AudioFlinger.cpp revision 571d49c1c316f5e07b74ed7b5df6bdec7cbc1a14
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 if (masterMute) { 1679 masterVolume = 0; 1680 } 1681#ifdef LVMX 1682 bool tracksConnectedChanged = false; 1683 bool stateChanged = false; 1684 1685 int audioOutputType = LifeVibes::getMixerType(mId, mType); 1686 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1687 { 1688 int activeTypes = 0; 1689 for (size_t i=0 ; i<count ; i++) { 1690 sp<Track> t = activeTracks[i].promote(); 1691 if (t == 0) continue; 1692 Track* const track = t.get(); 1693 int iTracktype=track->type(); 1694 activeTypes |= 1<<track->type(); 1695 } 1696 LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute); 1697 } 1698#endif 1699 // Delegate master volume control to effect in output mix effect chain if needed 1700 sp<EffectChain> chain = getEffectChain_l(AudioSystem::SESSION_OUTPUT_MIX); 1701 if (chain != 0) { 1702 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 1703 chain->setVolume_l(&v, &v); 1704 masterVolume = (float)((v + (1 << 23)) >> 24); 1705 chain.clear(); 1706 } 1707 1708 for (size_t i=0 ; i<count ; i++) { 1709 sp<Track> t = activeTracks[i].promote(); 1710 if (t == 0) continue; 1711 1712 Track* const track = t.get(); 1713 audio_track_cblk_t* cblk = track->cblk(); 1714 1715 // The first time a track is added we wait 1716 // for all its buffers to be filled before processing it 1717 mAudioMixer->setActiveTrack(track->name()); 1718 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 1719 !track->isPaused() && !track->isTerminated()) 1720 { 1721 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1722 1723 mixedTracks++; 1724 1725 // track->mainBuffer() != mMixBuffer means there is an effect chain 1726 // connected to the track 1727 chain.clear(); 1728 if (track->mainBuffer() != mMixBuffer) { 1729 chain = getEffectChain_l(track->sessionId()); 1730 // Delegate volume control to effect in track effect chain if needed 1731 if (chain != 0) { 1732 tracksWithEffect++; 1733 } else { 1734 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1735 track->name(), track->sessionId()); 1736 } 1737 } 1738 1739 1740 int param = AudioMixer::VOLUME; 1741 if (track->mFillingUpStatus == Track::FS_FILLED) { 1742 // no ramp for the first volume setting 1743 track->mFillingUpStatus = Track::FS_ACTIVE; 1744 if (track->mState == TrackBase::RESUMING) { 1745 track->mState = TrackBase::ACTIVE; 1746 param = AudioMixer::RAMP_VOLUME; 1747 } 1748 } else if (cblk->server != 0) { 1749 // If the track is stopped before the first frame was mixed, 1750 // do not apply ramp 1751 param = AudioMixer::RAMP_VOLUME; 1752 } 1753 1754 // compute volume for this track 1755 int16_t left, right, aux; 1756 if (track->isMuted() || track->isPausing() || 1757 mStreamTypes[track->type()].mute) { 1758 left = right = aux = 0; 1759 if (track->isPausing()) { 1760 track->setPaused(); 1761 } 1762 } else { 1763 // read original volumes with volume control 1764 float typeVolume = mStreamTypes[track->type()].volume; 1765#ifdef LVMX 1766 bool streamMute=false; 1767 // read the volume from the LivesVibes audio engine. 1768 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1769 { 1770 LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); 1771 if (streamMute) { 1772 typeVolume = 0; 1773 } 1774 } 1775#endif 1776 float v = masterVolume * typeVolume; 1777 uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12; 1778 uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12; 1779 1780 // Delegate volume control to effect in track effect chain if needed 1781 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1782 // Do not ramp volume is volume is controlled by effect 1783 param = AudioMixer::VOLUME; 1784 } 1785 1786 // Convert volumes from 8.24 to 4.12 format 1787 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1788 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1789 left = int16_t(v_clamped); 1790 v_clamped = (vr + (1 << 11)) >> 12; 1791 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1792 right = int16_t(v_clamped); 1793 1794 v_clamped = (uint32_t)(v * cblk->sendLevel); 1795 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1796 aux = int16_t(v_clamped); 1797 } 1798 1799#ifdef LVMX 1800 if ( tracksConnectedChanged || stateChanged ) 1801 { 1802 // only do the ramp when the volume is changed by the user / application 1803 param = AudioMixer::VOLUME; 1804 } 1805#endif 1806 1807 // XXX: these things DON'T need to be done each time 1808 mAudioMixer->setBufferProvider(track); 1809 mAudioMixer->enable(AudioMixer::MIXING); 1810 1811 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1812 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1813 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1814 mAudioMixer->setParameter( 1815 AudioMixer::TRACK, 1816 AudioMixer::FORMAT, (void *)track->format()); 1817 mAudioMixer->setParameter( 1818 AudioMixer::TRACK, 1819 AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); 1820 mAudioMixer->setParameter( 1821 AudioMixer::RESAMPLE, 1822 AudioMixer::SAMPLE_RATE, 1823 (void *)(cblk->sampleRate)); 1824 mAudioMixer->setParameter( 1825 AudioMixer::TRACK, 1826 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1827 mAudioMixer->setParameter( 1828 AudioMixer::TRACK, 1829 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1830 1831 // reset retry count 1832 track->mRetryCount = kMaxTrackRetries; 1833 mixerStatus = MIXER_TRACKS_READY; 1834 } else { 1835 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1836 if (track->isStopped()) { 1837 track->reset(); 1838 } 1839 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1840 // We have consumed all the buffers of this track. 1841 // Remove it from the list of active tracks. 1842 tracksToRemove->add(track); 1843 } else { 1844 // No buffers for this track. Give it a few chances to 1845 // fill a buffer, then remove it from active list. 1846 if (--(track->mRetryCount) <= 0) { 1847 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1848 tracksToRemove->add(track); 1849 } else if (mixerStatus != MIXER_TRACKS_READY) { 1850 mixerStatus = MIXER_TRACKS_ENABLED; 1851 } 1852 } 1853 mAudioMixer->disable(AudioMixer::MIXING); 1854 } 1855 } 1856 1857 // remove all the tracks that need to be... 1858 count = tracksToRemove->size(); 1859 if (UNLIKELY(count)) { 1860 for (size_t i=0 ; i<count ; i++) { 1861 const sp<Track>& track = tracksToRemove->itemAt(i); 1862 mActiveTracks.remove(track); 1863 if (track->mainBuffer() != mMixBuffer) { 1864 chain = getEffectChain_l(track->sessionId()); 1865 if (chain != 0) { 1866 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1867 chain->stopTrack(); 1868 } 1869 } 1870 if (track->isTerminated()) { 1871 mTracks.remove(track); 1872 deleteTrackName_l(track->mName); 1873 } 1874 } 1875 } 1876 1877 // mix buffer must be cleared if all tracks are connected to an 1878 // effect chain as in this case the mixer will not write to 1879 // mix buffer and track effects will accumulate into it 1880 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1881 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1882 } 1883 1884 return mixerStatus; 1885} 1886 1887void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1888{ 1889 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1890 this, streamType, mTracks.size()); 1891 Mutex::Autolock _l(mLock); 1892 1893 size_t size = mTracks.size(); 1894 for (size_t i = 0; i < size; i++) { 1895 sp<Track> t = mTracks[i]; 1896 if (t->type() == streamType) { 1897 t->mCblk->lock.lock(); 1898 t->mCblk->flags |= CBLK_INVALID_ON; 1899 t->mCblk->cv.signal(); 1900 t->mCblk->lock.unlock(); 1901 } 1902 } 1903} 1904 1905 1906// getTrackName_l() must be called with ThreadBase::mLock held 1907int AudioFlinger::MixerThread::getTrackName_l() 1908{ 1909 return mAudioMixer->getTrackName(); 1910} 1911 1912// deleteTrackName_l() must be called with ThreadBase::mLock held 1913void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1914{ 1915 LOGV("remove track (%d) and delete from mixer", name); 1916 mAudioMixer->deleteTrackName(name); 1917} 1918 1919// checkForNewParameters_l() must be called with ThreadBase::mLock held 1920bool AudioFlinger::MixerThread::checkForNewParameters_l() 1921{ 1922 bool reconfig = false; 1923 1924 while (!mNewParameters.isEmpty()) { 1925 status_t status = NO_ERROR; 1926 String8 keyValuePair = mNewParameters[0]; 1927 AudioParameter param = AudioParameter(keyValuePair); 1928 int value; 1929 1930 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1931 reconfig = true; 1932 } 1933 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1934 if (value != AudioSystem::PCM_16_BIT) { 1935 status = BAD_VALUE; 1936 } else { 1937 reconfig = true; 1938 } 1939 } 1940 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1941 if (value != AudioSystem::CHANNEL_OUT_STEREO) { 1942 status = BAD_VALUE; 1943 } else { 1944 reconfig = true; 1945 } 1946 } 1947 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1948 // do not accept frame count changes if tracks are open as the track buffer 1949 // size depends on frame count and correct behavior would not be garantied 1950 // if frame count is changed after track creation 1951 if (!mTracks.isEmpty()) { 1952 status = INVALID_OPERATION; 1953 } else { 1954 reconfig = true; 1955 } 1956 } 1957 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1958 // forward device change to effects that have requested to be 1959 // aware of attached audio device. 1960 mDevice = (uint32_t)value; 1961 for (size_t i = 0; i < mEffectChains.size(); i++) { 1962 mEffectChains[i]->setDevice_l(mDevice); 1963 } 1964 } 1965 1966 if (status == NO_ERROR) { 1967 status = mOutput->setParameters(keyValuePair); 1968 if (!mStandby && status == INVALID_OPERATION) { 1969 mOutput->standby(); 1970 mStandby = true; 1971 mBytesWritten = 0; 1972 status = mOutput->setParameters(keyValuePair); 1973 } 1974 if (status == NO_ERROR && reconfig) { 1975 delete mAudioMixer; 1976 readOutputParameters(); 1977 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1978 for (size_t i = 0; i < mTracks.size() ; i++) { 1979 int name = getTrackName_l(); 1980 if (name < 0) break; 1981 mTracks[i]->mName = name; 1982 // limit track sample rate to 2 x new output sample rate 1983 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1984 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 1985 } 1986 } 1987 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 1988 } 1989 } 1990 1991 mNewParameters.removeAt(0); 1992 1993 mParamStatus = status; 1994 mParamCond.signal(); 1995 mWaitWorkCV.wait(mLock); 1996 } 1997 return reconfig; 1998} 1999 2000status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2001{ 2002 const size_t SIZE = 256; 2003 char buffer[SIZE]; 2004 String8 result; 2005 2006 PlaybackThread::dumpInternals(fd, args); 2007 2008 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2009 result.append(buffer); 2010 write(fd, result.string(), result.size()); 2011 return NO_ERROR; 2012} 2013 2014uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2015{ 2016 return (uint32_t)(mOutput->latency() * 1000) / 2; 2017} 2018 2019uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2020{ 2021 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2022} 2023 2024// ---------------------------------------------------------------------------- 2025AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2026 : PlaybackThread(audioFlinger, output, id, device) 2027{ 2028 mType = PlaybackThread::DIRECT; 2029} 2030 2031AudioFlinger::DirectOutputThread::~DirectOutputThread() 2032{ 2033} 2034 2035 2036static inline int16_t clamp16(int32_t sample) 2037{ 2038 if ((sample>>15) ^ (sample>>31)) 2039 sample = 0x7FFF ^ (sample>>31); 2040 return sample; 2041} 2042 2043static inline 2044int32_t mul(int16_t in, int16_t v) 2045{ 2046#if defined(__arm__) && !defined(__thumb__) 2047 int32_t out; 2048 asm( "smulbb %[out], %[in], %[v] \n" 2049 : [out]"=r"(out) 2050 : [in]"%r"(in), [v]"r"(v) 2051 : ); 2052 return out; 2053#else 2054 return in * int32_t(v); 2055#endif 2056} 2057 2058void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2059{ 2060 // Do not apply volume on compressed audio 2061 if (!AudioSystem::isLinearPCM(mFormat)) { 2062 return; 2063 } 2064 2065 // convert to signed 16 bit before volume calculation 2066 if (mFormat == AudioSystem::PCM_8_BIT) { 2067 size_t count = mFrameCount * mChannelCount; 2068 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2069 int16_t *dst = mMixBuffer + count-1; 2070 while(count--) { 2071 *dst-- = (int16_t)(*src--^0x80) << 8; 2072 } 2073 } 2074 2075 size_t frameCount = mFrameCount; 2076 int16_t *out = mMixBuffer; 2077 if (ramp) { 2078 if (mChannelCount == 1) { 2079 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2080 int32_t vlInc = d / (int32_t)frameCount; 2081 int32_t vl = ((int32_t)mLeftVolShort << 16); 2082 do { 2083 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2084 out++; 2085 vl += vlInc; 2086 } while (--frameCount); 2087 2088 } else { 2089 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2090 int32_t vlInc = d / (int32_t)frameCount; 2091 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2092 int32_t vrInc = d / (int32_t)frameCount; 2093 int32_t vl = ((int32_t)mLeftVolShort << 16); 2094 int32_t vr = ((int32_t)mRightVolShort << 16); 2095 do { 2096 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2097 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2098 out += 2; 2099 vl += vlInc; 2100 vr += vrInc; 2101 } while (--frameCount); 2102 } 2103 } else { 2104 if (mChannelCount == 1) { 2105 do { 2106 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2107 out++; 2108 } while (--frameCount); 2109 } else { 2110 do { 2111 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2112 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2113 out += 2; 2114 } while (--frameCount); 2115 } 2116 } 2117 2118 // convert back to unsigned 8 bit after volume calculation 2119 if (mFormat == AudioSystem::PCM_8_BIT) { 2120 size_t count = mFrameCount * mChannelCount; 2121 int16_t *src = mMixBuffer; 2122 uint8_t *dst = (uint8_t *)mMixBuffer; 2123 while(count--) { 2124 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2125 } 2126 } 2127 2128 mLeftVolShort = leftVol; 2129 mRightVolShort = rightVol; 2130} 2131 2132bool AudioFlinger::DirectOutputThread::threadLoop() 2133{ 2134 uint32_t mixerStatus = MIXER_IDLE; 2135 sp<Track> trackToRemove; 2136 sp<Track> activeTrack; 2137 nsecs_t standbyTime = systemTime(); 2138 int8_t *curBuf; 2139 size_t mixBufferSize = mFrameCount*mFrameSize; 2140 uint32_t activeSleepTime = activeSleepTimeUs(); 2141 uint32_t idleSleepTime = idleSleepTimeUs(); 2142 uint32_t sleepTime = idleSleepTime; 2143 // use shorter standby delay as on normal output to release 2144 // hardware resources as soon as possible 2145 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2146 2147 while (!exitPending()) 2148 { 2149 bool rampVolume; 2150 uint16_t leftVol; 2151 uint16_t rightVol; 2152 Vector< sp<EffectChain> > effectChains; 2153 2154 processConfigEvents(); 2155 2156 mixerStatus = MIXER_IDLE; 2157 2158 { // scope for the mLock 2159 2160 Mutex::Autolock _l(mLock); 2161 2162 if (checkForNewParameters_l()) { 2163 mixBufferSize = mFrameCount*mFrameSize; 2164 activeSleepTime = activeSleepTimeUs(); 2165 idleSleepTime = idleSleepTimeUs(); 2166 standbyDelay = microseconds(activeSleepTime*2); 2167 } 2168 2169 // put audio hardware into standby after short delay 2170 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2171 mSuspended) { 2172 // wait until we have something to do... 2173 if (!mStandby) { 2174 LOGV("Audio hardware entering standby, mixer %p\n", this); 2175 mOutput->standby(); 2176 mStandby = true; 2177 mBytesWritten = 0; 2178 } 2179 2180 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2181 // we're about to wait, flush the binder command buffer 2182 IPCThreadState::self()->flushCommands(); 2183 2184 if (exitPending()) break; 2185 2186 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2187 mWaitWorkCV.wait(mLock); 2188 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2189 2190 if (mMasterMute == false) { 2191 char value[PROPERTY_VALUE_MAX]; 2192 property_get("ro.audio.silent", value, "0"); 2193 if (atoi(value)) { 2194 LOGD("Silence is golden"); 2195 setMasterMute(true); 2196 } 2197 } 2198 2199 standbyTime = systemTime() + standbyDelay; 2200 sleepTime = idleSleepTime; 2201 continue; 2202 } 2203 } 2204 2205 effectChains = mEffectChains; 2206 2207 // find out which tracks need to be processed 2208 if (mActiveTracks.size() != 0) { 2209 sp<Track> t = mActiveTracks[0].promote(); 2210 if (t == 0) continue; 2211 2212 Track* const track = t.get(); 2213 audio_track_cblk_t* cblk = track->cblk(); 2214 2215 // The first time a track is added we wait 2216 // for all its buffers to be filled before processing it 2217 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 2218 !track->isPaused() && !track->isTerminated()) 2219 { 2220 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2221 2222 if (track->mFillingUpStatus == Track::FS_FILLED) { 2223 track->mFillingUpStatus = Track::FS_ACTIVE; 2224 mLeftVolFloat = mRightVolFloat = 0; 2225 mLeftVolShort = mRightVolShort = 0; 2226 if (track->mState == TrackBase::RESUMING) { 2227 track->mState = TrackBase::ACTIVE; 2228 rampVolume = true; 2229 } 2230 } else if (cblk->server != 0) { 2231 // If the track is stopped before the first frame was mixed, 2232 // do not apply ramp 2233 rampVolume = true; 2234 } 2235 // compute volume for this track 2236 float left, right; 2237 if (track->isMuted() || mMasterMute || track->isPausing() || 2238 mStreamTypes[track->type()].mute) { 2239 left = right = 0; 2240 if (track->isPausing()) { 2241 track->setPaused(); 2242 } 2243 } else { 2244 float typeVolume = mStreamTypes[track->type()].volume; 2245 float v = mMasterVolume * typeVolume; 2246 float v_clamped = v * cblk->volume[0]; 2247 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2248 left = v_clamped/MAX_GAIN; 2249 v_clamped = v * cblk->volume[1]; 2250 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2251 right = v_clamped/MAX_GAIN; 2252 } 2253 2254 if (left != mLeftVolFloat || right != mRightVolFloat) { 2255 mLeftVolFloat = left; 2256 mRightVolFloat = right; 2257 2258 // If audio HAL implements volume control, 2259 // force software volume to nominal value 2260 if (mOutput->setVolume(left, right) == NO_ERROR) { 2261 left = 1.0f; 2262 right = 1.0f; 2263 } 2264 2265 // Convert volumes from float to 8.24 2266 uint32_t vl = (uint32_t)(left * (1 << 24)); 2267 uint32_t vr = (uint32_t)(right * (1 << 24)); 2268 2269 // Delegate volume control to effect in track effect chain if needed 2270 // only one effect chain can be present on DirectOutputThread, so if 2271 // there is one, the track is connected to it 2272 if (!effectChains.isEmpty()) { 2273 // Do not ramp volume is volume is controlled by effect 2274 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2275 rampVolume = false; 2276 } 2277 } 2278 2279 // Convert volumes from 8.24 to 4.12 format 2280 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2281 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2282 leftVol = (uint16_t)v_clamped; 2283 v_clamped = (vr + (1 << 11)) >> 12; 2284 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2285 rightVol = (uint16_t)v_clamped; 2286 } else { 2287 leftVol = mLeftVolShort; 2288 rightVol = mRightVolShort; 2289 rampVolume = false; 2290 } 2291 2292 // reset retry count 2293 track->mRetryCount = kMaxTrackRetriesDirect; 2294 activeTrack = t; 2295 mixerStatus = MIXER_TRACKS_READY; 2296 } else { 2297 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2298 if (track->isStopped()) { 2299 track->reset(); 2300 } 2301 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2302 // We have consumed all the buffers of this track. 