AudioFlinger.cpp revision 60e182437228312cc28469a5b0dfde77ac848e1a
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/IServiceManager.h> 28#include <utils/Log.h> 29#include <binder/Parcel.h> 30#include <binder/IPCThreadState.h> 31#include <utils/String16.h> 32#include <utils/threads.h> 33 34#include <cutils/properties.h> 35 36#include <media/AudioTrack.h> 37#include <media/AudioRecord.h> 38 39#include <private/media/AudioTrackShared.h> 40#include <private/media/AudioEffectShared.h> 41#include <hardware_legacy/AudioHardwareInterface.h> 42 43#include "AudioMixer.h" 44#include "AudioFlinger.h" 45 46#ifdef WITH_A2DP 47#include "A2dpAudioInterface.h" 48#endif 49 50#ifdef LVMX 51#include "lifevibes.h" 52#endif 53 54#include <media/EffectsFactoryApi.h> 55#include <media/EffectVisualizerApi.h> 56 57// ---------------------------------------------------------------------------- 58// the sim build doesn't have gettid 59 60#ifndef HAVE_GETTID 61# define gettid getpid 62#endif 63 64// ---------------------------------------------------------------------------- 65 66extern const char * const gEffectLibPath; 67 68namespace android { 69 70static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 71static const char* kHardwareLockedString = "Hardware lock is taken\n"; 72 73//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 74static const float MAX_GAIN = 4096.0f; 75static const float MAX_GAIN_INT = 0x1000; 76 77// retry counts for buffer fill timeout 78// 50 * ~20msecs = 1 second 79static const int8_t kMaxTrackRetries = 50; 80static const int8_t kMaxTrackStartupRetries = 50; 81// allow less retry attempts on direct output thread. 82// direct outputs can be a scarce resource in audio hardware and should 83// be released as quickly as possible. 84static const int8_t kMaxTrackRetriesDirect = 2; 85 86static const int kDumpLockRetries = 50; 87static const int kDumpLockSleep = 20000; 88 89static const nsecs_t kWarningThrottle = seconds(5); 90 91 92#define AUDIOFLINGER_SECURITY_ENABLED 1 93 94// ---------------------------------------------------------------------------- 95 96static bool recordingAllowed() { 97#ifndef HAVE_ANDROID_OS 98 return true; 99#endif 100#if AUDIOFLINGER_SECURITY_ENABLED 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#else 106 if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO"))) 107 LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest"); 108 return true; 109#endif 110} 111 112static bool settingsAllowed() { 113#ifndef HAVE_ANDROID_OS 114 return true; 115#endif 116#if AUDIOFLINGER_SECURITY_ENABLED 117 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 118 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 119 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 120 return ok; 121#else 122 if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"))) 123 LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest"); 124 return true; 125#endif 126} 127 128// ---------------------------------------------------------------------------- 129 130AudioFlinger::AudioFlinger() 131 : BnAudioFlinger(), 132 mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1) 133{ 134 mHardwareStatus = AUDIO_HW_IDLE; 135 136 mAudioHardware = AudioHardwareInterface::create(); 137 138 mHardwareStatus = AUDIO_HW_INIT; 139 if (mAudioHardware->initCheck() == NO_ERROR) { 140 // open 16-bit output stream for s/w mixer 141 mMode = AudioSystem::MODE_NORMAL; 142 setMode(mMode); 143 144 setMasterVolume(1.0f); 145 setMasterMute(false); 146 } else { 147 LOGE("Couldn't even initialize the stubbed audio hardware!"); 148 } 149#ifdef LVMX 150 LifeVibes::init(); 151 mLifeVibesClientPid = -1; 152#endif 153} 154 155AudioFlinger::~AudioFlinger() 156{ 157 while (!mRecordThreads.isEmpty()) { 158 // closeInput() will remove first entry from mRecordThreads 159 closeInput(mRecordThreads.keyAt(0)); 160 } 161 while (!mPlaybackThreads.isEmpty()) { 162 // closeOutput() will remove first entry from mPlaybackThreads 163 closeOutput(mPlaybackThreads.keyAt(0)); 164 } 165 if (mAudioHardware) { 166 delete mAudioHardware; 167 } 168} 169 170 171 172status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 173{ 174 const size_t SIZE = 256; 175 char buffer[SIZE]; 176 String8 result; 177 178 result.append("Clients:\n"); 179 for (size_t i = 0; i < mClients.size(); ++i) { 180 wp<Client> wClient = mClients.valueAt(i); 181 if (wClient != 0) { 182 sp<Client> client = wClient.promote(); 183 if (client != 0) { 184 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 185 result.append(buffer); 186 } 187 } 188 } 189 write(fd, result.string(), result.size()); 190 return NO_ERROR; 191} 192 193 194status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 195{ 196 const size_t SIZE = 256; 197 char buffer[SIZE]; 198 String8 result; 199 int hardwareStatus = mHardwareStatus; 200 201 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 202 result.append(buffer); 203 write(fd, result.string(), result.size()); 204 return NO_ERROR; 205} 206 207status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 208{ 209 const size_t SIZE = 256; 210 char buffer[SIZE]; 211 String8 result; 212 snprintf(buffer, SIZE, "Permission Denial: " 213 "can't dump AudioFlinger from pid=%d, uid=%d\n", 214 IPCThreadState::self()->getCallingPid(), 215 IPCThreadState::self()->getCallingUid()); 216 result.append(buffer); 217 write(fd, result.string(), result.size()); 218 return NO_ERROR; 219} 220 221static bool tryLock(Mutex& mutex) 222{ 223 bool locked = false; 224 for (int i = 0; i < kDumpLockRetries; ++i) { 225 if (mutex.tryLock() == NO_ERROR) { 226 locked = true; 227 break; 228 } 229 usleep(kDumpLockSleep); 230 } 231 return locked; 232} 233 234status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 235{ 236 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 237 dumpPermissionDenial(fd, args); 238 } else { 239 // get state of hardware lock 240 bool hardwareLocked = tryLock(mHardwareLock); 241 if (!hardwareLocked) { 242 String8 result(kHardwareLockedString); 243 write(fd, result.string(), result.size()); 244 } else { 245 mHardwareLock.unlock(); 246 } 247 248 bool locked = tryLock(mLock); 249 250 // failed to lock - AudioFlinger is probably deadlocked 251 if (!locked) { 252 String8 result(kDeadlockedString); 253 write(fd, result.string(), result.size()); 254 } 255 256 dumpClients(fd, args); 257 dumpInternals(fd, args); 258 259 // dump playback threads 260 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 261 mPlaybackThreads.valueAt(i)->dump(fd, args); 262 } 263 264 // dump record threads 265 for (size_t i = 0; i < mRecordThreads.size(); i++) { 266 mRecordThreads.valueAt(i)->dump(fd, args); 267 } 268 269 if (mAudioHardware) { 270 mAudioHardware->dumpState(fd, args); 271 } 272 if (locked) mLock.unlock(); 273 } 274 return NO_ERROR; 275} 276 277 278// IAudioFlinger interface 279 280 281sp<IAudioTrack> AudioFlinger::createTrack( 282 pid_t pid, 283 int streamType, 284 uint32_t sampleRate, 285 int format, 286 int channelCount, 287 int frameCount, 288 uint32_t flags, 289 const sp<IMemory>& sharedBuffer, 290 int output, 291 int *sessionId, 292 status_t *status) 293{ 294 sp<PlaybackThread::Track> track; 295 sp<TrackHandle> trackHandle; 296 sp<Client> client; 297 wp<Client> wclient; 298 status_t lStatus; 299 int lSessionId; 300 301 if (streamType >= AudioSystem::NUM_STREAM_TYPES) { 302 LOGE("invalid stream type"); 303 lStatus = BAD_VALUE; 304 goto Exit; 305 } 306 307 { 308 Mutex::Autolock _l(mLock); 309 PlaybackThread *thread = checkPlaybackThread_l(output); 310 PlaybackThread *effectThread = NULL; 311 if (thread == NULL) { 312 LOGE("unknown output thread"); 313 lStatus = BAD_VALUE; 314 goto Exit; 315 } 316 317 wclient = mClients.valueFor(pid); 318 319 if (wclient != NULL) { 320 client = wclient.promote(); 321 } else { 322 client = new Client(this, pid); 323 mClients.add(pid, client); 324 } 325 326 LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 327 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 328 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 329 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 330 if (mPlaybackThreads.keyAt(i) != output) { 331 // prevent same audio session on different output threads 332 uint32_t sessions = t->hasAudioSession(*sessionId); 333 if (sessions & PlaybackThread::TRACK_SESSION) { 334 lStatus = BAD_VALUE; 335 goto Exit; 336 } 337 // check if an effect with same session ID is waiting for a track to be created 338 if (sessions & PlaybackThread::EFFECT_SESSION) { 339 effectThread = t.get(); 340 } 341 } 342 } 343 lSessionId = *sessionId; 344 } else { 345 // if no audio session id is provided, create one here 346 lSessionId = nextUniqueId(); 347 if (sessionId != NULL) { 348 *sessionId = lSessionId; 349 } 350 } 351 LOGV("createTrack() lSessionId: %d", lSessionId); 352 353 track = thread->createTrack_l(client, streamType, sampleRate, format, 354 channelCount, frameCount, sharedBuffer, lSessionId, &lStatus); 355 356 // move effect chain to this output thread if an effect on same session was waiting 357 // for a track to be created 358 if (lStatus == NO_ERROR && effectThread != NULL) { 359 Mutex::Autolock _dl(thread->mLock); 360 Mutex::Autolock _sl(effectThread->mLock); 361 moveEffectChain_l(lSessionId, effectThread, thread, true); 362 } 363 } 364 if (lStatus == NO_ERROR) { 365 trackHandle = new TrackHandle(track); 366 } else { 367 // remove local strong reference to Client before deleting the Track so that the Client 368 // destructor is called by the TrackBase destructor with mLock held 369 client.clear(); 370 track.clear(); 371 } 372 373Exit: 374 if(status) { 375 *status = lStatus; 376 } 377 return trackHandle; 378} 379 380uint32_t AudioFlinger::sampleRate(int output) const 381{ 382 Mutex::Autolock _l(mLock); 383 PlaybackThread *thread = checkPlaybackThread_l(output); 384 if (thread == NULL) { 385 LOGW("sampleRate() unknown thread %d", output); 386 return 0; 387 } 388 return thread->sampleRate(); 389} 390 391int AudioFlinger::channelCount(int output) const 392{ 393 Mutex::Autolock _l(mLock); 394 PlaybackThread *thread = checkPlaybackThread_l(output); 395 if (thread == NULL) { 396 LOGW("channelCount() unknown thread %d", output); 397 return 0; 398 } 399 return thread->channelCount(); 400} 401 402int AudioFlinger::format(int output) const 403{ 404 Mutex::Autolock _l(mLock); 405 PlaybackThread *thread = checkPlaybackThread_l(output); 406 if (thread == NULL) { 407 LOGW("format() unknown thread %d", output); 408 return 0; 409 } 410 return thread->format(); 411} 412 413size_t AudioFlinger::frameCount(int output) const 414{ 415 Mutex::Autolock _l(mLock); 416 PlaybackThread *thread = checkPlaybackThread_l(output); 417 if (thread == NULL) { 418 LOGW("frameCount() unknown thread %d", output); 419 return 0; 420 } 421 return thread->frameCount(); 422} 423 424uint32_t AudioFlinger::latency(int output) const 425{ 426 Mutex::Autolock _l(mLock); 427 PlaybackThread *thread = checkPlaybackThread_l(output); 428 if (thread == NULL) { 429 LOGW("latency() unknown thread %d", output); 430 return 0; 431 } 432 return thread->latency(); 433} 434 435status_t AudioFlinger::setMasterVolume(float value) 436{ 437 // check calling permissions 438 if (!settingsAllowed()) { 439 return PERMISSION_DENIED; 440 } 441 442 // when hw supports master volume, don't scale in sw mixer 443 AutoMutex lock(mHardwareLock); 444 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 445 if (mAudioHardware->setMasterVolume(value) == NO_ERROR) { 446 value = 1.0f; 447 } 448 mHardwareStatus = AUDIO_HW_IDLE; 449 450 mMasterVolume = value; 451 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 452 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 453 454 return NO_ERROR; 455} 456 457status_t AudioFlinger::setMode(int mode) 458{ 459 status_t ret; 460 461 // check calling permissions 462 if (!settingsAllowed()) { 463 return PERMISSION_DENIED; 464 } 465 if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { 466 LOGW("Illegal value: setMode(%d)", mode); 467 return BAD_VALUE; 468 } 469 470 { // scope for the lock 471 AutoMutex lock(mHardwareLock); 472 mHardwareStatus = AUDIO_HW_SET_MODE; 473 ret = mAudioHardware->setMode(mode); 474 mHardwareStatus = AUDIO_HW_IDLE; 475 } 476 477 if (NO_ERROR == ret) { 478 Mutex::Autolock _l(mLock); 479 mMode = mode; 480 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 481 mPlaybackThreads.valueAt(i)->setMode(mode); 482#ifdef LVMX 483 LifeVibes::setMode(mode); 484#endif 485 } 486 487 return ret; 488} 489 490status_t AudioFlinger::setMicMute(bool state) 491{ 492 // check calling permissions 493 if (!settingsAllowed()) { 494 return PERMISSION_DENIED; 495 } 496 497 AutoMutex lock(mHardwareLock); 498 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 499 status_t ret = mAudioHardware->setMicMute(state); 500 mHardwareStatus = AUDIO_HW_IDLE; 501 return ret; 502} 503 504bool AudioFlinger::getMicMute() const 505{ 506 bool state = AudioSystem::MODE_INVALID; 507 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 508 mAudioHardware->getMicMute(&state); 509 mHardwareStatus = AUDIO_HW_IDLE; 510 return state; 511} 512 513status_t AudioFlinger::setMasterMute(bool muted) 514{ 515 // check calling permissions 516 if (!settingsAllowed()) { 517 return PERMISSION_DENIED; 518 } 519 520 mMasterMute = muted; 521 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 522 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 523 524 return NO_ERROR; 525} 526 527float AudioFlinger::masterVolume() const 528{ 529 return mMasterVolume; 530} 531 532bool AudioFlinger::masterMute() const 533{ 534 return mMasterMute; 535} 536 537status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 538{ 539 // check calling permissions 540 if (!settingsAllowed()) { 541 return PERMISSION_DENIED; 542 } 543 544 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { 545 return BAD_VALUE; 546 } 547 548 AutoMutex lock(mLock); 549 PlaybackThread *thread = NULL; 550 if (output) { 551 thread = checkPlaybackThread_l(output); 552 if (thread == NULL) { 553 return BAD_VALUE; 554 } 555 } 556 557 mStreamTypes[stream].volume = value; 558 559 if (thread == NULL) { 560 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 561 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 562 } 563 } else { 564 thread->setStreamVolume(stream, value); 565 } 566 567 return NO_ERROR; 568} 569 570status_t AudioFlinger::setStreamMute(int stream, bool muted) 571{ 572 // check calling permissions 573 if (!settingsAllowed()) { 574 return PERMISSION_DENIED; 575 } 576 577 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || 578 uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { 579 return BAD_VALUE; 580 } 581 582 mStreamTypes[stream].mute = muted; 583 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 584 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 585 586 return NO_ERROR; 587} 588 589float AudioFlinger::streamVolume(int stream, int output) const 590{ 591 if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { 592 return 0.0f; 593 } 594 595 AutoMutex lock(mLock); 596 float volume; 597 if (output) { 598 PlaybackThread *thread = checkPlaybackThread_l(output); 599 if (thread == NULL) { 600 return 0.0f; 601 } 602 volume = thread->streamVolume(stream); 603 } else { 604 volume = mStreamTypes[stream].volume; 605 } 606 607 return volume; 608} 609 610bool AudioFlinger::streamMute(int stream) const 611{ 612 if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { 613 return true; 614 } 615 616 return mStreamTypes[stream].mute; 617} 618 619bool AudioFlinger::isStreamActive(int stream) const 620{ 621 Mutex::Autolock _l(mLock); 622 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 623 if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) { 624 return true; 625 } 626 } 627 return false; 628} 629 630status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 631{ 632 status_t result; 633 634 LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 635 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 636 // check calling permissions 637 if (!settingsAllowed()) { 638 return PERMISSION_DENIED; 639 } 640 641#ifdef LVMX 642 AudioParameter param = AudioParameter(keyValuePairs); 643 LifeVibes::setParameters(ioHandle,keyValuePairs); 644 String8 key = String8(AudioParameter::keyRouting); 645 int device; 646 if (NO_ERROR != param.getInt(key, device)) { 647 device = -1; 648 } 649 650 key = String8(LifevibesTag); 651 String8 value; 652 int musicEnabled = -1; 653 if (NO_ERROR == param.get(key, value)) { 654 if (value == LifevibesEnable) { 655 mLifeVibesClientPid = IPCThreadState::self()->getCallingPid(); 656 musicEnabled = 1; 657 } else if (value == LifevibesDisable) { 658 mLifeVibesClientPid = -1; 659 musicEnabled = 0; 660 } 661 } 662#endif 663 664 // ioHandle == 0 means the parameters are global to the audio hardware interface 665 if (ioHandle == 0) { 666 AutoMutex lock(mHardwareLock); 667 mHardwareStatus = AUDIO_SET_PARAMETER; 668 result = mAudioHardware->setParameters(keyValuePairs); 669#ifdef LVMX 670 if (musicEnabled != -1) { 671 LifeVibes::enableMusic((bool) musicEnabled); 672 } 673#endif 674 mHardwareStatus = AUDIO_HW_IDLE; 675 return result; 676 } 677 678 // hold a strong ref on thread in case closeOutput() or closeInput() is called 679 // and the thread is exited once the lock is released 680 sp<ThreadBase> thread; 681 { 682 Mutex::Autolock _l(mLock); 683 thread = checkPlaybackThread_l(ioHandle); 684 if (thread == NULL) { 685 thread = checkRecordThread_l(ioHandle); 686 } 687 } 688 if (thread != NULL) { 689 result = thread->setParameters(keyValuePairs); 690#ifdef LVMX 691 if ((NO_ERROR == result) && (device != -1)) { 692 LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device); 693 } 694#endif 695 return result; 696 } 697 return BAD_VALUE; 698} 699 700String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 701{ 702// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 703// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 704 705 if (ioHandle == 0) { 706 return mAudioHardware->getParameters(keys); 707 } 708 709 Mutex::Autolock _l(mLock); 710 711 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 712 if (playbackThread != NULL) { 713 return playbackThread->getParameters(keys); 714 } 715 RecordThread *recordThread = checkRecordThread_l(ioHandle); 716 if (recordThread != NULL) { 717 return recordThread->getParameters(keys); 718 } 719 return String8(""); 720} 721 722size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 723{ 724 return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); 725} 726 727unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 728{ 729 if (ioHandle == 0) { 730 return 0; 731 } 732 733 Mutex::Autolock _l(mLock); 734 735 RecordThread *recordThread = checkRecordThread_l(ioHandle); 736 if (recordThread != NULL) { 737 return recordThread->getInputFramesLost(); 738 } 739 return 0; 740} 741 742status_t AudioFlinger::setVoiceVolume(float value) 743{ 744 // check calling permissions 745 if (!settingsAllowed()) { 746 return PERMISSION_DENIED; 747 } 748 749 AutoMutex lock(mHardwareLock); 750 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 751 status_t ret = mAudioHardware->setVoiceVolume(value); 752 mHardwareStatus = AUDIO_HW_IDLE; 753 754 return ret; 755} 756 757status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 758{ 759 status_t status; 760 761 Mutex::Autolock _l(mLock); 762 763 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 764 if (playbackThread != NULL) { 765 return playbackThread->getRenderPosition(halFrames, dspFrames); 766 } 767 768 return BAD_VALUE; 769} 770 771void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 772{ 773 774 Mutex::Autolock _l(mLock); 775 776 int pid = IPCThreadState::self()->getCallingPid(); 777 if (mNotificationClients.indexOfKey(pid) < 0) { 778 sp<NotificationClient> notificationClient = new NotificationClient(this, 779 client, 780 pid); 781 LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 782 783 mNotificationClients.add(pid, notificationClient); 784 785 sp<IBinder> binder = client->asBinder(); 786 binder->linkToDeath(notificationClient); 787 788 // the config change is always sent from playback or record threads to avoid deadlock 789 // with AudioSystem::gLock 790 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 791 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 792 } 793 794 for (size_t i = 0; i < mRecordThreads.size(); i++) { 795 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 796 } 797 } 798} 799 800void AudioFlinger::removeNotificationClient(pid_t pid) 801{ 802 Mutex::Autolock _l(mLock); 803 804 int index = mNotificationClients.indexOfKey(pid); 805 if (index >= 0) { 806 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 807 LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 808#ifdef LVMX 809 if (pid == mLifeVibesClientPid) { 810 LOGV("Disabling lifevibes"); 811 LifeVibes::enableMusic(false); 812 mLifeVibesClientPid = -1; 813 } 814#endif 815 mNotificationClients.removeItem(pid); 816 } 817} 818 819// audioConfigChanged_l() must be called with AudioFlinger::mLock held 820void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 821{ 822 size_t size = mNotificationClients.size(); 823 for (size_t i = 0; i < size; i++) { 824 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 825 } 826} 827 828// removeClient_l() must be called with AudioFlinger::mLock held 829void AudioFlinger::removeClient_l(pid_t pid) 830{ 831 LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 832 mClients.removeItem(pid); 833} 834 835 836// ---------------------------------------------------------------------------- 837 838AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id) 839 : Thread(false), 840 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 841 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) 842{ 843} 844 845AudioFlinger::ThreadBase::~ThreadBase() 846{ 847 mParamCond.broadcast(); 848 mNewParameters.clear(); 849} 850 851void AudioFlinger::ThreadBase::exit() 852{ 853 // keep a strong ref on ourself so that we wont get 854 // destroyed in the middle of requestExitAndWait() 855 sp <ThreadBase> strongMe = this; 856 857 LOGV("ThreadBase::exit"); 858 { 859 AutoMutex lock(&mLock); 860 mExiting = true; 861 requestExit(); 862 mWaitWorkCV.signal(); 863 } 864 requestExitAndWait(); 865} 866 867uint32_t AudioFlinger::ThreadBase::sampleRate() const 868{ 869 return mSampleRate; 870} 871 872int AudioFlinger::ThreadBase::channelCount() const 873{ 874 return (int)mChannelCount; 875} 876 877int AudioFlinger::ThreadBase::format() const 878{ 879 return mFormat; 880} 881 882size_t AudioFlinger::ThreadBase::frameCount() const 883{ 884 return mFrameCount; 885} 886 887status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 888{ 889 status_t status; 890 891 LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 892 Mutex::Autolock _l(mLock); 893 894 mNewParameters.add(keyValuePairs); 895 mWaitWorkCV.signal(); 896 // wait condition with timeout in case the thread loop has exited 897 // before the request could be processed 898 if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { 899 status = mParamStatus; 900 mWaitWorkCV.signal(); 901 } else { 902 status = TIMED_OUT; 903 } 904 return status; 905} 906 907void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 908{ 909 Mutex::Autolock _l(mLock); 910 sendConfigEvent_l(event, param); 911} 912 913// sendConfigEvent_l() must