AudioFlinger.cpp revision af59ce2407fa4e6e5d8f1664a4df2daf1badd407
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 = suspendSleepTimeUs(); 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() && 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 uint32_t vl, vr, va; 1756 if (track->isMuted() || track->isPausing() || 1757 mStreamTypes[track->type()].mute) { 1758 vl = vr = va = 0; 1759 if (track->isPausing()) { 1760 track->setPaused(); 1761 } 1762 } else { 1763 1764 // read original volumes with volume control 1765 float typeVolume = mStreamTypes[track->type()].volume; 1766#ifdef LVMX 1767 bool streamMute=false; 1768 // read the volume from the LivesVibes audio engine. 1769 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1770 { 1771 LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); 1772 if (streamMute) { 1773 typeVolume = 0; 1774 } 1775 } 1776#endif 1777 float v = masterVolume * typeVolume; 1778 vl = (uint32_t)(v * cblk->volume[0]) << 12; 1779 vr = (uint32_t)(v * cblk->volume[1]) << 12; 1780 1781 va = (uint32_t)(v * cblk->sendLevel); 1782 } 1783 // Delegate volume control to effect in track effect chain if needed 1784 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1785 // Do not ramp volume if volume is controlled by effect 1786 param = AudioMixer::VOLUME; 1787 track->mHasVolumeController = true; 1788 } else { 1789 // force no volume ramp when volume controller was just disabled or removed 1790 // from effect chain to avoid volume spike 1791 if (track->mHasVolumeController) { 1792 param = AudioMixer::VOLUME; 1793 } 1794 track->mHasVolumeController = false; 1795 } 1796 1797 // Convert volumes from 8.24 to 4.12 format 1798 int16_t left, right, aux; 1799 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1800 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1801 left = int16_t(v_clamped); 1802 v_clamped = (vr + (1 << 11)) >> 12; 1803 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1804 right = int16_t(v_clamped); 1805 1806 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 1807 aux = int16_t(va); 1808 1809#ifdef LVMX 1810 if ( tracksConnectedChanged || stateChanged ) 1811 { 1812 // only do the ramp when the volume is changed by the user / application 1813 param = AudioMixer::VOLUME; 1814 } 1815#endif 1816 1817 // XXX: these things DON'T need to be done each time 1818 mAudioMixer->setBufferProvider(track); 1819 mAudioMixer->enable(AudioMixer::MIXING); 1820 1821 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1822 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1823 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1824 mAudioMixer->setParameter( 1825 AudioMixer::TRACK, 1826 AudioMixer::FORMAT, (void *)track->format()); 1827 mAudioMixer->setParameter( 1828 AudioMixer::TRACK, 1829 AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); 1830 mAudioMixer->setParameter( 1831 AudioMixer::RESAMPLE, 1832 AudioMixer::SAMPLE_RATE, 1833 (void *)(cblk->sampleRate)); 1834 mAudioMixer->setParameter( 1835 AudioMixer::TRACK, 1836 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1837 mAudioMixer->setParameter( 1838 AudioMixer::TRACK, 1839 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1840 1841 // reset retry count 1842 track->mRetryCount = kMaxTrackRetries; 1843 mixerStatus = MIXER_TRACKS_READY; 1844 } else { 1845 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1846 if (track->isStopped()) { 1847 track->reset(); 1848 } 1849 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1850 // We have consumed all the buffers of this track. 1851 // Remove it from the list of active tracks. 1852 tracksToRemove->add(track); 1853 } else { 1854 // No buffers for this track. Give it a few chances to 1855 // fill a buffer, then remove it from active list. 1856 if (--(track->mRetryCount) <= 0) { 1857 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1858 tracksToRemove->add(track); 1859 // indicate to client process that the track was disabled because of underrun 1860 cblk->flags |= CBLK_DISABLED_ON; 1861 } else if (mixerStatus != MIXER_TRACKS_READY) { 1862 mixerStatus = MIXER_TRACKS_ENABLED; 1863 } 1864 } 1865 mAudioMixer->disable(AudioMixer::MIXING); 1866 } 1867 } 1868 1869 // remove all the tracks that need to be... 1870 count = tracksToRemove->size(); 1871 if (UNLIKELY(count)) { 1872 for (size_t i=0 ; i<count ; i++) { 1873 const sp<Track>& track = tracksToRemove->itemAt(i); 1874 mActiveTracks.remove(track); 1875 if (track->mainBuffer() != mMixBuffer) { 1876 chain = getEffectChain_l(track->sessionId()); 1877 if (chain != 0) { 1878 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1879 chain->stopTrack(); 1880 } 1881 } 1882 if (track->isTerminated()) { 1883 mTracks.remove(track); 1884 deleteTrackName_l(track->mName); 1885 } 1886 } 1887 } 1888 1889 // mix buffer must be cleared if all tracks are connected to an 1890 // effect chain as in this case the mixer will not write to 1891 // mix buffer and track effects will accumulate into it 1892 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1893 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1894 } 1895 1896 return mixerStatus; 1897} 1898 1899void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1900{ 1901 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1902 this, streamType, mTracks.size()); 1903 Mutex::Autolock _l(mLock); 1904 1905 size_t size = mTracks.size(); 1906 for (size_t i = 0; i < size; i++) { 1907 sp<Track> t = mTracks[i]; 1908 if (t->type() == streamType) { 1909 t->mCblk->lock.lock(); 1910 t->mCblk->flags |= CBLK_INVALID_ON; 1911 t->mCblk->cv.signal(); 1912 t->mCblk->lock.unlock(); 1913 } 1914 } 1915} 1916 1917 1918// getTrackName_l() must be called with ThreadBase::mLock held 1919int AudioFlinger::MixerThread::getTrackName_l() 1920{ 1921 return mAudioMixer->getTrackName(); 1922} 1923 1924// deleteTrackName_l() must be called with ThreadBase::mLock held 1925void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1926{ 1927 LOGV("remove track (%d) and delete from mixer", name); 1928 mAudioMixer->deleteTrackName(name); 1929} 1930 1931// checkForNewParameters_l() must be called with ThreadBase::mLock held 1932bool AudioFlinger::MixerThread::checkForNewParameters_l() 1933{ 1934 bool reconfig = false; 1935 1936 while (!mNewParameters.isEmpty()) { 1937 status_t status = NO_ERROR; 1938 String8 keyValuePair = mNewParameters[0]; 1939 AudioParameter param = AudioParameter(keyValuePair); 1940 int value; 1941 1942 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1943 reconfig = true; 1944 } 1945 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1946 if (value != AudioSystem::PCM_16_BIT) { 1947 status = BAD_VALUE; 1948 } else { 1949 reconfig = true; 1950 } 1951 } 1952 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1953 if (value != AudioSystem::CHANNEL_OUT_STEREO) { 1954 status = BAD_VALUE; 1955 } else { 1956 reconfig = true; 1957 } 1958 } 1959 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1960 // do not accept frame count changes if tracks are open as the track buffer 1961 // size depends on frame count and correct behavior would not be garantied 1962 // if frame count is changed after track creation 1963 if (!mTracks.isEmpty()) { 1964 status = INVALID_OPERATION; 1965 } else { 1966 reconfig = true; 1967 } 1968 } 1969 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1970 // forward device change to effects that have requested to be 1971 // aware of attached audio device. 1972 mDevice = (uint32_t)value; 1973 for (size_t i = 0; i < mEffectChains.size(); i++) { 1974 mEffectChains[i]->setDevice_l(mDevice); 1975 } 1976 } 1977 1978 if (status == NO_ERROR) { 1979 status = mOutput->setParameters(keyValuePair); 1980 if (!mStandby && status == INVALID_OPERATION) { 1981 mOutput->standby(); 1982 mStandby = true; 1983 mBytesWritten = 0; 1984 status = mOutput->setParameters(keyValuePair); 1985 } 1986 if (status == NO_ERROR && reconfig) { 1987 delete mAudioMixer; 1988 readOutputParameters(); 1989 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1990 for (size_t i = 0; i < mTracks.size() ; i++) { 1991 int name = getTrackName_l(); 1992 if (name < 0) break; 1993 mTracks[i]->mName = name; 1994 // limit track sample rate to 2 x new output sample rate 1995 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1996 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 1997 } 1998 } 1999 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2000 } 2001 } 2002 2003 mNewParameters.removeAt(0); 2004 2005 mParamStatus = status; 2006 mParamCond.signal(); 2007 mWaitWorkCV.wait(mLock); 2008 } 2009 return reconfig; 2010} 2011 2012status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2013{ 2014 const size_t SIZE = 256; 2015 char buffer[SIZE]; 2016 String8 result; 2017 2018 PlaybackThread::dumpInternals(fd, args); 2019 2020 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2021 result.append(buffer); 2022 write(fd, result.string(), result.size()); 2023 return NO_ERROR; 2024} 2025 2026uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2027{ 2028 return (uint32_t)(mOutput->latency() * 1000) / 2; 2029} 2030 2031uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2032{ 2033 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2034} 2035 2036uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2037{ 2038 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2039} 2040 2041// ---------------------------------------------------------------------------- 2042AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2043 : PlaybackThread(audioFlinger, output, id, device) 2044{ 2045 mType = PlaybackThread::DIRECT; 2046} 2047 2048AudioFlinger::DirectOutputThread::~DirectOutputThread() 2049{ 2050} 2051 2052 2053static inline int16_t clamp16(int32_t sample) 2054{ 2055 if ((sample>>15) ^ (sample>>31)) 2056 sample = 0x7FFF ^ (sample>>31); 2057 return sample; 2058} 2059 2060static inline 2061int32_t mul(int16_t in, int16_t v) 2062{ 2063#if defined(__arm__) && !defined(__thumb__) 2064 int32_t out; 2065 asm( "smulbb %[out], %[in], %[v] \n" 2066 : [out]"=r"(out) 2067 : [in]"%r"(in), [v]"r"(v) 2068 : ); 2069 return out; 2070#else 2071 return in * int32_t(v); 2072#endif 2073} 2074 2075void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2076{ 2077 // Do not apply volume on compressed audio 2078 if (!AudioSystem::isLinearPCM(mFormat)) { 2079 return; 2080 } 2081 2082 // convert to signed 16 bit before volume calculation 2083 if (mFormat == AudioSystem::PCM_8_BIT) { 2084 size_t count = mFrameCount * mChannelCount; 2085 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2086 int16_t *dst = mMixBuffer + count-1; 2087 while(count--) { 2088 *dst-- = (int16_t)(*src--^0x80) << 8; 2089 } 2090 } 2091 2092 size_t frameCount = mFrameCount; 2093 int16_t *out = mMixBuffer; 2094 if (ramp) { 2095 if (mChannelCount == 1) { 2096 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2097 int32_t vlInc = d / (int32_t)frameCount; 2098 int32_t vl = ((int32_t)mLeftVolShort << 16); 2099 do { 2100 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2101 out++; 2102 vl += vlInc; 2103 } while (--frameCount); 2104 2105 } else { 2106 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2107 int32_t vlInc = d / (int32_t)frameCount; 2108 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2109 int32_t vrInc = d / (int32_t)frameCount; 2110 int32_t vl = ((int32_t)mLeftVolShort << 16); 2111 int32_t vr = ((int32_t)mRightVolShort << 16); 2112 do { 2113 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2114 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2115 out += 2; 2116 vl += vlInc; 2117 vr += vrInc; 2118 } while (--frameCount); 2119 } 2120 } else { 2121 if (mChannelCount == 1) { 2122 do { 2123 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2124 out++; 2125 } while (--frameCount); 2126 } else { 2127 do { 2128 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2129 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2130 out += 2; 2131 } while (--frameCount); 2132 } 2133 } 2134 2135 // convert back to unsigned 8 bit after volume calculation 2136 if (mFormat == AudioSystem::PCM_8_BIT) { 2137 size_t count = mFrameCount * mChannelCount; 2138 int16_t *src = mMixBuffer; 2139 uint8_t *dst = (uint8_t *)mMixBuffer; 2140 while(count--) { 2141 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2142 } 2143 } 2144 2145 mLeftVolShort = leftVol; 2146 mRightVolShort = rightVol; 2147} 2148 2149bool AudioFlinger::DirectOutputThread::threadLoop() 2150{ 2151 uint32_t mixerStatus = MIXER_IDLE; 2152 sp<Track> trackToRemove; 2153 sp<Track> activeTrack; 2154 nsecs_t standbyTime = systemTime(); 2155 int8_t *curBuf; 2156 size_t mixBufferSize = mFrameCount*mFrameSize; 2157 uint32_t activeSleepTime = activeSleepTimeUs(); 2158 uint32_t idleSleepTime = idleSleepTimeUs(); 2159 uint32_t sleepTime = idleSleepTime; 2160 // use shorter standby delay as on normal output to release 2161 // hardware resources as soon as possible 2162 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2163 2164 while (!exitPending()) 2165 { 2166 bool rampVolume; 2167 uint16_t leftVol; 2168 uint16_t rightVol; 2169 Vector< sp<EffectChain> > effectChains; 2170 2171 processConfigEvents(); 2172 2173 mixerStatus = MIXER_IDLE; 2174 2175 { // scope for the mLock 2176 2177 Mutex::Autolock _l(mLock); 2178 2179 if (checkForNewParameters_l()) { 2180 mixBufferSize = mFrameCount*mFrameSize; 2181 activeSleepTime = activeSleepTimeUs(); 2182 idleSleepTime = idleSleepTimeUs(); 2183 standbyDelay = microseconds(activeSleepTime*2); 2184 } 2185 2186 // put audio hardware into standby after short delay 2187 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2188 mSuspended) { 2189 // wait until we have something to do... 2190 if (!mStandby) { 2191 LOGV("Audio hardware entering standby, mixer %p\n", this); 2192 mOutput->standby(); 2193 mStandby = true; 2194 mBytesWritten = 0; 2195 } 2196 2197 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2198 // we're about to wait, flush the binder command buffer 2199 IPCThreadState::self()->flushCommands(); 2200 2201 if (exitPending()) break; 2202 2203 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2204 mWaitWorkCV.wait(mLock); 2205 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2206 2207 if (mMasterMute == false) { 2208 char value[PROPERTY_VALUE_MAX]; 2209 property_get("ro.audio.silent", value, "0"); 2210 if (atoi(value)) { 2211 LOGD("Silence is golden"); 2212 setMasterMute(true); 2213 } 2214 } 2215 2216 standbyTime = systemTime() + standbyDelay; 2217 sleepTime = idleSleepTime; 2218 continue; 2219 } 2220 } 2221 2222 effectChains = mEffectChains; 2223 2224 // find out which tracks need to be processed 2225 if (mActiveTracks.size() != 0) { 2226 sp<Track> t = mActiveTracks[0].promote(); 2227 if (t == 0) continue; 2228 2229 Track* const track = t.get(); 2230 audio_track_cblk_t* cblk = track->cblk(); 2231 2232 // The first time a track is added we wait 2233 // for all its buffers to be filled before processing it 2234 if (cblk->framesReady() && track->isReady() && 2235 !track->isPaused() && !track->isTerminated()) 2236 { 2237 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2238 2239 if (track->mFillingUpStatus == Track::FS_FILLED) { 2240 track->mFillingUpStatus = Track::FS_ACTIVE; 2241 mLeftVolFloat = mRightVolFloat = 0; 2242 mLeftVolShort = mRightVolShort = 0; 2243 if (track->mState == TrackBase::RESUMING) { 2244 track->mState = TrackBase::ACTIVE; 2245 rampVolume = true; 2246 } 2247 } else if (cblk->server != 0) { 2248 // If the track is stopped before the first frame was mixed, 2249 // do not apply ramp 2250 rampVolume = true; 2251 } 2252 // compute volume for this track 2253 float left, right; 2254 if (track->isMuted() || mMasterMute || track->isPausing() || 2255 mStreamTypes[track->type()].mute) { 2256 left = right = 0; 2257 if (track->isPausing()) { 2258 track->setPaused(); 2259 } 2260 } else { 2261 float typeVolume = mStreamTypes[track->type()].volume; 2262 float v = mMasterVolume * typeVolume; 2263 float v_clamped = v * cblk->volume[0]; 2264 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2265 left = v_clamped/MAX_GAIN; 2266 v_clamped = v * cblk->volume[1]; 2267 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2268 right = v_clamped/MAX_GAIN; 2269 } 2270 2271 if (left != mLeftVolFloat || right != mRightVolFloat) { 2272 mLeftVolFloat = left; 2273 mRightVolFloat = right; 2274 2275 // If audio HAL implements volume control, 2276 // force software volume to nominal value 2277 if (mOutput->setVolume(left, right) == NO_ERROR) { 2278 left = 1.0f; 2279 right = 1.0f; 2280 } 2281 2282 // Convert volumes from float to 8.24 2283 uint32_t vl = (uint32_t)(left * (1 << 24)); 2284 uint32_t vr = (uint32_t)(right * (1 << 24)); 2285 2286 // Delegate volume control to effect in track effect chain if needed 2287 // only one effect chain can be present on DirectOutputThread, so if 2288 // there is one, the track is connected to it 2289 if (!effectChains.isEmpty()) { 2290 // Do not ramp volume if volume is controlled by effect 2291 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2292 rampVolume = false; 2293 } 2294 } 2295 2296 // Convert volumes from 8.24 to 4.12 format 2297 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2298 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2299 leftVol = (uint16_t)v_clamped; 2300 v_clamped = (vr + (1 << 11)) >> 12; 2301 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2302 rightVol = (uint16_t)v_clamped; 2303 } else { 2304 leftVol = mLeftVolShort; 2305 rightVol = mRightVolShort; 2306 rampVolume = false; 2307 } 2308 2309 // reset retry count 2310 track->mRetryCount = kMaxTrackRetriesDirect; 2311 activeTrack = t; 2312 mixerStatus = MIXER_TRACKS_READY; 2313 } else { 2314 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2315 if (track->isStopped()) { 2316 track->reset(); 2317 } 2318 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2319 // We have consumed all the buffers of this track. 