2303 // Remove it from the list of active tracks. 2304 trackToRemove = track; 2305 } else { 2306 // No buffers for this track. Give it a few chances to 2307 // fill a buffer, then remove it from active list. 2308 if (--(track->mRetryCount) <= 0) { 2309 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2310 trackToRemove = track; 2311 } else { 2312 mixerStatus = MIXER_TRACKS_ENABLED; 2313 } 2314 } 2315 } 2316 } 2317 2318 // remove all the tracks that need to be... 2319 if (UNLIKELY(trackToRemove != 0)) { 2320 mActiveTracks.remove(trackToRemove); 2321 if (!effectChains.isEmpty()) { 2322 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2323 trackToRemove->sessionId()); 2324 effectChains[0]->stopTrack(); 2325 } 2326 if (trackToRemove->isTerminated()) { 2327 mTracks.remove(trackToRemove); 2328 deleteTrackName_l(trackToRemove->mName); 2329 } 2330 } 2331 2332 lockEffectChains_l(effectChains); 2333 } 2334 2335 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2336 AudioBufferProvider::Buffer buffer; 2337 size_t frameCount = mFrameCount; 2338 curBuf = (int8_t *)mMixBuffer; 2339 // output audio to hardware 2340 while (frameCount) { 2341 buffer.frameCount = frameCount; 2342 activeTrack->getNextBuffer(&buffer); 2343 if (UNLIKELY(buffer.raw == 0)) { 2344 memset(curBuf, 0, frameCount * mFrameSize); 2345 break; 2346 } 2347 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2348 frameCount -= buffer.frameCount; 2349 curBuf += buffer.frameCount * mFrameSize; 2350 activeTrack->releaseBuffer(&buffer); 2351 } 2352 sleepTime = 0; 2353 standbyTime = systemTime() + standbyDelay; 2354 } else { 2355 if (sleepTime == 0) { 2356 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2357 sleepTime = activeSleepTime; 2358 } else { 2359 sleepTime = idleSleepTime; 2360 } 2361 } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { 2362 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2363 sleepTime = 0; 2364 } 2365 } 2366 2367 if (mSuspended) { 2368 sleepTime = idleSleepTime; 2369 } 2370 // sleepTime == 0 means we must write to audio hardware 2371 if (sleepTime == 0) { 2372 if (mixerStatus == MIXER_TRACKS_READY) { 2373 applyVolume(leftVol, rightVol, rampVolume); 2374 } 2375 for (size_t i = 0; i < effectChains.size(); i ++) { 2376 effectChains[i]->process_l(); 2377 } 2378 unlockEffectChains(effectChains); 2379 2380 mLastWriteTime = systemTime(); 2381 mInWrite = true; 2382 mBytesWritten += mixBufferSize; 2383 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 2384 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2385 mNumWrites++; 2386 mInWrite = false; 2387 mStandby = false; 2388 } else { 2389 unlockEffectChains(effectChains); 2390 usleep(sleepTime); 2391 } 2392 2393 // finally let go of removed track, without the lock held 2394 // since we can't guarantee the destructors won't acquire that 2395 // same lock. 2396 trackToRemove.clear(); 2397 activeTrack.clear(); 2398 2399 // Effect chains will be actually deleted here if they were removed from 2400 // mEffectChains list during mixing or effects processing 2401 effectChains.clear(); 2402 } 2403 2404 if (!mStandby) { 2405 mOutput->standby(); 2406 } 2407 2408 LOGV("DirectOutputThread %p exiting", this); 2409 return false; 2410} 2411 2412// getTrackName_l() must be called with ThreadBase::mLock held 2413int AudioFlinger::DirectOutputThread::getTrackName_l() 2414{ 2415 return 0; 2416} 2417 2418// deleteTrackName_l() must be called with ThreadBase::mLock held 2419void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2420{ 2421} 2422 2423// checkForNewParameters_l() must be called with ThreadBase::mLock held 2424bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2425{ 2426 bool reconfig = false; 2427 2428 while (!mNewParameters.isEmpty()) { 2429 status_t status = NO_ERROR; 2430 String8 keyValuePair = mNewParameters[0]; 2431 AudioParameter param = AudioParameter(keyValuePair); 2432 int value; 2433 2434 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2435 // do not accept frame count changes if tracks are open as the track buffer 2436 // size depends on frame count and correct behavior would not be garantied 2437 // if frame count is changed after track creation 2438 if (!mTracks.isEmpty()) { 2439 status = INVALID_OPERATION; 2440 } else { 2441 reconfig = true; 2442 } 2443 } 2444 if (status == NO_ERROR) { 2445 status = mOutput->setParameters(keyValuePair); 2446 if (!mStandby && status == INVALID_OPERATION) { 2447 mOutput->standby(); 2448 mStandby = true; 2449 mBytesWritten = 0; 2450 status = mOutput->setParameters(keyValuePair); 2451 } 2452 if (status == NO_ERROR && reconfig) { 2453 readOutputParameters(); 2454 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2455 } 2456 } 2457 2458 mNewParameters.removeAt(0); 2459 2460 mParamStatus = status; 2461 mParamCond.signal(); 2462 mWaitWorkCV.wait(mLock); 2463 } 2464 return reconfig; 2465} 2466 2467uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2468{ 2469 uint32_t time; 2470 if (AudioSystem::isLinearPCM(mFormat)) { 2471 time = (uint32_t)(mOutput->latency() * 1000) / 2; 2472 } else { 2473 time = 10000; 2474 } 2475 return time; 2476} 2477 2478uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2479{ 2480 uint32_t time; 2481 if (AudioSystem::isLinearPCM(mFormat)) { 2482 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2483 } else { 2484 time = 10000; 2485 } 2486 return time; 2487} 2488 2489// ---------------------------------------------------------------------------- 2490 2491AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2492 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2493{ 2494 mType = PlaybackThread::DUPLICATING; 2495 addOutputTrack(mainThread); 2496} 2497 2498AudioFlinger::DuplicatingThread::~DuplicatingThread() 2499{ 2500 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2501 mOutputTracks[i]->destroy(); 2502 } 2503 mOutputTracks.clear(); 2504} 2505 2506bool AudioFlinger::DuplicatingThread::threadLoop() 2507{ 2508 Vector< sp<Track> > tracksToRemove; 2509 uint32_t mixerStatus = MIXER_IDLE; 2510 nsecs_t standbyTime = systemTime(); 2511 size_t mixBufferSize = mFrameCount*mFrameSize; 2512 SortedVector< sp<OutputTrack> > outputTracks; 2513 uint32_t writeFrames = 0; 2514 uint32_t activeSleepTime = activeSleepTimeUs(); 2515 uint32_t idleSleepTime = idleSleepTimeUs(); 2516 uint32_t sleepTime = idleSleepTime; 2517 Vector< sp<EffectChain> > effectChains; 2518 2519 while (!exitPending()) 2520 { 2521 processConfigEvents(); 2522 2523 mixerStatus = MIXER_IDLE; 2524 { // scope for the mLock 2525 2526 Mutex::Autolock _l(mLock); 2527 2528 if (checkForNewParameters_l()) { 2529 mixBufferSize = mFrameCount*mFrameSize; 2530 updateWaitTime(); 2531 activeSleepTime = activeSleepTimeUs(); 2532 idleSleepTime = idleSleepTimeUs(); 2533 } 2534 2535 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2536 2537 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2538 outputTracks.add(mOutputTracks[i]); 2539 } 2540 2541 // put audio hardware into standby after short delay 2542 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2543 mSuspended) { 2544 if (!mStandby) { 2545 for (size_t i = 0; i < outputTracks.size(); i++) { 2546 outputTracks[i]->stop(); 2547 } 2548 mStandby = true; 2549 mBytesWritten = 0; 2550 } 2551 2552 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2553 // we're about to wait, flush the binder command buffer 2554 IPCThreadState::self()->flushCommands(); 2555 outputTracks.clear(); 2556 2557 if (exitPending()) break; 2558 2559 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2560 mWaitWorkCV.wait(mLock); 2561 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2562 if (mMasterMute == false) { 2563 char value[PROPERTY_VALUE_MAX]; 2564 property_get("ro.audio.silent", value, "0"); 2565 if (atoi(value)) { 2566 LOGD("Silence is golden"); 2567 setMasterMute(true); 2568 } 2569 } 2570 2571 standbyTime = systemTime() + kStandbyTimeInNsecs; 2572 sleepTime = idleSleepTime; 2573 continue; 2574 } 2575 } 2576 2577 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2578 2579 // prevent any changes in effect chain list and in each effect chain 2580 // during mixing and effect process as the audio buffers could be deleted 2581 // or modified if an effect is created or deleted 2582 lockEffectChains_l(effectChains); 2583 } 2584 2585 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2586 // mix buffers... 2587 if (outputsReady(outputTracks)) { 2588 mAudioMixer->process(); 2589 } else { 2590 memset(mMixBuffer, 0, mixBufferSize); 2591 } 2592 sleepTime = 0; 2593 writeFrames = mFrameCount; 2594 } else { 2595 if (sleepTime == 0) { 2596 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2597 sleepTime = activeSleepTime; 2598 } else { 2599 sleepTime = idleSleepTime; 2600 } 2601 } else if (mBytesWritten != 0) { 2602 // flush remaining overflow buffers in output tracks 2603 for (size_t i = 0; i < outputTracks.size(); i++) { 2604 if (outputTracks[i]->isActive()) { 2605 sleepTime = 0; 2606 writeFrames = 0; 2607 memset(mMixBuffer, 0, mixBufferSize); 2608 break; 2609 } 2610 } 2611 } 2612 } 2613 2614 if (mSuspended) { 2615 sleepTime = idleSleepTime; 2616 } 2617 // sleepTime == 0 means we must write to audio hardware 2618 if (sleepTime == 0) { 2619 for (size_t i = 0; i < effectChains.size(); i ++) { 2620 effectChains[i]->process_l(); 2621 } 2622 // enable changes in effect chain 2623 unlockEffectChains(effectChains); 2624 2625 standbyTime = systemTime() + kStandbyTimeInNsecs; 2626 for (size_t i = 0; i < outputTracks.size(); i++) { 2627 outputTracks[i]->write(mMixBuffer, writeFrames); 2628 } 2629 mStandby = false; 2630 mBytesWritten += mixBufferSize; 2631 } else { 2632 // enable changes in effect chain 2633 unlockEffectChains(effectChains); 2634 usleep(sleepTime); 2635 } 2636 2637 // finally let go of all our tracks, without the lock held 2638 // since we can't guarantee the destructors won't acquire that 2639 // same lock. 2640 tracksToRemove.clear(); 2641 outputTracks.clear(); 2642 2643 // Effect chains will be actually deleted here if they were removed from 2644 // mEffectChains list during mixing or effects processing 2645 effectChains.clear(); 2646 } 2647 2648 return false; 2649} 2650 2651void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2652{ 2653 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2654 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2655 this, 2656 mSampleRate, 2657 mFormat, 2658 mChannelCount, 2659 frameCount); 2660 if (outputTrack->cblk() != NULL) { 2661 thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); 2662 mOutputTracks.add(outputTrack); 2663 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2664 updateWaitTime(); 2665 } 2666} 2667 2668void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2669{ 2670 Mutex::Autolock _l(mLock); 2671 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2672 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2673 mOutputTracks[i]->destroy(); 2674 mOutputTracks.removeAt(i); 2675 updateWaitTime(); 2676 return; 2677 } 2678 } 2679 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2680} 2681 2682void AudioFlinger::DuplicatingThread::updateWaitTime() 2683{ 2684 mWaitTimeMs = UINT_MAX; 2685 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2686 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2687 if (strong != NULL) { 2688 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2689 if (waitTimeMs < mWaitTimeMs) { 2690 mWaitTimeMs = waitTimeMs; 2691 } 2692 } 2693 } 2694} 2695 2696 2697bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2698{ 2699 for (size_t i = 0; i < outputTracks.size(); i++) { 2700 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2701 if (thread == 0) { 2702 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2703 return false; 2704 } 2705 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2706 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2707 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2708 return false; 2709 } 2710 } 2711 return true; 2712} 2713 2714uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2715{ 2716 return (mWaitTimeMs * 1000) / 2; 2717} 2718 2719// ---------------------------------------------------------------------------- 2720 2721// TrackBase constructor must be called with AudioFlinger::mLock held 2722AudioFlinger::ThreadBase::TrackBase::TrackBase( 2723 const wp<ThreadBase>& thread, 2724 const sp<Client>& client, 2725 uint32_t sampleRate, 2726 int format, 2727 int channelCount, 2728 int frameCount, 2729 uint32_t flags, 2730 const sp<IMemory>& sharedBuffer, 2731 int sessionId) 2732 : RefBase(), 2733 mThread(thread), 2734 mClient(client), 2735 mCblk(0), 2736 mFrameCount(0), 2737 mState(IDLE), 2738 mClientTid(-1), 2739 mFormat(format), 2740 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2741 mSessionId(sessionId) 2742{ 2743 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2744 2745 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2746 size_t size = sizeof(audio_track_cblk_t); 2747 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2748 if (sharedBuffer == 0) { 2749 size += bufferSize; 2750 } 2751 2752 if (client != NULL) { 2753 mCblkMemory = client->heap()->allocate(size); 2754 if (mCblkMemory != 0) { 2755 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2756 if (mCblk) { // construct the shared structure in-place. 2757 new(mCblk) audio_track_cblk_t(); 2758 // clear all buffers 2759 mCblk->frameCount = frameCount; 2760 mCblk->sampleRate = sampleRate; 2761 mCblk->channelCount = (uint8_t)channelCount; 2762 if (sharedBuffer == 0) { 2763 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2764 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2765 // Force underrun condition to avoid false underrun callback until first data is 2766 // written to buffer 2767 mCblk->flags = CBLK_UNDERRUN_ON; 2768 } else { 2769 mBuffer = sharedBuffer->pointer(); 2770 } 2771 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2772 } 2773 } else { 2774 LOGE("not enough memory for AudioTrack size=%u", size); 2775 client->heap()->dump("AudioTrack"); 2776 return; 2777 } 2778 } else { 2779 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2780 if (mCblk) { // construct the shared structure in-place. 2781 new(mCblk) audio_track_cblk_t(); 2782 // clear all buffers 2783 mCblk->frameCount = frameCount; 2784 mCblk->sampleRate = sampleRate; 2785 mCblk->channelCount = (uint8_t)channelCount; 2786 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2787 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2788 // Force underrun condition to avoid false underrun callback until first data is 2789 // written to buffer 2790 mCblk->flags = CBLK_UNDERRUN_ON; 2791 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2792 } 2793 } 2794} 2795 2796AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2797{ 2798 if (mCblk) { 2799 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2800 if (mClient == NULL) { 2801 delete mCblk; 2802 } 2803 } 2804 mCblkMemory.clear(); // and free the shared memory 2805 if (mClient != NULL) { 2806 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2807 mClient.clear(); 2808 } 2809} 2810 2811void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2812{ 2813 buffer->raw = 0; 2814 mFrameCount = buffer->frameCount; 2815 step(); 2816 buffer->frameCount = 0; 2817} 2818 2819bool AudioFlinger::ThreadBase::TrackBase::step() { 2820 bool result; 2821 audio_track_cblk_t* cblk = this->cblk(); 2822 2823 result = cblk->stepServer(mFrameCount); 2824 if (!result) { 2825 LOGV("stepServer failed acquiring cblk mutex"); 2826 mFlags |= STEPSERVER_FAILED; 2827 } 2828 return result; 2829} 2830 2831void AudioFlinger::ThreadBase::TrackBase::reset() { 2832 audio_track_cblk_t* cblk = this->cblk(); 2833 2834 cblk->user = 0; 2835 cblk->server = 0; 2836 cblk->userBase = 0; 2837 cblk->serverBase = 0; 2838 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2839 LOGV("TrackBase::reset"); 2840} 2841 2842sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2843{ 2844 return mCblkMemory; 2845} 2846 2847int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2848 return (int)mCblk->sampleRate; 2849} 2850 2851int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2852 return (int)mCblk->channelCount; 2853} 2854 2855void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2856 audio_track_cblk_t* cblk = this->cblk(); 2857 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2858 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2859 2860 // Check validity of returned pointer in case the track control block would have been corrupted. 