AudioFlinger.cpp revision 25cbe0ecd6df8be7e40537c5d85c82f105038479
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() || track->isStopped()) && 1719 !track->isPaused() && !track->isTerminated()) 1720 { 1721 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1722 1723 mixedTracks++; 1724 1725 // track->mainBuffer() != mMixBuffer means there is an effect chain 1726 // connected to the track 1727 chain.clear(); 1728 if (track->mainBuffer() != mMixBuffer) { 1729 chain = getEffectChain_l(track->sessionId()); 1730 // Delegate volume control to effect in track effect chain if needed 1731 if (chain != 0) { 1732 tracksWithEffect++; 1733 } else { 1734 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1735 track->name(), track->sessionId()); 1736 } 1737 } 1738 1739 1740 int param = AudioMixer::VOLUME; 1741 if (track->mFillingUpStatus == Track::FS_FILLED) { 1742 // no ramp for the first volume setting 1743 track->mFillingUpStatus = Track::FS_ACTIVE; 1744 if (track->mState == TrackBase::RESUMING) { 1745 track->mState = TrackBase::ACTIVE; 1746 param = AudioMixer::RAMP_VOLUME; 1747 } 1748 } else if (cblk->server != 0) { 1749 // If the track is stopped before the first frame was mixed, 1750 // do not apply ramp 1751 param = AudioMixer::RAMP_VOLUME; 1752 } 1753 1754 // compute volume for this track 1755 int16_t left, right, aux; 1756 if (track->isMuted() || track->isPausing() || 1757 mStreamTypes[track->type()].mute) { 1758 left = right = aux = 0; 1759 if (track->isPausing()) { 1760 track->setPaused(); 1761 } 1762 } else { 1763 // read original volumes with volume control 1764 float typeVolume = mStreamTypes[track->type()].volume; 1765#ifdef LVMX 1766 bool streamMute=false; 1767 // read the volume from the LivesVibes audio engine. 1768 if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) 1769 { 1770 LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); 1771 if (streamMute) { 1772 typeVolume = 0; 1773 } 1774 } 1775#endif 1776 float v = masterVolume * typeVolume; 1777 uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12; 1778 uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12; 1779 1780 // Delegate volume control to effect in track effect chain if needed 1781 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1782 // Do not ramp volume is volume is controlled by effect 1783 param = AudioMixer::VOLUME; 1784 } 1785 1786 // Convert volumes from 8.24 to 4.12 format 1787 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1788 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1789 left = int16_t(v_clamped); 1790 v_clamped = (vr + (1 << 11)) >> 12; 1791 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1792 right = int16_t(v_clamped); 1793 1794 v_clamped = (uint32_t)(v * cblk->sendLevel); 1795 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1796 aux = int16_t(v_clamped); 1797 } 1798 1799#ifdef LVMX 1800 if ( tracksConnectedChanged || stateChanged ) 1801 { 1802 // only do the ramp when the volume is changed by the user / application 1803 param = AudioMixer::VOLUME; 1804 } 1805#endif 1806 1807 // XXX: these things DON'T need to be done each time 1808 mAudioMixer->setBufferProvider(track); 1809 mAudioMixer->enable(AudioMixer::MIXING); 1810 1811 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1812 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1813 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1814 mAudioMixer->setParameter( 1815 AudioMixer::TRACK, 1816 AudioMixer::FORMAT, (void *)track->format()); 1817 mAudioMixer->setParameter( 1818 AudioMixer::TRACK, 1819 AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); 1820 mAudioMixer->setParameter( 1821 AudioMixer::RESAMPLE, 1822 AudioMixer::SAMPLE_RATE, 1823 (void *)(cblk->sampleRate)); 1824 mAudioMixer->setParameter( 1825 AudioMixer::TRACK, 1826 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1827 mAudioMixer->setParameter( 1828 AudioMixer::TRACK, 1829 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1830 1831 // reset retry count 1832 track->mRetryCount = kMaxTrackRetries; 1833 mixerStatus = MIXER_TRACKS_READY; 1834 } else { 1835 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1836 if (track->isStopped()) { 1837 track->reset(); 1838 } 1839 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1840 // We have consumed all the buffers of this track. 1841 // Remove it from the list of active tracks. 1842 tracksToRemove->add(track); 1843 } else { 1844 // No buffers for this track. Give it a few chances to 1845 // fill a buffer, then remove it from active list. 1846 if (--(track->mRetryCount) <= 0) { 1847 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1848 tracksToRemove->add(track); 1849 } else if (mixerStatus != MIXER_TRACKS_READY) { 1850 mixerStatus = MIXER_TRACKS_ENABLED; 1851 } 1852 } 1853 mAudioMixer->disable(AudioMixer::MIXING); 1854 } 1855 } 1856 1857 // remove all the tracks that need to be... 1858 count = tracksToRemove->size(); 1859 if (UNLIKELY(count)) { 1860 for (size_t i=0 ; i<count ; i++) { 1861 const sp<Track>& track = tracksToRemove->itemAt(i); 1862 mActiveTracks.remove(track); 1863 if (track->mainBuffer() != mMixBuffer) { 1864 chain = getEffectChain_l(track->sessionId()); 1865 if (chain != 0) { 1866 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1867 chain->stopTrack(); 1868 } 1869 } 1870 if (track->isTerminated()) { 1871 mTracks.remove(track); 1872 deleteTrackName_l(track->mName); 1873 } 1874 } 1875 } 1876 1877 // mix buffer must be cleared if all tracks are connected to an 1878 // effect chain as in this case the mixer will not write to 1879 // mix buffer and track effects will accumulate into it 1880 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1881 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1882 } 1883 1884 return mixerStatus; 1885} 1886 1887void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1888{ 1889 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1890 this, streamType, mTracks.size()); 1891 Mutex::Autolock _l(mLock); 1892 1893 size_t size = mTracks.size(); 1894 for (size_t i = 0; i < size; i++) { 1895 sp<Track> t = mTracks[i]; 1896 if (t->type() == streamType) { 1897 t->mCblk->lock.lock(); 1898 t->mCblk->flags |= CBLK_INVALID_ON; 1899 t->mCblk->cv.signal(); 1900 t->mCblk->lock.unlock(); 1901 } 1902 } 1903} 1904 1905 1906// getTrackName_l() must be called with ThreadBase::mLock held 1907int AudioFlinger::MixerThread::getTrackName_l() 1908{ 1909 return mAudioMixer->getTrackName(); 1910} 1911 1912// deleteTrackName_l() must be called with ThreadBase::mLock held 1913void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1914{ 1915 LOGV("remove track (%d) and delete from mixer", name); 1916 mAudioMixer->deleteTrackName(name); 1917} 1918 1919// checkForNewParameters_l() must be called with ThreadBase::mLock held 1920bool AudioFlinger::MixerThread::checkForNewParameters_l() 1921{ 1922 bool reconfig = false; 1923 1924 while (!mNewParameters.isEmpty()) { 1925 status_t status = NO_ERROR; 1926 String8 keyValuePair = mNewParameters[0]; 1927 AudioParameter param = AudioParameter(keyValuePair); 1928 int value; 1929 1930 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1931 reconfig = true; 1932 } 1933 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1934 if (value != AudioSystem::PCM_16_BIT) { 1935 status = BAD_VALUE; 1936 } else { 1937 reconfig = true; 1938 } 1939 } 1940 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1941 if (value != AudioSystem::CHANNEL_OUT_STEREO) { 1942 status = BAD_VALUE; 1943 } else { 1944 reconfig = true; 1945 } 1946 } 1947 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1948 // do not accept frame count changes if tracks are open as the track buffer 1949 // size depends on frame count and correct behavior would not be garantied 1950 // if frame count is changed after track creation 1951 if (!mTracks.isEmpty()) { 1952 status = INVALID_OPERATION; 1953 } else { 1954 reconfig = true; 1955 } 1956 } 1957 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1958 // forward device change to effects that have requested to be 1959 // aware of attached audio device. 1960 mDevice = (uint32_t)value; 1961 for (size_t i = 0; i < mEffectChains.size(); i++) { 1962 mEffectChains[i]->setDevice_l(mDevice); 1963 } 1964 } 1965 1966 if (status == NO_ERROR) { 1967 status = mOutput->setParameters(keyValuePair); 1968 if (!mStandby && status == INVALID_OPERATION) { 1969 mOutput->standby(); 1970 mStandby = true; 1971 mBytesWritten = 0; 1972 status = mOutput->setParameters(keyValuePair); 1973 } 1974 if (status == NO_ERROR && reconfig) { 1975 delete mAudioMixer; 1976 readOutputParameters(); 1977 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1978 for (size_t i = 0; i < mTracks.size() ; i++) { 1979 int name = getTrackName_l(); 1980 if (name < 0) break; 1981 mTracks[i]->mName = name; 1982 // limit track sample rate to 2 x new output sample rate 1983 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1984 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 1985 } 1986 } 1987 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 1988 } 1989 } 1990 1991 mNewParameters.removeAt(0); 1992 1993 mParamStatus = status; 1994 mParamCond.signal(); 1995 mWaitWorkCV.wait(mLock); 1996 } 1997 return reconfig; 1998} 1999 2000status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2001{ 2002 const size_t SIZE = 256; 2003 char buffer[SIZE]; 2004 String8 result; 2005 2006 PlaybackThread::dumpInternals(fd, args); 2007 2008 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2009 result.append(buffer); 2010 write(fd, result.string(), result.size()); 2011 return NO_ERROR; 2012} 2013 2014uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2015{ 2016 return (uint32_t)(mOutput->latency() * 1000) / 2; 2017} 2018 2019uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2020{ 2021 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2022} 2023 2024uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2025{ 2026 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2027} 2028 2029// ---------------------------------------------------------------------------- 2030AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2031 : PlaybackThread(audioFlinger, output, id, device) 2032{ 2033 mType = PlaybackThread::DIRECT; 2034} 2035 2036AudioFlinger::DirectOutputThread::~DirectOutputThread() 2037{ 2038} 2039 2040 2041static inline int16_t clamp16(int32_t sample) 2042{ 2043 if ((sample>>15) ^ (sample>>31)) 2044 sample = 0x7FFF ^ (sample>>31); 2045 return sample; 2046} 2047 2048static inline 2049int32_t mul(int16_t in, int16_t v) 2050{ 2051#if defined(__arm__) && !defined(__thumb__) 2052 int32_t out; 2053 asm( "smulbb %[out], %[in], %[v] \n" 2054 : [out]"=r"(out) 2055 : [in]"%r"(in), [v]"r"(v) 2056 : ); 2057 return out; 2058#else 2059 return in * int32_t(v); 2060#endif 2061} 2062 2063void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2064{ 2065 // Do not apply volume on compressed audio 2066 if (!AudioSystem::isLinearPCM(mFormat)) { 2067 return; 2068 } 2069 2070 // convert to signed 16 bit before volume calculation 2071 if (mFormat == AudioSystem::PCM_8_BIT) { 2072 size_t count = mFrameCount * mChannelCount; 2073 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2074 int16_t *dst = mMixBuffer + count-1; 2075 while(count--) { 2076 *dst-- = (int16_t)(*src--^0x80) << 8; 2077 } 2078 } 2079 2080 size_t frameCount = mFrameCount; 2081 int16_t *out = mMixBuffer; 2082 if (ramp) { 2083 if (mChannelCount == 1) { 2084 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2085 int32_t vlInc = d / (int32_t)frameCount; 2086 int32_t vl = ((int32_t)mLeftVolShort << 16); 2087 do { 2088 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2089 out++; 2090 vl += vlInc; 2091 } while (--frameCount); 2092 2093 } else { 2094 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2095 int32_t vlInc = d / (int32_t)frameCount; 2096 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2097 int32_t vrInc = d / (int32_t)frameCount; 2098 int32_t vl = ((int32_t)mLeftVolShort << 16); 2099 int32_t vr = ((int32_t)mRightVolShort << 16); 2100 do { 2101 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2102 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2103 out += 2; 2104 vl += vlInc; 2105 vr += vrInc; 2106 } while (--frameCount); 2107 } 2108 } else { 2109 if (mChannelCount == 1) { 2110 do { 2111 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2112 out++; 2113 } while (--frameCount); 2114 } else { 2115 do { 2116 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2117 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2118 out += 2; 2119 } while (--frameCount); 2120 } 2121 } 2122 2123 // convert back to unsigned 8 bit after volume calculation 2124 if (mFormat == AudioSystem::PCM_8_BIT) { 2125 size_t count = mFrameCount * mChannelCount; 2126 int16_t *src = mMixBuffer; 2127 uint8_t *dst = (uint8_t *)mMixBuffer; 2128 while(count--) { 2129 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2130 } 2131 } 2132 2133 mLeftVolShort = leftVol; 2134 mRightVolShort = rightVol; 2135} 2136 2137bool AudioFlinger::DirectOutputThread::threadLoop() 2138{ 2139 uint32_t mixerStatus = MIXER_IDLE; 2140 sp<Track> trackToRemove; 2141 sp<Track> activeTrack; 2142 nsecs_t standbyTime = systemTime(); 2143 int8_t *curBuf; 2144 size_t mixBufferSize = mFrameCount*mFrameSize; 2145 uint32_t activeSleepTime = activeSleepTimeUs(); 2146 uint32_t idleSleepTime = idleSleepTimeUs(); 2147 uint32_t sleepTime = idleSleepTime; 2148 // use shorter standby delay as on normal output to release 2149 // hardware resources as soon as possible 2150 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2151 2152 while (!exitPending()) 2153 { 2154 bool rampVolume; 2155 uint16_t leftVol; 2156 uint16_t rightVol; 2157 Vector< sp<EffectChain> > effectChains; 2158 2159 processConfigEvents(); 2160 2161 mixerStatus = MIXER_IDLE; 2162 2163 { // scope for the mLock 2164 2165 Mutex::Autolock _l(mLock); 2166 2167 if (checkForNewParameters_l()) { 2168 mixBufferSize = mFrameCount*mFrameSize; 2169 activeSleepTime = activeSleepTimeUs(); 2170 idleSleepTime = idleSleepTimeUs(); 2171 standbyDelay = microseconds(activeSleepTime*2); 2172 } 2173 2174 // put audio hardware into standby after short delay 2175 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2176 mSuspended) { 2177 // wait until we have something to do... 2178 if (!mStandby) { 2179 LOGV("Audio hardware entering standby, mixer %p\n", this); 2180 mOutput->standby(); 2181 mStandby = true; 2182 mBytesWritten = 0; 2183 } 2184 2185 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2186 // we're about to wait, flush the binder command buffer 2187 IPCThreadState::self()->flushCommands(); 2188 2189 if (exitPending()) break; 2190 2191 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2192 mWaitWorkCV.wait(mLock); 2193 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2194 2195 if (mMasterMute == false) { 2196 char value[PROPERTY_VALUE_MAX]; 2197 property_get("ro.audio.silent", value, "0"); 2198 if (atoi(value)) { 2199 LOGD("Silence is golden"); 2200 setMasterMute(true); 2201 } 2202 } 2203 2204 standbyTime = systemTime() + standbyDelay; 2205 sleepTime = idleSleepTime; 2206 continue; 2207 } 2208 } 2209 2210 effectChains = mEffectChains; 2211 2212 // find out which tracks need to be processed 2213 if (mActiveTracks.size() != 0) { 2214 sp<Track> t = mActiveTracks[0].promote(); 2215 if (t == 0) continue; 2216 2217 Track* const track = t.get(); 2218 audio_track_cblk_t* cblk = track->cblk(); 2219 2220 // The first time a track is added we wait 2221 // for all its buffers to be filled before processing it 2222 if (cblk->framesReady() && (track->isReady() || track->isStopped()) && 2223 !track->isPaused() && !track->isTerminated()) 2224 { 2225 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2226 2227 if (track->mFillingUpStatus == Track::FS_FILLED) { 2228 track->mFillingUpStatus = Track::FS_ACTIVE; 2229 mLeftVolFloat = mRightVolFloat = 0; 2230 mLeftVolShort = mRightVolShort = 0; 2231 if (track->mState == TrackBase::RESUMING) { 2232 track->mState = TrackBase::ACTIVE; 2233 rampVolume = true; 2234 } 2235 } else if (cblk->server != 0) { 2236 // If the track is stopped before the first frame was mixed, 2237 // do not apply ramp 2238 rampVolume = true; 2239 } 2240 // compute volume for this track 2241 float left, right; 2242 if (track->isMuted() || mMasterMute || track->isPausing() || 2243 mStreamTypes[track->type()].mute) { 2244 left = right = 0; 2245 if (track->isPausing()) { 2246 track->setPaused(); 2247 } 2248 } else { 2249 float typeVolume = mStreamTypes[track->type()].volume; 2250 float v = mMasterVolume * typeVolume; 2251 float v_clamped = v * cblk->volume[0]; 2252 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2253 left = v_clamped/MAX_GAIN; 2254 v_clamped = v * cblk->volume[1]; 2255 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2256 right = v_clamped/MAX_GAIN; 2257 } 2258 2259 if (left != mLeftVolFloat || right != mRightVolFloat) { 2260 mLeftVolFloat = left; 2261 mRightVolFloat = right; 2262 2263 // If audio HAL implements volume control, 2264 // force software volume to nominal value 2265 if (mOutput->setVolume(left, right) == NO_ERROR) { 2266 left = 1.0f; 2267 right = 1.0f; 2268 } 2269 2270 // Convert volumes from float to 8.24 2271 uint32_t vl = (uint32_t)(left * (1 << 24)); 2272 uint32_t vr = (uint32_t)(right * (1 << 24)); 2273 2274 // Delegate volume control to effect in track effect chain if needed 2275 // only one effect chain can be present on DirectOutputThread, so if 2276 // there is one, the track is connected to it 2277 if (!effectChains.isEmpty()) { 2278 // Do not ramp volume is volume is controlled by effect 2279 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2280 rampVolume = false; 2281 } 2282 } 2283 2284 // Convert volumes from 8.24 to 4.12 format 2285 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2286 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2287 leftVol = (uint16_t)v_clamped; 2288 v_clamped = (vr + (1 << 11)) >> 12; 2289 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2290 rightVol = (uint16_t)v_clamped; 2291 } else { 2292 leftVol = mLeftVolShort; 2293 rightVol = mRightVolShort; 2294 rampVolume = false; 2295 } 2296 2297 // reset retry count 2298 track->mRetryCount = kMaxTrackRetriesDirect; 2299 activeTrack = t; 2300 mixerStatus = MIXER_TRACKS_READY; 2301 } else { 2302 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2303 if (track->isStopped()) { 2304 track->reset(); 2305 } 2306 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2307 // We have consumed all the buffers of this track. 