2320 // Remove it from the list of active tracks. 2321 trackToRemove = track; 2322 } else { 2323 // No buffers for this track. Give it a few chances to 2324 // fill a buffer, then remove it from active list. 2325 if (--(track->mRetryCount) <= 0) { 2326 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2327 trackToRemove = track; 2328 } else { 2329 mixerStatus = MIXER_TRACKS_ENABLED; 2330 } 2331 } 2332 } 2333 } 2334 2335 // remove all the tracks that need to be... 2336 if (UNLIKELY(trackToRemove != 0)) { 2337 mActiveTracks.remove(trackToRemove); 2338 if (!effectChains.isEmpty()) { 2339 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2340 trackToRemove->sessionId()); 2341 effectChains[0]->stopTrack(); 2342 } 2343 if (trackToRemove->isTerminated()) { 2344 mTracks.remove(trackToRemove); 2345 deleteTrackName_l(trackToRemove->mName); 2346 } 2347 } 2348 2349 lockEffectChains_l(effectChains); 2350 } 2351 2352 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2353 AudioBufferProvider::Buffer buffer; 2354 size_t frameCount = mFrameCount; 2355 curBuf = (int8_t *)mMixBuffer; 2356 // output audio to hardware 2357 while (frameCount) { 2358 buffer.frameCount = frameCount; 2359 activeTrack->getNextBuffer(&buffer); 2360 if (UNLIKELY(buffer.raw == 0)) { 2361 memset(curBuf, 0, frameCount * mFrameSize); 2362 break; 2363 } 2364 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2365 frameCount -= buffer.frameCount; 2366 curBuf += buffer.frameCount * mFrameSize; 2367 activeTrack->releaseBuffer(&buffer); 2368 } 2369 sleepTime = 0; 2370 standbyTime = systemTime() + standbyDelay; 2371 } else { 2372 if (sleepTime == 0) { 2373 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2374 sleepTime = activeSleepTime; 2375 } else { 2376 sleepTime = idleSleepTime; 2377 } 2378 } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { 2379 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2380 sleepTime = 0; 2381 } 2382 } 2383 2384 if (mSuspended) { 2385 sleepTime = suspendSleepTimeUs(); 2386 } 2387 // sleepTime == 0 means we must write to audio hardware 2388 if (sleepTime == 0) { 2389 if (mixerStatus == MIXER_TRACKS_READY) { 2390 applyVolume(leftVol, rightVol, rampVolume); 2391 } 2392 for (size_t i = 0; i < effectChains.size(); i ++) { 2393 effectChains[i]->process_l(); 2394 } 2395 unlockEffectChains(effectChains); 2396 2397 mLastWriteTime = systemTime(); 2398 mInWrite = true; 2399 mBytesWritten += mixBufferSize; 2400 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 2401 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2402 mNumWrites++; 2403 mInWrite = false; 2404 mStandby = false; 2405 } else { 2406 unlockEffectChains(effectChains); 2407 usleep(sleepTime); 2408 } 2409 2410 // finally let go of removed track, without the lock held 2411 // since we can't guarantee the destructors won't acquire that 2412 // same lock. 2413 trackToRemove.clear(); 2414 activeTrack.clear(); 2415 2416 // Effect chains will be actually deleted here if they were removed from 2417 // mEffectChains list during mixing or effects processing 2418 effectChains.clear(); 2419 } 2420 2421 if (!mStandby) { 2422 mOutput->standby(); 2423 } 2424 2425 LOGV("DirectOutputThread %p exiting", this); 2426 return false; 2427} 2428 2429// getTrackName_l() must be called with ThreadBase::mLock held 2430int AudioFlinger::DirectOutputThread::getTrackName_l() 2431{ 2432 return 0; 2433} 2434 2435// deleteTrackName_l() must be called with ThreadBase::mLock held 2436void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2437{ 2438} 2439 2440// checkForNewParameters_l() must be called with ThreadBase::mLock held 2441bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2442{ 2443 bool reconfig = false; 2444 2445 while (!mNewParameters.isEmpty()) { 2446 status_t status = NO_ERROR; 2447 String8 keyValuePair = mNewParameters[0]; 2448 AudioParameter param = AudioParameter(keyValuePair); 2449 int value; 2450 2451 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2452 // do not accept frame count changes if tracks are open as the track buffer 2453 // size depends on frame count and correct behavior would not be garantied 2454 // if frame count is changed after track creation 2455 if (!mTracks.isEmpty()) { 2456 status = INVALID_OPERATION; 2457 } else { 2458 reconfig = true; 2459 } 2460 } 2461 if (status == NO_ERROR) { 2462 status = mOutput->setParameters(keyValuePair); 2463 if (!mStandby && status == INVALID_OPERATION) { 2464 mOutput->standby(); 2465 mStandby = true; 2466 mBytesWritten = 0; 2467 status = mOutput->setParameters(keyValuePair); 2468 } 2469 if (status == NO_ERROR && reconfig) { 2470 readOutputParameters(); 2471 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2472 } 2473 } 2474 2475 mNewParameters.removeAt(0); 2476 2477 mParamStatus = status; 2478 mParamCond.signal(); 2479 mWaitWorkCV.wait(mLock); 2480 } 2481 return reconfig; 2482} 2483 2484uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2485{ 2486 uint32_t time; 2487 if (AudioSystem::isLinearPCM(mFormat)) { 2488 time = (uint32_t)(mOutput->latency() * 1000) / 2; 2489 } else { 2490 time = 10000; 2491 } 2492 return time; 2493} 2494 2495uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2496{ 2497 uint32_t time; 2498 if (AudioSystem::isLinearPCM(mFormat)) { 2499 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2500 } else { 2501 time = 10000; 2502 } 2503 return time; 2504} 2505 2506uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2507{ 2508 uint32_t time; 2509 if (AudioSystem::isLinearPCM(mFormat)) { 2510 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2511 } else { 2512 time = 10000; 2513 } 2514 return time; 2515} 2516 2517 2518// ---------------------------------------------------------------------------- 2519 2520AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2521 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2522{ 2523 mType = PlaybackThread::DUPLICATING; 2524 addOutputTrack(mainThread); 2525} 2526 2527AudioFlinger::DuplicatingThread::~DuplicatingThread() 2528{ 2529 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2530 mOutputTracks[i]->destroy(); 2531 } 2532 mOutputTracks.clear(); 2533} 2534 2535bool AudioFlinger::DuplicatingThread::threadLoop() 2536{ 2537 Vector< sp<Track> > tracksToRemove; 2538 uint32_t mixerStatus = MIXER_IDLE; 2539 nsecs_t standbyTime = systemTime(); 2540 size_t mixBufferSize = mFrameCount*mFrameSize; 2541 SortedVector< sp<OutputTrack> > outputTracks; 2542 uint32_t writeFrames = 0; 2543 uint32_t activeSleepTime = activeSleepTimeUs(); 2544 uint32_t idleSleepTime = idleSleepTimeUs(); 2545 uint32_t sleepTime = idleSleepTime; 2546 Vector< sp<EffectChain> > effectChains; 2547 2548 while (!exitPending()) 2549 { 2550 processConfigEvents(); 2551 2552 mixerStatus = MIXER_IDLE; 2553 { // scope for the mLock 2554 2555 Mutex::Autolock _l(mLock); 2556 2557 if (checkForNewParameters_l()) { 2558 mixBufferSize = mFrameCount*mFrameSize; 2559 updateWaitTime(); 2560 activeSleepTime = activeSleepTimeUs(); 2561 idleSleepTime = idleSleepTimeUs(); 2562 } 2563 2564 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2565 2566 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2567 outputTracks.add(mOutputTracks[i]); 2568 } 2569 2570 // put audio hardware into standby after short delay 2571 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2572 mSuspended) { 2573 if (!mStandby) { 2574 for (size_t i = 0; i < outputTracks.size(); i++) { 2575 outputTracks[i]->stop(); 2576 } 2577 mStandby = true; 2578 mBytesWritten = 0; 2579 } 2580 2581 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2582 // we're about to wait, flush the binder command buffer 2583 IPCThreadState::self()->flushCommands(); 2584 outputTracks.clear(); 2585 2586 if (exitPending()) break; 2587 2588 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2589 mWaitWorkCV.wait(mLock); 2590 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2591 if (mMasterMute == false) { 2592 char value[PROPERTY_VALUE_MAX]; 2593 property_get("ro.audio.silent", value, "0"); 2594 if (atoi(value)) { 2595 LOGD("Silence is golden"); 2596 setMasterMute(true); 2597 } 2598 } 2599 2600 standbyTime = systemTime() + kStandbyTimeInNsecs; 2601 sleepTime = idleSleepTime; 2602 continue; 2603 } 2604 } 2605 2606 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2607 2608 // prevent any changes in effect chain list and in each effect chain 2609 // during mixing and effect process as the audio buffers could be deleted 2610 // or modified if an effect is created or deleted 2611 lockEffectChains_l(effectChains); 2612 } 2613 2614 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2615 // mix buffers... 2616 if (outputsReady(outputTracks)) { 2617 mAudioMixer->process(); 2618 } else { 2619 memset(mMixBuffer, 0, mixBufferSize); 2620 } 2621 sleepTime = 0; 2622 writeFrames = mFrameCount; 2623 } else { 2624 if (sleepTime == 0) { 2625 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2626 sleepTime = activeSleepTime; 2627 } else { 2628 sleepTime = idleSleepTime; 2629 } 2630 } else if (mBytesWritten != 0) { 2631 // flush remaining overflow buffers in output tracks 2632 for (size_t i = 0; i < outputTracks.size(); i++) { 2633 if (outputTracks[i]->isActive()) { 2634 sleepTime = 0; 2635 writeFrames = 0; 2636 memset(mMixBuffer, 0, mixBufferSize); 2637 break; 2638 } 2639 } 2640 } 2641 } 2642 2643 if (mSuspended) { 2644 sleepTime = suspendSleepTimeUs(); 2645 } 2646 // sleepTime == 0 means we must write to audio hardware 2647 if (sleepTime == 0) { 2648 for (size_t i = 0; i < effectChains.size(); i ++) { 2649 effectChains[i]->process_l(); 2650 } 2651 // enable changes in effect chain 2652 unlockEffectChains(effectChains); 2653 2654 standbyTime = systemTime() + kStandbyTimeInNsecs; 2655 for (size_t i = 0; i < outputTracks.size(); i++) { 2656 outputTracks[i]->write(mMixBuffer, writeFrames); 2657 } 2658 mStandby = false; 2659 mBytesWritten += mixBufferSize; 2660 } else { 2661 // enable changes in effect chain 2662 unlockEffectChains(effectChains); 2663 usleep(sleepTime); 2664 } 2665 2666 // finally let go of all our tracks, without the lock held 2667 // since we can't guarantee the destructors won't acquire that 2668 // same lock. 2669 tracksToRemove.clear(); 2670 outputTracks.clear(); 2671 2672 // Effect chains will be actually deleted here if they were removed from 2673 // mEffectChains list during mixing or effects processing 2674 effectChains.clear(); 2675 } 2676 2677 return false; 2678} 2679 2680void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2681{ 2682 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2683 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2684 this, 2685 mSampleRate, 2686 mFormat, 2687 mChannelCount, 2688 frameCount); 2689 if (outputTrack->cblk() != NULL) { 2690 thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); 2691 mOutputTracks.add(outputTrack); 2692 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2693 updateWaitTime(); 2694 } 2695} 2696 2697void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2698{ 2699 Mutex::Autolock _l(mLock); 2700 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2701 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2702 mOutputTracks[i]->destroy(); 2703 mOutputTracks.removeAt(i); 2704 updateWaitTime(); 2705 return; 2706 } 2707 } 2708 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2709} 2710 2711void AudioFlinger::DuplicatingThread::updateWaitTime() 2712{ 2713 mWaitTimeMs = UINT_MAX; 2714 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2715 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2716 if (strong != NULL) { 2717 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2718 if (waitTimeMs < mWaitTimeMs) { 2719 mWaitTimeMs = waitTimeMs; 2720 } 2721 } 2722 } 2723} 2724 2725 2726bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2727{ 2728 for (size_t i = 0; i < outputTracks.size(); i++) { 2729 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2730 if (thread == 0) { 2731 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2732 return false; 2733 } 2734 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2735 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2736 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2737 return false; 2738 } 2739 } 2740 return true; 2741} 2742 2743uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2744{ 2745 return (mWaitTimeMs * 1000) / 2; 2746} 2747 2748// ---------------------------------------------------------------------------- 2749 2750// TrackBase constructor must be called with AudioFlinger::mLock held 2751AudioFlinger::ThreadBase::TrackBase::TrackBase( 2752 const wp<ThreadBase>& thread, 2753 const sp<Client>& client, 2754 uint32_t sampleRate, 2755 int format, 2756 int channelCount, 2757 int frameCount, 2758 uint32_t flags, 2759 const sp<IMemory>& sharedBuffer, 2760 int sessionId) 2761 : RefBase(), 2762 mThread(thread), 2763 mClient(client), 2764 mCblk(0), 2765 mFrameCount(0), 2766 mState(IDLE), 2767 mClientTid(-1), 2768 mFormat(format), 2769 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2770 mSessionId(sessionId) 2771{ 2772 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2773 2774 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2775 size_t size = sizeof(audio_track_cblk_t); 2776 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2777 if (sharedBuffer == 0) { 2778 size += bufferSize; 2779 } 2780 2781 if (client != NULL) { 2782 mCblkMemory = client->heap()->allocate(size); 2783 if (mCblkMemory != 0) { 2784 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2785 if (mCblk) { // construct the shared structure in-place. 2786 new(mCblk) audio_track_cblk_t(); 2787 // clear all buffers 2788 mCblk->frameCount = frameCount; 2789 mCblk->sampleRate = sampleRate; 2790 mCblk->channelCount = (uint8_t)channelCount; 2791 if (sharedBuffer == 0) { 2792 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2793 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2794 // Force underrun condition to avoid false underrun callback until first data is 2795 // written to buffer (other flags are cleared) 2796 mCblk->flags = CBLK_UNDERRUN_ON; 2797 } else { 2798 mBuffer = sharedBuffer->pointer(); 2799 } 2800 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2801 } 2802 } else { 2803 LOGE("not enough memory for AudioTrack size=%u", size); 2804 client->heap()->dump("AudioTrack"); 2805 return; 2806 } 2807 } else { 2808 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2809 if (mCblk) { // construct the shared structure in-place. 2810 new(mCblk) audio_track_cblk_t(); 2811 // clear all buffers 2812 mCblk->frameCount = frameCount; 2813 mCblk->sampleRate = sampleRate; 2814 mCblk->channelCount = (uint8_t)channelCount; 2815 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2816 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2817 // Force underrun condition to avoid false underrun callback until first data is 2818 // written to buffer (other flags are cleared) 2819 mCblk->flags = CBLK_UNDERRUN_ON; 2820 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2821 } 2822 } 2823} 2824 2825AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2826{ 2827 if (mCblk) { 2828 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2829 if (mClient == NULL) { 2830 delete mCblk; 2831 } 2832 } 2833 mCblkMemory.clear(); // and free the shared memory 2834 if (mClient != NULL) { 2835 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2836 mClient.clear(); 2837 } 2838} 2839 2840void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2841{ 2842 buffer->raw = 0; 2843 mFrameCount = buffer->frameCount; 2844 step(); 2845 buffer->frameCount = 0; 2846} 2847 2848bool AudioFlinger::ThreadBase::TrackBase::step() { 2849 bool result; 2850 audio_track_cblk_t* cblk = this->cblk(); 2851 2852 result = cblk->stepServer(mFrameCount); 2853 if (!result) { 2854 LOGV("stepServer failed acquiring cblk mutex"); 2855 mFlags |= STEPSERVER_FAILED; 2856 } 2857 return result; 2858} 2859 2860void AudioFlinger::ThreadBase::TrackBase::reset() { 2861 audio_track_cblk_t* cblk = this->cblk(); 2862 2863 cblk->user = 0; 2864 cblk->server = 0; 2865 cblk->userBase = 0; 2866 cblk->serverBase = 0; 2867 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2868 LOGV("TrackBase::reset"); 2869} 2870 2871sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2872{ 2873 return mCblkMemory; 2874} 2875 2876int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2877 return (int)mCblk->sampleRate; 2878} 2879 2880int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2881 return (int)mCblk->channelCount; 2882} 2883 2884void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2885 audio_track_cblk_t* cblk = this->cblk(); 2886 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2887 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2888 2889 // Check validity of returned pointer in case the track control block would have been corrupted. 