2861 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2862 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2863 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2864 server %d, serverBase %d, user %d, userBase %d, channelCount %d", 2865 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2866 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); 2867 return 0; 2868 } 2869 2870 return bufferStart; 2871} 2872 2873// ---------------------------------------------------------------------------- 2874 2875// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2876AudioFlinger::PlaybackThread::Track::Track( 2877 const wp<ThreadBase>& thread, 2878 const sp<Client>& client, 2879 int streamType, 2880 uint32_t sampleRate, 2881 int format, 2882 int channelCount, 2883 int frameCount, 2884 const sp<IMemory>& sharedBuffer, 2885 int sessionId) 2886 : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), 2887 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0) 2888{ 2889 if (mCblk != NULL) { 2890 sp<ThreadBase> baseThread = thread.promote(); 2891 if (baseThread != 0) { 2892 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2893 mName = playbackThread->getTrackName_l(); 2894 mMainBuffer = playbackThread->mixBuffer(); 2895 } 2896 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2897 if (mName < 0) { 2898 LOGE("no more track names available"); 2899 } 2900 mVolume[0] = 1.0f; 2901 mVolume[1] = 1.0f; 2902 mStreamType = streamType; 2903 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2904 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2905 mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); 2906 } 2907} 2908 2909AudioFlinger::PlaybackThread::Track::~Track() 2910{ 2911 LOGV("PlaybackThread::Track destructor"); 2912 sp<ThreadBase> thread = mThread.promote(); 2913 if (thread != 0) { 2914 Mutex::Autolock _l(thread->mLock); 2915 mState = TERMINATED; 2916 } 2917} 2918 2919void AudioFlinger::PlaybackThread::Track::destroy() 2920{ 2921 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2922 // by removing it from mTracks vector, so there is a risk that this Tracks's 2923 // desctructor is called. As the destructor needs to lock mLock, 2924 // we must acquire a strong reference on this Track before locking mLock 2925 // here so that the destructor is called only when exiting this function. 2926 // On the other hand, as long as Track::destroy() is only called by 2927 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2928 // this Track with its member mTrack. 2929 sp<Track> keep(this); 2930 { // scope for mLock 2931 sp<ThreadBase> thread = mThread.promote(); 2932 if (thread != 0) { 2933 if (!isOutputTrack()) { 2934 if (mState == ACTIVE || mState == RESUMING) { 2935 AudioSystem::stopOutput(thread->id(), 2936 (AudioSystem::stream_type)mStreamType, 2937 mSessionId); 2938 } 2939 AudioSystem::releaseOutput(thread->id()); 2940 } 2941 Mutex::Autolock _l(thread->mLock); 2942 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2943 playbackThread->destroyTrack_l(this); 2944 } 2945 } 2946} 2947 2948void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2949{ 2950 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", 2951 mName - AudioMixer::TRACK0, 2952 (mClient == NULL) ? getpid() : mClient->pid(), 2953 mStreamType, 2954 mFormat, 2955 mCblk->channelCount, 2956 mSessionId, 2957 mFrameCount, 2958 mState, 2959 mMute, 2960 mFillingUpStatus, 2961 mCblk->sampleRate, 2962 mCblk->volume[0], 2963 mCblk->volume[1], 2964 mCblk->server, 2965 mCblk->user, 2966 (int)mMainBuffer, 2967 (int)mAuxBuffer); 2968} 2969 2970status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 2971{ 2972 audio_track_cblk_t* cblk = this->cblk(); 2973 uint32_t framesReady; 2974 uint32_t framesReq = buffer->frameCount; 2975 2976 // Check if last stepServer failed, try to step now 2977 if (mFlags & TrackBase::STEPSERVER_FAILED) { 2978 if (!step()) goto getNextBuffer_exit; 2979 LOGV("stepServer recovered"); 2980 mFlags &= ~TrackBase::STEPSERVER_FAILED; 2981 } 2982 2983 framesReady = cblk->framesReady(); 2984 2985 if (LIKELY(framesReady)) { 2986 uint32_t s = cblk->server; 2987 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 2988 2989 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 2990 if (framesReq > framesReady) { 2991 framesReq = framesReady; 2992 } 2993 if (s + framesReq > bufferEnd) { 2994 framesReq = bufferEnd - s; 2995 } 2996 2997 buffer->raw = getBuffer(s, framesReq); 2998 if (buffer->raw == 0) goto getNextBuffer_exit; 2999 3000 buffer->frameCount = framesReq; 3001 return NO_ERROR; 3002 } 3003 3004getNextBuffer_exit: 3005 buffer->raw = 0; 3006 buffer->frameCount = 0; 3007 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3008 return NOT_ENOUGH_DATA; 3009} 3010 3011bool AudioFlinger::PlaybackThread::Track::isReady() const { 3012 if (mFillingUpStatus != FS_FILLING) return true; 3013 3014 if (mCblk->framesReady() >= mCblk->frameCount || 3015 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3016 mFillingUpStatus = FS_FILLED; 3017 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3018 return true; 3019 } 3020 return false; 3021} 3022 3023status_t AudioFlinger::PlaybackThread::Track::start() 3024{ 3025 status_t status = NO_ERROR; 3026 LOGV("start(%d), calling thread %d session %d", 3027 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3028 sp<ThreadBase> thread = mThread.promote(); 3029 if (thread != 0) { 3030 Mutex::Autolock _l(thread->mLock); 3031 int state = mState; 3032 // here the track could be either new, or restarted 3033 // in both cases "unstop" the track 3034 if (mState == PAUSED) { 3035 mState = TrackBase::RESUMING; 3036 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3037 } else { 3038 mState = TrackBase::ACTIVE; 3039 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3040 } 3041 3042 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3043 thread->mLock.unlock(); 3044 status = AudioSystem::startOutput(thread->id(), 3045 (AudioSystem::stream_type)mStreamType, 3046 mSessionId); 3047 thread->mLock.lock(); 3048 } 3049 if (status == NO_ERROR) { 3050 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3051 playbackThread->addTrack_l(this); 3052 } else { 3053 mState = state; 3054 } 3055 } else { 3056 status = BAD_VALUE; 3057 } 3058 return status; 3059} 3060 3061void AudioFlinger::PlaybackThread::Track::stop() 3062{ 3063 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3064 sp<ThreadBase> thread = mThread.promote(); 3065 if (thread != 0) { 3066 Mutex::Autolock _l(thread->mLock); 3067 int state = mState; 3068 if (mState > STOPPED) { 3069 mState = STOPPED; 3070 // If the track is not active (PAUSED and buffers full), flush buffers 3071 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3072 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3073 reset(); 3074 } 3075 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3076 } 3077 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3078 thread->mLock.unlock(); 3079 AudioSystem::stopOutput(thread->id(), 3080 (AudioSystem::stream_type)mStreamType, 3081 mSessionId); 3082 thread->mLock.lock(); 3083 } 3084 } 3085} 3086 3087void AudioFlinger::PlaybackThread::Track::pause() 3088{ 3089 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3090 sp<ThreadBase> thread = mThread.promote(); 3091 if (thread != 0) { 3092 Mutex::Autolock _l(thread->mLock); 3093 if (mState == ACTIVE || mState == RESUMING) { 3094 mState = PAUSING; 3095 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3096 if (!isOutputTrack()) { 3097 thread->mLock.unlock(); 3098 AudioSystem::stopOutput(thread->id(), 3099 (AudioSystem::stream_type)mStreamType, 3100 mSessionId); 3101 thread->mLock.lock(); 3102 } 3103 } 3104 } 3105} 3106 3107void AudioFlinger::PlaybackThread::Track::flush() 3108{ 3109 LOGV("flush(%d)", mName); 3110 sp<ThreadBase> thread = mThread.promote(); 3111 if (thread != 0) { 3112 Mutex::Autolock _l(thread->mLock); 3113 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3114 return; 3115 } 3116 // No point remaining in PAUSED state after a flush => go to 3117 // STOPPED state 3118 mState = STOPPED; 3119 3120 mCblk->lock.lock(); 3121 // NOTE: reset() will reset cblk->user and cblk->server with 3122 // the risk that at the same time, the AudioMixer is trying to read 3123 // data. In this case, getNextBuffer() would return a NULL pointer 3124 // as audio buffer => the AudioMixer code MUST always test that pointer 3125 // returned by getNextBuffer() is not NULL! 3126 reset(); 3127 mCblk->lock.unlock(); 3128 } 3129} 3130 3131void AudioFlinger::PlaybackThread::Track::reset() 3132{ 3133 // Do not reset twice to avoid discarding data written just after a flush and before 3134 // the audioflinger thread detects the track is stopped. 3135 if (!mResetDone) { 3136 TrackBase::reset(); 3137 // Force underrun condition to avoid false underrun callback until first data is 3138 // written to buffer 3139 mCblk->flags |= CBLK_UNDERRUN_ON; 3140 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3141 mFillingUpStatus = FS_FILLING; 3142 mResetDone = true; 3143 } 3144} 3145 3146void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3147{ 3148 mMute = muted; 3149} 3150 3151void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3152{ 3153 mVolume[0] = left; 3154 mVolume[1] = right; 3155} 3156 3157status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3158{ 3159 status_t status = DEAD_OBJECT; 3160 sp<ThreadBase> thread = mThread.promote(); 3161 if (thread != 0) { 3162 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3163 status = playbackThread->attachAuxEffect(this, EffectId); 3164 } 3165 return status; 3166} 3167 3168void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3169{ 3170 mAuxEffectId = EffectId; 3171 mAuxBuffer = buffer; 3172} 3173 3174// ---------------------------------------------------------------------------- 3175 3176// RecordTrack constructor must be called with AudioFlinger::mLock held 3177AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3178 const wp<ThreadBase>& thread, 3179 const sp<Client>& client, 3180 uint32_t sampleRate, 3181 int format, 3182 int channelCount, 3183 int frameCount, 3184 uint32_t flags, 3185 int sessionId) 3186 : TrackBase(thread, client, sampleRate, format, 3187 channelCount, frameCount, flags, 0, sessionId), 3188 mOverflow(false) 3189{ 3190 if (mCblk != NULL) { 3191 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3192 if (format == AudioSystem::PCM_16_BIT) { 3193 mCblk->frameSize = channelCount * sizeof(int16_t); 3194 } else if (format == AudioSystem::PCM_8_BIT) { 3195 mCblk->frameSize = channelCount * sizeof(int8_t); 3196 } else { 3197 mCblk->frameSize = sizeof(int8_t); 3198 } 3199 } 3200} 3201 3202AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3203{ 3204 sp<ThreadBase> thread = mThread.promote(); 3205 if (thread != 0) { 3206 AudioSystem::releaseInput(thread->id()); 3207 } 3208} 3209 3210status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3211{ 3212 audio_track_cblk_t* cblk = this->cblk(); 3213 uint32_t framesAvail; 3214 uint32_t framesReq = buffer->frameCount; 3215 3216 // Check if last stepServer failed, try to step now 3217 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3218 if (!step()) goto getNextBuffer_exit; 3219 LOGV("stepServer recovered"); 3220 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3221 } 3222 3223 framesAvail = cblk->framesAvailable_l(); 3224 3225 if (LIKELY(framesAvail)) { 3226 uint32_t s = cblk->server; 3227 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3228 3229 if (framesReq > framesAvail) { 3230 framesReq = framesAvail; 3231 } 3232 if (s + framesReq > bufferEnd) { 3233 framesReq = bufferEnd - s; 3234 } 3235 3236 buffer->raw = getBuffer(s, framesReq); 3237 if (buffer->raw == 0) goto getNextBuffer_exit; 3238 3239 buffer->frameCount = framesReq; 3240 return NO_ERROR; 3241 } 3242 3243getNextBuffer_exit: 3244 buffer->raw = 0; 3245 buffer->frameCount = 0; 3246 return NOT_ENOUGH_DATA; 3247} 3248 3249status_t AudioFlinger::RecordThread::RecordTrack::start() 3250{ 3251 sp<ThreadBase> thread = mThread.promote(); 3252 if (thread != 0) { 3253 RecordThread *recordThread = (RecordThread *)thread.get(); 3254 return recordThread->start(this); 3255 } else { 3256 return BAD_VALUE; 3257 } 3258} 3259 3260void AudioFlinger::RecordThread::RecordTrack::stop() 3261{ 3262 sp<ThreadBase> thread = mThread.promote(); 3263 if (thread != 0) { 3264 RecordThread *recordThread = (RecordThread *)thread.get(); 3265 recordThread->stop(this); 3266 TrackBase::reset(); 3267 // Force overerrun condition to avoid false overrun callback until first data is 3268 // read from buffer 3269 mCblk->flags |= CBLK_UNDERRUN_ON; 3270 } 3271} 3272 3273void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3274{ 3275 snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", 3276 (mClient == NULL) ? getpid() : mClient->pid(), 3277 mFormat, 3278 mCblk->channelCount, 3279 mSessionId, 3280 mFrameCount, 3281 mState, 3282 mCblk->sampleRate, 3283 mCblk->server, 3284 mCblk->user); 3285} 3286 3287 3288// ---------------------------------------------------------------------------- 3289 3290AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3291 const wp<ThreadBase>& thread, 3292 DuplicatingThread *sourceThread, 3293 uint32_t sampleRate, 3294 int format, 3295 int channelCount, 3296 int frameCount) 3297 : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), 3298 mActive(false), mSourceThread(sourceThread) 3299{ 3300 3301 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3302 if (mCblk != NULL) { 3303 mCblk->flags |= CBLK_DIRECTION_OUT; 3304 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3305 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3306 mOutBuffer.frameCount = 0; 3307 playbackThread->mTracks.add(this); 3308 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", 3309 mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); 3310 } else { 3311 LOGW("Error creating output track on thread %p", playbackThread); 3312 } 3313} 3314 3315AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3316{ 3317 clearBufferQueue(); 3318} 3319 3320status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3321{ 3322 status_t status = Track::start(); 3323 if (status != NO_ERROR) { 3324 return status; 3325 } 3326 3327 mActive = true; 3328 mRetryCount = 127; 3329 return status; 3330} 3331 3332void AudioFlinger::PlaybackThread::OutputTrack::stop() 3333{ 3334 Track::stop(); 3335 clearBufferQueue(); 3336 mOutBuffer.frameCount = 0; 3337 mActive = false; 3338} 3339 3340bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3341{ 3342 Buffer *pInBuffer; 3343 Buffer inBuffer; 3344 uint32_t channelCount = mCblk->channelCount; 3345 bool outputBufferFull = false; 3346 inBuffer.frameCount = frames; 3347 inBuffer.i16 = data; 3348 3349 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3350 3351 if (!mActive && frames != 0) { 3352 start(); 3353 sp<ThreadBase> thread = mThread.promote(); 3354 if (thread != 0) { 3355 MixerThread *mixerThread = (MixerThread *)thread.get(); 3356 if (mCblk->frameCount > frames){ 3357 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3358 uint32_t startFrames = (mCblk->frameCount - frames); 3359 pInBuffer = new Buffer; 3360 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3361 pInBuffer->frameCount = startFrames; 3362 pInBuffer->i16 = pInBuffer->mBuffer; 3363 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3364 mBufferQueue.add(pInBuffer); 3365 } else { 3366 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3367 } 3368 } 3369 } 3370 } 3371 3372 while (waitTimeLeftMs) { 3373 // First write pending buffers, then new data 3374 if (mBufferQueue.size()) { 3375 pInBuffer = mBufferQueue.itemAt(0); 3376 } else { 3377 pInBuffer = &inBuffer; 3378 } 3379 3380 if (pInBuffer->frameCount == 0) { 3381 break; 3382 } 3383 3384 if (mOutBuffer.frameCount == 0) { 3385 mOutBuffer.frameCount = pInBuffer->frameCount; 3386 nsecs_t startTime = systemTime(); 3387 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3388 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3389 outputBufferFull = true; 3390 break; 3391 } 3392 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3393 if (waitTimeLeftMs >= waitTimeMs) { 3394 waitTimeLeftMs -= waitTimeMs; 3395 } else { 3396 waitTimeLeftMs = 0; 3397 } 3398 } 3399 3400 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3401 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3402 mCblk->stepUser(outFrames); 3403 pInBuffer->frameCount -= outFrames; 3404 pInBuffer->i16 += outFrames * channelCount; 3405 mOutBuffer.frameCount -= outFrames; 3406 mOutBuffer.i16 += outFrames * channelCount; 3407 3408 if (pInBuffer->frameCount == 0) { 3409 if (mBufferQueue.size()) { 3410 mBufferQueue.removeAt(0); 3411 delete [] pInBuffer->mBuffer; 3412 delete pInBuffer; 3413 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3414 } else { 3415 break; 3416 } 3417 } 3418 } 3419 3420 // If we could not write all frames, allocate a buffer and queue it for next time. 3421 if (inBuffer.frameCount) { 3422 sp<ThreadBase> thread = mThread.promote(); 3423 if (thread != 0 && !thread->standby()) { 3424 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3425 pInBuffer = new Buffer; 3426 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3427 pInBuffer->frameCount = inBuffer.frameCount; 3428 pInBuffer->i16 = pInBuffer->mBuffer; 3429 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3430 mBufferQueue.add(pInBuffer); 3431 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3432 } else { 3433 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3434 } 3435 } 3436 } 3437 3438 // Calling write() with a 0 length buffer, means that no more data will be written: 3439 // If no more buffers are pending, fill output track buffer to make sure it is started 3440 // by output mixer. 