be called with ThreadBase::mLock held 914void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 915{ 916 ConfigEvent *configEvent = new ConfigEvent(); 917 configEvent->mEvent = event; 918 configEvent->mParam = param; 919 mConfigEvents.add(configEvent); 920 LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 921 mWaitWorkCV.signal(); 922} 923 924void AudioFlinger::ThreadBase::processConfigEvents() 925{ 926 mLock.lock(); 927 while(!mConfigEvents.isEmpty()) { 928 LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 929 ConfigEvent *configEvent = mConfigEvents[0]; 930 mConfigEvents.removeAt(0); 931 // release mLock before locking AudioFlinger mLock: lock order is always 932 // AudioFlinger then ThreadBase to avoid cross deadlock 933 mLock.unlock(); 934 mAudioFlinger->mLock.lock(); 935 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 936 mAudioFlinger->mLock.unlock(); 937 delete configEvent; 938 mLock.lock(); 939 } 940 mLock.unlock(); 941} 942 943status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 944{ 945 const size_t SIZE = 256; 946 char buffer[SIZE]; 947 String8 result; 948 949 bool locked = tryLock(mLock); 950 if (!locked) { 951 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 952 write(fd, buffer, strlen(buffer)); 953 } 954 955 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 956 result.append(buffer); 957 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 958 result.append(buffer); 959 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 960 result.append(buffer); 961 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 962 result.append(buffer); 963 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 964 result.append(buffer); 965 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 966 result.append(buffer); 967 968 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 969 result.append(buffer); 970 result.append(" Index Command"); 971 for (size_t i = 0; i < mNewParameters.size(); ++i) { 972 snprintf(buffer, SIZE, "\n %02d ", i); 973 result.append(buffer); 974 result.append(mNewParameters[i]); 975 } 976 977 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 978 result.append(buffer); 979 snprintf(buffer, SIZE, " Index event param\n"); 980 result.append(buffer); 981 for (size_t i = 0; i < mConfigEvents.size(); i++) { 982 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 983 result.append(buffer); 984 } 985 result.append("\n"); 986 987 write(fd, result.string(), result.size()); 988 989 if (locked) { 990 mLock.unlock(); 991 } 992 return NO_ERROR; 993} 994 995 996// ---------------------------------------------------------------------------- 997 998AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 999 : ThreadBase(audioFlinger, id), 1000 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1001 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1002 mDevice(device) 1003{ 1004 readOutputParameters(); 1005 1006 mMasterVolume = mAudioFlinger->masterVolume(); 1007 mMasterMute = mAudioFlinger->masterMute(); 1008 1009 for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { 1010 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1011 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1012 } 1013} 1014 1015AudioFlinger::PlaybackThread::~PlaybackThread() 1016{ 1017 delete [] mMixBuffer; 1018} 1019 1020status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1021{ 1022 dumpInternals(fd, args); 1023 dumpTracks(fd, args); 1024 dumpEffectChains(fd, args); 1025 return NO_ERROR; 1026} 1027 1028status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1029{ 1030 const size_t SIZE = 256; 1031 char buffer[SIZE]; 1032 String8 result; 1033 1034 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1035 result.append(buffer); 1036 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1037 for (size_t i = 0; i < mTracks.size(); ++i) { 1038 sp<Track> track = mTracks[i]; 1039 if (track != 0) { 1040 track->dump(buffer, SIZE); 1041 result.append(buffer); 1042 } 1043 } 1044 1045 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1046 result.append(buffer); 1047 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1048 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1049 wp<Track> wTrack = mActiveTracks[i]; 1050 if (wTrack != 0) { 1051 sp<Track> track = wTrack.promote(); 1052 if (track != 0) { 1053 track->dump(buffer, SIZE); 1054 result.append(buffer); 1055 } 1056 } 1057 } 1058 write(fd, result.string(), result.size()); 1059 return NO_ERROR; 1060} 1061 1062status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args) 1063{ 1064 const size_t SIZE = 256; 1065 char buffer[SIZE]; 1066 String8 result; 1067 1068 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1069 write(fd, buffer, strlen(buffer)); 1070 1071 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1072 sp<EffectChain> chain = mEffectChains[i]; 1073 if (chain != 0) { 1074 chain->dump(fd, args); 1075 } 1076 } 1077 return NO_ERROR; 1078} 1079 1080status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1081{ 1082 const size_t SIZE = 256; 1083 char buffer[SIZE]; 1084 String8 result; 1085 1086 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1087 result.append(buffer); 1088 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1089 result.append(buffer); 1090 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1091 result.append(buffer); 1092 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1093 result.append(buffer); 1094 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1095 result.append(buffer); 1096 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1097 result.append(buffer); 1098 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1099 result.append(buffer); 1100 write(fd, result.string(), result.size()); 1101 1102 dumpBase(fd, args); 1103 1104 return NO_ERROR; 1105} 1106 1107// Thread virtuals 1108status_t AudioFlinger::PlaybackThread::readyToRun() 1109{ 1110 if (mSampleRate == 0) { 1111 LOGE("No working audio driver found."); 1112 return NO_INIT; 1113 } 1114 LOGI("AudioFlinger's thread %p ready to run", this); 1115 return NO_ERROR; 1116} 1117 1118void AudioFlinger::PlaybackThread::onFirstRef() 1119{ 1120 const size_t SIZE = 256; 1121 char buffer[SIZE]; 1122 1123 snprintf(buffer, SIZE, "Playback Thread %p", this); 1124 1125 run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); 1126} 1127 1128// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1129sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1130 const sp<AudioFlinger::Client>& client, 1131 int streamType, 1132 uint32_t sampleRate, 1133 int format, 1134 int channelCount, 1135 int frameCount, 1136 const sp<IMemory>& sharedBuffer, 1137 int sessionId, 1138 status_t *status) 1139{ 1140 sp<Track> track; 1141 status_t lStatus; 1142 1143 if (mType == DIRECT) { 1144 if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) { 1145 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", 1146 sampleRate, format, channelCount, mOutput); 1147 lStatus = BAD_VALUE; 1148 goto Exit; 1149 } 1150 } else { 1151 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1152 if (sampleRate > mSampleRate*2) { 1153 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1154 lStatus = BAD_VALUE; 1155 goto Exit; 1156 } 1157 } 1158 1159 if (mOutput == 0) { 1160 LOGE("Audio driver not initialized."); 1161 lStatus = NO_INIT; 1162 goto Exit; 1163 } 1164 1165 { // scope for mLock 1166 Mutex::Autolock _l(mLock); 1167 1168 // all tracks in same audio session must share the same routing strategy otherwise 1169 // conflicts will happen when tracks are moved from one output to another by audio policy 1170 // manager 1171 uint32_t strategy = 1172 AudioSystem::getStrategyForStream((AudioSystem::stream_type)streamType); 1173 for (size_t i = 0; i < mTracks.size(); ++i) { 1174 sp<Track> t = mTracks[i]; 1175 if (t != 0) { 1176 if (sessionId == t->sessionId() && 1177 strategy != AudioSystem::getStrategyForStream((AudioSystem::stream_type)t->type())) { 1178 lStatus = BAD_VALUE; 1179 goto Exit; 1180 } 1181 } 1182 } 1183 1184 track = new Track(this, client, streamType, sampleRate, format, 1185 channelCount, frameCount, sharedBuffer, sessionId); 1186 if (track->getCblk() == NULL || track->name() < 0) { 1187 lStatus = NO_MEMORY; 1188 goto Exit; 1189 } 1190 mTracks.add(track); 1191 1192 sp<EffectChain> chain = getEffectChain_l(sessionId); 1193 if (chain != 0) { 1194 LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1195 track->setMainBuffer(chain->inBuffer()); 1196 chain->setStrategy(AudioSystem::getStrategyForStream((AudioSystem::stream_type)track->type())); 1197 } 1198 } 1199 lStatus = NO_ERROR; 1200 1201Exit: 1202 if(status) { 1203 *status = lStatus; 1204 } 1205 return track; 1206} 1207 1208uint32_t AudioFlinger::PlaybackThread::latency() const 1209{ 1210 if (mOutput) { 1211 return mOutput->latency(); 1212 } 1213 else { 1214 return 0; 1215 } 1216} 1217 1218status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1219{ 1220#ifdef LVMX 1221 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1222 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1223 LifeVibes::setMasterVolume(audioOutputType, value); 1224 } 1225#endif 1226 mMasterVolume = value; 1227 return NO_ERROR; 1228} 1229 1230status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1231{ 1232#ifdef LVMX 1233 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1234 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1235 LifeVibes::setMasterMute(audioOutputType, muted); 1236 } 1237#endif 1238 mMasterMute = muted; 1239 return NO_ERROR; 1240} 1241 1242float AudioFlinger::PlaybackThread::masterVolume() const 1243{ 1244 return mMasterVolume; 1245} 1246 1247bool AudioFlinger::PlaybackThread::masterMute() const 1248{ 1249 return mMasterMute; 1250} 1251 1252status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1253{ 1254#ifdef LVMX 1255 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1256 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1257 LifeVibes::setStreamVolume(audioOutputType, stream, value); 1258 } 1259#endif 1260 mStreamTypes[stream].volume = value; 1261 return NO_ERROR; 1262} 1263 1264status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1265{ 1266#ifdef LVMX 1267 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1268 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1269 LifeVibes::setStreamMute(audioOutputType, stream, muted); 1270 } 1271#endif 1272 mStreamTypes[stream].mute = muted; 1273 return NO_ERROR; 1274} 1275 1276float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1277{ 1278 return mStreamTypes[stream].volume; 1279} 1280 1281bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1282{ 1283 return mStreamTypes[stream].mute; 1284} 1285 1286bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const 1287{ 1288 Mutex::Autolock _l(mLock); 1289 size_t count = mActiveTracks.size(); 1290 for (size_t i = 0 ; i < count ; ++i) { 1291 sp<Track> t = mActiveTracks[i].promote(); 1292 if (t == 0) continue; 1293 Track* const track = t.get(); 1294 if (t->type() == stream) 1295 return true; 1296 } 1297 return false; 1298} 1299 1300// addTrack_l() must be called with ThreadBase::mLock held 1301status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1302{ 1303 status_t status = ALREADY_EXISTS; 1304 1305 // set retry count for buffer fill 1306 track->mRetryCount = kMaxTrackStartupRetries; 1307 if (mActiveTracks.indexOf(track) < 0) { 1308 // the track is newly added, make sure it fills up all its 1309 // buffers before playing. This is to ensure the client will 1310 // effectively get the latency it requested. 1311 track->mFillingUpStatus = Track::FS_FILLING; 1312 track->mResetDone = false; 1313 mActiveTracks.add(track); 1314 if (track->mainBuffer() != mMixBuffer) { 1315 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1316 if (chain != 0) { 1317 LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1318 chain->startTrack(); 1319 } 1320 } 1321 1322 status = NO_ERROR; 1323 } 1324 1325 LOGV("mWaitWorkCV.broadcast"); 1326 mWaitWorkCV.broadcast(); 1327 1328 return status; 1329} 1330 1331// destroyTrack_l() must be called with ThreadBase::mLock held 1332void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1333{ 1334 track->mState = TrackBase::TERMINATED; 1335 if (mActiveTracks.indexOf(track) < 0) { 1336 mTracks.remove(track); 1337 deleteTrackName_l(track->name()); 1338 } 1339} 1340 1341String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1342{ 1343 return mOutput->getParameters(keys); 1344} 1345 1346// destroyTrack_l() must be called with AudioFlinger::mLock held 1347void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1348 AudioSystem::OutputDescriptor desc; 1349 void *param2 = 0; 1350 1351 LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1352 1353 switch (event) { 1354 case AudioSystem::OUTPUT_OPENED: 1355 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1356 desc.channels = mChannels; 1357 desc.samplingRate = mSampleRate; 1358 desc.format = mFormat; 1359 desc.frameCount = mFrameCount; 1360 desc.latency = latency(); 1361 param2 = &desc; 1362 break; 1363 1364 case AudioSystem::STREAM_CONFIG_CHANGED: 1365 param2 = ¶m; 1366 case AudioSystem::OUTPUT_CLOSED: 1367 default: 1368 break; 1369 } 1370 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1371} 1372 1373void AudioFlinger::PlaybackThread::readOutputParameters() 1374{ 1375 mSampleRate = mOutput->sampleRate(); 1376 mChannels = mOutput->channels(); 1377 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 1378 mFormat = mOutput->format(); 1379 mFrameSize = (uint16_t)mOutput->frameSize(); 1380 mFrameCount = mOutput->bufferSize() / mFrameSize; 1381 1382 // FIXME - Current mixer implementation only supports stereo output: Always 1383 // Allocate a stereo buffer even if HW output is mono. 1384 if (mMixBuffer != NULL) delete[] mMixBuffer; 1385 mMixBuffer = new int16_t[mFrameCount * 2]; 1386 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1387 1388 // force reconfiguration of effect chains and engines to take new buffer size and audio 1389 // parameters into account 1390 // Note that mLock is not held when readOutputParameters() is called from the constructor 1391 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1392 // matter. 1393 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1394 Vector< sp<EffectChain> > effectChains = mEffectChains; 1395 for (size_t i = 0; i < effectChains.size(); i ++) { 1396 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1397 } 1398} 1399 1400status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1401{ 1402 if (halFrames == 0 || dspFrames == 0) { 1403 return BAD_VALUE; 1404 } 1405 if (mOutput == 0) { 1406 return INVALID_OPERATION; 1407 } 1408 *halFrames = mBytesWritten/mOutput->frameSize(); 1409 1410 return mOutput->getRenderPosition(dspFrames); 1411} 1412 1413uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1414{ 1415 Mutex::Autolock _l(mLock); 1416 uint32_t result = 0; 1417 if (getEffectChain_l(sessionId) != 0) { 1418 result = EFFECT_SESSION; 1419 } 1420 1421 for (size_t i = 0; i < mTracks.size(); ++i) { 1422 sp<Track> track = mTracks[i]; 1423 if (sessionId == track->sessionId() && 1424 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1425 result |= TRACK_SESSION; 1426 break; 1427 } 1428 } 1429 1430 return result; 1431} 1432 1433uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1434{ 1435 // session AudioSystem::SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1436 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1437 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 1438 return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 1439 } 1440 for (size_t i = 0; i < mTracks.size(); i++) { 1441 sp<Track> track = mTracks[i]; 1442 if (sessionId == track->sessionId() && 1443 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1444 return AudioSystem::getStrategyForStream((AudioSystem::stream_type) track->type()); 1445 } 1446 } 1447 return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 1448} 1449 1450sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId) 1451{ 1452 Mutex::Autolock _l(mLock); 1453 return getEffectChain_l(sessionId); 1454} 1455 1456sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) 1457{ 1458 sp<EffectChain> chain; 1459 1460 size_t size = mEffectChains.size(); 1461 for (size_t i = 0; i < size; i++) { 1462 if (mEffectChains[i]->sessionId() == sessionId) { 1463 chain = mEffectChains[i]; 1464 break; 1465 } 1466 } 1467 return chain; 1468} 1469 1470void AudioFlinger::PlaybackThread::setMode(uint32_t mode) 1471{ 1472 Mutex::Autolock _l(mLock); 1473 size_t size = mEffectChains.size(); 1474 for (size_t i = 0; i < size; i++) { 1475 mEffectChains[i]->setMode_l(mode); 1476 } 1477} 1478 1479// ---------------------------------------------------------------------------- 1480 1481AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1482 : PlaybackThread(audioFlinger, output, id, device), 1483 mAudioMixer(0) 1484{ 1485 mType = PlaybackThread::MIXER; 1486 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1487 1488 // FIXME - Current mixer implementation only supports stereo output 1489 if (mChannelCount == 1) { 1490 LOGE("Invalid audio hardware channel count"); 1491 } 1492} 1493 1494AudioFlinger::MixerThread::~MixerThread() 1495{ 1496 delete mAudioMixer; 1497} 1498 1499bool AudioFlinger::MixerThread::threadLoop() 1500{ 1501 Vector< sp<Track> > tracksToRemove; 1502 uint32_t mixerStatus = MIXER_IDLE; 1503 nsecs_t standbyTime = systemTime(); 1504 size_t mixBufferSize = mFrameCount * mFrameSize; 1505 // FIXME: Relaxed timing because of a certain device that can't meet latency 1506 // Should be reduced to 2x after the vendor fixes the driver issue 1507 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1508 nsecs_t lastWarning = 0; 1509 bool longStandbyExit = false; 1510 uint32_t activeSleepTime = activeSleepTimeUs(); 1511 uint32_t idleSleepTime = idleSleepTimeUs(); 1512 uint32_t sleepTime = idleSleepTime; 1513 Vector< sp<EffectChain> > effectChains; 1514 1515 while (!exitPending()) 1516 { 1517 processConfigEvents(); 1518 1519 mixerStatus = MIXER_IDLE; 1520 { // scope for mLock 1521 1522 Mutex::Autolock _l(mLock); 1523 1524 if (checkForNewParameters_l()) { 1525 mixBufferSize = mFrameCount * mFrameSize; 1526 // FIXME: Relaxed timing because of a certain device that can't meet latency 1527 // Should be reduced to 2x after the vendor fixes the driver issue 1528 maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1529 activeSleepTime = activeSleepTimeUs(); 1530 idleSleepTime = idleSleepTimeUs(); 1531 } 1532 1533 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1534 1535 // put audio hardware into standby after short delay 1536 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1537 mSuspended) { 1538 if (!mStandby) { 1539 LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1540 mOutput->standby(); 1541 mStandby = true; 1542 mBytesWritten = 0; 1543 } 1544 1545 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1546 // we're about to wait, flush the binder command buffer 1547 IPCThreadState::self()->flushCommands(); 1548 1549 if (exitPending()) break; 1550 1551 // wait until we have something to do... 1552 LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1553 mWaitWorkCV.wait(mLock); 1554 LOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1555 1556 if (mMasterMute == false) { 1557 char value[PROPERTY_VALUE_MAX]; 1558 property_get("ro.audio.silent", value, "0"); 1559 if (atoi(value)) { 1560 LOGD("Silence is golden"); 1561 setMasterMute(true); 1562 } 1563 } 1564 1565 standbyTime = systemTime() + kStandbyTimeInNsecs; 1566 sleepTime = idleSleepTime; 1567 continue; 1568 } 1569 } 1570 1571 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1572 1573 // prevent any changes in effect chain list and in each effect chain 1574 // during mixing and effect process as the audio buffers could be deleted 1575 // or modified if an effect is created or deleted 1576 lockEffectChains_l(effectChains); 1577 } 1578 1579 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1580 // mix buffers... 1581 mAudioMixer->process(); 1582 sleepTime = 0; 1583 standbyTime = systemTime() + kStandbyTimeInNsecs; 1584 //TODO: delay standby when effects have a tail 1585 } else { 1586 // If no tracks are ready, sleep once for the duration of an output 1587 // buffer size, then write 0s to the output 1588 if (sleepTime == 0) { 1589 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1590 sleepTime = activeSleepTime; 1591 } else { 1592 sleepTime = idleSleepTime; 1593 } 1594 } else if (mBytesWritten != 0 || 1595 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1596 memset (mMixBuffer, 0, mixBufferSize); 1597 sleepTime = 0; 1598 LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1599 } 1600 // TODO add standby time extension fct of effect tail 1601 } 1602 1603 if (mSuspended) { 1604 sleepTime = idleSleepTime; 1605 } 1606 // sleepTime == 0 means we must write to audio hardware 1607 if (sleepTime == 0) { 1608 for (size_t i = 0; i < effectChains.size(); i ++) { 1609 effectChains[i]->process_l(); 1610 } 1611 // enable changes in effect chain 1612 unlockEffectChains(effectChains); 1613#ifdef LVMX 1614 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1615 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { 1616 LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize); 1617 } 1618#endif 1619 mLastWriteTime = systemTime(); 1620 mInWrite = true; 1621 mBytesWritten += mixBufferSize; 1622 1623 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 1624 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 1625 mNumWrites++; 1626 mInWrite = false; 1627 nsecs_t now = systemTime(); 1628 nsecs_t delta = now - mLastWriteTime; 1629 if (delta > maxPeriod) { 1630 mNumDelayedWrites++; 1631 if ((now - lastWarning) > kWarningThrottle) { 1632 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 1633 ns2ms(delta), mNumDelayedWrites, this); 1634 lastWarning = now; 1635 } 1636 if (mStandby) { 1637 longStandbyExit = true; 1638 } 1639 } 1640 mStandby = false; 1641 } else { 1642 // enable changes in effect chain 1643 unlockEffectChains(effectChains); 1644 usleep(sleepTime); 1645 } 1646 1647 // finally let go of all our tracks, without the lock held 1648 // since we can't guarantee the destructors won't acquire that 1649 // same lock. 1650 tracksToRemove.clear(); 1651 1652 // Effect chains will be actually deleted here if they were removed from 1653 // mEffectChains list during mixing or effects processing 1654 effectChains.clear(); 1655 } 1656 1657 if (!mStandby) { 1658 mOutput->standby(); 1659 } 1660 1661 LOGV("MixerThread %p exiting", this); 1662 return false; 1663} 1664 1665// prepareTracks_l() must be called with ThreadBase::mLock held 1666uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 1667{ 1668 1669 uint32_t mixerStatus = MIXER_IDLE; 1670 // find out which tracks need to be processed 1671 size_t count = activeTracks.size(); 1672 size_t mixedTracks = 0; 1673 size_t tracksWithEffect = 0; 1674 1675 float masterVolume = mMasterVolume; 1676 bool masterMute = mMasterMute; 1677 1678#ifdef LVMX 1679 bool tracksConnectedChanged = false; 1680 bool stateChanged = false; 1681 1682 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1683 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1684 { 1685 int activeTypes = 0; 1686 for (size_t i=0 ; i<count ; i++) { 1687 sp<Track> t = activeTracks[i].promote(); 1688 if (t == 0) continue; 1689 Track* const track = t.get(); 1690 int iTracktype=track->type(); 1691 activeTypes |= 1<<track->type(); 1692 } 1693 LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute); 1694 } 1695#endif 1696 // Delegate master volume control to effect in output mix effect chain if needed 1697 sp<EffectChain> chain = getEffectChain_l(AudioSystem::SESSION_OUTPUT_MIX); 1698 if (chain != 0) { 1699 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 1700 chain->setVolume_l(&v, &v); 1701 masterVolume = (float)((v + (1 << 23)) >> 24); 1702 chain.clear(); 1703 } 1704 1705 for (size_t i=0 ; i<count ; i++) { 1706 sp<Track> t = activeTracks[i].promote(); 1707 if (t == 0) continue; 1708 1709 Track* const track = t.get(); 1710 audio_track_cblk_t* cblk = track->cblk(); 1711 1712 // The first time a track is added we wait 1713 // for all its buffers to be filled before processing it 1714 mAudioMixer->setActiveTrack(track->name()); 1715 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 1716 !track->isPaused() && !track->isTerminated()) 1717 { 1718 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1719 1720 mixedTracks++; 1721 1722 // track->mainBuffer() != mMixBuffer means there is an effect chain 1723 // connected to the track 1724 chain.clear(); 1725 if (track->mainBuffer() != mMixBuffer) { 1726 chain = getEffectChain_l(track->sessionId()); 1727 // Delegate volume control to effect in track effect chain if needed 1728 if (chain != 0) { 1729 tracksWithEffect++; 1730 } else { 1731 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1732 track->name(), track->sessionId()); 1733 } 1734 } 1735 1736 1737 int param = AudioMixer::VOLUME; 1738 if (track->mFillingUpStatus == Track::FS_FILLED) { 1739 // no ramp for the first volume setting 1740 track->mFillingUpStatus = Track::FS_ACTIVE; 1741 if (track->mState == TrackBase::RESUMING) { 1742 track->mState = TrackBase::ACTIVE; 1743 param = AudioMixer::RAMP_VOLUME; 1744 } 1745 } else if (cblk->server != 0) { 1746 // If the track is stopped before the first frame was mixed, 1747 // do not apply ramp 1748 param = AudioMixer::RAMP_VOLUME; 1749 } 1750 1751 // compute volume for this track 1752 int16_t left, right, aux; 1753 if (track->isMuted() || masterMute || track->isPausing() || 1754 mStreamTypes[track->type()].mute) { 1755 left = right = aux = 0; 1756 if (track->isPausing()) { 1757 track->setPaused(); 1758 } 1759 } else { 1760 // read original volumes with volume control 1761 float typeVolume = mStreamTypes[track->type()].volume; 1762#ifdef LVMX 1763 bool streamMute=false; 1764 // read the volume from the LivesVibes audio engine. 