calling permissions 4119 if (!recordingAllowed()) { 4120 lStatus = PERMISSION_DENIED; 4121 goto Exit; 4122 } 4123 4124 // add client to list 4125 { // scope for mLock 4126 Mutex::Autolock _l(mLock); 4127 thread = checkRecordThread_l(input); 4128 if (thread == NULL) { 4129 lStatus = BAD_VALUE; 4130 goto Exit; 4131 } 4132 4133 wclient = mClients.valueFor(pid); 4134 if (wclient != NULL) { 4135 client = wclient.promote(); 4136 } else { 4137 client = new Client(this, pid); 4138 mClients.add(pid, client); 4139 } 4140 4141 // If no audio session id is provided, create one here 4142 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4143 lSessionId = *sessionId; 4144 } else { 4145 lSessionId = nextUniqueId(); 4146 if (sessionId != NULL) { 4147 *sessionId = lSessionId; 4148 } 4149 } 4150 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4151 recordTrack = thread->createRecordTrack_l(client, 4152 sampleRate, 4153 format, 4154 channelMask, 4155 frameCount, 4156 flags, 4157 lSessionId, 4158 &lStatus); 4159 } 4160 if (lStatus != NO_ERROR) { 4161 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4162 // destructor is called by the TrackBase destructor with mLock held 4163 client.clear(); 4164 recordTrack.clear(); 4165 goto Exit; 4166 } 4167 4168 // return to handle to client 4169 recordHandle = new RecordHandle(recordTrack); 4170 lStatus = NO_ERROR; 4171 4172 Exit: 4173 if (status) { 4174 *status = lStatus; 4175 } 4176 return recordHandle; 4177 } 4178 4179 // ---------------------------------------------------------------------------- 4180 4181 AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4182 : BnAudioRecord(), 4183 mRecordTrack(recordTrack) 4184 { 4185 } 4186 4187 AudioFlinger::RecordHandle::~RecordHandle() { 4188 stop(); 4189 } 4190 4191 status_t AudioFlinger::RecordHandle::start() { 4192 LOGV("RecordHandle::start()"); 4193 return mRecordTrack->start(); 4194 } 4195 4196 void AudioFlinger::RecordHandle::stop() { 4197 LOGV("RecordHandle::stop()"); 4198 mRecordTrack->stop(); 4199 } 4200 4201 sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4202 return mRecordTrack->getCblk(); 4203 } 4204 4205 status_t AudioFlinger::RecordHandle::onTransact( 4206 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4207 { 4208 return BnAudioRecord::onTransact(code, data, reply, flags); 4209 } 4210 4211 // ---------------------------------------------------------------------------- 4212 4213 AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4214 AudioStreamIn *input, 4215 uint32_t sampleRate, 4216 uint32_t channels, 4217 int id, 4218 uint32_t device) : 4219 ThreadBase(audioFlinger, id, device), 4220 mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 4221 { 4222 mType = ThreadBase::RECORD; 4223 4224 snprintf(mName, kNameLength, "AudioIn_%d", id); 4225 4226 mReqChannelCount = popcount(channels); 4227 mReqSampleRate = sampleRate; 4228 readInputParameters(); 4229 } 4230 4231 4232 AudioFlinger::RecordThread::~RecordThread() 4233 { 4234 delete[] mRsmpInBuffer; 4235 if (mResampler != 0) { 4236 delete mResampler; 4237 delete[] mRsmpOutBuffer; 4238 } 4239 } 4240 4241 void AudioFlinger::RecordThread::onFirstRef() 4242 { 4243 run(mName, PRIORITY_URGENT_AUDIO); 4244 } 4245 4246 status_t AudioFlinger::RecordThread::readyToRun() 4247 { 4248 status_t status = initCheck(); 4249 LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4250 return status; 4251 } 4252 4253 bool AudioFlinger::RecordThread::threadLoop() 4254 { 4255 AudioBufferProvider::Buffer buffer; 4256 sp<RecordTrack> activeTrack; 4257 Vector< sp<EffectChain> > effectChains; 4258 4259 nsecs_t lastWarning = 0; 4260 4261 acquireWakeLock(); 4262 4263 // start recording 4264 while (!exitPending()) { 4265 4266 processConfigEvents(); 4267 4268 { // scope for mLock 4269 Mutex::Autolock _l(mLock); 4270 checkForNewParameters_l(); 4271 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4272 if (!mStandby) { 4273 mInput->stream->common.standby(&mInput->stream->common); 4274 mStandby = true; 4275 } 4276 4277 if (exitPending()) break; 4278 4279 releaseWakeLock_l(); 4280 LOGV("RecordThread: loop stopping"); 4281 // go to sleep 4282 mWaitWorkCV.wait(mLock); 4283 LOGV("RecordThread: loop starting"); 4284 acquireWakeLock_l(); 4285 continue; 4286 } 4287 if (mActiveTrack != 0) { 4288 if (mActiveTrack->mState == TrackBase::PAUSING) { 4289 if (!mStandby) { 4290 mInput->stream->common.standby(&mInput->stream->common); 4291 mStandby = true; 4292 } 4293 mActiveTrack.clear(); 4294 mStartStopCond.broadcast(); 4295 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4296 if (mReqChannelCount != mActiveTrack->channelCount()) { 4297 mActiveTrack.clear(); 4298 mStartStopCond.broadcast(); 4299 } else if (mBytesRead != 0) { 4300 // record start succeeds only if first read from audio input 4301 // succeeds 4302 if (mBytesRead > 0) { 4303 mActiveTrack->mState = TrackBase::ACTIVE; 4304 } else { 4305 mActiveTrack.clear(); 4306 } 4307 mStartStopCond.broadcast(); 4308 } 4309 mStandby = false; 4310 } 4311 } 4312 lockEffectChains_l(effectChains); 4313 } 4314 4315 if (mActiveTrack != 0) { 4316 if (mActiveTrack->mState != TrackBase::ACTIVE && 4317 mActiveTrack->mState != TrackBase::RESUMING) { 4318 unlockEffectChains(effectChains); 4319 usleep(kRecordThreadSleepUs); 4320 continue; 4321 } 4322 for (size_t i = 0; i < effectChains.size(); i ++) { 4323 effectChains[i]->process_l(); 4324 } 4325 4326 buffer.frameCount = mFrameCount; 4327 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4328 size_t framesOut = buffer.frameCount; 4329 if (mResampler == 0) { 4330 // no resampling 4331 while (framesOut) { 4332 size_t framesIn = mFrameCount - mRsmpInIndex; 4333 if (framesIn) { 4334 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4335 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4336 if (framesIn > framesOut) 4337 framesIn = framesOut; 4338 mRsmpInIndex += framesIn; 4339 framesOut -= framesIn; 4340 if ((int)mChannelCount == mReqChannelCount || 4341 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4342 memcpy(dst, src, framesIn * mFrameSize); 4343 } else { 4344 int16_t *src16 = (int16_t *)src; 4345 int16_t *dst16 = (int16_t *)dst; 4346 if (mChannelCount == 1) { 4347 while (framesIn--) { 4348 *dst16++ = *src16; 4349 *dst16++ = *src16++; 4350 } 4351 } else { 4352 while (framesIn--) { 4353 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4354 src16 += 2; 4355 } 4356 } 4357 } 4358 } 4359 if (framesOut && mFrameCount == mRsmpInIndex) { 4360 if (framesOut == mFrameCount && 4361 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4362 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4363 framesOut = 0; 4364 } else { 4365 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4366 mRsmpInIndex = 0; 4367 } 4368 if (mBytesRead < 0) { 4369 LOGE("Error reading audio input"); 4370 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4371 // Force input into standby so that it tries to 4372 // recover at next read attempt 4373 mInput->stream->common.standby(&mInput->stream->common); 4374 usleep(kRecordThreadSleepUs); 4375 } 4376 mRsmpInIndex = mFrameCount; 4377 framesOut = 0; 4378 buffer.frameCount = 0; 4379 } 4380 } 4381 } 4382 } else { 4383 // resampling 4384 4385 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4386 // alter output frame count as if we were expecting stereo samples 4387 if (mChannelCount == 1 && mReqChannelCount == 1) { 4388 framesOut >>= 1; 4389 } 4390 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4391 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4392 // are 32 bit aligned which should be always true. 4393 if (mChannelCount == 2 && mReqChannelCount == 1) { 4394 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4395 // the resampler always outputs stereo samples: do post stereo to mono conversion 4396 int16_t *src = (int16_t *)mRsmpOutBuffer; 4397 int16_t *dst = buffer.i16; 4398 while (framesOut--) { 4399 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4400 src += 2; 4401 } 4402 } else { 4403 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4404 } 4405 4406 } 4407 mActiveTrack->releaseBuffer(&buffer); 4408 mActiveTrack->overflow(); 4409 } 4410 // client isn't retrieving buffers fast enough 4411 else { 4412 if (!mActiveTrack->setOverflow()) { 4413 nsecs_t now = systemTime(); 4414 if ((now - lastWarning) > kWarningThrottle) { 4415 LOGW("RecordThread: buffer overflow"); 4416 lastWarning = now; 4417 } 4418 } 4419 // Release the processor for a while before asking for a new buffer. 4420 // This will give the application more chance to read from the buffer and 4421 // clear the overflow. 4422 usleep(kRecordThreadSleepUs); 4423 } 4424 } 4425 // enable changes in effect chain 4426 unlockEffectChains(effectChains); 4427 effectChains.clear(); 4428 } 4429 4430 if (!mStandby) { 4431 mInput->stream->common.standby(&mInput->stream->common); 4432 } 4433 mActiveTrack.clear(); 4434 4435 mStartStopCond.broadcast(); 4436 4437 releaseWakeLock(); 4438 4439 LOGV("RecordThread %p exiting", this); 4440 return false; 4441 } 4442 4443 4444 sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4445 const sp<AudioFlinger::Client>& client, 4446 uint32_t sampleRate, 4447 int format, 4448 int channelMask, 4449 int frameCount, 4450 uint32_t flags, 4451 int sessionId, 4452 status_t *status) 4453 { 4454 sp<RecordTrack> track; 4455 status_t lStatus; 4456 4457 lStatus = initCheck(); 4458 if (lStatus != NO_ERROR) { 4459 LOGE("Audio driver not initialized."); 4460 goto Exit; 4461 } 4462 4463 { // scope for mLock 4464 Mutex::Autolock _l(mLock); 4465 4466 track = new RecordTrack(this, client, sampleRate, 4467 format, channelMask, frameCount, flags, sessionId); 4468 4469 if (track->getCblk() == NULL) { 4470 lStatus = NO_MEMORY; 4471 goto Exit; 4472 } 4473 4474 mTrack = track.get(); 4475 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4476 bool suspend = audio_is_bluetooth_sco_device( 4477 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4478 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4479 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4480 } 4481 lStatus = NO_ERROR; 4482 4483 Exit: 4484 if (status) { 4485 *status = lStatus; 4486 } 4487 return track; 4488 } 4489 4490 status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4491 { 4492 LOGV("RecordThread::start"); 4493 sp <ThreadBase> strongMe = this; 4494 status_t status = NO_ERROR; 4495 { 4496 AutoMutex lock(&mLock); 4497 if (mActiveTrack != 0) { 4498 if (recordTrack != mActiveTrack.get()) { 4499 status = -EBUSY; 4500 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4501 mActiveTrack->mState = TrackBase::ACTIVE; 4502 } 4503 return status; 4504 } 4505 4506 recordTrack->mState = TrackBase::IDLE; 4507 mActiveTrack = recordTrack; 4508 mLock.unlock(); 4509 status_t status = AudioSystem::startInput(mId); 4510 mLock.lock(); 4511 if (status != NO_ERROR) { 4512 mActiveTrack.clear(); 4513 return status; 4514 } 4515 mRsmpInIndex = mFrameCount; 4516 mBytesRead = 0; 4517 if (mResampler != NULL) { 4518 mResampler->reset(); 4519 } 4520 mActiveTrack->mState = TrackBase::RESUMING; 4521 // signal thread to start 4522 LOGV("Signal record thread"); 4523 mWaitWorkCV.signal(); 4524 // do not wait for mStartStopCond if exiting 4525 if (mExiting) { 4526 mActiveTrack.clear(); 4527 status = INVALID_OPERATION; 4528 goto startError; 4529 } 4530 mStartStopCond.wait(mLock); 4531 if (mActiveTrack == 0) { 4532 LOGV("Record failed to start"); 4533 status = BAD_VALUE; 4534 goto startError; 4535 } 4536 LOGV("Record started OK"); 4537 return status; 4538 } 4539 startError: 4540 AudioSystem::stopInput(mId); 4541 return status; 4542 } 4543 4544 void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4545 LOGV("RecordThread::stop"); 4546 sp <ThreadBase> strongMe = this; 4547 { 4548 AutoMutex lock(&mLock); 4549 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4550 mActiveTrack->mState = TrackBase::PAUSING; 4551 // do not wait for mStartStopCond if exiting 4552 if (mExiting) { 4553 return; 4554 } 4555 mStartStopCond.wait(mLock); 4556 // if we have been restarted, recordTrack == mActiveTrack.get() here 4557 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4558 mLock.unlock(); 4559 AudioSystem::stopInput(mId); 4560 mLock.lock(); 4561 LOGV("Record stopped OK"); 4562 } 4563 } 4564 } 4565 } 4566 4567 status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4568 { 4569 const size_t SIZE = 256; 4570 char buffer[SIZE]; 4571 String8 result; 4572 pid_t pid = 0; 4573 4574 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4575 result.append(buffer); 4576 4577 if (mActiveTrack != 0) { 4578 result.append("Active Track:\n"); 4579 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4580 mActiveTrack->dump(buffer, SIZE); 4581 result.append(buffer); 4582 4583 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4584 result.append(buffer); 4585 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4586 result.append(buffer); 4587 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4588 result.append(buffer); 4589 