2308 // Remove it from the list of active tracks. 2309 trackToRemove = track; 2310 } else { 2311 // No buffers for this track. Give it a few chances to 2312 // fill a buffer, then remove it from active list. 2313 if (--(track->mRetryCount) <= 0) { 2314 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2315 trackToRemove = track; 2316 } else { 2317 mixerStatus = MIXER_TRACKS_ENABLED; 2318 } 2319 } 2320 } 2321 } 2322 2323 // remove all the tracks that need to be... 2324 if (UNLIKELY(trackToRemove != 0)) { 2325 mActiveTracks.remove(trackToRemove); 2326 if (!effectChains.isEmpty()) { 2327 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2328 trackToRemove->sessionId()); 2329 effectChains[0]->stopTrack(); 2330 } 2331 if (trackToRemove->isTerminated()) { 2332 mTracks.remove(trackToRemove); 2333 deleteTrackName_l(trackToRemove->mName); 2334 } 2335 } 2336 2337 lockEffectChains_l(effectChains); 2338 } 2339 2340 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2341 AudioBufferProvider::Buffer buffer; 2342 size_t frameCount = mFrameCount; 2343 curBuf = (int8_t *)mMixBuffer; 2344 // output audio to hardware 2345 while (frameCount) { 2346 buffer.frameCount = frameCount; 2347 activeTrack->getNextBuffer(&buffer); 2348 if (UNLIKELY(buffer.raw == 0)) { 2349 memset(curBuf, 0, frameCount * mFrameSize); 2350 break; 2351 } 2352 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2353 frameCount -= buffer.frameCount; 2354 curBuf += buffer.frameCount * mFrameSize; 2355 activeTrack->releaseBuffer(&buffer); 2356 } 2357 sleepTime = 0; 2358 standbyTime = systemTime() + standbyDelay; 2359 } else { 2360 if (sleepTime == 0) { 2361 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2362 sleepTime = activeSleepTime; 2363 } else { 2364 sleepTime = idleSleepTime; 2365 } 2366 } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { 2367 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2368 sleepTime = 0; 2369 } 2370 } 2371 2372 if (mSuspended) { 2373 sleepTime = suspendSleepTimeUs(); 2374 } 2375 // sleepTime == 0 means we must write to audio hardware 2376 if (sleepTime == 0) { 2377 if (mixerStatus == MIXER_TRACKS_READY) { 2378 applyVolume(leftVol, rightVol, rampVolume); 2379 } 2380 for (size_t i = 0; i < effectChains.size(); i ++) { 2381 effectChains[i]->process_l(); 2382 } 2383 unlockEffectChains(effectChains); 2384 2385 mLastWriteTime = systemTime(); 2386 mInWrite = true; 2387 mBytesWritten += mixBufferSize; 2388 int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); 2389 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2390 mNumWrites++; 2391 mInWrite = false; 2392 mStandby = false; 2393 } else { 2394 unlockEffectChains(effectChains); 2395 usleep(sleepTime); 2396 } 2397 2398 // finally let go of removed track, without the lock held 2399 // since we can't guarantee the destructors won't acquire that 2400 // same lock. 2401 trackToRemove.clear(); 2402 activeTrack.clear(); 2403 2404 // Effect chains will be actually deleted here if they were removed from 2405 // mEffectChains list during mixing or effects processing 2406 effectChains.clear(); 2407 } 2408 2409 if (!mStandby) { 2410 mOutput->standby(); 2411 } 2412 2413 LOGV("DirectOutputThread %p exiting", this); 2414 return false; 2415} 2416 2417// getTrackName_l() must be called with ThreadBase::mLock held 2418int AudioFlinger::DirectOutputThread::getTrackName_l() 2419{ 2420 return 0; 2421} 2422 2423// deleteTrackName_l() must be called with ThreadBase::mLock held 2424void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2425{ 2426} 2427 2428// checkForNewParameters_l() must be called with ThreadBase::mLock held 2429bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2430{ 2431 bool reconfig = false; 2432 2433 while (!mNewParameters.isEmpty()) { 2434 status_t status = NO_ERROR; 2435 String8 keyValuePair = mNewParameters[0]; 2436 AudioParameter param = AudioParameter(keyValuePair); 2437 int value; 2438 2439 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2440 // do not accept frame count changes if tracks are open as the track buffer 2441 // size depends on frame count and correct behavior would not be garantied 2442 // if frame count is changed after track creation 2443 if (!mTracks.isEmpty()) { 2444 status = INVALID_OPERATION; 2445 } else { 2446 reconfig = true; 2447 } 2448 } 2449 if (status == NO_ERROR) { 2450 status = mOutput->setParameters(keyValuePair); 2451 if (!mStandby && status == INVALID_OPERATION) { 2452 mOutput->standby(); 2453 mStandby = true; 2454 mBytesWritten = 0; 2455 status = mOutput->setParameters(keyValuePair); 2456 } 2457 if (status == NO_ERROR && reconfig) { 2458 readOutputParameters(); 2459 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2460 } 2461 } 2462 2463 mNewParameters.removeAt(0); 2464 2465 mParamStatus = status; 2466 mParamCond.signal(); 2467 mWaitWorkCV.wait(mLock); 2468 } 2469 return reconfig; 2470} 2471 2472uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2473{ 2474 uint32_t time; 2475 if (AudioSystem::isLinearPCM(mFormat)) { 2476 time = (uint32_t)(mOutput->latency() * 1000) / 2; 2477 } else { 2478 time = 10000; 2479 } 2480 return time; 2481} 2482 2483uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2484{ 2485 uint32_t time; 2486 if (AudioSystem::isLinearPCM(mFormat)) { 2487 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2488 } else { 2489 time = 10000; 2490 } 2491 return time; 2492} 2493 2494uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2495{ 2496 uint32_t time; 2497 if (AudioSystem::isLinearPCM(mFormat)) { 2498 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2499 } else { 2500 time = 10000; 2501 } 2502 return time; 2503} 2504 2505 2506// ---------------------------------------------------------------------------- 2507 2508AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2509 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2510{ 2511 mType = PlaybackThread::DUPLICATING; 2512 addOutputTrack(mainThread); 2513} 2514 2515AudioFlinger::DuplicatingThread::~DuplicatingThread() 2516{ 2517 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2518 mOutputTracks[i]->destroy(); 2519 } 2520 mOutputTracks.clear(); 2521} 2522 2523bool AudioFlinger::DuplicatingThread::threadLoop() 2524{ 2525 Vector< sp<Track> > tracksToRemove; 2526 uint32_t mixerStatus = MIXER_IDLE; 2527 nsecs_t standbyTime = systemTime(); 2528 size_t mixBufferSize = mFrameCount*mFrameSize; 2529 SortedVector< sp<OutputTrack> > outputTracks; 2530 uint32_t writeFrames = 0; 2531 uint32_t activeSleepTime = activeSleepTimeUs(); 2532 uint32_t idleSleepTime = idleSleepTimeUs(); 2533 uint32_t sleepTime = idleSleepTime; 2534 Vector< sp<EffectChain> > effectChains; 2535 2536 while (!exitPending()) 2537 { 2538 processConfigEvents(); 2539 2540 mixerStatus = MIXER_IDLE; 2541 { // scope for the mLock 2542 2543 Mutex::Autolock _l(mLock); 2544 2545 if (checkForNewParameters_l()) { 2546 mixBufferSize = mFrameCount*mFrameSize; 2547 updateWaitTime(); 2548 activeSleepTime = activeSleepTimeUs(); 2549 idleSleepTime = idleSleepTimeUs(); 2550 } 2551 2552 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2553 2554 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2555 outputTracks.add(mOutputTracks[i]); 2556 } 2557 2558 // put audio hardware into standby after short delay 2559 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2560 mSuspended) { 2561 if (!mStandby) { 2562 for (size_t i = 0; i < outputTracks.size(); i++) { 2563 outputTracks[i]->stop(); 2564 } 2565 mStandby = true; 2566 mBytesWritten = 0; 2567 } 2568 2569 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2570 // we're about to wait, flush the binder command buffer 2571 IPCThreadState::self()->flushCommands(); 2572 outputTracks.clear(); 2573 2574 if (exitPending()) break; 2575 2576 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2577 mWaitWorkCV.wait(mLock); 2578 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2579 if (mMasterMute == false) { 2580 char value[PROPERTY_VALUE_MAX]; 2581 property_get("ro.audio.silent", value, "0"); 2582 if (atoi(value)) { 2583 LOGD("Silence is golden"); 2584 setMasterMute(true); 2585 } 2586 } 2587 2588 standbyTime = systemTime() + kStandbyTimeInNsecs; 2589 sleepTime = idleSleepTime; 2590 continue; 2591 } 2592 } 2593 2594 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2595 2596 // prevent any changes in effect chain list and in each effect chain 2597 // during mixing and effect process as the audio buffers could be deleted 2598 // or modified if an effect is created or deleted 2599 lockEffectChains_l(effectChains); 2600 } 2601 2602 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2603 // mix buffers... 2604 if (outputsReady(outputTracks)) { 2605 mAudioMixer->process(); 2606 } else { 2607 memset(mMixBuffer, 0, mixBufferSize); 2608 } 2609 sleepTime = 0; 2610 writeFrames = mFrameCount; 2611 } else { 2612 if (sleepTime == 0) { 2613 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2614 sleepTime = activeSleepTime; 2615 } else { 2616 sleepTime = idleSleepTime; 2617 } 2618 } else if (mBytesWritten != 0) { 2619 // flush remaining overflow buffers in output tracks 2620 for (size_t i = 0; i < outputTracks.size(); i++) { 2621 if (outputTracks[i]->isActive()) { 2622 sleepTime = 0; 2623 writeFrames = 0; 2624 memset(mMixBuffer, 0, mixBufferSize); 2625 break; 2626 } 2627 } 2628 } 2629 } 2630 2631 if (mSuspended) { 2632 sleepTime = suspendSleepTimeUs(); 2633 } 2634 // sleepTime == 0 means we must write to audio hardware 2635 if (sleepTime == 0) { 2636 for (size_t i = 0; i < effectChains.size(); i ++) { 2637 effectChains[i]->process_l(); 2638 } 2639 // enable changes in effect chain 2640 unlockEffectChains(effectChains); 2641 2642 standbyTime = systemTime() + kStandbyTimeInNsecs; 2643 for (size_t i = 0; i < outputTracks.size(); i++) { 2644 outputTracks[i]->write(mMixBuffer, writeFrames); 2645 } 2646 mStandby = false; 2647 mBytesWritten += mixBufferSize; 2648 } else { 2649 // enable changes in effect chain 2650 unlockEffectChains(effectChains); 2651 usleep(sleepTime); 2652 } 2653 2654 // finally let go of all our tracks, without the lock held 2655 // since we can't guarantee the destructors won't acquire that 2656 // same lock. 2657 tracksToRemove.clear(); 2658 outputTracks.clear(); 2659 2660 // Effect chains will be actually deleted here if they were removed from 2661 // mEffectChains list during mixing or effects processing 2662 effectChains.clear(); 2663 } 2664 2665 return false; 2666} 2667 2668void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2669{ 2670 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2671 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2672 this, 2673 mSampleRate, 2674 mFormat, 2675 mChannelCount, 2676 frameCount); 2677 if (outputTrack->cblk() != NULL) { 2678 thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); 2679 mOutputTracks.add(outputTrack); 2680 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2681 updateWaitTime(); 2682 } 2683} 2684 2685void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2686{ 2687 Mutex::Autolock _l(mLock); 2688 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2689 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2690 mOutputTracks[i]->destroy(); 2691 mOutputTracks.removeAt(i); 2692 updateWaitTime(); 2693 return; 2694 } 2695 } 2696 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2697} 2698 2699void AudioFlinger::DuplicatingThread::updateWaitTime() 2700{ 2701 mWaitTimeMs = UINT_MAX; 2702 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2703 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2704 if (strong != NULL) { 2705 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2706 if (waitTimeMs < mWaitTimeMs) { 2707 mWaitTimeMs = waitTimeMs; 2708 } 2709 } 2710 } 2711} 2712 2713 2714bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2715{ 2716 for (size_t i = 0; i < outputTracks.size(); i++) { 2717 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2718 if (thread == 0) { 2719 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2720 return false; 2721 } 2722 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2723 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2724 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2725 return false; 2726 } 2727 } 2728 return true; 2729} 2730 2731uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2732{ 2733 return (mWaitTimeMs * 1000) / 2; 2734} 2735 2736// ---------------------------------------------------------------------------- 2737 2738// TrackBase constructor must be called with AudioFlinger::mLock held 2739AudioFlinger::ThreadBase::TrackBase::TrackBase( 2740 const wp<ThreadBase>& thread, 2741 const sp<Client>& client, 2742 uint32_t sampleRate, 2743 int format, 2744 int channelCount, 2745 int frameCount, 2746 uint32_t flags, 2747 const sp<IMemory>& sharedBuffer, 2748 int sessionId) 2749 : RefBase(), 2750 mThread(thread), 2751 mClient(client), 2752 mCblk(0), 2753 mFrameCount(0), 2754 mState(IDLE), 2755 mClientTid(-1), 2756 mFormat(format), 2757 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2758 mSessionId(sessionId) 2759{ 2760 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2761 2762 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2763 size_t size = sizeof(audio_track_cblk_t); 2764 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2765 if (sharedBuffer == 0) { 2766 size += bufferSize; 2767 } 2768 2769 if (client != NULL) { 2770 mCblkMemory = client->heap()->allocate(size); 2771 if (mCblkMemory != 0) { 2772 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2773 if (mCblk) { // construct the shared structure in-place. 2774 new(mCblk) audio_track_cblk_t(); 2775 // clear all buffers 2776 mCblk->frameCount = frameCount; 2777 mCblk->sampleRate = sampleRate; 2778 mCblk->channelCount = (uint8_t)channelCount; 2779 if (sharedBuffer == 0) { 2780 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2781 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2782 // Force underrun condition to avoid false underrun callback until first data is 2783 // written to buffer 2784 mCblk->flags = CBLK_UNDERRUN_ON; 2785 } else { 2786 mBuffer = sharedBuffer->pointer(); 2787 } 2788 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2789 } 2790 } else { 2791 LOGE("not enough memory for AudioTrack size=%u", size); 2792 client->heap()->dump("AudioTrack"); 2793 return; 2794 } 2795 } else { 2796 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2797 if (mCblk) { // construct the shared structure in-place. 2798 new(mCblk) audio_track_cblk_t(); 2799 // clear all buffers 2800 mCblk->frameCount = frameCount; 2801 mCblk->sampleRate = sampleRate; 2802 mCblk->channelCount = (uint8_t)channelCount; 2803 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2804 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2805 // Force underrun condition to avoid false underrun callback until first data is 2806 // written to buffer 2807 mCblk->flags = CBLK_UNDERRUN_ON; 2808 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2809 } 2810 } 2811} 2812 2813AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2814{ 2815 if (mCblk) { 2816 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2817 if (mClient == NULL) { 2818 delete mCblk; 2819 } 2820 } 2821 mCblkMemory.clear(); // and free the shared memory 2822 if (mClient != NULL) { 2823 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2824 mClient.clear(); 2825 } 2826} 2827 2828void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2829{ 2830 buffer->raw = 0; 2831 mFrameCount = buffer->frameCount; 2832 step(); 2833 buffer->frameCount = 0; 2834} 2835 2836bool AudioFlinger::ThreadBase::TrackBase::step() { 2837 bool result; 2838 audio_track_cblk_t* cblk = this->cblk(); 2839 2840 result = cblk->stepServer(mFrameCount); 2841 if (!result) { 2842 LOGV("stepServer failed acquiring cblk mutex"); 2843 mFlags |= STEPSERVER_FAILED; 2844 } 2845 return result; 2846} 2847 2848void AudioFlinger::ThreadBase::TrackBase::reset() { 2849 audio_track_cblk_t* cblk = this->cblk(); 2850 2851 cblk->user = 0; 2852 cblk->server = 0; 2853 cblk->userBase = 0; 2854 cblk->serverBase = 0; 2855 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2856 LOGV("TrackBase::reset"); 2857} 2858 2859sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2860{ 2861 return mCblkMemory; 2862} 2863 2864int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2865 return (int)mCblk->sampleRate; 2866} 2867 2868int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2869 return (int)mCblk->channelCount; 2870} 2871 2872void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2873 audio_track_cblk_t* cblk = this->cblk(); 2874 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2875 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2876 2877 // Check validity of returned pointer in case the track control block would have been corrupted. 