1765 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1766 { 1767 LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); 1768 if (streamMute) { 1769 typeVolume = 0; 1770 } 1771 } 1772#endif 1773 float v = masterVolume * typeVolume; 1774 uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12; 1775 uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12; 1776 1777 // Delegate volume control to effect in track effect chain if needed 1778 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1779 // Do not ramp volume is volume is controlled by effect 1780 param = AudioMixer::VOLUME; 1781 } 1782 1783 // Convert volumes from 8.24 to 4.12 format 1784 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1785 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1786 left = int16_t(v_clamped); 1787 v_clamped = (vr + (1 << 11)) >> 12; 1788 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1789 right = int16_t(v_clamped); 1790 1791 v_clamped = (uint32_t)(v * cblk->sendLevel); 1792 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1793 aux = int16_t(v_clamped); 1794 } 1795 1796#ifdef LVMX 1797 if ( tracksConnectedChanged || stateChanged ) 1798 { 1799 // only do the ramp when the volume is changed by the user / application 1800 param = AudioMixer::VOLUME; 1801 } 1802#endif 1803 1804 // XXX: these things DON'T need to be done each time 1805 mAudioMixer->setBufferProvider(track); 1806 mAudioMixer->enable(AudioMixer::MIXING); 1807 1808 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1809 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1810 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1811 mAudioMixer->setParameter( 1812 AudioMixer::TRACK, 1813 AudioMixer::FORMAT, (void *)track->format()); 1814 mAudioMixer->setParameter( 1815 AudioMixer::TRACK, 1816 AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); 1817 mAudioMixer->setParameter( 1818 AudioMixer::RESAMPLE, 1819 AudioMixer::SAMPLE_RATE, 1820 (void *)(cblk->sampleRate)); 1821 mAudioMixer->setParameter( 1822 AudioMixer::TRACK, 1823 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1824 mAudioMixer->setParameter( 1825 AudioMixer::TRACK, 1826 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1827 1828 // reset retry count 1829 track->mRetryCount = kMaxTrackRetries; 1830 mixerStatus = MIXER_TRACKS_READY; 1831 } else { 1832 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1833 if (track->isStopped()) { 1834 track->reset(); 1835 } 1836 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1837 // We have consumed all the buffers of this track. 1838 // Remove it from the list of active tracks. 1839 tracksToRemove->add(track); 1840 } else { 1841 // No buffers for this track. Give it a few chances to 1842 // fill a buffer, then remove it from active list. 1843 if (--(track->mRetryCount) <= 0) { 1844 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1845 tracksToRemove->add(track); 1846 } else if (mixerStatus != MIXER_TRACKS_READY) { 1847 mixerStatus = MIXER_TRACKS_ENABLED; 1848 } 1849 } 1850 mAudioMixer->disable(AudioMixer::MIXING); 1851 } 1852 } 1853 1854 // remove all the tracks that need to be... 1855 count = tracksToRemove->size(); 1856 if (UNLIKELY(count)) { 1857 for (size_t i=0 ; i<count ; i++) { 1858 const sp<Track>& track = tracksToRemove->itemAt(i); 1859 mActiveTracks.remove(track); 1860 if (track->mainBuffer() != mMixBuffer) { 1861 chain = getEffectChain_l(track->sessionId()); 1862 if (chain != 0) { 1863 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1864 chain->stopTrack(); 1865 } 1866 } 1867 if (track->isTerminated()) { 1868 mTracks.remove(track); 1869 deleteTrackName_l(track->mName); 1870 } 1871 } 1872 } 1873 1874 // mix buffer must be cleared if all tracks are connected to an 1875 // effect chain as in this case the mixer will not write to 1876 // mix buffer and track effects will accumulate into it 1877 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1878 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1879 } 1880 1881 return mixerStatus; 1882} 1883 1884void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1885{ 1886 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1887 this, streamType, mTracks.size()); 1888 Mutex::Autolock _l(mLock); 1889 1890 size_t size = mTracks.size(); 1891 for (size_t i = 0; i < size; i++) { 1892 sp<Track> t = mTracks[i]; 1893 if (t->type() == streamType) { 1894 t->mCblk->lock.lock(); 1895 t->mCblk->flags |= CBLK_INVALID_ON; 1896 t->mCblk->cv.signal(); 1897 t->mCblk->lock.unlock(); 1898 } 1899 } 1900} 1901 1902 1903// getTrackName_l() must be called with ThreadBase::mLock held 1904int AudioFlinger::MixerThread::getTrackName_l() 1905{ 1906 return mAudioMixer->getTrackName(); 1907} 1908 1909// deleteTrackName_l() must be called with ThreadBase::mLock held 1910void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1911{ 1912 LOGV("remove track (%d) and delete from mixer", name); 1913 mAudioMixer->deleteTrackName(name); 1914} 1915 1916// checkForNewParameters_l() must be called with ThreadBase::mLock held 1917bool AudioFlinger::MixerThread::checkForNewParameters_l() 1918{ 1919 bool reconfig = false; 1920 1921 while (!mNewParameters.isEmpty()) { 1922 status_t status = NO_ERROR; 1923 String8 keyValuePair = mNewParameters[0]; 1924 AudioParameter param = AudioParameter(keyValuePair); 1925 int value; 1926 1927 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1928 reconfig = true; 1929 } 1930 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1931 if (value != AudioSystem::PCM_16_BIT) { 1932 status = BAD_VALUE; 1933 } else { 1934 reconfig = true; 1935 } 1936 } 1937 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1938 if (value != AudioSystem::CHANNEL_OUT_STEREO) { 1939 status = BAD_VALUE; 1940 } else { 1941 reconfig = true; 1942 } 1943 } 1944 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1945 // do not accept frame count changes if tracks are open as the track buffer 1946 // size depends on frame count and correct behavior would not be garantied 1947 // if frame count is changed after track creation 1948 if (!mTracks.isEmpty()) { 1949 status = INVALID_OPERATION; 1950 } else { 1951 reconfig = true; 1952 } 1953 } 1954 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1955 // forward device change to effects that have requested to be 1956 // aware of attached audio device. 1957 mDevice = (uint32_t)value; 1958 for (size_t i = 0; i < mEffectChains.size(); i++) { 1959 mEffectChains[i]->setDevice_l(mDevice); 1960 } 1961 } 1962 1963 if (status == NO_ERROR) { 1964 status = mOutput->setParameters(keyValuePair); 1965 if (!mStandby && status == INVALID_OPERATION) { 1966 mOutput->standby(); 1967 mStandby = true; 1968 mBytesWritten = 0; 1969 status = mOutput->setParameters(keyValuePair); 1970 } 1971 if (status == NO_ERROR && reconfig) { 1972 delete mAudioMixer; 1973 readOutputParameters(); 1974 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1975 for (size_t i = 0; i < mTracks.size() ; i++) { 1976 int name = getTrackName_l(); 1977 if (name < 0) break; 1978 mTracks[i]->mName = name; 1979 // limit track sample rate to 2 x new output sample rate 1980 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1981 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 1982 } 1983 } 1984 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 1985 } 1986 } 1987 1988 mNewParameters.removeAt(0); 1989 1990 mParamStatus = status; 1991 mParamCond.signal(); 1992 mWaitWorkCV.wait(mLock); 1993 } 1994 return reconfig; 1995} 1996 1997status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 1998{ 1999 const size_t SIZE = 256; 2000 char buffer[SIZE]; 2001 String8 result; 2002 2003 PlaybackThread::dumpInternals(fd, args); 2004 2005 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2006 result.append(buffer); 2007 write(fd, result.string(), result.size()); 2008 return NO_ERROR; 2009} 2010 2011uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2012{ 2013 return (uint32_t)(mOutput->latency() * 1000) / 2; 2014} 2015 2016uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2017{ 2018 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2019} 2020 2021// ---------------------------------------------------------------------------- 2022AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2023 : PlaybackThread(audioFlinger, output, id, device) 2024{ 2025 mType = PlaybackThread::DIRECT; 2026} 2027 2028AudioFlinger::DirectOutputThread::~DirectOutputThread() 2029{ 2030} 2031 2032 2033static inline int16_t clamp16(int32_t sample) 2034{ 2035 if ((sample>>15) ^ (sample>>31)) 2036 sample = 0x7FFF ^ (sample>>31); 2037 return sample; 2038} 2039 2040static inline 2041int32_t mul(int16_t in, int16_t v) 2042{ 2043#if defined(__arm__) && !defined(__thumb__) 2044 int32_t out; 2045 asm( "smulbb %[out], %[in], %[v] \n" 2046 : [out]"=r"(out) 2047 : [in]"%r"(in), [v]"r"(v) 2048 : ); 2049 return out; 2050#else 2051 return in * int32_t(v); 2052#endif 2053} 2054 2055void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2056{ 2057 // Do not apply volume on compressed audio 2058 if (!AudioSystem::isLinearPCM(mFormat)) { 2059 return; 2060 } 2061 2062 // convert to signed 16 bit before volume calculation 2063 if (mFormat == AudioSystem::PCM_8_BIT) { 2064 size_t count = mFrameCount * mChannelCount; 2065 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2066 int16_t *dst = mMixBuffer + count-1; 2067 while(count--) { 2068 *dst-- = (int16_t)(*src--^0x80) << 8; 2069 } 2070 } 2071 2072 size_t frameCount = mFrameCount; 2073 int16_t *out = mMixBuffer; 2074 if (ramp) { 2075 if (mChannelCount == 1) { 2076 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2077 int32_t vlInc = d / (int32_t)frameCount; 2078 int32_t vl = ((int32_t)mLeftVolShort << 16); 2079 do { 2080 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2081 out++; 2082 vl += vlInc; 2083 } while (--frameCount); 2084 2085 } else { 2086 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2087 int32_t vlInc = d / (int32_t)frameCount; 2088 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2089 int32_t vrInc = d / (int32_t)frameCount; 2090 int32_t vl = ((int32_t)mLeftVolShort << 16); 2091 int32_t vr = ((int32_t)mRightVolShort << 16); 2092 do { 2093 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2094 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2095 out += 2; 2096 vl += vlInc; 2097 vr += vrInc; 2098 } while (--frameCount); 2099 } 2100 } else { 2101 if (mChannelCount == 1) { 2102 do { 2103 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2104 out++; 2105 } while (--frameCount); 2106 } else { 2107 do { 2108 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2109 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2110 out += 2; 2111 } while (--frameCount); 2112 } 2113 } 2114 2115 // convert back to unsigned 8 bit after volume calculation 2116 if (mFormat == AudioSystem::PCM_8_BIT) { 2117 size_t count = mFrameCount * mChannelCount; 2118 int16_t *src = mMixBuffer; 2119 uint8_t *dst = (uint8_t *)mMixBuffer; 2120 while(count--) { 2121 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2122 } 2123 } 2124 2125 mLeftVolShort = leftVol; 2126 mRightVolShort = rightVol; 2127} 2128 2129bool AudioFlinger::DirectOutputThread::threadLoop() 2130{ 2131 uint32_t mixerStatus = MIXER_IDLE; 2132 sp<Track> trackToRemove; 2133 sp<Track> activeTrack; 2134 nsecs_t standbyTime = systemTime(); 2135 int8_t *curBuf; 2136 size_t mixBufferSize = mFrameCount*mFrameSize; 2137 uint32_t activeSleepTime = activeSleepTimeUs(); 2138 uint32_t idleSleepTime = idleSleepTimeUs(); 2139 uint32_t sleepTime = idleSleepTime; 2140 // use shorter standby delay as on normal output to release 2141 // hardware resources as soon as possible 2142 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2143 2144 while (!exitPending()) 2145 { 2146 bool rampVolume; 2147 uint16_t leftVol; 2148 uint16_t rightVol; 2149 Vector< sp<EffectChain> > effectChains; 2150 2151 processConfigEvents(); 2152 2153 mixerStatus = MIXER_IDLE; 2154 2155 { // scope for the mLock 2156 2157 Mutex::Autolock _l(mLock); 2158 2159 if (checkForNewParameters_l()) { 2160 mixBufferSize = mFrameCount*mFrameSize; 2161 activeSleepTime = activeSleepTimeUs(); 2162 idleSleepTime = idleSleepTimeUs(); 2163 standbyDelay = microseconds(activeSleepTime*2); 2164 } 2165 2166 // put audio hardware into standby after short delay 2167 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2168 mSuspended) { 2169 // wait until we have something to do... 2170 if (!mStandby) { 2171 LOGV("Audio hardware entering standby, mixer %p\n", this); 2172 mOutput->standby(); 2173 mStandby = true; 2174 mBytesWritten = 0; 2175 } 2176 2177 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2178 // we're about to wait, flush the binder command buffer 2179 IPCThreadState::self()->flushCommands(); 2180 2181 if (exitPending()) break; 2182 2183 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2184 mWaitWorkCV.wait(mLock); 2185 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2186 2187 if (mMasterMute == false) { 2188 char value[PROPERTY_VALUE_MAX]; 2189 property_get("ro.audio.silent", value, "0"); 2190 if (atoi(value)) { 2191 LOGD("Silence is golden"); 2192 setMasterMute(true); 2193 } 2194 } 2195 2196 standbyTime = systemTime() + standbyDelay; 2197 sleepTime = idleSleepTime; 2198 continue; 2199 } 2200 } 2201 2202 effectChains = mEffectChains; 2203 2204 // find out which tracks need to be processed 2205 if (mActiveTracks.size() != 0) { 2206 sp<Track> t = mActiveTracks[0].promote(); 2207 if (t == 0) continue; 2208 2209 Track* const track = t.get(); 2210 audio_track_cblk_t* cblk = track->cblk(); 2211 2212 // The first time a track is added we wait 2213 // for all its buffers to be filled before processing it 2214 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 2215 !track->isPaused() && !track->isTerminated()) 2216 { 2217 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2218 2219 if (track->mFillingUpStatus == Track::FS_FILLED) { 2220 track->mFillingUpStatus = Track::FS_ACTIVE; 2221 mLeftVolFloat = mRightVolFloat = 0; 2222 mLeftVolShort = mRightVolShort = 0; 2223 if (track->mState == TrackBase::RESUMING) { 2224 track->mState = TrackBase::ACTIVE; 2225 rampVolume = true; 2226 } 2227 } else if (cblk->server != 0) { 2228 // If the track is stopped before the first frame was mixed, 2229 // do not apply ramp 2230 rampVolume = true; 2231 } 2232 // compute volume for this track 2233 float left, right; 2234 if (track->isMuted() || mMasterMute || track->isPausing() || 2235 mStreamTypes[track->type()].mute) { 2236 left = right = 0; 2237 if (track->isPausing()) { 2238 track->setPaused(); 2239 } 2240 } else { 2241 float typeVolume = mStreamTypes[track->type()].volume; 2242 float v = mMasterVolume * typeVolume; 2243 float v_clamped = v * cblk->volume[0]; 2244 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2245 left = v_clamped/MAX_GAIN; 2246 v_clamped = v * cblk->volume[1]; 2247 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2248 right = v_clamped/MAX_GAIN; 2249 } 2250 2251 if (left != mLeftVolFloat || right != mRightVolFloat) { 2252 mLeftVolFloat = left; 2253 mRightVolFloat = right; 2254 2255 // If audio HAL implements volume control, 2256 // force software volume to nominal value 2257 if (mOutput->setVolume(left, right) == NO_ERROR) { 2258 left = 1.0f; 2259 right = 1.0f; 2260 } 2261 2262 // Convert volumes from float to 8.24 2263 uint32_t vl = (uint32_t)(left * (1 << 24)); 2264 uint32_t vr = (uint32_t)(right * (1 << 24)); 2265 2266 // Delegate volume control to effect in track effect chain if needed 2267 // only one effect chain can be present on DirectOutputThread, so if 2268 // there is one, the track is connected to it 2269 if (!effectChains.isEmpty()) { 2270 // Do not ramp volume is volume is controlled by effect 2271 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2272 rampVolume = false; 2273 } 2274 } 2275 2276 // Convert volumes from 8.24 to 4.12 format 2277 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2278 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2279 leftVol = (uint16_t)v_clamped; 2280 v_clamped = (vr + (1 << 11)) >> 12; 2281 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2282 rightVol = (uint16_t)v_clamped; 2283 } else { 2284 leftVol = mLeftVolShort; 2285 rightVol = mRightVolShort; 2286 rampVolume = false; 2287 } 2288 2289 // reset retry count 2290 track->mRetryCount = kMaxTrackRetriesDirect; 2291 activeTrack = t; 2292 mixerStatus = MIXER_TRACKS_READY; 2293 } else { 2294 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2295 if (track->isStopped()) { 2296 track->reset(); 2297 } 2298 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2299 // We have consumed all the buffers of this track. 2300 // Remove it from the list of active tracks. 2301 trackToRemove = track; 2302 } else { 2303 // No buffers for this track. Give it a few chances to 2304 // fill a buffer, then remove it from active list. 2305 if (--(track->mRetryCount) <= 0) { 2306 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2307 trackToRemove = track; 2308 } else { 2309 mixerStatus = MIXER_TRACKS_ENABLED; 2310 } 2311 } 2312 } 2313 } 2314 2315 // remove all the tracks that need to be... 2316 if (UNLIKELY(trackToRemove != 0)) { 2317 mActiveTracks.remove(trackToRemove); 2318 if (!effectChains.isEmpty()) { 2319 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2320 trackToRemove->sessionId()); 2321 effectChains[0]->stopTrack(); 2322 } 2323 if (trackToRemove->isTerminated()) { 2324 mTracks.remove(trackToRemove); 2325 deleteTrackName_l(trackToRemove->mName); 2326 } 2327 } 2328 2329 lockEffectChains_l(effectChains); 2330 } 2331 2332 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2333 AudioBufferProvider::Buffer buffer; 2334 size_t frameCount = mFrameCount; 2335 curBuf = (int8_t *)mMixBuffer; 2336 // output audio to hardware 2337 while (frameCount) { 2338 buffer.frameCount = frameCount; 2339 activeTrack->getNextBuffer(&buffer); 2340 if (UNLIKELY(buffer.raw == 0)) { 2341 memset(curBuf, 0, frameCount * mFrameSize); 2342 break; 2343 } 2344 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2345 frameCount -= buffer.frameCount; 2346 curBuf += buffer.frameCount * mFrameSize; 2347 activeTrack->releaseBuffer(&buffer); 2348 } 2349 sleepTime = 0; 2350 standbyTime = systemTime() + standbyDelay; 2351 } else { 2352 if (sleepTime == 0) { 2353 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2354 sleepTime = activeSleepTime; 2355 } else { 2356 sleepTime = idleSleepTime; 2357 } 2358 } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { 2359 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2360 sleepTime = 0; 2361 } 2362 } 2363 2364 if (mSuspended) { 2365 sleepTime = idleSleepTime; 2366 } 2367 // sleepTime == 0 means we must write to audio hardware 2368 if (sleepTime == 0) { 2369 if (mixerStatus == MIXER_TRACKS_READY) { 2370 applyVolume(leftVol, rightVol, rampVolume); 2371 } 2372 for (size_t i = 0; i < effectChains.size(); i ++) { 2373 effectChains[i]->process_l(); 2374 } 2375 unlockEffectChains(effectChains); 2376 2377 mLastWriteTime = systemTime(); 2378 mInWrite = true; 2379 mBytesWritten += mixBufferSize; 2380 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 2381 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2382 mNumWrites++; 2383 mInWrite = false; 2384 mStandby = false; 2385 } else { 2386 unlockEffectChains(effectChains); 2387 usleep(sleepTime); 2388 } 2389 2390 // finally let go of removed track, without the lock held 2391 // since we can't guarantee the destructors won't acquire that 2392 // same lock. 2393 trackToRemove.clear(); 2394 activeTrack.clear(); 2395 2396 // Effect chains will be actually deleted here if they were removed from 2397 // mEffectChains list during mixing or effects processing 2398 effectChains.clear(); 2399 } 2400 2401 if (!mStandby) { 2402 mOutput->standby(); 2403 } 2404 2405 LOGV("DirectOutputThread %p exiting", this); 2406 return false; 2407} 2408 2409// getTrackName_l() must be called with ThreadBase::mLock held 2410int AudioFlinger::DirectOutputThread::getTrackName_l() 2411{ 2412 return 0; 2413} 2414 2415// deleteTrackName_l() must be called with ThreadBase::mLock held 2416void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2417{ 2418} 2419 2420// checkForNewParameters_l() must be called with ThreadBase::mLock held 2421bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2422{ 2423 bool reconfig = false; 2424 2425 while (!mNewParameters.isEmpty()) { 2426 status_t status = NO_ERROR; 2427 String8 keyValuePair = mNewParameters[0]; 2428 AudioParameter param = AudioParameter(keyValuePair); 2429 int value; 2430 2431 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2432 // do not accept frame count changes if tracks are open as the track buffer 2433 // size