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4590 result.append(buffer); 4591 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4592 result.append(buffer); 4593 4594 4595 } else { 4596 result.append("No record client\n"); 4597 } 4598 write(fd, result.string(), result.size()); 4599 4600 dumpBase(fd, args); 4601 dumpEffectChains(fd, args); 4602 4603 return NO_ERROR; 4604 } 4605 4606 status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4607 { 4608 size_t framesReq = buffer->frameCount; 4609 size_t framesReady = mFrameCount - mRsmpInIndex; 4610 int channelCount; 4611 4612 if (framesReady == 0) { 4613 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4614 if (mBytesRead < 0) { 4615 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4616 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4617 // Force input into standby so that it tries to 4618 // recover at next read attempt 4619 mInput->stream->common.standby(&mInput->stream->common); 4620 usleep(kRecordThreadSleepUs); 4621 } 4622 buffer->raw = 0; 4623 buffer->frameCount = 0; 4624 return NOT_ENOUGH_DATA; 4625 } 4626 mRsmpInIndex = 0; 4627 framesReady = mFrameCount; 4628 } 4629 4630 if (framesReq > framesReady) { 4631 framesReq = framesReady; 4632 } 4633 4634 if (mChannelCount == 1 && mReqChannelCount == 2) { 4635 channelCount = 1; 4636 } else { 4637 channelCount = 2; 4638 } 4639 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4640 buffer->frameCount = framesReq; 4641 return NO_ERROR; 4642 } 4643 4644 void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4645 { 4646 mRsmpInIndex += buffer->frameCount; 4647 buffer->frameCount = 0; 4648 } 4649 4650 bool AudioFlinger::RecordThread::checkForNewParameters_l() 4651 { 4652 bool reconfig = false; 4653 4654 while (!mNewParameters.isEmpty()) { 4655 status_t status = NO_ERROR; 4656 String8 keyValuePair = mNewParameters[0]; 4657 AudioParameter param = AudioParameter(keyValuePair); 4658 int value; 4659 int reqFormat = mFormat; 4660 int reqSamplingRate = mReqSampleRate; 4661 int reqChannelCount = mReqChannelCount; 4662 4663 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4664 reqSamplingRate = value; 4665 reconfig = true; 4666 } 4667 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4668 reqFormat = value; 4669 reconfig = true; 4670 } 4671 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4672 reqChannelCount = popcount(value); 4673 reconfig = true; 4674 } 4675 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4676 // do not accept frame count changes if tracks are open as the track buffer 4677 // size depends on frame count and correct behavior would not be garantied 4678 // if frame count is changed after track creation 4679 if (mActiveTrack != 0) { 4680 status = INVALID_OPERATION; 4681 } else { 4682 reconfig = true; 4683 } 4684 } 4685 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4686 // forward device change to effects that have requested to be 4687 // aware of attached audio device. 4688 for (size_t i = 0; i < mEffectChains.size(); i++) { 4689 mEffectChains[i]->setDevice_l(value); 4690 } 4691 // store input device and output device but do not forward output device to audio HAL. 4692 // Note that status is ignored by the caller for output device 4693 // (see AudioFlinger::setParameters() 4694 if (value & AUDIO_DEVICE_OUT_ALL) { 4695 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4696 status = BAD_VALUE; 4697 } else { 4698 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4699 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4700 if (mTrack != NULL) { 4701 bool suspend = audio_is_bluetooth_sco_device( 4702 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4703 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4704 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4705 } 4706 } 4707 mDevice |= (uint32_t)value; 4708 } 4709 if (status == NO_ERROR) { 4710 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4711 if (status == INVALID_OPERATION) { 4712 mInput->stream->common.standby(&mInput->stream->common); 4713 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4714 } 4715 if (reconfig) { 4716 if (status == BAD_VALUE && 4717 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4718 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4719 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4720 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4721 (reqChannelCount < 3)) { 4722 status = NO_ERROR; 4723 } 4724 if (status == NO_ERROR) { 4725 readInputParameters(); 4726 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4727 } 4728 } 4729 } 4730 4731 mNewParameters.removeAt(0); 4732 4733 mParamStatus = status; 4734 mParamCond.signal(); 4735 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4736 // already timed out waiting for the status and will never signal the condition. 4737 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 4738 } 4739 return reconfig; 4740 } 4741 4742 String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4743 { 4744 char *s; 4745 String8 out_s8 = String8(); 4746 4747 Mutex::Autolock _l(mLock); 4748 if (initCheck() != NO_ERROR) { 4749 return out_s8; 4750 } 4751 4752 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4753 out_s8 = String8(s); 4754 free(s); 4755 return out_s8; 4756 } 4757 4758 void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4759 AudioSystem::OutputDescriptor desc; 4760 void *param2 = 0; 4761 4762 switch (event) { 4763 case AudioSystem::INPUT_OPENED: 4764 case AudioSystem::INPUT_CONFIG_CHANGED: 4765 desc.channels = mChannelMask; 4766 desc.samplingRate = mSampleRate; 4767 desc.format = mFormat; 4768 desc.frameCount = mFrameCount; 4769 desc.latency = 0; 4770 param2 = &desc; 4771 break; 4772 4773 case AudioSystem::INPUT_CLOSED: 4774 default: 4775 break; 4776 } 4777 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4778 } 4779 4780 void AudioFlinger::RecordThread::readInputParameters() 4781 { 4782 if (mRsmpInBuffer) delete mRsmpInBuffer; 4783 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4784 if (mResampler) delete mResampler; 4785 mResampler = 0; 4786 4787 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4788 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4789 mChannelCount = (uint16_t)popcount(mChannelMask); 4790 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4791 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4792 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4793 mFrameCount = mInputBytes / mFrameSize; 4794 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4795 4796 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4797 { 4798 int channelCount; 4799 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4800 // stereo to mono post process as the resampler always outputs stereo. 4801 if (mChannelCount == 1 && mReqChannelCount == 2) { 4802 channelCount = 1; 4803 } else { 4804 channelCount = 2; 4805 } 4806 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4807 mResampler->setSampleRate(mSampleRate); 4808 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4809 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4810 4811 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4812 if (mChannelCount == 1 && mReqChannelCount == 1) { 4813 mFrameCount >>= 1; 4814 } 4815 4816 } 4817 mRsmpInIndex = mFrameCount; 4818 } 4819 4820 unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4821 { 4822 Mutex::Autolock _l(mLock); 4823 if (initCheck() != NO_ERROR) { 4824 return 0; 4825 } 4826 4827 return mInput->stream->get_input_frames_lost(mInput->stream); 4828 } 4829 4830 uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4831 { 4832 Mutex::Autolock _l(mLock); 4833 uint32_t result = 0; 4834 if (getEffectChain_l(sessionId) != 0) { 4835 result = EFFECT_SESSION; 4836 } 4837 4838 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4839 result |= TRACK_SESSION; 4840 } 4841 4842 return result; 4843 } 4844 4845 AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4846 { 4847 Mutex::Autolock _l(mLock); 4848 return mTrack; 4849 } 4850 4851 AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4852 { 4853 Mutex::Autolock _l(mLock); 4854 return mInput; 4855 } 4856 4857 AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4858 { 4859 Mutex::Autolock _l(mLock); 4860 AudioStreamIn *input = mInput; 4861 mInput = NULL; 4862 return input; 4863 } 4864 4865 // this method must always be called either with ThreadBase mLock held or inside the thread loop 4866 audio_stream_t* AudioFlinger::RecordThread::stream() 4867 { 4868 if (mInput == NULL) { 4869 return NULL; 4870 } 4871 return &mInput->stream->common; 4872 } 4873 4874 4875 // ---------------------------------------------------------------------------- 4876 4877 int AudioFlinger::openOutput(uint32_t *pDevices, 4878 uint32_t *pSamplingRate, 4879 uint32_t *pFormat, 4880 uint32_t *pChannels, 4881 uint32_t *pLatencyMs, 4882 uint32_t flags) 4883 { 4884 status_t status; 4885 PlaybackThread *thread = NULL; 4886 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4887 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4888 uint32_t format = pFormat ? *pFormat : 0; 4889 uint32_t channels = pChannels ? *pChannels : 0; 4890 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4891 audio_stream_out_t *outStream; 4892 audio_hw_device_t *outHwDev; 4893 4894 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4895 pDevices ? *pDevices : 0, 4896 samplingRate, 4897 format, 4898 channels, 4899 flags); 4900 4901 if (pDevices == NULL || *pDevices == 0) { 4902 return 0; 4903 } 4904 4905 Mutex::Autolock _l(mLock); 4906 4907 outHwDev = findSuitableHwDev_l(*pDevices); 4908 if (outHwDev == NULL) 4909 return 0; 4910 4911 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4912 &channels, &samplingRate, &outStream); 4913 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4914 outStream, 4915 samplingRate, 4916 format, 4917 channels, 4918 status); 4919 4920 mHardwareStatus = AUDIO_HW_IDLE; 4921 if (outStream != NULL) { 4922 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4923 int id = nextUniqueId(); 4924 4925 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4926 (format != AUDIO_FORMAT_PCM_16_BIT) || 4927 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4928 thread = new DirectOutputThread(this, output, id, *pDevices); 4929 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4930 } else { 4931 thread = new MixerThread(this, output, id, *pDevices); 4932 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4933 } 4934 mPlaybackThreads.add(id, thread); 4935 4936 if (pSamplingRate) *pSamplingRate = samplingRate; 4937 if (pFormat) *pFormat = format; 4938 if (pChannels) *pChannels = channels; 4939 if (pLatencyMs) *pLatencyMs = thread->latency(); 4940 4941 // notify client processes of the new output creation 4942 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4943 return id; 4944 } 4945 4946 return 0; 4947 } 4948 4949 int AudioFlinger::openDuplicateOutput(int output1, int output2) 4950 { 4951 Mutex::Autolock _l(mLock); 4952 MixerThread *thread1 = checkMixerThread_l(output1); 4953 MixerThread *thread2 = checkMixerThread_l(output2); 4954 4955 if (thread1 == NULL || thread2 == NULL) { 4956 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4957 return 0; 4958 } 4959 4960 int id = nextUniqueId(); 4961 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4962 thread->addOutputTrack(thread2); 4963 mPlaybackThreads.add(id, thread); 4964 // notify client processes of the new output creation 4965 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4966 return id; 4967 } 4968 4969 status_t AudioFlinger::closeOutput(int output) 4970 { 4971 // keep strong reference on the playback thread so that 4972 // it is not destroyed while exit() is executed 4973 sp <PlaybackThread> thread; 4974 { 4975 Mutex::Autolock _l(mLock); 4976 thread = checkPlaybackThread_l(output); 4977 if (thread == NULL) { 4978 return BAD_VALUE; 4979 } 4980 4981 LOGV("closeOutput() %d", output); 4982 4983 if (thread->type() == ThreadBase::MIXER) { 4984 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4985 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4986 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4987 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4988 } 4989 } 4990 } 4991 void *param2 = 0; 4992 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4993 mPlaybackThreads.removeItem(output); 4994 } 4995 thread->exit(); 4996 4997 if (thread->type() != ThreadBase::DUPLICATING) { 4998 AudioStreamOut *out = thread->clearOutput(); 4999 // from now on thread->mOutput is NULL 5000 out->hwDev->close_output_stream(out->hwDev, out->stream); 5001 delete out; 5002 } 5003 return NO_ERROR; 5004 } 5005 5006 status_t AudioFlinger::suspendOutput(int output) 5007 { 5008 Mutex::Autolock _l(mLock); 5009 PlaybackThread *thread = checkPlaybackThread_l(output); 5010 5011 if (thread == NULL) { 5012 return BAD_VALUE; 5013 } 5014 5015 LOGV("suspendOutput() %d", output); 5016 thread->suspend(); 5017 5018 return NO_ERROR; 5019 } 5020 5021 status_t AudioFlinger::restoreOutput(int output) 5022 { 5023 Mutex::Autolock _l(mLock); 5024 PlaybackThread *thread = checkPlaybackThread_l(output); 5025 5026 if (thread == NULL) { 5027 return BAD_VALUE; 5028 } 5029 5030 LOGV("restoreOutput() %d", output); 5031 5032 thread->restore(); 5033 5034 return NO_ERROR; 5035 } 5036 5037 int AudioFlinger::openInput(uint32_t *pDevices, 5038 uint32_t *pSamplingRate, 5039 uint32_t *pFormat, 5040 uint32_t *pChannels, 5041 uint32_t acoustics) 5042 { 5043 status_t status; 5044 RecordThread *thread = NULL; 5045 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5046 uint32_t format = pFormat ? *pFormat : 0; 5047 uint32_t channels = pChannels ? *pChannels : 0; 5048 uint32_t reqSamplingRate = samplingRate; 5049 uint32_t reqFormat = format; 5050 uint32_t reqChannels = channels; 5051 audio_stream_in_t *inStream; 5052 audio_hw_device_t *inHwDev; 5053 5054 if (pDevices == NULL || *pDevices == 0) { 5055 return 0; 5056 } 5057 5058 Mutex::Autolock _l(mLock); 5059 5060 inHwDev = findSuitableHwDev_l(*pDevices); 5061 if (inHwDev == NULL) 5062 return 0; 5063 5064 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5065 &channels, &samplingRate, 5066 (audio_in_acoustics_t)acoustics, 5067 &inStream); 5068 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5069 inStream, 5070 samplingRate, 5071 format, 5072 channels, 5073 acoustics, 5074 status); 5075 5076 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5077 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5078 // or stereo to mono conversions on 16 bit PCM inputs. 