3441 if (frames == 0 && mBufferQueue.size() == 0) { 3442 if (mCblk->user < mCblk->frameCount) { 3443 frames = mCblk->frameCount - mCblk->user; 3444 pInBuffer = new Buffer; 3445 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3446 pInBuffer->frameCount = frames; 3447 pInBuffer->i16 = pInBuffer->mBuffer; 3448 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3449 mBufferQueue.add(pInBuffer); 3450 } else if (mActive) { 3451 stop(); 3452 } 3453 } 3454 3455 return outputBufferFull; 3456} 3457 3458status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3459{ 3460 int active; 3461 status_t result; 3462 audio_track_cblk_t* cblk = mCblk; 3463 uint32_t framesReq = buffer->frameCount; 3464 3465// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3466 buffer->frameCount = 0; 3467 3468 uint32_t framesAvail = cblk->framesAvailable(); 3469 3470 3471 if (framesAvail == 0) { 3472 Mutex::Autolock _l(cblk->lock); 3473 goto start_loop_here; 3474 while (framesAvail == 0) { 3475 active = mActive; 3476 if (UNLIKELY(!active)) { 3477 LOGV("Not active and NO_MORE_BUFFERS"); 3478 return AudioTrack::NO_MORE_BUFFERS; 3479 } 3480 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3481 if (result != NO_ERROR) { 3482 return AudioTrack::NO_MORE_BUFFERS; 3483 } 3484 // read the server count again 3485 start_loop_here: 3486 framesAvail = cblk->framesAvailable_l(); 3487 } 3488 } 3489 3490// if (framesAvail < framesReq) { 3491// return AudioTrack::NO_MORE_BUFFERS; 3492// } 3493 3494 if (framesReq > framesAvail) { 3495 framesReq = framesAvail; 3496 } 3497 3498 uint32_t u = cblk->user; 3499 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3500 3501 if (u + framesReq > bufferEnd) { 3502 framesReq = bufferEnd - u; 3503 } 3504 3505 buffer->frameCount = framesReq; 3506 buffer->raw = (void *)cblk->buffer(u); 3507 return NO_ERROR; 3508} 3509 3510 3511void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3512{ 3513 size_t size = mBufferQueue.size(); 3514 Buffer *pBuffer; 3515 3516 for (size_t i = 0; i < size; i++) { 3517 pBuffer = mBufferQueue.itemAt(i); 3518 delete [] pBuffer->mBuffer; 3519 delete pBuffer; 3520 } 3521 mBufferQueue.clear(); 3522} 3523 3524// ---------------------------------------------------------------------------- 3525 3526AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3527 : RefBase(), 3528 mAudioFlinger(audioFlinger), 3529 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3530 mPid(pid) 3531{ 3532 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3533} 3534 3535// Client destructor must be called with AudioFlinger::mLock held 3536AudioFlinger::Client::~Client() 3537{ 3538 mAudioFlinger->removeClient_l(mPid); 3539} 3540 3541const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3542{ 3543 return mMemoryDealer; 3544} 3545 3546// ---------------------------------------------------------------------------- 3547 3548AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3549 const sp<IAudioFlingerClient>& client, 3550 pid_t pid) 3551 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3552{ 3553} 3554 3555AudioFlinger::NotificationClient::~NotificationClient() 3556{ 3557 mClient.clear(); 3558} 3559 3560void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3561{ 3562 sp<NotificationClient> keep(this); 3563 { 3564 mAudioFlinger->removeNotificationClient(mPid); 3565 } 3566} 3567 3568// ---------------------------------------------------------------------------- 3569 3570AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3571 : BnAudioTrack(), 3572 mTrack(track) 3573{ 3574} 3575 3576AudioFlinger::TrackHandle::~TrackHandle() { 3577 // just stop the track on deletion, associated resources 3578 // will be freed from the main thread once all pending buffers have 3579 // been played. Unless it's not in the active track list, in which 3580 // case we free everything now... 3581 mTrack->destroy(); 3582} 3583 3584status_t AudioFlinger::TrackHandle::start() { 3585 return mTrack->start(); 3586} 3587 3588void AudioFlinger::TrackHandle::stop() { 3589 mTrack->stop(); 3590} 3591 3592void AudioFlinger::TrackHandle::flush() { 3593 mTrack->flush(); 3594} 3595 3596void AudioFlinger::TrackHandle::mute(bool e) { 3597 mTrack->mute(e); 3598} 3599 3600void AudioFlinger::TrackHandle::pause() { 3601 mTrack->pause(); 3602} 3603 3604void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3605 mTrack->setVolume(left, right); 3606} 3607 3608sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3609 return mTrack->getCblk(); 3610} 3611 3612status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3613{ 3614 return mTrack->attachAuxEffect(EffectId); 3615} 3616 3617status_t AudioFlinger::TrackHandle::onTransact( 3618 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3619{ 3620 return BnAudioTrack::onTransact(code, data, reply, flags); 3621} 3622 3623// ---------------------------------------------------------------------------- 3624 3625sp<IAudioRecord> AudioFlinger::openRecord( 3626 pid_t pid, 3627 int input, 3628 uint32_t sampleRate, 3629 int format, 3630 int channelCount, 3631 int frameCount, 3632 uint32_t flags, 3633 int *sessionId, 3634 status_t *status) 3635{ 3636 sp<RecordThread::RecordTrack> recordTrack; 3637 sp<RecordHandle> recordHandle; 3638 sp<Client> client; 3639 wp<Client> wclient; 3640 status_t lStatus; 3641 RecordThread *thread; 3642 size_t inFrameCount; 3643 int lSessionId; 3644 3645 // check calling permissions 3646 if (!recordingAllowed()) { 3647 lStatus = PERMISSION_DENIED; 3648 goto Exit; 3649 } 3650 3651 // add client to list 3652 { // scope for mLock 3653 Mutex::Autolock _l(mLock); 3654 thread = checkRecordThread_l(input); 3655 if (thread == NULL) { 3656 lStatus = BAD_VALUE; 3657 goto Exit; 3658 } 3659 3660 wclient = mClients.valueFor(pid); 3661 if (wclient != NULL) { 3662 client = wclient.promote(); 3663 } else { 3664 client = new Client(this, pid); 3665 mClients.add(pid, client); 3666 } 3667 3668 // If no audio session id is provided, create one here 3669 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 3670 lSessionId = *sessionId; 3671 } else { 3672 lSessionId = nextUniqueId(); 3673 if (sessionId != NULL) { 3674 *sessionId = lSessionId; 3675 } 3676 } 3677 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3678 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3679 format, channelCount, frameCount, flags, lSessionId); 3680 } 3681 if (recordTrack->getCblk() == NULL) { 3682 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3683 // destructor is called by the TrackBase destructor with mLock held 3684 client.clear(); 3685 recordTrack.clear(); 3686 lStatus = NO_MEMORY; 3687 goto Exit; 3688 } 3689 3690 // return to handle to client 3691 recordHandle = new RecordHandle(recordTrack); 3692 lStatus = NO_ERROR; 3693 3694Exit: 3695 if (status) { 3696 *status = lStatus; 3697 } 3698 return recordHandle; 3699} 3700 3701// ---------------------------------------------------------------------------- 3702 3703AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3704 : BnAudioRecord(), 3705 mRecordTrack(recordTrack) 3706{ 3707} 3708 3709AudioFlinger::RecordHandle::~RecordHandle() { 3710 stop(); 3711} 3712 3713status_t AudioFlinger::RecordHandle::start() { 3714 LOGV("RecordHandle::start()"); 3715 return mRecordTrack->start(); 3716} 3717 3718void AudioFlinger::RecordHandle::stop() { 3719 LOGV("RecordHandle::stop()"); 3720 mRecordTrack->stop(); 3721} 3722 3723sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3724 return mRecordTrack->getCblk(); 3725} 3726 3727status_t AudioFlinger::RecordHandle::onTransact( 3728 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3729{ 3730 return BnAudioRecord::onTransact(code, data, reply, flags); 3731} 3732 3733// ---------------------------------------------------------------------------- 3734 3735AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : 3736 ThreadBase(audioFlinger, id), 3737 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3738{ 3739 mReqChannelCount = AudioSystem::popCount(channels); 3740 mReqSampleRate = sampleRate; 3741 readInputParameters(); 3742} 3743 3744 3745AudioFlinger::RecordThread::~RecordThread() 3746{ 3747 delete[] mRsmpInBuffer; 3748 if (mResampler != 0) { 3749 delete mResampler; 3750 delete[] mRsmpOutBuffer; 3751 } 3752} 3753 3754void AudioFlinger::RecordThread::onFirstRef() 3755{ 3756 const size_t SIZE = 256; 3757 char buffer[SIZE]; 3758 3759 snprintf(buffer, SIZE, "Record Thread %p", this); 3760 3761 run(buffer, PRIORITY_URGENT_AUDIO); 3762} 3763 3764bool AudioFlinger::RecordThread::threadLoop() 3765{ 3766 AudioBufferProvider::Buffer buffer; 3767 sp<RecordTrack> activeTrack; 3768 3769 // start recording 3770 while (!exitPending()) { 3771 3772 processConfigEvents(); 3773 3774 { // scope for mLock 3775 Mutex::Autolock _l(mLock); 3776 checkForNewParameters_l(); 3777 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3778 if (!mStandby) { 3779 mInput->standby(); 3780 mStandby = true; 3781 } 3782 3783 if (exitPending()) break; 3784 3785 LOGV("RecordThread: loop stopping"); 3786 // go to sleep 3787 mWaitWorkCV.wait(mLock); 3788 LOGV("RecordThread: loop starting"); 3789 continue; 3790 } 3791 if (mActiveTrack != 0) { 3792 if (mActiveTrack->mState == TrackBase::PAUSING) { 3793 if (!mStandby) { 3794 mInput->standby(); 3795 mStandby = true; 3796 } 3797 mActiveTrack.clear(); 3798 mStartStopCond.broadcast(); 3799 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3800 if (mReqChannelCount != mActiveTrack->channelCount()) { 3801 mActiveTrack.clear(); 3802 mStartStopCond.broadcast(); 3803 } else if (mBytesRead != 0) { 3804 // record start succeeds only if first read from audio input 3805 // succeeds 3806 if (mBytesRead > 0) { 3807 mActiveTrack->mState = TrackBase::ACTIVE; 3808 } else { 3809 mActiveTrack.clear(); 3810 } 3811 mStartStopCond.broadcast(); 3812 } 3813 mStandby = false; 3814 } 3815 } 3816 } 3817 3818 if (mActiveTrack != 0) { 3819 if (mActiveTrack->mState != TrackBase::ACTIVE && 3820 mActiveTrack->mState != TrackBase::RESUMING) { 3821 usleep(5000); 3822 continue; 3823 } 3824 buffer.frameCount = mFrameCount; 3825 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3826 size_t framesOut = buffer.frameCount; 3827 if (mResampler == 0) { 3828 // no resampling 3829 while (framesOut) { 3830 size_t framesIn = mFrameCount - mRsmpInIndex; 3831 if (framesIn) { 3832 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3833 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3834 if (framesIn > framesOut) 3835 framesIn = framesOut; 3836 mRsmpInIndex += framesIn; 3837 framesOut -= framesIn; 3838 if ((int)mChannelCount == mReqChannelCount || 3839 mFormat != AudioSystem::PCM_16_BIT) { 3840 memcpy(dst, src, framesIn * mFrameSize); 3841 } else { 3842 int16_t *src16 = (int16_t *)src; 3843 int16_t *dst16 = (int16_t *)dst; 3844 if (mChannelCount == 1) { 3845 while (framesIn--) { 3846 *dst16++ = *src16; 3847 *dst16++ = *src16++; 3848 } 3849 } else { 3850 while (framesIn--) { 3851 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3852 src16 += 2; 3853 } 3854 } 3855 } 3856 } 3857 if (framesOut && mFrameCount == mRsmpInIndex) { 3858 if (framesOut == mFrameCount && 3859 ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { 3860 mBytesRead = mInput->read(buffer.raw, mInputBytes); 3861 framesOut = 0; 3862 } else { 3863 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 3864 mRsmpInIndex = 0; 3865 } 3866 if (mBytesRead < 0) { 3867 LOGE("Error reading audio input"); 3868 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3869 // Force input into standby so that it tries to 3870 // recover at next read attempt 3871 mInput->standby(); 3872 usleep(5000); 3873 } 3874 mRsmpInIndex = mFrameCount; 3875 framesOut = 0; 3876 buffer.frameCount = 0; 3877 } 3878 } 3879 } 3880 } else { 3881 // resampling 3882 3883 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3884 // alter output frame count as if we were expecting stereo samples 3885 if (mChannelCount == 1 && mReqChannelCount == 1) { 3886 framesOut >>= 1; 3887 } 3888 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3889 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3890 // are 32 bit aligned which should be always true. 3891 if (mChannelCount == 2 && mReqChannelCount == 1) { 3892 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3893 // the resampler always outputs stereo samples: do post stereo to mono conversion 3894 int16_t *src = (int16_t *)mRsmpOutBuffer; 3895 int16_t *dst = buffer.i16; 3896 while (framesOut--) { 3897 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3898 src += 2; 3899 } 3900 } else { 3901 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3902 } 3903 3904 } 3905 mActiveTrack->releaseBuffer(&buffer); 3906 mActiveTrack->overflow(); 3907 } 3908 // client isn't retrieving buffers fast enough 3909 else { 3910 if (!mActiveTrack->setOverflow()) 3911 LOGW("RecordThread: buffer overflow"); 3912 // Release the processor for a while before asking for a new buffer. 3913 // This will give the application more chance to read from the buffer and 3914 // clear the overflow. 3915 usleep(5000); 3916 } 3917 } 3918 } 3919 3920 if (!mStandby) { 3921 mInput->standby(); 3922 } 3923 mActiveTrack.clear(); 3924 3925 mStartStopCond.broadcast(); 3926 3927 LOGV("RecordThread %p exiting", this); 3928 return false; 3929} 3930 3931status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3932{ 3933 LOGV("RecordThread::start"); 3934 sp <ThreadBase> strongMe = this; 3935 status_t status = NO_ERROR; 3936 { 3937 AutoMutex lock(&mLock); 3938 if (mActiveTrack != 0) { 3939 if (recordTrack != mActiveTrack.get()) { 3940 status = -EBUSY; 3941 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3942 mActiveTrack->mState = TrackBase::ACTIVE; 3943 } 3944 return status; 3945 } 3946 3947 recordTrack->mState = TrackBase::IDLE; 3948 mActiveTrack = recordTrack; 3949 mLock.unlock(); 3950 status_t status = AudioSystem::startInput(mId); 3951 mLock.lock(); 3952 if (status != NO_ERROR) { 3953 mActiveTrack.clear(); 3954 return status; 3955 } 3956 mActiveTrack->mState = TrackBase::RESUMING; 3957 mRsmpInIndex = mFrameCount; 3958 mBytesRead = 0; 3959 // signal thread to start 3960 LOGV("Signal record thread"); 3961 mWaitWorkCV.signal(); 3962 // do not wait for mStartStopCond if exiting 3963 if (mExiting) { 3964 mActiveTrack.clear(); 3965 status = INVALID_OPERATION; 3966 goto startError; 3967 } 3968 mStartStopCond.wait(mLock); 3969 if (mActiveTrack == 0) { 3970 LOGV("Record failed to start"); 3971 status = BAD_VALUE; 3972 goto startError; 3973 } 3974 LOGV("Record started OK"); 3975 return status; 3976 } 3977startError: 3978 AudioSystem::stopInput(mId); 3979 return status; 3980} 3981 3982void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 3983 LOGV("RecordThread::stop"); 3984 sp <ThreadBase> strongMe = this; 3985 { 3986 AutoMutex lock(&mLock); 3987 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 3988 mActiveTrack->mState = TrackBase::PAUSING; 3989 // do not wait for mStartStopCond if exiting 3990 if (mExiting) { 3991 return; 3992 } 3993 mStartStopCond.wait(mLock); 3994 // if we have been restarted, recordTrack == mActiveTrack.get() here 3995 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 3996 mLock.unlock(); 3997 AudioSystem::stopInput(mId); 3998 mLock.lock(); 3999 LOGV("Record stopped OK"); 4000 } 4001 } 4002 } 4003} 4004 4005status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4006{ 4007 const size_t SIZE = 256; 4008 char buffer[SIZE]; 4009 String8 result; 4010 pid_t pid = 0; 4011 4012 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4013 result.append(buffer); 4014 4015 if (mActiveTrack != 0) { 4016 result.append("Active Track:\n"); 4017 result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); 4018 mActiveTrack->dump(buffer, SIZE); 4019 result.append(buffer); 4020 4021 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4022 result.append(buffer); 4023 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4024 result.append(buffer); 4025 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4026 result.append(buffer); 4027 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4028 result.append(buffer); 4029 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4030 result.append(buffer); 4031 4032 4033 } else { 4034 