depends on frame count and correct behavior would not be garantied 2434 // if frame count is changed after track creation 2435 if (!mTracks.isEmpty()) { 2436 status = INVALID_OPERATION; 2437 } else { 2438 reconfig = true; 2439 } 2440 } 2441 if (status == NO_ERROR) { 2442 status = mOutput->setParameters(keyValuePair); 2443 if (!mStandby && status == INVALID_OPERATION) { 2444 mOutput->standby(); 2445 mStandby = true; 2446 mBytesWritten = 0; 2447 status = mOutput->setParameters(keyValuePair); 2448 } 2449 if (status == NO_ERROR && reconfig) { 2450 readOutputParameters(); 2451 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2452 } 2453 } 2454 2455 mNewParameters.removeAt(0); 2456 2457 mParamStatus = status; 2458 mParamCond.signal(); 2459 mWaitWorkCV.wait(mLock); 2460 } 2461 return reconfig; 2462} 2463 2464uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2465{ 2466 uint32_t time; 2467 if (AudioSystem::isLinearPCM(mFormat)) { 2468 time = (uint32_t)(mOutput->latency() * 1000) / 2; 2469 } else { 2470 time = 10000; 2471 } 2472 return time; 2473} 2474 2475uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2476{ 2477 uint32_t time; 2478 if (AudioSystem::isLinearPCM(mFormat)) { 2479 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2480 } else { 2481 time = 10000; 2482 } 2483 return time; 2484} 2485 2486// ---------------------------------------------------------------------------- 2487 2488AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2489 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2490{ 2491 mType = PlaybackThread::DUPLICATING; 2492 addOutputTrack(mainThread); 2493} 2494 2495AudioFlinger::DuplicatingThread::~DuplicatingThread() 2496{ 2497 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2498 mOutputTracks[i]->destroy(); 2499 } 2500 mOutputTracks.clear(); 2501} 2502 2503bool AudioFlinger::DuplicatingThread::threadLoop() 2504{ 2505 Vector< sp<Track> > tracksToRemove; 2506 uint32_t mixerStatus = MIXER_IDLE; 2507 nsecs_t standbyTime = systemTime(); 2508 size_t mixBufferSize = mFrameCount*mFrameSize; 2509 SortedVector< sp<OutputTrack> > outputTracks; 2510 uint32_t writeFrames = 0; 2511 uint32_t activeSleepTime = activeSleepTimeUs(); 2512 uint32_t idleSleepTime = idleSleepTimeUs(); 2513 uint32_t sleepTime = idleSleepTime; 2514 Vector< sp<EffectChain> > effectChains; 2515 2516 while (!exitPending()) 2517 { 2518 processConfigEvents(); 2519 2520 mixerStatus = MIXER_IDLE; 2521 { // scope for the mLock 2522 2523 Mutex::Autolock _l(mLock); 2524 2525 if (checkForNewParameters_l()) { 2526 mixBufferSize = mFrameCount*mFrameSize; 2527 updateWaitTime(); 2528 activeSleepTime = activeSleepTimeUs(); 2529 idleSleepTime = idleSleepTimeUs(); 2530 } 2531 2532 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2533 2534 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2535 outputTracks.add(mOutputTracks[i]); 2536 } 2537 2538 // put audio hardware into standby after short delay 2539 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2540 mSuspended) { 2541 if (!mStandby) { 2542 for (size_t i = 0; i < outputTracks.size(); i++) { 2543 outputTracks[i]->stop(); 2544 } 2545 mStandby = true; 2546 mBytesWritten = 0; 2547 } 2548 2549 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2550 // we're about to wait, flush the binder command buffer 2551 IPCThreadState::self()->flushCommands(); 2552 outputTracks.clear(); 2553 2554 if (exitPending()) break; 2555 2556 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2557 mWaitWorkCV.wait(mLock); 2558 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2559 if (mMasterMute == false) { 2560 char value[PROPERTY_VALUE_MAX]; 2561 property_get("ro.audio.silent", value, "0"); 2562 if (atoi(value)) { 2563 LOGD("Silence is golden"); 2564 setMasterMute(true); 2565 } 2566 } 2567 2568 standbyTime = systemTime() + kStandbyTimeInNsecs; 2569 sleepTime = idleSleepTime; 2570 continue; 2571 } 2572 } 2573 2574 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2575 2576 // prevent any changes in effect chain list and in each effect chain 2577 // during mixing and effect process as the audio buffers could be deleted 2578 // or modified if an effect is created or deleted 2579 lockEffectChains_l(effectChains); 2580 } 2581 2582 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2583 // mix buffers... 2584 if (outputsReady(outputTracks)) { 2585 mAudioMixer->process(); 2586 } else { 2587 memset(mMixBuffer, 0, mixBufferSize); 2588 } 2589 sleepTime = 0; 2590 writeFrames = mFrameCount; 2591 } else { 2592 if (sleepTime == 0) { 2593 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2594 sleepTime = activeSleepTime; 2595 } else { 2596 sleepTime = idleSleepTime; 2597 } 2598 } else if (mBytesWritten != 0) { 2599 // flush remaining overflow buffers in output tracks 2600 for (size_t i = 0; i < outputTracks.size(); i++) { 2601 if (outputTracks[i]->isActive()) { 2602 sleepTime = 0; 2603 writeFrames = 0; 2604 memset(mMixBuffer, 0, mixBufferSize); 2605 break; 2606 } 2607 } 2608 } 2609 } 2610 2611 if (mSuspended) { 2612 sleepTime = idleSleepTime; 2613 } 2614 // sleepTime == 0 means we must write to audio hardware 2615 if (sleepTime == 0) { 2616 for (size_t i = 0; i < effectChains.size(); i ++) { 2617 effectChains[i]->process_l(); 2618 } 2619 // enable changes in effect chain 2620 unlockEffectChains(effectChains); 2621 2622 standbyTime = systemTime() + kStandbyTimeInNsecs; 2623 for (size_t i = 0; i < outputTracks.size(); i++) { 2624 outputTracks[i]->write(mMixBuffer, writeFrames); 2625 } 2626 mStandby = false; 2627 mBytesWritten += mixBufferSize; 2628 } else { 2629 // enable changes in effect chain 2630 unlockEffectChains(effectChains); 2631 usleep(sleepTime); 2632 } 2633 2634 // finally let go of all our tracks, without the lock held 2635 // since we can't guarantee the destructors won't acquire that 2636 // same lock. 2637 tracksToRemove.clear(); 2638 outputTracks.clear(); 2639 2640 // Effect chains will be actually deleted here if they were removed from 2641 // mEffectChains list during mixing or effects processing 2642 effectChains.clear(); 2643 } 2644 2645 return false; 2646} 2647 2648void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2649{ 2650 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2651 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2652 this, 2653 mSampleRate, 2654 mFormat, 2655 mChannelCount, 2656 frameCount); 2657 if (outputTrack->cblk() != NULL) { 2658 thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); 2659 mOutputTracks.add(outputTrack); 2660 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2661 updateWaitTime(); 2662 } 2663} 2664 2665void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2666{ 2667 Mutex::Autolock _l(mLock); 2668 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2669 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2670 mOutputTracks[i]->destroy(); 2671 mOutputTracks.removeAt(i); 2672 updateWaitTime(); 2673 return; 2674 } 2675 } 2676 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2677} 2678 2679void AudioFlinger::DuplicatingThread::updateWaitTime() 2680{ 2681 mWaitTimeMs = UINT_MAX; 2682 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2683 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2684 if (strong != NULL) { 2685 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2686 if (waitTimeMs < mWaitTimeMs) { 2687 mWaitTimeMs = waitTimeMs; 2688 } 2689 } 2690 } 2691} 2692 2693 2694bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2695{ 2696 for (size_t i = 0; i < outputTracks.size(); i++) { 2697 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2698 if (thread == 0) { 2699 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2700 return false; 2701 } 2702 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2703 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2704 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2705 return false; 2706 } 2707 } 2708 return true; 2709} 2710 2711uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2712{ 2713 return (mWaitTimeMs * 1000) / 2; 2714} 2715 2716// ---------------------------------------------------------------------------- 2717 2718// TrackBase constructor must be called with AudioFlinger::mLock held 2719AudioFlinger::ThreadBase::TrackBase::TrackBase( 2720 const wp<ThreadBase>& thread, 2721 const sp<Client>& client, 2722 uint32_t sampleRate, 2723 int format, 2724 int channelCount, 2725 int frameCount, 2726 uint32_t flags, 2727 const sp<IMemory>& sharedBuffer, 2728 int sessionId) 2729 : RefBase(), 2730 mThread(thread), 2731 mClient(client), 2732 mCblk(0), 2733 mFrameCount(0), 2734 mState(IDLE), 2735 mClientTid(-1), 2736 mFormat(format), 2737 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2738 mSessionId(sessionId) 2739{ 2740 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2741 2742 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2743 size_t size = sizeof(audio_track_cblk_t); 2744 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2745 if (sharedBuffer == 0) { 2746 size += bufferSize; 2747 } 2748 2749 if (client != NULL) { 2750 mCblkMemory = client->heap()->allocate(size); 2751 if (mCblkMemory != 0) { 2752 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2753 if (mCblk) { // construct the shared structure in-place. 2754 new(mCblk) audio_track_cblk_t(); 2755 // clear all buffers 2756 mCblk->frameCount = frameCount; 2757 mCblk->sampleRate = sampleRate; 2758 mCblk->channelCount = (uint8_t)channelCount; 2759 if (sharedBuffer == 0) { 2760 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2761 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2762 // Force underrun condition to avoid false underrun callback until first data is 2763 // written to buffer 2764 mCblk->flags = CBLK_UNDERRUN_ON; 2765 } else { 2766 mBuffer = sharedBuffer->pointer(); 2767 } 2768 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2769 } 2770 } else { 2771 LOGE("not enough memory for AudioTrack size=%u", size); 2772 client->heap()->dump("AudioTrack"); 2773 return; 2774 } 2775 } else { 2776 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2777 if (mCblk) { // construct the shared structure in-place. 2778 new(mCblk) audio_track_cblk_t(); 2779 // clear all buffers 2780 mCblk->frameCount = frameCount; 2781 mCblk->sampleRate = sampleRate; 2782 mCblk->channelCount = (uint8_t)channelCount; 2783 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2784 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2785 // Force underrun condition to avoid false underrun callback until first data is 2786 // written to buffer 2787 mCblk->flags = CBLK_UNDERRUN_ON; 2788 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2789 } 2790 } 2791} 2792 2793AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2794{ 2795 if (mCblk) { 2796 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2797 if (mClient == NULL) { 2798 delete mCblk; 2799 } 2800 } 2801 mCblkMemory.clear(); // and free the shared memory 2802 if (mClient != NULL) { 2803 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2804 mClient.clear(); 2805 } 2806} 2807 2808void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2809{ 2810 buffer->raw = 0; 2811 mFrameCount = buffer->frameCount; 2812 step(); 2813 buffer->frameCount = 0; 2814} 2815 2816bool AudioFlinger::ThreadBase::TrackBase::step() { 2817 bool result; 2818 audio_track_cblk_t* cblk = this->cblk(); 2819 2820 result = cblk->stepServer(mFrameCount); 2821 if (!result) { 2822 LOGV("stepServer failed acquiring cblk mutex"); 2823 mFlags |= STEPSERVER_FAILED; 2824 } 2825 return result; 2826} 2827 2828void AudioFlinger::ThreadBase::TrackBase::reset() { 2829 audio_track_cblk_t* cblk = this->cblk(); 2830 2831 cblk->user = 0; 2832 cblk->server = 0; 2833 cblk->userBase = 0; 2834 cblk->serverBase = 0; 2835 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2836 LOGV("TrackBase::reset"); 2837} 2838 2839sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2840{ 2841 return mCblkMemory; 2842} 2843 2844int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2845 return (int)mCblk->sampleRate; 2846} 2847 2848int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2849 return (int)mCblk->channelCount; 2850} 2851 2852void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2853 audio_track_cblk_t* cblk = this->cblk(); 2854 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2855 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2856 2857 // Check validity of returned pointer in case the track control block would have been corrupted. 2858 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2859 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2860 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2861 server %d, serverBase %d, user %d, userBase %d, channelCount %d", 2862 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2863 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); 2864 return 0; 2865 } 2866 2867 return bufferStart; 2868} 2869 2870// ---------------------------------------------------------------------------- 2871 2872// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2873AudioFlinger::PlaybackThread::Track::Track( 2874 const wp<ThreadBase>& thread, 2875 const sp<Client>& client, 2876 int streamType, 2877 uint32_t sampleRate, 2878 int format, 2879 int channelCount, 2880 int frameCount, 2881 const sp<IMemory>& sharedBuffer, 2882 int sessionId) 2883 : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), 2884 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0) 2885{ 2886 if (mCblk != NULL) { 2887 sp<ThreadBase> baseThread = thread.promote(); 2888 if (baseThread != 0) { 2889 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2890 mName = playbackThread->getTrackName_l(); 2891 mMainBuffer = playbackThread->mixBuffer(); 2892 } 2893 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2894 if (mName < 0) { 2895 LOGE("no more track names available"); 2896 } 2897 mVolume[0] = 1.0f; 2898 mVolume[1] = 1.0f; 2899 mStreamType = streamType; 2900 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2901 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2902 mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); 2903 } 2904} 2905 2906AudioFlinger::PlaybackThread::Track::~Track() 2907{ 2908 LOGV("PlaybackThread::Track destructor"); 2909 sp<ThreadBase> thread = mThread.promote(); 2910 if (thread != 0) { 2911 Mutex::Autolock _l(thread->mLock); 2912 mState = TERMINATED; 2913 } 2914} 2915 2916void AudioFlinger::PlaybackThread::Track::destroy() 2917{ 2918 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2919 // by removing it from mTracks vector, so there is a risk that this Tracks's 2920 // desctructor is called. As the destructor needs to lock mLock, 2921 // we must acquire a strong reference on this Track before locking mLock 2922 // here so that the destructor is called only when exiting this function. 2923 // On the other hand, as long as Track::destroy() is only called by 2924 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2925 // this Track with its member mTrack. 2926 sp<Track> keep(this); 2927 { // scope for mLock 2928 sp<ThreadBase> thread = mThread.promote(); 2929 if (thread != 0) { 2930 if (!isOutputTrack()) { 2931 if (mState == ACTIVE || mState == RESUMING) { 2932 AudioSystem::stopOutput(thread->id(), 2933 (AudioSystem::stream_type)mStreamType, 2934 mSessionId); 2935 } 2936 AudioSystem::releaseOutput(thread->id()); 2937 } 2938 Mutex::Autolock _l(thread->mLock); 2939 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2940 playbackThread->destroyTrack_l(this); 2941 } 2942 } 2943} 2944 2945void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2946{ 2947 snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 2948 mName - AudioMixer::TRACK0, 2949 (mClient == NULL) ? getpid() : mClient->pid(), 2950 mStreamType, 2951 mFormat, 2952 mCblk->channelCount, 2953 mSessionId, 2954 mFrameCount, 2955 mState, 2956 mMute, 2957 mFillingUpStatus, 2958 mCblk->sampleRate, 2959 mCblk->volume[0], 2960 mCblk->volume[1], 2961 mCblk->server, 2962 mCblk->user, 2963 (int)mMainBuffer, 2964 (int)mAuxBuffer); 2965} 2966 2967status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 2968{ 2969 audio_track_cblk_t* cblk = this->cblk(); 2970 uint32_t framesReady; 2971 uint32_t framesReq = buffer->frameCount; 2972 2973 // Check if last stepServer failed, try to step now 2974 if (mFlags & TrackBase::STEPSERVER_FAILED) { 2975 if (!step()) goto getNextBuffer_exit; 2976 LOGV("stepServer recovered"); 2977 mFlags &= ~TrackBase::STEPSERVER_FAILED; 2978 } 2979 2980 framesReady = cblk->framesReady(); 2981 2982 if (LIKELY(framesReady)) { 2983 uint32_t s = cblk->server; 2984 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 2985 2986 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 2987 if (framesReq > framesReady) { 2988 framesReq = framesReady; 2989 } 2990 if (s + framesReq > bufferEnd) { 2991 framesReq = bufferEnd - s; 2992 } 2993 2994 buffer->raw = getBuffer(s, framesReq); 2995 if (buffer->raw == 0) goto getNextBuffer_exit; 2996 2997 buffer->frameCount = framesReq; 2998 return NO_ERROR; 2999 } 3000 3001getNextBuffer_exit: 3002 buffer->raw = 0; 3003 buffer->frameCount = 0; 3004 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3005 return NOT_ENOUGH_DATA; 3006} 3007 3008bool AudioFlinger::PlaybackThread::Track::isReady() const { 3009 if (mFillingUpStatus != FS_FILLING) return true; 3010 3011 if (mCblk->framesReady() >= mCblk->frameCount || 3012 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3013 mFillingUpStatus = FS_FILLED; 3014 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3015 return true; 3016 } 3017 return false; 3018} 3019 3020status_t AudioFlinger::PlaybackThread::Track::start() 3021{ 3022 status_t status = NO_ERROR; 3023 LOGV("start(%d), calling thread %d session %d", 3024 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3025 sp<ThreadBase> thread = mThread.promote(); 3026 if (thread != 0) { 3027 Mutex::Autolock _l(thread->mLock); 3028 int state = mState; 3029 // here the track could be either new, or restarted 3030 // in both cases "unstop" the track 3031 if (mState == PAUSED) { 3032 mState = TrackBase::RESUMING; 3033 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3034 } else { 3035 mState = TrackBase::ACTIVE; 3036 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3037 } 3038 3039 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3040 thread->mLock.unlock(); 3041 status = AudioSystem::startOutput(thread->id(), 3042 (AudioSystem::stream_type)mStreamType, 3043 mSessionId); 3044 thread->mLock.lock(); 3045 } 3046 if (status == NO_ERROR) { 3047 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3048 playbackThread->addTrack_l(this); 3049 } else { 3050 mState = state; 3051 } 3052 } else { 3053 status = BAD_VALUE; 3054 } 3055 return status; 3056} 3057 3058void AudioFlinger::PlaybackThread::Track::stop() 3059{ 3060 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3061 sp<ThreadBase> thread = mThread.promote(); 3062 if (thread != 0) { 3063 Mutex::Autolock _l(thread->mLock); 3064 int state = mState; 3065 if (mState > STOPPED) { 3066 mState = STOPPED; 3067 // If the track is not active (PAUSED and buffers full), flush buffers 3068 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3069 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3070 reset(); 3071 } 3072 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3073 } 3074 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3075 thread->mLock.unlock(); 3076 AudioSystem::stopOutput(thread->id(), 3077 (AudioSystem::stream_type)mStreamType, 3078 mSessionId); 3079 thread->mLock.lock(); 3080 } 3081 } 3082} 3083 3084void AudioFlinger::PlaybackThread::Track::pause() 3085{ 3086 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3087 sp<ThreadBase> thread = mThread.promote(); 3088 if (thread != 0) { 3089 Mutex::Autolock _l(thread->mLock); 3090 if (mState == ACTIVE || mState == RESUMING) { 3091 mState = PAUSING; 3092 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3093 if (!isOutputTrack()) { 3094 thread->mLock.unlock(); 3095 AudioSystem::stopOutput(thread->id(), 3096 (AudioSystem::stream_type)mStreamType, 3097 mSessionId); 3098 thread->mLock.lock(); 3099 } 3100 } 3101 } 3102} 3103 3104void AudioFlinger::PlaybackThread::Track::flush() 3105{ 3106 LOGV("flush(%d)", mName); 3107 sp<ThreadBase> thread = mThread.promote(); 3108 if (thread != 0) { 3109 Mutex::Autolock _l(thread->mLock); 3110 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3111 return; 3112 } 3113 // No point remaining in PAUSED state after a flush => go to 3114 // STOPPED state 3115 mState = STOPPED; 3116 3117 mCblk->lock.lock(); 3118 // NOTE: reset() will reset cblk->user and cblk->server with 3119 // the risk that at the same time, the AudioMixer is trying to read 3120 // data. In this case, getNextBuffer() would return a NULL pointer 3121 // as audio buffer => the AudioMixer code MUST always test that pointer 3122 // returned by getNextBuffer() is not NULL! 3123 reset(); 3124 mCblk->lock.unlock(); 3125 } 3126} 3127 3128void AudioFlinger::PlaybackThread::Track::reset() 3129{ 3130 // Do not reset twice to avoid discarding data written just after a flush and before 3131 // the audioflinger thread detects the track is stopped. 