5079 if (inStream == NULL && status == BAD_VALUE && 5080 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5081 (samplingRate <= 2 * reqSamplingRate) && 5082 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5083 LOGV("openInput() reopening with proposed sampling rate and channels"); 5084 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5085 &channels, &samplingRate, 5086 (audio_in_acoustics_t)acoustics, 5087 &inStream); 5088 } 5089 5090 if (inStream != NULL) { 5091 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5092 5093 int id = nextUniqueId(); 5094 // Start record thread 5095 // RecorThread require both input and output device indication to forward to audio 5096 // pre processing modules 5097 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5098 thread = new RecordThread(this, 5099 input, 5100 reqSamplingRate, 5101 reqChannels, 5102 id, 5103 device); 5104 mRecordThreads.add(id, thread); 5105 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 5106 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5107 if (pFormat) *pFormat = format; 5108 if (pChannels) *pChannels = reqChannels; 5109 5110 input->stream->common.standby(&input->stream->common); 5111 5112 // notify client processes of the new input creation 5113 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5114 return id; 5115 } 5116 5117 return 0; 5118 } 5119 5120 status_t AudioFlinger::closeInput(int input) 5121 { 5122 // keep strong reference on the record thread so that 5123 // it is not destroyed while exit() is executed 5124 sp <RecordThread> thread; 5125 { 5126 Mutex::Autolock _l(mLock); 5127 thread = checkRecordThread_l(input); 5128 if (thread == NULL) { 5129 return BAD_VALUE; 5130 } 5131 5132 LOGV("closeInput() %d", input); 5133 void *param2 = 0; 5134 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5135 mRecordThreads.removeItem(input); 5136 } 5137 thread->exit(); 5138 5139 AudioStreamIn *in = thread->clearInput(); 5140 // from now on thread->mInput is NULL 5141 in->hwDev->close_input_stream(in->hwDev, in->stream); 5142 delete in; 5143 5144 return NO_ERROR; 5145 } 5146 5147 status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 5148 { 5149 Mutex::Autolock _l(mLock); 5150 MixerThread *dstThread = checkMixerThread_l(output); 5151 if (dstThread == NULL) { 5152 LOGW("setStreamOutput() bad output id %d", output); 5153 return BAD_VALUE; 5154 } 5155 5156 LOGV("setStreamOutput() stream %d to output %d", stream, output); 5157 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5158 5159 dstThread->setStreamValid(stream, true); 5160 5161 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5162 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5163 if (thread != dstThread && 5164 thread->type() != ThreadBase::DIRECT) { 5165 MixerThread *srcThread = (MixerThread *)thread; 5166 srcThread->setStreamValid(stream, false); 5167 srcThread->invalidateTracks(stream); 5168 } 5169 } 5170 5171 return NO_ERROR; 5172 } 5173 5174 5175 int AudioFlinger::newAudioSessionId() 5176 { 5177 return nextUniqueId(); 5178 } 5179 5180 void AudioFlinger::acquireAudioSessionId(int audioSession) 5181 { 5182 Mutex::Autolock _l(mLock); 5183 int caller = IPCThreadState::self()->getCallingPid(); 5184 LOGV("acquiring %d from %d", audioSession, caller); 5185 int num = mAudioSessionRefs.size(); 5186 for (int i = 0; i< num; i++) { 5187 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5188 if (ref->sessionid == audioSession && ref->pid == caller) { 5189 ref->cnt++; 5190 LOGV(" incremented refcount to %d", ref->cnt); 5191 return; 5192 } 5193 } 5194 AudioSessionRef *ref = new AudioSessionRef(); 5195 ref->sessionid = audioSession; 5196 ref->pid = caller; 5197 ref->cnt = 1; 5198 mAudioSessionRefs.push(ref); 5199 LOGV(" added new entry for %d", ref->sessionid); 5200 } 5201 5202 void AudioFlinger::releaseAudioSessionId(int audioSession) 5203 { 5204 Mutex::Autolock _l(mLock); 5205 int caller = IPCThreadState::self()->getCallingPid(); 5206 LOGV("releasing %d from %d", audioSession, caller); 5207 int num = mAudioSessionRefs.size(); 5208 for (int i = 0; i< num; i++) { 5209 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5210 if (ref->sessionid == audioSession && ref->pid == caller) { 5211 ref->cnt--; 5212 LOGV(" decremented refcount to %d", ref->cnt); 5213 if (ref->cnt == 0) { 5214 mAudioSessionRefs.removeAt(i); 5215 delete ref; 5216 purgeStaleEffects_l(); 5217 } 5218 return; 5219 } 5220 } 5221 LOGW("session id %d not found for pid %d", audioSession, caller); 5222 } 5223 5224 void AudioFlinger::purgeStaleEffects_l() { 5225 5226 LOGV("purging stale effects"); 5227 5228 Vector< sp<EffectChain> > chains; 5229 5230 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5231 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5232 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5233 sp<EffectChain> ec = t->mEffectChains[j]; 5234 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5235 chains.push(ec); 5236 } 5237 } 5238 } 5239 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5240 sp<RecordThread> t = mRecordThreads.valueAt(i); 5241 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5242 sp<EffectChain> ec = t->mEffectChains[j]; 5243 chains.push(ec); 5244 } 5245 } 5246 5247 for (size_t i = 0; i < chains.size(); i++) { 5248 sp<EffectChain> ec = chains[i]; 5249 int sessionid = ec->sessionId(); 5250 sp<ThreadBase> t = ec->mThread.promote(); 5251 if (t == 0) { 5252 continue; 5253 } 5254 size_t numsessionrefs = mAudioSessionRefs.size(); 5255 bool found = false; 5256 for (size_t k = 0; k < numsessionrefs; k++) { 5257 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5258 if (ref->sessionid == sessionid) { 5259 LOGV(" session %d still exists for %d with %d refs", 5260 sessionid, ref->pid, ref->cnt); 5261 found = true; 5262 break; 5263 } 5264 } 5265 if (!found) { 5266 // remove all effects from the chain 5267 while (ec->mEffects.size()) { 5268 sp<EffectModule> effect = ec->mEffects[0]; 5269 effect->unPin(); 5270 Mutex::Autolock _l (t->mLock); 5271 t->removeEffect_l(effect); 5272 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5273 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5274 if (handle != 0) { 5275 handle->mEffect.clear(); 5276 if (handle->mHasControl && handle->mEnabled) { 5277 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5278 } 5279 } 5280 } 5281 AudioSystem::unregisterEffect(effect->id()); 5282 } 5283 } 5284 } 5285 return; 5286 } 5287 5288 // checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5289 AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5290 { 5291 PlaybackThread *thread = NULL; 5292 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5293 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5294 } 5295 return thread; 5296 } 5297 5298 // checkMixerThread_l() must be called with AudioFlinger::mLock held 5299 AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5300 { 5301 PlaybackThread *thread = checkPlaybackThread_l(output); 5302 if (thread != NULL) { 5303 if (thread->type() == ThreadBase::DIRECT) { 5304 thread = NULL; 5305 } 5306 } 5307 return (MixerThread *)thread; 5308 } 5309 5310 // checkRecordThread_l() must be called with AudioFlinger::mLock held 5311 AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5312 { 5313 RecordThread *thread = NULL; 5314 if (mRecordThreads.indexOfKey(input) >= 0) { 5315 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5316 } 5317 return thread; 5318 } 5319 5320 uint32_t AudioFlinger::nextUniqueId() 5321 { 5322 return android_atomic_inc(&mNextUniqueId); 5323 } 5324 5325 AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5326 { 5327 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5328 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5329 AudioStreamOut *output = thread->getOutput(); 5330 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5331 return thread; 5332 } 5333 } 5334 return NULL; 5335 } 5336 5337 uint32_t AudioFlinger::primaryOutputDevice_l() 5338 { 5339 PlaybackThread *thread = primaryPlaybackThread_l(); 5340 5341 if (thread == NULL) { 5342 return 0; 5343 } 5344 5345 return thread->device(); 5346 } 5347 5348 5349 // ---------------------------------------------------------------------------- 5350 // Effect management 5351 // ---------------------------------------------------------------------------- 5352 5353 5354 status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5355 { 5356 Mutex::Autolock _l(mLock); 5357 return EffectQueryNumberEffects(numEffects); 5358 } 5359 5360 status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5361 { 5362 Mutex::Autolock _l(mLock); 5363 return EffectQueryEffect(index, descriptor); 5364 } 5365 5366 status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5367 { 5368 Mutex::Autolock _l(mLock); 5369 return EffectGetDescriptor(pUuid, descriptor); 5370 } 5371 5372 5373 sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5374 effect_descriptor_t *pDesc, 5375 const sp<IEffectClient>& effectClient, 5376 int32_t priority, 5377 int io, 5378 int sessionId, 5379 status_t *status, 5380 int *id, 5381 int *enabled) 5382 { 5383 status_t lStatus = NO_ERROR; 5384 sp<EffectHandle> handle; 5385 effect_descriptor_t desc; 5386 sp<Client> client; 5387 wp<Client> wclient; 5388 5389 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5390 pid, effectClient.get(), priority, sessionId, io); 5391 5392 if (pDesc == NULL) { 5393 lStatus = BAD_VALUE; 5394 goto Exit; 5395 } 5396 5397 // check audio settings permission for global effects 5398 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5399 lStatus = PERMISSION_DENIED; 5400 goto Exit; 5401 } 5402 5403 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5404 // that can only be created by audio policy manager (running in same process) 5405 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5406 lStatus = PERMISSION_DENIED; 5407 goto Exit; 5408 } 5409 5410 if (io == 0) { 5411 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5412 // output must be specified by AudioPolicyManager when using session 5413 // AUDIO_SESSION_OUTPUT_STAGE 5414 lStatus = BAD_VALUE; 5415 goto Exit; 5416 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5417 // if the output returned by getOutputForEffect() is removed before we lock the 5418 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5419 // and we will exit safely 5420 io = AudioSystem::getOutputForEffect(&desc); 5421 } 5422 } 5423 5424 { 5425 Mutex::Autolock _l(mLock); 5426 5427 5428 if (!EffectIsNullUuid(&pDesc->uuid)) { 5429 // if uuid is specified, request effect descriptor 5430 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5431 if (lStatus < 0) { 5432 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5433 goto Exit; 5434 } 5435 } else { 5436 // if uuid is not specified, look for an available implementation 5437 // of the required type in effect factory 5438 if (EffectIsNullUuid(&pDesc->type)) { 5439 LOGW("createEffect() no effect type"); 5440 lStatus = BAD_VALUE; 5441 goto Exit; 5442 } 5443 uint32_t numEffects = 0; 5444 effect_descriptor_t d; 5445 d.flags = 0; // prevent compiler warning 5446 bool found = false; 5447 5448 lStatus = EffectQueryNumberEffects(&numEffects); 5449 if (lStatus < 0) { 5450 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5451 goto Exit; 5452 } 5453 for (uint32_t i = 0; i < numEffects; i++) { 5454 lStatus = EffectQueryEffect(i, &desc); 5455 if (lStatus < 0) { 5456 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5457 continue; 5458 } 5459 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5460 // If matching type found save effect descriptor. If the session is 5461 // 0 and the effect is not auxiliary, continue enumeration in case 5462 // an auxiliary version of this effect type is available 5463 found = true; 5464 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5465 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5466 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5467 break; 5468 } 5469 } 5470 } 5471 if (!found) { 5472 lStatus = BAD_VALUE; 5473 LOGW("createEffect() effect not found"); 5474 goto Exit; 5475 } 5476 // For same effect type, chose auxiliary version over insert version if 5477 // connect to output mix (Compliance to OpenSL ES) 5478 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5479 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5480 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5481 } 5482 } 5483 5484 // Do not allow auxiliary effects on a session different from 0 (output mix) 5485 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5486 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5487 lStatus = INVALID_OPERATION; 5488 goto Exit; 5489 } 5490 5491 // check recording permission for visualizer 5492 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5493 !recordingAllowed()) { 5494 lStatus = PERMISSION_DENIED; 5495 goto Exit; 5496 } 5497 5498 // return effect descriptor 5499 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5500 5501 // If output is not specified try to find a matching audio session ID in one of the 5502 // output threads. 