2890 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2891 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2892 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2893 server %d, serverBase %d, user %d, userBase %d, channelCount %d", 2894 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2895 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); 2896 return 0; 2897 } 2898 2899 return bufferStart; 2900} 2901 2902// ---------------------------------------------------------------------------- 2903 2904// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2905AudioFlinger::PlaybackThread::Track::Track( 2906 const wp<ThreadBase>& thread, 2907 const sp<Client>& client, 2908 int streamType, 2909 uint32_t sampleRate, 2910 int format, 2911 int channelCount, 2912 int frameCount, 2913 const sp<IMemory>& sharedBuffer, 2914 int sessionId) 2915 : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), 2916 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 2917 mAuxEffectId(0), mHasVolumeController(false) 2918{ 2919 if (mCblk != NULL) { 2920 sp<ThreadBase> baseThread = thread.promote(); 2921 if (baseThread != 0) { 2922 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2923 mName = playbackThread->getTrackName_l(); 2924 mMainBuffer = playbackThread->mixBuffer(); 2925 } 2926 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2927 if (mName < 0) { 2928 LOGE("no more track names available"); 2929 } 2930 mVolume[0] = 1.0f; 2931 mVolume[1] = 1.0f; 2932 mStreamType = streamType; 2933 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2934 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2935 mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); 2936 } 2937} 2938 2939AudioFlinger::PlaybackThread::Track::~Track() 2940{ 2941 LOGV("PlaybackThread::Track destructor"); 2942 sp<ThreadBase> thread = mThread.promote(); 2943 if (thread != 0) { 2944 Mutex::Autolock _l(thread->mLock); 2945 mState = TERMINATED; 2946 } 2947} 2948 2949void AudioFlinger::PlaybackThread::Track::destroy() 2950{ 2951 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2952 // by removing it from mTracks vector, so there is a risk that this Tracks's 2953 // desctructor is called. As the destructor needs to lock mLock, 2954 // we must acquire a strong reference on this Track before locking mLock 2955 // here so that the destructor is called only when exiting this function. 2956 // On the other hand, as long as Track::destroy() is only called by 2957 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2958 // this Track with its member mTrack. 2959 sp<Track> keep(this); 2960 { // scope for mLock 2961 sp<ThreadBase> thread = mThread.promote(); 2962 if (thread != 0) { 2963 if (!isOutputTrack()) { 2964 if (mState == ACTIVE || mState == RESUMING) { 2965 AudioSystem::stopOutput(thread->id(), 2966 (AudioSystem::stream_type)mStreamType, 2967 mSessionId); 2968 } 2969 AudioSystem::releaseOutput(thread->id()); 2970 } 2971 Mutex::Autolock _l(thread->mLock); 2972 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2973 playbackThread->destroyTrack_l(this); 2974 } 2975 } 2976} 2977 2978void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2979{ 2980 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", 2981 mName - AudioMixer::TRACK0, 2982 (mClient == NULL) ? getpid() : mClient->pid(), 2983 mStreamType, 2984 mFormat, 2985 mCblk->channelCount, 2986 mSessionId, 2987 mFrameCount, 2988 mState, 2989 mMute, 2990 mFillingUpStatus, 2991 mCblk->sampleRate, 2992 mCblk->volume[0], 2993 mCblk->volume[1], 2994 mCblk->server, 2995 mCblk->user, 2996 (int)mMainBuffer, 2997 (int)mAuxBuffer); 2998} 2999 3000status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3001{ 3002 audio_track_cblk_t* cblk = this->cblk(); 3003 uint32_t framesReady; 3004 uint32_t framesReq = buffer->frameCount; 3005 3006 // Check if last stepServer failed, try to step now 3007 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3008 if (!step()) goto getNextBuffer_exit; 3009 LOGV("stepServer recovered"); 3010 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3011 } 3012 3013 framesReady = cblk->framesReady(); 3014 3015 if (LIKELY(framesReady)) { 3016 uint32_t s = cblk->server; 3017 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3018 3019 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3020 if (framesReq > framesReady) { 3021 framesReq = framesReady; 3022 } 3023 if (s + framesReq > bufferEnd) { 3024 framesReq = bufferEnd - s; 3025 } 3026 3027 buffer->raw = getBuffer(s, framesReq); 3028 if (buffer->raw == 0) goto getNextBuffer_exit; 3029 3030 buffer->frameCount = framesReq; 3031 return NO_ERROR; 3032 } 3033 3034getNextBuffer_exit: 3035 buffer->raw = 0; 3036 buffer->frameCount = 0; 3037 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3038 return NOT_ENOUGH_DATA; 3039} 3040 3041bool AudioFlinger::PlaybackThread::Track::isReady() const { 3042 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3043 3044 if (mCblk->framesReady() >= mCblk->frameCount || 3045 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3046 mFillingUpStatus = FS_FILLED; 3047 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3048 return true; 3049 } 3050 return false; 3051} 3052 3053status_t AudioFlinger::PlaybackThread::Track::start() 3054{ 3055 status_t status = NO_ERROR; 3056 LOGV("start(%d), calling thread %d session %d", 3057 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3058 sp<ThreadBase> thread = mThread.promote(); 3059 if (thread != 0) { 3060 Mutex::Autolock _l(thread->mLock); 3061 int state = mState; 3062 // here the track could be either new, or restarted 3063 // in both cases "unstop" the track 3064 if (mState == PAUSED) { 3065 mState = TrackBase::RESUMING; 3066 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3067 } else { 3068 mState = TrackBase::ACTIVE; 3069 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3070 } 3071 3072 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3073 thread->mLock.unlock(); 3074 status = AudioSystem::startOutput(thread->id(), 3075 (AudioSystem::stream_type)mStreamType, 3076 mSessionId); 3077 thread->mLock.lock(); 3078 } 3079 if (status == NO_ERROR) { 3080 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3081 playbackThread->addTrack_l(this); 3082 } else { 3083 mState = state; 3084 } 3085 } else { 3086 status = BAD_VALUE; 3087 } 3088 return status; 3089} 3090 3091void AudioFlinger::PlaybackThread::Track::stop() 3092{ 3093 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3094 sp<ThreadBase> thread = mThread.promote(); 3095 if (thread != 0) { 3096 Mutex::Autolock _l(thread->mLock); 3097 int state = mState; 3098 if (mState > STOPPED) { 3099 mState = STOPPED; 3100 // If the track is not active (PAUSED and buffers full), flush buffers 3101 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3102 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3103 reset(); 3104 } 3105 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3106 } 3107 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3108 thread->mLock.unlock(); 3109 AudioSystem::stopOutput(thread->id(), 3110 (AudioSystem::stream_type)mStreamType, 3111 mSessionId); 3112 thread->mLock.lock(); 3113 } 3114 } 3115} 3116 3117void AudioFlinger::PlaybackThread::Track::pause() 3118{ 3119 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3120 sp<ThreadBase> thread = mThread.promote(); 3121 if (thread != 0) { 3122 Mutex::Autolock _l(thread->mLock); 3123 if (mState == ACTIVE || mState == RESUMING) { 3124 mState = PAUSING; 3125 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3126 if (!isOutputTrack()) { 3127 thread->mLock.unlock(); 3128 AudioSystem::stopOutput(thread->id(), 3129 (AudioSystem::stream_type)mStreamType, 3130 mSessionId); 3131 thread->mLock.lock(); 3132 } 3133 } 3134 } 3135} 3136 3137void AudioFlinger::PlaybackThread::Track::flush() 3138{ 3139 LOGV("flush(%d)", mName); 3140 sp<ThreadBase> thread = mThread.promote(); 3141 if (thread != 0) { 3142 Mutex::Autolock _l(thread->mLock); 3143 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3144 return; 3145 } 3146 // No point remaining in PAUSED state after a flush => go to 3147 // STOPPED state 3148 mState = STOPPED; 3149 3150 mCblk->lock.lock(); 3151 // NOTE: reset() will reset cblk->user and cblk->server with 3152 // the risk that at the same time, the AudioMixer is trying to read 3153 // data. In this case, getNextBuffer() would return a NULL pointer 3154 // as audio buffer => the AudioMixer code MUST always test that pointer 3155 // returned by getNextBuffer() is not NULL! 3156 reset(); 3157 mCblk->lock.unlock(); 3158 } 3159} 3160 3161void AudioFlinger::PlaybackThread::Track::reset() 3162{ 3163 // Do not reset twice to avoid discarding data written just after a flush and before 3164 // the audioflinger thread detects the track is stopped. 3165 if (!mResetDone) { 3166 TrackBase::reset(); 3167 // Force underrun condition to avoid false underrun callback until first data is 3168 // written to buffer 3169 mCblk->flags |= CBLK_UNDERRUN_ON; 3170 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3171 mFillingUpStatus = FS_FILLING; 3172 mResetDone = true; 3173 } 3174} 3175 3176void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3177{ 3178 mMute = muted; 3179} 3180 3181void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3182{ 3183 mVolume[0] = left; 3184 mVolume[1] = right; 3185} 3186 3187status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3188{ 3189 status_t status = DEAD_OBJECT; 3190 sp<ThreadBase> thread = mThread.promote(); 3191 if (thread != 0) { 3192 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3193 status = playbackThread->attachAuxEffect(this, EffectId); 3194 } 3195 return status; 3196} 3197 3198void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3199{ 3200 mAuxEffectId = EffectId; 3201 mAuxBuffer = buffer; 3202} 3203 3204// ---------------------------------------------------------------------------- 3205 3206// RecordTrack constructor must be called with AudioFlinger::mLock held 3207AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3208 const wp<ThreadBase>& thread, 3209 const sp<Client>& client, 3210 uint32_t sampleRate, 3211 int format, 3212 int channelCount, 3213 int frameCount, 3214 uint32_t flags, 3215 int sessionId) 3216 : TrackBase(thread, client, sampleRate, format, 3217 channelCount, frameCount, flags, 0, sessionId), 3218 mOverflow(false) 3219{ 3220 if (mCblk != NULL) { 3221 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3222 if (format == AudioSystem::PCM_16_BIT) { 3223 mCblk->frameSize = channelCount * sizeof(int16_t); 3224 } else if (format == AudioSystem::PCM_8_BIT) { 3225 mCblk->frameSize = channelCount * sizeof(int8_t); 3226 } else { 3227 mCblk->frameSize = sizeof(int8_t); 3228 } 3229 } 3230} 3231 3232AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3233{ 3234 sp<ThreadBase> thread = mThread.promote(); 3235 if (thread != 0) { 3236 AudioSystem::releaseInput(thread->id()); 3237 } 3238} 3239 3240status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3241{ 3242 audio_track_cblk_t* cblk = this->cblk(); 3243 uint32_t framesAvail; 3244 uint32_t framesReq = buffer->frameCount; 3245 3246 // Check if last stepServer failed, try to step now 3247 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3248 if (!step()) goto getNextBuffer_exit; 3249 LOGV("stepServer recovered"); 3250 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3251 } 3252 3253 framesAvail = cblk->framesAvailable_l(); 3254 3255 if (LIKELY(framesAvail)) { 3256 uint32_t s = cblk->server; 3257 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3258 3259 if (framesReq > framesAvail) { 3260 framesReq = framesAvail; 3261 } 3262 if (s + framesReq > bufferEnd) { 3263 framesReq = bufferEnd - s; 3264 } 3265 3266 buffer->raw = getBuffer(s, framesReq); 3267 if (buffer->raw == 0) goto getNextBuffer_exit; 3268 3269 buffer->frameCount = framesReq; 3270 return NO_ERROR; 3271 } 3272 3273getNextBuffer_exit: 3274 buffer->raw = 0; 3275 buffer->frameCount = 0; 3276 return NOT_ENOUGH_DATA; 3277} 3278 3279status_t AudioFlinger::RecordThread::RecordTrack::start() 3280{ 3281 sp<ThreadBase> thread = mThread.promote(); 3282 if (thread != 0) { 3283 RecordThread *recordThread = (RecordThread *)thread.get(); 3284 return recordThread->start(this); 3285 } else { 3286 return BAD_VALUE; 3287 } 3288} 3289 3290void AudioFlinger::RecordThread::RecordTrack::stop() 3291{ 3292 sp<ThreadBase> thread = mThread.promote(); 3293 if (thread != 0) { 3294 RecordThread *recordThread = (RecordThread *)thread.get(); 3295 recordThread->stop(this); 3296 TrackBase::reset(); 3297 // Force overerrun condition to avoid false overrun callback until first data is 3298 // read from buffer 3299 mCblk->flags |= CBLK_UNDERRUN_ON; 3300 } 3301} 3302 3303void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3304{ 3305 snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", 3306 (mClient == NULL) ? getpid() : mClient->pid(), 3307 mFormat, 3308 mCblk->channelCount, 3309 mSessionId, 3310 mFrameCount, 3311 mState, 3312 mCblk->sampleRate, 3313 mCblk->server, 3314 mCblk->user); 3315} 3316 3317 3318// ---------------------------------------------------------------------------- 3319 3320AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3321 const wp<ThreadBase>& thread, 3322 DuplicatingThread *sourceThread, 3323 uint32_t sampleRate, 3324 int format, 3325 int channelCount, 3326 int frameCount) 3327 : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), 3328 mActive(false), mSourceThread(sourceThread) 3329{ 3330 3331 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3332 if (mCblk != NULL) { 3333 mCblk->flags |= CBLK_DIRECTION_OUT; 3334 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3335 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3336 mOutBuffer.frameCount = 0; 3337 playbackThread->mTracks.add(this); 3338 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", 3339 mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); 3340 } else { 3341 LOGW("Error creating output track on thread %p", playbackThread); 3342 } 3343} 3344 3345AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3346{ 3347 clearBufferQueue(); 3348} 3349 3350status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3351{ 3352 status_t status = Track::start(); 3353 if (status != NO_ERROR) { 3354 return status; 3355 } 3356 3357 mActive = true; 3358 mRetryCount = 127; 3359 return status; 3360} 3361 3362void AudioFlinger::PlaybackThread::OutputTrack::stop() 3363{ 3364 Track::stop(); 3365 clearBufferQueue(); 3366 mOutBuffer.frameCount = 0; 3367 mActive = false; 3368} 3369 3370bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3371{ 3372 Buffer *pInBuffer; 3373 Buffer inBuffer; 3374 uint32_t channelCount = mCblk->channelCount; 3375 bool outputBufferFull = false; 3376 inBuffer.frameCount = frames; 3377 inBuffer.i16 = data; 3378 3379 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3380 3381 if (!mActive && frames != 0) { 3382 start(); 3383 sp<ThreadBase> thread = mThread.promote(); 3384 if (thread != 0) { 3385 MixerThread *mixerThread = (MixerThread *)thread.get(); 3386 if (mCblk->frameCount > frames){ 3387 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3388 uint32_t startFrames = (mCblk->frameCount - frames); 3389 pInBuffer = new Buffer; 3390 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3391 pInBuffer->frameCount = startFrames; 3392 pInBuffer->i16 = pInBuffer->mBuffer; 3393 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3394 mBufferQueue.add(pInBuffer); 3395 } else { 3396 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3397 } 3398 } 3399 } 3400 } 3401 3402 while (waitTimeLeftMs) { 3403 // First write pending buffers, then new data 3404 if (mBufferQueue.size()) { 3405 pInBuffer = mBufferQueue.itemAt(0); 3406 } else { 3407 pInBuffer = &inBuffer; 3408 } 3409 3410 if (pInBuffer->frameCount == 0) { 3411 break; 3412 } 3413 3414 if (mOutBuffer.frameCount == 0) { 3415 mOutBuffer.frameCount = pInBuffer->frameCount; 3416 nsecs_t startTime = systemTime(); 3417 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3418 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3419 outputBufferFull = true; 3420 break; 3421 } 3422 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3423 if (waitTimeLeftMs >= waitTimeMs) { 3424 waitTimeLeftMs -= waitTimeMs; 3425 } else { 3426 waitTimeLeftMs = 0; 3427 } 3428 } 3429 3430 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3431 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3432 mCblk->stepUser(outFrames); 3433 pInBuffer->frameCount -= outFrames; 3434 pInBuffer->i16 += outFrames * channelCount; 3435 mOutBuffer.frameCount -= outFrames; 3436 mOutBuffer.i16 += outFrames * channelCount; 3437 3438 if (pInBuffer->frameCount == 0) { 3439 if (mBufferQueue.size()) { 3440 mBufferQueue.removeAt(0); 3441 delete [] pInBuffer->mBuffer; 3442 delete pInBuffer; 3443 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3444 } else { 3445 break; 3446 } 3447 } 3448 } 3449 3450 // If we could not write all frames, allocate a buffer and queue it for next time. 3451 if (inBuffer.frameCount) { 3452 sp<ThreadBase> thread = mThread.promote(); 3453 if (thread != 0 && !thread->standby()) { 3454 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3455 pInBuffer = new Buffer; 3456 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3457 pInBuffer->frameCount = inBuffer.frameCount; 3458 pInBuffer->i16 = pInBuffer->mBuffer; 3459 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3460 mBufferQueue.add(pInBuffer); 3461 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3462 } else { 3463 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3464 } 3465 } 3466 } 3467 3468 // Calling write() with a 0 length buffer, means that no more data will be written: 3469 // If no more buffers are pending, fill output track buffer to make sure it is started 3470 // by output mixer. 