3132 if (!mResetDone) { 3133 TrackBase::reset(); 3134 // Force underrun condition to avoid false underrun callback until first data is 3135 // written to buffer 3136 mCblk->flags |= CBLK_UNDERRUN_ON; 3137 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3138 mFillingUpStatus = FS_FILLING; 3139 mResetDone = true; 3140 } 3141} 3142 3143void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3144{ 3145 mMute = muted; 3146} 3147 3148void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3149{ 3150 mVolume[0] = left; 3151 mVolume[1] = right; 3152} 3153 3154status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3155{ 3156 status_t status = DEAD_OBJECT; 3157 sp<ThreadBase> thread = mThread.promote(); 3158 if (thread != 0) { 3159 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3160 status = playbackThread->attachAuxEffect(this, EffectId); 3161 } 3162 return status; 3163} 3164 3165void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3166{ 3167 mAuxEffectId = EffectId; 3168 mAuxBuffer = buffer; 3169} 3170 3171// ---------------------------------------------------------------------------- 3172 3173// RecordTrack constructor must be called with AudioFlinger::mLock held 3174AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3175 const wp<ThreadBase>& thread, 3176 const sp<Client>& client, 3177 uint32_t sampleRate, 3178 int format, 3179 int channelCount, 3180 int frameCount, 3181 uint32_t flags, 3182 int sessionId) 3183 : TrackBase(thread, client, sampleRate, format, 3184 channelCount, frameCount, flags, 0, sessionId), 3185 mOverflow(false) 3186{ 3187 if (mCblk != NULL) { 3188 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3189 if (format == AudioSystem::PCM_16_BIT) { 3190 mCblk->frameSize = channelCount * sizeof(int16_t); 3191 } else if (format == AudioSystem::PCM_8_BIT) { 3192 mCblk->frameSize = channelCount * sizeof(int8_t); 3193 } else { 3194 mCblk->frameSize = sizeof(int8_t); 3195 } 3196 } 3197} 3198 3199AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3200{ 3201 sp<ThreadBase> thread = mThread.promote(); 3202 if (thread != 0) { 3203 AudioSystem::releaseInput(thread->id()); 3204 } 3205} 3206 3207status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3208{ 3209 audio_track_cblk_t* cblk = this->cblk(); 3210 uint32_t framesAvail; 3211 uint32_t framesReq = buffer->frameCount; 3212 3213 // Check if last stepServer failed, try to step now 3214 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3215 if (!step()) goto getNextBuffer_exit; 3216 LOGV("stepServer recovered"); 3217 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3218 } 3219 3220 framesAvail = cblk->framesAvailable_l(); 3221 3222 if (LIKELY(framesAvail)) { 3223 uint32_t s = cblk->server; 3224 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3225 3226 if (framesReq > framesAvail) { 3227 framesReq = framesAvail; 3228 } 3229 if (s + framesReq > bufferEnd) { 3230 framesReq = bufferEnd - s; 3231 } 3232 3233 buffer->raw = getBuffer(s, framesReq); 3234 if (buffer->raw == 0) goto getNextBuffer_exit; 3235 3236 buffer->frameCount = framesReq; 3237 return NO_ERROR; 3238 } 3239 3240getNextBuffer_exit: 3241 buffer->raw = 0; 3242 buffer->frameCount = 0; 3243 return NOT_ENOUGH_DATA; 3244} 3245 3246status_t AudioFlinger::RecordThread::RecordTrack::start() 3247{ 3248 sp<ThreadBase> thread = mThread.promote(); 3249 if (thread != 0) { 3250 RecordThread *recordThread = (RecordThread *)thread.get(); 3251 return recordThread->start(this); 3252 } else { 3253 return BAD_VALUE; 3254 } 3255} 3256 3257void AudioFlinger::RecordThread::RecordTrack::stop() 3258{ 3259 sp<ThreadBase> thread = mThread.promote(); 3260 if (thread != 0) { 3261 RecordThread *recordThread = (RecordThread *)thread.get(); 3262 recordThread->stop(this); 3263 TrackBase::reset(); 3264 // Force overerrun condition to avoid false overrun callback until first data is 3265 // read from buffer 3266 mCblk->flags |= CBLK_UNDERRUN_ON; 3267 } 3268} 3269 3270void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3271{ 3272 snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", 3273 (mClient == NULL) ? getpid() : mClient->pid(), 3274 mFormat, 3275 mCblk->channelCount, 3276 mSessionId, 3277 mFrameCount, 3278 mState, 3279 mCblk->sampleRate, 3280 mCblk->server, 3281 mCblk->user); 3282} 3283 3284 3285// ---------------------------------------------------------------------------- 3286 3287AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3288 const wp<ThreadBase>& thread, 3289 DuplicatingThread *sourceThread, 3290 uint32_t sampleRate, 3291 int format, 3292 int channelCount, 3293 int frameCount) 3294 : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), 3295 mActive(false), mSourceThread(sourceThread) 3296{ 3297 3298 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3299 if (mCblk != NULL) { 3300 mCblk->flags |= CBLK_DIRECTION_OUT; 3301 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3302 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3303 mOutBuffer.frameCount = 0; 3304 playbackThread->mTracks.add(this); 3305 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", 3306 mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); 3307 } else { 3308 LOGW("Error creating output track on thread %p", playbackThread); 3309 } 3310} 3311 3312AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3313{ 3314 clearBufferQueue(); 3315} 3316 3317status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3318{ 3319 status_t status = Track::start(); 3320 if (status != NO_ERROR) { 3321 return status; 3322 } 3323 3324 mActive = true; 3325 mRetryCount = 127; 3326 return status; 3327} 3328 3329void AudioFlinger::PlaybackThread::OutputTrack::stop() 3330{ 3331 Track::stop(); 3332 clearBufferQueue(); 3333 mOutBuffer.frameCount = 0; 3334 mActive = false; 3335} 3336 3337bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3338{ 3339 Buffer *pInBuffer; 3340 Buffer inBuffer; 3341 uint32_t channelCount = mCblk->channelCount; 3342 bool outputBufferFull = false; 3343 inBuffer.frameCount = frames; 3344 inBuffer.i16 = data; 3345 3346 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3347 3348 if (!mActive && frames != 0) { 3349 start(); 3350 sp<ThreadBase> thread = mThread.promote(); 3351 if (thread != 0) { 3352 MixerThread *mixerThread = (MixerThread *)thread.get(); 3353 if (mCblk->frameCount > frames){ 3354 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3355 uint32_t startFrames = (mCblk->frameCount - frames); 3356 pInBuffer = new Buffer; 3357 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3358 pInBuffer->frameCount = startFrames; 3359 pInBuffer->i16 = pInBuffer->mBuffer; 3360 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3361 mBufferQueue.add(pInBuffer); 3362 } else { 3363 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3364 } 3365 } 3366 } 3367 } 3368 3369 while (waitTimeLeftMs) { 3370 // First write pending buffers, then new data 3371 if (mBufferQueue.size()) { 3372 pInBuffer = mBufferQueue.itemAt(0); 3373 } else { 3374 pInBuffer = &inBuffer; 3375 } 3376 3377 if (pInBuffer->frameCount == 0) { 3378 break; 3379 } 3380 3381 if (mOutBuffer.frameCount == 0) { 3382 mOutBuffer.frameCount = pInBuffer->frameCount; 3383 nsecs_t startTime = systemTime(); 3384 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3385 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3386 outputBufferFull = true; 3387 break; 3388 } 3389 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3390 if (waitTimeLeftMs >= waitTimeMs) { 3391 waitTimeLeftMs -= waitTimeMs; 3392 } else { 3393 waitTimeLeftMs = 0; 3394 } 3395 } 3396 3397 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3398 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3399 mCblk->stepUser(outFrames); 3400 pInBuffer->frameCount -= outFrames; 3401 pInBuffer->i16 += outFrames * channelCount; 3402 mOutBuffer.frameCount -= outFrames; 3403 mOutBuffer.i16 += outFrames * channelCount; 3404 3405 if (pInBuffer->frameCount == 0) { 3406 if (mBufferQueue.size()) { 3407 mBufferQueue.removeAt(0); 3408 delete [] pInBuffer->mBuffer; 3409 delete pInBuffer; 3410 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3411 } else { 3412 break; 3413 } 3414 } 3415 } 3416 3417 // If we could not write all frames, allocate a buffer and queue it for next time. 3418 if (inBuffer.frameCount) { 3419 sp<ThreadBase> thread = mThread.promote(); 3420 if (thread != 0 && !thread->standby()) { 3421 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3422 pInBuffer = new Buffer; 3423 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3424 pInBuffer->frameCount = inBuffer.frameCount; 3425 pInBuffer->i16 = pInBuffer->mBuffer; 3426 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3427 mBufferQueue.add(pInBuffer); 3428 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3429 } else { 3430 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3431 } 3432 } 3433 } 3434 3435 // Calling write() with a 0 length buffer, means that no more data will be written: 3436 // If no more buffers are pending, fill output track buffer to make sure it is started 3437 // by output mixer. 3438 if (frames == 0 && mBufferQueue.size() == 0) { 3439 if (mCblk->user < mCblk->frameCount) { 3440 frames = mCblk->frameCount - mCblk->user; 3441 pInBuffer = new Buffer; 3442 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3443 pInBuffer->frameCount = frames; 3444 pInBuffer->i16 = pInBuffer->mBuffer; 3445 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3446 mBufferQueue.add(pInBuffer); 3447 } else if (mActive) { 3448 stop(); 3449 } 3450 } 3451 3452 return outputBufferFull; 3453} 3454 3455status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3456{ 3457 int active; 3458 status_t result; 3459 audio_track_cblk_t* cblk = mCblk; 3460 uint32_t framesReq = buffer->frameCount; 3461 3462// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3463 buffer->frameCount = 0; 3464 3465 uint32_t framesAvail = cblk->framesAvailable(); 3466 3467 3468 if (framesAvail == 0) { 3469 Mutex::Autolock _l(cblk->lock); 3470 goto start_loop_here; 3471 while (framesAvail == 0) { 3472 active = mActive; 3473 if (UNLIKELY(!active)) { 3474 LOGV("Not active and NO_MORE_BUFFERS"); 3475 return AudioTrack::NO_MORE_BUFFERS; 3476 } 3477 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3478 if (result != NO_ERROR) { 3479 return AudioTrack::NO_MORE_BUFFERS; 3480 } 3481 // read the server count again 3482 start_loop_here: 3483 framesAvail = cblk->framesAvailable_l(); 3484 } 3485 } 3486 3487// if (framesAvail < framesReq) { 3488// return AudioTrack::NO_MORE_BUFFERS; 3489// } 3490 3491 if (framesReq > framesAvail) { 3492 framesReq = framesAvail; 3493 } 3494 3495 uint32_t u = cblk->user; 3496 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3497 3498 if (u + framesReq > bufferEnd) { 3499 framesReq = bufferEnd - u; 3500 } 3501 3502 buffer->frameCount = framesReq; 3503 buffer->raw = (void *)cblk->buffer(u); 3504 return NO_ERROR; 3505} 3506 3507 3508void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3509{ 3510 size_t size = mBufferQueue.size(); 3511 Buffer *pBuffer; 3512 3513 for (size_t i = 0; i < size; i++) { 3514 pBuffer = mBufferQueue.itemAt(i); 3515 delete [] pBuffer->mBuffer; 3516 delete pBuffer; 3517 } 3518 mBufferQueue.clear(); 3519} 3520 3521// ---------------------------------------------------------------------------- 3522 3523AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3524 : RefBase(), 3525 mAudioFlinger(audioFlinger), 3526 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3527 mPid(pid) 3528{ 3529 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3530} 3531 3532// Client destructor must be called with AudioFlinger::mLock held 3533AudioFlinger::Client::~Client() 3534{ 3535 mAudioFlinger->removeClient_l(mPid); 3536} 3537 3538const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3539{ 3540 return mMemoryDealer; 3541} 3542 3543// ---------------------------------------------------------------------------- 3544 3545AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3546 const sp<IAudioFlingerClient>& client, 3547 pid_t pid) 3548 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3549{ 3550} 3551 3552AudioFlinger::NotificationClient::~NotificationClient() 3553{ 3554 mClient.clear(); 3555} 3556 3557void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3558{ 3559 sp<NotificationClient> keep(this); 3560 { 3561 mAudioFlinger->removeNotificationClient(mPid); 3562 } 3563} 3564 3565// ---------------------------------------------------------------------------- 3566 3567AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3568 : BnAudioTrack(), 3569 mTrack(track) 3570{ 3571} 3572 3573AudioFlinger::TrackHandle::~TrackHandle() { 3574 // just stop the track on deletion, associated resources 3575 // will be freed from the main thread once all pending buffers have 3576 // been played. Unless it's not in the active track list, in which 3577 // case we free everything now... 3578 mTrack->destroy(); 3579} 3580 3581status_t AudioFlinger::TrackHandle::start() { 3582 return mTrack->start(); 3583} 3584 3585void AudioFlinger::TrackHandle::stop() { 3586 mTrack->stop(); 3587} 3588 3589void AudioFlinger::TrackHandle::flush() { 3590 mTrack->flush(); 3591} 3592 3593void AudioFlinger::TrackHandle::mute(bool e) { 3594 mTrack->mute(e); 3595} 3596 3597void AudioFlinger::TrackHandle::pause() { 3598 mTrack->pause(); 3599} 3600 3601void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3602 mTrack->setVolume(left, right); 3603} 3604 3605sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3606 return mTrack->getCblk(); 3607} 3608 3609status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3610{ 3611 return mTrack->attachAuxEffect(EffectId); 3612} 3613 3614status_t AudioFlinger::TrackHandle::onTransact( 3615 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3616{ 3617 return BnAudioTrack::onTransact(code, data, reply, flags); 3618} 3619 3620// ---------------------------------------------------------------------------- 3621 3622sp<IAudioRecord> AudioFlinger::openRecord( 3623 pid_t pid, 3624 int input, 3625 uint32_t sampleRate, 3626 int format, 3627 int channelCount, 3628 int frameCount, 3629 uint32_t flags, 3630 int *sessionId, 3631 status_t *status) 3632{ 3633 sp<RecordThread::RecordTrack> recordTrack; 3634 sp<RecordHandle> recordHandle; 3635 sp<Client> client; 3636 wp<Client> wclient; 3637 status_t lStatus; 3638 RecordThread *thread; 3639 size_t inFrameCount; 3640 int lSessionId; 3641 3642 // check calling permissions 3643 if (!recordingAllowed()) { 3644 lStatus = PERMISSION_DENIED; 3645 goto Exit; 3646 } 3647 3648 // add client to list 3649 { // scope for mLock 3650 Mutex::Autolock _l(mLock); 3651 thread = checkRecordThread_l(input); 3652 if (thread == NULL) { 3653 lStatus = BAD_VALUE; 3654 goto Exit; 3655 } 3656 3657 wclient = mClients.valueFor(pid); 3658 if (wclient != NULL) { 3659 client = wclient.promote(); 3660 } else { 3661 client = new Client(this, pid); 3662 mClients.add(pid, client); 3663 } 3664 3665 // If no audio session id is provided, create one here 3666 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 3667 lSessionId = *sessionId; 3668 } else { 3669 lSessionId = nextUniqueId(); 3670 if (sessionId != NULL) { 3671 *sessionId = lSessionId; 3672 } 3673 } 3674 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3675 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3676 format, channelCount, frameCount, flags, lSessionId); 3677 } 3678 if (recordTrack->getCblk() == NULL) { 3679 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3680 // destructor is called by the TrackBase destructor with mLock held 3681 client.clear(); 3682 recordTrack.clear(); 3683 lStatus = NO_MEMORY; 3684 goto Exit; 3685 } 3686 3687 // return to handle to client 3688 recordHandle = new RecordHandle(recordTrack); 3689 lStatus = NO_ERROR; 3690 3691Exit: 3692 if (status) { 3693 *status = lStatus; 3694 } 3695 return recordHandle; 3696} 3697 3698// ---------------------------------------------------------------------------- 3699 3700AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3701 : BnAudioRecord(), 3702 mRecordTrack(recordTrack) 3703{ 3704} 3705 3706AudioFlinger::RecordHandle::~RecordHandle() { 3707 stop(); 3708} 3709 3710status_t AudioFlinger::RecordHandle::start() { 3711 LOGV("RecordHandle::start()"); 3712 return mRecordTrack->start(); 3713} 3714 3715void AudioFlinger::RecordHandle::stop() { 3716 LOGV("RecordHandle::stop()"); 3717 mRecordTrack->stop(); 3718} 3719 3720sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3721 return mRecordTrack->getCblk(); 3722} 3723 3724status_t AudioFlinger::RecordHandle::onTransact( 3725 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3726{ 3727 return BnAudioRecord::onTransact(code, data, reply, flags); 3728} 3729 3730// ---------------------------------------------------------------------------- 3731 3732AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : 3733 ThreadBase(audioFlinger, id), 3734 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3735{ 3736 mReqChannelCount = AudioSystem::popCount(channels); 3737 mReqSampleRate = sampleRate; 3738 readInputParameters(); 3739} 3740 3741 3742AudioFlinger::RecordThread::~RecordThread() 3743{ 3744 delete[] mRsmpInBuffer; 3745 if (mResampler != 0) { 3746 delete mResampler; 3747 delete[] mRsmpOutBuffer; 3748 } 3749} 3750 3751void AudioFlinger::RecordThread::onFirstRef() 3752{ 3753 const size_t SIZE = 256; 3754 char buffer[SIZE]; 3755 3756 snprintf(buffer, SIZE, "Record Thread %p", this); 3757 3758 run(buffer, PRIORITY_URGENT_AUDIO); 3759} 3760 3761bool AudioFlinger::RecordThread::threadLoop() 3762{ 3763 AudioBufferProvider::Buffer buffer; 3764 sp<RecordTrack> activeTrack; 3765 3766 // start recording 3767 while (!exitPending()) { 3768 3769 processConfigEvents(); 3770 3771 { // scope for mLock 3772 Mutex::Autolock _l(mLock); 3773 checkForNewParameters_l(); 3774 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3775 if (!mStandby) { 3776 mInput->standby(); 3777 mStandby = true; 3778 } 3779 3780 if (exitPending()) break; 3781 3782 LOGV("RecordThread: loop stopping"); 3783 // go to sleep 3784 mWaitWorkCV.wait(mLock); 3785 LOGV("RecordThread: loop starting"); 3786 continue; 3787 } 3788 if (mActiveTrack != 0) { 3789 if (mActiveTrack->mState == TrackBase::PAUSING) { 3790 if (!mStandby) { 3791 mInput->standby(); 3792 mStandby = true; 3793 } 3794 mActiveTrack.clear(); 3795 mStartStopCond.broadcast(); 3796 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3797 if (mReqChannelCount != mActiveTrack->channelCount()) { 3798 mActiveTrack.clear(); 3799 mStartStopCond.broadcast(); 3800 } else if (mBytesRead != 0) { 3801 // record start succeeds only if first read from audio input 3802 // succeeds 3803 if (mBytesRead > 0) { 3804 mActiveTrack->mState = TrackBase::ACTIVE; 3805 } else { 3806 mActiveTrack.clear(); 3807 } 3808 mStartStopCond.broadcast(); 3809 } 3810 mStandby = false; 3811 } 3812 } 3813 } 3814 3815 if (mActiveTrack != 0) { 3816 if (mActiveTrack->mState != TrackBase::ACTIVE && 3817 mActiveTrack->mState != TrackBase::RESUMING) { 3818 usleep(5000); 3819 continue; 3820 } 3821 buffer.frameCount = mFrameCount; 3822 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3823 size_t framesOut = buffer.frameCount; 3824 if (mResampler == 0) { 3825 // no resampling 3826 while (framesOut) { 3827 size_t framesIn = mFrameCount - mRsmpInIndex; 3828 if (framesIn) { 3829 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3830 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3831 if (framesIn > framesOut) 3832 framesIn = framesOut; 3833 mRsmpInIndex += framesIn; 3834 framesOut -= framesIn; 3835 if ((int)mChannelCount == mReqChannelCount || 3836 mFormat != AudioSystem::PCM_16_BIT) { 3837 memcpy(dst, src, framesIn * mFrameSize); 3838 } else { 3839 int16_t *src16 = (int16_t *)src; 3840 int16_t *dst16 = (int16_t *)dst; 3841 if (mChannelCount == 1) { 3842 while (framesIn--) { 3843 *dst16++ = *src16; 3844 *dst16++ = *src16++; 3845 } 3846 } else { 3847 while (framesIn--) { 3848 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3849 src16 += 2; 3850 } 3851 } 3852 } 3853 } 3854 if (framesOut && mFrameCount == mRsmpInIndex) { 3855 if (framesOut == mFrameCount && 3856 ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { 3857 mBytesRead = mInput->read(buffer.raw, mInputBytes); 3858 framesOut = 0; 3859 } else { 3860 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 3861 mRsmpInIndex = 0; 3862 } 3863 if (mBytesRead < 0) { 3864 LOGE("Error reading audio input"); 3865 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3866 // Force input into standby so that it tries to 3867 // recover at next read attempt 3868 mInput->standby(); 3869 usleep(5000); 3870 } 3871 mRsmpInIndex = mFrameCount; 3872 framesOut = 0; 3873 buffer.frameCount = 0; 3874 } 3875 } 3876 } 3877 } else { 3878 // resampling 3879 3880 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3881 // alter output frame count as if we were expecting stereo samples 3882 if (mChannelCount == 1 && mReqChannelCount == 1) { 3883 framesOut >>= 1; 3884 } 3885 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3886 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3887 // are 32 bit aligned which should be always true. 3888 if (mChannelCount == 2 && mReqChannelCount == 1) { 3889 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3890 // the resampler always outputs stereo samples: do post stereo to mono conversion 3891 int16_t *src = (int16_t *)mRsmpOutBuffer; 3892 int16_t *dst = buffer.i16; 3893 while (framesOut--) { 3894 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3895 src += 2; 3896 } 3897 } else { 3898 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3899 } 3900 3901 } 3902 mActiveTrack->releaseBuffer(&buffer); 3903 mActiveTrack->overflow(); 3904 } 3905 // client isn't retrieving buffers fast enough 3906 else { 3907 if (!mActiveTrack->setOverflow()) 3908 LOGW("RecordThread: buffer overflow"); 3909 // Release the processor for a while before asking for a new buffer. 3910 // This will give the application more chance to read from the buffer and 3911 // clear the overflow. 