5503 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5504 // because of code checking output when entering the function. 5505 // Note: io is never 0 when creating an effect on an input 5506 if (io == 0) { 5507 // look for the thread where the specified audio session is present 5508 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5509 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5510 io = mPlaybackThreads.keyAt(i); 5511 break; 5512 } 5513 } 5514 if (io == 0) { 5515 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5516 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5517 io = mRecordThreads.keyAt(i); 5518 break; 5519 } 5520 } 5521 } 5522 // If no output thread contains the requested session ID, default to 5523 // first output. The effect chain will be moved to the correct output 5524 // thread when a track with the same session ID is created 5525 if (io == 0 && mPlaybackThreads.size()) { 5526 io = mPlaybackThreads.keyAt(0); 5527 } 5528 LOGV("createEffect() got io %d for effect %s", io, desc.name); 5529 } 5530 ThreadBase *thread = checkRecordThread_l(io); 5531 if (thread == NULL) { 5532 thread = checkPlaybackThread_l(io); 5533 if (thread == NULL) { 5534 LOGE("createEffect() unknown output thread"); 5535 lStatus = BAD_VALUE; 5536 goto Exit; 5537 } 5538 } 5539 5540 wclient = mClients.valueFor(pid); 5541 5542 if (wclient != NULL) { 5543 client = wclient.promote(); 5544 } else { 5545 client = new Client(this, pid); 5546 mClients.add(pid, client); 5547 } 5548 5549 // create effect on selected output thread 5550 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5551 &desc, enabled, &lStatus); 5552 if (handle != 0 && id != NULL) { 5553 *id = handle->id(); 5554 } 5555 } 5556 5557 Exit: 5558 if(status) { 5559 *status = lStatus; 5560 } 5561 return handle; 5562 } 5563 5564 status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5565 { 5566 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5567 sessionId, srcOutput, dstOutput); 5568 Mutex::Autolock _l(mLock); 5569 if (srcOutput == dstOutput) { 5570 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 5571 return NO_ERROR; 5572 } 5573 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5574 if (srcThread == NULL) { 5575 LOGW("moveEffects() bad srcOutput %d", srcOutput); 5576 return BAD_VALUE; 5577 } 5578 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5579 if (dstThread == NULL) { 5580 LOGW("moveEffects() bad dstOutput %d", dstOutput); 5581 return BAD_VALUE; 5582 } 5583 5584 Mutex::Autolock _dl(dstThread->mLock); 5585 Mutex::Autolock _sl(srcThread->mLock); 5586 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5587 5588 return NO_ERROR; 5589 } 5590 5591 // moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5592 status_t AudioFlinger::moveEffectChain_l(int sessionId, 5593 AudioFlinger::PlaybackThread *srcThread, 5594 AudioFlinger::PlaybackThread *dstThread, 5595 bool reRegister) 5596 { 5597 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5598 sessionId, srcThread, dstThread); 5599 5600 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5601 if (chain == 0) { 5602 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5603 sessionId, srcThread); 5604 return INVALID_OPERATION; 5605 } 5606 5607 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5608 // so that a new chain is created with correct parameters when first effect is added. This is 5609 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5610 // removed. 5611 srcThread->removeEffectChain_l(chain); 5612 5613 // transfer all effects one by one so that new effect chain is created on new thread with 5614 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5615 int dstOutput = dstThread->id(); 5616 sp<EffectChain> dstChain; 5617 uint32_t strategy = 0; // prevent compiler warning 5618 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5619 while (effect != 0) { 5620 srcThread->removeEffect_l(effect); 5621 dstThread->addEffect_l(effect); 5622 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5623 if (effect->state() == EffectModule::ACTIVE || 5624 effect->state() == EffectModule::STOPPING) { 5625 effect->start(); 5626 } 5627 // if the move request is not received from audio policy manager, the effect must be 5628 // re-registered with the new strategy and output 5629 if (dstChain == 0) { 5630 dstChain = effect->chain().promote(); 5631 if (dstChain == 0) { 5632 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5633 srcThread->addEffect_l(effect); 5634 return NO_INIT; 5635 } 5636 strategy = dstChain->strategy(); 5637 } 5638 if (reRegister) { 5639 AudioSystem::unregisterEffect(effect->id()); 5640 AudioSystem::registerEffect(&effect->desc(), 5641 dstOutput, 5642 strategy, 5643 sessionId, 5644 effect->id()); 5645 } 5646 effect = chain->getEffectFromId_l(0); 5647 } 5648 5649 return NO_ERROR; 5650 } 5651 5652 5653 // PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5654 sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5655 const sp<AudioFlinger::Client>& client, 5656 const sp<IEffectClient>& effectClient, 5657 int32_t priority, 5658 int sessionId, 5659 effect_descriptor_t *desc, 5660 int *enabled, 5661 status_t *status 5662 ) 5663 { 5664 sp<EffectModule> effect; 5665 sp<EffectHandle> handle; 5666 status_t lStatus; 5667 sp<EffectChain> chain; 5668 bool chainCreated = false; 5669 bool effectCreated = false; 5670 bool effectRegistered = false; 5671 5672 lStatus = initCheck(); 5673 if (lStatus != NO_ERROR) { 5674 LOGW("createEffect_l() Audio driver not initialized."); 5675 goto Exit; 5676 } 5677 5678 // Do not allow effects with session ID 0 on direct output or duplicating threads 5679 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5680 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5681 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5682 desc->name, sessionId); 5683 lStatus = BAD_VALUE; 5684 goto Exit; 5685 } 5686 // Only Pre processor effects are allowed on input threads and only on input threads 5687 if ((mType == RECORD && 5688 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5689 (mType != RECORD && 5690 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5691 LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5692 desc->name, desc->flags, mType); 5693 lStatus = BAD_VALUE; 5694 goto Exit; 5695 } 5696 5697 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5698 5699 { // scope for mLock 5700 Mutex::Autolock _l(mLock); 5701 5702 // check for existing effect chain with the requested audio session 5703 chain = getEffectChain_l(sessionId); 5704 if (chain == 0) { 5705 // create a new chain for this session 5706 LOGV("createEffect_l() new effect chain for session %d", sessionId); 5707 chain = new EffectChain(this, sessionId); 5708 addEffectChain_l(chain); 5709 chain->setStrategy(getStrategyForSession_l(sessionId)); 5710 chainCreated = true; 5711 } else { 5712 effect = chain->getEffectFromDesc_l(desc); 5713 } 5714 5715 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 5716 5717 if (effect == 0) { 5718 int id = mAudioFlinger->nextUniqueId(); 5719 // Check CPU and memory usage 5720 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5721 if (lStatus != NO_ERROR) { 5722 goto Exit; 5723 } 5724 effectRegistered = true; 5725 // create a new effect module if none present in the chain 5726 effect = new EffectModule(this, chain, desc, id, sessionId); 5727 lStatus = effect->status(); 5728 if (lStatus != NO_ERROR) { 5729 goto Exit; 5730 } 5731 lStatus = chain->addEffect_l(effect); 5732 if (lStatus != NO_ERROR) { 5733 goto Exit; 5734 } 5735 effectCreated = true; 5736 5737 effect->setDevice(mDevice); 5738 effect->setMode(mAudioFlinger->getMode()); 5739 } 5740 // create effect handle and connect it to effect module 5741 handle = new EffectHandle(effect, client, effectClient, priority); 5742 lStatus = effect->addHandle(handle); 5743 if (enabled) { 5744 *enabled = (int)effect->isEnabled(); 5745 } 5746 } 5747 5748 Exit: 5749 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5750 Mutex::Autolock _l(mLock); 5751 if (effectCreated) { 5752 chain->removeEffect_l(effect); 5753 } 5754 if (effectRegistered) { 5755 AudioSystem::unregisterEffect(effect->id()); 5756 } 5757 if (chainCreated) { 5758 removeEffectChain_l(chain); 5759 } 5760 handle.clear(); 5761 } 5762 5763 if(status) { 5764 *status = lStatus; 5765 } 5766 return handle; 5767 } 5768 5769 sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5770 { 5771 sp<EffectModule> effect; 5772 5773 sp<EffectChain> chain = getEffectChain_l(sessionId); 5774 if (chain != 0) { 5775 effect = chain->getEffectFromId_l(effectId); 5776 } 5777 return effect; 5778 } 5779 5780 // PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5781 // PlaybackThread::mLock held 5782 status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5783 { 5784 // check for existing effect chain with the requested audio session 5785 int sessionId = effect->sessionId(); 5786 sp<EffectChain> chain = getEffectChain_l(sessionId); 5787 bool chainCreated = false; 5788 5789 if (chain == 0) { 5790 // create a new chain for this session 5791 LOGV("addEffect_l() new effect chain for session %d", sessionId); 5792 chain = new EffectChain(this, sessionId); 5793 addEffectChain_l(chain); 5794 chain->setStrategy(getStrategyForSession_l(sessionId)); 5795 chainCreated = true; 5796 } 5797 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5798 5799 if (chain->getEffectFromId_l(effect->id()) != 0) { 5800 LOGW("addEffect_l() %p effect %s already present in chain %p", 5801 this, effect->desc().name, chain.get()); 5802 return BAD_VALUE; 5803 } 5804 5805 status_t status = chain->addEffect_l(effect); 5806 if (status != NO_ERROR) { 5807 if (chainCreated) { 5808 removeEffectChain_l(chain); 5809 } 5810 return status; 5811 } 5812 5813 effect->setDevice(mDevice); 5814 effect->setMode(mAudioFlinger->getMode()); 5815 return NO_ERROR; 5816 } 5817 5818 void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5819 5820 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5821 effect_descriptor_t desc = effect->desc(); 5822 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5823 detachAuxEffect_l(effect->id()); 5824 } 5825 5826 sp<EffectChain> chain = effect->chain().promote(); 5827 if (chain != 0) { 5828 // remove effect chain if removing last effect 5829 if (chain->removeEffect_l(effect) == 0) { 5830 removeEffectChain_l(chain); 5831 } 5832 } else { 5833 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5834 } 5835 } 5836 5837 void AudioFlinger::ThreadBase::lockEffectChains_l( 5838 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5839 { 5840 effectChains = mEffectChains; 5841 for (size_t i = 0; i < mEffectChains.size(); i++) { 5842 mEffectChains[i]->lock(); 5843 } 5844 } 5845 5846 void AudioFlinger::ThreadBase::unlockEffectChains( 5847 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5848 { 5849 for (size_t i = 0; i < effectChains.size(); i++) { 5850 effectChains[i]->unlock(); 5851 } 5852 } 5853 5854 sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5855 { 5856 Mutex::Autolock _l(mLock); 5857 return getEffectChain_l(sessionId); 5858 } 5859 5860 sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5861 { 5862 sp<EffectChain> chain; 5863 5864 size_t size = mEffectChains.size(); 5865 for (size_t i = 0; i < size; i++) { 5866 if (mEffectChains[i]->sessionId() == sessionId) { 5867 chain = mEffectChains[i]; 5868 break; 5869 } 5870 } 5871 return chain; 5872 } 5873 5874 void AudioFlinger::ThreadBase::setMode(uint32_t mode) 5875 { 5876 Mutex::Autolock _l(mLock); 5877 size_t size = mEffectChains.size(); 5878 for (size_t i = 0; i < size; i++) { 5879 mEffectChains[i]->setMode_l(mode); 5880 } 5881 } 5882 5883 void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5884 const wp<EffectHandle>& handle, 5885 bool unpiniflast) { 5886 5887 Mutex::Autolock _l(mLock); 5888 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5889 // delete the effect module if removing last handle on it 5890 if (effect->removeHandle(handle) == 0) { 5891 if (!effect->isPinned() || unpiniflast) { 5892 removeEffect_l(effect); 5893 AudioSystem::unregisterEffect(effect->id()); 5894 } 5895 } 5896 } 5897 5898 status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5899 { 5900 int session = chain->sessionId(); 5901 int16_t *buffer = mMixBuffer; 5902 bool ownsBuffer = false; 5903 5904 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5905 if (session > 0) { 5906 // Only one effect chain can be present in direct output thread and it uses 5907 // the mix buffer as input 5908 if (mType != DIRECT) { 5909 size_t numSamples = mFrameCount * mChannelCount; 5910 buffer = new int16_t[numSamples]; 5911 memset(buffer, 0, numSamples * sizeof(int16_t)); 5912 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5913 ownsBuffer = true; 5914 } 5915 5916 // Attach all tracks with same session ID to this chain. 