2878 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2879 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2880 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2881 server %d, serverBase %d, user %d, userBase %d, channelCount %d", 2882 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2883 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); 2884 return 0; 2885 } 2886 2887 return bufferStart; 2888} 2889 2890// ---------------------------------------------------------------------------- 2891 2892// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2893AudioFlinger::PlaybackThread::Track::Track( 2894 const wp<ThreadBase>& thread, 2895 const sp<Client>& client, 2896 int streamType, 2897 uint32_t sampleRate, 2898 int format, 2899 int channelCount, 2900 int frameCount, 2901 const sp<IMemory>& sharedBuffer, 2902 int sessionId) 2903 : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), 2904 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0) 2905{ 2906 if (mCblk != NULL) { 2907 sp<ThreadBase> baseThread = thread.promote(); 2908 if (baseThread != 0) { 2909 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2910 mName = playbackThread->getTrackName_l(); 2911 mMainBuffer = playbackThread->mixBuffer(); 2912 } 2913 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2914 if (mName < 0) { 2915 LOGE("no more track names available"); 2916 } 2917 mVolume[0] = 1.0f; 2918 mVolume[1] = 1.0f; 2919 mStreamType = streamType; 2920 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2921 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2922 mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); 2923 } 2924} 2925 2926AudioFlinger::PlaybackThread::Track::~Track() 2927{ 2928 LOGV("PlaybackThread::Track destructor"); 2929 sp<ThreadBase> thread = mThread.promote(); 2930 if (thread != 0) { 2931 Mutex::Autolock _l(thread->mLock); 2932 mState = TERMINATED; 2933 } 2934} 2935 2936void AudioFlinger::PlaybackThread::Track::destroy() 2937{ 2938 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2939 // by removing it from mTracks vector, so there is a risk that this Tracks's 2940 // desctructor is called. As the destructor needs to lock mLock, 2941 // we must acquire a strong reference on this Track before locking mLock 2942 // here so that the destructor is called only when exiting this function. 2943 // On the other hand, as long as Track::destroy() is only called by 2944 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2945 // this Track with its member mTrack. 2946 sp<Track> keep(this); 2947 { // scope for mLock 2948 sp<ThreadBase> thread = mThread.promote(); 2949 if (thread != 0) { 2950 if (!isOutputTrack()) { 2951 if (mState == ACTIVE || mState == RESUMING) { 2952 AudioSystem::stopOutput(thread->id(), 2953 (AudioSystem::stream_type)mStreamType, 2954 mSessionId); 2955 } 2956 AudioSystem::releaseOutput(thread->id()); 2957 } 2958 Mutex::Autolock _l(thread->mLock); 2959 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2960 playbackThread->destroyTrack_l(this); 2961 } 2962 } 2963} 2964 2965void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2966{ 2967 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", 2968 mName - AudioMixer::TRACK0, 2969 (mClient == NULL) ? getpid() : mClient->pid(), 2970 mStreamType, 2971 mFormat, 2972 mCblk->channelCount, 2973 mSessionId, 2974 mFrameCount, 2975 mState, 2976 mMute, 2977 mFillingUpStatus, 2978 mCblk->sampleRate, 2979 mCblk->volume[0], 2980 mCblk->volume[1], 2981 mCblk->server, 2982 mCblk->user, 2983 (int)mMainBuffer, 2984 (int)mAuxBuffer); 2985} 2986 2987status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 2988{ 2989 audio_track_cblk_t* cblk = this->cblk(); 2990 uint32_t framesReady; 2991 uint32_t framesReq = buffer->frameCount; 2992 2993 // Check if last stepServer failed, try to step now 2994 if (mFlags & TrackBase::STEPSERVER_FAILED) { 2995 if (!step()) goto getNextBuffer_exit; 2996 LOGV("stepServer recovered"); 2997 mFlags &= ~TrackBase::STEPSERVER_FAILED; 2998 } 2999 3000 framesReady = cblk->framesReady(); 3001 3002 if (LIKELY(framesReady)) { 3003 uint32_t s = cblk->server; 3004 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3005 3006 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3007 if (framesReq > framesReady) { 3008 framesReq = framesReady; 3009 } 3010 if (s + framesReq > bufferEnd) { 3011 framesReq = bufferEnd - s; 3012 } 3013 3014 buffer->raw = getBuffer(s, framesReq); 3015 if (buffer->raw == 0) goto getNextBuffer_exit; 3016 3017 buffer->frameCount = framesReq; 3018 return NO_ERROR; 3019 } 3020 3021getNextBuffer_exit: 3022 buffer->raw = 0; 3023 buffer->frameCount = 0; 3024 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3025 return NOT_ENOUGH_DATA; 3026} 3027 3028bool AudioFlinger::PlaybackThread::Track::isReady() const { 3029 if (mFillingUpStatus != FS_FILLING) return true; 3030 3031 if (mCblk->framesReady() >= mCblk->frameCount || 3032 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3033 mFillingUpStatus = FS_FILLED; 3034 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3035 return true; 3036 } 3037 return false; 3038} 3039 3040status_t AudioFlinger::PlaybackThread::Track::start() 3041{ 3042 status_t status = NO_ERROR; 3043 LOGV("start(%d), calling thread %d session %d", 3044 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3045 sp<ThreadBase> thread = mThread.promote(); 3046 if (thread != 0) { 3047 Mutex::Autolock _l(thread->mLock); 3048 int state = mState; 3049 // here the track could be either new, or restarted 3050 // in both cases "unstop" the track 3051 if (mState == PAUSED) { 3052 mState = TrackBase::RESUMING; 3053 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3054 } else { 3055 mState = TrackBase::ACTIVE; 3056 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3057 } 3058 3059 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3060 thread->mLock.unlock(); 3061 status = AudioSystem::startOutput(thread->id(), 3062 (AudioSystem::stream_type)mStreamType, 3063 mSessionId); 3064 thread->mLock.lock(); 3065 } 3066 if (status == NO_ERROR) { 3067 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3068 playbackThread->addTrack_l(this); 3069 } else { 3070 mState = state; 3071 } 3072 } else { 3073 status = BAD_VALUE; 3074 } 3075 return status; 3076} 3077 3078void AudioFlinger::PlaybackThread::Track::stop() 3079{ 3080 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3081 sp<ThreadBase> thread = mThread.promote(); 3082 if (thread != 0) { 3083 Mutex::Autolock _l(thread->mLock); 3084 int state = mState; 3085 if (mState > STOPPED) { 3086 mState = STOPPED; 3087 // If the track is not active (PAUSED and buffers full), flush buffers 3088 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3089 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3090 reset(); 3091 } 3092 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3093 } 3094 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3095 thread->mLock.unlock(); 3096 AudioSystem::stopOutput(thread->id(), 3097 (AudioSystem::stream_type)mStreamType, 3098 mSessionId); 3099 thread->mLock.lock(); 3100 } 3101 } 3102} 3103 3104void AudioFlinger::PlaybackThread::Track::pause() 3105{ 3106 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3107 sp<ThreadBase> thread = mThread.promote(); 3108 if (thread != 0) { 3109 Mutex::Autolock _l(thread->mLock); 3110 if (mState == ACTIVE || mState == RESUMING) { 3111 mState = PAUSING; 3112 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3113 if (!isOutputTrack()) { 3114 thread->mLock.unlock(); 3115 AudioSystem::stopOutput(thread->id(), 3116 (AudioSystem::stream_type)mStreamType, 3117 mSessionId); 3118 thread->mLock.lock(); 3119 } 3120 } 3121 } 3122} 3123 3124void AudioFlinger::PlaybackThread::Track::flush() 3125{ 3126 LOGV("flush(%d)", mName); 3127 sp<ThreadBase> thread = mThread.promote(); 3128 if (thread != 0) { 3129 Mutex::Autolock _l(thread->mLock); 3130 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3131 return; 3132 } 3133 // No point remaining in PAUSED state after a flush => go to 3134 // STOPPED state 3135 mState = STOPPED; 3136 3137 mCblk->lock.lock(); 3138 // NOTE: reset() will reset cblk->user and cblk->server with 3139 // the risk that at the same time, the AudioMixer is trying to read 3140 // data. In this case, getNextBuffer() would return a NULL pointer 3141 // as audio buffer => the AudioMixer code MUST always test that pointer 3142 // returned by getNextBuffer() is not NULL! 3143 reset(); 3144 mCblk->lock.unlock(); 3145 } 3146} 3147 3148void AudioFlinger::PlaybackThread::Track::reset() 3149{ 3150 // Do not reset twice to avoid discarding data written just after a flush and before 3151 // the audioflinger thread detects the track is stopped. 3152 if (!mResetDone) { 3153 TrackBase::reset(); 3154 // Force underrun condition to avoid false underrun callback until first data is 3155 // written to buffer 3156 mCblk->flags |= CBLK_UNDERRUN_ON; 3157 mCblk->flags &= ~CBLK_FORCEREADY_MSK; 3158 mFillingUpStatus = FS_FILLING; 3159 mResetDone = true; 3160 } 3161} 3162 3163void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3164{ 3165 mMute = muted; 3166} 3167 3168void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3169{ 3170 mVolume[0] = left; 3171 mVolume[1] = right; 3172} 3173 3174status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3175{ 3176 status_t status = DEAD_OBJECT; 3177 sp<ThreadBase> thread = mThread.promote(); 3178 if (thread != 0) { 3179 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3180 status = playbackThread->attachAuxEffect(this, EffectId); 3181 } 3182 return status; 3183} 3184 3185void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3186{ 3187 mAuxEffectId = EffectId; 3188 mAuxBuffer = buffer; 3189} 3190 3191// ---------------------------------------------------------------------------- 3192 3193// RecordTrack constructor must be called with AudioFlinger::mLock held 3194AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3195 const wp<ThreadBase>& thread, 3196 const sp<Client>& client, 3197 uint32_t sampleRate, 3198 int format, 3199 int channelCount, 3200 int frameCount, 3201 uint32_t flags, 3202 int sessionId) 3203 : TrackBase(thread, client, sampleRate, format, 3204 channelCount, frameCount, flags, 0, sessionId), 3205 mOverflow(false) 3206{ 3207 if (mCblk != NULL) { 3208 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3209 if (format == AudioSystem::PCM_16_BIT) { 3210 mCblk->frameSize = channelCount * sizeof(int16_t); 3211 } else if (format == AudioSystem::PCM_8_BIT) { 3212 mCblk->frameSize = channelCount * sizeof(int8_t); 3213 } else { 3214 mCblk->frameSize = sizeof(int8_t); 3215 } 3216 } 3217} 3218 3219AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3220{ 3221 sp<ThreadBase> thread = mThread.promote(); 3222 if (thread != 0) { 3223 AudioSystem::releaseInput(thread->id()); 3224 } 3225} 3226 3227status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3228{ 3229 audio_track_cblk_t* cblk = this->cblk(); 3230 uint32_t framesAvail; 3231 uint32_t framesReq = buffer->frameCount; 3232 3233 // Check if last stepServer failed, try to step now 3234 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3235 if (!step()) goto getNextBuffer_exit; 3236 LOGV("stepServer recovered"); 3237 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3238 } 3239 3240 framesAvail = cblk->framesAvailable_l(); 3241 3242 if (LIKELY(framesAvail)) { 3243 uint32_t s = cblk->server; 3244 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3245 3246 if (framesReq > framesAvail) { 3247 framesReq = framesAvail; 3248 } 3249 if (s + framesReq > bufferEnd) { 3250 framesReq = bufferEnd - s; 3251 } 3252 3253 buffer->raw = getBuffer(s, framesReq); 3254 if (buffer->raw == 0) goto getNextBuffer_exit; 3255 3256 buffer->frameCount = framesReq; 3257 return NO_ERROR; 3258 } 3259 3260getNextBuffer_exit: 3261 buffer->raw = 0; 3262 buffer->frameCount = 0; 3263 return NOT_ENOUGH_DATA; 3264} 3265 3266status_t AudioFlinger::RecordThread::RecordTrack::start() 3267{ 3268 sp<ThreadBase> thread = mThread.promote(); 3269 if (thread != 0) { 3270 RecordThread *recordThread = (RecordThread *)thread.get(); 3271 return recordThread->start(this); 3272 } else { 3273 return BAD_VALUE; 3274 } 3275} 3276 3277void AudioFlinger::RecordThread::RecordTrack::stop() 3278{ 3279 sp<ThreadBase> thread = mThread.promote(); 3280 if (thread != 0) { 3281 RecordThread *recordThread = (RecordThread *)thread.get(); 3282 recordThread->stop(this); 3283 TrackBase::reset(); 3284 // Force overerrun condition to avoid false overrun callback until first data is 3285 // read from buffer 3286 mCblk->flags |= CBLK_UNDERRUN_ON; 3287 } 3288} 3289 3290void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3291{ 3292 snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", 3293 (mClient == NULL) ? getpid() : mClient->pid(), 3294 mFormat, 3295 mCblk->channelCount, 3296 mSessionId, 3297 mFrameCount, 3298 mState, 3299 mCblk->sampleRate, 3300 mCblk->server, 3301 mCblk->user); 3302} 3303 3304 3305// ---------------------------------------------------------------------------- 3306 3307AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3308 const wp<ThreadBase>& thread, 3309 DuplicatingThread *sourceThread, 3310 uint32_t sampleRate, 3311 int format, 3312 int channelCount, 3313 int frameCount) 3314 : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), 3315 mActive(false), mSourceThread(sourceThread) 3316{ 3317 3318 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3319 if (mCblk != NULL) { 3320 mCblk->flags |= CBLK_DIRECTION_OUT; 3321 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3322 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3323 mOutBuffer.frameCount = 0; 3324 playbackThread->mTracks.add(this); 3325 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", 3326 mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); 3327 } else { 3328 LOGW("Error creating output track on thread %p", playbackThread); 3329 } 3330} 3331 3332AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3333{ 3334 clearBufferQueue(); 3335} 3336 3337status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3338{ 3339 status_t status = Track::start(); 3340 if (status != NO_ERROR) { 3341 return status; 3342 } 3343 3344 mActive = true; 3345 mRetryCount = 127; 3346 return status; 3347} 3348 3349void AudioFlinger::PlaybackThread::OutputTrack::stop() 3350{ 3351 Track::stop(); 3352 clearBufferQueue(); 3353 mOutBuffer.frameCount = 0; 3354 mActive = false; 3355} 3356 3357bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3358{ 3359 Buffer *pInBuffer; 3360 Buffer inBuffer; 3361 uint32_t channelCount = mCblk->channelCount; 3362 bool outputBufferFull = false; 3363 inBuffer.frameCount = frames; 3364 inBuffer.i16 = data; 3365 3366 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3367 3368 if (!mActive && frames != 0) { 3369 start(); 3370 sp<ThreadBase> thread = mThread.promote(); 3371 if (thread != 0) { 3372 MixerThread *mixerThread = (MixerThread *)thread.get(); 3373 if (mCblk->frameCount > frames){ 3374 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3375 uint32_t startFrames = (mCblk->frameCount - frames); 3376 pInBuffer = new Buffer; 3377 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3378 pInBuffer->frameCount = startFrames; 3379 pInBuffer->i16 = pInBuffer->mBuffer; 3380 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3381 mBufferQueue.add(pInBuffer); 3382 } else { 3383 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3384 } 3385 } 3386 } 3387 } 3388 3389 while (waitTimeLeftMs) { 3390 // First write pending buffers, then new data 3391 if (mBufferQueue.size()) { 3392 pInBuffer = mBufferQueue.itemAt(0); 3393 } else { 3394 pInBuffer = &inBuffer; 3395 } 3396 3397 if (pInBuffer->frameCount == 0) { 3398 break; 3399 } 3400 3401 if (mOutBuffer.frameCount == 0) { 3402 mOutBuffer.frameCount = pInBuffer->frameCount; 3403 nsecs_t startTime = systemTime(); 3404 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3405 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3406 outputBufferFull = true; 3407 break; 3408 } 3409 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3410 if (waitTimeLeftMs >= waitTimeMs) { 3411 waitTimeLeftMs -= waitTimeMs; 3412 } else { 3413 waitTimeLeftMs = 0; 3414 } 3415 } 3416 3417 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3418 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3419 mCblk->stepUser(outFrames); 3420 pInBuffer->frameCount -= outFrames; 3421 pInBuffer->i16 += outFrames * channelCount; 3422 mOutBuffer.frameCount -= outFrames; 3423 mOutBuffer.i16 += outFrames * channelCount; 3424 3425 if (pInBuffer->frameCount == 0) { 3426 if (mBufferQueue.size()) { 3427 mBufferQueue.removeAt(0); 3428 delete [] pInBuffer->mBuffer; 3429 delete pInBuffer; 3430 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3431 } else { 3432 break; 3433 } 3434 } 3435 } 3436 3437 // If we could not write all frames, allocate a buffer and queue it for next time. 3438 if (inBuffer.frameCount) { 3439 sp<ThreadBase> thread = mThread.promote(); 3440 if (thread != 0 && !thread->standby()) { 3441 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3442 pInBuffer = new Buffer; 3443 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3444 pInBuffer->frameCount = inBuffer.frameCount; 3445 pInBuffer->i16 = pInBuffer->mBuffer; 3446 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3447 mBufferQueue.add(pInBuffer); 3448 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3449 } else { 3450 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3451 } 3452 } 3453 } 3454 3455 // Calling write() with a 0 length buffer, means that no more data will be written: 3456 // If no more buffers are pending, fill output track buffer to make sure it is started 3457 // by output mixer. 