3471 if (frames == 0 && mBufferQueue.size() == 0) { 3472 if (mCblk->user < mCblk->frameCount) { 3473 frames = mCblk->frameCount - mCblk->user; 3474 pInBuffer = new Buffer; 3475 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3476 pInBuffer->frameCount = frames; 3477 pInBuffer->i16 = pInBuffer->mBuffer; 3478 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3479 mBufferQueue.add(pInBuffer); 3480 } else if (mActive) { 3481 stop(); 3482 } 3483 } 3484 3485 return outputBufferFull; 3486} 3487 3488status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3489{ 3490 int active; 3491 status_t result; 3492 audio_track_cblk_t* cblk = mCblk; 3493 uint32_t framesReq = buffer->frameCount; 3494 3495// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3496 buffer->frameCount = 0; 3497 3498 uint32_t framesAvail = cblk->framesAvailable(); 3499 3500 3501 if (framesAvail == 0) { 3502 Mutex::Autolock _l(cblk->lock); 3503 goto start_loop_here; 3504 while (framesAvail == 0) { 3505 active = mActive; 3506 if (UNLIKELY(!active)) { 3507 LOGV("Not active and NO_MORE_BUFFERS"); 3508 return AudioTrack::NO_MORE_BUFFERS; 3509 } 3510 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3511 if (result != NO_ERROR) { 3512 return AudioTrack::NO_MORE_BUFFERS; 3513 } 3514 // read the server count again 3515 start_loop_here: 3516 framesAvail = cblk->framesAvailable_l(); 3517 } 3518 } 3519 3520// if (framesAvail < framesReq) { 3521// return AudioTrack::NO_MORE_BUFFERS; 3522// } 3523 3524 if (framesReq > framesAvail) { 3525 framesReq = framesAvail; 3526 } 3527 3528 uint32_t u = cblk->user; 3529 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3530 3531 if (u + framesReq > bufferEnd) { 3532 framesReq = bufferEnd - u; 3533 } 3534 3535 buffer->frameCount = framesReq; 3536 buffer->raw = (void *)cblk->buffer(u); 3537 return NO_ERROR; 3538} 3539 3540 3541void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3542{ 3543 size_t size = mBufferQueue.size(); 3544 Buffer *pBuffer; 3545 3546 for (size_t i = 0; i < size; i++) { 3547 pBuffer = mBufferQueue.itemAt(i); 3548 delete [] pBuffer->mBuffer; 3549 delete pBuffer; 3550 } 3551 mBufferQueue.clear(); 3552} 3553 3554// ---------------------------------------------------------------------------- 3555 3556AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3557 : RefBase(), 3558 mAudioFlinger(audioFlinger), 3559 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3560 mPid(pid) 3561{ 3562 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3563} 3564 3565// Client destructor must be called with AudioFlinger::mLock held 3566AudioFlinger::Client::~Client() 3567{ 3568 mAudioFlinger->removeClient_l(mPid); 3569} 3570 3571const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3572{ 3573 return mMemoryDealer; 3574} 3575 3576// ---------------------------------------------------------------------------- 3577 3578AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3579 const sp<IAudioFlingerClient>& client, 3580 pid_t pid) 3581 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3582{ 3583} 3584 3585AudioFlinger::NotificationClient::~NotificationClient() 3586{ 3587 mClient.clear(); 3588} 3589 3590void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3591{ 3592 sp<NotificationClient> keep(this); 3593 { 3594 mAudioFlinger->removeNotificationClient(mPid); 3595 } 3596} 3597 3598// ---------------------------------------------------------------------------- 3599 3600AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3601 : BnAudioTrack(), 3602 mTrack(track) 3603{ 3604} 3605 3606AudioFlinger::TrackHandle::~TrackHandle() { 3607 // just stop the track on deletion, associated resources 3608 // will be freed from the main thread once all pending buffers have 3609 // been played. Unless it's not in the active track list, in which 3610 // case we free everything now... 3611 mTrack->destroy(); 3612} 3613 3614status_t AudioFlinger::TrackHandle::start() { 3615 return mTrack->start(); 3616} 3617 3618void AudioFlinger::TrackHandle::stop() { 3619 mTrack->stop(); 3620} 3621 3622void AudioFlinger::TrackHandle::flush() { 3623 mTrack->flush(); 3624} 3625 3626void AudioFlinger::TrackHandle::mute(bool e) { 3627 mTrack->mute(e); 3628} 3629 3630void AudioFlinger::TrackHandle::pause() { 3631 mTrack->pause(); 3632} 3633 3634void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3635 mTrack->setVolume(left, right); 3636} 3637 3638sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3639 return mTrack->getCblk(); 3640} 3641 3642status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3643{ 3644 return mTrack->attachAuxEffect(EffectId); 3645} 3646 3647status_t AudioFlinger::TrackHandle::onTransact( 3648 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3649{ 3650 return BnAudioTrack::onTransact(code, data, reply, flags); 3651} 3652 3653// ---------------------------------------------------------------------------- 3654 3655sp<IAudioRecord> AudioFlinger::openRecord( 3656 pid_t pid, 3657 int input, 3658 uint32_t sampleRate, 3659 int format, 3660 int channelCount, 3661 int frameCount, 3662 uint32_t flags, 3663 int *sessionId, 3664 status_t *status) 3665{ 3666 sp<RecordThread::RecordTrack> recordTrack; 3667 sp<RecordHandle> recordHandle; 3668 sp<Client> client; 3669 wp<Client> wclient; 3670 status_t lStatus; 3671 RecordThread *thread; 3672 size_t inFrameCount; 3673 int lSessionId; 3674 3675 // check calling permissions 3676 if (!recordingAllowed()) { 3677 lStatus = PERMISSION_DENIED; 3678 goto Exit; 3679 } 3680 3681 // add client to list 3682 { // scope for mLock 3683 Mutex::Autolock _l(mLock); 3684 thread = checkRecordThread_l(input); 3685 if (thread == NULL) { 3686 lStatus = BAD_VALUE; 3687 goto Exit; 3688 } 3689 3690 wclient = mClients.valueFor(pid); 3691 if (wclient != NULL) { 3692 client = wclient.promote(); 3693 } else { 3694 client = new Client(this, pid); 3695 mClients.add(pid, client); 3696 } 3697 3698 // If no audio session id is provided, create one here 3699 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 3700 lSessionId = *sessionId; 3701 } else { 3702 lSessionId = nextUniqueId(); 3703 if (sessionId != NULL) { 3704 *sessionId = lSessionId; 3705 } 3706 } 3707 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3708 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3709 format, channelCount, frameCount, flags, lSessionId); 3710 } 3711 if (recordTrack->getCblk() == NULL) { 3712 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3713 // destructor is called by the TrackBase destructor with mLock held 3714 client.clear(); 3715 recordTrack.clear(); 3716 lStatus = NO_MEMORY; 3717 goto Exit; 3718 } 3719 3720 // return to handle to client 3721 recordHandle = new RecordHandle(recordTrack); 3722 lStatus = NO_ERROR; 3723 3724Exit: 3725 if (status) { 3726 *status = lStatus; 3727 } 3728 return recordHandle; 3729} 3730 3731// ---------------------------------------------------------------------------- 3732 3733AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3734 : BnAudioRecord(), 3735 mRecordTrack(recordTrack) 3736{ 3737} 3738 3739AudioFlinger::RecordHandle::~RecordHandle() { 3740 stop(); 3741} 3742 3743status_t AudioFlinger::RecordHandle::start() { 3744 LOGV("RecordHandle::start()"); 3745 return mRecordTrack->start(); 3746} 3747 3748void AudioFlinger::RecordHandle::stop() { 3749 LOGV("RecordHandle::stop()"); 3750 mRecordTrack->stop(); 3751} 3752 3753sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3754 return mRecordTrack->getCblk(); 3755} 3756 3757status_t AudioFlinger::RecordHandle::onTransact( 3758 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3759{ 3760 return BnAudioRecord::onTransact(code, data, reply, flags); 3761} 3762 3763// ---------------------------------------------------------------------------- 3764 3765AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : 3766 ThreadBase(audioFlinger, id), 3767 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3768{ 3769 mReqChannelCount = AudioSystem::popCount(channels); 3770 mReqSampleRate = sampleRate; 3771 readInputParameters(); 3772} 3773 3774 3775AudioFlinger::RecordThread::~RecordThread() 3776{ 3777 delete[] mRsmpInBuffer; 3778 if (mResampler != 0) { 3779 delete mResampler; 3780 delete[] mRsmpOutBuffer; 3781 } 3782} 3783 3784void AudioFlinger::RecordThread::onFirstRef() 3785{ 3786 const size_t SIZE = 256; 3787 char buffer[SIZE]; 3788 3789 snprintf(buffer, SIZE, "Record Thread %p", this); 3790 3791 run(buffer, PRIORITY_URGENT_AUDIO); 3792} 3793 3794bool AudioFlinger::RecordThread::threadLoop() 3795{ 3796 AudioBufferProvider::Buffer buffer; 3797 sp<RecordTrack> activeTrack; 3798 3799 nsecs_t lastWarning = 0; 3800 3801 // start recording 3802 while (!exitPending()) { 3803 3804 processConfigEvents(); 3805 3806 { // scope for mLock 3807 Mutex::Autolock _l(mLock); 3808 checkForNewParameters_l(); 3809 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3810 if (!mStandby) { 3811 mInput->standby(); 3812 mStandby = true; 3813 } 3814 3815 if (exitPending()) break; 3816 3817 LOGV("RecordThread: loop stopping"); 3818 // go to sleep 3819 mWaitWorkCV.wait(mLock); 3820 LOGV("RecordThread: loop starting"); 3821 continue; 3822 } 3823 if (mActiveTrack != 0) { 3824 if (mActiveTrack->mState == TrackBase::PAUSING) { 3825 if (!mStandby) { 3826 mInput->standby(); 3827 mStandby = true; 3828 } 3829 mActiveTrack.clear(); 3830 mStartStopCond.broadcast(); 3831 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3832 if (mReqChannelCount != mActiveTrack->channelCount()) { 3833 mActiveTrack.clear(); 3834 mStartStopCond.broadcast(); 3835 } else if (mBytesRead != 0) { 3836 // record start succeeds only if first read from audio input 3837 // succeeds 3838 if (mBytesRead > 0) { 3839 mActiveTrack->mState = TrackBase::ACTIVE; 3840 } else { 3841 mActiveTrack.clear(); 3842 } 3843 mStartStopCond.broadcast(); 3844 } 3845 mStandby = false; 3846 } 3847 } 3848 } 3849 3850 if (mActiveTrack != 0) { 3851 if (mActiveTrack->mState != TrackBase::ACTIVE && 3852 mActiveTrack->mState != TrackBase::RESUMING) { 3853 usleep(5000); 3854 continue; 3855 } 3856 buffer.frameCount = mFrameCount; 3857 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3858 size_t framesOut = buffer.frameCount; 3859 if (mResampler == 0) { 3860 // no resampling 3861 while (framesOut) { 3862 size_t framesIn = mFrameCount - mRsmpInIndex; 3863 if (framesIn) { 3864 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3865 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3866 if (framesIn > framesOut) 3867 framesIn = framesOut; 3868 mRsmpInIndex += framesIn; 3869 framesOut -= framesIn; 3870 if ((int)mChannelCount == mReqChannelCount || 3871 mFormat != AudioSystem::PCM_16_BIT) { 3872 memcpy(dst, src, framesIn * mFrameSize); 3873 } else { 3874 int16_t *src16 = (int16_t *)src; 3875 int16_t *dst16 = (int16_t *)dst; 3876 if (mChannelCount == 1) { 3877 while (framesIn--) { 3878 *dst16++ = *src16; 3879 *dst16++ = *src16++; 3880 } 3881 } else { 3882 while (framesIn--) { 3883 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3884 src16 += 2; 3885 } 3886 } 3887 } 3888 } 3889 if (framesOut && mFrameCount == mRsmpInIndex) { 3890 if (framesOut == mFrameCount && 3891 ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { 3892 mBytesRead = mInput->read(buffer.raw, mInputBytes); 3893 framesOut = 0; 3894 } else { 3895 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 3896 mRsmpInIndex = 0; 3897 } 3898 if (mBytesRead < 0) { 3899 LOGE("Error reading audio input"); 3900 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3901 // Force input into standby so that it tries to 3902 // recover at next read attempt 3903 mInput->standby(); 3904 usleep(5000); 3905 } 3906 mRsmpInIndex = mFrameCount; 3907 framesOut = 0; 3908 buffer.frameCount = 0; 3909 } 3910 } 3911 } 3912 } else { 3913 // resampling 3914 3915 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3916 // alter output frame count as if we were expecting stereo samples 3917 if (mChannelCount == 1 && mReqChannelCount == 1) { 3918 framesOut >>= 1; 3919 } 3920 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3921 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3922 // are 32 bit aligned which should be always true. 3923 if (mChannelCount == 2 && mReqChannelCount == 1) { 3924 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3925 // the resampler always outputs stereo samples: do post stereo to mono conversion 3926 int16_t *src = (int16_t *)mRsmpOutBuffer; 3927 int16_t *dst = buffer.i16; 3928 while (framesOut--) { 3929 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3930 src += 2; 3931 } 3932 } else { 3933 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3934 } 3935 3936 } 3937 mActiveTrack->releaseBuffer(&buffer); 3938 mActiveTrack->overflow(); 3939 } 3940 // client isn't retrieving buffers fast enough 3941 else { 3942 if (!mActiveTrack->setOverflow()) { 3943 nsecs_t now = systemTime(); 3944 if ((now - lastWarning) > kWarningThrottle) { 3945 LOGW("RecordThread: buffer overflow"); 3946 lastWarning = now; 3947 } 3948 } 3949 // Release the processor for a while before asking for a new buffer. 3950 // This will give the application more chance to read from the buffer and 3951 // clear the overflow. 3952 usleep(5000); 3953 } 3954 } 3955 } 3956 3957 if (!mStandby) { 3958 mInput->standby(); 3959 } 3960 mActiveTrack.clear(); 3961 3962 mStartStopCond.broadcast(); 3963 3964 LOGV("RecordThread %p exiting", this); 3965 return false; 3966} 3967 3968status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3969{ 3970 LOGV("RecordThread::start"); 3971 sp <ThreadBase> strongMe = this; 3972 status_t status = NO_ERROR; 3973 { 3974 AutoMutex lock(&mLock); 3975 if (mActiveTrack != 0) { 3976 if (recordTrack != mActiveTrack.get()) { 3977 status = -EBUSY; 3978 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3979 mActiveTrack->mState = TrackBase::ACTIVE; 3980 } 3981 return status; 3982 } 3983 3984 recordTrack->mState = TrackBase::IDLE; 3985 mActiveTrack = recordTrack; 3986 mLock.unlock(); 3987 status_t status = AudioSystem::startInput(mId); 3988 mLock.lock(); 3989 if (status != NO_ERROR) { 3990 mActiveTrack.clear(); 3991 return status; 3992 } 3993 mActiveTrack->mState = TrackBase::RESUMING; 3994 mRsmpInIndex = mFrameCount; 3995 mBytesRead = 0; 3996 // signal thread to start 3997 LOGV("Signal record thread"); 3998 mWaitWorkCV.signal(); 3999 // do not wait for mStartStopCond if exiting 4000 if (mExiting) { 4001 mActiveTrack.clear(); 4002 status = INVALID_OPERATION; 4003 goto startError; 4004 } 4005 mStartStopCond.wait(mLock); 4006 if (mActiveTrack == 0) { 4007 LOGV("Record failed to start"); 4008 status = BAD_VALUE; 4009 goto startError; 4010 } 4011 LOGV("Record started OK"); 4012 return status; 4013 } 4014startError: 4015 AudioSystem::stopInput(mId); 4016 return status; 4017} 4018 4019void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4020 LOGV("RecordThread::stop"); 4021 sp <ThreadBase> strongMe = this; 4022 { 4023 AutoMutex lock(&mLock); 4024 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4025 mActiveTrack->mState = TrackBase::PAUSING; 4026 // do not wait for mStartStopCond if exiting 4027 if (mExiting) { 4028 return; 4029 } 4030 mStartStopCond.wait(mLock); 4031 // if we have been restarted, recordTrack == mActiveTrack.get() here 4032 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4033 mLock.unlock(); 4034 AudioSystem::stopInput(mId); 4035 mLock.lock(); 4036 LOGV("Record stopped OK"); 4037 } 4038 } 4039 } 4040} 4041 4042status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4043{ 4044 const size_t SIZE = 256; 4045 char buffer[SIZE]; 4046 String8 result; 4047 pid_t pid = 0; 4048 4049 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4050 result.append(buffer); 4051 4052 if (mActiveTrack != 0) { 4053 result.append("Active Track:\n"); 4054 result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); 4055 mActiveTrack->dump(buffer, SIZE); 4056 result.append(buffer); 4057 4058 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4059 result.append(buffer); 4060 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4061 result.append(buffer); 4062 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4063 result.append(buffer); 4064 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4065 result.append(buffer); 4066 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4067 result.append(buffer); 4068 4069 4070 } else { 4071 result.append("No record client\n"); 4072 } 4073 write(fd, result.string(), result.size()); 4074 4075 dumpBase(fd, args); 4076 4077 return NO_ERROR; 4078} 4079 4080status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4081{ 4082 size_t framesReq = buffer->frameCount; 4083 size_t framesReady = mFrameCount - mRsmpInIndex; 4084 int channelCount; 4085 4086 if (framesReady == 0) { 4087 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 4088 if (mBytesRead < 0) { 4089 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4090 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4091 // Force input into standby so that it tries to 4092 // recover at next read attempt 4093 mInput->standby(); 4094 usleep(5000); 4095 } 4096 buffer->raw = 0; 4097 buffer->frameCount = 0; 4098 return NOT_ENOUGH_DATA; 4099 } 4100 mRsmpInIndex = 0; 4101 framesReady = mFrameCount; 4102 } 4103 4104 if (framesReq > framesReady) { 4105 framesReq = framesReady; 4106 } 4107 4108 if (mChannelCount == 1 && mReqChannelCount == 2) { 4109 channelCount = 1; 4110 } else { 4111 channelCount = 2; 4112 } 4113 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4114 buffer->frameCount = framesReq; 4115 return NO_ERROR; 4116} 4117 4118void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4119{ 4120 mRsmpInIndex += buffer->frameCount; 4121 buffer->frameCount = 0; 4122} 4123 4124bool AudioFlinger::RecordThread::checkForNewParameters_l() 4125{ 4126 bool reconfig = false; 4127 4128 while (!mNewParameters.isEmpty()) { 4129 status_t status = NO_ERROR; 4130 String8 keyValuePair = mNewParameters[0]; 4131 AudioParameter param = AudioParameter(keyValuePair); 4132 int value; 4133 int reqFormat = mFormat; 4134 int reqSamplingRate = mReqSampleRate; 4135 int reqChannelCount = mReqChannelCount; 4136 4137 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4138 reqSamplingRate = value; 4139 reconfig = true; 4140 } 4141 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4142 reqFormat = value; 4143 reconfig = true; 4144 } 4145 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4146 reqChannelCount = AudioSystem::popCount(value); 4147 reconfig = true; 4148 } 4149 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4150 // do not accept frame count changes if tracks are open as the track buffer 4151 // size depends on frame count and correct behavior would not be garantied 4152 // if frame count is changed after track creation 4153 if (mActiveTrack != 0) { 4154 status = INVALID_OPERATION; 4155 } else { 4156 reconfig = true; 4157 } 4158 } 4159 if (status == NO_ERROR) { 4160 status = mInput->setParameters(keyValuePair); 4161 if (status == INVALID_OPERATION) { 4162 mInput->standby(); 4163 status = mInput->setParameters(keyValuePair); 4164 } 4165 if (reconfig) { 4166 if (status == BAD_VALUE && 4167 reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && 4168 ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && 4169 (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { 4170 status = NO_ERROR; 4171 } 4172 if (status == NO_ERROR) { 4173 readInputParameters(); 4174 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4175 } 4176 } 4177 } 4178 4179 mNewParameters.removeAt(0); 4180 4181 mParamStatus = status; 4182 mParamCond.signal(); 4183 mWaitWorkCV.wait(mLock); 4184 } 4185 return reconfig; 4186} 4187 4188String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4189{ 4190 return mInput->getParameters(keys); 4191} 4192 4193void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4194 AudioSystem::OutputDescriptor desc; 4195 void *param2 = 0; 4196 4197 switch (event) { 4198 case AudioSystem::INPUT_OPENED: 4199 case AudioSystem::INPUT_CONFIG_CHANGED: 4200 desc.channels = mChannels; 4201 desc.samplingRate = mSampleRate; 4202 desc.format = mFormat; 4203 desc.frameCount = mFrameCount; 4204 desc.latency = 0; 4205 param2 = &desc; 4206 break; 4207 4208 case AudioSystem::INPUT_CLOSED: 4209 default: 4210 break; 4211 } 4212 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4213} 4214 4215void AudioFlinger::RecordThread::readInputParameters() 4216{ 4217 if (mRsmpInBuffer) delete mRsmpInBuffer; 4218 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4219 if (mResampler) delete mResampler; 4220 mResampler = 0; 4221 4222 mSampleRate = mInput->sampleRate(); 4223 mChannels = mInput->channels(); 4224 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 4225 mFormat = mInput->format(); 4226 mFrameSize = (uint16_t)mInput->frameSize(); 4227 mInputBytes = mInput->bufferSize(); 4228 mFrameCount = mInputBytes / mFrameSize; 4229 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4230 4231 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4232 { 4233 int channelCount; 4234 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4235 // stereo to mono post process as the resampler always outputs stereo. 4236 if (mChannelCount == 1 && mReqChannelCount == 2) { 4237 channelCount = 1; 4238 } else { 4239 channelCount = 2; 4240 } 4241 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4242 mResampler->setSampleRate(mSampleRate); 4243 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4244 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4245 4246 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4247 if (mChannelCount == 1 && mReqChannelCount == 1) { 4248 mFrameCount >>= 1; 4249 } 4250 4251 } 4252 mRsmpInIndex = mFrameCount; 4253} 4254 4255unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4256{ 4257 return mInput->getInputFramesLost(); 4258} 4259 4260// ---------------------------------------------------------------------------- 4261 4262int AudioFlinger::openOutput(uint32_t *pDevices, 4263 uint32_t *pSamplingRate, 4264 uint32_t *pFormat, 4265 uint32_t *pChannels, 4266 uint32_t *pLatencyMs, 4267 uint32_t flags) 4268{ 4269 status_t status; 4270 PlaybackThread *thread = NULL; 4271 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4272 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4273 uint32_t format = pFormat ? *pFormat : 0; 4274 uint32_t channels = pChannels ? *pChannels : 0; 4275 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4276 4277 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4278 pDevices ? *pDevices : 0, 4279 samplingRate, 4280 format, 4281 channels, 4282 flags); 4283 4284 if (pDevices == NULL || *pDevices == 0) { 4285 return 0; 4286 } 4287 Mutex::Autolock _l(mLock); 4288 4289 AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, 4290 (int *)&format, 4291 &channels, 4292 &samplingRate, 4293 &status); 4294 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4295 output, 4296 samplingRate, 4297 format, 4298 channels, 4299 status); 4300 4301 mHardwareStatus = AUDIO_HW_IDLE; 4302 if (output != 0) { 4303 int id = nextUniqueId(); 4304 if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || 4305 (format != AudioSystem::PCM_16_BIT) || 4306 (channels != AudioSystem::CHANNEL_OUT_STEREO)) { 4307 thread = new DirectOutputThread(this, output, id, *pDevices); 4308 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4309 } else { 4310 thread = new MixerThread(this, output, id, *pDevices); 4311 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4312 4313#ifdef LVMX 4314 unsigned bitsPerSample = 4315 (format == AudioSystem::PCM_16_BIT) ? 16 : 4316 ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); 4317 unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; 4318 int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); 4319 4320 LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); 4321 LifeVibes::setDevice(audioOutputType, *pDevices); 4322#endif 4323 4324 } 4325 mPlaybackThreads.add(id, thread); 4326 4327 if (pSamplingRate) *pSamplingRate = samplingRate; 4328 if (pFormat) *pFormat = format; 4329 if (pChannels) *pChannels = channels; 4330 if (pLatencyMs) *pLatencyMs = thread->latency(); 4331 4332 // notify client processes of the new output creation 4333 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4334 return id; 4335 } 4336 4337 return 0; 4338} 4339 4340int AudioFlinger::openDuplicateOutput(int output1, int output2) 4341{ 4342 Mutex::Autolock _l(mLock); 4343 MixerThread *thread1 = checkMixerThread_l(output1); 4344 MixerThread *thread2 = checkMixerThread_l(output2); 4345 4346 if (thread1 == NULL || thread2 == NULL) { 4347 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4348 return 0; 4349 } 4350 4351 int id = nextUniqueId(); 4352 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4353 thread->addOutputTrack(thread2); 4354 mPlaybackThreads.add(id, thread); 4355 // notify client processes of the new output creation 4356 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4357 return id; 4358} 4359 4360status_t AudioFlinger::closeOutput(int output) 4361{ 4362 // keep strong reference on the playback thread so that 4363 // it is not destroyed while exit() is executed 4364 sp <PlaybackThread> thread; 4365 { 4366 Mutex::Autolock _l(mLock); 4367 thread = checkPlaybackThread_l(output); 4368 if (thread == NULL) { 4369 return BAD_VALUE; 4370 } 4371 4372 LOGV("closeOutput() %d", output); 4373 4374 if (thread->type() == PlaybackThread::MIXER) { 4375 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4376 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4377 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4378 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4379 } 4380 } 4381 } 4382 void *param2 = 0; 4383 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4384 mPlaybackThreads.removeItem(output); 4385 } 4386 thread->exit(); 4387 4388 if (thread->type() != PlaybackThread::DUPLICATING) { 4389 mAudioHardware->closeOutputStream(thread->getOutput()); 4390 } 4391 return NO_ERROR; 4392} 4393 4394status_t AudioFlinger::suspendOutput(int output) 4395{ 4396 Mutex::Autolock _l(mLock); 4397 PlaybackThread *thread = checkPlaybackThread_l(output); 4398 4399 if (thread == NULL) { 4400 return BAD_VALUE; 4401 } 4402 4403 LOGV("suspendOutput() %d", output); 4404 thread->suspend(); 4405 4406 return NO_ERROR; 4407} 4408 4409status_t AudioFlinger::restoreOutput(int output) 4410{ 4411 Mutex::Autolock _l(mLock); 4412 PlaybackThread *thread = checkPlaybackThread_l(output); 4413 4414 if (thread == NULL) { 4415 return BAD_VALUE; 4416 } 4417 4418 LOGV("restoreOutput() %d", output); 4419 4420 thread->restore(); 4421 4422 return NO_ERROR; 4423} 4424 4425int AudioFlinger::openInput(uint32_t *pDevices, 4426 uint32_t *pSamplingRate, 4427 uint32_t *pFormat, 4428 uint32_t *pChannels, 4429 uint32_t acoustics) 4430{ 4431 status_t status; 4432 RecordThread *thread = NULL; 4433 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4434 uint32_t format = pFormat ? *pFormat : 0; 4435 uint32_t channels = pChannels ? *pChannels : 0; 4436 uint32_t reqSamplingRate = samplingRate; 4437 uint32_t reqFormat = format; 4438 uint32_t reqChannels = channels; 4439 4440 if (pDevices == NULL || *pDevices == 0) { 4441 return 0; 4442 } 4443 Mutex::Autolock _l(mLock); 4444 4445 AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, 4446 (int *)&format, 4447 &channels, 4448 &samplingRate, 4449 &status, 4450 (AudioSystem::audio_in_acoustics)acoustics); 4451 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4452 input, 4453 samplingRate, 4454 format, 4455 channels, 4456 acoustics, 4457 status); 4458 4459 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4460 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4461 // or stereo to mono conversions on 16 bit PCM inputs. 4462 if (input == 0 && status == BAD_VALUE && 4463 reqFormat == format && format == AudioSystem::PCM_16_BIT && 4464 (samplingRate <= 2 * reqSamplingRate) && 4465 (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { 4466 LOGV("openInput() reopening with proposed sampling rate and channels"); 4467 input = mAudioHardware->openInputStream(*pDevices, 4468 (int *)&format, 4469 &channels, 4470 &samplingRate, 4471 &status, 4472 (AudioSystem::audio_in_acoustics)acoustics); 4473 } 4474 4475 if (input != 0) { 4476 int id = nextUniqueId(); 4477 // Start record thread 4478 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4479 mRecordThreads.add(id, thread); 4480 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4481 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4482 if (pFormat) *pFormat = format; 4483 if (pChannels) *pChannels = reqChannels; 4484 4485 input->standby(); 4486 4487 // notify client processes of the new input creation 4488 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4489 return id; 4490 } 4491 4492 return 0; 4493} 4494 4495status_t AudioFlinger::closeInput(int input) 4496{ 4497 // keep strong reference on the record thread so that 4498 // it is not destroyed while exit() is executed 4499 sp <RecordThread> thread; 4500 { 4501 Mutex::Autolock _l(mLock); 4502 thread = checkRecordThread_l(input); 4503 if (thread == NULL) { 4504 return BAD_VALUE; 4505 } 4506 4507 LOGV("closeInput() %d", input); 4508 void *param2 = 0; 4509 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4510 mRecordThreads.removeItem(input); 4511 } 4512 thread->exit(); 4513 4514 mAudioHardware->closeInputStream(thread->getInput()); 4515 4516 return NO_ERROR; 4517} 4518 4519status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4520{ 4521 Mutex::Autolock _l(mLock); 4522 MixerThread *dstThread = checkMixerThread_l(output); 4523 if (dstThread == NULL) { 4524 LOGW("setStreamOutput() bad output id %d", output); 4525 return BAD_VALUE; 4526 } 4527 4528 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4529 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4530 4531 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4532 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4533 if (thread != dstThread && 4534 thread->type() != PlaybackThread::DIRECT) { 4535 MixerThread *srcThread = (MixerThread *)thread; 4536 srcThread->invalidateTracks(stream); 4537 } 4538 } 4539 4540 return NO_ERROR; 4541} 4542 4543 4544int AudioFlinger::newAudioSessionId() 4545{ 4546 return nextUniqueId(); 4547} 4548 4549// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4550AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4551{ 4552 PlaybackThread *thread = NULL; 4553 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4554 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4555 } 4556 return thread; 4557} 4558 4559// checkMixerThread_l() must be called with AudioFlinger::mLock held 4560AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4561{ 4562 PlaybackThread *thread = checkPlaybackThread_l(output); 4563 if (thread != NULL) { 4564 if (thread->type() == PlaybackThread::DIRECT) { 4565 thread = NULL; 4566 } 4567 } 4568 return (MixerThread *)thread; 4569} 4570 4571// checkRecordThread_l() must be called with AudioFlinger::mLock held 4572AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4573{ 4574 RecordThread *thread = NULL; 4575 if (mRecordThreads.indexOfKey(input) >= 0) { 4576 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4577 } 4578 return thread; 4579} 4580 4581int AudioFlinger::nextUniqueId() 4582{ 4583 return android_atomic_inc(&mNextUniqueId); 4584} 4585 4586// ---------------------------------------------------------------------------- 4587// Effect management 4588// ---------------------------------------------------------------------------- 4589 4590 4591status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) 4592{ 4593 // check calling permissions 4594 if (!settingsAllowed()) { 4595 return PERMISSION_DENIED; 4596 } 4597 // only allow libraries loaded from /system/lib/soundfx for now 4598 if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { 4599 return PERMISSION_DENIED; 4600 } 4601 4602 Mutex::Autolock _l(mLock); 4603 return EffectLoadLibrary(libPath, handle); 4604} 4605 4606status_t AudioFlinger::unloadEffectLibrary(int handle) 4607{ 4608 // check calling permissions 4609 if (!settingsAllowed()) { 4610 return PERMISSION_DENIED; 4611 } 4612 4613 Mutex::Autolock _l(mLock); 4614 return EffectUnloadLibrary(handle); 4615} 4616 4617status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4618{ 4619 Mutex::Autolock _l(mLock); 4620 return EffectQueryNumberEffects(numEffects); 4621} 4622 4623status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4624{ 4625 Mutex::Autolock _l(mLock); 4626 return EffectQueryEffect(index, descriptor); 4627} 4628 4629status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4630{ 4631 Mutex::Autolock _l(mLock); 4632 return EffectGetDescriptor(pUuid, descriptor); 4633} 4634 4635 4636// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4637static const effect_uuid_t VISUALIZATION_UUID_ = 4638 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4639 4640sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4641 effect_descriptor_t *pDesc, 4642 const sp<IEffectClient>& effectClient, 4643 int32_t priority, 4644 int output, 4645 int sessionId, 4646 status_t *status, 4647 int *id, 4648 int *enabled) 4649{ 4650 status_t lStatus = NO_ERROR; 4651 sp<EffectHandle> handle; 4652 effect_interface_t itfe; 4653 effect_descriptor_t desc; 4654 sp<Client> client; 4655 wp<Client> wclient; 4656 4657 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4658 pid, effectClient.get(), priority, sessionId, output); 4659 4660 if (pDesc == NULL) { 4661 lStatus = BAD_VALUE; 4662 goto Exit; 4663 } 4664 4665 // check audio settings permission for global effects 4666 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && !settingsAllowed()) { 4667 lStatus = PERMISSION_DENIED; 4668 goto Exit; 4669 } 4670 4671 // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects 4672 // that can only be created by audio policy manager (running in same process) 4673 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && getpid() != pid) { 4674 lStatus = PERMISSION_DENIED; 4675 goto Exit; 4676 } 4677 4678 // check recording permission for visualizer 4679 if ((memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4680 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) && 4681 !recordingAllowed()) { 4682 lStatus = PERMISSION_DENIED; 4683 goto Exit; 4684 } 4685 4686 if (output == 0) { 4687 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) { 4688 // output must be specified by AudioPolicyManager when using session 4689 // AudioSystem::SESSION_OUTPUT_STAGE 4690 lStatus = BAD_VALUE; 4691 goto Exit; 4692 } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 4693 // if the output returned by getOutputForEffect() is removed before we lock the 4694 // mutex below, the call to checkPlaybackThread_l(output) below will detect it 4695 // and we will exit safely 4696 output = AudioSystem::getOutputForEffect(&desc); 4697 } 4698 } 4699 4700 { 4701 Mutex::Autolock _l(mLock); 4702 4703 4704 if (!EffectIsNullUuid(&pDesc->uuid)) { 4705 // if uuid is specified, request effect descriptor 4706 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4707 if (lStatus < 0) { 4708 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4709 goto Exit; 4710 } 4711 } else { 4712 // if uuid is not specified, look for an available implementation 4713 // of the required type in effect factory 4714 if (EffectIsNullUuid(&pDesc->type)) { 4715 LOGW("createEffect() no effect type"); 4716 lStatus = BAD_VALUE; 4717 goto Exit; 4718 } 4719 uint32_t numEffects = 0; 4720 effect_descriptor_t d; 4721 bool found = false; 4722 4723 lStatus = EffectQueryNumberEffects(&numEffects); 4724 if (lStatus < 0) { 4725 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4726 goto Exit; 4727 } 4728 for (uint32_t i = 0; i < numEffects; i++) { 4729 lStatus = EffectQueryEffect(i, &desc); 4730 if (lStatus < 0) { 4731 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4732 continue; 4733 } 4734 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4735 // If matching type found save effect descriptor. If the session is 4736 // 0 and the effect is not auxiliary, continue enumeration in case 4737 // an auxiliary version of this effect type is available 4738 found = true; 4739 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4740 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX || 4741 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4742 break; 4743 } 4744 } 4745 } 4746 if (!found) { 4747 lStatus = BAD_VALUE; 4748 LOGW("createEffect() effect not found"); 4749 goto Exit; 4750 } 4751 // For same effect type, chose auxiliary version over insert version if 4752 // connect to output mix (Compliance to OpenSL ES) 4753 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && 4754 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4755 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4756 } 4757 } 4758 4759 // Do not allow auxiliary effects on a session different from 0 (output mix) 4760 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX && 4761 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4762 lStatus = INVALID_OPERATION; 4763 goto Exit; 4764 } 4765 4766 // return effect descriptor 4767 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4768 4769 // If output is not specified try to find a matching audio session ID in one of the 4770 // output threads. 