3912 usleep(5000); 3913 } 3914 } 3915 } 3916 3917 if (!mStandby) { 3918 mInput->standby(); 3919 } 3920 mActiveTrack.clear(); 3921 3922 mStartStopCond.broadcast(); 3923 3924 LOGV("RecordThread %p exiting", this); 3925 return false; 3926} 3927 3928status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3929{ 3930 LOGV("RecordThread::start"); 3931 sp <ThreadBase> strongMe = this; 3932 status_t status = NO_ERROR; 3933 { 3934 AutoMutex lock(&mLock); 3935 if (mActiveTrack != 0) { 3936 if (recordTrack != mActiveTrack.get()) { 3937 status = -EBUSY; 3938 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3939 mActiveTrack->mState = TrackBase::ACTIVE; 3940 } 3941 return status; 3942 } 3943 3944 recordTrack->mState = TrackBase::IDLE; 3945 mActiveTrack = recordTrack; 3946 mLock.unlock(); 3947 status_t status = AudioSystem::startInput(mId); 3948 mLock.lock(); 3949 if (status != NO_ERROR) { 3950 mActiveTrack.clear(); 3951 return status; 3952 } 3953 mActiveTrack->mState = TrackBase::RESUMING; 3954 mRsmpInIndex = mFrameCount; 3955 mBytesRead = 0; 3956 // signal thread to start 3957 LOGV("Signal record thread"); 3958 mWaitWorkCV.signal(); 3959 // do not wait for mStartStopCond if exiting 3960 if (mExiting) { 3961 mActiveTrack.clear(); 3962 status = INVALID_OPERATION; 3963 goto startError; 3964 } 3965 mStartStopCond.wait(mLock); 3966 if (mActiveTrack == 0) { 3967 LOGV("Record failed to start"); 3968 status = BAD_VALUE; 3969 goto startError; 3970 } 3971 LOGV("Record started OK"); 3972 return status; 3973 } 3974startError: 3975 AudioSystem::stopInput(mId); 3976 return status; 3977} 3978 3979void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 3980 LOGV("RecordThread::stop"); 3981 sp <ThreadBase> strongMe = this; 3982 { 3983 AutoMutex lock(&mLock); 3984 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 3985 mActiveTrack->mState = TrackBase::PAUSING; 3986 // do not wait for mStartStopCond if exiting 3987 if (mExiting) { 3988 return; 3989 } 3990 mStartStopCond.wait(mLock); 3991 // if we have been restarted, recordTrack == mActiveTrack.get() here 3992 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 3993 mLock.unlock(); 3994 AudioSystem::stopInput(mId); 3995 mLock.lock(); 3996 LOGV("Record stopped OK"); 3997 } 3998 } 3999 } 4000} 4001 4002status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4003{ 4004 const size_t SIZE = 256; 4005 char buffer[SIZE]; 4006 String8 result; 4007 pid_t pid = 0; 4008 4009 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4010 result.append(buffer); 4011 4012 if (mActiveTrack != 0) { 4013 result.append("Active Track:\n"); 4014 result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); 4015 mActiveTrack->dump(buffer, SIZE); 4016 result.append(buffer); 4017 4018 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4019 result.append(buffer); 4020 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4021 result.append(buffer); 4022 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4023 result.append(buffer); 4024 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4025 result.append(buffer); 4026 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4027 result.append(buffer); 4028 4029 4030 } else { 4031 result.append("No record client\n"); 4032 } 4033 write(fd, result.string(), result.size()); 4034 4035 dumpBase(fd, args); 4036 4037 return NO_ERROR; 4038} 4039 4040status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4041{ 4042 size_t framesReq = buffer->frameCount; 4043 size_t framesReady = mFrameCount - mRsmpInIndex; 4044 int channelCount; 4045 4046 if (framesReady == 0) { 4047 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 4048 if (mBytesRead < 0) { 4049 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4050 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4051 // Force input into standby so that it tries to 4052 // recover at next read attempt 4053 mInput->standby(); 4054 usleep(5000); 4055 } 4056 buffer->raw = 0; 4057 buffer->frameCount = 0; 4058 return NOT_ENOUGH_DATA; 4059 } 4060 mRsmpInIndex = 0; 4061 framesReady = mFrameCount; 4062 } 4063 4064 if (framesReq > framesReady) { 4065 framesReq = framesReady; 4066 } 4067 4068 if (mChannelCount == 1 && mReqChannelCount == 2) { 4069 channelCount = 1; 4070 } else { 4071 channelCount = 2; 4072 } 4073 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4074 buffer->frameCount = framesReq; 4075 return NO_ERROR; 4076} 4077 4078void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4079{ 4080 mRsmpInIndex += buffer->frameCount; 4081 buffer->frameCount = 0; 4082} 4083 4084bool AudioFlinger::RecordThread::checkForNewParameters_l() 4085{ 4086 bool reconfig = false; 4087 4088 while (!mNewParameters.isEmpty()) { 4089 status_t status = NO_ERROR; 4090 String8 keyValuePair = mNewParameters[0]; 4091 AudioParameter param = AudioParameter(keyValuePair); 4092 int value; 4093 int reqFormat = mFormat; 4094 int reqSamplingRate = mReqSampleRate; 4095 int reqChannelCount = mReqChannelCount; 4096 4097 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4098 reqSamplingRate = value; 4099 reconfig = true; 4100 } 4101 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4102 reqFormat = value; 4103 reconfig = true; 4104 } 4105 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4106 reqChannelCount = AudioSystem::popCount(value); 4107 reconfig = true; 4108 } 4109 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4110 // do not accept frame count changes if tracks are open as the track buffer 4111 // size depends on frame count and correct behavior would not be garantied 4112 // if frame count is changed after track creation 4113 if (mActiveTrack != 0) { 4114 status = INVALID_OPERATION; 4115 } else { 4116 reconfig = true; 4117 } 4118 } 4119 if (status == NO_ERROR) { 4120 status = mInput->setParameters(keyValuePair); 4121 if (status == INVALID_OPERATION) { 4122 mInput->standby(); 4123 status = mInput->setParameters(keyValuePair); 4124 } 4125 if (reconfig) { 4126 if (status == BAD_VALUE && 4127 reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && 4128 ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && 4129 (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { 4130 status = NO_ERROR; 4131 } 4132 if (status == NO_ERROR) { 4133 readInputParameters(); 4134 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4135 } 4136 } 4137 } 4138 4139 mNewParameters.removeAt(0); 4140 4141 mParamStatus = status; 4142 mParamCond.signal(); 4143 mWaitWorkCV.wait(mLock); 4144 } 4145 return reconfig; 4146} 4147 4148String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4149{ 4150 return mInput->getParameters(keys); 4151} 4152 4153void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4154 AudioSystem::OutputDescriptor desc; 4155 void *param2 = 0; 4156 4157 switch (event) { 4158 case AudioSystem::INPUT_OPENED: 4159 case AudioSystem::INPUT_CONFIG_CHANGED: 4160 desc.channels = mChannels; 4161 desc.samplingRate = mSampleRate; 4162 desc.format = mFormat; 4163 desc.frameCount = mFrameCount; 4164 desc.latency = 0; 4165 param2 = &desc; 4166 break; 4167 4168 case AudioSystem::INPUT_CLOSED: 4169 default: 4170 break; 4171 } 4172 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4173} 4174 4175void AudioFlinger::RecordThread::readInputParameters() 4176{ 4177 if (mRsmpInBuffer) delete mRsmpInBuffer; 4178 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4179 if (mResampler) delete mResampler; 4180 mResampler = 0; 4181 4182 mSampleRate = mInput->sampleRate(); 4183 mChannels = mInput->channels(); 4184 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 4185 mFormat = mInput->format(); 4186 mFrameSize = (uint16_t)mInput->frameSize(); 4187 mInputBytes = mInput->bufferSize(); 4188 mFrameCount = mInputBytes / mFrameSize; 4189 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4190 4191 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4192 { 4193 int channelCount; 4194 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4195 // stereo to mono post process as the resampler always outputs stereo. 4196 if (mChannelCount == 1 && mReqChannelCount == 2) { 4197 channelCount = 1; 4198 } else { 4199 channelCount = 2; 4200 } 4201 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4202 mResampler->setSampleRate(mSampleRate); 4203 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4204 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4205 4206 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4207 if (mChannelCount == 1 && mReqChannelCount == 1) { 4208 mFrameCount >>= 1; 4209 } 4210 4211 } 4212 mRsmpInIndex = mFrameCount; 4213} 4214 4215unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4216{ 4217 return mInput->getInputFramesLost(); 4218} 4219 4220// ---------------------------------------------------------------------------- 4221 4222int AudioFlinger::openOutput(uint32_t *pDevices, 4223 uint32_t *pSamplingRate, 4224 uint32_t *pFormat, 4225 uint32_t *pChannels, 4226 uint32_t *pLatencyMs, 4227 uint32_t flags) 4228{ 4229 status_t status; 4230 PlaybackThread *thread = NULL; 4231 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4232 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4233 uint32_t format = pFormat ? *pFormat : 0; 4234 uint32_t channels = pChannels ? *pChannels : 0; 4235 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4236 4237 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4238 pDevices ? *pDevices : 0, 4239 samplingRate, 4240 format, 4241 channels, 4242 flags); 4243 4244 if (pDevices == NULL || *pDevices == 0) { 4245 return 0; 4246 } 4247 Mutex::Autolock _l(mLock); 4248 4249 AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, 4250 (int *)&format, 4251 &channels, 4252 &samplingRate, 4253 &status); 4254 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4255 output, 4256 samplingRate, 4257 format, 4258 channels, 4259 status); 4260 4261 mHardwareStatus = AUDIO_HW_IDLE; 4262 if (output != 0) { 4263 int id = nextUniqueId(); 4264 if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || 4265 (format != AudioSystem::PCM_16_BIT) || 4266 (channels != AudioSystem::CHANNEL_OUT_STEREO)) { 4267 thread = new DirectOutputThread(this, output, id, *pDevices); 4268 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4269 } else { 4270 thread = new MixerThread(this, output, id, *pDevices); 4271 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4272 4273#ifdef LVMX 4274 unsigned bitsPerSample = 4275 (format == AudioSystem::PCM_16_BIT) ? 16 : 4276 ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); 4277 unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; 4278 int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); 4279 4280 LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); 4281 LifeVibes::setDevice(audioOutputType, *pDevices); 4282#endif 4283 4284 } 4285 mPlaybackThreads.add(id, thread); 4286 4287 if (pSamplingRate) *pSamplingRate = samplingRate; 4288 if (pFormat) *pFormat = format; 4289 if (pChannels) *pChannels = channels; 4290 if (pLatencyMs) *pLatencyMs = thread->latency(); 4291 4292 // notify client processes of the new output creation 4293 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4294 return id; 4295 } 4296 4297 return 0; 4298} 4299 4300int AudioFlinger::openDuplicateOutput(int output1, int output2) 4301{ 4302 Mutex::Autolock _l(mLock); 4303 MixerThread *thread1 = checkMixerThread_l(output1); 4304 MixerThread *thread2 = checkMixerThread_l(output2); 4305 4306 if (thread1 == NULL || thread2 == NULL) { 4307 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4308 return 0; 4309 } 4310 4311 int id = nextUniqueId(); 4312 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4313 thread->addOutputTrack(thread2); 4314 mPlaybackThreads.add(id, thread); 4315 // notify client processes of the new output creation 4316 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4317 return id; 4318} 4319 4320status_t AudioFlinger::closeOutput(int output) 4321{ 4322 // keep strong reference on the playback thread so that 4323 // it is not destroyed while exit() is executed 4324 sp <PlaybackThread> thread; 4325 { 4326 Mutex::Autolock _l(mLock); 4327 thread = checkPlaybackThread_l(output); 4328 if (thread == NULL) { 4329 return BAD_VALUE; 4330 } 4331 4332 LOGV("closeOutput() %d", output); 4333 4334 if (thread->type() == PlaybackThread::MIXER) { 4335 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4336 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4337 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4338 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4339 } 4340 } 4341 } 4342 void *param2 = 0; 4343 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4344 mPlaybackThreads.removeItem(output); 4345 } 4346 thread->exit(); 4347 4348 if (thread->type() != PlaybackThread::DUPLICATING) { 4349 mAudioHardware->closeOutputStream(thread->getOutput()); 4350 } 4351 return NO_ERROR; 4352} 4353 4354status_t AudioFlinger::suspendOutput(int output) 4355{ 4356 Mutex::Autolock _l(mLock); 4357 PlaybackThread *thread = checkPlaybackThread_l(output); 4358 4359 if (thread == NULL) { 4360 return BAD_VALUE; 4361 } 4362 4363 LOGV("suspendOutput() %d", output); 4364 thread->suspend(); 4365 4366 return NO_ERROR; 4367} 4368 4369status_t AudioFlinger::restoreOutput(int output) 4370{ 4371 Mutex::Autolock _l(mLock); 4372 PlaybackThread *thread = checkPlaybackThread_l(output); 4373 4374 if (thread == NULL) { 4375 return BAD_VALUE; 4376 } 4377 4378 LOGV("restoreOutput() %d", output); 4379 4380 thread->restore(); 4381 4382 return NO_ERROR; 4383} 4384 4385int AudioFlinger::openInput(uint32_t *pDevices, 4386 uint32_t *pSamplingRate, 4387 uint32_t *pFormat, 4388 uint32_t *pChannels, 4389 uint32_t acoustics) 4390{ 4391 status_t status; 4392 RecordThread *thread = NULL; 4393 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4394 uint32_t format = pFormat ? *pFormat : 0; 4395 uint32_t channels = pChannels ? *pChannels : 0; 4396 uint32_t reqSamplingRate = samplingRate; 4397 uint32_t reqFormat = format; 4398 uint32_t reqChannels = channels; 4399 4400 if (pDevices == NULL || *pDevices == 0) { 4401 return 0; 4402 } 4403 Mutex::Autolock _l(mLock); 4404 4405 AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, 4406 (int *)&format, 4407 &channels, 4408 &samplingRate, 4409 &status, 4410 (AudioSystem::audio_in_acoustics)acoustics); 4411 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4412 input, 4413 samplingRate, 4414 format, 4415 channels, 4416 acoustics, 4417 status); 4418 4419 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4420 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4421 // or stereo to mono conversions on 16 bit PCM inputs. 4422 if (input == 0 && status == BAD_VALUE && 4423 reqFormat == format && format == AudioSystem::PCM_16_BIT && 4424 (samplingRate <= 2 * reqSamplingRate) && 4425 (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { 4426 LOGV("openInput() reopening with proposed sampling rate and channels"); 4427 input = mAudioHardware->openInputStream(*pDevices, 4428 (int *)&format, 4429 &channels, 4430 &samplingRate, 4431 &status, 4432 (AudioSystem::audio_in_acoustics)acoustics); 4433 } 4434 4435 if (input != 0) { 4436 int id = nextUniqueId(); 4437 // Start record thread 4438 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4439 mRecordThreads.add(id, thread); 4440 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4441 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4442 if (pFormat) *pFormat = format; 4443 if (pChannels) *pChannels = reqChannels; 4444 4445 input->standby(); 4446 4447 // notify client processes of the new input creation 4448 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4449 return id; 4450 } 4451 4452 return 0; 4453} 4454 4455status_t AudioFlinger::closeInput(int input) 4456{ 4457 // keep strong reference on the record thread so that 4458 // it is not destroyed while exit() is executed 4459 sp <RecordThread> thread; 4460 { 4461 Mutex::Autolock _l(mLock); 4462 thread = checkRecordThread_l(input); 4463 if (thread == NULL) { 4464 return BAD_VALUE; 4465 } 4466 4467 LOGV("closeInput() %d", input); 4468 void *param2 = 0; 4469 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4470 mRecordThreads.removeItem(input); 4471 } 4472 thread->exit(); 4473 4474 mAudioHardware->closeInputStream(thread->getInput()); 4475 4476 return NO_ERROR; 4477} 4478 4479status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4480{ 4481 Mutex::Autolock _l(mLock); 4482 MixerThread *dstThread = checkMixerThread_l(output); 4483 if (dstThread == NULL) { 4484 LOGW("setStreamOutput() bad output id %d", output); 4485 return BAD_VALUE; 4486 } 4487 4488 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4489 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4490 4491 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4492 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4493 if (thread != dstThread && 4494 thread->type() != PlaybackThread::DIRECT) { 4495 MixerThread *srcThread = (MixerThread *)thread; 4496 srcThread->invalidateTracks(stream); 4497 } 4498 } 4499 4500 return NO_ERROR; 4501} 4502 4503 4504int AudioFlinger::newAudioSessionId() 4505{ 4506 return nextUniqueId(); 4507} 4508 4509// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4510AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4511{ 4512 PlaybackThread *thread = NULL; 4513 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4514 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4515 } 4516 return thread; 4517} 4518 4519// checkMixerThread_l() must be called with AudioFlinger::mLock held 4520AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4521{ 4522 PlaybackThread *thread = checkPlaybackThread_l(output); 4523 if (thread != NULL) { 4524 if (thread->type() == PlaybackThread::DIRECT) { 4525 thread = NULL; 4526 } 4527 } 4528 return (MixerThread *)thread; 4529} 4530 4531// checkRecordThread_l() must be called with AudioFlinger::mLock held 4532AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4533{ 4534 RecordThread *thread = NULL; 4535 if (mRecordThreads.indexOfKey(input) >= 0) { 4536 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4537 } 4538 return thread; 4539} 4540 4541int AudioFlinger::nextUniqueId() 4542{ 4543 return android_atomic_inc(&mNextUniqueId); 4544} 4545 4546// ---------------------------------------------------------------------------- 4547// Effect management 4548// ---------------------------------------------------------------------------- 4549 4550 4551status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) 4552{ 4553 // check calling permissions 4554 if (!settingsAllowed()) { 4555 return PERMISSION_DENIED; 4556 } 4557 // only allow libraries loaded from /system/lib/soundfx for now 4558 if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { 4559 return PERMISSION_DENIED; 4560 } 4561 4562 Mutex::Autolock _l(mLock); 4563 return EffectLoadLibrary(libPath, handle); 4564} 4565 4566status_t AudioFlinger::unloadEffectLibrary(int handle) 4567{ 4568 // check calling permissions 4569 if (!settingsAllowed()) { 4570 return PERMISSION_DENIED; 4571 } 4572 4573 Mutex::Autolock _l(mLock); 4574 return EffectUnloadLibrary(handle); 4575} 4576 4577status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4578{ 4579 Mutex::Autolock _l(mLock); 4580 return EffectQueryNumberEffects(numEffects); 4581} 4582 4583status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4584{ 4585 Mutex::Autolock _l(mLock); 4586 return EffectQueryEffect(index, descriptor); 4587} 4588 4589status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4590{ 4591 Mutex::Autolock _l(mLock); 4592 return EffectGetDescriptor(pUuid, descriptor); 4593} 4594 4595 4596// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4597static const effect_uuid_t VISUALIZATION_UUID_ = 4598 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4599 4600sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4601 effect_descriptor_t *pDesc, 4602 const sp<IEffectClient>& effectClient, 4603 int32_t priority, 4604 int output, 4605 int sessionId, 4606 status_t *status, 4607 int *id, 4608 int *enabled) 4609{ 4610 status_t lStatus = NO_ERROR; 4611 sp<EffectHandle> handle; 4612 effect_interface_t itfe; 4613 effect_descriptor_t desc; 4614 sp<Client> client; 4615 wp<Client> wclient; 4616 4617 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4618 pid, effectClient.get(), priority, sessionId, output); 4619 4620 if (pDesc == NULL) { 4621 lStatus = BAD_VALUE; 4622 goto Exit; 4623 } 4624 4625 { 4626 Mutex::Autolock _l(mLock); 4627 4628 // check recording permission for visualizer 4629 if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4630 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) { 4631 if (!recordingAllowed()) { 4632 lStatus = PERMISSION_DENIED; 4633 goto Exit; 4634 } 4635 } 4636 4637 if (!EffectIsNullUuid(&pDesc->uuid)) { 4638 // if uuid is specified, request effect descriptor 4639 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4640 if (lStatus < 0) { 4641 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4642 goto Exit; 4643 } 4644 } else { 4645 // if uuid is not specified, look for an available implementation 4646 // of the required type in effect factory 4647 if (EffectIsNullUuid(&pDesc->type)) { 4648 LOGW("createEffect() no effect type"); 4649 lStatus = BAD_VALUE; 4650 goto Exit; 4651 } 4652 uint32_t numEffects = 0; 4653 effect_descriptor_t d; 4654 bool found = false; 4655 4656 lStatus = EffectQueryNumberEffects(&numEffects); 4657 if (lStatus < 0) { 4658 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4659 goto Exit; 4660 } 4661 for (uint32_t i = 0; i < numEffects; i++) { 4662 lStatus = EffectQueryEffect(i, &desc); 4663 if (lStatus < 0) { 4664 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4665 continue; 4666 } 4667 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4668 // If matching type found save effect descriptor. If the session is 4669 // 0 and the effect is not auxiliary, continue enumeration in case 4670 // an auxiliary version of this effect type is available 4671 found = true; 4672 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4673 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX || 4674 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4675 break; 4676 } 4677 } 4678 } 4679 if (!found) { 4680 lStatus = BAD_VALUE; 4681 LOGW("createEffect() effect not found"); 4682 goto Exit; 4683 } 4684 // For same effect type, chose auxiliary version over insert version if 4685 // connect to output mix (Compliance to OpenSL ES) 4686 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && 4687 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4688 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4689 } 4690 } 4691 4692 // Do not allow auxiliary effects on a session different from 0 (output mix) 4693 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX && 4694 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4695 lStatus = INVALID_OPERATION; 4696 goto Exit; 4697 } 4698 4699 // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects 4700 // that can only be created by audio policy manager (running in same process) 4701 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && 4702 getpid() != IPCThreadState::self()->getCallingPid()) { 4703 lStatus = INVALID_OPERATION; 4704 goto Exit; 4705 } 4706 4707 // return effect descriptor 4708 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4709 4710 // If output is not specified try to find a matching audio session ID in one of the 4711 // output threads. 4712 // TODO: allow attachment of effect to inputs 4713 if (output == 0) { 4714 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) { 4715 // output must be specified by AudioPolicyManager when using session 4716 // AudioSystem::SESSION_OUTPUT_STAGE 4717 lStatus = BAD_VALUE; 4718 goto Exit; 4719 } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 4720 output = AudioSystem::getOutputForEffect(&desc); 4721 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4722 } else { 4723 // look for the thread where the specified audio session is present 4724 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4725 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4726 output = mPlaybackThreads.keyAt(i); 4727 break; 4728 } 4729 } 4730 // If no output thread contains the requested session ID, default to 4731 // first output. The effect chain will be moved to the correct output 4732 // thread when a track with the same session ID is created 4733 if (output == 0 && mPlaybackThreads.size()) { 4734 output = mPlaybackThreads.keyAt(0); 4735 } 4736 } 4737 } 4738 PlaybackThread *thread = checkPlaybackThread_l(output); 4739 if (thread == NULL) { 4740 LOGE("createEffect() unknown output thread"); 4741 lStatus = BAD_VALUE; 4742 goto Exit; 4743 } 4744 4745 wclient = mClients.valueFor(pid); 4746 4747 if (wclient != NULL) { 4748 client = wclient.promote(); 4749 } else { 4750 client = new Client(this, pid); 4751 mClients.add(pid, client); 4752 } 4753 4754 // create effect on selected output trhead 4755 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4756 &desc, enabled, &lStatus); 4757 if (handle != 0 && id != NULL) { 4758 *id = handle->id(); 4759 } 4760 } 4761 4762Exit: 4763 if(status) { 4764 *status = lStatus; 4765 } 4766 return handle; 4767} 4768 4769status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4770{ 4771 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4772 session, srcOutput, dstOutput); 4773 Mutex::Autolock _l(mLock); 4774 if (srcOutput == dstOutput) { 4775 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4776 return NO_ERROR; 4777 } 4778 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4779 if (srcThread == NULL) { 4780 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4781 return BAD_VALUE; 4782 } 4783 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4784 if (dstThread == NULL) { 4785 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4786 return BAD_VALUE; 4787 } 4788 4789 Mutex::Autolock _dl(dstThread->mLock); 4790 Mutex::Autolock _sl(srcThread->mLock); 4791 moveEffectChain_l(session, srcThread, dstThread, false); 4792 4793 return NO_ERROR; 4794} 4795 4796// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4797status_t AudioFlinger::moveEffectChain_l(int session, 4798 AudioFlinger::PlaybackThread *srcThread, 4799 AudioFlinger::PlaybackThread *dstThread, 4800 bool reRegister) 4801{ 4802 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4803 session, srcThread, dstThread); 4804 4805 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4806 if (chain == 0) { 4807 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4808 session, srcThread); 4809 return INVALID_OPERATION; 4810 } 4811 4812 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 4813 // so that a new chain is created with correct parameters when first effect is added. This is 4814 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4815 // removed. 