result.append("No record client\n"); 4035 } 4036 write(fd, result.string(), result.size()); 4037 4038 dumpBase(fd, args); 4039 4040 return NO_ERROR; 4041} 4042 4043status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4044{ 4045 size_t framesReq = buffer->frameCount; 4046 size_t framesReady = mFrameCount - mRsmpInIndex; 4047 int channelCount; 4048 4049 if (framesReady == 0) { 4050 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 4051 if (mBytesRead < 0) { 4052 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4053 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4054 // Force input into standby so that it tries to 4055 // recover at next read attempt 4056 mInput->standby(); 4057 usleep(5000); 4058 } 4059 buffer->raw = 0; 4060 buffer->frameCount = 0; 4061 return NOT_ENOUGH_DATA; 4062 } 4063 mRsmpInIndex = 0; 4064 framesReady = mFrameCount; 4065 } 4066 4067 if (framesReq > framesReady) { 4068 framesReq = framesReady; 4069 } 4070 4071 if (mChannelCount == 1 && mReqChannelCount == 2) { 4072 channelCount = 1; 4073 } else { 4074 channelCount = 2; 4075 } 4076 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4077 buffer->frameCount = framesReq; 4078 return NO_ERROR; 4079} 4080 4081void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4082{ 4083 mRsmpInIndex += buffer->frameCount; 4084 buffer->frameCount = 0; 4085} 4086 4087bool AudioFlinger::RecordThread::checkForNewParameters_l() 4088{ 4089 bool reconfig = false; 4090 4091 while (!mNewParameters.isEmpty()) { 4092 status_t status = NO_ERROR; 4093 String8 keyValuePair = mNewParameters[0]; 4094 AudioParameter param = AudioParameter(keyValuePair); 4095 int value; 4096 int reqFormat = mFormat; 4097 int reqSamplingRate = mReqSampleRate; 4098 int reqChannelCount = mReqChannelCount; 4099 4100 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4101 reqSamplingRate = value; 4102 reconfig = true; 4103 } 4104 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4105 reqFormat = value; 4106 reconfig = true; 4107 } 4108 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4109 reqChannelCount = AudioSystem::popCount(value); 4110 reconfig = true; 4111 } 4112 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4113 // do not accept frame count changes if tracks are open as the track buffer 4114 // size depends on frame count and correct behavior would not be garantied 4115 // if frame count is changed after track creation 4116 if (mActiveTrack != 0) { 4117 status = INVALID_OPERATION; 4118 } else { 4119 reconfig = true; 4120 } 4121 } 4122 if (status == NO_ERROR) { 4123 status = mInput->setParameters(keyValuePair); 4124 if (status == INVALID_OPERATION) { 4125 mInput->standby(); 4126 status = mInput->setParameters(keyValuePair); 4127 } 4128 if (reconfig) { 4129 if (status == BAD_VALUE && 4130 reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && 4131 ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && 4132 (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { 4133 status = NO_ERROR; 4134 } 4135 if (status == NO_ERROR) { 4136 readInputParameters(); 4137 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4138 } 4139 } 4140 } 4141 4142 mNewParameters.removeAt(0); 4143 4144 mParamStatus = status; 4145 mParamCond.signal(); 4146 mWaitWorkCV.wait(mLock); 4147 } 4148 return reconfig; 4149} 4150 4151String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4152{ 4153 return mInput->getParameters(keys); 4154} 4155 4156void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4157 AudioSystem::OutputDescriptor desc; 4158 void *param2 = 0; 4159 4160 switch (event) { 4161 case AudioSystem::INPUT_OPENED: 4162 case AudioSystem::INPUT_CONFIG_CHANGED: 4163 desc.channels = mChannels; 4164 desc.samplingRate = mSampleRate; 4165 desc.format = mFormat; 4166 desc.frameCount = mFrameCount; 4167 desc.latency = 0; 4168 param2 = &desc; 4169 break; 4170 4171 case AudioSystem::INPUT_CLOSED: 4172 default: 4173 break; 4174 } 4175 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4176} 4177 4178void AudioFlinger::RecordThread::readInputParameters() 4179{ 4180 if (mRsmpInBuffer) delete mRsmpInBuffer; 4181 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4182 if (mResampler) delete mResampler; 4183 mResampler = 0; 4184 4185 mSampleRate = mInput->sampleRate(); 4186 mChannels = mInput->channels(); 4187 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 4188 mFormat = mInput->format(); 4189 mFrameSize = (uint16_t)mInput->frameSize(); 4190 mInputBytes = mInput->bufferSize(); 4191 mFrameCount = mInputBytes / mFrameSize; 4192 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4193 4194 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4195 { 4196 int channelCount; 4197 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4198 // stereo to mono post process as the resampler always outputs stereo. 4199 if (mChannelCount == 1 && mReqChannelCount == 2) { 4200 channelCount = 1; 4201 } else { 4202 channelCount = 2; 4203 } 4204 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4205 mResampler->setSampleRate(mSampleRate); 4206 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4207 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4208 4209 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4210 if (mChannelCount == 1 && mReqChannelCount == 1) { 4211 mFrameCount >>= 1; 4212 } 4213 4214 } 4215 mRsmpInIndex = mFrameCount; 4216} 4217 4218unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4219{ 4220 return mInput->getInputFramesLost(); 4221} 4222 4223// ---------------------------------------------------------------------------- 4224 4225int AudioFlinger::openOutput(uint32_t *pDevices, 4226 uint32_t *pSamplingRate, 4227 uint32_t *pFormat, 4228 uint32_t *pChannels, 4229 uint32_t *pLatencyMs, 4230 uint32_t flags) 4231{ 4232 status_t status; 4233 PlaybackThread *thread = NULL; 4234 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4235 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4236 uint32_t format = pFormat ? *pFormat : 0; 4237 uint32_t channels = pChannels ? *pChannels : 0; 4238 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4239 4240 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4241 pDevices ? *pDevices : 0, 4242 samplingRate, 4243 format, 4244 channels, 4245 flags); 4246 4247 if (pDevices == NULL || *pDevices == 0) { 4248 return 0; 4249 } 4250 Mutex::Autolock _l(mLock); 4251 4252 AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, 4253 (int *)&format, 4254 &channels, 4255 &samplingRate, 4256 &status); 4257 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4258 output, 4259 samplingRate, 4260 format, 4261 channels, 4262 status); 4263 4264 mHardwareStatus = AUDIO_HW_IDLE; 4265 if (output != 0) { 4266 int id = nextUniqueId(); 4267 if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || 4268 (format != AudioSystem::PCM_16_BIT) || 4269 (channels != AudioSystem::CHANNEL_OUT_STEREO)) { 4270 thread = new DirectOutputThread(this, output, id, *pDevices); 4271 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4272 } else { 4273 thread = new MixerThread(this, output, id, *pDevices); 4274 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4275 4276#ifdef LVMX 4277 unsigned bitsPerSample = 4278 (format == AudioSystem::PCM_16_BIT) ? 16 : 4279 ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); 4280 unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; 4281 int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); 4282 4283 LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); 4284 LifeVibes::setDevice(audioOutputType, *pDevices); 4285#endif 4286 4287 } 4288 mPlaybackThreads.add(id, thread); 4289 4290 if (pSamplingRate) *pSamplingRate = samplingRate; 4291 if (pFormat) *pFormat = format; 4292 if (pChannels) *pChannels = channels; 4293 if (pLatencyMs) *pLatencyMs = thread->latency(); 4294 4295 // notify client processes of the new output creation 4296 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4297 return id; 4298 } 4299 4300 return 0; 4301} 4302 4303int AudioFlinger::openDuplicateOutput(int output1, int output2) 4304{ 4305 Mutex::Autolock _l(mLock); 4306 MixerThread *thread1 = checkMixerThread_l(output1); 4307 MixerThread *thread2 = checkMixerThread_l(output2); 4308 4309 if (thread1 == NULL || thread2 == NULL) { 4310 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4311 return 0; 4312 } 4313 4314 int id = nextUniqueId(); 4315 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4316 thread->addOutputTrack(thread2); 4317 mPlaybackThreads.add(id, thread); 4318 // notify client processes of the new output creation 4319 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4320 return id; 4321} 4322 4323status_t AudioFlinger::closeOutput(int output) 4324{ 4325 // keep strong reference on the playback thread so that 4326 // it is not destroyed while exit() is executed 4327 sp <PlaybackThread> thread; 4328 { 4329 Mutex::Autolock _l(mLock); 4330 thread = checkPlaybackThread_l(output); 4331 if (thread == NULL) { 4332 return BAD_VALUE; 4333 } 4334 4335 LOGV("closeOutput() %d", output); 4336 4337 if (thread->type() == PlaybackThread::MIXER) { 4338 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4339 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4340 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4341 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4342 } 4343 } 4344 } 4345 void *param2 = 0; 4346 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4347 mPlaybackThreads.removeItem(output); 4348 } 4349 thread->exit(); 4350 4351 if (thread->type() != PlaybackThread::DUPLICATING) { 4352 mAudioHardware->closeOutputStream(thread->getOutput()); 4353 } 4354 return NO_ERROR; 4355} 4356 4357status_t AudioFlinger::suspendOutput(int output) 4358{ 4359 Mutex::Autolock _l(mLock); 4360 PlaybackThread *thread = checkPlaybackThread_l(output); 4361 4362 if (thread == NULL) { 4363 return BAD_VALUE; 4364 } 4365 4366 LOGV("suspendOutput() %d", output); 4367 thread->suspend(); 4368 4369 return NO_ERROR; 4370} 4371 4372status_t AudioFlinger::restoreOutput(int output) 4373{ 4374 Mutex::Autolock _l(mLock); 4375 PlaybackThread *thread = checkPlaybackThread_l(output); 4376 4377 if (thread == NULL) { 4378 return BAD_VALUE; 4379 } 4380 4381 LOGV("restoreOutput() %d", output); 4382 4383 thread->restore(); 4384 4385 return NO_ERROR; 4386} 4387 4388int AudioFlinger::openInput(uint32_t *pDevices, 4389 uint32_t *pSamplingRate, 4390 uint32_t *pFormat, 4391 uint32_t *pChannels, 4392 uint32_t acoustics) 4393{ 4394 status_t status; 4395 RecordThread *thread = NULL; 4396 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4397 uint32_t format = pFormat ? *pFormat : 0; 4398 uint32_t channels = pChannels ? *pChannels : 0; 4399 uint32_t reqSamplingRate = samplingRate; 4400 uint32_t reqFormat = format; 4401 uint32_t reqChannels = channels; 4402 4403 if (pDevices == NULL || *pDevices == 0) { 4404 return 0; 4405 } 4406 Mutex::Autolock _l(mLock); 4407 4408 AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, 4409 (int *)&format, 4410 &channels, 4411 &samplingRate, 4412 &status, 4413 (AudioSystem::audio_in_acoustics)acoustics); 4414 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4415 input, 4416 samplingRate, 4417 format, 4418 channels, 4419 acoustics, 4420 status); 4421 4422 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4423 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4424 // or stereo to mono conversions on 16 bit PCM inputs. 4425 if (input == 0 && status == BAD_VALUE && 4426 reqFormat == format && format == AudioSystem::PCM_16_BIT && 4427 (samplingRate <= 2 * reqSamplingRate) && 4428 (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { 4429 LOGV("openInput() reopening with proposed sampling rate and channels"); 4430 input = mAudioHardware->openInputStream(*pDevices, 4431 (int *)&format, 4432 &channels, 4433 &samplingRate, 4434 &status, 4435 (AudioSystem::audio_in_acoustics)acoustics); 4436 } 4437 4438 if (input != 0) { 4439 int id = nextUniqueId(); 4440 // Start record thread 4441 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4442 mRecordThreads.add(id, thread); 4443 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4444 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4445 if (pFormat) *pFormat = format; 4446 if (pChannels) *pChannels = reqChannels; 4447 4448 input->standby(); 4449 4450 // notify client processes of the new input creation 4451 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4452 return id; 4453 } 4454 4455 return 0; 4456} 4457 4458status_t AudioFlinger::closeInput(int input) 4459{ 4460 // keep strong reference on the record thread so that 4461 // it is not destroyed while exit() is executed 4462 sp <RecordThread> thread; 4463 { 4464 Mutex::Autolock _l(mLock); 4465 thread = checkRecordThread_l(input); 4466 if (thread == NULL) { 4467 return BAD_VALUE; 4468 } 4469 4470 LOGV("closeInput() %d", input); 4471 void *param2 = 0; 4472 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4473 mRecordThreads.removeItem(input); 4474 } 4475 thread->exit(); 4476 4477 mAudioHardware->closeInputStream(thread->getInput()); 4478 4479 return NO_ERROR; 4480} 4481 4482status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4483{ 4484 Mutex::Autolock _l(mLock); 4485 MixerThread *dstThread = checkMixerThread_l(output); 4486 if (dstThread == NULL) { 4487 LOGW("setStreamOutput() bad output id %d", output); 4488 return BAD_VALUE; 4489 } 4490 4491 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4492 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4493 4494 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4495 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4496 if (thread != dstThread && 4497 thread->type() != PlaybackThread::DIRECT) { 4498 MixerThread *srcThread = (MixerThread *)thread; 4499 srcThread->invalidateTracks(stream); 4500 } 4501 } 4502 4503 return NO_ERROR; 4504} 4505 4506 4507int AudioFlinger::newAudioSessionId() 4508{ 4509 return nextUniqueId(); 4510} 4511 4512// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4513AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4514{ 4515 PlaybackThread *thread = NULL; 4516 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4517 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4518 } 4519 return thread; 4520} 4521 4522// checkMixerThread_l() must be called with AudioFlinger::mLock held 4523AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4524{ 4525 PlaybackThread *thread = checkPlaybackThread_l(output); 4526 if (thread != NULL) { 4527 if (thread->type() == PlaybackThread::DIRECT) { 4528 thread = NULL; 4529 } 4530 } 4531 return (MixerThread *)thread; 4532} 4533 4534// checkRecordThread_l() must be called with AudioFlinger::mLock held 4535AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4536{ 4537 RecordThread *thread = NULL; 4538 if (mRecordThreads.indexOfKey(input) >= 0) { 4539 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4540 } 4541 return thread; 4542} 4543 4544int AudioFlinger::nextUniqueId() 4545{ 4546 return android_atomic_inc(&mNextUniqueId); 4547} 4548 4549// ---------------------------------------------------------------------------- 4550// Effect management 4551// ---------------------------------------------------------------------------- 4552 4553 4554status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) 4555{ 4556 // check calling permissions 4557 if (!settingsAllowed()) { 4558 return PERMISSION_DENIED; 4559 } 4560 // only allow libraries loaded from /system/lib/soundfx for now 4561 if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { 4562 return PERMISSION_DENIED; 4563 } 4564 4565 Mutex::Autolock _l(mLock); 4566 return EffectLoadLibrary(libPath, handle); 4567} 4568 4569status_t AudioFlinger::unloadEffectLibrary(int handle) 4570{ 4571 // check calling permissions 4572 if (!settingsAllowed()) { 4573 return PERMISSION_DENIED; 4574 } 4575 4576 Mutex::Autolock _l(mLock); 4577 return EffectUnloadLibrary(handle); 4578} 4579 4580status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4581{ 4582 Mutex::Autolock _l(mLock); 4583 return EffectQueryNumberEffects(numEffects); 4584} 4585 4586status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4587{ 4588 Mutex::Autolock _l(mLock); 4589 return EffectQueryEffect(index, descriptor); 4590} 4591 4592status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4593{ 4594 Mutex::Autolock _l(mLock); 4595 return EffectGetDescriptor(pUuid, descriptor); 4596} 4597 4598 4599// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4600static const effect_uuid_t VISUALIZATION_UUID_ = 4601 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4602 4603sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4604 effect_descriptor_t *pDesc, 4605 const sp<IEffectClient>& effectClient, 4606 int32_t priority, 4607 int output, 4608 int sessionId, 4609 status_t *status, 4610 int *id, 4611 int *enabled) 4612{ 4613 status_t lStatus = NO_ERROR; 4614 sp<EffectHandle> handle; 4615 effect_interface_t itfe; 4616 effect_descriptor_t desc; 4617 sp<Client> client; 4618 wp<Client> wclient; 4619 4620 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4621 pid, effectClient.get(), priority, sessionId, output); 4622 4623 if (pDesc == NULL) { 4624 lStatus = BAD_VALUE; 4625 goto Exit; 4626 } 4627 4628 { 4629 Mutex::Autolock _l(mLock); 4630 4631 // check recording permission for visualizer 4632 if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4633 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) { 4634 if (!recordingAllowed()) { 4635 lStatus = PERMISSION_DENIED; 4636 goto Exit; 4637 } 4638 } 4639 4640 if (!EffectIsNullUuid(&pDesc->uuid)) { 4641 // if uuid