4771 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 4772 // because of code checking output when entering the function. 4773 if (output == 0) { 4774 // look for the thread where the specified audio session is present 4775 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4776 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4777 output = mPlaybackThreads.keyAt(i); 4778 break; 4779 } 4780 } 4781 // If no output thread contains the requested session ID, default to 4782 // first output. The effect chain will be moved to the correct output 4783 // thread when a track with the same session ID is created 4784 if (output == 0 && mPlaybackThreads.size()) { 4785 output = mPlaybackThreads.keyAt(0); 4786 } 4787 } 4788 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4789 PlaybackThread *thread = checkPlaybackThread_l(output); 4790 if (thread == NULL) { 4791 LOGE("createEffect() unknown output thread"); 4792 lStatus = BAD_VALUE; 4793 goto Exit; 4794 } 4795 4796 // TODO: allow attachment of effect to inputs 4797 4798 wclient = mClients.valueFor(pid); 4799 4800 if (wclient != NULL) { 4801 client = wclient.promote(); 4802 } else { 4803 client = new Client(this, pid); 4804 mClients.add(pid, client); 4805 } 4806 4807 // create effect on selected output trhead 4808 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4809 &desc, enabled, &lStatus); 4810 if (handle != 0 && id != NULL) { 4811 *id = handle->id(); 4812 } 4813 } 4814 4815Exit: 4816 if(status) { 4817 *status = lStatus; 4818 } 4819 return handle; 4820} 4821 4822status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4823{ 4824 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4825 session, srcOutput, dstOutput); 4826 Mutex::Autolock _l(mLock); 4827 if (srcOutput == dstOutput) { 4828 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4829 return NO_ERROR; 4830 } 4831 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4832 if (srcThread == NULL) { 4833 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4834 return BAD_VALUE; 4835 } 4836 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4837 if (dstThread == NULL) { 4838 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4839 return BAD_VALUE; 4840 } 4841 4842 Mutex::Autolock _dl(dstThread->mLock); 4843 Mutex::Autolock _sl(srcThread->mLock); 4844 moveEffectChain_l(session, srcThread, dstThread, false); 4845 4846 return NO_ERROR; 4847} 4848 4849// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4850status_t AudioFlinger::moveEffectChain_l(int session, 4851 AudioFlinger::PlaybackThread *srcThread, 4852 AudioFlinger::PlaybackThread *dstThread, 4853 bool reRegister) 4854{ 4855 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4856 session, srcThread, dstThread); 4857 4858 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4859 if (chain == 0) { 4860 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4861 session, srcThread); 4862 return INVALID_OPERATION; 4863 } 4864 4865 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 4866 // so that a new chain is created with correct parameters when first effect is added. This is 4867 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4868 // removed. 4869 srcThread->removeEffectChain_l(chain); 4870 4871 // transfer all effects one by one so that new effect chain is created on new thread with 4872 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4873 int dstOutput = dstThread->id(); 4874 sp<EffectChain> dstChain; 4875 uint32_t strategy; 4876 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4877 while (effect != 0) { 4878 srcThread->removeEffect_l(effect); 4879 dstThread->addEffect_l(effect); 4880 // if the move request is not received from audio policy manager, the effect must be 4881 // re-registered with the new strategy and output 4882 if (dstChain == 0) { 4883 dstChain = effect->chain().promote(); 4884 if (dstChain == 0) { 4885 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 4886 srcThread->addEffect_l(effect); 4887 return NO_INIT; 4888 } 4889 strategy = dstChain->strategy(); 4890 } 4891 if (reRegister) { 4892 AudioSystem::unregisterEffect(effect->id()); 4893 AudioSystem::registerEffect(&effect->desc(), 4894 dstOutput, 4895 strategy, 4896 session, 4897 effect->id()); 4898 } 4899 effect = chain->getEffectFromId_l(0); 4900 } 4901 4902 return NO_ERROR; 4903} 4904 4905// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4906sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4907 const sp<AudioFlinger::Client>& client, 4908 const sp<IEffectClient>& effectClient, 4909 int32_t priority, 4910 int sessionId, 4911 effect_descriptor_t *desc, 4912 int *enabled, 4913 status_t *status 4914 ) 4915{ 4916 sp<EffectModule> effect; 4917 sp<EffectHandle> handle; 4918 status_t lStatus; 4919 sp<Track> track; 4920 sp<EffectChain> chain; 4921 bool chainCreated = false; 4922 bool effectCreated = false; 4923 bool effectRegistered = false; 4924 4925 if (mOutput == 0) { 4926 LOGW("createEffect_l() Audio driver not initialized."); 4927 lStatus = NO_INIT; 4928 goto Exit; 4929 } 4930 4931 // Do not allow auxiliary effect on session other than 0 4932 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4933 sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 4934 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4935 desc->name, sessionId); 4936 lStatus = BAD_VALUE; 4937 goto Exit; 4938 } 4939 4940 // Do not allow effects with session ID 0 on direct output or duplicating threads 4941 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4942 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) { 4943 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4944 desc->name, sessionId); 4945 lStatus = BAD_VALUE; 4946 goto Exit; 4947 } 4948 4949 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4950 4951 { // scope for mLock 4952 Mutex::Autolock _l(mLock); 4953 4954 // check for existing effect chain with the requested audio session 4955 chain = getEffectChain_l(sessionId); 4956 if (chain == 0) { 4957 // create a new chain for this session 4958 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4959 chain = new EffectChain(this, sessionId); 4960 addEffectChain_l(chain); 4961 chain->setStrategy(getStrategyForSession_l(sessionId)); 4962 chainCreated = true; 4963 } else { 4964 effect = chain->getEffectFromDesc_l(desc); 4965 } 4966 4967 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4968 4969 if (effect == 0) { 4970 int id = mAudioFlinger->nextUniqueId(); 4971 // Check CPU and memory usage 4972 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4973 if (lStatus != NO_ERROR) { 4974 goto Exit; 4975 } 4976 effectRegistered = true; 4977 // create a new effect module if none present in the chain 4978 effect = new EffectModule(this, chain, desc, id, sessionId); 4979 lStatus = effect->status(); 4980 if (lStatus != NO_ERROR) { 4981 goto Exit; 4982 } 4983 lStatus = chain->addEffect_l(effect); 4984 if (lStatus != NO_ERROR) { 4985 goto Exit; 4986 } 4987 effectCreated = true; 4988 4989 effect->setDevice(mDevice); 4990 effect->setMode(mAudioFlinger->getMode()); 4991 } 4992 // create effect handle and connect it to effect module 4993 handle = new EffectHandle(effect, client, effectClient, priority); 4994 lStatus = effect->addHandle(handle); 4995 if (enabled) { 4996 *enabled = (int)effect->isEnabled(); 4997 } 4998 } 4999 5000Exit: 5001 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5002 Mutex::Autolock _l(mLock); 5003 if (effectCreated) { 5004 chain->removeEffect_l(effect); 5005 } 5006 if (effectRegistered) { 5007 AudioSystem::unregisterEffect(effect->id()); 5008 } 5009 if (chainCreated) { 5010 removeEffectChain_l(chain); 5011 } 5012 handle.clear(); 5013 } 5014 5015 if(status) { 5016 *status = lStatus; 5017 } 5018 return handle; 5019} 5020 5021// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5022// PlaybackThread::mLock held 5023status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 5024{ 5025 // check for existing effect chain with the requested audio session 5026 int sessionId = effect->sessionId(); 5027 sp<EffectChain> chain = getEffectChain_l(sessionId); 5028 bool chainCreated = false; 5029 5030 if (chain == 0) { 5031 // create a new chain for this session 5032 LOGV("addEffect_l() new effect chain for session %d", sessionId); 5033 chain = new EffectChain(this, sessionId); 5034 addEffectChain_l(chain); 5035 chain->setStrategy(getStrategyForSession_l(sessionId)); 5036 chainCreated = true; 5037 } 5038 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5039 5040 if (chain->getEffectFromId_l(effect->id()) != 0) { 5041 LOGW("addEffect_l() %p effect %s already present in chain %p", 5042 this, effect->desc().name, chain.get()); 5043 return BAD_VALUE; 5044 } 5045 5046 status_t status = chain->addEffect_l(effect); 5047 if (status != NO_ERROR) { 5048 if (chainCreated) { 5049 removeEffectChain_l(chain); 5050 } 5051 return status; 5052 } 5053 5054 effect->setDevice(mDevice); 5055 effect->setMode(mAudioFlinger->getMode()); 5056 return NO_ERROR; 5057} 5058 5059void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 5060 5061 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5062 effect_descriptor_t desc = effect->desc(); 5063 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5064 detachAuxEffect_l(effect->id()); 5065 } 5066 5067 sp<EffectChain> chain = effect->chain().promote(); 5068 if (chain != 0) { 5069 // remove effect chain if removing last effect 5070 if (chain->removeEffect_l(effect) == 0) { 5071 removeEffectChain_l(chain); 5072 } 5073 } else { 5074 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5075 } 5076} 5077 5078void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 5079 const wp<EffectHandle>& handle) { 5080 Mutex::Autolock _l(mLock); 5081 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5082 // delete the effect module if removing last handle on it 5083 if (effect->removeHandle(handle) == 0) { 5084 removeEffect_l(effect); 5085 AudioSystem::unregisterEffect(effect->id()); 5086 } 5087} 5088 5089status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5090{ 5091 int session = chain->sessionId(); 5092 int16_t *buffer = mMixBuffer; 5093 bool ownsBuffer = false; 5094 5095 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5096 if (session > 0) { 5097 // Only one effect chain can be present in direct output thread and it uses 5098 // the mix buffer as input 5099 if (mType != DIRECT) { 5100 size_t numSamples = mFrameCount * mChannelCount; 5101 buffer = new int16_t[numSamples]; 5102 memset(buffer, 0, numSamples * sizeof(int16_t)); 5103 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5104 ownsBuffer = true; 5105 } 5106 5107 // Attach all tracks with same session ID to this chain. 5108 for (size_t i = 0; i < mTracks.size(); ++i) { 5109 sp<Track> track = mTracks[i]; 5110 if (session == track->sessionId()) { 5111 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5112 track->setMainBuffer(buffer); 5113 } 5114 } 5115 5116 // indicate all active tracks in the chain 5117 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5118 sp<Track> track = mActiveTracks[i].promote(); 5119 if (track == 0) continue; 5120 if (session == track->sessionId()) { 5121 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5122 chain->startTrack(); 5123 } 5124 } 5125 } 5126 5127 chain->setInBuffer(buffer, ownsBuffer); 5128 chain->setOutBuffer(mMixBuffer); 5129 // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect 5130 // chains list in order to be processed last as it contains output stage effects 5131 // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before 5132 // session AudioSystem::SESSION_OUTPUT_STAGE to be processed 5133 // after track specific effects and before output stage 5134 // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and 5135 // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX 5136 // Effect chain for other sessions are inserted at beginning of effect 5137 // chains list to be processed before output mix effects. Relative order between other 5138 // sessions is not important 5139 size_t size = mEffectChains.size(); 5140 size_t i = 0; 5141 for (i = 0; i < size; i++) { 5142 if (mEffectChains[i]->sessionId() < session) break; 5143 } 5144 mEffectChains.insertAt(chain, i); 5145 5146 return NO_ERROR; 5147} 5148 5149size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5150{ 5151 int session = chain->sessionId(); 5152 5153 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5154 5155 for (size_t i = 0; i < mEffectChains.size(); i++) { 5156 if (chain == mEffectChains[i]) { 5157 mEffectChains.removeAt(i); 5158 // detach all tracks with same session ID from this chain 5159 for (size_t i = 0; i < mTracks.size(); ++i) { 5160 sp<Track> track = mTracks[i]; 5161 if (session == track->sessionId()) { 5162 track->setMainBuffer(mMixBuffer); 5163 } 5164 } 5165 break; 5166 } 5167 } 5168 return mEffectChains.size(); 5169} 5170 5171void AudioFlinger::PlaybackThread::lockEffectChains_l( 5172 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5173{ 5174 effectChains = mEffectChains; 5175 for (size_t i = 0; i < mEffectChains.size(); i++) { 5176 mEffectChains[i]->lock(); 5177 } 5178} 5179 5180void AudioFlinger::PlaybackThread::unlockEffectChains( 5181 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5182{ 5183 for (size_t i = 0; i < effectChains.size(); i++) { 5184 effectChains[i]->unlock(); 5185 } 5186} 5187 5188 5189sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5190{ 5191 sp<EffectModule> effect; 5192 5193 sp<EffectChain> chain = getEffectChain_l(sessionId); 5194 if (chain != 0) { 5195 effect = chain->getEffectFromId_l(effectId); 5196 } 5197 return effect; 5198} 5199 5200status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5201 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5202{ 5203 Mutex::Autolock _l(mLock); 5204 return attachAuxEffect_l(track, EffectId); 5205} 5206 5207status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5208 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5209{ 5210 status_t status = NO_ERROR; 5211 5212 if (EffectId == 0) { 5213 track->setAuxBuffer(0, NULL); 5214 } else { 5215 // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX 5216 sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId); 5217 if (effect != 0) { 5218 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5219 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5220 } else { 5221 status = INVALID_OPERATION; 5222 } 5223 } else { 5224 status = BAD_VALUE; 5225 } 5226 } 5227 return status; 5228} 5229 5230void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5231{ 5232 for (size_t i = 0; i < mTracks.size(); ++i) { 5233 sp<Track> track = mTracks[i]; 5234 if (track->auxEffectId() == effectId) { 5235 attachAuxEffect_l(track, 0); 5236 } 5237 } 5238} 5239 5240// ---------------------------------------------------------------------------- 5241// EffectModule implementation 5242// ---------------------------------------------------------------------------- 5243 5244#undef LOG_TAG 5245#define LOG_TAG "AudioFlinger::EffectModule" 5246 5247AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5248 const wp<AudioFlinger::EffectChain>& chain, 5249 effect_descriptor_t *desc, 5250 int id, 5251 int sessionId) 5252 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5253 mStatus(NO_INIT), mState(IDLE) 5254{ 5255 LOGV("Constructor %p", this); 5256 int lStatus; 5257 sp<ThreadBase> thread = mThread.promote(); 5258 if (thread == 0) { 5259 return; 5260 } 5261 PlaybackThread *p = (PlaybackThread *)thread.get(); 5262 5263 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5264 5265 // create effect engine from effect factory 5266 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5267 5268 if (mStatus != NO_ERROR) { 5269 return; 5270 } 5271 lStatus = init(); 5272 if (lStatus < 0) { 5273 mStatus = lStatus; 5274 goto Error; 5275 } 5276 5277 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5278 return; 5279Error: 5280 EffectRelease(mEffectInterface); 5281 mEffectInterface = NULL; 5282 LOGV("Constructor Error %d", mStatus); 5283} 5284 5285AudioFlinger::EffectModule::~EffectModule() 5286{ 5287 LOGV("Destructor %p", this); 5288 if (mEffectInterface != NULL) { 5289 // release effect engine 5290 EffectRelease(mEffectInterface); 5291 } 5292} 5293 5294status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5295{ 5296 status_t status; 5297 5298 Mutex::Autolock _l(mLock); 5299 // First handle in mHandles has highest priority and controls the effect module 5300 int priority = handle->priority(); 5301 size_t size = mHandles.size(); 5302 sp<EffectHandle> h; 5303 size_t i; 5304 for (i = 0; i < size; i++) { 5305 h = mHandles[i].promote(); 5306 if (h == 0) continue; 5307 if (h->priority() <= priority) break; 5308 } 5309 // if inserted in first place, move effect control from previous owner to this handle 5310 if (i == 0) { 5311 if (h != 0) { 5312 h->setControl(false, true); 5313 } 5314 handle->setControl(true, false); 5315 status = NO_ERROR; 5316 } else { 5317 status = ALREADY_EXISTS; 5318 } 5319 mHandles.insertAt(handle, i); 5320 return status; 5321} 5322 5323size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5324{ 5325 Mutex::Autolock _l(mLock); 5326 size_t size = mHandles.size(); 5327 size_t i; 5328 for (i = 0; i < size; i++) { 5329 if (mHandles[i] == handle) break; 5330 } 5331 if (i == size) { 5332 return size; 5333 } 5334 mHandles.removeAt(i); 5335 size = mHandles.size(); 5336 // if removed from first place, move effect control from this handle to next in line 5337 if (i == 0 && size != 0) { 5338 sp<EffectHandle> h = mHandles[0].promote(); 5339 if (h != 0) { 5340 h->setControl(true, true); 5341 } 5342 } 5343 5344 // Release effect engine here so that it is done immediately. Otherwise it will be released 5345 // by the destructor when the last strong reference on the this object is released which can 5346 // happen after next process is called on this effect. 