4816 srcThread->removeEffectChain_l(chain); 4817 4818 // transfer all effects one by one so that new effect chain is created on new thread with 4819 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4820 int dstOutput = dstThread->id(); 4821 sp<EffectChain> dstChain; 4822 uint32_t strategy; 4823 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4824 while (effect != 0) { 4825 srcThread->removeEffect_l(effect); 4826 dstThread->addEffect_l(effect); 4827 // if the move request is not received from audio policy manager, the effect must be 4828 // re-registered with the new strategy and output 4829 if (dstChain == 0) { 4830 dstChain = effect->chain().promote(); 4831 if (dstChain == 0) { 4832 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 4833 srcThread->addEffect_l(effect); 4834 return NO_INIT; 4835 } 4836 strategy = dstChain->strategy(); 4837 } 4838 if (reRegister) { 4839 AudioSystem::unregisterEffect(effect->id()); 4840 AudioSystem::registerEffect(&effect->desc(), 4841 dstOutput, 4842 strategy, 4843 session, 4844 effect->id()); 4845 } 4846 effect = chain->getEffectFromId_l(0); 4847 } 4848 4849 return NO_ERROR; 4850} 4851 4852// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4853sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4854 const sp<AudioFlinger::Client>& client, 4855 const sp<IEffectClient>& effectClient, 4856 int32_t priority, 4857 int sessionId, 4858 effect_descriptor_t *desc, 4859 int *enabled, 4860 status_t *status 4861 ) 4862{ 4863 sp<EffectModule> effect; 4864 sp<EffectHandle> handle; 4865 status_t lStatus; 4866 sp<Track> track; 4867 sp<EffectChain> chain; 4868 bool chainCreated = false; 4869 bool effectCreated = false; 4870 bool effectRegistered = false; 4871 4872 if (mOutput == 0) { 4873 LOGW("createEffect_l() Audio driver not initialized."); 4874 lStatus = NO_INIT; 4875 goto Exit; 4876 } 4877 4878 // Do not allow auxiliary effect on session other than 0 4879 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4880 sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 4881 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4882 desc->name, sessionId); 4883 lStatus = BAD_VALUE; 4884 goto Exit; 4885 } 4886 4887 // Do not allow effects with session ID 0 on direct output or duplicating threads 4888 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4889 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) { 4890 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4891 desc->name, sessionId); 4892 lStatus = BAD_VALUE; 4893 goto Exit; 4894 } 4895 4896 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4897 4898 { // scope for mLock 4899 Mutex::Autolock _l(mLock); 4900 4901 // check for existing effect chain with the requested audio session 4902 chain = getEffectChain_l(sessionId); 4903 if (chain == 0) { 4904 // create a new chain for this session 4905 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4906 chain = new EffectChain(this, sessionId); 4907 addEffectChain_l(chain); 4908 chain->setStrategy(getStrategyForSession_l(sessionId)); 4909 chainCreated = true; 4910 } else { 4911 effect = chain->getEffectFromDesc_l(desc); 4912 } 4913 4914 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4915 4916 if (effect == 0) { 4917 int id = mAudioFlinger->nextUniqueId(); 4918 // Check CPU and memory usage 4919 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4920 if (lStatus != NO_ERROR) { 4921 goto Exit; 4922 } 4923 effectRegistered = true; 4924 // create a new effect module if none present in the chain 4925 effect = new EffectModule(this, chain, desc, id, sessionId); 4926 lStatus = effect->status(); 4927 if (lStatus != NO_ERROR) { 4928 goto Exit; 4929 } 4930 lStatus = chain->addEffect_l(effect); 4931 if (lStatus != NO_ERROR) { 4932 goto Exit; 4933 } 4934 effectCreated = true; 4935 4936 effect->setDevice(mDevice); 4937 effect->setMode(mAudioFlinger->getMode()); 4938 } 4939 // create effect handle and connect it to effect module 4940 handle = new EffectHandle(effect, client, effectClient, priority); 4941 lStatus = effect->addHandle(handle); 4942 if (enabled) { 4943 *enabled = (int)effect->isEnabled(); 4944 } 4945 } 4946 4947Exit: 4948 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 4949 Mutex::Autolock _l(mLock); 4950 if (effectCreated) { 4951 chain->removeEffect_l(effect); 4952 } 4953 if (effectRegistered) { 4954 AudioSystem::unregisterEffect(effect->id()); 4955 } 4956 if (chainCreated) { 4957 removeEffectChain_l(chain); 4958 } 4959 handle.clear(); 4960 } 4961 4962 if(status) { 4963 *status = lStatus; 4964 } 4965 return handle; 4966} 4967 4968// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 4969// PlaybackThread::mLock held 4970status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 4971{ 4972 // check for existing effect chain with the requested audio session 4973 int sessionId = effect->sessionId(); 4974 sp<EffectChain> chain = getEffectChain_l(sessionId); 4975 bool chainCreated = false; 4976 4977 if (chain == 0) { 4978 // create a new chain for this session 4979 LOGV("addEffect_l() new effect chain for session %d", sessionId); 4980 chain = new EffectChain(this, sessionId); 4981 addEffectChain_l(chain); 4982 chain->setStrategy(getStrategyForSession_l(sessionId)); 4983 chainCreated = true; 4984 } 4985 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 4986 4987 if (chain->getEffectFromId_l(effect->id()) != 0) { 4988 LOGW("addEffect_l() %p effect %s already present in chain %p", 4989 this, effect->desc().name, chain.get()); 4990 return BAD_VALUE; 4991 } 4992 4993 status_t status = chain->addEffect_l(effect); 4994 if (status != NO_ERROR) { 4995 if (chainCreated) { 4996 removeEffectChain_l(chain); 4997 } 4998 return status; 4999 } 5000 5001 effect->setDevice(mDevice); 5002 effect->setMode(mAudioFlinger->getMode()); 5003 return NO_ERROR; 5004} 5005 5006void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 5007 5008 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5009 effect_descriptor_t desc = effect->desc(); 5010 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5011 detachAuxEffect_l(effect->id()); 5012 } 5013 5014 sp<EffectChain> chain = effect->chain().promote(); 5015 if (chain != 0) { 5016 // remove effect chain if removing last effect 5017 if (chain->removeEffect_l(effect) == 0) { 5018 removeEffectChain_l(chain); 5019 } 5020 } else { 5021 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5022 } 5023} 5024 5025void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 5026 const wp<EffectHandle>& handle) { 5027 Mutex::Autolock _l(mLock); 5028 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5029 // delete the effect module if removing last handle on it 5030 if (effect->removeHandle(handle) == 0) { 5031 removeEffect_l(effect); 5032 AudioSystem::unregisterEffect(effect->id()); 5033 } 5034} 5035 5036status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5037{ 5038 int session = chain->sessionId(); 5039 int16_t *buffer = mMixBuffer; 5040 bool ownsBuffer = false; 5041 5042 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5043 if (session > 0) { 5044 // Only one effect chain can be present in direct output thread and it uses 5045 // the mix buffer as input 5046 if (mType != DIRECT) { 5047 size_t numSamples = mFrameCount * mChannelCount; 5048 buffer = new int16_t[numSamples]; 5049 memset(buffer, 0, numSamples * sizeof(int16_t)); 5050 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5051 ownsBuffer = true; 5052 } 5053 5054 // Attach all tracks with same session ID to this chain. 5055 for (size_t i = 0; i < mTracks.size(); ++i) { 5056 sp<Track> track = mTracks[i]; 5057 if (session == track->sessionId()) { 5058 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5059 track->setMainBuffer(buffer); 5060 } 5061 } 5062 5063 // indicate all active tracks in the chain 5064 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5065 sp<Track> track = mActiveTracks[i].promote(); 5066 if (track == 0) continue; 5067 if (session == track->sessionId()) { 5068 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5069 chain->startTrack(); 5070 } 5071 } 5072 } 5073 5074 chain->setInBuffer(buffer, ownsBuffer); 5075 chain->setOutBuffer(mMixBuffer); 5076 // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect 5077 // chains list in order to be processed last as it contains output stage effects 5078 // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before 5079 // session AudioSystem::SESSION_OUTPUT_STAGE to be processed 5080 // after track specific effects and before output stage 5081 // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and 5082 // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX 5083 // Effect chain for other sessions are inserted at beginning of effect 5084 // chains list to be processed before output mix effects. Relative order between other 5085 // sessions is not important 5086 size_t size = mEffectChains.size(); 5087 size_t i = 0; 5088 for (i = 0; i < size; i++) { 5089 if (mEffectChains[i]->sessionId() < session) break; 5090 } 5091 mEffectChains.insertAt(chain, i); 5092 5093 return NO_ERROR; 5094} 5095 5096size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5097{ 5098 int session = chain->sessionId(); 5099 5100 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5101 5102 for (size_t i = 0; i < mEffectChains.size(); i++) { 5103 if (chain == mEffectChains[i]) { 5104 mEffectChains.removeAt(i); 5105 // detach all tracks with same session ID from this chain 5106 for (size_t i = 0; i < mTracks.size(); ++i) { 5107 sp<Track> track = mTracks[i]; 5108 if (session == track->sessionId()) { 5109 track->setMainBuffer(mMixBuffer); 5110 } 5111 } 5112 break; 5113 } 5114 } 5115 return mEffectChains.size(); 5116} 5117 5118void AudioFlinger::PlaybackThread::lockEffectChains_l( 5119 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5120{ 5121 effectChains = mEffectChains; 5122 for (size_t i = 0; i < mEffectChains.size(); i++) { 5123 mEffectChains[i]->lock(); 5124 } 5125} 5126 5127void AudioFlinger::PlaybackThread::unlockEffectChains( 5128 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5129{ 5130 for (size_t i = 0; i < effectChains.size(); i++) { 5131 effectChains[i]->unlock(); 5132 } 5133} 5134 5135 5136sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5137{ 5138 sp<EffectModule> effect; 5139 5140 sp<EffectChain> chain = getEffectChain_l(sessionId); 5141 if (chain != 0) { 5142 effect = chain->getEffectFromId_l(effectId); 5143 } 5144 return effect; 5145} 5146 5147status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5148 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5149{ 5150 Mutex::Autolock _l(mLock); 5151 return attachAuxEffect_l(track, EffectId); 5152} 5153 5154status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5155 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5156{ 5157 status_t status = NO_ERROR; 5158 5159 if (EffectId == 0) { 5160 track->setAuxBuffer(0, NULL); 5161 } else { 5162 // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX 5163 sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId); 5164 if (effect != 0) { 5165 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5166 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5167 } else { 5168 status = INVALID_OPERATION; 5169 } 5170 } else { 5171 status = BAD_VALUE; 5172 } 5173 } 5174 return status; 5175} 5176 5177void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5178{ 5179 for (size_t i = 0; i < mTracks.size(); ++i) { 5180 sp<Track> track = mTracks[i]; 5181 if (track->auxEffectId() == effectId) { 5182 attachAuxEffect_l(track, 0); 5183 } 5184 } 5185} 5186 5187// ---------------------------------------------------------------------------- 5188// EffectModule implementation 5189// ---------------------------------------------------------------------------- 5190 5191#undef LOG_TAG 5192#define LOG_TAG "AudioFlinger::EffectModule" 5193 5194AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5195 const wp<AudioFlinger::EffectChain>& chain, 5196 effect_descriptor_t *desc, 5197 int id, 5198 int sessionId) 5199 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5200 mStatus(NO_INIT), mState(IDLE) 5201{ 5202 LOGV("Constructor %p", this); 5203 int lStatus; 5204 sp<ThreadBase> thread = mThread.promote(); 5205 if (thread == 0) { 5206 return; 5207 } 5208 PlaybackThread *p = (PlaybackThread *)thread.get(); 5209 5210 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5211 5212 // create effect engine from effect factory 5213 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5214 5215 if (mStatus != NO_ERROR) { 5216 return; 5217 } 5218 lStatus = init(); 5219 if (lStatus < 0) { 5220 mStatus = lStatus; 5221 goto Error; 5222 } 5223 5224 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5225 return; 5226Error: 5227 EffectRelease(mEffectInterface); 5228 mEffectInterface = NULL; 5229 LOGV("Constructor Error %d", mStatus); 5230} 5231 5232AudioFlinger::EffectModule::~EffectModule() 5233{ 5234 LOGV("Destructor %p", this); 5235 if (mEffectInterface != NULL) { 5236 // release effect engine 5237 EffectRelease(mEffectInterface); 5238 } 5239} 5240 5241status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5242{ 5243 status_t status; 5244 5245 Mutex::Autolock _l(mLock); 5246 // First handle in mHandles has highest priority and controls the effect module 5247 int priority = handle->priority(); 5248 size_t size = mHandles.size(); 5249 sp<EffectHandle> h; 5250 size_t i; 5251 for (i = 0; i < size; i++) { 5252 h = mHandles[i].promote(); 5253 if (h == 0) continue; 5254 if (h->priority() <= priority) break; 5255 } 5256 // if inserted in first place, move effect control from previous owner to this handle 5257 if (i == 0) { 5258 if (h != 0) { 5259 h->setControl(false, true); 5260 } 5261 handle->setControl(true, false); 5262 status = NO_ERROR; 5263 } else { 5264 status = ALREADY_EXISTS; 5265 } 5266 mHandles.insertAt(handle, i); 5267 return status; 5268} 5269 5270size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5271{ 5272 Mutex::Autolock _l(mLock); 5273 size_t size = mHandles.size(); 5274 size_t i; 5275 for (i = 0; i < size; i++) { 5276 if (mHandles[i] == handle) break; 5277 } 5278 if (i == size) { 5279 return size; 5280 } 5281 mHandles.removeAt(i); 5282 size = mHandles.size(); 5283 // if removed from first place, move effect control from this handle to next in line 5284 if (i == 0 && size != 0) { 5285 sp<EffectHandle> h = mHandles[0].promote(); 5286 if (h != 0) { 5287 h->setControl(true, true); 5288 } 5289 } 5290 5291 return size; 5292} 5293 5294void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5295{ 5296 // keep a strong reference on this EffectModule to avoid calling the 5297 // destructor before we exit 5298 sp<EffectModule> keep(this); 5299 { 5300 sp<ThreadBase> thread = mThread.promote(); 5301 if (thread != 0) { 5302 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5303 playbackThread->disconnectEffect(keep, handle); 5304 } 5305 } 5306} 5307 5308void AudioFlinger::EffectModule::updateState() { 5309 Mutex::Autolock _l(mLock); 5310 5311 switch (mState) { 5312 case RESTART: 5313 reset_l(); 5314 // FALL THROUGH 5315 5316 case STARTING: 5317 // clear auxiliary effect input buffer for next accumulation 5318 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5319 memset(mConfig.inputCfg.buffer.raw, 5320 0, 5321 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5322 } 5323 start_l(); 5324 mState = ACTIVE; 5325 break; 5326 case STOPPING: 5327 stop_l(); 5328 mDisableWaitCnt = mMaxDisableWaitCnt; 5329 mState = STOPPED; 5330 break; 5331 case STOPPED: 5332 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5333 // turn off sequence. 5334 if (--mDisableWaitCnt == 0) { 5335 reset_l(); 5336 mState = IDLE; 5337 } 5338 break; 5339 default: //IDLE , ACTIVE 5340 break; 5341 } 5342} 5343 5344void AudioFlinger::EffectModule::process() 5345{ 5346 Mutex::Autolock _l(mLock); 5347 5348 if (mEffectInterface == NULL || 5349 mConfig.inputCfg.buffer.raw == NULL || 5350 mConfig.outputCfg.buffer.raw == NULL) { 5351 return; 5352 } 5353 5354 if (mState == ACTIVE || mState == STOPPING || mState == STOPPED) { 5355 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5356 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5357 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5358 mConfig.inputCfg.buffer.s32, 5359 mConfig.inputCfg.buffer.frameCount/2); 5360 } 5361 5362 // do the actual processing in the effect engine 5363 int ret = (*mEffectInterface)->process(mEffectInterface, 5364 &mConfig.inputCfg.buffer, 5365 &mConfig.outputCfg.buffer); 5366 5367 // force transition to IDLE state when engine is ready 5368 if (mState == STOPPED && ret == -ENODATA) { 5369 mDisableWaitCnt = 1; 5370 } 5371 5372 // clear auxiliary effect input buffer for next accumulation 5373 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5374 memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5375 } 5376 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5377 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ 5378 // If an insert effect is idle and input buffer is different from output buffer, copy input to 5379 // output 5380 sp<EffectChain> chain = mChain.promote(); 5381 if (chain != 0 && chain->activeTracks() != 0) { 5382 size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); 5383 if (mConfig.inputCfg.channels == CHANNEL_STEREO) { 5384 size *= 2; 5385 } 5386 memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); 5387 } 5388 } 5389} 5390 5391void AudioFlinger::EffectModule::reset_l() 5392{ 5393 if (mEffectInterface == NULL) { 5394 return; 5395 } 5396 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5397} 5398 5399status_t AudioFlinger::EffectModule::configure() 5400{ 5401 uint32_t channels; 5402 if (mEffectInterface == NULL) { 5403 return NO_INIT; 5404 } 5405 5406 sp<ThreadBase> thread = mThread.promote(); 5407 if (thread == 0) { 5408 return DEAD_OBJECT; 5409 } 5410 5411 // TODO: handle configuration of effects replacing track process 5412 if (thread->channelCount() == 1) { 5413 channels = CHANNEL_MONO; 5414 } else { 5415 channels = CHANNEL_STEREO; 5416 } 5417 5418 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5419 mConfig.inputCfg.channels = CHANNEL_MONO; 5420 } else { 5421 mConfig.inputCfg.channels = channels; 5422 } 5423 mConfig.outputCfg.channels = channels; 5424 mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; 5425 mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; 5426 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5427 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5428 mConfig.inputCfg.bufferProvider.cookie = NULL; 5429 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5430 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5431 mConfig.outputCfg.bufferProvider.cookie = NULL; 5432 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5433 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5434 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5435 // Insert effect: 5436 // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE, 5437 // always overwrites output buffer: input buffer == output buffer 5438 // - in other sessions: 5439 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5440 // other effect: overwrites output buffer: input buffer == output buffer 5441 // Auxiliary effect: 5442 // accumulates in output buffer: input buffer != output buffer 5443 // Therefore: accumulate <=> input buffer != output buffer 5444 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5445 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5446 } else { 5447 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5448 } 5449 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5450 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5451 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5452 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5453 5454 LOGV("configure() %p thread %p buffer %p framecount %d", 5455 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5456 5457 status_t cmdStatus; 5458 uint32_t size = sizeof(int); 5459 status_t status = (*mEffectInterface)->command(mEffectInterface, 5460 EFFECT_CMD_CONFIGURE, 5461 sizeof(effect_config_t), 5462 &mConfig, 5463 &size, 5464 &cmdStatus); 5465 if (status == 0) { 5466 status = cmdStatus; 5467 } 5468 5469 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5470 (1000 * mConfig.outputCfg.buffer.frameCount); 5471 5472 return status; 5473} 5474 5475status_t AudioFlinger::EffectModule::init() 5476{ 5477 Mutex::Autolock _l(mLock); 5478 if (mEffectInterface == NULL) { 5479 return NO_INIT; 5480 } 5481 status_t cmdStatus; 5482 uint32_t size = sizeof(status_t); 5483 status_t status = (*mEffectInterface)->command(mEffectInterface, 5484 EFFECT_CMD_INIT, 5485 0, 5486 NULL, 5487 &size, 5488 &cmdStatus); 5489 if (status == 0) { 5490 status = cmdStatus; 5491 } 5492 return status; 5493} 5494 5495status_t AudioFlinger::EffectModule::start_l() 5496{ 5497 if (mEffectInterface == NULL) { 5498 return NO_INIT; 5499 } 5500 status_t cmdStatus; 5501 uint32_t size = sizeof(status_t); 5502 status_t status = (*mEffectInterface)->command(mEffectInterface, 5503 EFFECT_CMD_ENABLE, 5504 0, 5505 NULL, 5506 &size, 5507 &cmdStatus); 5508 if (status == 0) { 5509 status = cmdStatus; 5510 } 5511 return status; 5512} 5513 5514status_t AudioFlinger::EffectModule::stop_l() 5515{ 5516 if (mEffectInterface == NULL) { 5517 return NO_INIT; 5518 } 5519 status_t cmdStatus; 5520 uint32_t size = sizeof(status_t); 5521 status_t status = (*mEffectInterface)->command(mEffectInterface, 5522 EFFECT_CMD_DISABLE, 5523 0, 5524 NULL, 5525 &size, 5526 &cmdStatus); 5527 if (status == 0) { 5528 status = cmdStatus; 5529 } 5530 return status; 5531} 5532 5533status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5534 uint32_t cmdSize, 5535 void *pCmdData, 5536 uint32_t *replySize, 5537 void *pReplyData) 5538{ 5539 Mutex::Autolock _l(mLock); 5540// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5541 5542 if (mEffectInterface == NULL) { 5543 return NO_INIT; 5544 } 5545 status_t status = (*mEffectInterface)->command(mEffectInterface, 5546 cmdCode, 5547 cmdSize, 5548 pCmdData, 5549 replySize, 5550 pReplyData); 5551 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5552 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5553 for (size_t i = 1; i < mHandles.size(); i++) { 5554 sp<EffectHandle> h = mHandles[i].promote(); 5555 if (h != 0) { 5556 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5557 } 5558 } 5559 } 5560 return status; 5561} 5562 5563status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5564{ 5565 Mutex::Autolock _l(mLock); 5566 LOGV("setEnabled %p enabled %d", this, enabled); 5567 5568 if (enabled != isEnabled()) { 5569 switch (mState) { 5570 // going from disabled to enabled 5571 case IDLE: 5572 mState = STARTING; 5573 break; 5574 case STOPPED: 5575 mState = RESTART; 5576 break; 5577 case STOPPING: 5578 mState = ACTIVE; 5579 break; 5580 5581 // going from enabled to disabled 5582 case RESTART: 5583 case STARTING: 5584 mState = IDLE; 5585 break; 5586 case ACTIVE: 5587 mState = STOPPING; 5588 break; 5589 } 5590 for (size_t i = 1; i < mHandles.size(); i++) { 5591 sp<EffectHandle> h = mHandles[i].promote(); 5592 if (h != 0) { 5593 h->setEnabled(enabled); 5594 } 5595 } 5596 } 5597 return NO_ERROR; 5598} 5599 5600bool AudioFlinger::EffectModule::isEnabled() 5601{ 5602 switch (mState) { 5603 case RESTART: 5604 case STARTING: 5605 case ACTIVE: 5606 return true; 5607 case IDLE: 5608 case STOPPING: 5609 case STOPPED: 5610 default: 5611 return false; 5612 } 5613} 5614 5615status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5616{ 5617 Mutex::Autolock _l(mLock); 5618 status_t status = NO_ERROR; 5619 5620 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5621 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5622 if ((mState >= ACTIVE) && 5623 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5624 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5625 status_t cmdStatus; 5626 uint32_t volume[2]; 5627 uint32_t *pVolume = NULL; 5628 uint32_t size = sizeof(volume); 5629 volume[0] = *left; 5630 volume[1] = *right; 5631 if (controller) { 5632 pVolume = volume; 5633 } 5634 status = (*mEffectInterface)->command(mEffectInterface, 5635 EFFECT_CMD_SET_VOLUME, 5636 size, 5637 volume, 5638 &size, 5639 pVolume); 5640 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5641 *left = volume[0]; 5642 *right = volume[1]; 5643 } 5644 } 5645 return status; 5646} 5647 5648status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5649{ 5650 Mutex::Autolock _l(mLock); 5651 status_t status = NO_ERROR; 5652 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5653 // convert device bit field from AudioSystem to EffectApi format. 