3458 if (frames == 0 && mBufferQueue.size() == 0) { 3459 if (mCblk->user < mCblk->frameCount) { 3460 frames = mCblk->frameCount - mCblk->user; 3461 pInBuffer = new Buffer; 3462 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3463 pInBuffer->frameCount = frames; 3464 pInBuffer->i16 = pInBuffer->mBuffer; 3465 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3466 mBufferQueue.add(pInBuffer); 3467 } else if (mActive) { 3468 stop(); 3469 } 3470 } 3471 3472 return outputBufferFull; 3473} 3474 3475status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3476{ 3477 int active; 3478 status_t result; 3479 audio_track_cblk_t* cblk = mCblk; 3480 uint32_t framesReq = buffer->frameCount; 3481 3482// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3483 buffer->frameCount = 0; 3484 3485 uint32_t framesAvail = cblk->framesAvailable(); 3486 3487 3488 if (framesAvail == 0) { 3489 Mutex::Autolock _l(cblk->lock); 3490 goto start_loop_here; 3491 while (framesAvail == 0) { 3492 active = mActive; 3493 if (UNLIKELY(!active)) { 3494 LOGV("Not active and NO_MORE_BUFFERS"); 3495 return AudioTrack::NO_MORE_BUFFERS; 3496 } 3497 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3498 if (result != NO_ERROR) { 3499 return AudioTrack::NO_MORE_BUFFERS; 3500 } 3501 // read the server count again 3502 start_loop_here: 3503 framesAvail = cblk->framesAvailable_l(); 3504 } 3505 } 3506 3507// if (framesAvail < framesReq) { 3508// return AudioTrack::NO_MORE_BUFFERS; 3509// } 3510 3511 if (framesReq > framesAvail) { 3512 framesReq = framesAvail; 3513 } 3514 3515 uint32_t u = cblk->user; 3516 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3517 3518 if (u + framesReq > bufferEnd) { 3519 framesReq = bufferEnd - u; 3520 } 3521 3522 buffer->frameCount = framesReq; 3523 buffer->raw = (void *)cblk->buffer(u); 3524 return NO_ERROR; 3525} 3526 3527 3528void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3529{ 3530 size_t size = mBufferQueue.size(); 3531 Buffer *pBuffer; 3532 3533 for (size_t i = 0; i < size; i++) { 3534 pBuffer = mBufferQueue.itemAt(i); 3535 delete [] pBuffer->mBuffer; 3536 delete pBuffer; 3537 } 3538 mBufferQueue.clear(); 3539} 3540 3541// ---------------------------------------------------------------------------- 3542 3543AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3544 : RefBase(), 3545 mAudioFlinger(audioFlinger), 3546 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3547 mPid(pid) 3548{ 3549 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3550} 3551 3552// Client destructor must be called with AudioFlinger::mLock held 3553AudioFlinger::Client::~Client() 3554{ 3555 mAudioFlinger->removeClient_l(mPid); 3556} 3557 3558const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3559{ 3560 return mMemoryDealer; 3561} 3562 3563// ---------------------------------------------------------------------------- 3564 3565AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3566 const sp<IAudioFlingerClient>& client, 3567 pid_t pid) 3568 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3569{ 3570} 3571 3572AudioFlinger::NotificationClient::~NotificationClient() 3573{ 3574 mClient.clear(); 3575} 3576 3577void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3578{ 3579 sp<NotificationClient> keep(this); 3580 { 3581 mAudioFlinger->removeNotificationClient(mPid); 3582 } 3583} 3584 3585// ---------------------------------------------------------------------------- 3586 3587AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3588 : BnAudioTrack(), 3589 mTrack(track) 3590{ 3591} 3592 3593AudioFlinger::TrackHandle::~TrackHandle() { 3594 // just stop the track on deletion, associated resources 3595 // will be freed from the main thread once all pending buffers have 3596 // been played. Unless it's not in the active track list, in which 3597 // case we free everything now... 3598 mTrack->destroy(); 3599} 3600 3601status_t AudioFlinger::TrackHandle::start() { 3602 return mTrack->start(); 3603} 3604 3605void AudioFlinger::TrackHandle::stop() { 3606 mTrack->stop(); 3607} 3608 3609void AudioFlinger::TrackHandle::flush() { 3610 mTrack->flush(); 3611} 3612 3613void AudioFlinger::TrackHandle::mute(bool e) { 3614 mTrack->mute(e); 3615} 3616 3617void AudioFlinger::TrackHandle::pause() { 3618 mTrack->pause(); 3619} 3620 3621void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3622 mTrack->setVolume(left, right); 3623} 3624 3625sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3626 return mTrack->getCblk(); 3627} 3628 3629status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3630{ 3631 return mTrack->attachAuxEffect(EffectId); 3632} 3633 3634status_t AudioFlinger::TrackHandle::onTransact( 3635 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3636{ 3637 return BnAudioTrack::onTransact(code, data, reply, flags); 3638} 3639 3640// ---------------------------------------------------------------------------- 3641 3642sp<IAudioRecord> AudioFlinger::openRecord( 3643 pid_t pid, 3644 int input, 3645 uint32_t sampleRate, 3646 int format, 3647 int channelCount, 3648 int frameCount, 3649 uint32_t flags, 3650 int *sessionId, 3651 status_t *status) 3652{ 3653 sp<RecordThread::RecordTrack> recordTrack; 3654 sp<RecordHandle> recordHandle; 3655 sp<Client> client; 3656 wp<Client> wclient; 3657 status_t lStatus; 3658 RecordThread *thread; 3659 size_t inFrameCount; 3660 int lSessionId; 3661 3662 // check calling permissions 3663 if (!recordingAllowed()) { 3664 lStatus = PERMISSION_DENIED; 3665 goto Exit; 3666 } 3667 3668 // add client to list 3669 { // scope for mLock 3670 Mutex::Autolock _l(mLock); 3671 thread = checkRecordThread_l(input); 3672 if (thread == NULL) { 3673 lStatus = BAD_VALUE; 3674 goto Exit; 3675 } 3676 3677 wclient = mClients.valueFor(pid); 3678 if (wclient != NULL) { 3679 client = wclient.promote(); 3680 } else { 3681 client = new Client(this, pid); 3682 mClients.add(pid, client); 3683 } 3684 3685 // If no audio session id is provided, create one here 3686 if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 3687 lSessionId = *sessionId; 3688 } else { 3689 lSessionId = nextUniqueId(); 3690 if (sessionId != NULL) { 3691 *sessionId = lSessionId; 3692 } 3693 } 3694 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3695 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3696 format, channelCount, frameCount, flags, lSessionId); 3697 } 3698 if (recordTrack->getCblk() == NULL) { 3699 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3700 // destructor is called by the TrackBase destructor with mLock held 3701 client.clear(); 3702 recordTrack.clear(); 3703 lStatus = NO_MEMORY; 3704 goto Exit; 3705 } 3706 3707 // return to handle to client 3708 recordHandle = new RecordHandle(recordTrack); 3709 lStatus = NO_ERROR; 3710 3711Exit: 3712 if (status) { 3713 *status = lStatus; 3714 } 3715 return recordHandle; 3716} 3717 3718// ---------------------------------------------------------------------------- 3719 3720AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3721 : BnAudioRecord(), 3722 mRecordTrack(recordTrack) 3723{ 3724} 3725 3726AudioFlinger::RecordHandle::~RecordHandle() { 3727 stop(); 3728} 3729 3730status_t AudioFlinger::RecordHandle::start() { 3731 LOGV("RecordHandle::start()"); 3732 return mRecordTrack->start(); 3733} 3734 3735void AudioFlinger::RecordHandle::stop() { 3736 LOGV("RecordHandle::stop()"); 3737 mRecordTrack->stop(); 3738} 3739 3740sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3741 return mRecordTrack->getCblk(); 3742} 3743 3744status_t AudioFlinger::RecordHandle::onTransact( 3745 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3746{ 3747 return BnAudioRecord::onTransact(code, data, reply, flags); 3748} 3749 3750// ---------------------------------------------------------------------------- 3751 3752AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : 3753 ThreadBase(audioFlinger, id), 3754 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3755{ 3756 mReqChannelCount = AudioSystem::popCount(channels); 3757 mReqSampleRate = sampleRate; 3758 readInputParameters(); 3759} 3760 3761 3762AudioFlinger::RecordThread::~RecordThread() 3763{ 3764 delete[] mRsmpInBuffer; 3765 if (mResampler != 0) { 3766 delete mResampler; 3767 delete[] mRsmpOutBuffer; 3768 } 3769} 3770 3771void AudioFlinger::RecordThread::onFirstRef() 3772{ 3773 const size_t SIZE = 256; 3774 char buffer[SIZE]; 3775 3776 snprintf(buffer, SIZE, "Record Thread %p", this); 3777 3778 run(buffer, PRIORITY_URGENT_AUDIO); 3779} 3780 3781bool AudioFlinger::RecordThread::threadLoop() 3782{ 3783 AudioBufferProvider::Buffer buffer; 3784 sp<RecordTrack> activeTrack; 3785 3786 // start recording 3787 while (!exitPending()) { 3788 3789 processConfigEvents(); 3790 3791 { // scope for mLock 3792 Mutex::Autolock _l(mLock); 3793 checkForNewParameters_l(); 3794 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3795 if (!mStandby) { 3796 mInput->standby(); 3797 mStandby = true; 3798 } 3799 3800 if (exitPending()) break; 3801 3802 LOGV("RecordThread: loop stopping"); 3803 // go to sleep 3804 mWaitWorkCV.wait(mLock); 3805 LOGV("RecordThread: loop starting"); 3806 continue; 3807 } 3808 if (mActiveTrack != 0) { 3809 if (mActiveTrack->mState == TrackBase::PAUSING) { 3810 if (!mStandby) { 3811 mInput->standby(); 3812 mStandby = true; 3813 } 3814 mActiveTrack.clear(); 3815 mStartStopCond.broadcast(); 3816 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3817 if (mReqChannelCount != mActiveTrack->channelCount()) { 3818 mActiveTrack.clear(); 3819 mStartStopCond.broadcast(); 3820 } else if (mBytesRead != 0) { 3821 // record start succeeds only if first read from audio input 3822 // succeeds 3823 if (mBytesRead > 0) { 3824 mActiveTrack->mState = TrackBase::ACTIVE; 3825 } else { 3826 mActiveTrack.clear(); 3827 } 3828 mStartStopCond.broadcast(); 3829 } 3830 mStandby = false; 3831 } 3832 } 3833 } 3834 3835 if (mActiveTrack != 0) { 3836 if (mActiveTrack->mState != TrackBase::ACTIVE && 3837 mActiveTrack->mState != TrackBase::RESUMING) { 3838 usleep(5000); 3839 continue; 3840 } 3841 buffer.frameCount = mFrameCount; 3842 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3843 size_t framesOut = buffer.frameCount; 3844 if (mResampler == 0) { 3845 // no resampling 3846 while (framesOut) { 3847 size_t framesIn = mFrameCount - mRsmpInIndex; 3848 if (framesIn) { 3849 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3850 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3851 if (framesIn > framesOut) 3852 framesIn = framesOut; 3853 mRsmpInIndex += framesIn; 3854 framesOut -= framesIn; 3855 if ((int)mChannelCount == mReqChannelCount || 3856 mFormat != AudioSystem::PCM_16_BIT) { 3857 memcpy(dst, src, framesIn * mFrameSize); 3858 } else { 3859 int16_t *src16 = (int16_t *)src; 3860 int16_t *dst16 = (int16_t *)dst; 3861 if (mChannelCount == 1) { 3862 while (framesIn--) { 3863 *dst16++ = *src16; 3864 *dst16++ = *src16++; 3865 } 3866 } else { 3867 while (framesIn--) { 3868 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3869 src16 += 2; 3870 } 3871 } 3872 } 3873 } 3874 if (framesOut && mFrameCount == mRsmpInIndex) { 3875 if (framesOut == mFrameCount && 3876 ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { 3877 mBytesRead = mInput->read(buffer.raw, mInputBytes); 3878 framesOut = 0; 3879 } else { 3880 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 3881 mRsmpInIndex = 0; 3882 } 3883 if (mBytesRead < 0) { 3884 LOGE("Error reading audio input"); 3885 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3886 // Force input into standby so that it tries to 3887 // recover at next read attempt 3888 mInput->standby(); 3889 usleep(5000); 3890 } 3891 mRsmpInIndex = mFrameCount; 3892 framesOut = 0; 3893 buffer.frameCount = 0; 3894 } 3895 } 3896 } 3897 } else { 3898 // resampling 3899 3900 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3901 // alter output frame count as if we were expecting stereo samples 3902 if (mChannelCount == 1 && mReqChannelCount == 1) { 3903 framesOut >>= 1; 3904 } 3905 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3906 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3907 // are 32 bit aligned which should be always true. 3908 if (mChannelCount == 2 && mReqChannelCount == 1) { 3909 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3910 // the resampler always outputs stereo samples: do post stereo to mono conversion 3911 int16_t *src = (int16_t *)mRsmpOutBuffer; 3912 int16_t *dst = buffer.i16; 3913 while (framesOut--) { 3914 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3915 src += 2; 3916 } 3917 } else { 3918 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3919 } 3920 3921 } 3922 mActiveTrack->releaseBuffer(&buffer); 3923 mActiveTrack->overflow(); 3924 } 3925 // client isn't retrieving buffers fast enough 3926 else { 3927 if (!mActiveTrack->setOverflow()) 3928 LOGW("RecordThread: buffer overflow"); 3929 // Release the processor for a while before asking for a new buffer. 3930 // This will give the application more chance to read from the buffer and 3931 // clear the overflow. 3932 usleep(5000); 3933 } 3934 } 3935 } 3936 3937 if (!mStandby) { 3938 mInput->standby(); 3939 } 3940 mActiveTrack.clear(); 3941 3942 mStartStopCond.broadcast(); 3943 3944 LOGV("RecordThread %p exiting", this); 3945 return false; 3946} 3947 3948status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3949{ 3950 LOGV("RecordThread::start"); 3951 sp <ThreadBase> strongMe = this; 3952 status_t status = NO_ERROR; 3953 { 3954 AutoMutex lock(&mLock); 3955 if (mActiveTrack != 0) { 3956 if (recordTrack != mActiveTrack.get()) { 3957 status = -EBUSY; 3958 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3959 mActiveTrack->mState = TrackBase::ACTIVE; 3960 } 3961 return status; 3962 } 3963 3964 recordTrack->mState = TrackBase::IDLE; 3965 mActiveTrack = recordTrack; 3966 mLock.unlock(); 3967 status_t status = AudioSystem::startInput(mId); 3968 mLock.lock(); 3969 if (status != NO_ERROR) { 3970 mActiveTrack.clear(); 3971 return status; 3972 } 3973 mActiveTrack->mState = TrackBase::RESUMING; 3974 mRsmpInIndex = mFrameCount; 3975 mBytesRead = 0; 3976 // signal thread to start 3977 LOGV("Signal record thread"); 3978 mWaitWorkCV.signal(); 3979 // do not wait for mStartStopCond if exiting 3980 if (mExiting) { 3981 mActiveTrack.clear(); 3982 status = INVALID_OPERATION; 3983 goto startError; 3984 } 3985 mStartStopCond.wait(mLock); 3986 if (mActiveTrack == 0) { 3987 LOGV("Record failed to start"); 3988 status = BAD_VALUE; 3989 goto startError; 3990 } 3991 LOGV("Record started OK"); 3992 return status; 3993 } 3994startError: 3995 AudioSystem::stopInput(mId); 3996 return status; 3997} 3998 3999void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4000 LOGV("RecordThread::stop"); 4001 sp <ThreadBase> strongMe = this; 4002 { 4003 AutoMutex lock(&mLock); 4004 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4005 mActiveTrack->mState = TrackBase::PAUSING; 4006 // do not wait for mStartStopCond if exiting 4007 if (mExiting) { 4008 return; 4009 } 4010 mStartStopCond.wait(mLock); 4011 // if we have been restarted, recordTrack == mActiveTrack.get() here 4012 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4013 mLock.unlock(); 4014 AudioSystem::stopInput(mId); 4015 mLock.lock(); 4016 LOGV("Record stopped OK"); 4017 } 4018 } 4019 } 4020} 4021 4022status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4023{ 4024 const size_t SIZE = 256; 4025 char buffer[SIZE]; 4026 String8 result; 4027 pid_t pid = 0; 4028 4029 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4030 result.append(buffer); 4031 4032 if (mActiveTrack != 0) { 4033 result.append("Active Track:\n"); 4034 result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); 4035 mActiveTrack->dump(buffer, SIZE); 4036 result.append(buffer); 