5917 for (size_t i = 0; i < mTracks.size(); ++i) { 5918 sp<Track> track = mTracks[i]; 5919 if (session == track->sessionId()) { 5920 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5921 track->setMainBuffer(buffer); 5922 chain->incTrackCnt(); 5923 } 5924 } 5925 5926 // indicate all active tracks in the chain 5927 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5928 sp<Track> track = mActiveTracks[i].promote(); 5929 if (track == 0) continue; 5930 if (session == track->sessionId()) { 5931 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5932 chain->incActiveTrackCnt(); 5933 } 5934 } 5935 } 5936 5937 chain->setInBuffer(buffer, ownsBuffer); 5938 chain->setOutBuffer(mMixBuffer); 5939 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5940 // chains list in order to be processed last as it contains output stage effects 5941 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5942 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5943 // after track specific effects and before output stage 5944 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5945 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5946 // Effect chain for other sessions are inserted at beginning of effect 5947 // chains list to be processed before output mix effects. Relative order between other 5948 // sessions is not important 5949 size_t size = mEffectChains.size(); 5950 size_t i = 0; 5951 for (i = 0; i < size; i++) { 5952 if (mEffectChains[i]->sessionId() < session) break; 5953 } 5954 mEffectChains.insertAt(chain, i); 5955 checkSuspendOnAddEffectChain_l(chain); 5956 5957 return NO_ERROR; 5958 } 5959 5960 size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5961 { 5962 int session = chain->sessionId(); 5963 5964 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5965 5966 for (size_t i = 0; i < mEffectChains.size(); i++) { 5967 if (chain == mEffectChains[i]) { 5968 mEffectChains.removeAt(i); 5969 // detach all active tracks from the chain 5970 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5971 sp<Track> track = mActiveTracks[i].promote(); 5972 if (track == 0) continue; 5973 if (session == track->sessionId()) { 5974 LOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5975 chain.get(), session); 5976 chain->decActiveTrackCnt(); 5977 } 5978 } 5979 5980 // detach all tracks with same session ID from this chain 5981 for (size_t i = 0; i < mTracks.size(); ++i) { 5982 sp<Track> track = mTracks[i]; 5983 if (session == track->sessionId()) { 5984 track->setMainBuffer(mMixBuffer); 5985 chain->decTrackCnt(); 5986 } 5987 } 5988 break; 5989 } 5990 } 5991 return mEffectChains.size(); 5992 } 5993 5994 status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5995 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5996 { 5997 Mutex::Autolock _l(mLock); 5998 return attachAuxEffect_l(track, EffectId); 5999 } 6000 6001 status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6002 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6003 { 6004 status_t status = NO_ERROR; 6005 6006 if (EffectId == 0) { 6007 track->setAuxBuffer(0, NULL); 6008 } else { 6009 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6010 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6011 if (effect != 0) { 6012 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6013 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6014 } else { 6015 status = INVALID_OPERATION; 6016 } 6017 } else { 6018 status = BAD_VALUE; 6019 } 6020 } 6021 return status; 6022 } 6023 6024 void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6025 { 6026 for (size_t i = 0; i < mTracks.size(); ++i) { 6027 sp<Track> track = mTracks[i]; 6028 if (track->auxEffectId() == effectId) { 6029 attachAuxEffect_l(track, 0); 6030 } 6031 } 6032 } 6033 6034 status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6035 { 6036 // only one chain per input thread 6037 if (mEffectChains.size() != 0) { 6038 return INVALID_OPERATION; 6039 } 6040 LOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6041 6042 chain->setInBuffer(NULL); 6043 chain->setOutBuffer(NULL); 6044 6045 checkSuspendOnAddEffectChain_l(chain); 6046 6047 mEffectChains.add(chain); 6048 6049 return NO_ERROR; 6050 } 6051 6052 size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6053 { 6054 LOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6055 LOGW_IF(mEffectChains.size() != 1, 6056 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6057 chain.get(), mEffectChains.size(), this); 6058 if (mEffectChains.size() == 1) { 6059 mEffectChains.removeAt(0); 6060 } 6061 return 0; 6062 } 6063 6064 // ---------------------------------------------------------------------------- 6065 // EffectModule implementation 6066 // ---------------------------------------------------------------------------- 6067 6068 #undef LOG_TAG 6069 #define LOG_TAG "AudioFlinger::EffectModule" 6070 6071 AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6072 const wp<AudioFlinger::EffectChain>& chain, 6073 effect_descriptor_t *desc, 6074 int id, 6075 int sessionId) 6076 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6077 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6078 { 6079 LOGV("Constructor %p", this); 6080 int lStatus; 6081 sp<ThreadBase> thread = mThread.promote(); 6082 if (thread == 0) { 6083 return; 6084 } 6085 6086 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6087 6088 // create effect engine from effect factory 6089 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6090 6091 if (mStatus != NO_ERROR) { 6092 return; 6093 } 6094 lStatus = init(); 6095 if (lStatus < 0) { 6096 mStatus = lStatus; 6097 goto Error; 6098 } 6099 6100 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6101 mPinned = true; 6102 } 6103 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6104 return; 6105 Error: 6106 EffectRelease(mEffectInterface); 6107 mEffectInterface = NULL; 6108 LOGV("Constructor Error %d", mStatus); 6109 } 6110 6111 AudioFlinger::EffectModule::~EffectModule() 6112 { 6113 LOGV("Destructor %p", this); 6114 if (mEffectInterface != NULL) { 6115 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6116 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6117 sp<ThreadBase> thread = mThread.promote(); 6118 if (thread != 0) { 6119 audio_stream_t *stream = thread->stream(); 6120 if (stream != NULL) { 6121 stream->remove_audio_effect(stream, mEffectInterface); 6122 } 6123 } 6124 } 6125 // release effect engine 6126 EffectRelease(mEffectInterface); 6127 } 6128 } 6129 6130 status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6131 { 6132 status_t status; 6133 6134 Mutex::Autolock _l(mLock); 6135 // First handle in mHandles has highest priority and controls the effect module 6136 int priority = handle->priority(); 6137 size_t size = mHandles.size(); 6138 sp<EffectHandle> h; 6139 size_t i; 6140 for (i = 0; i < size; i++) { 6141 h = mHandles[i].promote(); 6142 if (h == 0) continue; 6143 if (h->priority() <= priority) break; 6144 } 6145 // if inserted in first place, move effect control from previous owner to this handle 6146 if (i == 0) { 6147 bool enabled = false; 6148 if (h != 0) { 6149 enabled = h->enabled(); 6150 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6151 } 6152 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6153 status = NO_ERROR; 6154 } else { 6155 status = ALREADY_EXISTS; 6156 } 6157 LOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6158 mHandles.insertAt(handle, i); 6159 return status; 6160 } 6161 6162 size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6163 { 6164 Mutex::Autolock _l(mLock); 6165 size_t size = mHandles.size(); 6166 size_t i; 6167 for (i = 0; i < size; i++) { 6168 if (mHandles[i] == handle) break; 6169 } 6170 if (i == size) { 6171 return size; 6172 } 6173 LOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6174 6175 bool enabled = false; 6176 EffectHandle *hdl = handle.unsafe_get(); 6177 if (hdl) { 6178 LOGV("removeHandle() unsafe_get OK"); 6179 enabled = hdl->enabled(); 6180 } 6181 mHandles.removeAt(i); 6182 size = mHandles.size(); 6183 // if removed from first place, move effect control from this handle to next in line 6184 if (i == 0 && size != 0) { 6185 sp<EffectHandle> h = mHandles[0].promote(); 6186 if (h != 0) { 6187 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6188 } 6189 } 6190 6191 // Prevent calls to process() and other functions on effect interface from now on. 6192 // The effect engine will be released by the destructor when the last strong reference on 6193 // this object is released which can happen after next process is called. 6194 if (size == 0 && !mPinned) { 6195 mState = DESTROYED; 6196 } 6197 6198 return size; 6199 } 6200 6201 sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6202 { 6203 Mutex::Autolock _l(mLock); 6204 sp<EffectHandle> handle; 6205 if (mHandles.size() != 0) { 6206 handle = mHandles[0].promote(); 6207 } 6208 return handle; 6209 } 6210 6211 void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6212 { 6213 LOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6214 // keep a strong reference on this EffectModule to avoid calling the 6215 // destructor before we exit 6216 sp<EffectModule> keep(this); 6217 { 6218 sp<ThreadBase> thread = mThread.promote(); 6219 if (thread != 0) { 6220 thread->disconnectEffect(keep, handle, unpiniflast); 6221 } 6222 } 6223 } 6224 6225 void AudioFlinger::EffectModule::updateState() { 6226 Mutex::Autolock _l(mLock); 6227 6228 switch (mState) { 6229 case RESTART: 6230 reset_l(); 6231 // FALL THROUGH 6232 6233 case STARTING: 6234 // clear auxiliary effect input buffer for next accumulation 6235 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6236 memset(mConfig.inputCfg.buffer.raw, 6237 0, 6238 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6239 } 6240 start_l(); 6241 mState = ACTIVE; 6242 break; 6243 case STOPPING: 6244 stop_l(); 6245 mDisableWaitCnt = mMaxDisableWaitCnt; 6246 mState = STOPPED; 6247 break; 6248 case STOPPED: 6249 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6250 // turn off sequence. 