is specified, request effect descriptor 4642 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4643 if (lStatus < 0) { 4644 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4645 goto Exit; 4646 } 4647 } else { 4648 // if uuid is not specified, look for an available implementation 4649 // of the required type in effect factory 4650 if (EffectIsNullUuid(&pDesc->type)) { 4651 LOGW("createEffect() no effect type"); 4652 lStatus = BAD_VALUE; 4653 goto Exit; 4654 } 4655 uint32_t numEffects = 0; 4656 effect_descriptor_t d; 4657 bool found = false; 4658 4659 lStatus = EffectQueryNumberEffects(&numEffects); 4660 if (lStatus < 0) { 4661 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4662 goto Exit; 4663 } 4664 for (uint32_t i = 0; i < numEffects; i++) { 4665 lStatus = EffectQueryEffect(i, &desc); 4666 if (lStatus < 0) { 4667 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4668 continue; 4669 } 4670 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4671 // If matching type found save effect descriptor. If the session is 4672 // 0 and the effect is not auxiliary, continue enumeration in case 4673 // an auxiliary version of this effect type is available 4674 found = true; 4675 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4676 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX || 4677 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4678 break; 4679 } 4680 } 4681 } 4682 if (!found) { 4683 lStatus = BAD_VALUE; 4684 LOGW("createEffect() effect not found"); 4685 goto Exit; 4686 } 4687 // For same effect type, chose auxiliary version over insert version if 4688 // connect to output mix (Compliance to OpenSL ES) 4689 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && 4690 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4691 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4692 } 4693 } 4694 4695 // Do not allow auxiliary effects on a session different from 0 (output mix) 4696 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX && 4697 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4698 lStatus = INVALID_OPERATION; 4699 goto Exit; 4700 } 4701 4702 // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects 4703 // that can only be created by audio policy manager (running in same process) 4704 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && 4705 getpid() != IPCThreadState::self()->getCallingPid()) { 4706 lStatus = INVALID_OPERATION; 4707 goto Exit; 4708 } 4709 4710 // return effect descriptor 4711 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4712 4713 // If output is not specified try to find a matching audio session ID in one of the 4714 // output threads. 4715 // TODO: allow attachment of effect to inputs 4716 if (output == 0) { 4717 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) { 4718 // output must be specified by AudioPolicyManager when using session 4719 // AudioSystem::SESSION_OUTPUT_STAGE 4720 lStatus = BAD_VALUE; 4721 goto Exit; 4722 } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 4723 output = AudioSystem::getOutputForEffect(&desc); 4724 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4725 } else { 4726 // look for the thread where the specified audio session is present 4727 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4728 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4729 output = mPlaybackThreads.keyAt(i); 4730 break; 4731 } 4732 } 4733 // If no output thread contains the requested session ID, default to 4734 // first output. The effect chain will be moved to the correct output 4735 // thread when a track with the same session ID is created 4736 if (output == 0 && mPlaybackThreads.size()) { 4737 output = mPlaybackThreads.keyAt(0); 4738 } 4739 } 4740 } 4741 PlaybackThread *thread = checkPlaybackThread_l(output); 4742 if (thread == NULL) { 4743 LOGE("createEffect() unknown output thread"); 4744 lStatus = BAD_VALUE; 4745 goto Exit; 4746 } 4747 4748 wclient = mClients.valueFor(pid); 4749 4750 if (wclient != NULL) { 4751 client = wclient.promote(); 4752 } else { 4753 client = new Client(this, pid); 4754 mClients.add(pid, client); 4755 } 4756 4757 // create effect on selected output trhead 4758 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4759 &desc, enabled, &lStatus); 4760 if (handle != 0 && id != NULL) { 4761 *id = handle->id(); 4762 } 4763 } 4764 4765Exit: 4766 if(status) { 4767 *status = lStatus; 4768 } 4769 return handle; 4770} 4771 4772status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4773{ 4774 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4775 session, srcOutput, dstOutput); 4776 Mutex::Autolock _l(mLock); 4777 if (srcOutput == dstOutput) { 4778 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4779 return NO_ERROR; 4780 } 4781 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4782 if (srcThread == NULL) { 4783 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4784 return BAD_VALUE; 4785 } 4786 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4787 if (dstThread == NULL) { 4788 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4789 return BAD_VALUE; 4790 } 4791 4792 Mutex::Autolock _dl(dstThread->mLock); 4793 Mutex::Autolock _sl(srcThread->mLock); 4794 moveEffectChain_l(session, srcThread, dstThread, false); 4795 4796 return NO_ERROR; 4797} 4798 4799// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4800status_t AudioFlinger::moveEffectChain_l(int session, 4801 AudioFlinger::PlaybackThread *srcThread, 4802 AudioFlinger::PlaybackThread *dstThread, 4803 bool reRegister) 4804{ 4805 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4806 session, srcThread, dstThread); 4807 4808 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4809 if (chain == 0) { 4810 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4811 session, srcThread); 4812 return INVALID_OPERATION; 4813 } 4814 4815 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 4816 // so that a new chain is created with correct parameters when first effect is added. This is 4817 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4818 // removed. 4819 srcThread->removeEffectChain_l(chain); 4820 4821 // transfer all effects one by one so that new effect chain is created on new thread with 4822 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4823 int dstOutput = dstThread->id(); 4824 sp<EffectChain> dstChain; 4825 uint32_t strategy; 4826 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4827 while (effect != 0) { 4828 srcThread->removeEffect_l(effect); 4829 dstThread->addEffect_l(effect); 4830 // if the move request is not received from audio policy manager, the effect must be 4831 // re-registered with the new strategy and output 4832 if (dstChain == 0) { 4833 dstChain = effect->chain().promote(); 4834 if (dstChain == 0) { 4835 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 4836 srcThread->addEffect_l(effect); 4837 return NO_INIT; 4838 } 4839 strategy = dstChain->strategy(); 4840 } 4841 if (reRegister) { 4842 AudioSystem::unregisterEffect(effect->id()); 4843 AudioSystem::registerEffect(&effect->desc(), 4844 dstOutput, 4845 strategy, 4846 session, 4847 effect->id()); 4848 } 4849 effect = chain->getEffectFromId_l(0); 4850 } 4851 4852 return NO_ERROR; 4853} 4854 4855// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4856sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4857 const sp<AudioFlinger::Client>& client, 4858 const sp<IEffectClient>& effectClient, 4859 int32_t priority, 4860 int sessionId, 4861 effect_descriptor_t *desc, 4862 int *enabled, 4863 status_t *status 4864 ) 4865{ 4866 sp<EffectModule> effect; 4867 sp<EffectHandle> handle; 4868 status_t lStatus; 4869 sp<Track> track; 4870 sp<EffectChain> chain; 4871 bool chainCreated = false; 4872 bool effectCreated = false; 4873 bool effectRegistered = false; 4874 4875 if (mOutput == 0) { 4876 LOGW("createEffect_l() Audio driver not initialized."); 4877 lStatus = NO_INIT; 4878 goto Exit; 4879 } 4880 4881 // Do not allow auxiliary effect on session other than 0 4882 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4883 sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 4884 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4885 desc->name, sessionId); 4886 lStatus = BAD_VALUE; 4887 goto Exit; 4888 } 4889 4890 // Do not allow effects with session ID 0 on direct output or duplicating threads 4891 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4892 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) { 4893 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4894 desc->name, sessionId); 4895 lStatus = BAD_VALUE; 4896 goto Exit; 4897 } 4898 4899 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4900 4901 { // scope for mLock 4902 Mutex::Autolock _l(mLock); 4903 4904 // check for existing effect chain with the requested audio session 4905 chain = getEffectChain_l(sessionId); 4906 if (chain == 0) { 4907 // create a new chain for this session 4908 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4909 chain = new EffectChain(this, sessionId); 4910 addEffectChain_l(chain); 4911 chain->setStrategy(getStrategyForSession_l(sessionId)); 4912 chainCreated = true; 4913 } else { 4914 effect = chain->getEffectFromDesc_l(desc); 4915 } 4916 4917 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4918 4919 if (effect == 0) { 4920 int id = mAudioFlinger->nextUniqueId(); 4921 // Check CPU and memory usage 4922 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4923 if (lStatus != NO_ERROR) { 4924 goto Exit; 4925 } 4926 effectRegistered = true; 4927 // create a new effect module if none present in the chain 4928 effect = new EffectModule(this, chain, desc, id, sessionId); 4929 lStatus = effect->status(); 4930 if (lStatus != NO_ERROR) { 4931 goto Exit; 4932 } 4933 lStatus = chain->addEffect_l(effect); 4934 if (lStatus != NO_ERROR) { 4935 goto Exit; 4936 } 4937 effectCreated = true; 4938 4939 effect->setDevice(mDevice); 4940 effect->setMode(mAudioFlinger->getMode()); 4941 } 4942 // create effect handle and connect it to effect module 4943 handle = new EffectHandle(effect, client, effectClient, priority); 4944 lStatus = effect->addHandle(handle); 4945 if (enabled) { 4946 *enabled = (int)effect->isEnabled(); 4947 } 4948 } 4949 4950Exit: 4951 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 4952 Mutex::Autolock _l(mLock); 4953 if (effectCreated) { 4954 chain->removeEffect_l(effect); 4955 } 4956 if (effectRegistered) { 4957 AudioSystem::unregisterEffect(effect->id()); 4958 } 4959 if (chainCreated) { 4960 removeEffectChain_l(chain); 4961 } 4962 handle.clear(); 4963 } 4964 4965 if(status) { 4966 *status = lStatus; 4967 } 4968 return handle; 4969} 4970 4971// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 4972// PlaybackThread::mLock held 4973status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 4974{ 4975 // check for existing effect chain with the requested audio session 4976 int sessionId = effect->sessionId(); 4977 sp<EffectChain> chain = getEffectChain_l(sessionId); 4978 bool chainCreated = false; 4979 4980 if (chain == 0) { 4981 // create a new chain for this session 4982 LOGV("addEffect_l() new effect chain for session %d", sessionId); 4983 chain = new EffectChain(this, sessionId); 4984 addEffectChain_l(chain); 4985 chain->setStrategy(getStrategyForSession_l(sessionId)); 4986 chainCreated = true; 4987 } 4988 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 4989 4990 if (chain->getEffectFromId_l(effect->id()) != 0) { 4991 LOGW("addEffect_l() %p effect %s already present in chain %p", 4992 this, effect->desc().name, chain.get()); 4993 return BAD_VALUE; 4994 } 4995 4996 status_t status = chain->addEffect_l(effect); 4997 if (status != NO_ERROR) { 4998 if (chainCreated) { 4999 removeEffectChain_l(chain); 5000 } 5001 return status; 5002 } 5003 5004 effect->setDevice(mDevice); 5005 effect->setMode(mAudioFlinger->getMode()); 5006 return NO_ERROR; 5007} 5008 5009void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 5010 5011 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5012 effect_descriptor_t desc = effect->desc(); 5013 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5014 detachAuxEffect_l(effect->id()); 5015 } 5016 5017 sp<EffectChain> chain = effect->chain().promote(); 5018 if (chain != 0) { 5019 // remove effect chain if removing last effect 5020 if (chain->removeEffect_l(effect) == 0) { 5021 removeEffectChain_l(chain); 5022 } 5023 } else { 5024 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5025 } 5026} 5027 5028void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 5029 const wp<EffectHandle>& handle) { 5030 Mutex::Autolock _l(mLock); 5031 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5032 // delete the effect module if removing last handle on it 5033 if (effect->removeHandle(handle) == 0) { 5034 removeEffect_l(effect); 5035 AudioSystem::unregisterEffect(effect->id()); 5036 } 5037} 5038 5039status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5040{ 5041 int session = chain->sessionId(); 5042 int16_t *buffer = mMixBuffer; 5043 bool ownsBuffer = false; 5044 5045 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5046 if (session > 0) { 5047 // Only one effect chain can be present in direct output thread and it uses 5048 // the mix buffer as input 5049 if (mType != DIRECT) { 5050 size_t numSamples = mFrameCount * mChannelCount; 5051 buffer = new int16_t[numSamples]; 5052 memset(buffer, 0, numSamples * sizeof(int16_t)); 5053 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5054 ownsBuffer = true; 5055 } 5056 5057 // Attach all tracks with same session ID to this chain. 5058 for (size_t i = 0; i < mTracks.size(); ++i) { 5059 sp<Track> track = mTracks[i]; 5060 if (session == track->sessionId()) { 5061 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5062 track->setMainBuffer(buffer); 5063 } 5064 } 5065 5066 // indicate all active tracks in the chain 5067 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5068 sp<Track> track = mActiveTracks[i].promote(); 5069 if (track == 0) continue; 5070 if (session == track->sessionId()) { 5071 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5072 chain->startTrack(); 5073 } 5074 } 5075 } 5076 5077 chain->setInBuffer(buffer, ownsBuffer); 5078 chain->setOutBuffer(mMixBuffer); 5079 // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect 5080 // chains list in order to be processed last as it contains output stage effects 5081 // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before 5082 // session AudioSystem::SESSION_OUTPUT_STAGE to be processed 5083 // after track specific effects and before output stage 5084 // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and 5085 // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX 5086 // Effect chain for other sessions are inserted at beginning of effect 5087 // chains list to be processed before output mix effects. Relative order between other 5088 // sessions is not important 5089 size_t size = mEffectChains.size(); 5090 size_t i = 0; 5091 for (i = 0; i < size; i++) { 5092 if (mEffectChains[i]->sessionId() < session) break; 5093 } 5094 mEffectChains.insertAt(chain, i); 5095 5096 return NO_ERROR; 5097} 5098 5099size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5100{ 5101 int session = chain->sessionId(); 5102 5103 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5104 5105 for (size_t i = 0; i < mEffectChains.size(); i++) { 5106 if (chain == mEffectChains[i]) { 5107 mEffectChains.removeAt(i); 5108 // detach all tracks with same session ID from this chain 5109 for (size_t i = 0; i < mTracks.size(); ++i) { 5110 sp<Track> track = mTracks[i]; 5111 if (session == track->sessionId()) { 5112 track->setMainBuffer(mMixBuffer); 5113 } 5114 } 5115 break; 5116 } 5117 } 5118 return mEffectChains.size(); 5119} 5120 5121void AudioFlinger::PlaybackThread::lockEffectChains_l( 5122 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5123{ 5124 effectChains = mEffectChains; 5125 for (size_t i = 0; i < mEffectChains.size(); i++) { 5126 mEffectChains[i]->lock(); 5127 } 5128} 5129 5130void AudioFlinger::PlaybackThread::unlockEffectChains( 5131 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5132{ 5133 for (size_t i = 0; i < effectChains.size(); i++) { 5134 effectChains[i]->unlock(); 5135 } 5136} 5137 5138 5139sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5140{ 5141 sp<EffectModule> effect; 5142 5143 sp<EffectChain> chain = getEffectChain_l(sessionId); 5144 if (chain != 0) { 5145 effect = chain->getEffectFromId_l(effectId); 5146 } 5147 return effect; 5148} 5149 5150status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5151 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5152{ 5153 Mutex::Autolock _l(mLock); 5154 return attachAuxEffect_l(track, EffectId); 5155} 5156 5157status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5158 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5159{ 5160 status_t status = NO_ERROR; 5161 5162 if (EffectId == 0) { 5163 track->setAuxBuffer(0, NULL); 5164 } else { 5165 // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX 5166 sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId); 5167 if (effect != 0) { 5168 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5169 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5170 } else { 5171 status = INVALID_OPERATION; 5172 } 5173 } else { 5174 status = BAD_VALUE; 5175 } 5176 } 5177 return status; 5178} 5179 5180void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5181{ 5182 for (size_t i = 0; i < mTracks.size(); ++i) { 5183 sp<Track> track = mTracks[i]; 5184 if (track->auxEffectId() == effectId) { 5185 attachAuxEffect_l(track, 0); 5186 } 5187 } 5188} 5189 5190// ---------------------------------------------------------------------------- 5191// EffectModule implementation 5192// ---------------------------------------------------------------------------- 5193 5194#undef LOG_TAG 5195#define LOG_TAG "AudioFlinger::EffectModule" 5196 5197AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5198 const wp<AudioFlinger::EffectChain>& chain, 5199 effect_descriptor_t *desc, 5200 int id, 5201 int sessionId) 5202 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5203 mStatus(NO_INIT), mState(IDLE) 5204{ 5205 LOGV("Constructor %p", this); 5206 int lStatus; 5207 sp<ThreadBase> thread = mThread.promote(); 5208 if (thread == 0) { 5209 return; 5210 } 5211 PlaybackThread *p = (PlaybackThread *)thread.get(); 5212 5213 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5214 5215 // create effect engine from effect factory 5216 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5217 5218 if (mStatus != NO_ERROR) { 5219 return; 5220 } 5221 lStatus = init(); 5222 if (lStatus < 0) { 5223 mStatus = lStatus; 5224 goto Error; 5225 } 5226 5227 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5228 return; 5229Error: 5230 EffectRelease(mEffectInterface); 5231 mEffectInterface = NULL; 5232 LOGV("Constructor Error %d", mStatus); 5233} 5234 5235AudioFlinger::EffectModule::~EffectModule() 5236{ 5237 LOGV("Destructor %p", this); 5238 if (mEffectInterface != NULL) { 5239 // release effect engine 5240 EffectRelease(mEffectInterface); 5241 } 5242} 5243 5244status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5245{ 5246 status_t status; 5247 5248 Mutex::Autolock _l(mLock); 5249 // First handle in mHandles has highest priority and controls the effect module 5250 int priority = handle->priority(); 5251 size_t size = mHandles.size(); 5252 sp<EffectHandle> h; 5253 size_t i; 5254 for (i = 0; i < size; i++) { 5255 h = mHandles[i].promote(); 5256 if (h == 0) continue; 5257 if (h->priority() <= priority) break; 5258 } 5259 // if inserted in first place, move effect control from previous owner to this handle 5260 if (i == 0) { 5261 if (h != 0) { 5262 h->setControl(false, true); 5263 } 5264 handle->setControl(true, false); 5265 status = NO_ERROR; 5266 } else { 5267 status = ALREADY_EXISTS; 5268 } 5269 mHandles.insertAt(handle, i); 5270 return status; 5271} 5272 5273size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5274{ 5275 Mutex::Autolock _l(mLock); 5276 size_t size = mHandles.size(); 5277 size_t i; 5278 for (i = 0; i < size; i++) { 5279 if (mHandles[i] == handle) break; 5280 } 5281 if (i == size) { 5282 return size; 5283 } 5284 mHandles.removeAt(i); 5285 size = mHandles.size(); 5286 // if removed from first place, move effect control from this handle to next in line 5287 if (i == 0 && size != 0) { 5288 sp<EffectHandle> h = mHandles[0].promote(); 5289 if (h != 0) { 5290 h->setControl(true, true); 5291 } 5292 } 5293 5294 return size; 5295} 5296 5297void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5298{ 5299 // keep a strong reference on this EffectModule to avoid calling the 5300 // destructor before we exit 5301 sp<EffectModule> keep(this); 5302 { 5303 sp<ThreadBase> thread = mThread.promote(); 5304 if (thread != 0) { 5305 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5306 playbackThread->disconnectEffect(keep, handle); 5307 } 5308 } 5309} 5310 5311void AudioFlinger::EffectModule::updateState() { 5312 Mutex::Autolock _l(mLock); 5313 5314 switch (mState) { 5315 case RESTART: 5316 reset_l(); 5317 // FALL THROUGH 5318 5319 case STARTING: 5320 // clear auxiliary effect input buffer for next accumulation 5321 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5322 memset(mConfig.inputCfg.buffer.raw, 5323 0, 5324 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5325 } 5326 start_l(); 5327 mState = ACTIVE; 5328 break; 5329 case STOPPING: 5330 stop_l(); 5331 mDisableWaitCnt = mMaxDisableWaitCnt; 5332 mState = STOPPED; 5333 break; 5334 case STOPPED: 5335 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5336 // turn off sequence. 5337 if (--mDisableWaitCnt == 0) { 5338 reset_l(); 5339 mState = IDLE; 5340 } 5341 break; 5342 default: //IDLE , ACTIVE 5343 break; 5344 } 5345} 5346 5347void AudioFlinger::EffectModule::process() 5348{ 5349 Mutex::Autolock _l(mLock); 5350 5351 if (mEffectInterface == NULL || 5352 mConfig.inputCfg.buffer.raw == NULL || 5353 mConfig.outputCfg.buffer.raw == NULL) { 5354 return; 5355 } 5356 5357 if (mState == ACTIVE || mState == STOPPING || mState == STOPPED || mState == RESTART) { 5358 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5359 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5360 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5361 mConfig.inputCfg.buffer.s32, 5362 mConfig.inputCfg.buffer.frameCount/2); 5363 } 5364 5365 // do the actual processing in the effect engine 5366 int ret = (*mEffectInterface)->process(mEffectInterface, 5367 &mConfig.inputCfg.buffer, 5368 &mConfig.outputCfg.buffer); 5369 5370 // force transition to IDLE state when engine is ready 5371 if (mState == STOPPED && ret == -ENODATA) { 5372 mDisableWaitCnt = 1; 5373 } 5374 5375 // clear auxiliary effect input buffer for next accumulation 5376 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5377 memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5378 } 5379 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5380 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ 5381 // If an insert effect is idle and input buffer is different from output buffer, copy input to 5382 // output 5383 sp<EffectChain> chain = mChain.promote(); 5384 if (chain != 0 && chain->activeTracks() != 0) { 5385 size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); 5386 if (mConfig.inputCfg.channels == CHANNEL_STEREO) { 5387 size *= 2; 5388 } 5389 memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); 5390 } 5391 } 5392} 5393 5394void AudioFlinger::EffectModule::reset_l() 5395{ 5396 if (mEffectInterface == NULL) { 5397 return; 5398 } 5399 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5400} 5401 5402status_t AudioFlinger::EffectModule::configure() 5403{ 5404 uint32_t channels; 5405 if (mEffectInterface == NULL) { 5406 return NO_INIT; 5407 } 5408 5409 sp<ThreadBase> thread = mThread.promote(); 5410 if (thread == 0) { 5411 return DEAD_OBJECT; 5412 } 5413 5414 // TODO: handle configuration of effects replacing track process 5415 if (thread->channelCount() == 1) { 5416 channels = CHANNEL_MONO; 5417 } else { 5418 channels = CHANNEL_STEREO; 5419 } 5420 5421 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5422 mConfig.inputCfg.channels = CHANNEL_MONO; 5423 } else { 5424 mConfig.inputCfg.channels = channels; 5425 } 5426 mConfig.outputCfg.channels = channels; 5427 mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; 5428 mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; 5429 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5430 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5431 mConfig.inputCfg.bufferProvider.cookie = NULL; 5432 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5433 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5434 mConfig.outputCfg.bufferProvider.cookie = NULL; 5435 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5436 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5437 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5438 // Insert effect: 5439 // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE, 5440 // always overwrites output buffer: input buffer == output buffer 5441 // - in other sessions: 5442 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5443 // other effect: overwrites output buffer: input buffer == output buffer 5444 // Auxiliary effect: 5445 // accumulates in output buffer: input buffer != output buffer 5446 // Therefore: accumulate <=> input buffer != output buffer 5447 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5448 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5449 } else { 5450 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5451 } 5452 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5453 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5454 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5455 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5456 5457 LOGV("configure() %p thread %p buffer %p framecount %d", 5458 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5459 5460 status_t cmdStatus; 5461 uint32_t size = sizeof(int); 5462 status_t status = (*mEffectInterface)->command(mEffectInterface, 5463 EFFECT_CMD_CONFIGURE, 5464 sizeof(effect_config_t), 5465 &mConfig, 5466 &size, 5467 &cmdStatus); 5468 if (status == 0) { 5469 status = cmdStatus; 5470 } 5471 5472 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5473 (1000 * mConfig.outputCfg.buffer.frameCount); 5474 5475 return status; 5476} 5477 5478status_t AudioFlinger::EffectModule::init() 5479{ 5480 Mutex::Autolock _l(mLock); 5481 if (mEffectInterface == NULL) { 5482 return NO_INIT; 5483 } 5484 status_t cmdStatus; 5485 uint32_t size = sizeof(status_t); 5486 status_t status = (*mEffectInterface)->command(mEffectInterface, 5487 EFFECT_CMD_INIT, 5488 0, 5489 NULL, 5490 &size, 5491 &cmdStatus); 5492 if (status == 0) { 5493 status = cmdStatus; 5494 } 5495 return status; 5496} 5497 5498status_t AudioFlinger::EffectModule::start_l() 5499{ 5500 if (mEffectInterface == NULL) { 5501 return NO_INIT; 5502 } 5503 status_t cmdStatus; 5504 uint32_t size = sizeof(status_t); 5505 status_t status = (*mEffectInterface)->command(mEffectInterface, 5506 EFFECT_CMD_ENABLE, 5507 0, 5508 NULL, 5509 &size, 5510 &cmdStatus); 5511 if (status == 0) { 5512 status = cmdStatus; 5513 } 5514 return status; 5515} 5516 5517status_t AudioFlinger::EffectModule::stop_l() 5518{ 5519 if (mEffectInterface == NULL) { 5520 return NO_INIT; 5521 } 5522 status_t cmdStatus; 5523 uint32_t size = sizeof(status_t); 5524 status_t status = (*mEffectInterface)->command(mEffectInterface, 5525 EFFECT_CMD_DISABLE, 5526 0, 5527 NULL, 5528 &size, 5529 &cmdStatus); 5530 if (status == 0) { 5531 status = cmdStatus; 5532 } 5533 return status; 5534} 5535 5536status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5537 uint32_t cmdSize, 5538 void *pCmdData, 5539 uint32_t *replySize, 5540 void *pReplyData) 5541{ 5542 Mutex::Autolock _l(mLock); 5543// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5544 5545 if (mEffectInterface == NULL) { 5546 return NO_INIT; 5547 } 5548 status_t status = (*mEffectInterface)->command(mEffectInterface, 5549 cmdCode, 5550 cmdSize, 5551 pCmdData, 5552 replySize, 5553 pReplyData); 5554 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5555 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5556 for (size_t i = 1; i < mHandles.size(); i++) { 5557 sp<EffectHandle> h = mHandles[i].promote(); 5558 if (h != 0) { 5559 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5560 } 5561 } 5562 } 5563 return status; 5564} 5565 5566status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5567{ 5568 Mutex::Autolock _l(mLock); 5569 LOGV("setEnabled %p enabled %d", this, enabled); 5570 5571 if (enabled != isEnabled()) { 5572 switch (mState) { 5573 // going from disabled to enabled 5574 case IDLE: 5575 mState = STARTING; 5576 break; 5577 case STOPPED: 5578 mState = RESTART; 5579 break; 5580 case STOPPING: 5581 mState = ACTIVE; 5582 break; 5583 5584 // going from enabled to disabled 5585 case RESTART: 5586 case STARTING: 5587 mState = IDLE; 5588 break; 5589 case ACTIVE: 5590 mState = STOPPING; 5591 break; 5592 } 5593 for (size_t i = 1; i < mHandles.size(); i++) { 5594 sp<EffectHandle> h = mHandles[i].promote(); 5595 if (h != 0) { 5596 h->setEnabled(enabled); 5597 } 5598 } 5599 } 5600 return NO_ERROR; 5601} 5602 5603bool AudioFlinger::EffectModule::isEnabled() 5604{ 5605 switch (mState) { 5606 case RESTART: 5607 case STARTING: 5608 case ACTIVE: 5609 return true; 5610 case IDLE: 5611 case STOPPING: 5612 case STOPPED: 5613 default: 5614 return false; 5615 } 5616} 5617 5618status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5619{ 5620 Mutex::Autolock _l(mLock); 5621 status_t status = NO_ERROR; 5622 5623 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5624 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5625 if ((mState >= ACTIVE) && 5626 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5627 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5628 status_t cmdStatus; 5629 uint32_t volume[2]; 5630 uint32_t *pVolume = NULL; 5631 uint32_t size = sizeof(volume); 5632 volume[0] = *left; 5633 volume[1] = *right; 5634 if (controller) { 5635 pVolume = volume; 5636 } 5637 status = (*mEffectInterface)->command(mEffectInterface, 5638 EFFECT_CMD_SET_VOLUME, 5639 size, 5640 volume, 5641 &size, 5642 pVolume); 5643 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5644 *left = volume[0]; 5645 *right = volume[1]; 5646 } 5647 } 5648 return status; 5649} 5650 5651status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5652{ 5653 Mutex::Autolock _l(mLock); 5654 status_t status = NO_ERROR; 5655 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5656 // convert device bit field from AudioSystem to EffectApi format. 5657 device = deviceAudioSystemToEffectApi(device); 5658 if (device == 0) { 5659 return BAD_VALUE; 5660 } 5661 status_t cmdStatus; 5662 uint32_t size = sizeof(status_t); 5663 status = (*mEffectInterface)->command(mEffectInterface, 5664 EFFECT_CMD_SET_DEVICE, 5665 sizeof(uint32_t), 5666 &device, 5667 &size, 5668 &cmdStatus); 5669 if (status == NO_ERROR) { 5670 status = cmdStatus; 5671 } 5672 } 5673 return status; 5674} 5675 5676status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5677{ 5678 Mutex::Autolock _l(mLock); 5679 status_t status = NO_ERROR; 5680 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5681 // convert audio mode from AudioSystem to EffectApi format. 5682 int effectMode = modeAudioSystemToEffectApi(mode); 5683 if (effectMode < 0) { 5684 return BAD_VALUE; 5685 } 5686 status_t cmdStatus; 5687 uint32_t size = sizeof(status_t); 5688 status = (*mEffectInterface)->command(mEffectInterface, 5689 EFFECT_CMD_SET_AUDIO_MODE, 5690 sizeof(int), 5691 &effectMode, 5692 &size, 5693 &cmdStatus); 5694 if (status == NO_ERROR) { 5695 status = cmdStatus; 5696 } 5697 } 5698 return status; 5699} 5700 5701// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified 5702const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { 5703 DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE 5704 DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER 5705 DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET 5706 DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE 5707 DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO 5708 DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET 5709 DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT 5710 DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP 5711 DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES 5712 DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER 5713 DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL 5714}; 5715 5716uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) 5717{ 5718 uint32_t deviceOut = 0; 5719 while (device) { 5720 const uint32_t i = 31 - __builtin_clz(device); 5721 device &= ~(1 << i); 5722 if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { 5723 LOGE("device convertion error for AudioSystem device 0x%08x", device); 5724 return 0; 5725 } 5726 deviceOut |= (uint32_t)sDeviceConvTable[i]; 5727 } 5728 return deviceOut; 5729} 5730 5731// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified 5732const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { 5733 AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL 5734 AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE 5735 AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL 5736}; 5737 5738int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) 5739{ 5740 int modeOut = -1; 5741 if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { 5742 modeOut = (int)sModeConvTable[mode]; 5743 } 5744 return modeOut; 5745} 5746 5747status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5748{ 5749 const size_t SIZE = 256; 5750 char buffer[SIZE]; 5751 String8 result; 5752 5753 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5754 result.append(buffer); 5755 5756 bool locked = tryLock(mLock); 5757 // failed to lock - AudioFlinger is probably deadlocked 5758 if (!locked) { 5759 result.append("\t\tCould not lock Fx mutex:\n"); 5760 } 5761 5762 result.append("\t\tSession Status State Engine:\n"); 5763 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5764 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5765 result.append(buffer); 5766 5767 result.append("\t\tDescriptor:\n"); 5768 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5769 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5770 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5771 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5772 result.append(buffer); 5773 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5774 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5775 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5776 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5777 result.append(buffer); 5778 snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", 5779 mDescriptor.apiVersion, 5780 mDescriptor.flags); 5781 result.append(buffer); 5782 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5783 mDescriptor.name); 5784 result.append(buffer); 5785 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5786 