4037 4038 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4039 result.append(buffer); 4040 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4041 result.append(buffer); 4042 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4043 result.append(buffer); 4044 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4045 result.append(buffer); 4046 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4047 result.append(buffer); 4048 4049 4050 } else { 4051 result.append("No record client\n"); 4052 } 4053 write(fd, result.string(), result.size()); 4054 4055 dumpBase(fd, args); 4056 4057 return NO_ERROR; 4058} 4059 4060status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4061{ 4062 size_t framesReq = buffer->frameCount; 4063 size_t framesReady = mFrameCount - mRsmpInIndex; 4064 int channelCount; 4065 4066 if (framesReady == 0) { 4067 mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); 4068 if (mBytesRead < 0) { 4069 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4070 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4071 // Force input into standby so that it tries to 4072 // recover at next read attempt 4073 mInput->standby(); 4074 usleep(5000); 4075 } 4076 buffer->raw = 0; 4077 buffer->frameCount = 0; 4078 return NOT_ENOUGH_DATA; 4079 } 4080 mRsmpInIndex = 0; 4081 framesReady = mFrameCount; 4082 } 4083 4084 if (framesReq > framesReady) { 4085 framesReq = framesReady; 4086 } 4087 4088 if (mChannelCount == 1 && mReqChannelCount == 2) { 4089 channelCount = 1; 4090 } else { 4091 channelCount = 2; 4092 } 4093 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4094 buffer->frameCount = framesReq; 4095 return NO_ERROR; 4096} 4097 4098void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4099{ 4100 mRsmpInIndex += buffer->frameCount; 4101 buffer->frameCount = 0; 4102} 4103 4104bool AudioFlinger::RecordThread::checkForNewParameters_l() 4105{ 4106 bool reconfig = false; 4107 4108 while (!mNewParameters.isEmpty()) { 4109 status_t status = NO_ERROR; 4110 String8 keyValuePair = mNewParameters[0]; 4111 AudioParameter param = AudioParameter(keyValuePair); 4112 int value; 4113 int reqFormat = mFormat; 4114 int reqSamplingRate = mReqSampleRate; 4115 int reqChannelCount = mReqChannelCount; 4116 4117 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4118 reqSamplingRate = value; 4119 reconfig = true; 4120 } 4121 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4122 reqFormat = value; 4123 reconfig = true; 4124 } 4125 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4126 reqChannelCount = AudioSystem::popCount(value); 4127 reconfig = true; 4128 } 4129 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4130 // do not accept frame count changes if tracks are open as the track buffer 4131 // size depends on frame count and correct behavior would not be garantied 4132 // if frame count is changed after track creation 4133 if (mActiveTrack != 0) { 4134 status = INVALID_OPERATION; 4135 } else { 4136 reconfig = true; 4137 } 4138 } 4139 if (status == NO_ERROR) { 4140 status = mInput->setParameters(keyValuePair); 4141 if (status == INVALID_OPERATION) { 4142 mInput->standby(); 4143 status = mInput->setParameters(keyValuePair); 4144 } 4145 if (reconfig) { 4146 if (status == BAD_VALUE && 4147 reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && 4148 ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && 4149 (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { 4150 status = NO_ERROR; 4151 } 4152 if (status == NO_ERROR) { 4153 readInputParameters(); 4154 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4155 } 4156 } 4157 } 4158 4159 mNewParameters.removeAt(0); 4160 4161 mParamStatus = status; 4162 mParamCond.signal(); 4163 mWaitWorkCV.wait(mLock); 4164 } 4165 return reconfig; 4166} 4167 4168String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4169{ 4170 return mInput->getParameters(keys); 4171} 4172 4173void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4174 AudioSystem::OutputDescriptor desc; 4175 void *param2 = 0; 4176 4177 switch (event) { 4178 case AudioSystem::INPUT_OPENED: 4179 case AudioSystem::INPUT_CONFIG_CHANGED: 4180 desc.channels = mChannels; 4181 desc.samplingRate = mSampleRate; 4182 desc.format = mFormat; 4183 desc.frameCount = mFrameCount; 4184 desc.latency = 0; 4185 param2 = &desc; 4186 break; 4187 4188 case AudioSystem::INPUT_CLOSED: 4189 default: 4190 break; 4191 } 4192 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4193} 4194 4195void AudioFlinger::RecordThread::readInputParameters() 4196{ 4197 if (mRsmpInBuffer) delete mRsmpInBuffer; 4198 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4199 if (mResampler) delete mResampler; 4200 mResampler = 0; 4201 4202 mSampleRate = mInput->sampleRate(); 4203 mChannels = mInput->channels(); 4204 mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); 4205 mFormat = mInput->format(); 4206 mFrameSize = (uint16_t)mInput->frameSize(); 4207 mInputBytes = mInput->bufferSize(); 4208 mFrameCount = mInputBytes / mFrameSize; 4209 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4210 4211 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4212 { 4213 int channelCount; 4214 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4215 // stereo to mono post process as the resampler always outputs stereo. 4216 if (mChannelCount == 1 && mReqChannelCount == 2) { 4217 channelCount = 1; 4218 } else { 4219 channelCount = 2; 4220 } 4221 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4222 mResampler->setSampleRate(mSampleRate); 4223 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4224 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4225 4226 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4227 if (mChannelCount == 1 && mReqChannelCount == 1) { 4228 mFrameCount >>= 1; 4229 } 4230 4231 } 4232 mRsmpInIndex = mFrameCount; 4233} 4234 4235unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4236{ 4237 return mInput->getInputFramesLost(); 4238} 4239 4240// ---------------------------------------------------------------------------- 4241 4242int AudioFlinger::openOutput(uint32_t *pDevices, 4243 uint32_t *pSamplingRate, 4244 uint32_t *pFormat, 4245 uint32_t *pChannels, 4246 uint32_t *pLatencyMs, 4247 uint32_t flags) 4248{ 4249 status_t status; 4250 PlaybackThread *thread = NULL; 4251 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4252 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4253 uint32_t format = pFormat ? *pFormat : 0; 4254 uint32_t channels = pChannels ? *pChannels : 0; 4255 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4256 4257 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4258 pDevices ? *pDevices : 0, 4259 samplingRate, 4260 format, 4261 channels, 4262 flags); 4263 4264 if (pDevices == NULL || *pDevices == 0) { 4265 return 0; 4266 } 4267 Mutex::Autolock _l(mLock); 4268 4269 AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, 4270 (int *)&format, 4271 &channels, 4272 &samplingRate, 4273 &status); 4274 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4275 output, 4276 samplingRate, 4277 format, 4278 channels, 4279 status); 4280 4281 mHardwareStatus = AUDIO_HW_IDLE; 4282 if (output != 0) { 4283 int id = nextUniqueId(); 4284 if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || 4285 (format != AudioSystem::PCM_16_BIT) || 4286 (channels != AudioSystem::CHANNEL_OUT_STEREO)) { 4287 thread = new DirectOutputThread(this, output, id, *pDevices); 4288 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4289 } else { 4290 thread = new MixerThread(this, output, id, *pDevices); 4291 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4292 4293#ifdef LVMX 4294 unsigned bitsPerSample = 4295 (format == AudioSystem::PCM_16_BIT) ? 16 : 4296 ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); 4297 unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; 4298 int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); 4299 4300 LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); 4301 LifeVibes::setDevice(audioOutputType, *pDevices); 4302#endif 4303 4304 } 4305 mPlaybackThreads.add(id, thread); 4306 4307 if (pSamplingRate) *pSamplingRate = samplingRate; 4308 if (pFormat) *pFormat = format; 4309 if (pChannels) *pChannels = channels; 4310 if (pLatencyMs) *pLatencyMs = thread->latency(); 4311 4312 // notify client processes of the new output creation 4313 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4314 return id; 4315 } 4316 4317 return 0; 4318} 4319 4320int AudioFlinger::openDuplicateOutput(int output1, int output2) 4321{ 4322 Mutex::Autolock _l(mLock); 4323 MixerThread *thread1 = checkMixerThread_l(output1); 4324 MixerThread *thread2 = checkMixerThread_l(output2); 4325 4326 if (thread1 == NULL || thread2 == NULL) { 4327 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4328 return 0; 4329 } 4330 4331 int id = nextUniqueId(); 4332 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4333 thread->addOutputTrack(thread2); 4334 mPlaybackThreads.add(id, thread); 4335 // notify client processes of the new output creation 4336 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4337 return id; 4338} 4339 4340status_t AudioFlinger::closeOutput(int output) 4341{ 4342 // keep strong reference on the playback thread so that 4343 // it is not destroyed while exit() is executed 4344 sp <PlaybackThread> thread; 4345 { 4346 Mutex::Autolock _l(mLock); 4347 thread = checkPlaybackThread_l(output); 4348 if (thread == NULL) { 4349 return BAD_VALUE; 4350 } 4351 4352 LOGV("closeOutput() %d", output); 4353 4354 if (thread->type() == PlaybackThread::MIXER) { 4355 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4356 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4357 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4358 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4359 } 4360 } 4361 } 4362 void *param2 = 0; 4363 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4364 mPlaybackThreads.removeItem(output); 4365 } 4366 thread->exit(); 4367 4368 if (thread->type() != PlaybackThread::DUPLICATING) { 4369 mAudioHardware->closeOutputStream(thread->getOutput()); 4370 } 4371 return NO_ERROR; 4372} 4373 4374status_t AudioFlinger::suspendOutput(int output) 4375{ 4376 Mutex::Autolock _l(mLock); 4377 PlaybackThread *thread = checkPlaybackThread_l(output); 4378 4379 if (thread == NULL) { 4380 return BAD_VALUE; 4381 } 4382 4383 LOGV("suspendOutput() %d", output); 4384 thread->suspend(); 4385 4386 return NO_ERROR; 4387} 4388 4389status_t AudioFlinger::restoreOutput(int output) 4390{ 4391 Mutex::Autolock _l(mLock); 4392 PlaybackThread *thread = checkPlaybackThread_l(output); 4393 4394 if (thread == NULL) { 4395 return BAD_VALUE; 4396 } 4397 4398 LOGV("restoreOutput() %d", output); 4399 4400 thread->restore(); 4401 4402 return NO_ERROR; 4403} 4404 4405int AudioFlinger::openInput(uint32_t *pDevices, 4406 uint32_t *pSamplingRate, 4407 uint32_t *pFormat, 4408 uint32_t *pChannels, 4409 uint32_t acoustics) 4410{ 4411 status_t status; 4412 RecordThread *thread = NULL; 4413 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4414 uint32_t format = pFormat ? *pFormat : 0; 4415 uint32_t channels = pChannels ? *pChannels : 0; 4416 uint32_t reqSamplingRate = samplingRate; 4417 uint32_t reqFormat = format; 4418 uint32_t reqChannels = channels; 4419 4420 if (pDevices == NULL || *pDevices == 0) { 4421 return 0; 4422 } 4423 Mutex::Autolock _l(mLock); 4424 4425 AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, 4426 (int *)&format, 4427 &channels, 4428 &samplingRate, 4429 &status, 4430 (AudioSystem::audio_in_acoustics)acoustics); 4431 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4432 input, 4433 samplingRate, 4434 format, 4435 channels, 4436 acoustics, 4437 status); 4438 4439 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4440 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4441 // or stereo to mono conversions on 16 bit PCM inputs. 4442 if (input == 0 && status == BAD_VALUE && 4443 reqFormat == format && format == AudioSystem::PCM_16_BIT && 4444 (samplingRate <= 2 * reqSamplingRate) && 4445 (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { 4446 LOGV("openInput() reopening with proposed sampling rate and channels"); 4447 input = mAudioHardware->openInputStream(*pDevices, 4448 (int *)&format, 4449 &channels, 4450 &samplingRate, 4451 &status, 4452 (AudioSystem::audio_in_acoustics)acoustics); 4453 } 4454 4455 if (input != 0) { 4456 int id = nextUniqueId(); 4457 // Start record thread 4458 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4459 mRecordThreads.add(id, thread); 4460 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4461 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4462 if (pFormat) *pFormat = format; 4463 if (pChannels) *pChannels = reqChannels; 4464 4465 input->standby(); 4466 4467 // notify client processes of the new input creation 4468 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4469 return id; 4470 } 4471 4472 return 0; 4473} 4474 4475status_t AudioFlinger::closeInput(int input) 4476{ 4477 // keep strong reference on the record thread so that 4478 // it is not destroyed while exit() is executed 4479 sp <RecordThread> thread; 4480 { 4481 Mutex::Autolock _l(mLock); 4482 thread = checkRecordThread_l(input); 4483 if (thread == NULL) { 4484 return BAD_VALUE; 4485 } 4486 4487 LOGV("closeInput() %d", input); 4488 void *param2 = 0; 4489 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4490 mRecordThreads.removeItem(input); 4491 } 4492 thread->exit(); 4493 4494 mAudioHardware->closeInputStream(thread->getInput()); 4495 4496 return NO_ERROR; 4497} 4498 4499status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4500{ 4501 Mutex::Autolock _l(mLock); 4502 MixerThread *dstThread = checkMixerThread_l(output); 4503 if (dstThread == NULL) { 4504 LOGW("setStreamOutput() bad output id %d", output); 4505 return BAD_VALUE; 4506 } 4507 4508 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4509 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4510 4511 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4512 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4513 if (thread != dstThread && 4514 thread->type() != PlaybackThread::DIRECT) { 4515 MixerThread *srcThread = (MixerThread *)thread; 4516 srcThread->invalidateTracks(stream); 4517 } 4518 } 4519 4520 return NO_ERROR; 4521} 4522 4523 4524int AudioFlinger::newAudioSessionId() 4525{ 4526 return nextUniqueId(); 4527} 4528 4529// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4530AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4531{ 4532 PlaybackThread *thread = NULL; 4533 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4534 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4535 } 4536 return thread; 4537} 4538 4539// checkMixerThread_l() must be called with AudioFlinger::mLock held 4540AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4541{ 4542 PlaybackThread *thread = checkPlaybackThread_l(output); 4543 if (thread != NULL) { 4544 if (thread->type() == PlaybackThread::DIRECT) { 4545 thread = NULL; 4546 } 4547 } 4548 return (MixerThread *)thread; 4549} 4550 4551// checkRecordThread_l() must be called with AudioFlinger::mLock held 4552AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4553{ 4554 RecordThread *thread = NULL; 4555 if (mRecordThreads.indexOfKey(input) >= 0) { 4556 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4557 } 4558 return thread; 4559} 4560 4561int AudioFlinger::nextUniqueId() 4562{ 4563 return android_atomic_inc(&mNextUniqueId); 4564} 4565 4566// ---------------------------------------------------------------------------- 4567// Effect management 4568// ---------------------------------------------------------------------------- 4569 4570 4571status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) 4572{ 4573 // check calling permissions 4574 if (!settingsAllowed()) { 4575 return PERMISSION_DENIED; 4576 } 4577 // only allow libraries loaded from /system/lib/soundfx for now 4578 if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { 4579 return PERMISSION_DENIED; 4580 } 4581 4582 Mutex::Autolock _l(mLock); 4583 return EffectLoadLibrary(libPath, handle); 4584} 4585 4586status_t AudioFlinger::unloadEffectLibrary(int handle) 4587{ 4588 // check calling permissions 4589 if (!settingsAllowed()) { 4590 return PERMISSION_DENIED; 4591 } 4592 4593 Mutex::Autolock _l(mLock); 4594 return EffectUnloadLibrary(handle); 4595} 4596 4597status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4598{ 4599 Mutex::Autolock _l(mLock); 4600 return EffectQueryNumberEffects(numEffects); 4601} 4602 4603status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4604{ 4605 Mutex::Autolock _l(mLock); 4606 return EffectQueryEffect(index, descriptor); 4607} 4608 4609status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4610{ 4611 Mutex::Autolock _l(mLock); 4612 return EffectGetDescriptor(pUuid, descriptor); 4613} 4614 4615 4616// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4617static const effect_uuid_t VISUALIZATION_UUID_ = 4618 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4619 4620sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4621 effect_descriptor_t *pDesc, 4622 const sp<IEffectClient>& effectClient, 4623 int32_t priority, 4624 int output, 4625 