6251 if (--mDisableWaitCnt == 0) { 6252 reset_l(); 6253 mState = IDLE; 6254 } 6255 break; 6256 default: //IDLE , ACTIVE, DESTROYED 6257 break; 6258 } 6259 } 6260 6261 void AudioFlinger::EffectModule::process() 6262 { 6263 Mutex::Autolock _l(mLock); 6264 6265 if (mState == DESTROYED || mEffectInterface == NULL || 6266 mConfig.inputCfg.buffer.raw == NULL || 6267 mConfig.outputCfg.buffer.raw == NULL) { 6268 return; 6269 } 6270 6271 if (isProcessEnabled()) { 6272 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6273 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6274 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 6275 mConfig.inputCfg.buffer.s32, 6276 mConfig.inputCfg.buffer.frameCount/2); 6277 } 6278 6279 // do the actual processing in the effect engine 6280 int ret = (*mEffectInterface)->process(mEffectInterface, 6281 &mConfig.inputCfg.buffer, 6282 &mConfig.outputCfg.buffer); 6283 6284 // force transition to IDLE state when engine is ready 6285 if (mState == STOPPED && ret == -ENODATA) { 6286 mDisableWaitCnt = 1; 6287 } 6288 6289 // clear auxiliary effect input buffer for next accumulation 6290 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6291 memset(mConfig.inputCfg.buffer.raw, 0, 6292 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6293 } 6294 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6295 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6296 // If an insert effect is idle and input buffer is different from output buffer, 6297 // accumulate input onto output 6298 sp<EffectChain> chain = mChain.promote(); 6299 if (chain != 0 && chain->activeTrackCnt() != 0) { 6300 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6301 int16_t *in = mConfig.inputCfg.buffer.s16; 6302 int16_t *out = mConfig.outputCfg.buffer.s16; 6303 for (size_t i = 0; i < frameCnt; i++) { 6304 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6305 } 6306 } 6307 } 6308 } 6309 6310 void AudioFlinger::EffectModule::reset_l() 6311 { 6312 if (mEffectInterface == NULL) { 6313 return; 6314 } 6315 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6316 } 6317 6318 status_t AudioFlinger::EffectModule::configure() 6319 { 6320 uint32_t channels; 6321 if (mEffectInterface == NULL) { 6322 return NO_INIT; 6323 } 6324 6325 sp<ThreadBase> thread = mThread.promote(); 6326 if (thread == 0) { 6327 return DEAD_OBJECT; 6328 } 6329 6330 // TODO: handle configuration of effects replacing track process 6331 if (thread->channelCount() == 1) { 6332 channels = AUDIO_CHANNEL_OUT_MONO; 6333 } else { 6334 channels = AUDIO_CHANNEL_OUT_STEREO; 6335 } 6336 6337 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6338 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6339 } else { 6340 mConfig.inputCfg.channels = channels; 6341 } 6342 mConfig.outputCfg.channels = channels; 6343 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6344 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6345 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6346 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6347 mConfig.inputCfg.bufferProvider.cookie = NULL; 6348 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6349 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6350 mConfig.outputCfg.bufferProvider.cookie = NULL; 6351 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6352 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6353 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6354 // Insert effect: 6355 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6356 // always overwrites output buffer: input buffer == output buffer 6357 // - in other sessions: 6358 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6359 // other effect: overwrites output buffer: input buffer == output buffer 6360 // Auxiliary effect: 6361 // accumulates in output buffer: input buffer != output buffer 6362 // Therefore: accumulate <=> input buffer != output buffer 6363 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6364 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6365 } else { 6366 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6367 } 6368 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6369 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6370 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6371 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6372 6373 LOGV("configure() %p thread %p buffer %p framecount %d", 6374 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6375 6376 status_t cmdStatus; 6377 uint32_t size = sizeof(int); 6378 status_t status = (*mEffectInterface)->command(mEffectInterface, 6379 EFFECT_CMD_CONFIGURE, 6380 sizeof(effect_config_t), 6381 &mConfig, 6382 &size, 6383 &cmdStatus); 6384 if (status == 0) { 6385 status = cmdStatus; 6386 } 6387 6388 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6389 (1000 * mConfig.outputCfg.buffer.frameCount); 6390 6391 return status; 6392 } 6393 6394 status_t AudioFlinger::EffectModule::init() 6395 { 6396 Mutex::Autolock _l(mLock); 6397 if (mEffectInterface == NULL) { 6398 return NO_INIT; 6399 } 6400 status_t cmdStatus; 6401 uint32_t size = sizeof(status_t); 6402 status_t status = (*mEffectInterface)->command(mEffectInterface, 6403 EFFECT_CMD_INIT, 6404 0, 6405 NULL, 6406 &size, 6407 &cmdStatus); 6408 if (status == 0) { 6409 status = cmdStatus; 6410 } 6411 return status; 6412 } 6413 6414 status_t AudioFlinger::EffectModule::start() 6415 { 6416 Mutex::Autolock _l(mLock); 6417 return start_l(); 6418 } 6419 6420 status_t AudioFlinger::EffectModule::start_l() 6421 { 6422 if (mEffectInterface == NULL) { 6423 return NO_INIT; 6424 } 6425 status_t cmdStatus; 6426 uint32_t size = sizeof(status_t); 6427 status_t status = (*mEffectInterface)->command(mEffectInterface, 6428 EFFECT_CMD_ENABLE, 6429 0, 6430 NULL, 6431 &size, 6432 &cmdStatus); 6433 if (status == 0) { 6434 status = cmdStatus; 6435 } 6436 if (status == 0 && 6437 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6438 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6439 sp<ThreadBase> thread = mThread.promote(); 6440 if (thread != 0) { 6441 audio_stream_t *stream = thread->stream(); 6442 if (stream != NULL) { 6443 stream->add_audio_effect(stream, mEffectInterface); 6444 } 6445 } 6446 } 6447 return status; 6448 } 6449 6450 status_t AudioFlinger::EffectModule::stop() 6451 { 6452 Mutex::Autolock _l(mLock); 6453 return stop_l(); 6454 } 6455 6456 status_t AudioFlinger::EffectModule::stop_l() 6457 { 6458 if (mEffectInterface == NULL) { 6459 return NO_INIT; 6460 } 6461 status_t cmdStatus; 6462 uint32_t size = sizeof(status_t); 6463 status_t status = (*mEffectInterface)->command(mEffectInterface, 6464 EFFECT_CMD_DISABLE, 6465 0, 6466 NULL, 6467 &size, 6468 &cmdStatus); 6469 if (status == 0) { 6470 status = cmdStatus; 6471 } 6472 if (status == 0 && 6473 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6474 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6475 sp<ThreadBase> thread = mThread.promote(); 6476 if (thread != 0) { 6477 audio_stream_t *stream = thread->stream(); 6478 if (stream != NULL) { 6479 stream->remove_audio_effect(stream, mEffectInterface); 6480 } 6481 } 6482 } 6483 return status; 6484 } 6485 6486 status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6487 uint32_t cmdSize, 6488 void *pCmdData, 6489 uint32_t *replySize, 6490 void *pReplyData) 6491 { 6492 Mutex::Autolock _l(mLock); 6493 // LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6494 6495 if (mState == DESTROYED || mEffectInterface == NULL) { 6496 return NO_INIT; 6497 } 6498 status_t status = (*mEffectInterface)->command(mEffectInterface, 6499 cmdCode, 6500 cmdSize, 6501 pCmdData, 6502 replySize, 6503 pReplyData); 6504 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6505 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6506 for (size_t i = 1; i < mHandles.size(); i++) { 6507 sp<EffectHandle> h = mHandles[i].promote(); 6508 if (h != 0) { 6509 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6510 } 6511 } 6512 } 6513 return status; 6514 } 6515 6516 status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6517 { 6518 6519 Mutex::Autolock _l(mLock); 6520 LOGV("setEnabled %p enabled %d", this, enabled); 6521 6522 if (enabled != isEnabled()) { 6523 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6524 if (enabled && status != NO_ERROR) { 6525 return status; 6526 } 6527 6528 switch (mState) { 6529 // going from disabled to enabled 6530 case IDLE: 6531 mState = STARTING; 6532 break; 6533 case STOPPED: 6534 mState = RESTART; 6535 break; 6536 case STOPPING: 6537 mState = ACTIVE; 6538 break; 6539 6540 // going from enabled to disabled 6541 case RESTART: 6542 mState = STOPPED; 6543 break; 6544 case STARTING: 6545 mState = IDLE; 6546 break; 6547 case ACTIVE: 6548 mState = STOPPING; 6549 break; 6550 case DESTROYED: 6551 return NO_ERROR; // simply ignore as we are being destroyed 6552 } 6553 for (size_t i = 1; i < mHandles.size(); i++) { 6554 sp<EffectHandle> h = mHandles[i].promote(); 6555 if (h != 0) { 6556 h->setEnabled(enabled); 6557 } 6558 } 6559 } 6560 return NO_ERROR; 6561 } 6562 6563 bool AudioFlinger::EffectModule::isEnabled() 6564 { 6565 switch (mState) { 6566 case RESTART: 6567 case STARTING: 6568 case ACTIVE: 6569 return true; 6570 case IDLE: 6571 case STOPPING: 6572 case STOPPED: 6573 case DESTROYED: 6574 default: 6575 return false; 6576 } 6577 } 6578 6579 bool AudioFlinger::EffectModule::isProcessEnabled() 6580 { 6581 switch (mState) { 6582 case RESTART: 6583 case ACTIVE: 6584 case STOPPING: 6585 case STOPPED: 6586 return true; 6587 case IDLE: 6588 case STARTING: 6589 case DESTROYED: 6590 default: 6591 return false; 6592 } 6593 } 6594 6595 status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6596 { 6597 Mutex::Autolock _l(mLock); 6598 status_t status = NO_ERROR; 6599 6600 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6601 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6602 if (isProcessEnabled() && 6603 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6604 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6605 status_t cmdStatus; 6606 uint32_t volume[2]; 6607 uint32_t *pVolume = NULL; 6608 uint32_t size = sizeof(volume); 6609 volume[0] = *left; 6610 volume[1] = *right; 6611 if (controller) { 6612 pVolume = volume; 6613 } 6614 status = (*mEffectInterface)->command(mEffectInterface, 6615 EFFECT_CMD_SET_VOLUME, 6616 size, 6617 volume, 6618 &size, 6619 pVolume); 6620 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6621 *left = volume[0]; 6622 *right = volume[1]; 6623 } 6624 } 6625 return status; 6626 } 6627 6628 status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6629 { 6630 Mutex::Autolock _l(mLock); 6631 status_t status = NO_ERROR; 6632 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6633 // audio pre processing modules on RecordThread can receive both output and 6634 // input device indication in the same call 6635 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6636 if (dev) { 6637 status_t cmdStatus; 6638 uint32_t size = sizeof(status_t); 6639 6640 status = (*mEffectInterface)->command(mEffectInterface, 6641 EFFECT_CMD_SET_DEVICE, 6642 sizeof(uint32_t), 6643 &dev, 6644 &size, 6645 &cmdStatus); 6646 if (status == NO_ERROR) { 6647 status = cmdStatus; 6648 } 6649 } 6650 dev = device & AUDIO_DEVICE_IN_ALL; 6651 if (dev) { 6652 status_t cmdStatus; 6653 uint32_t size = sizeof(status_t); 6654 6655 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6656 EFFECT_CMD_SET_INPUT_DEVICE, 6657 sizeof(uint32_t), 6658 &dev, 6659 &size, 6660 &cmdStatus); 6661 if (status2 == NO_ERROR) { 6662 status2 = cmdStatus; 6663 } 6664 if (status == NO_ERROR) { 6665 status = status2; 6666 } 6667 } 6668 } 6669 return status; 6670 } 6671 6672 status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 6673 { 6674 Mutex::Autolock _l(mLock); 6675 status_t status = NO_ERROR; 6676 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6677 status_t cmdStatus; 6678 uint32_t size = sizeof(status_t); 6679 status = (*mEffectInterface)->command(mEffectInterface, 6680 EFFECT_CMD_SET_AUDIO_MODE, 6681 sizeof(int), 6682 &mode, 6683 &size, 6684 &cmdStatus); 6685 if (status == NO_ERROR) { 6686 status = cmdStatus; 6687 } 6688 } 6689 return status; 6690 } 6691 6692 void AudioFlinger::EffectModule::setSuspended(bool suspended) 6693 { 6694 Mutex::Autolock _l(mLock); 6695 mSuspended = suspended; 6696 } 6697 bool AudioFlinger::EffectModule::suspended() 6698 { 6699 Mutex::Autolock _l(mLock); 6700 return mSuspended; 6701 } 6702 6703 status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6704 { 6705 const size_t SIZE = 256; 6706 char buffer[SIZE]; 6707 String8 result; 6708 6709 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6710 result.append(buffer); 6711 6712 bool locked = tryLock(mLock); 6713 // failed to lock - AudioFlinger is probably deadlocked 6714 if (!locked) { 6715 result.append("\t\tCould not lock Fx mutex:\n"); 6716 } 6717 6718 result.append("\t\tSession Status State Engine:\n"); 6719 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6720 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6721 result.append(buffer); 6722 6723 result.append("\t\tDescriptor:\n"); 6724 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6725 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6726 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6727 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6728 result.append(buffer); 6729 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6730 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6731 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6732 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6733 result.append(buffer); 6734 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6735 mDescriptor.apiVersion, 6736 mDescriptor.flags); 6737 result.append(buffer); 6738 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6739 mDescriptor.name); 6740 result.append(buffer); 6741 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6742 mDescriptor.implementor); 6743 result.append(buffer); 6744 6745 result.append("\t\t- Input configuration:\n"); 6746 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6747 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6748 (uint32_t)mConfig.inputCfg.buffer.raw, 6749 mConfig.inputCfg.buffer.frameCount, 6750 mConfig.inputCfg.samplingRate, 6751 mConfig.inputCfg.channels, 6752 mConfig.inputCfg.format); 6753 result.append(buffer); 6754 6755 result.append("\t\t- Output configuration:\n"); 6756 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6757 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6758 (uint32_t)mConfig.outputCfg.buffer.raw, 