mDescriptor.implementor); 5787 result.append(buffer); 5788 5789 result.append("\t\t- Input configuration:\n"); 5790 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5791 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5792 (uint32_t)mConfig.inputCfg.buffer.raw, 5793 mConfig.inputCfg.buffer.frameCount, 5794 mConfig.inputCfg.samplingRate, 5795 mConfig.inputCfg.channels, 5796 mConfig.inputCfg.format); 5797 result.append(buffer); 5798 5799 result.append("\t\t- Output configuration:\n"); 5800 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5801 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5802 (uint32_t)mConfig.outputCfg.buffer.raw, 5803 mConfig.outputCfg.buffer.frameCount, 5804 mConfig.outputCfg.samplingRate, 5805 mConfig.outputCfg.channels, 5806 mConfig.outputCfg.format); 5807 result.append(buffer); 5808 5809 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5810 result.append(buffer); 5811 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5812 for (size_t i = 0; i < mHandles.size(); ++i) { 5813 sp<EffectHandle> handle = mHandles[i].promote(); 5814 if (handle != 0) { 5815 handle->dump(buffer, SIZE); 5816 result.append(buffer); 5817 } 5818 } 5819 5820 result.append("\n"); 5821 5822 write(fd, result.string(), result.length()); 5823 5824 if (locked) { 5825 mLock.unlock(); 5826 } 5827 5828 return NO_ERROR; 5829} 5830 5831// ---------------------------------------------------------------------------- 5832// EffectHandle implementation 5833// ---------------------------------------------------------------------------- 5834 5835#undef LOG_TAG 5836#define LOG_TAG "AudioFlinger::EffectHandle" 5837 5838AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5839 const sp<AudioFlinger::Client>& client, 5840 const sp<IEffectClient>& effectClient, 5841 int32_t priority) 5842 : BnEffect(), 5843 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5844{ 5845 LOGV("constructor %p", this); 5846 5847 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5848 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5849 if (mCblkMemory != 0) { 5850 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5851 5852 if (mCblk) { 5853 new(mCblk) effect_param_cblk_t(); 5854 mBuffer = (uint8_t *)mCblk + bufOffset; 5855 } 5856 } else { 5857 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5858 return; 5859 } 5860} 5861 5862AudioFlinger::EffectHandle::~EffectHandle() 5863{ 5864 LOGV("Destructor %p", this); 5865 disconnect(); 5866} 5867 5868status_t AudioFlinger::EffectHandle::enable() 5869{ 5870 if (!mHasControl) return INVALID_OPERATION; 5871 if (mEffect == 0) return DEAD_OBJECT; 5872 5873 return mEffect->setEnabled(true); 5874} 5875 5876status_t AudioFlinger::EffectHandle::disable() 5877{ 5878 if (!mHasControl) return INVALID_OPERATION; 5879 if (mEffect == NULL) return DEAD_OBJECT; 5880 5881 return mEffect->setEnabled(false); 5882} 5883 5884void AudioFlinger::EffectHandle::disconnect() 5885{ 5886 if (mEffect == 0) { 5887 return; 5888 } 5889 mEffect->disconnect(this); 5890 // release sp on module => module destructor can be called now 5891 mEffect.clear(); 5892 if (mCblk) { 5893 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5894 } 5895 mCblkMemory.clear(); // and free the shared memory 5896 if (mClient != 0) { 5897 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5898 mClient.clear(); 5899 } 5900} 5901 5902status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5903 uint32_t cmdSize, 5904 void *pCmdData, 5905 uint32_t *replySize, 5906 void *pReplyData) 5907{ 5908// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5909// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5910 5911 // only get parameter command is permitted for applications not controlling the effect 5912 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5913 return INVALID_OPERATION; 5914 } 5915 if (mEffect == 0) return DEAD_OBJECT; 5916 5917 // handle commands that are not forwarded transparently to effect engine 5918 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5919 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5920 // no risk to block the whole media server process or mixer threads is we are stuck here 5921 Mutex::Autolock _l(mCblk->lock); 5922 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5923 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 5924 mCblk->serverIndex = 0; 5925 mCblk->clientIndex = 0; 5926 return BAD_VALUE; 5927 } 5928 status_t status = NO_ERROR; 5929 while (mCblk->serverIndex < mCblk->clientIndex) { 5930 int reply; 5931 uint32_t rsize = sizeof(int); 5932 int *p = (int *)(mBuffer + mCblk->serverIndex); 5933 int size = *p++; 5934 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 5935 LOGW("command(): invalid parameter block size"); 5936 break; 5937 } 5938 effect_param_t *param = (effect_param_t *)p; 5939 if (param->psize == 0 || param->vsize == 0) { 5940 LOGW("command(): null parameter or value size"); 5941 mCblk->serverIndex += size; 5942 continue; 5943 } 5944 uint32_t psize = sizeof(effect_param_t) + 5945 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 5946 param->vsize; 5947 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 5948 psize, 5949 p, 5950 &rsize, 5951 &reply); 5952 if (ret == NO_ERROR) { 5953 if (reply != NO_ERROR) { 5954 status = reply; 5955 } 5956 } else { 5957 status = ret; 5958 } 5959 mCblk->serverIndex += size; 5960 } 5961 mCblk->serverIndex = 0; 5962 mCblk->clientIndex = 0; 5963 return status; 5964 } else if (cmdCode == EFFECT_CMD_ENABLE) { 5965 return enable(); 5966 } else if (cmdCode == EFFECT_CMD_DISABLE) { 5967 return disable(); 5968 } 5969 5970 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 5971} 5972 5973sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 5974 return mCblkMemory; 5975} 5976 5977void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 5978{ 5979 LOGV("setControl %p control %d", this, hasControl); 5980 5981 mHasControl = hasControl; 5982 if (signal && mEffectClient != 0) { 5983 mEffectClient->controlStatusChanged(hasControl); 5984 } 5985} 5986 5987void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 5988 uint32_t cmdSize, 5989 void *pCmdData, 5990 uint32_t replySize, 5991 void *pReplyData) 5992{ 5993 if (mEffectClient != 0) { 5994 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 5995 } 5996} 5997 5998 5999 6000void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6001{ 6002 if (mEffectClient != 0) { 6003 mEffectClient->enableStatusChanged(enabled); 6004 } 6005} 6006 6007status_t AudioFlinger::EffectHandle::onTransact( 6008 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6009{ 6010 return BnEffect::onTransact(code, data, reply, flags); 6011} 6012 6013 6014void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6015{ 6016 bool locked = tryLock(mCblk->lock); 6017 6018 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6019 (mClient == NULL) ? getpid() : mClient->pid(), 6020 mPriority, 6021 mHasControl, 6022 !locked, 6023 mCblk->clientIndex, 6024 mCblk->serverIndex 6025 ); 6026 6027 if (locked) { 6028 mCblk->lock.unlock(); 6029 } 6030} 6031 6032#undef LOG_TAG 6033#define LOG_TAG "AudioFlinger::EffectChain" 6034 6035AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6036 int sessionId) 6037 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), 6038 mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 6039 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 6040{ 6041 mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 6042} 6043 6044AudioFlinger::EffectChain::~EffectChain() 6045{ 6046 if (mOwnInBuffer) { 6047 delete mInBuffer; 6048 } 6049 6050} 6051 6052// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6053sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6054{ 6055 sp<EffectModule> effect; 6056 size_t size = mEffects.size(); 6057 6058 for (size_t i = 0; i < size; i++) { 6059 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6060 effect = mEffects[i]; 6061 break; 6062 } 6063 } 6064 return effect; 6065} 6066 6067// getEffectFromId_l() must be called with PlaybackThread::mLock held 6068sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6069{ 6070 sp<EffectModule> effect; 6071 size_t size = mEffects.size(); 6072 6073 for (size_t i = 0; i < size; i++) { 6074 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6075 if (id == 0 || mEffects[i]->id() == id) { 6076 effect = mEffects[i]; 6077 break; 6078 } 6079 } 6080 return effect; 6081} 6082 6083// Must be called with EffectChain::mLock locked 6084void AudioFlinger::EffectChain::process_l() 6085{ 6086 size_t size = mEffects.size(); 6087 for (size_t i = 0; i < size; i++) { 6088 mEffects[i]->process(); 6089 } 6090 for (size_t i = 0; i < size; i++) { 6091 mEffects[i]->updateState(); 6092 } 6093 // if no track is active, input buffer must be cleared here as the mixer process 6094 // will not do it 6095 if (mSessionId > 0 && activeTracks() == 0) { 6096 sp<ThreadBase> thread = mThread.promote(); 6097 if (thread != 0) { 6098 size_t numSamples = thread->frameCount() * thread->channelCount(); 6099 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6100 } 6101 } 6102} 6103 6104// addEffect_l() must be called with PlaybackThread::mLock held 6105status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6106{ 6107 effect_descriptor_t desc = effect->desc(); 6108 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6109 6110 Mutex::Autolock _l(mLock); 6111 effect->setChain(this); 6112 sp<ThreadBase> thread = mThread.promote(); 6113 if (thread == 0) { 6114 return NO_INIT; 6115 } 6116 effect->setThread(thread); 6117 6118 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6119 // Auxiliary effects are inserted at the beginning of mEffects vector as 6120 // they are processed first and accumulated in chain input buffer 6121 mEffects.insertAt(effect, 0); 6122 6123 // the input buffer for auxiliary effect contains mono samples in 6124 // 32 bit format. This is to avoid saturation in AudoMixer 6125 // accumulation stage. Saturation is done in EffectModule::process() before 6126 // calling the process in effect engine 6127 size_t numSamples = thread->frameCount(); 6128 int32_t *buffer = new int32_t[numSamples]; 6129 memset(buffer, 0, numSamples * sizeof(int32_t)); 6130 effect->setInBuffer((int16_t *)buffer); 6131 // auxiliary effects output samples to chain input buffer for further processing 6132 // by insert effects 6133 effect->setOutBuffer(mInBuffer); 6134 } else { 6135 // Insert effects are inserted at the end of mEffects vector as they are processed 6136 // after track and auxiliary effects. 6137 // Insert effect order as a function of indicated preference: 6138 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6139 // another effect is present 6140 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6141 // last effect claiming first position 6142 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6143 // first effect claiming last position 6144 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6145 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6146 // already present 6147 6148 int size = (int)mEffects.size(); 6149 int idx_insert = size; 6150 int idx_insert_first = -1; 6151 int idx_insert_last = -1; 6152 6153 for (int i = 0; i < size; i++) { 6154 effect_descriptor_t d = mEffects[i]->desc(); 6155 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6156 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6157 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6158 // check invalid effect chaining combinations 6159 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6160 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6161 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6162 return INVALID_OPERATION; 6163 } 6164 // remember position of first insert effect and by default 6165 // select this as insert position for new effect 6166 if (idx_insert == size) { 6167 idx_insert = i; 6168 } 6169 // remember position of last insert effect claiming 6170 // first position 6171 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6172 idx_insert_first = i; 6173 } 6174 // remember position of first insert effect claiming 6175 // last position 6176 if (iPref == EFFECT_FLAG_INSERT_LAST && 6177 idx_insert_last == -1) { 6178 idx_insert_last = i; 6179 } 6180 } 6181 } 6182 6183 // modify idx_insert from first position if needed 6184 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6185 if (idx_insert_last != -1) { 6186 idx_insert = idx_insert_last; 6187 } else { 6188 idx_insert = size; 6189 } 6190 } else { 6191 if (idx_insert_first != -1) { 6192 idx_insert = idx_insert_first + 1; 6193 } 6194 } 6195 6196 // always read samples from chain input buffer 6197 effect->setInBuffer(mInBuffer); 6198 6199 // if last effect in the chain, output samples to chain 6200 // output buffer, otherwise to chain input buffer 6201 if (idx_insert == size) { 6202 if (idx_insert != 0) { 6203 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6204 mEffects[idx_insert-1]->configure(); 6205 } 6206 effect->setOutBuffer(mOutBuffer); 6207 } else { 6208 effect->setOutBuffer(mInBuffer); 6209 } 6210 mEffects.insertAt(effect, idx_insert); 6211 6212 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6213 } 6214 effect->configure(); 6215 return NO_ERROR; 6216} 6217 6218// removeEffect_l() must be called with PlaybackThread::mLock held 6219size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6220{ 6221 Mutex::Autolock _l(mLock); 6222 int size = (int)mEffects.size(); 6223 int i; 6224 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6225 6226 for (i = 0; i < size; i++) { 6227 if (effect == mEffects[i]) { 6228 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6229 delete[] effect->inBuffer(); 6230 } else { 6231 if (i == size - 1 && i != 0) { 6232 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6233 mEffects[i - 1]->configure(); 6234 } 6235 } 6236 mEffects.removeAt(i); 6237 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6238 break; 6239 } 6240 } 6241 6242 return mEffects.size(); 6243} 6244 6245// setDevice_l() must be called with PlaybackThread::mLock held 6246void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6247{ 6248 size_t size = mEffects.size(); 6249 for (size_t i = 0; i < size; i++) { 6250 mEffects[i]->setDevice(device); 6251 } 6252} 6253 6254// setMode_l() must be called with PlaybackThread::mLock held 6255void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6256{ 6257 size_t size = mEffects.size(); 6258 for (size_t i = 0; i < size; i++) { 6259 mEffects[i]->setMode(mode); 6260 } 6261} 6262 6263// setVolume_l() must be called with PlaybackThread::mLock held 6264bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6265{ 6266 uint32_t newLeft = *left; 6267 uint32_t newRight = *right; 6268 bool hasControl = false; 6269 int ctrlIdx = -1; 6270 size_t size = mEffects.size(); 6271 6272 // first update volume controller 6273 for (size_t i = size; i > 0; i--) { 6274 if ((mEffects[i - 1]->state() >= EffectModule::ACTIVE) && 6275 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6276 ctrlIdx = i - 1; 6277 hasControl = true; 6278 break; 6279 } 6280 } 6281 6282 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6283 if (hasControl) { 6284 *left = mNewLeftVolume; 6285 *right = mNewRightVolume; 6286 } 6287 return hasControl; 6288 } 6289 6290 if (mVolumeCtrlIdx != -1) { 6291 hasControl = true; 6292 } 6293 mVolumeCtrlIdx = ctrlIdx; 6294 mLeftVolume = newLeft; 6295 mRightVolume = newRight; 6296 6297 // second get volume update from volume controller 6298 if (ctrlIdx >= 0) { 6299 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6300 mNewLeftVolume = newLeft; 6301 mNewRightVolume = newRight; 6302 } 6303 // then indicate volume to all other effects in chain. 6304 // Pass altered volume to effects before volume controller 6305 // and requested volume to effects after controller 6306 uint32_t lVol = newLeft; 6307 uint32_t rVol = newRight; 6308 6309 for (size_t i = 0; i < size; i++) { 6310 if ((int)i == ctrlIdx) continue; 6311 // this also works for ctrlIdx == -1 when there is no volume controller 6312 if ((int)i > ctrlIdx) { 6313 lVol = *left; 6314 rVol = *right; 6315 } 6316 mEffects[i]->setVolume(&lVol, &rVol, false); 6317 } 6318 *left = newLeft; 6319 *right = newRight; 6320 6321 return hasControl; 6322} 6323 6324status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6325{ 6326 const size_t SIZE = 256; 6327 char buffer[SIZE]; 6328 String8 result; 6329 6330 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6331 result.append(buffer); 6332 6333 bool locked = tryLock(mLock); 6334 // failed to lock - AudioFlinger is probably deadlocked 6335 if (!locked) { 6336 result.append("\tCould not lock mutex:\n"); 6337 } 6338 6339 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6340 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6341 mEffects.size(), 6342 (uint32_t)mInBuffer, 6343 (uint32_t)mOutBuffer, 6344 mActiveTrackCnt); 6345 result.append(buffer); 6346 write(fd, result.string(), result.size()); 6347 6348 for (size_t i = 0; i < mEffects.size(); ++i) { 6349 sp<EffectModule> effect = mEffects[i]; 6350 if (effect != 0) { 6351 effect->dump(fd, args); 6352 } 6353 } 6354 6355 if (locked) { 6356 mLock.unlock(); 6357 } 6358 6359 return NO_ERROR; 6360} 6361 6362#undef LOG_TAG 6363#define LOG_TAG "AudioFlinger" 6364 6365// ---------------------------------------------------------------------------- 6366 6367status_t AudioFlinger::onTransact( 6368 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6369{ 6370 return BnAudioFlinger::onTransact(code, data, reply, flags); 6371} 6372 6373}; // namespace android 6374