5347 if (size == 0 && mEffectInterface != NULL) { 5348 // release effect engine 5349 EffectRelease(mEffectInterface); 5350 mEffectInterface = NULL; 5351 } 5352 5353 return size; 5354} 5355 5356void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5357{ 5358 // keep a strong reference on this EffectModule to avoid calling the 5359 // destructor before we exit 5360 sp<EffectModule> keep(this); 5361 { 5362 sp<ThreadBase> thread = mThread.promote(); 5363 if (thread != 0) { 5364 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5365 playbackThread->disconnectEffect(keep, handle); 5366 } 5367 } 5368} 5369 5370void AudioFlinger::EffectModule::updateState() { 5371 Mutex::Autolock _l(mLock); 5372 5373 switch (mState) { 5374 case RESTART: 5375 reset_l(); 5376 // FALL THROUGH 5377 5378 case STARTING: 5379 // clear auxiliary effect input buffer for next accumulation 5380 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5381 memset(mConfig.inputCfg.buffer.raw, 5382 0, 5383 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5384 } 5385 start_l(); 5386 mState = ACTIVE; 5387 break; 5388 case STOPPING: 5389 stop_l(); 5390 mDisableWaitCnt = mMaxDisableWaitCnt; 5391 mState = STOPPED; 5392 break; 5393 case STOPPED: 5394 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5395 // turn off sequence. 5396 if (--mDisableWaitCnt == 0) { 5397 reset_l(); 5398 mState = IDLE; 5399 } 5400 break; 5401 default: //IDLE , ACTIVE 5402 break; 5403 } 5404} 5405 5406void AudioFlinger::EffectModule::process() 5407{ 5408 Mutex::Autolock _l(mLock); 5409 5410 if (mEffectInterface == NULL || 5411 mConfig.inputCfg.buffer.raw == NULL || 5412 mConfig.outputCfg.buffer.raw == NULL) { 5413 return; 5414 } 5415 5416 if (isProcessEnabled()) { 5417 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5418 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5419 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5420 mConfig.inputCfg.buffer.s32, 5421 mConfig.inputCfg.buffer.frameCount/2); 5422 } 5423 5424 // do the actual processing in the effect engine 5425 int ret = (*mEffectInterface)->process(mEffectInterface, 5426 &mConfig.inputCfg.buffer, 5427 &mConfig.outputCfg.buffer); 5428 5429 // force transition to IDLE state when engine is ready 5430 if (mState == STOPPED && ret == -ENODATA) { 5431 mDisableWaitCnt = 1; 5432 } 5433 5434 // clear auxiliary effect input buffer for next accumulation 5435 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5436 memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5437 } 5438 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5439 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ 5440 // If an insert effect is idle and input buffer is different from output buffer, copy input to 5441 // output 5442 sp<EffectChain> chain = mChain.promote(); 5443 if (chain != 0 && chain->activeTracks() != 0) { 5444 size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); 5445 if (mConfig.inputCfg.channels == CHANNEL_STEREO) { 5446 size *= 2; 5447 } 5448 memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); 5449 } 5450 } 5451} 5452 5453void AudioFlinger::EffectModule::reset_l() 5454{ 5455 if (mEffectInterface == NULL) { 5456 return; 5457 } 5458 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5459} 5460 5461status_t AudioFlinger::EffectModule::configure() 5462{ 5463 uint32_t channels; 5464 if (mEffectInterface == NULL) { 5465 return NO_INIT; 5466 } 5467 5468 sp<ThreadBase> thread = mThread.promote(); 5469 if (thread == 0) { 5470 return DEAD_OBJECT; 5471 } 5472 5473 // TODO: handle configuration of effects replacing track process 5474 if (thread->channelCount() == 1) { 5475 channels = CHANNEL_MONO; 5476 } else { 5477 channels = CHANNEL_STEREO; 5478 } 5479 5480 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5481 mConfig.inputCfg.channels = CHANNEL_MONO; 5482 } else { 5483 mConfig.inputCfg.channels = channels; 5484 } 5485 mConfig.outputCfg.channels = channels; 5486 mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; 5487 mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; 5488 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5489 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5490 mConfig.inputCfg.bufferProvider.cookie = NULL; 5491 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5492 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5493 mConfig.outputCfg.bufferProvider.cookie = NULL; 5494 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5495 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5496 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5497 // Insert effect: 5498 // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE, 5499 // always overwrites output buffer: input buffer == output buffer 5500 // - in other sessions: 5501 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5502 // other effect: overwrites output buffer: input buffer == output buffer 5503 // Auxiliary effect: 5504 // accumulates in output buffer: input buffer != output buffer 5505 // Therefore: accumulate <=> input buffer != output buffer 5506 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5507 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5508 } else { 5509 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5510 } 5511 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5512 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5513 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5514 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5515 5516 LOGV("configure() %p thread %p buffer %p framecount %d", 5517 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5518 5519 status_t cmdStatus; 5520 uint32_t size = sizeof(int); 5521 status_t status = (*mEffectInterface)->command(mEffectInterface, 5522 EFFECT_CMD_CONFIGURE, 5523 sizeof(effect_config_t), 5524 &mConfig, 5525 &size, 5526 &cmdStatus); 5527 if (status == 0) { 5528 status = cmdStatus; 5529 } 5530 5531 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5532 (1000 * mConfig.outputCfg.buffer.frameCount); 5533 5534 return status; 5535} 5536 5537status_t AudioFlinger::EffectModule::init() 5538{ 5539 Mutex::Autolock _l(mLock); 5540 if (mEffectInterface == NULL) { 5541 return NO_INIT; 5542 } 5543 status_t cmdStatus; 5544 uint32_t size = sizeof(status_t); 5545 status_t status = (*mEffectInterface)->command(mEffectInterface, 5546 EFFECT_CMD_INIT, 5547 0, 5548 NULL, 5549 &size, 5550 &cmdStatus); 5551 if (status == 0) { 5552 status = cmdStatus; 5553 } 5554 return status; 5555} 5556 5557status_t AudioFlinger::EffectModule::start_l() 5558{ 5559 if (mEffectInterface == NULL) { 5560 return NO_INIT; 5561 } 5562 status_t cmdStatus; 5563 uint32_t size = sizeof(status_t); 5564 status_t status = (*mEffectInterface)->command(mEffectInterface, 5565 EFFECT_CMD_ENABLE, 5566 0, 5567 NULL, 5568 &size, 5569 &cmdStatus); 5570 if (status == 0) { 5571 status = cmdStatus; 5572 } 5573 return status; 5574} 5575 5576status_t AudioFlinger::EffectModule::stop_l() 5577{ 5578 if (mEffectInterface == NULL) { 5579 return NO_INIT; 5580 } 5581 status_t cmdStatus; 5582 uint32_t size = sizeof(status_t); 5583 status_t status = (*mEffectInterface)->command(mEffectInterface, 5584 EFFECT_CMD_DISABLE, 5585 0, 5586 NULL, 5587 &size, 5588 &cmdStatus); 5589 if (status == 0) { 5590 status = cmdStatus; 5591 } 5592 return status; 5593} 5594 5595status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5596 uint32_t cmdSize, 5597 void *pCmdData, 5598 uint32_t *replySize, 5599 void *pReplyData) 5600{ 5601 Mutex::Autolock _l(mLock); 5602// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5603 5604 if (mEffectInterface == NULL) { 5605 return NO_INIT; 5606 } 5607 status_t status = (*mEffectInterface)->command(mEffectInterface, 5608 cmdCode, 5609 cmdSize, 5610 pCmdData, 5611 replySize, 5612 pReplyData); 5613 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5614 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5615 for (size_t i = 1; i < mHandles.size(); i++) { 5616 sp<EffectHandle> h = mHandles[i].promote(); 5617 if (h != 0) { 5618 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5619 } 5620 } 5621 } 5622 return status; 5623} 5624 5625status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5626{ 5627 Mutex::Autolock _l(mLock); 5628 LOGV("setEnabled %p enabled %d", this, enabled); 5629 5630 if (enabled != isEnabled()) { 5631 switch (mState) { 5632 // going from disabled to enabled 5633 case IDLE: 5634 mState = STARTING; 5635 break; 5636 case STOPPED: 5637 mState = RESTART; 5638 break; 5639 case STOPPING: 5640 mState = ACTIVE; 5641 break; 5642 5643 // going from enabled to disabled 5644 case RESTART: 5645 mState = STOPPED; 5646 break; 5647 case STARTING: 5648 mState = IDLE; 5649 break; 5650 case ACTIVE: 5651 mState = STOPPING; 5652 break; 5653 } 5654 for (size_t i = 1; i < mHandles.size(); i++) { 5655 sp<EffectHandle> h = mHandles[i].promote(); 5656 if (h != 0) { 5657 h->setEnabled(enabled); 5658 } 5659 } 5660 } 5661 return NO_ERROR; 5662} 5663 5664bool AudioFlinger::EffectModule::isEnabled() 5665{ 5666 switch (mState) { 5667 case RESTART: 5668 case STARTING: 5669 case ACTIVE: 5670 return true; 5671 case IDLE: 5672 case STOPPING: 5673 case STOPPED: 5674 default: 5675 return false; 5676 } 5677} 5678 5679bool AudioFlinger::EffectModule::isProcessEnabled() 5680{ 5681 switch (mState) { 5682 case RESTART: 5683 case ACTIVE: 5684 case STOPPING: 5685 case STOPPED: 5686 return true; 5687 case IDLE: 5688 case STARTING: 5689 default: 5690 return false; 5691 } 5692} 5693 5694status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5695{ 5696 Mutex::Autolock _l(mLock); 5697 status_t status = NO_ERROR; 5698 5699 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5700 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5701 if (isProcessEnabled() && 5702 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5703 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5704 status_t cmdStatus; 5705 uint32_t volume[2]; 5706 uint32_t *pVolume = NULL; 5707 uint32_t size = sizeof(volume); 5708 volume[0] = *left; 5709 volume[1] = *right; 5710 if (controller) { 5711 pVolume = volume; 5712 } 5713 status = (*mEffectInterface)->command(mEffectInterface, 5714 EFFECT_CMD_SET_VOLUME, 5715 size, 5716 volume, 5717 &size, 5718 pVolume); 5719 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5720 *left = volume[0]; 5721 *right = volume[1]; 5722 } 5723 } 5724 return status; 5725} 5726 5727status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5728{ 5729 Mutex::Autolock _l(mLock); 5730 status_t status = NO_ERROR; 5731 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5732 // convert device bit field from AudioSystem to EffectApi format. 5733 device = deviceAudioSystemToEffectApi(device); 5734 if (device == 0) { 5735 return BAD_VALUE; 5736 } 5737 status_t cmdStatus; 5738 uint32_t size = sizeof(status_t); 5739 status = (*mEffectInterface)->command(mEffectInterface, 5740 EFFECT_CMD_SET_DEVICE, 5741 sizeof(uint32_t), 5742 &device, 5743 &size, 5744 &cmdStatus); 5745 if (status == NO_ERROR) { 5746 status = cmdStatus; 5747 } 5748 } 5749 return status; 5750} 5751 5752status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5753{ 5754 Mutex::Autolock _l(mLock); 5755 status_t status = NO_ERROR; 5756 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5757 // convert audio mode from AudioSystem to EffectApi format. 5758 int effectMode = modeAudioSystemToEffectApi(mode); 5759 if (effectMode < 0) { 5760 return BAD_VALUE; 5761 } 5762 status_t cmdStatus; 5763 uint32_t size = sizeof(status_t); 5764 status = (*mEffectInterface)->command(mEffectInterface, 5765 EFFECT_CMD_SET_AUDIO_MODE, 5766 sizeof(int), 5767 &effectMode, 5768 &size, 5769 &cmdStatus); 5770 if (status == NO_ERROR) { 5771 status = cmdStatus; 5772 } 5773 } 5774 return status; 5775} 5776 5777// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified 5778const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { 5779 DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE 5780 DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER 5781 DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET 5782 DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE 5783 DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO 5784 DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET 5785 DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT 5786 DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP 5787 DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES 5788 DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER 5789 DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL 5790}; 5791 5792uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) 5793{ 5794 uint32_t deviceOut = 0; 5795 while (device) { 5796 const uint32_t i = 31 - __builtin_clz(device); 5797 device &= ~(1 << i); 5798 if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { 5799 LOGE("device convertion error for AudioSystem device 0x%08x", device); 5800 return 0; 5801 } 5802 deviceOut |= (uint32_t)sDeviceConvTable[i]; 5803 } 5804 return deviceOut; 5805} 5806 5807// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified 5808const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { 5809 AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL 5810 AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE 5811 AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL 5812}; 5813 5814int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) 5815{ 5816 int modeOut = -1; 5817 if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { 5818 modeOut = (int)sModeConvTable[mode]; 5819 } 5820 return modeOut; 5821} 5822 5823status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5824{ 5825 const size_t SIZE = 256; 5826 char buffer[SIZE]; 5827 String8 result; 5828 5829 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5830 result.append(buffer); 5831 5832 bool locked = tryLock(mLock); 5833 // failed to lock - AudioFlinger is probably deadlocked 5834 if (!locked) { 5835 result.append("\t\tCould not lock Fx mutex:\n"); 5836 } 5837 5838 result.append("\t\tSession Status State Engine:\n"); 5839 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5840 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5841 result.append(buffer); 5842 5843 result.append("\t\tDescriptor:\n"); 5844 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5845 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5846 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5847 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5848 result.append(buffer); 5849 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5850 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5851 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5852 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5853 result.append(buffer); 5854 snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", 5855 mDescriptor.apiVersion, 5856 mDescriptor.flags); 5857 result.append(buffer); 5858 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5859 mDescriptor.name); 5860 result.append(buffer); 5861 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5862 mDescriptor.implementor); 5863 result.append(buffer); 5864 5865 result.append("\t\t- Input configuration:\n"); 5866 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5867 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5868 (uint32_t)mConfig.inputCfg.buffer.raw, 5869 mConfig.inputCfg.buffer.frameCount, 5870 mConfig.inputCfg.samplingRate, 5871 mConfig.inputCfg.channels, 5872 mConfig.inputCfg.format); 5873 result.append(buffer); 5874 5875 result.append("\t\t- Output configuration:\n"); 5876 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5877 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5878 (uint32_t)mConfig.outputCfg.buffer.raw, 5879 mConfig.outputCfg.buffer.frameCount, 5880 mConfig.outputCfg.samplingRate, 5881 mConfig.outputCfg.channels, 5882 mConfig.outputCfg.format); 5883 result.append(buffer); 5884 5885 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5886 result.append(buffer); 5887 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5888 for (size_t i = 0; i < mHandles.size(); ++i) { 5889 sp<EffectHandle> handle = mHandles[i].promote(); 5890 if (handle != 0) { 5891 handle->dump(buffer, SIZE); 5892 result.append(buffer); 5893 } 5894 } 5895 5896 result.append("\n"); 5897 5898 write(fd, result.string(), result.length()); 5899 5900 if (locked) { 5901 mLock.unlock(); 5902 } 5903 5904 return NO_ERROR; 5905} 5906 5907// ---------------------------------------------------------------------------- 5908// EffectHandle implementation 5909// ---------------------------------------------------------------------------- 5910 5911#undef LOG_TAG 5912#define LOG_TAG "AudioFlinger::EffectHandle" 5913 5914AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5915 const sp<AudioFlinger::Client>& client, 5916 const sp<IEffectClient>& effectClient, 5917 int32_t priority) 5918 : BnEffect(), 5919 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5920{ 5921 LOGV("constructor %p", this); 5922 5923 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5924 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5925 if (mCblkMemory != 0) { 5926 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5927 5928 if (mCblk) { 5929 new(mCblk) effect_param_cblk_t(); 5930 mBuffer = (uint8_t *)mCblk + bufOffset; 5931 } 5932 } else { 5933 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5934 return; 5935 } 5936} 5937 5938AudioFlinger::EffectHandle::~EffectHandle() 5939{ 5940 LOGV("Destructor %p", this); 5941 disconnect(); 5942} 5943 5944status_t AudioFlinger::EffectHandle::enable() 5945{ 5946 if (!mHasControl) return INVALID_OPERATION; 5947 if (mEffect == 0) return DEAD_OBJECT; 5948 5949 return mEffect->setEnabled(true); 5950} 5951 5952status_t AudioFlinger::EffectHandle::disable() 5953{ 5954 if (!mHasControl) return INVALID_OPERATION; 5955 if (mEffect == NULL) return DEAD_OBJECT; 5956 5957 return mEffect->setEnabled(false); 5958} 5959 5960void AudioFlinger::EffectHandle::disconnect() 5961{ 5962 if (mEffect == 0) { 5963 return; 5964 } 5965 mEffect->disconnect(this); 5966 // release sp on module => module destructor can be called now 5967 mEffect.clear(); 5968 if (mCblk) { 5969 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5970 } 5971 mCblkMemory.clear(); // and free the shared memory 5972 if (mClient != 0) { 5973 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5974 mClient.clear(); 5975 } 5976} 5977 5978status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5979 uint32_t cmdSize, 5980 void *pCmdData, 5981 uint32_t *replySize, 5982 void *pReplyData) 5983{ 5984// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5985// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5986 5987 // only get parameter command is permitted for applications not controlling the effect 5988 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5989 return INVALID_OPERATION; 5990 } 5991 if (mEffect == 0) return DEAD_OBJECT; 5992 5993 // handle commands that are not forwarded transparently to effect engine 5994 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5995 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5996 // no risk to block the whole media server process or mixer threads is we are stuck here 5997 Mutex::Autolock _l(mCblk->lock); 5998 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5999 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6000 mCblk->serverIndex = 0; 6001 mCblk->clientIndex = 0; 6002 return BAD_VALUE; 6003 } 6004 status_t status = NO_ERROR; 6005 while (mCblk->serverIndex < mCblk->clientIndex) { 6006 int reply; 6007 uint32_t rsize = sizeof(int); 6008 int *p = (int *)(mBuffer + mCblk->serverIndex); 6009 int size = *p++; 6010 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6011 LOGW("command(): invalid parameter block size"); 6012 break; 6013 } 6014 effect_param_t *param = (effect_param_t *)p; 6015 if (param->psize == 0 || param->vsize == 0) { 6016 LOGW("command(): null parameter or value size"); 6017 mCblk->serverIndex += size; 6018 continue; 6019 } 6020 uint32_t psize = sizeof(effect_param_t) + 6021 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6022 param->vsize; 6023 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6024 psize, 6025 p, 6026 &rsize, 6027 &reply); 6028 // stop at first error encountered 6029 if (ret != NO_ERROR) { 6030 status = ret; 6031 *(int *)pReplyData = reply; 6032 break; 6033 } else if (reply != NO_ERROR) { 6034 *(int *)pReplyData = reply; 6035 break; 6036 } 6037 mCblk->serverIndex += size; 6038 } 6039 mCblk->serverIndex = 0; 6040 mCblk->clientIndex = 0; 6041 return status; 6042 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6043 *(int *)pReplyData = NO_ERROR; 6044 return enable(); 6045 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6046 *(int *)pReplyData = NO_ERROR; 6047 return disable(); 6048 } 6049 6050 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6051} 6052 6053sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6054 return mCblkMemory; 6055} 6056 6057void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 6058{ 6059 LOGV("setControl %p control %d", this, hasControl); 6060 6061 mHasControl = hasControl; 6062 if (signal && mEffectClient != 0) { 6063 mEffectClient->controlStatusChanged(hasControl); 6064 } 6065} 6066 6067void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6068 uint32_t cmdSize, 6069 void *pCmdData, 6070 uint32_t replySize, 6071 void *pReplyData) 6072{ 6073 if (mEffectClient != 0) { 6074 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6075 } 6076} 6077 6078 6079 6080void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6081{ 6082 if (mEffectClient != 0) { 6083 mEffectClient->enableStatusChanged(enabled); 6084 } 6085} 6086 6087status_t AudioFlinger::EffectHandle::onTransact( 6088 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6089{ 6090 return BnEffect::onTransact(code, data, reply, flags); 6091} 6092 6093 6094void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6095{ 6096 bool locked = tryLock(mCblk->lock); 6097 6098 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6099 (mClient == NULL) ? getpid() : mClient->pid(), 6100 mPriority, 6101 mHasControl, 6102 !locked, 6103 mCblk->clientIndex, 6104 mCblk->serverIndex 6105 ); 6106 6107 if (locked) { 6108 mCblk->lock.unlock(); 6109 } 6110} 6111 6112#undef LOG_TAG 6113#define LOG_TAG "AudioFlinger::EffectChain" 6114 6115AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6116 int sessionId) 6117 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), 6118 mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 6119 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 6120{ 6121 mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 6122} 6123 6124AudioFlinger::EffectChain::~EffectChain() 6125{ 6126 if (mOwnInBuffer) { 6127 delete mInBuffer; 6128 } 6129 6130} 6131 6132// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6133sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6134{ 6135 sp<EffectModule> effect; 6136 size_t size = mEffects.size(); 6137 6138 for (size_t i = 0; i < size; i++) { 6139 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6140 effect = mEffects[i]; 6141 break; 6142 } 6143 } 6144 return effect; 6145} 6146 6147// getEffectFromId_l() must be called with PlaybackThread::mLock held 6148sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6149{ 6150 sp<EffectModule> effect; 6151 size_t size = mEffects.size(); 6152 6153 for (size_t i = 0; i < size; i++) { 6154 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6155 if (id == 0 || mEffects[i]->id() == id) { 6156 effect = mEffects[i]; 6157 break; 6158 } 6159 } 6160 return effect; 6161} 6162 6163// Must be called with EffectChain::mLock locked 6164void AudioFlinger::EffectChain::process_l() 6165{ 6166 sp<ThreadBase> thread = mThread.promote(); 6167 if (thread == 0) { 6168 LOGW("process_l(): cannot promote mixer thread"); 6169 return; 6170 } 6171 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 6172 bool isGlobalSession = (mSessionId == AudioSystem::SESSION_OUTPUT_MIX) || 6173 (mSessionId == AudioSystem::SESSION_OUTPUT_STAGE); 6174 bool tracksOnSession = false; 6175 if (!isGlobalSession) { 6176 tracksOnSession = 6177 playbackThread->hasAudioSession(mSessionId) & PlaybackThread::TRACK_SESSION; 6178 } 6179 6180 size_t size = mEffects.size(); 6181 // do not process effect if no track is present in same audio session 6182 if (isGlobalSession || tracksOnSession) { 6183 for (size_t i = 0; i < size; i++) { 6184 mEffects[i]->process(); 6185 } 6186 } 6187 for (size_t i = 0; i < size; i++) { 6188 mEffects[i]->updateState(); 6189 } 6190 // if no track is active, input buffer must be cleared here as the mixer process 6191 // will not do it 6192 if (tracksOnSession && 6193 activeTracks() == 0) { 6194 size_t numSamples = playbackThread->frameCount() * playbackThread->channelCount(); 6195 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6196 } 6197} 6198 6199// addEffect_l() must be called with PlaybackThread::mLock held 6200status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6201{ 6202 effect_descriptor_t desc = effect->desc(); 6203 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6204 6205 Mutex::Autolock _l(mLock); 6206 effect->setChain(this); 6207 sp<ThreadBase> thread = mThread.promote(); 6208 if (thread == 0) { 6209 return NO_INIT; 6210 } 6211 effect->setThread(thread); 6212 6213 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6214 // Auxiliary effects are inserted at the beginning of mEffects vector as 6215 // they are processed first and accumulated in chain input buffer 6216 mEffects.insertAt(effect, 0); 6217 6218 // the input buffer for auxiliary effect contains mono samples in 6219 // 32 bit format. This is to avoid saturation in AudoMixer 6220 // accumulation stage. Saturation is done in EffectModule::process() before 6221 // calling the process in effect engine 6222 size_t numSamples = thread->frameCount(); 6223 int32_t *buffer = new int32_t[numSamples]; 6224 memset(buffer, 0, numSamples * sizeof(int32_t)); 6225 effect->setInBuffer((int16_t *)buffer); 6226 // auxiliary effects output samples to chain input buffer for further processing 6227 // by insert effects 6228 effect->setOutBuffer(mInBuffer); 6229 } else { 6230 // Insert effects are inserted at the end of mEffects vector as they are processed 6231 // after track and auxiliary effects. 6232 // Insert effect order as a function of indicated preference: 6233 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6234 // another effect is present 6235 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6236 // last effect claiming first position 6237 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6238 // first effect claiming last position 6239 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6240 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6241 // already present 6242 6243 int size = (int)mEffects.size(); 6244 int idx_insert = size; 6245 int idx_insert_first = -1; 6246 int idx_insert_last = -1; 6247 6248 for (int i = 0; i < size; i++) { 6249 effect_descriptor_t d = mEffects[i]->desc(); 6250 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6251 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6252 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6253 // check invalid effect chaining combinations 6254 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6255 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6256 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6257 return INVALID_OPERATION; 6258 } 6259 // remember position of first insert effect and by default 6260 // select this as insert position for new effect 6261 if (idx_insert == size) { 6262 idx_insert = i; 6263 } 6264 // remember position of last insert effect claiming 6265 // first position 6266 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6267 idx_insert_first = i; 6268 } 6269 // remember position of first insert effect claiming 6270 // last position 6271 if (iPref == EFFECT_FLAG_INSERT_LAST && 6272 idx_insert_last == -1) { 6273 idx_insert_last = i; 6274 } 6275 } 6276 } 6277 6278 // modify idx_insert from first position if needed 6279 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6280 if (idx_insert_last != -1) { 6281 idx_insert = idx_insert_last; 6282 } else { 6283 idx_insert = size; 6284 } 6285 } else { 6286 if (idx_insert_first != -1) { 6287 idx_insert = idx_insert_first + 1; 6288 } 6289 } 6290 6291 // always read samples from chain input buffer 6292 effect->setInBuffer(mInBuffer); 6293 6294 // if last effect in the chain, output samples to chain 6295 // output buffer, otherwise to chain input buffer 6296 if (idx_insert == size) { 6297 if (idx_insert != 0) { 6298 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6299 mEffects[idx_insert-1]->configure(); 6300 } 6301 effect->setOutBuffer(mOutBuffer); 6302 } else { 6303 effect->setOutBuffer(mInBuffer); 6304 } 6305 mEffects.insertAt(effect, idx_insert); 6306 6307 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6308 } 6309 effect->configure(); 6310 return NO_ERROR; 6311} 6312 6313// removeEffect_l() must be called with PlaybackThread::mLock held 6314size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6315{ 6316 Mutex::Autolock _l(mLock); 6317 int size = (int)mEffects.size(); 6318 int i; 6319 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6320 6321 for (i = 0; i < size; i++) { 6322 if (effect == mEffects[i]) { 6323 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6324 delete[] effect->inBuffer(); 6325 } else { 6326 if (i == size - 1 && i != 0) { 6327 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6328 mEffects[i - 1]->configure(); 6329 } 6330 } 6331 mEffects.removeAt(i); 6332 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6333 break; 6334 } 6335 } 6336 6337 return mEffects.size(); 6338} 6339 6340// setDevice_l() must be called with PlaybackThread::mLock held 6341void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6342{ 6343 size_t size = mEffects.size(); 6344 for (size_t i = 0; i < size; i++) { 6345 mEffects[i]->setDevice(device); 6346 } 6347} 6348 6349// setMode_l() must be called with PlaybackThread::mLock held 6350void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6351{ 6352 size_t size = mEffects.size(); 6353 for (size_t i = 0; i < size; i++) { 6354 mEffects[i]->setMode(mode); 6355 } 6356} 6357 6358// setVolume_l() must be called with PlaybackThread::mLock held 6359bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6360{ 6361 uint32_t newLeft = *left; 6362 uint32_t newRight = *right; 6363 bool hasControl = false; 6364 int ctrlIdx = -1; 6365 size_t size = mEffects.size(); 6366 6367 // first update volume controller 6368 for (size_t i = size; i > 0; i--) { 6369 if (mEffects[i - 1]->isProcessEnabled() && 6370 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6371 ctrlIdx = i - 1; 6372 hasControl = true; 6373 break; 6374 } 6375 } 6376 6377 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6378 if (hasControl) { 6379 *left = mNewLeftVolume; 6380 *right = mNewRightVolume; 6381 } 6382 return hasControl; 6383 } 6384 6385 mVolumeCtrlIdx = ctrlIdx; 6386 mLeftVolume = newLeft; 6387 mRightVolume = newRight; 6388 6389 // second get volume update from volume controller 6390 if (ctrlIdx >= 0) { 6391 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6392 mNewLeftVolume = newLeft; 6393 mNewRightVolume = newRight; 6394 } 6395 // then indicate volume to all other effects in chain. 6396 // Pass altered volume to effects before volume controller 6397 // and requested volume to effects after controller 6398 uint32_t lVol = newLeft; 6399 uint32_t rVol = newRight; 6400 6401 for (size_t i = 0; i < size; i++) { 6402 if ((int)i == ctrlIdx) continue; 6403 // this also works for ctrlIdx == -1 when there is no volume controller 6404 if ((int)i > ctrlIdx) { 6405 lVol = *left; 6406 rVol = *right; 6407 } 6408 mEffects[i]->setVolume(&lVol, &rVol, false); 6409 } 6410 *left = newLeft; 6411 *right = newRight; 6412 6413 return hasControl; 6414} 6415 6416status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6417{ 6418 const size_t SIZE = 256; 6419 char buffer[SIZE]; 6420 String8 result; 6421 6422 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6423 result.append(buffer); 6424 6425 bool locked = tryLock(mLock); 6426 // failed to lock - AudioFlinger is probably deadlocked 6427 if (!locked) { 6428 result.append("\tCould not lock mutex:\n"); 6429 } 6430 6431 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6432 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6433 mEffects.size(), 6434 (uint32_t)mInBuffer, 6435 (uint32_t)mOutBuffer, 6436 mActiveTrackCnt); 6437 result.append(buffer); 6438 write(fd, result.string(), result.size()); 6439 6440 for (size_t i = 0; i < mEffects.size(); ++i) { 6441 sp<EffectModule> effect = mEffects[i]; 6442 if (effect != 0) { 6443 effect->dump(fd, args); 6444 } 6445 } 6446 6447 if (locked) { 6448 mLock.unlock(); 6449 } 6450 6451 return NO_ERROR; 6452} 6453 6454#undef LOG_TAG 6455#define LOG_TAG "AudioFlinger" 6456 6457// ---------------------------------------------------------------------------- 6458 6459status_t AudioFlinger::onTransact( 6460 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6461{ 6462 return BnAudioFlinger::onTransact(code, data, reply, flags); 6463} 6464 6465}; // namespace android 6466