int sessionId, 4626 status_t *status, 4627 int *id, 4628 int *enabled) 4629{ 4630 status_t lStatus = NO_ERROR; 4631 sp<EffectHandle> handle; 4632 effect_interface_t itfe; 4633 effect_descriptor_t desc; 4634 sp<Client> client; 4635 wp<Client> wclient; 4636 4637 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4638 pid, effectClient.get(), priority, sessionId, output); 4639 4640 if (pDesc == NULL) { 4641 lStatus = BAD_VALUE; 4642 goto Exit; 4643 } 4644 4645 { 4646 Mutex::Autolock _l(mLock); 4647 4648 // check recording permission for visualizer 4649 if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4650 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) { 4651 if (!recordingAllowed()) { 4652 lStatus = PERMISSION_DENIED; 4653 goto Exit; 4654 } 4655 } 4656 4657 if (!EffectIsNullUuid(&pDesc->uuid)) { 4658 // if uuid is specified, request effect descriptor 4659 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4660 if (lStatus < 0) { 4661 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4662 goto Exit; 4663 } 4664 } else { 4665 // if uuid is not specified, look for an available implementation 4666 // of the required type in effect factory 4667 if (EffectIsNullUuid(&pDesc->type)) { 4668 LOGW("createEffect() no effect type"); 4669 lStatus = BAD_VALUE; 4670 goto Exit; 4671 } 4672 uint32_t numEffects = 0; 4673 effect_descriptor_t d; 4674 bool found = false; 4675 4676 lStatus = EffectQueryNumberEffects(&numEffects); 4677 if (lStatus < 0) { 4678 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4679 goto Exit; 4680 } 4681 for (uint32_t i = 0; i < numEffects; i++) { 4682 lStatus = EffectQueryEffect(i, &desc); 4683 if (lStatus < 0) { 4684 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4685 continue; 4686 } 4687 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4688 // If matching type found save effect descriptor. If the session is 4689 // 0 and the effect is not auxiliary, continue enumeration in case 4690 // an auxiliary version of this effect type is available 4691 found = true; 4692 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4693 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX || 4694 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4695 break; 4696 } 4697 } 4698 } 4699 if (!found) { 4700 lStatus = BAD_VALUE; 4701 LOGW("createEffect() effect not found"); 4702 goto Exit; 4703 } 4704 // For same effect type, chose auxiliary version over insert version if 4705 // connect to output mix (Compliance to OpenSL ES) 4706 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && 4707 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4708 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4709 } 4710 } 4711 4712 // Do not allow auxiliary effects on a session different from 0 (output mix) 4713 if (sessionId != AudioSystem::SESSION_OUTPUT_MIX && 4714 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4715 lStatus = INVALID_OPERATION; 4716 goto Exit; 4717 } 4718 4719 // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects 4720 // that can only be created by audio policy manager (running in same process) 4721 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && 4722 getpid() != IPCThreadState::self()->getCallingPid()) { 4723 lStatus = INVALID_OPERATION; 4724 goto Exit; 4725 } 4726 4727 // return effect descriptor 4728 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4729 4730 // If output is not specified try to find a matching audio session ID in one of the 4731 // output threads. 4732 // TODO: allow attachment of effect to inputs 4733 if (output == 0) { 4734 if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) { 4735 // output must be specified by AudioPolicyManager when using session 4736 // AudioSystem::SESSION_OUTPUT_STAGE 4737 lStatus = BAD_VALUE; 4738 goto Exit; 4739 } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { 4740 output = AudioSystem::getOutputForEffect(&desc); 4741 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4742 } else { 4743 // look for the thread where the specified audio session is present 4744 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4745 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4746 output = mPlaybackThreads.keyAt(i); 4747 break; 4748 } 4749 } 4750 // If no output thread contains the requested session ID, default to 4751 // first output. The effect chain will be moved to the correct output 4752 // thread when a track with the same session ID is created 4753 if (output == 0 && mPlaybackThreads.size()) { 4754 output = mPlaybackThreads.keyAt(0); 4755 } 4756 } 4757 } 4758 PlaybackThread *thread = checkPlaybackThread_l(output); 4759 if (thread == NULL) { 4760 LOGE("createEffect() unknown output thread"); 4761 lStatus = BAD_VALUE; 4762 goto Exit; 4763 } 4764 4765 wclient = mClients.valueFor(pid); 4766 4767 if (wclient != NULL) { 4768 client = wclient.promote(); 4769 } else { 4770 client = new Client(this, pid); 4771 mClients.add(pid, client); 4772 } 4773 4774 // create effect on selected output trhead 4775 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4776 &desc, enabled, &lStatus); 4777 if (handle != 0 && id != NULL) { 4778 *id = handle->id(); 4779 } 4780 } 4781 4782Exit: 4783 if(status) { 4784 *status = lStatus; 4785 } 4786 return handle; 4787} 4788 4789status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4790{ 4791 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4792 session, srcOutput, dstOutput); 4793 Mutex::Autolock _l(mLock); 4794 if (srcOutput == dstOutput) { 4795 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4796 return NO_ERROR; 4797 } 4798 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4799 if (srcThread == NULL) { 4800 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4801 return BAD_VALUE; 4802 } 4803 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4804 if (dstThread == NULL) { 4805 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4806 return BAD_VALUE; 4807 } 4808 4809 Mutex::Autolock _dl(dstThread->mLock); 4810 Mutex::Autolock _sl(srcThread->mLock); 4811 moveEffectChain_l(session, srcThread, dstThread, false); 4812 4813 return NO_ERROR; 4814} 4815 4816// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4817status_t AudioFlinger::moveEffectChain_l(int session, 4818 AudioFlinger::PlaybackThread *srcThread, 4819 AudioFlinger::PlaybackThread *dstThread, 4820 bool reRegister) 4821{ 4822 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4823 session, srcThread, dstThread); 4824 4825 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4826 if (chain == 0) { 4827 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4828 session, srcThread); 4829 return INVALID_OPERATION; 4830 } 4831 4832 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 4833 // so that a new chain is created with correct parameters when first effect is added. This is 4834 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4835 // removed. 4836 srcThread->removeEffectChain_l(chain); 4837 4838 // transfer all effects one by one so that new effect chain is created on new thread with 4839 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4840 int dstOutput = dstThread->id(); 4841 sp<EffectChain> dstChain; 4842 uint32_t strategy; 4843 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4844 while (effect != 0) { 4845 srcThread->removeEffect_l(effect); 4846 dstThread->addEffect_l(effect); 4847 // if the move request is not received from audio policy manager, the effect must be 4848 // re-registered with the new strategy and output 4849 if (dstChain == 0) { 4850 dstChain = effect->chain().promote(); 4851 if (dstChain == 0) { 4852 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 4853 srcThread->addEffect_l(effect); 4854 return NO_INIT; 4855 } 4856 strategy = dstChain->strategy(); 4857 } 4858 if (reRegister) { 4859 AudioSystem::unregisterEffect(effect->id()); 4860 AudioSystem::registerEffect(&effect->desc(), 4861 dstOutput, 4862 strategy, 4863 session, 4864 effect->id()); 4865 } 4866 effect = chain->getEffectFromId_l(0); 4867 } 4868 4869 return NO_ERROR; 4870} 4871 4872// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4873sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4874 const sp<AudioFlinger::Client>& client, 4875 const sp<IEffectClient>& effectClient, 4876 int32_t priority, 4877 int sessionId, 4878 effect_descriptor_t *desc, 4879 int *enabled, 4880 status_t *status 4881 ) 4882{ 4883 sp<EffectModule> effect; 4884 sp<EffectHandle> handle; 4885 status_t lStatus; 4886 sp<Track> track; 4887 sp<EffectChain> chain; 4888 bool chainCreated = false; 4889 bool effectCreated = false; 4890 bool effectRegistered = false; 4891 4892 if (mOutput == 0) { 4893 LOGW("createEffect_l() Audio driver not initialized."); 4894 lStatus = NO_INIT; 4895 goto Exit; 4896 } 4897 4898 // Do not allow auxiliary effect on session other than 0 4899 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4900 sessionId != AudioSystem::SESSION_OUTPUT_MIX) { 4901 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4902 desc->name, sessionId); 4903 lStatus = BAD_VALUE; 4904 goto Exit; 4905 } 4906 4907 // Do not allow effects with session ID 0 on direct output or duplicating threads 4908 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4909 if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) { 4910 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4911 desc->name, sessionId); 4912 lStatus = BAD_VALUE; 4913 goto Exit; 4914 } 4915 4916 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4917 4918 { // scope for mLock 4919 Mutex::Autolock _l(mLock); 4920 4921 // check for existing effect chain with the requested audio session 4922 chain = getEffectChain_l(sessionId); 4923 if (chain == 0) { 4924 // create a new chain for this session 4925 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4926 chain = new EffectChain(this, sessionId); 4927 addEffectChain_l(chain); 4928 chain->setStrategy(getStrategyForSession_l(sessionId)); 4929 chainCreated = true; 4930 } else { 4931 effect = chain->getEffectFromDesc_l(desc); 4932 } 4933 4934 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4935 4936 if (effect == 0) { 4937 int id = mAudioFlinger->nextUniqueId(); 4938 // Check CPU and memory usage 4939 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4940 if (lStatus != NO_ERROR) { 4941 goto Exit; 4942 } 4943 effectRegistered = true; 4944 // create a new effect module if none present in the chain 4945 effect = new EffectModule(this, chain, desc, id, sessionId); 4946 lStatus = effect->status(); 4947 if (lStatus != NO_ERROR) { 4948 goto Exit; 4949 } 4950 lStatus = chain->addEffect_l(effect); 4951 if (lStatus != NO_ERROR) { 4952 goto Exit; 4953 } 4954 effectCreated = true; 4955 4956 effect->setDevice(mDevice); 4957 effect->setMode(mAudioFlinger->getMode()); 4958 } 4959 // create effect handle and connect it to effect module 4960 handle = new EffectHandle(effect, client, effectClient, priority); 4961 lStatus = effect->addHandle(handle); 4962 if (enabled) { 4963 *enabled = (int)effect->isEnabled(); 4964 } 4965 } 4966 4967Exit: 4968 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 4969 Mutex::Autolock _l(mLock); 4970 if (effectCreated) { 4971 chain->removeEffect_l(effect); 4972 } 4973 if (effectRegistered) { 4974 AudioSystem::unregisterEffect(effect->id()); 4975 } 4976 if (chainCreated) { 4977 removeEffectChain_l(chain); 4978 } 4979 handle.clear(); 4980 } 4981 4982 if(status) { 4983 *status = lStatus; 4984 } 4985 return handle; 4986} 4987 4988// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 4989// PlaybackThread::mLock held 4990status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 4991{ 4992 // check for existing effect chain with the requested audio session 4993 int sessionId = effect->sessionId(); 4994 sp<EffectChain> chain = getEffectChain_l(sessionId); 4995 bool chainCreated = false; 4996 4997 if (chain == 0) { 4998 // create a new chain for this session 4999 LOGV("addEffect_l() new effect chain for session %d", sessionId); 5000 chain = new EffectChain(this, sessionId); 5001 addEffectChain_l(chain); 5002 chain->setStrategy(getStrategyForSession_l(sessionId)); 5003 chainCreated = true; 5004 } 5005 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5006 5007 if (chain->getEffectFromId_l(effect->id()) != 0) { 5008 LOGW("addEffect_l() %p effect %s already present in chain %p", 5009 this, effect->desc().name, chain.get()); 5010 return BAD_VALUE; 5011 } 5012 5013 status_t status = chain->addEffect_l(effect); 5014 if (status != NO_ERROR) { 5015 if (chainCreated) { 5016 removeEffectChain_l(chain); 5017 } 5018 return status; 5019 } 5020 5021 effect->setDevice(mDevice); 5022 effect->setMode(mAudioFlinger->getMode()); 5023 return NO_ERROR; 5024} 5025 5026void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 5027 5028 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5029 effect_descriptor_t desc = effect->desc(); 5030 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5031 detachAuxEffect_l(effect->id()); 5032 } 5033 5034 sp<EffectChain> chain = effect->chain().promote(); 5035 if (chain != 0) { 5036 // remove effect chain if removing last effect 5037 if (chain->removeEffect_l(effect) == 0) { 5038 removeEffectChain_l(chain); 5039 } 5040 } else { 5041 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5042 } 5043} 5044 5045void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 5046 const wp<EffectHandle>& handle) { 5047 Mutex::Autolock _l(mLock); 5048 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5049 // delete the effect module if removing last handle on it 5050 if (effect->removeHandle(handle) == 0) { 5051 removeEffect_l(effect); 5052 AudioSystem::unregisterEffect(effect->id()); 5053 } 5054} 5055 5056status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5057{ 5058 int session = chain->sessionId(); 5059 int16_t *buffer = mMixBuffer; 5060 bool ownsBuffer = false; 5061 5062 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5063 if (session > 0) { 5064 // Only one effect chain can be present in direct output thread and it uses 5065 // the mix buffer as input 5066 if (mType != DIRECT) { 5067 size_t numSamples = mFrameCount * mChannelCount; 5068 buffer = new int16_t[numSamples]; 5069 memset(buffer, 0, numSamples * sizeof(int16_t)); 5070 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5071 ownsBuffer = true; 5072 } 5073 5074 // Attach all tracks with same session ID to this chain. 5075 for (size_t i = 0; i < mTracks.size(); ++i) { 5076 sp<Track> track = mTracks[i]; 5077 if (session == track->sessionId()) { 5078 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5079 track->setMainBuffer(buffer); 5080 } 5081 } 5082 5083 // indicate all active tracks in the chain 5084 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5085 sp<Track> track = mActiveTracks[i].promote(); 5086 if (track == 0) continue; 5087 if (session == track->sessionId()) { 5088 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5089 chain->startTrack(); 5090 } 5091 } 5092 } 5093 5094 chain->setInBuffer(buffer, ownsBuffer); 5095 chain->setOutBuffer(mMixBuffer); 5096 // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect 5097 // chains list in order to be processed last as it contains output stage effects 5098 // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before 5099 // session AudioSystem::SESSION_OUTPUT_STAGE to be processed 5100 // after track specific effects and before output stage 5101 // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and 5102 // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX 5103 // Effect chain for other sessions are inserted at beginning of effect 5104 // chains list to be processed before output mix effects. Relative order between other 5105 // sessions is not important 5106 size_t size = mEffectChains.size(); 5107 size_t i = 0; 5108 for (i = 0; i < size; i++) { 5109 if (mEffectChains[i]->sessionId() < session) break; 5110 } 5111 mEffectChains.insertAt(chain, i); 5112 5113 return NO_ERROR; 5114} 5115 5116size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5117{ 5118 int session = chain->sessionId(); 5119 5120 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5121 5122 for (size_t i = 0; i < mEffectChains.size(); i++) { 5123 if (chain == mEffectChains[i]) { 5124 mEffectChains.removeAt(i); 5125 // detach all tracks with same session ID from this chain 5126 for (size_t i = 0; i < mTracks.size(); ++i) { 5127 sp<Track> track = mTracks[i]; 5128 if (session == track->sessionId()) { 5129 track->setMainBuffer(mMixBuffer); 5130 } 5131 } 5132 break; 5133 } 5134 } 5135 return mEffectChains.size(); 5136} 5137 5138void AudioFlinger::PlaybackThread::lockEffectChains_l( 5139 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5140{ 5141 effectChains = mEffectChains; 5142 for (size_t i = 0; i < mEffectChains.size(); i++) { 5143 mEffectChains[i]->lock(); 5144 } 5145} 5146 5147void AudioFlinger::PlaybackThread::unlockEffectChains( 5148 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5149{ 5150 for (size_t i = 0; i < effectChains.size(); i++) { 5151 effectChains[i]->unlock(); 5152 } 5153} 5154 5155 5156sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5157{ 5158 sp<EffectModule> effect; 5159 5160 sp<EffectChain> chain = getEffectChain_l(sessionId); 5161 if (chain != 0) { 5162 