6759 mConfig.outputCfg.buffer.frameCount, 6760 mConfig.outputCfg.samplingRate, 6761 mConfig.outputCfg.channels, 6762 mConfig.outputCfg.format); 6763 result.append(buffer); 6764 6765 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6766 result.append(buffer); 6767 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6768 for (size_t i = 0; i < mHandles.size(); ++i) { 6769 sp<EffectHandle> handle = mHandles[i].promote(); 6770 if (handle != 0) { 6771 handle->dump(buffer, SIZE); 6772 result.append(buffer); 6773 } 6774 } 6775 6776 result.append("\n"); 6777 6778 write(fd, result.string(), result.length()); 6779 6780 if (locked) { 6781 mLock.unlock(); 6782 } 6783 6784 return NO_ERROR; 6785 } 6786 6787 // ---------------------------------------------------------------------------- 6788 // EffectHandle implementation 6789 // ---------------------------------------------------------------------------- 6790 6791 #undef LOG_TAG 6792 #define LOG_TAG "AudioFlinger::EffectHandle" 6793 6794 AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6795 const sp<AudioFlinger::Client>& client, 6796 const sp<IEffectClient>& effectClient, 6797 int32_t priority) 6798 : BnEffect(), 6799 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6800 mPriority(priority), mHasControl(false), mEnabled(false) 6801 { 6802 LOGV("constructor %p", this); 6803 6804 if (client == 0) { 6805 return; 6806 } 6807 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6808 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6809 if (mCblkMemory != 0) { 6810 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6811 6812 if (mCblk) { 6813 new(mCblk) effect_param_cblk_t(); 6814 mBuffer = (uint8_t *)mCblk + bufOffset; 6815 } 6816 } else { 6817 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6818 return; 6819 } 6820 } 6821 6822 AudioFlinger::EffectHandle::~EffectHandle() 6823 { 6824 LOGV("Destructor %p", this); 6825 disconnect(false); 6826 LOGV("Destructor DONE %p", this); 6827 } 6828 6829 status_t AudioFlinger::EffectHandle::enable() 6830 { 6831 LOGV("enable %p", this); 6832 if (!mHasControl) return INVALID_OPERATION; 6833 if (mEffect == 0) return DEAD_OBJECT; 6834 6835 if (mEnabled) { 6836 return NO_ERROR; 6837 } 6838 6839 mEnabled = true; 6840 6841 sp<ThreadBase> thread = mEffect->thread().promote(); 6842 if (thread != 0) { 6843 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6844 } 6845 6846 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6847 if (mEffect->suspended()) { 6848 return NO_ERROR; 6849 } 6850 6851 status_t status = mEffect->setEnabled(true); 6852 if (status != NO_ERROR) { 6853 if (thread != 0) { 6854 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6855 } 6856 mEnabled = false; 6857 } 6858 return status; 6859 } 6860 6861 status_t AudioFlinger::EffectHandle::disable() 6862 { 6863 LOGV("disable %p", this); 6864 if (!mHasControl) return INVALID_OPERATION; 6865 if (mEffect == 0) return DEAD_OBJECT; 6866 6867 if (!mEnabled) { 6868 return NO_ERROR; 6869 } 6870 mEnabled = false; 6871 6872 if (mEffect->suspended()) { 6873 return NO_ERROR; 6874 } 6875 6876 status_t status = mEffect->setEnabled(false); 6877 6878 sp<ThreadBase> thread = mEffect->thread().promote(); 6879 if (thread != 0) { 6880 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6881 } 6882 6883 return status; 6884 } 6885 6886 void AudioFlinger::EffectHandle::disconnect() 6887 { 6888 disconnect(true); 6889 } 6890 6891 void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6892 { 6893 LOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6894 if (mEffect == 0) { 6895 return; 6896 } 6897 mEffect->disconnect(this, unpiniflast); 6898 6899 if (mHasControl && mEnabled) { 6900 sp<ThreadBase> thread = mEffect->thread().promote(); 6901 if (thread != 0) { 6902 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6903 } 6904 } 6905 6906 // release sp on module => module destructor can be called now 6907 mEffect.clear(); 6908 if (mClient != 0) { 6909 if (mCblk) { 6910 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6911 } 6912 mCblkMemory.clear(); // and free the shared memory 6913 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6914 mClient.clear(); 6915 } 6916 } 6917 6918 status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6919 uint32_t cmdSize, 6920 void *pCmdData, 6921 uint32_t *replySize, 6922 void *pReplyData) 6923 { 6924 // LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6925 // cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6926 6927 // only get parameter command is permitted for applications not controlling the effect 6928 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6929 return INVALID_OPERATION; 6930 } 6931 if (mEffect == 0) return DEAD_OBJECT; 6932 if (mClient == 0) return INVALID_OPERATION; 6933 6934 // handle commands that are not forwarded transparently to effect engine 6935 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6936 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6937 // no risk to block the whole media server process or mixer threads is we are stuck here 6938 Mutex::Autolock _l(mCblk->lock); 6939 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6940 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6941 mCblk->serverIndex = 0; 6942 mCblk->clientIndex = 0; 6943 return BAD_VALUE; 6944 } 6945 status_t status = NO_ERROR; 6946 while (mCblk->serverIndex < mCblk->clientIndex) { 6947 int reply; 6948 uint32_t rsize = sizeof(int); 6949 int *p = (int *)(mBuffer + mCblk->serverIndex); 6950 int size = *p++; 6951 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6952 LOGW("command(): invalid parameter block size"); 6953 break; 6954 } 6955 effect_param_t *param = (effect_param_t *)p; 6956 if (param->psize == 0 || param->vsize == 0) { 6957 LOGW("command(): null parameter or value size"); 6958 mCblk->serverIndex += size; 6959 continue; 6960 } 6961 uint32_t psize = sizeof(effect_param_t) + 6962 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6963 param->vsize; 6964 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6965 psize, 6966 p, 6967 &rsize, 6968 &reply); 6969 // stop at first error encountered 6970 if (ret != NO_ERROR) { 6971 status = ret; 6972 *(int *)pReplyData = reply; 6973 break; 6974 } else if (reply != NO_ERROR) { 6975 *(int *)pReplyData = reply; 6976 break; 6977 } 6978 mCblk->serverIndex += size; 6979 } 6980 mCblk->serverIndex = 0; 6981 mCblk->clientIndex = 0; 6982 return status; 6983 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6984 *(int *)pReplyData = NO_ERROR; 6985 return enable(); 6986 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6987 *(int *)pReplyData = NO_ERROR; 6988 return disable(); 6989 } 6990 6991 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6992 } 6993 6994 sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6995 return mCblkMemory; 6996 } 6997 6998 void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 6999 { 7000 LOGV("setControl %p control %d", this, hasControl); 7001 7002 mHasControl = hasControl; 7003 mEnabled = enabled; 7004 7005 if (signal && mEffectClient != 0) { 7006 mEffectClient->controlStatusChanged(hasControl); 7007 } 7008 } 7009 7010 void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7011 uint32_t cmdSize, 7012 void *pCmdData, 7013 uint32_t replySize, 7014 void *pReplyData) 7015 { 7016 if (mEffectClient != 0) { 7017 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7018 } 7019 } 7020 7021 7022 7023 void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7024 { 7025 if (mEffectClient != 0) { 7026 mEffectClient->enableStatusChanged(enabled); 7027 } 7028 } 7029 7030 status_t AudioFlinger::EffectHandle::onTransact( 7031 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7032 { 7033 return BnEffect::onTransact(code, data, reply, flags); 7034 } 7035 7036 7037 void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7038 { 7039 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7040 7041 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7042 (mClient == NULL) ? getpid() : mClient->pid(), 7043 mPriority, 7044 mHasControl, 7045 !locked, 7046 mCblk ? mCblk->clientIndex : 0, 7047 mCblk ? mCblk->serverIndex : 0 7048 ); 7049 7050 if (locked) { 7051 mCblk->lock.unlock(); 7052 } 7053 } 7054 7055 #undef LOG_TAG 7056 #define LOG_TAG "AudioFlinger::EffectChain" 7057 7058 AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7059 int sessionId) 7060 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7061 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7062 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7063 { 7064 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7065 sp<ThreadBase> thread = mThread.promote(); 7066 if (thread == 0) { 7067 return; 7068 } 7069 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7070 thread->frameCount(); 7071 } 7072 7073 AudioFlinger::EffectChain::~EffectChain() 7074 { 7075 if (mOwnInBuffer) { 7076 delete mInBuffer; 7077 } 7078 7079 } 7080 7081 // getEffectFromDesc_l() must be called with ThreadBase::mLock held 7082 sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7083 { 7084 sp<EffectModule> effect; 7085 size_t size = mEffects.size(); 7086 7087 for (size_t i = 0; i < size; i++) { 7088 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7089 effect = mEffects[i]; 7090 break; 7091 } 7092 } 7093 return effect; 7094 } 7095 7096 // getEffectFromId_l() must be called with ThreadBase::mLock held 7097 sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7098 { 7099 sp<EffectModule> effect; 7100 size_t size = mEffects.size(); 7101 7102 for (size_t i = 0; i < size; i++) { 7103 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7104 if (id == 0 || mEffects[i]->id() == id) { 7105 effect = mEffects[i]; 7106 break; 7107 } 7108 } 7109 return effect; 7110 } 7111 7112 // getEffectFromType_l() must be called with ThreadBase::mLock held 7113 sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7114 const effect_uuid_t *type) 7115 { 7116 sp<EffectModule> effect; 7117 size_t size = mEffects.size(); 7118 7119 for (size_t i = 0; i < size; i++) { 7120 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7121 effect = mEffects[i]; 7122 break; 7123 } 7124 } 7125 return effect; 7126 } 7127 7128 // Must be called with EffectChain::mLock locked 7129 void AudioFlinger::EffectChain::process_l() 7130 { 7131 sp<ThreadBase> thread = mThread.promote(); 7132 if (thread == 0) { 7133 LOGW("process_l(): cannot promote mixer thread"); 7134 return; 7135 } 7136 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7137 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7138 // always process effects unless no more tracks are on the session and the effect tail 7139 // has been rendered 7140 bool doProcess = true; 7141 if (!isGlobalSession) { 7142 bool tracksOnSession = (trackCnt() != 0); 7143 7144 if (!tracksOnSession && mTailBufferCount == 0) { 7145 doProcess = false; 7146 } 7147 7148 if (activeTrackCnt() == 0) { 7149 // if no track is active and the effect tail has not been rendered, 7150 // the input buffer must be cleared here as the mixer process will not do it 7151 if (tracksOnSession || mTailBufferCount > 0) { 7152 size_t numSamples = thread->frameCount() * thread->channelCount(); 7153 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7154 if (mTailBufferCount > 0) { 7155 mTailBufferCount--; 7156 } 7157 } 7158 } 7159 } 7160 7161 size_t size = mEffects.size(); 7162 if (doProcess) { 7163 for (size_t i = 0; i < size; i++) { 7164 mEffects[i]->process(); 7165 } 7166 } 7167 for (size_t i = 0; i < size; i++) { 7168 mEffects[i]->updateState(); 7169 } 7170 } 7171 7172 // addEffect_l() must be called with PlaybackThread::mLock held 7173 status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7174 { 7175 effect_descriptor_t desc = effect->desc(); 7176 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7177 7178 Mutex::Autolock _l(mLock); 7179 effect->setChain(this); 7180 sp<ThreadBase> thread = mThread.promote(); 7181 if (thread == 0) { 7182 return NO_INIT; 7183 } 7184 effect->setThread(thread); 7185 7186 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7187 // Auxiliary effects are inserted at the beginning of mEffects vector as 7188 // they are processed first and accumulated in chain input buffer 7189 mEffects.insertAt(effect, 0); 7190 7191 // the input buffer for auxiliary effect contains mono samples in 7192 // 32 bit format. This is to avoid saturation in AudoMixer 7193 // accumulation stage. Saturation is done in EffectModule::process() before 7194 // calling the process in effect engine 7195 size_t numSamples = thread->frameCount(); 7196 int32_t *buffer = new int32_t[numSamples]; 7197 memset(buffer, 0, numSamples * sizeof(int32_t)); 7198 effect->setInBuffer((int16_t *)buffer); 7199 // auxiliary effects output samples to chain input buffer for further processing 7200 // by insert effects 7201 effect->setOutBuffer(mInBuffer); 7202 } else { 7203 // Insert effects are inserted at the end of mEffects vector as they are processed 7204 // after track and auxiliary effects. 