5654 device = deviceAudioSystemToEffectApi(device); 5655 if (device == 0) { 5656 return BAD_VALUE; 5657 } 5658 status_t cmdStatus; 5659 uint32_t size = sizeof(status_t); 5660 status = (*mEffectInterface)->command(mEffectInterface, 5661 EFFECT_CMD_SET_DEVICE, 5662 sizeof(uint32_t), 5663 &device, 5664 &size, 5665 &cmdStatus); 5666 if (status == NO_ERROR) { 5667 status = cmdStatus; 5668 } 5669 } 5670 return status; 5671} 5672 5673status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5674{ 5675 Mutex::Autolock _l(mLock); 5676 status_t status = NO_ERROR; 5677 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5678 // convert audio mode from AudioSystem to EffectApi format. 5679 int effectMode = modeAudioSystemToEffectApi(mode); 5680 if (effectMode < 0) { 5681 return BAD_VALUE; 5682 } 5683 status_t cmdStatus; 5684 uint32_t size = sizeof(status_t); 5685 status = (*mEffectInterface)->command(mEffectInterface, 5686 EFFECT_CMD_SET_AUDIO_MODE, 5687 sizeof(int), 5688 &effectMode, 5689 &size, 5690 &cmdStatus); 5691 if (status == NO_ERROR) { 5692 status = cmdStatus; 5693 } 5694 } 5695 return status; 5696} 5697 5698// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified 5699const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { 5700 DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE 5701 DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER 5702 DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET 5703 DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE 5704 DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO 5705 DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET 5706 DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT 5707 DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP 5708 DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES 5709 DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER 5710 DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL 5711}; 5712 5713uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) 5714{ 5715 uint32_t deviceOut = 0; 5716 while (device) { 5717 const uint32_t i = 31 - __builtin_clz(device); 5718 device &= ~(1 << i); 5719 if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { 5720 LOGE("device convertion error for AudioSystem device 0x%08x", device); 5721 return 0; 5722 } 5723 deviceOut |= (uint32_t)sDeviceConvTable[i]; 5724 } 5725 return deviceOut; 5726} 5727 5728// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified 5729const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { 5730 AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL 5731 AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE 5732 AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL 5733}; 5734 5735int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) 5736{ 5737 int modeOut = -1; 5738 if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { 5739 modeOut = (int)sModeConvTable[mode]; 5740 } 5741 return modeOut; 5742} 5743 5744status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5745{ 5746 const size_t SIZE = 256; 5747 char buffer[SIZE]; 5748 String8 result; 5749 5750 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5751 result.append(buffer); 5752 5753 bool locked = tryLock(mLock); 5754 // failed to lock - AudioFlinger is probably deadlocked 5755 if (!locked) { 5756 result.append("\t\tCould not lock Fx mutex:\n"); 5757 } 5758 5759 result.append("\t\tSession Status State Engine:\n"); 5760 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5761 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5762 result.append(buffer); 5763 5764 result.append("\t\tDescriptor:\n"); 5765 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5766 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5767 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5768 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5769 result.append(buffer); 5770 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5771 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5772 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5773 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5774 result.append(buffer); 5775 snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", 5776 mDescriptor.apiVersion, 5777 mDescriptor.flags); 5778 result.append(buffer); 5779 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5780 mDescriptor.name); 5781 result.append(buffer); 5782 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5783 mDescriptor.implementor); 5784 result.append(buffer); 5785 5786 result.append("\t\t- Input configuration:\n"); 5787 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5788 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5789 (uint32_t)mConfig.inputCfg.buffer.raw, 5790 mConfig.inputCfg.buffer.frameCount, 5791 mConfig.inputCfg.samplingRate, 5792 mConfig.inputCfg.channels, 5793 mConfig.inputCfg.format); 5794 result.append(buffer); 5795 5796 result.append("\t\t- Output configuration:\n"); 5797 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5798 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5799 (uint32_t)mConfig.outputCfg.buffer.raw, 5800 mConfig.outputCfg.buffer.frameCount, 5801 mConfig.outputCfg.samplingRate, 5802 mConfig.outputCfg.channels, 5803 mConfig.outputCfg.format); 5804 result.append(buffer); 5805 5806 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5807 result.append(buffer); 5808 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5809 for (size_t i = 0; i < mHandles.size(); ++i) { 5810 sp<EffectHandle> handle = mHandles[i].promote(); 5811 if (handle != 0) { 5812 handle->dump(buffer, SIZE); 5813 result.append(buffer); 5814 } 5815 } 5816 5817 result.append("\n"); 5818 5819 write(fd, result.string(), result.length()); 5820 5821 if (locked) { 5822 mLock.unlock(); 5823 } 5824 5825 return NO_ERROR; 5826} 5827 5828// ---------------------------------------------------------------------------- 5829// EffectHandle implementation 5830// ---------------------------------------------------------------------------- 5831 5832#undef LOG_TAG 5833#define LOG_TAG "AudioFlinger::EffectHandle" 5834 5835AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5836 const sp<AudioFlinger::Client>& client, 5837 const sp<IEffectClient>& effectClient, 5838 int32_t priority) 5839 : BnEffect(), 5840 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5841{ 5842 LOGV("constructor %p", this); 5843 5844 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5845 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5846 if (mCblkMemory != 0) { 5847 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5848 5849 if (mCblk) { 5850 new(mCblk) effect_param_cblk_t(); 5851 mBuffer = (uint8_t *)mCblk + bufOffset; 5852 } 5853 } else { 5854 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5855 return; 5856 } 5857} 5858 5859AudioFlinger::EffectHandle::~EffectHandle() 5860{ 5861 LOGV("Destructor %p", this); 5862 disconnect(); 5863} 5864 5865status_t AudioFlinger::EffectHandle::enable() 5866{ 5867 if (!mHasControl) return INVALID_OPERATION; 5868 if (mEffect == 0) return DEAD_OBJECT; 5869 5870 return mEffect->setEnabled(true); 5871} 5872 5873status_t AudioFlinger::EffectHandle::disable() 5874{ 5875 if (!mHasControl) return INVALID_OPERATION; 5876 if (mEffect == NULL) return DEAD_OBJECT; 5877 5878 return mEffect->setEnabled(false); 5879} 5880 5881void AudioFlinger::EffectHandle::disconnect() 5882{ 5883 if (mEffect == 0) { 5884 return; 5885 } 5886 mEffect->disconnect(this); 5887 // release sp on module => module destructor can be called now 5888 mEffect.clear(); 5889 if (mCblk) { 5890 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5891 } 5892 mCblkMemory.clear(); // and free the shared memory 5893 if (mClient != 0) { 5894 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5895 mClient.clear(); 5896 } 5897} 5898 5899status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5900 uint32_t cmdSize, 5901 void *pCmdData, 5902 uint32_t *replySize, 5903 void *pReplyData) 5904{ 5905// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5906// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5907 5908 // only get parameter command is permitted for applications not controlling the effect 5909 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5910 return INVALID_OPERATION; 5911 } 5912 if (mEffect == 0) return DEAD_OBJECT; 5913 5914 // handle commands that are not forwarded transparently to effect engine 5915 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5916 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5917 // no risk to block the whole media server process or mixer threads is we are stuck here 5918 Mutex::Autolock _l(mCblk->lock); 5919 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5920 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 5921 mCblk->serverIndex = 0; 5922 mCblk->clientIndex = 0; 5923 return BAD_VALUE; 5924 } 5925 status_t status = NO_ERROR; 5926 while (mCblk->serverIndex < mCblk->clientIndex) { 5927 int reply; 5928 uint32_t rsize = sizeof(int); 5929 int *p = (int *)(mBuffer + mCblk->serverIndex); 5930 int size = *p++; 5931 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 5932 LOGW("command(): invalid parameter block size"); 5933 break; 5934 } 5935 effect_param_t *param = (effect_param_t *)p; 5936 if (param->psize == 0 || param->vsize == 0) { 5937 LOGW("command(): null parameter or value size"); 5938 mCblk->serverIndex += size; 5939 continue; 5940 } 5941 uint32_t psize = sizeof(effect_param_t) + 5942 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 5943 param->vsize; 5944 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 5945 psize, 5946 p, 5947 &rsize, 5948 &reply); 5949 if (ret == NO_ERROR) { 5950 if (reply != NO_ERROR) { 5951 status = reply; 5952 } 5953 } else { 5954 status = ret; 5955 } 5956 mCblk->serverIndex += size; 5957 } 5958 mCblk->serverIndex = 0; 5959 mCblk->clientIndex = 0; 5960 return status; 5961 } else if (cmdCode == EFFECT_CMD_ENABLE) { 5962 return enable(); 5963 } else if (cmdCode == EFFECT_CMD_DISABLE) { 5964 return disable(); 5965 } 5966 5967 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 5968} 5969 5970sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 5971 return mCblkMemory; 5972} 5973 5974void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 5975{ 5976 LOGV("setControl %p control %d", this, hasControl); 5977 5978 mHasControl = hasControl; 5979 if (signal && mEffectClient != 0) { 5980 mEffectClient->controlStatusChanged(hasControl); 5981 } 5982} 5983 5984void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 5985 uint32_t cmdSize, 5986 void *pCmdData, 5987 uint32_t replySize, 5988 void *pReplyData) 5989{ 5990 if (mEffectClient != 0) { 5991 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 5992 } 5993} 5994 5995 5996 5997void AudioFlinger::EffectHandle::setEnabled(bool enabled) 5998{ 5999 if (mEffectClient != 0) { 6000 mEffectClient->enableStatusChanged(enabled); 6001 } 6002} 6003 6004status_t AudioFlinger::EffectHandle::onTransact( 6005 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6006{ 6007 return BnEffect::onTransact(code, data, reply, flags); 6008} 6009 6010 6011void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6012{ 6013 bool locked = tryLock(mCblk->lock); 6014 6015 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6016 (mClient == NULL) ? getpid() : mClient->pid(), 6017 mPriority, 6018 mHasControl, 6019 !locked, 6020 mCblk->clientIndex, 6021 mCblk->serverIndex 6022 ); 6023 6024 if (locked) { 6025 mCblk->lock.unlock(); 6026 } 6027} 6028 6029#undef LOG_TAG 6030#define LOG_TAG "AudioFlinger::EffectChain" 6031 6032AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6033 int sessionId) 6034 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), 6035 mVolumeCtrlIdx(-1), mLeftVolume(0), mRightVolume(0), 6036 mNewLeftVolume(0), mNewRightVolume(0) 6037{ 6038 mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 6039} 6040 6041AudioFlinger::EffectChain::~EffectChain() 6042{ 6043 if (mOwnInBuffer) { 6044 delete mInBuffer; 6045 } 6046 6047} 6048 6049// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6050sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6051{ 6052 sp<EffectModule> effect; 6053 size_t size = mEffects.size(); 6054 6055 for (size_t i = 0; i < size; i++) { 6056 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6057 effect = mEffects[i]; 6058 break; 6059 } 6060 } 6061 return effect; 6062} 6063 6064// getEffectFromId_l() must be called with PlaybackThread::mLock held 6065sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6066{ 6067 sp<EffectModule> effect; 6068 size_t size = mEffects.size(); 6069 6070 for (size_t i = 0; i < size; i++) { 6071 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6072 if (id == 0 || mEffects[i]->id() == id) { 6073 effect = mEffects[i]; 6074 break; 6075 } 6076 } 6077 return effect; 6078} 6079 6080// Must be called with EffectChain::mLock locked 6081void AudioFlinger::EffectChain::process_l() 6082{ 6083 size_t size = mEffects.size(); 6084 for (size_t i = 0; i < size; i++) { 6085 mEffects[i]->process(); 6086 } 6087 for (size_t i = 0; i < size; i++) { 6088 mEffects[i]->updateState(); 6089 } 6090 // if no track is active, input buffer must be cleared here as the mixer process 6091 // will not do it 6092 if (mSessionId > 0 && activeTracks() == 0) { 6093 sp<ThreadBase> thread = mThread.promote(); 6094 if (thread != 0) { 6095 size_t numSamples = thread->frameCount() * thread->channelCount(); 6096 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6097 } 6098 } 6099} 6100 6101// addEffect_l() must be called with PlaybackThread::mLock held 6102status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6103{ 6104 effect_descriptor_t desc = effect->desc(); 6105 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6106 6107 Mutex::Autolock _l(mLock); 6108 effect->setChain(this); 6109 sp<ThreadBase> thread = mThread.promote(); 6110 if (thread == 0) { 6111 return NO_INIT; 6112 } 6113 effect->setThread(thread); 6114 6115 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6116 // Auxiliary effects are inserted at the beginning of mEffects vector as 6117 // they are processed first and accumulated in chain input buffer 6118 mEffects.insertAt(effect, 0); 6119 6120 // the input buffer for auxiliary effect contains mono samples in 6121 // 32 bit format. This is to avoid saturation in AudoMixer 6122 // accumulation stage. Saturation is done in EffectModule::process() before 6123 // calling the process in effect engine 6124 size_t numSamples = thread->frameCount(); 6125 int32_t *buffer = new int32_t[numSamples]; 6126 memset(buffer, 0, numSamples * sizeof(int32_t)); 6127 effect->setInBuffer((int16_t *)buffer); 6128 // auxiliary effects output samples to chain input buffer for further processing 6129 // by insert effects 6130 effect->setOutBuffer(mInBuffer); 6131 } else { 6132 // Insert effects are inserted at the end of mEffects vector as they are processed 6133 // after track and auxiliary effects. 6134 // Insert effect order as a function of indicated preference: 6135 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6136 // another effect is present 6137 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6138 // last effect claiming first position 6139 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6140 // first effect claiming last position 6141 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6142 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6143 // already present 6144 6145 int size = (int)mEffects.size(); 6146 int idx_insert = size; 6147 int idx_insert_first = -1; 6148 int idx_insert_last = -1; 6149 6150 for (int i = 0; i < size; i++) { 6151 effect_descriptor_t d = mEffects[i]->desc(); 6152 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6153 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6154 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6155 // check invalid effect chaining combinations 6156 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6157 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6158 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6159 return INVALID_OPERATION; 6160 } 6161 // remember position of first insert effect and by default 6162 // select this as insert position for new effect 6163 if (idx_insert == size) { 6164 idx_insert = i; 6165 } 6166 // remember position of last insert effect claiming 6167 // first position 6168 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6169 idx_insert_first = i; 6170 } 6171 // remember position of first insert effect claiming 6172 // last position 6173 if (iPref == EFFECT_FLAG_INSERT_LAST && 6174 idx_insert_last == -1) { 6175 idx_insert_last = i; 6176 } 6177 } 6178 } 6179 6180 // modify idx_insert from first position if needed 6181 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6182 if (idx_insert_last != -1) { 6183 idx_insert = idx_insert_last; 6184 } else { 6185 idx_insert = size; 6186 } 6187 } else { 6188 if (idx_insert_first != -1) { 6189 idx_insert = idx_insert_first + 1; 6190 } 6191 } 6192 6193 // always read samples from chain input buffer 6194 effect->setInBuffer(mInBuffer); 6195 6196 // if last effect in the chain, output samples to chain 6197 // output buffer, otherwise to chain input buffer 6198 if (idx_insert == size) { 6199 if (idx_insert != 0) { 6200 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6201 mEffects[idx_insert-1]->configure(); 6202 } 6203 effect->setOutBuffer(mOutBuffer); 6204 } else { 6205 effect->setOutBuffer(mInBuffer); 6206 } 6207 mEffects.insertAt(effect, idx_insert); 6208 6209 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6210 } 6211 effect->configure(); 6212 return NO_ERROR; 6213} 6214 6215// removeEffect_l() must be called with PlaybackThread::mLock held 6216size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6217{ 6218 Mutex::Autolock _l(mLock); 6219 int size = (int)mEffects.size(); 6220 int i; 6221 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6222 6223 for (i = 0; i < size; i++) { 6224 if (effect == mEffects[i]) { 6225 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6226 delete[] effect->inBuffer(); 6227 } else { 6228 if (i == size - 1 && i != 0) { 6229 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6230 mEffects[i - 1]->configure(); 6231 } 6232 } 6233 mEffects.removeAt(i); 6234 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6235 break; 6236 } 6237 } 6238 6239 return mEffects.size(); 6240} 6241 6242// setDevice_l() must be called with PlaybackThread::mLock held 6243void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6244{ 6245 size_t size = mEffects.size(); 6246 for (size_t i = 0; i < size; i++) { 6247 mEffects[i]->setDevice(device); 6248 } 6249} 6250 6251// setMode_l() must be called with PlaybackThread::mLock held 6252void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6253{ 6254 size_t size = mEffects.size(); 6255 for (size_t i = 0; i < size; i++) { 6256 mEffects[i]->setMode(mode); 6257 } 6258} 6259 6260// setVolume_l() must be called with PlaybackThread::mLock held 6261bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6262{ 6263 uint32_t newLeft = *left; 6264 uint32_t newRight = *right; 6265 bool hasControl = false; 6266 int ctrlIdx = -1; 6267 size_t size = mEffects.size(); 6268 6269 // first update volume controller 6270 for (size_t i = size; i > 0; i--) { 6271 if ((mEffects[i - 1]->state() >= EffectModule::ACTIVE) && 6272 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6273 ctrlIdx = i - 1; 6274 hasControl = true; 6275 break; 6276 } 6277 } 6278 6279 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6280 if (hasControl) { 6281 *left = mNewLeftVolume; 6282 *right = mNewRightVolume; 6283 } 6284 return hasControl; 6285 } 6286 6287 if (mVolumeCtrlIdx != -1) { 6288 hasControl = true; 6289 } 6290 mVolumeCtrlIdx = ctrlIdx; 6291 mLeftVolume = newLeft; 6292 mRightVolume = newRight; 6293 6294 // second get volume update from volume controller 6295 if (ctrlIdx >= 0) { 6296 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6297 mNewLeftVolume = newLeft; 6298 mNewRightVolume = newRight; 6299 } 6300 // then indicate volume to all other effects in chain. 6301 // Pass altered volume to effects before volume controller 6302 // and requested volume to effects after controller 6303 uint32_t lVol = newLeft; 6304 uint32_t rVol = newRight; 6305 6306 for (size_t i = 0; i < size; i++) { 6307 if ((int)i == ctrlIdx) continue; 6308 // this also works for ctrlIdx == -1 when there is no volume controller 6309 if ((int)i > ctrlIdx) { 6310 lVol = *left; 6311 rVol = *right; 6312 } 6313 mEffects[i]->setVolume(&lVol, &rVol, false); 6314 } 6315 *left = newLeft; 6316 *right = newRight; 6317 6318 return hasControl; 6319} 6320 6321status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6322{ 6323 const size_t SIZE = 256; 6324 char buffer[SIZE]; 6325 String8 result; 6326 6327 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6328 result.append(buffer); 6329 6330 bool locked = tryLock(mLock); 6331 // failed to lock - AudioFlinger is probably deadlocked 6332 if (!locked) { 6333 result.append("\tCould not lock mutex:\n"); 6334 } 6335 6336 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6337 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6338 mEffects.size(), 6339 (uint32_t)mInBuffer, 6340 (uint32_t)mOutBuffer, 6341 mActiveTrackCnt); 6342 result.append(buffer); 6343 write(fd, result.string(), result.size()); 6344 6345 for (size_t i = 0; i < mEffects.size(); ++i) { 6346 sp<EffectModule> effect = mEffects[i]; 6347 if (effect != 0) { 6348 effect->dump(fd, args); 6349 } 6350 } 6351 6352 if (locked) { 6353 mLock.unlock(); 6354 } 6355 6356 return NO_ERROR; 6357} 6358 6359#undef LOG_TAG 6360#define LOG_TAG "AudioFlinger" 6361 6362// ---------------------------------------------------------------------------- 6363 6364status_t AudioFlinger::onTransact( 6365 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6366{ 6367 return BnAudioFlinger::onTransact(code, data, reply, flags); 6368} 6369 6370}; // namespace android 6371