effect = chain->getEffectFromId_l(effectId); 5163 } 5164 return effect; 5165} 5166 5167status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5168 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5169{ 5170 Mutex::Autolock _l(mLock); 5171 return attachAuxEffect_l(track, EffectId); 5172} 5173 5174status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5175 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5176{ 5177 status_t status = NO_ERROR; 5178 5179 if (EffectId == 0) { 5180 track->setAuxBuffer(0, NULL); 5181 } else { 5182 // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX 5183 sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId); 5184 if (effect != 0) { 5185 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5186 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5187 } else { 5188 status = INVALID_OPERATION; 5189 } 5190 } else { 5191 status = BAD_VALUE; 5192 } 5193 } 5194 return status; 5195} 5196 5197void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5198{ 5199 for (size_t i = 0; i < mTracks.size(); ++i) { 5200 sp<Track> track = mTracks[i]; 5201 if (track->auxEffectId() == effectId) { 5202 attachAuxEffect_l(track, 0); 5203 } 5204 } 5205} 5206 5207// ---------------------------------------------------------------------------- 5208// EffectModule implementation 5209// ---------------------------------------------------------------------------- 5210 5211#undef LOG_TAG 5212#define LOG_TAG "AudioFlinger::EffectModule" 5213 5214AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5215 const wp<AudioFlinger::EffectChain>& chain, 5216 effect_descriptor_t *desc, 5217 int id, 5218 int sessionId) 5219 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5220 mStatus(NO_INIT), mState(IDLE) 5221{ 5222 LOGV("Constructor %p", this); 5223 int lStatus; 5224 sp<ThreadBase> thread = mThread.promote(); 5225 if (thread == 0) { 5226 return; 5227 } 5228 PlaybackThread *p = (PlaybackThread *)thread.get(); 5229 5230 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5231 5232 // create effect engine from effect factory 5233 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5234 5235 if (mStatus != NO_ERROR) { 5236 return; 5237 } 5238 lStatus = init(); 5239 if (lStatus < 0) { 5240 mStatus = lStatus; 5241 goto Error; 5242 } 5243 5244 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5245 return; 5246Error: 5247 EffectRelease(mEffectInterface); 5248 mEffectInterface = NULL; 5249 LOGV("Constructor Error %d", mStatus); 5250} 5251 5252AudioFlinger::EffectModule::~EffectModule() 5253{ 5254 LOGV("Destructor %p", this); 5255 if (mEffectInterface != NULL) { 5256 // release effect engine 5257 EffectRelease(mEffectInterface); 5258 } 5259} 5260 5261status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5262{ 5263 status_t status; 5264 5265 Mutex::Autolock _l(mLock); 5266 // First handle in mHandles has highest priority and controls the effect module 5267 int priority = handle->priority(); 5268 size_t size = mHandles.size(); 5269 sp<EffectHandle> h; 5270 size_t i; 5271 for (i = 0; i < size; i++) { 5272 h = mHandles[i].promote(); 5273 if (h == 0) continue; 5274 if (h->priority() <= priority) break; 5275 } 5276 // if inserted in first place, move effect control from previous owner to this handle 5277 if (i == 0) { 5278 if (h != 0) { 5279 h->setControl(false, true); 5280 } 5281 handle->setControl(true, false); 5282 status = NO_ERROR; 5283 } else { 5284 status = ALREADY_EXISTS; 5285 } 5286 mHandles.insertAt(handle, i); 5287 return status; 5288} 5289 5290size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5291{ 5292 Mutex::Autolock _l(mLock); 5293 size_t size = mHandles.size(); 5294 size_t i; 5295 for (i = 0; i < size; i++) { 5296 if (mHandles[i] == handle) break; 5297 } 5298 if (i == size) { 5299 return size; 5300 } 5301 mHandles.removeAt(i); 5302 size = mHandles.size(); 5303 // if removed from first place, move effect control from this handle to next in line 5304 if (i == 0 && size != 0) { 5305 sp<EffectHandle> h = mHandles[0].promote(); 5306 if (h != 0) { 5307 h->setControl(true, true); 5308 } 5309 } 5310 5311 return size; 5312} 5313 5314void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5315{ 5316 // keep a strong reference on this EffectModule to avoid calling the 5317 // destructor before we exit 5318 sp<EffectModule> keep(this); 5319 { 5320 sp<ThreadBase> thread = mThread.promote(); 5321 if (thread != 0) { 5322 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5323 playbackThread->disconnectEffect(keep, handle); 5324 } 5325 } 5326} 5327 5328void AudioFlinger::EffectModule::updateState() { 5329 Mutex::Autolock _l(mLock); 5330 5331 switch (mState) { 5332 case RESTART: 5333 reset_l(); 5334 // FALL THROUGH 5335 5336 case STARTING: 5337 // clear auxiliary effect input buffer for next accumulation 5338 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5339 memset(mConfig.inputCfg.buffer.raw, 5340 0, 5341 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5342 } 5343 start_l(); 5344 mState = ACTIVE; 5345 break; 5346 case STOPPING: 5347 stop_l(); 5348 mDisableWaitCnt = mMaxDisableWaitCnt; 5349 mState = STOPPED; 5350 break; 5351 case STOPPED: 5352 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5353 // turn off sequence. 5354 if (--mDisableWaitCnt == 0) { 5355 reset_l(); 5356 mState = IDLE; 5357 } 5358 break; 5359 default: //IDLE , ACTIVE 5360 break; 5361 } 5362} 5363 5364void AudioFlinger::EffectModule::process() 5365{ 5366 Mutex::Autolock _l(mLock); 5367 5368 if (mEffectInterface == NULL || 5369 mConfig.inputCfg.buffer.raw == NULL || 5370 mConfig.outputCfg.buffer.raw == NULL) { 5371 return; 5372 } 5373 5374 if (mState == ACTIVE || mState == STOPPING || mState == STOPPED || mState == RESTART) { 5375 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5376 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5377 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5378 mConfig.inputCfg.buffer.s32, 5379 mConfig.inputCfg.buffer.frameCount/2); 5380 } 5381 5382 // do the actual processing in the effect engine 5383 int ret = (*mEffectInterface)->process(mEffectInterface, 5384 &mConfig.inputCfg.buffer, 5385 &mConfig.outputCfg.buffer); 5386 5387 // force transition to IDLE state when engine is ready 5388 if (mState == STOPPED && ret == -ENODATA) { 5389 mDisableWaitCnt = 1; 5390 } 5391 5392 // clear auxiliary effect input buffer for next accumulation 5393 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5394 memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5395 } 5396 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5397 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ 5398 // If an insert effect is idle and input buffer is different from output buffer, copy input to 5399 // output 5400 sp<EffectChain> chain = mChain.promote(); 5401 if (chain != 0 && chain->activeTracks() != 0) { 5402 size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); 5403 if (mConfig.inputCfg.channels == CHANNEL_STEREO) { 5404 size *= 2; 5405 } 5406 memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); 5407 } 5408 } 5409} 5410 5411void AudioFlinger::EffectModule::reset_l() 5412{ 5413 if (mEffectInterface == NULL) { 5414 return; 5415 } 5416 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5417} 5418 5419status_t AudioFlinger::EffectModule::configure() 5420{ 5421 uint32_t channels; 5422 if (mEffectInterface == NULL) { 5423 return NO_INIT; 5424 } 5425 5426 sp<ThreadBase> thread = mThread.promote(); 5427 if (thread == 0) { 5428 return DEAD_OBJECT; 5429 } 5430 5431 // TODO: handle configuration of effects replacing track process 5432 if (thread->channelCount() == 1) { 5433 channels = CHANNEL_MONO; 5434 } else { 5435 channels = CHANNEL_STEREO; 5436 } 5437 5438 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5439 mConfig.inputCfg.channels = CHANNEL_MONO; 5440 } else { 5441 mConfig.inputCfg.channels = channels; 5442 } 5443 mConfig.outputCfg.channels = channels; 5444 mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; 5445 mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; 5446 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5447 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5448 mConfig.inputCfg.bufferProvider.cookie = NULL; 5449 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5450 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5451 mConfig.outputCfg.bufferProvider.cookie = NULL; 5452 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5453 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5454 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5455 // Insert effect: 5456 // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE, 5457 // always overwrites output buffer: input buffer == output buffer 5458 // - in other sessions: 5459 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5460 // other effect: overwrites output buffer: input buffer == output buffer 5461 // Auxiliary effect: 5462 // accumulates in output buffer: input buffer != output buffer 5463 // Therefore: accumulate <=> input buffer != output buffer 5464 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5465 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5466 } else { 5467 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5468 } 5469 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5470 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5471 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5472 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5473 5474 LOGV("configure() %p thread %p buffer %p framecount %d", 5475 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5476 5477 status_t cmdStatus; 5478 uint32_t size = sizeof(int); 5479 status_t status = (*mEffectInterface)->command(mEffectInterface, 5480 EFFECT_CMD_CONFIGURE, 5481 sizeof(effect_config_t), 5482 &mConfig, 5483 &size, 5484 &cmdStatus); 5485 if (status == 0) { 5486 status = cmdStatus; 5487 } 5488 5489 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5490 (1000 * mConfig.outputCfg.buffer.frameCount); 5491 5492 return status; 5493} 5494 5495status_t AudioFlinger::EffectModule::init() 5496{ 5497 Mutex::Autolock _l(mLock); 5498 if (mEffectInterface == NULL) { 5499 return NO_INIT; 5500 } 5501 status_t cmdStatus; 5502 uint32_t size = sizeof(status_t); 5503 status_t status = (*mEffectInterface)->command(mEffectInterface, 5504 EFFECT_CMD_INIT, 5505 0, 5506 NULL, 5507 &size, 5508 &cmdStatus); 5509 if (status == 0) { 5510 status = cmdStatus; 5511 } 5512 return status; 5513} 5514 5515status_t AudioFlinger::EffectModule::start_l() 5516{ 5517 if (mEffectInterface == NULL) { 5518 return NO_INIT; 5519 } 5520 status_t cmdStatus; 5521 uint32_t size = sizeof(status_t); 5522 status_t status = (*mEffectInterface)->command(mEffectInterface, 5523 EFFECT_CMD_ENABLE, 5524 0, 5525 NULL, 5526 &size, 5527 &cmdStatus); 5528 if (status == 0) { 5529 status = cmdStatus; 5530 } 5531 return status; 5532} 5533 5534status_t AudioFlinger::EffectModule::stop_l() 5535{ 5536 if (mEffectInterface == NULL) { 5537 return NO_INIT; 5538 } 5539 status_t cmdStatus; 5540 uint32_t size = sizeof(status_t); 5541 status_t status = (*mEffectInterface)->command(mEffectInterface, 5542 EFFECT_CMD_DISABLE, 5543 0, 5544 NULL, 5545 &size, 5546 &cmdStatus); 5547 if (status == 0) { 5548 status = cmdStatus; 5549 } 5550 return status; 5551} 5552 5553status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5554 uint32_t cmdSize, 5555 void *pCmdData, 5556 uint32_t *replySize, 5557 void *pReplyData) 5558{ 5559 Mutex::Autolock _l(mLock); 5560// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5561 5562 if (mEffectInterface == NULL) { 5563 return NO_INIT; 5564 } 5565 status_t status = (*mEffectInterface)->command(mEffectInterface, 5566 cmdCode, 5567 cmdSize, 5568 pCmdData, 5569 replySize, 5570 pReplyData); 5571 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5572 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5573 for (size_t i = 1; i < mHandles.size(); i++) { 5574 sp<EffectHandle> h = mHandles[i].promote(); 5575 if (h != 0) { 5576 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5577 } 5578 } 5579 } 5580 return status; 5581} 5582 5583status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5584{ 5585 Mutex::Autolock _l(mLock); 5586 LOGV("setEnabled %p enabled %d", this, enabled); 5587 5588 if (enabled != isEnabled()) { 5589 switch (mState) { 5590 // going from disabled to enabled 5591 case IDLE: 5592 mState = STARTING; 5593 break; 5594 case STOPPED: 5595 mState = RESTART; 5596 break; 5597 case STOPPING: 5598 mState = ACTIVE; 5599 break; 5600 5601 // going from enabled to disabled 5602 case RESTART: 5603 case STARTING: 5604 mState = IDLE; 5605 break; 5606 case ACTIVE: 5607 mState = STOPPING; 5608 break; 5609 } 5610 for (size_t i = 1; i < mHandles.size(); i++) { 5611 sp<EffectHandle> h = mHandles[i].promote(); 5612 if (h != 0) { 5613 h->setEnabled(enabled); 5614 } 5615 } 5616 } 5617 return NO_ERROR; 5618} 5619 5620bool AudioFlinger::EffectModule::isEnabled() 5621{ 5622 switch (mState) { 5623 case RESTART: 5624 case STARTING: 5625 case ACTIVE: 5626 return true; 5627 case IDLE: 5628 case STOPPING: 5629 case STOPPED: 5630 default: 5631 return false; 5632 } 5633} 5634 5635status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5636{ 5637 Mutex::Autolock _l(mLock); 5638 status_t status = NO_ERROR; 5639 5640 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5641 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5642 if ((mState >= ACTIVE) && 5643 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5644 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5645 status_t cmdStatus; 5646 uint32_t volume[2]; 5647 uint32_t *pVolume = NULL; 5648 uint32_t size = sizeof(volume); 5649 volume[0] = *left; 5650 volume[1] = *right; 5651 if (controller) { 5652 pVolume = volume; 5653 } 5654 status = (*mEffectInterface)->command(mEffectInterface, 5655 EFFECT_CMD_SET_VOLUME, 5656 size, 5657 volume, 5658 &size, 5659 pVolume); 5660 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5661 *left = volume[0]; 5662 *right = volume[1]; 5663 } 5664 } 5665 return status; 5666} 5667 5668status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5669{ 5670 Mutex::Autolock _l(mLock); 5671 status_t status = NO_ERROR; 5672 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5673 // convert device bit field from AudioSystem to EffectApi format. 5674 device = deviceAudioSystemToEffectApi(device); 5675 if (device == 0) { 5676 return BAD_VALUE; 5677 } 5678 status_t cmdStatus; 5679 uint32_t size = sizeof(status_t); 5680 status = (*mEffectInterface)->command(mEffectInterface, 5681 EFFECT_CMD_SET_DEVICE, 5682 sizeof(uint32_t), 5683 &device, 5684 &size, 5685 &cmdStatus); 5686 if (status == NO_ERROR) { 5687 status = cmdStatus; 5688 } 5689 } 5690 return status; 5691} 5692 5693status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5694{ 5695 Mutex::Autolock _l(mLock); 5696 status_t status = NO_ERROR; 5697 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5698 // convert audio mode from AudioSystem to EffectApi format. 