7205 // Insert effect order as a function of indicated preference: 7206 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7207 // another effect is present 7208 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7209 // last effect claiming first position 7210 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7211 // first effect claiming last position 7212 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7213 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7214 // already present 7215 7216 int size = (int)mEffects.size(); 7217 int idx_insert = size; 7218 int idx_insert_first = -1; 7219 int idx_insert_last = -1; 7220 7221 for (int i = 0; i < size; i++) { 7222 effect_descriptor_t d = mEffects[i]->desc(); 7223 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7224 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7225 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7226 // check invalid effect chaining combinations 7227 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7228 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7229 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7230 return INVALID_OPERATION; 7231 } 7232 // remember position of first insert effect and by default 7233 // select this as insert position for new effect 7234 if (idx_insert == size) { 7235 idx_insert = i; 7236 } 7237 // remember position of last insert effect claiming 7238 // first position 7239 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7240 idx_insert_first = i; 7241 } 7242 // remember position of first insert effect claiming 7243 // last position 7244 if (iPref == EFFECT_FLAG_INSERT_LAST && 7245 idx_insert_last == -1) { 7246 idx_insert_last = i; 7247 } 7248 } 7249 } 7250 7251 // modify idx_insert from first position if needed 7252 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7253 if (idx_insert_last != -1) { 7254 idx_insert = idx_insert_last; 7255 } else { 7256 idx_insert = size; 7257 } 7258 } else { 7259 if (idx_insert_first != -1) { 7260 idx_insert = idx_insert_first + 1; 7261 } 7262 } 7263 7264 // always read samples from chain input buffer 7265 effect->setInBuffer(mInBuffer); 7266 7267 // if last effect in the chain, output samples to chain 7268 // output buffer, otherwise to chain input buffer 7269 if (idx_insert == size) { 7270 if (idx_insert != 0) { 7271 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7272 mEffects[idx_insert-1]->configure(); 7273 } 7274 effect->setOutBuffer(mOutBuffer); 7275 } else { 7276 effect->setOutBuffer(mInBuffer); 7277 } 7278 mEffects.insertAt(effect, idx_insert); 7279 7280 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7281 } 7282 effect->configure(); 7283 return NO_ERROR; 7284 } 7285 7286 // removeEffect_l() must be called with PlaybackThread::mLock held 7287 size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7288 { 7289 Mutex::Autolock _l(mLock); 7290 int size = (int)mEffects.size(); 7291 int i; 7292 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7293 7294 for (i = 0; i < size; i++) { 7295 if (effect == mEffects[i]) { 7296 // calling stop here will remove pre-processing effect from the audio HAL. 7297 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7298 // the middle of a read from audio HAL 7299 if (mEffects[i]->state() == EffectModule::ACTIVE || 7300 mEffects[i]->state() == EffectModule::STOPPING) { 7301 mEffects[i]->stop(); 7302 } 7303 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7304 delete[] effect->inBuffer(); 7305 } else { 7306 if (i == size - 1 && i != 0) { 7307 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7308 mEffects[i - 1]->configure(); 7309 } 7310 } 7311 mEffects.removeAt(i); 7312 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7313 break; 7314 } 7315 } 7316 7317 return mEffects.size(); 7318 } 7319 7320 // setDevice_l() must be called with PlaybackThread::mLock held 7321 void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7322 { 7323 size_t size = mEffects.size(); 7324 for (size_t i = 0; i < size; i++) { 7325 mEffects[i]->setDevice(device); 7326 } 7327 } 7328 7329 // setMode_l() must be called with PlaybackThread::mLock held 7330 void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 7331 { 7332 size_t size = mEffects.size(); 7333 for (size_t i = 0; i < size; i++) { 7334 mEffects[i]->setMode(mode); 7335 } 7336 } 7337 7338 // setVolume_l() must be called with PlaybackThread::mLock held 7339 bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7340 { 7341 uint32_t newLeft = *left; 7342 uint32_t newRight = *right; 7343 bool hasControl = false; 7344 int ctrlIdx = -1; 7345 size_t size = mEffects.size(); 7346 7347 // first update volume controller 7348 for (size_t i = size; i > 0; i--) { 7349 if (mEffects[i - 1]->isProcessEnabled() && 7350 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7351 ctrlIdx = i - 1; 7352 hasControl = true; 7353 break; 7354 } 7355 } 7356 7357 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7358 if (hasControl) { 7359 *left = mNewLeftVolume; 7360 *right = mNewRightVolume; 7361 } 7362 return hasControl; 7363 } 7364 7365 mVolumeCtrlIdx = ctrlIdx; 7366 mLeftVolume = newLeft; 7367 mRightVolume = newRight; 7368 7369 // second get volume update from volume controller 7370 if (ctrlIdx >= 0) { 7371 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7372 mNewLeftVolume = newLeft; 7373 mNewRightVolume = newRight; 7374 } 7375 // then indicate volume to all other effects in chain. 7376 // Pass altered volume to effects before volume controller 7377 // and requested volume to effects after controller 7378 uint32_t lVol = newLeft; 7379 uint32_t rVol = newRight; 7380 7381 for (size_t i = 0; i < size; i++) { 7382 if ((int)i == ctrlIdx) continue; 7383 // this also works for ctrlIdx == -1 when there is no volume controller 7384 if ((int)i > ctrlIdx) { 7385 lVol = *left; 7386 rVol = *right; 7387 } 7388 mEffects[i]->setVolume(&lVol, &rVol, false); 7389 } 7390 *left = newLeft; 7391 *right = newRight; 7392 7393 return hasControl; 7394 } 7395 7396 status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7397 { 7398 const size_t SIZE = 256; 7399 char buffer[SIZE]; 7400 String8 result; 7401 7402 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7403 result.append(buffer); 7404 7405 bool locked = tryLock(mLock); 7406 // failed to lock - AudioFlinger is probably deadlocked 7407 if (!locked) { 7408 result.append("\tCould not lock mutex:\n"); 7409 } 7410 7411 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7412 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7413 mEffects.size(), 7414 (uint32_t)mInBuffer, 7415 (uint32_t)mOutBuffer, 7416 mActiveTrackCnt); 7417 result.append(buffer); 7418 write(fd, result.string(), result.size()); 7419 7420 for (size_t i = 0; i < mEffects.size(); ++i) { 7421 sp<EffectModule> effect = mEffects[i]; 7422 if (effect != 0) { 7423 effect->dump(fd, args); 7424 } 7425 } 7426 7427 if (locked) { 7428 mLock.unlock(); 7429 } 7430 7431 return NO_ERROR; 7432 } 7433 7434 // must be called with ThreadBase::mLock held 7435 void AudioFlinger::EffectChain::setEffectSuspended_l( 7436 const effect_uuid_t *type, bool suspend) 7437 { 7438 sp<SuspendedEffectDesc> desc; 7439 // use effect type UUID timelow as key as there is no real risk of identical 7440 // timeLow fields among effect type UUIDs. 7441 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7442 if (suspend) { 7443 if (index >= 0) { 7444 desc = mSuspendedEffects.valueAt(index); 7445 } else { 7446 desc = new SuspendedEffectDesc(); 7447 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7448 mSuspendedEffects.add(type->timeLow, desc); 7449 LOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7450 } 7451 if (desc->mRefCount++ == 0) { 7452 sp<EffectModule> effect = getEffectIfEnabled(type); 7453 if (effect != 0) { 7454 desc->mEffect = effect; 7455 effect->setSuspended(true); 7456 effect->setEnabled(false); 7457 } 7458 } 7459 } else { 7460 if (index < 0) { 7461 return; 7462 } 7463 desc = mSuspendedEffects.valueAt(index); 7464 if (desc->mRefCount <= 0) { 7465 LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7466 desc->mRefCount = 1; 7467 } 7468 if (--desc->mRefCount == 0) { 7469 LOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7470 if (desc->mEffect != 0) { 7471 sp<EffectModule> effect = desc->mEffect.promote(); 7472 if (effect != 0) { 7473 effect->setSuspended(false); 7474 sp<EffectHandle> handle = effect->controlHandle(); 7475 if (handle != 0) { 7476 effect->setEnabled(handle->enabled()); 7477 } 7478 } 7479 desc->mEffect.clear(); 7480 } 7481 mSuspendedEffects.removeItemsAt(index); 7482 } 7483 } 7484 } 7485 7486 // must be called with ThreadBase::mLock held 7487 void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7488 { 7489 sp<SuspendedEffectDesc> desc; 7490 7491 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7492 if (suspend) { 7493 if (index >= 0) { 7494 desc = mSuspendedEffects.valueAt(index); 7495 } else { 7496 desc = new SuspendedEffectDesc(); 7497 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7498 LOGV("setEffectSuspendedAll_l() add entry for 0"); 7499 } 7500 if (desc->mRefCount++ == 0) { 7501 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7502 for (size_t i = 0; i < effects.size(); i++) { 7503 setEffectSuspended_l(&effects[i]->desc().type, true); 7504 } 7505 } 7506 } else { 7507 if (index < 0) { 7508 return; 7509 } 7510 desc = mSuspendedEffects.valueAt(index); 7511 if (desc->mRefCount <= 0) { 7512 LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7513 desc->mRefCount = 1; 7514 } 7515 if (--desc->mRefCount == 0) { 7516 Vector<const effect_uuid_t *> types; 7517 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7518 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7519 continue; 7520 } 7521 types.add(&mSuspendedEffects.valueAt(i)->mType); 7522 } 7523 for (size_t i = 0; i < types.size(); i++) { 7524 setEffectSuspended_l(types[i], false); 7525 } 7526 LOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7527 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7528 } 7529 } 7530 } 7531 7532 7533 // The volume effect is used for automated tests only 7534 #ifndef OPENSL_ES_H_ 7535 static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7536 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7537 const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7538 #endif //OPENSL_ES_H_ 7539 7540 bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7541 { 7542 // auxiliary effects and visualizer are never suspended on output mix 7543 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7544 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7545 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7546 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7547 return false; 7548 } 7549 return true; 7550 } 7551 7552 Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7553 { 7554 Vector< sp<EffectModule> > effects; 7555 for (size_t i = 0; i < mEffects.size(); i++) { 7556 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7557 continue; 7558 } 7559 effects.add(mEffects[i]); 7560 } 7561 return effects; 7562 } 7563 7564 sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7565 const effect_uuid_t *type) 7566 { 7567 sp<EffectModule> effect; 7568 effect = getEffectFromType_l(type); 7569 if (effect != 0 && !effect->isEnabled()) { 7570 effect.clear(); 7571 } 7572 return effect; 7573 } 7574 7575 void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7576 bool enabled) 7577 { 7578 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7579 if (enabled) { 7580 if (index < 0) { 7581 // if the effect is not suspend check if all effects are suspended 7582 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7583 if (index < 0) { 7584 return; 7585 } 7586 if (!isEffectEligibleForSuspend(effect->desc())) { 7587 return; 7588 } 7589 setEffectSuspended_l(&effect->desc().type, enabled); 7590 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7591 if (index < 0) { 7592 LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7593 return; 7594 } 7595 } 7596 LOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7597 effect->desc().type.timeLow); 7598 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7599 // if effect is requested to suspended but was not yet enabled, supend it now. 7600 if (desc->mEffect == 0) { 7601 desc->mEffect = effect; 7602 effect->setEnabled(false); 7603 effect->setSuspended(true); 7604 } 7605 } else { 7606 if (index < 0) { 7607 return; 7608 } 7609 LOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7610 effect->desc().type.timeLow); 7611 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7612 desc->mEffect.clear(); 7613 effect->setSuspended(false); 7614 } 7615 } 7616 7617 #undef LOG_TAG 7618 #define LOG_TAG "AudioFlinger" 7619 7620 // ---------------------------------------------------------------------------- 7621 7622 status_t AudioFlinger::onTransact( 7623 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7624 { 7625 return BnAudioFlinger::onTransact(code, data, reply, flags); 7626 } 7627 7628 }; // namespace android 7629