5699 int effectMode = modeAudioSystemToEffectApi(mode); 5700 if (effectMode < 0) { 5701 return BAD_VALUE; 5702 } 5703 status_t cmdStatus; 5704 uint32_t size = sizeof(status_t); 5705 status = (*mEffectInterface)->command(mEffectInterface, 5706 EFFECT_CMD_SET_AUDIO_MODE, 5707 sizeof(int), 5708 &effectMode, 5709 &size, 5710 &cmdStatus); 5711 if (status == NO_ERROR) { 5712 status = cmdStatus; 5713 } 5714 } 5715 return status; 5716} 5717 5718// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified 5719const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { 5720 DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE 5721 DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER 5722 DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET 5723 DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE 5724 DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO 5725 DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET 5726 DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT 5727 DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP 5728 DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES 5729 DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER 5730 DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL 5731}; 5732 5733uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) 5734{ 5735 uint32_t deviceOut = 0; 5736 while (device) { 5737 const uint32_t i = 31 - __builtin_clz(device); 5738 device &= ~(1 << i); 5739 if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { 5740 LOGE("device convertion error for AudioSystem device 0x%08x", device); 5741 return 0; 5742 } 5743 deviceOut |= (uint32_t)sDeviceConvTable[i]; 5744 } 5745 return deviceOut; 5746} 5747 5748// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified 5749const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { 5750 AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL 5751 AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE 5752 AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL 5753}; 5754 5755int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) 5756{ 5757 int modeOut = -1; 5758 if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { 5759 modeOut = (int)sModeConvTable[mode]; 5760 } 5761 return modeOut; 5762} 5763 5764status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5765{ 5766 const size_t SIZE = 256; 5767 char buffer[SIZE]; 5768 String8 result; 5769 5770 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5771 result.append(buffer); 5772 5773 bool locked = tryLock(mLock); 5774 // failed to lock - AudioFlinger is probably deadlocked 5775 if (!locked) { 5776 result.append("\t\tCould not lock Fx mutex:\n"); 5777 } 5778 5779 result.append("\t\tSession Status State Engine:\n"); 5780 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5781 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5782 result.append(buffer); 5783 5784 result.append("\t\tDescriptor:\n"); 5785 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5786 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5787 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5788 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5789 result.append(buffer); 5790 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5791 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5792 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5793 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5794 result.append(buffer); 5795 snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", 5796 mDescriptor.apiVersion, 5797 mDescriptor.flags); 5798 result.append(buffer); 5799 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5800 mDescriptor.name); 5801 result.append(buffer); 5802 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5803 mDescriptor.implementor); 5804 result.append(buffer); 5805 5806 result.append("\t\t- Input configuration:\n"); 5807 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5808 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5809 (uint32_t)mConfig.inputCfg.buffer.raw, 5810 mConfig.inputCfg.buffer.frameCount, 5811 mConfig.inputCfg.samplingRate, 5812 mConfig.inputCfg.channels, 5813 mConfig.inputCfg.format); 5814 result.append(buffer); 5815 5816 result.append("\t\t- Output configuration:\n"); 5817 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5818 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5819 (uint32_t)mConfig.outputCfg.buffer.raw, 5820 mConfig.outputCfg.buffer.frameCount, 5821 mConfig.outputCfg.samplingRate, 5822 mConfig.outputCfg.channels, 5823 mConfig.outputCfg.format); 5824 result.append(buffer); 5825 5826 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5827 result.append(buffer); 5828 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5829 for (size_t i = 0; i < mHandles.size(); ++i) { 5830 sp<EffectHandle> handle = mHandles[i].promote(); 5831 if (handle != 0) { 5832 handle->dump(buffer, SIZE); 5833 result.append(buffer); 5834 } 5835 } 5836 5837 result.append("\n"); 5838 5839 write(fd, result.string(), result.length()); 5840 5841 if (locked) { 5842 mLock.unlock(); 5843 } 5844 5845 return NO_ERROR; 5846} 5847 5848// ---------------------------------------------------------------------------- 5849// EffectHandle implementation 5850// ---------------------------------------------------------------------------- 5851 5852#undef LOG_TAG 5853#define LOG_TAG "AudioFlinger::EffectHandle" 5854 5855AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5856 const sp<AudioFlinger::Client>& client, 5857 const sp<IEffectClient>& effectClient, 5858 int32_t priority) 5859 : BnEffect(), 5860 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5861{ 5862 LOGV("constructor %p", this); 5863 5864 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5865 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5866 if (mCblkMemory != 0) { 5867 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5868 5869 if (mCblk) { 5870 new(mCblk) effect_param_cblk_t(); 5871 mBuffer = (uint8_t *)mCblk + bufOffset; 5872 } 5873 } else { 5874 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5875 return; 5876 } 5877} 5878 5879AudioFlinger::EffectHandle::~EffectHandle() 5880{ 5881 LOGV("Destructor %p", this); 5882 disconnect(); 5883} 5884 5885status_t AudioFlinger::EffectHandle::enable() 5886{ 5887 if (!mHasControl) return INVALID_OPERATION; 5888 if (mEffect == 0) return DEAD_OBJECT; 5889 5890 return mEffect->setEnabled(true); 5891} 5892 5893status_t AudioFlinger::EffectHandle::disable() 5894{ 5895 if (!mHasControl) return INVALID_OPERATION; 5896 if (mEffect == NULL) return DEAD_OBJECT; 5897 5898 return mEffect->setEnabled(false); 5899} 5900 5901void AudioFlinger::EffectHandle::disconnect() 5902{ 5903 if (mEffect == 0) { 5904 return; 5905 } 5906 mEffect->disconnect(this); 5907 // release sp on module => module destructor can be called now 5908 mEffect.clear(); 5909 if (mCblk) { 5910 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5911 } 5912 mCblkMemory.clear(); // and free the shared memory 5913 if (mClient != 0) { 5914 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5915 mClient.clear(); 5916 } 5917} 5918 5919status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5920 uint32_t cmdSize, 5921 void *pCmdData, 5922 uint32_t *replySize, 5923 void *pReplyData) 5924{ 5925// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5926// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5927 5928 // only get parameter command is permitted for applications not controlling the effect 5929 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5930 return INVALID_OPERATION; 5931 } 5932 if (mEffect == 0) return DEAD_OBJECT; 5933 5934 // handle commands that are not forwarded transparently to effect engine 5935 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5936 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5937 // no risk to block the whole media server process or mixer threads is we are stuck here 5938 Mutex::Autolock _l(mCblk->lock); 5939 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5940 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 5941 mCblk->serverIndex = 0; 5942 mCblk->clientIndex = 0; 5943 return BAD_VALUE; 5944 } 5945 status_t status = NO_ERROR; 5946 while (mCblk->serverIndex < mCblk->clientIndex) { 5947 int reply; 5948 uint32_t rsize = sizeof(int); 5949 int *p = (int *)(mBuffer + mCblk->serverIndex); 5950 int size = *p++; 5951 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 5952 LOGW("command(): invalid parameter block size"); 5953 break; 5954 } 5955 effect_param_t *param = (effect_param_t *)p; 5956 if (param->psize == 0 || param->vsize == 0) { 5957 LOGW("command(): null parameter or value size"); 5958 mCblk->serverIndex += size; 5959 continue; 5960 } 5961 uint32_t psize = sizeof(effect_param_t) + 5962 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 5963 param->vsize; 5964 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 5965 psize, 5966 p, 5967 &rsize, 5968 &reply); 5969 if (ret == NO_ERROR) { 5970 if (reply != NO_ERROR) { 5971 status = reply; 5972 } 5973 } else { 5974 status = ret; 5975 } 5976 mCblk->serverIndex += size; 5977 } 5978 mCblk->serverIndex = 0; 5979 mCblk->clientIndex = 0; 5980 return status; 5981 } else if (cmdCode == EFFECT_CMD_ENABLE) { 5982 return enable(); 5983 } else if (cmdCode == EFFECT_CMD_DISABLE) { 5984 return disable(); 5985 } 5986 5987 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 5988} 5989 5990sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 5991 return mCblkMemory; 5992} 5993 5994void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 5995{ 5996 LOGV("setControl %p control %d", this, hasControl); 5997 5998 mHasControl = hasControl; 5999 if (signal && mEffectClient != 0) { 6000 mEffectClient->controlStatusChanged(hasControl); 6001 } 6002} 6003 6004void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6005 uint32_t cmdSize, 6006 void *pCmdData, 6007 uint32_t replySize, 6008 void *pReplyData) 6009{ 6010 if (mEffectClient != 0) { 6011 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6012 } 6013} 6014 6015 6016 6017void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6018{ 6019 if (mEffectClient != 0) { 6020 mEffectClient->enableStatusChanged(enabled); 6021 } 6022} 6023 6024status_t AudioFlinger::EffectHandle::onTransact( 6025 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6026{ 6027 return BnEffect::onTransact(code, data, reply, flags); 6028} 6029 6030 6031void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6032{ 6033 bool locked = tryLock(mCblk->lock); 6034 6035 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6036 (mClient == NULL) ? getpid() : mClient->pid(), 6037 mPriority, 6038 mHasControl, 6039 !locked, 6040 mCblk->clientIndex, 6041 mCblk->serverIndex 6042 ); 6043 6044 if (locked) { 6045 mCblk->lock.unlock(); 6046 } 6047} 6048 6049#undef LOG_TAG 6050#define LOG_TAG "AudioFlinger::EffectChain" 6051 6052AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6053 int sessionId) 6054 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), 6055 mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 6056 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 6057{ 6058 mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC); 6059} 6060 6061AudioFlinger::EffectChain::~EffectChain() 6062{ 6063 if (mOwnInBuffer) { 6064 delete mInBuffer; 6065 } 6066 6067} 6068 6069// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6070sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6071{ 6072 sp<EffectModule> effect; 6073 size_t size = mEffects.size(); 6074 6075 for (size_t i = 0; i < size; i++) { 6076 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6077 effect = mEffects[i]; 6078 break; 6079 } 6080 } 6081 return effect; 6082} 6083 6084// getEffectFromId_l() must be called with PlaybackThread::mLock held 6085sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6086{ 6087 sp<EffectModule> effect; 6088 size_t size = mEffects.size(); 6089 6090 for (size_t i = 0; i < size; i++) { 6091 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6092 if (id == 0 || mEffects[i]->id() == id) { 6093 effect = mEffects[i]; 6094 break; 6095 } 6096 } 6097 return effect; 6098} 6099 6100// Must be called with EffectChain::mLock locked 6101void AudioFlinger::EffectChain::process_l() 6102{ 6103 size_t size = mEffects.size(); 6104 for (size_t i = 0; i < size; i++) { 6105 mEffects[i]->process(); 6106 } 6107 for (size_t i = 0; i < size; i++) { 6108 mEffects[i]->updateState(); 6109 } 6110 // if no track is active, input buffer must be cleared here as the mixer process 6111 // will not do it 6112 if (mSessionId > 0 && activeTracks() == 0) { 6113 sp<ThreadBase> thread = mThread.promote(); 6114 if (thread != 0) { 6115 size_t numSamples = thread->frameCount() * thread->channelCount(); 6116 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6117 } 6118 } 6119} 6120 6121// addEffect_l() must be called with PlaybackThread::mLock held 6122status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6123{ 6124 effect_descriptor_t desc = effect->desc(); 6125 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6126 6127 Mutex::Autolock _l(mLock); 6128 effect->setChain(this); 6129 sp<ThreadBase> thread = mThread.promote(); 6130 if (thread == 0) { 6131 return NO_INIT; 6132 } 6133 effect->setThread(thread); 6134 6135 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6136 // Auxiliary effects are inserted at the beginning of mEffects vector as 6137 // they are processed first and accumulated in chain input buffer 6138 mEffects.insertAt(effect, 0); 6139 6140 // the input buffer for auxiliary effect contains mono samples in 6141 // 32 bit format. This is to avoid saturation in AudoMixer 6142 // accumulation stage. Saturation is done in EffectModule::process() before 6143 // calling the process in effect engine 6144 size_t numSamples = thread->frameCount(); 6145 int32_t *buffer = new int32_t[numSamples]; 6146 memset(buffer, 0, numSamples * sizeof(int32_t)); 6147 effect->setInBuffer((int16_t *)buffer); 6148 // auxiliary effects output samples to chain input buffer for further processing 6149 // by insert effects 6150 effect->setOutBuffer(mInBuffer); 6151 } else { 6152 // Insert effects are inserted at the end of mEffects vector as they are processed 6153 // after track and auxiliary effects. 6154 // Insert effect order as a function of indicated preference: 6155 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6156 // another effect is present 6157 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6158 // last effect claiming first position 6159 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6160 // first effect claiming last position 6161 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6162 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6163 // already present 6164 6165 int size = (int)mEffects.size(); 6166 int idx_insert = size; 6167 int idx_insert_first = -1; 6168 int idx_insert_last = -1; 6169 6170 for (int i = 0; i < size; i++) { 6171 effect_descriptor_t d = mEffects[i]->desc(); 6172 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6173 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6174 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6175 // check invalid effect chaining combinations 6176 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6177 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6178 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6179 return INVALID_OPERATION; 6180 } 6181 // remember position of first insert effect and by default 6182 // select this as insert position for new effect 6183 if (idx_insert == size) { 6184 idx_insert = i; 6185 } 6186 // remember position of last insert effect claiming 6187 // first position 6188 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6189 idx_insert_first = i; 6190 } 6191 // remember position of first insert effect claiming 6192 // last position 6193 if (iPref == EFFECT_FLAG_INSERT_LAST && 6194 idx_insert_last == -1) { 6195 idx_insert_last = i; 6196 } 6197 } 6198 } 6199 6200 // modify idx_insert from first position if needed 6201 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6202 if (idx_insert_last != -1) { 6203 idx_insert = idx_insert_last; 6204 } else { 6205 idx_insert = size; 6206 } 6207 } else { 6208 if (idx_insert_first != -1) { 6209 idx_insert = idx_insert_first + 1; 6210 } 6211 } 6212 6213 // always read samples from chain input buffer 6214 effect->setInBuffer(mInBuffer); 6215 6216 // if last effect in the chain, output samples to chain 6217 // output buffer, otherwise to chain input buffer 6218 if (idx_insert == size) { 6219 if (idx_insert != 0) { 6220 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6221 mEffects[idx_insert-1]->configure(); 6222 } 6223 effect->setOutBuffer(mOutBuffer); 6224 } else { 6225 effect->setOutBuffer(mInBuffer); 6226 } 6227 mEffects.insertAt(effect, idx_insert); 6228 6229 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6230 } 6231 effect->configure(); 6232 return NO_ERROR; 6233} 6234 6235// removeEffect_l() must be called with PlaybackThread::mLock held 6236size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6237{ 6238 Mutex::Autolock _l(mLock); 6239 int size = (int)mEffects.size(); 6240 int i; 6241 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6242 6243 for (i = 0; i < size; i++) { 6244 if (effect == mEffects[i]) { 6245 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6246 delete[] effect->inBuffer(); 6247 } else { 6248 if (i == size - 1 && i != 0) { 6249 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6250 mEffects[i - 1]->configure(); 6251 } 6252 } 6253 mEffects.removeAt(i); 6254 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6255 break; 6256 } 6257 } 6258 6259 return mEffects.size(); 6260} 6261 6262// setDevice_l() must be called with PlaybackThread::mLock held 6263void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6264{ 6265 size_t size = mEffects.size(); 6266 for (size_t i = 0; i < size; i++) { 6267 mEffects[i]->setDevice(device); 6268 } 6269} 6270 6271// setMode_l() must be called with PlaybackThread::mLock held 6272void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6273{ 6274 size_t size = mEffects.size(); 6275 for (size_t i = 0; i < size; i++) { 6276 mEffects[i]->setMode(mode); 6277 } 6278} 6279 6280// setVolume_l() must be called with PlaybackThread::mLock held 6281bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6282{ 6283 uint32_t newLeft = *left; 6284 uint32_t newRight = *right; 6285 bool hasControl = false; 6286 int ctrlIdx = -1; 6287 size_t size = mEffects.size(); 6288 6289 // first update volume controller 6290 for (size_t i = size; i > 0; i--) { 6291 if ((mEffects[i - 1]->state() >= EffectModule::ACTIVE) && 6292 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6293 ctrlIdx = i - 1; 6294 hasControl = true; 6295 break; 6296 } 6297 } 6298 6299 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6300 if (hasControl) { 6301 *left = mNewLeftVolume; 6302 *right = mNewRightVolume; 6303 } 6304 return hasControl; 6305 } 6306 6307 if (mVolumeCtrlIdx != -1) { 6308 hasControl = true; 6309 } 6310 mVolumeCtrlIdx = ctrlIdx; 6311 mLeftVolume = newLeft; 6312 mRightVolume = newRight; 6313 6314 // second get volume update from volume controller 6315 if (ctrlIdx >= 0) { 6316 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6317 mNewLeftVolume = newLeft; 6318 mNewRightVolume = newRight; 6319 } 6320 // then indicate volume to all other effects in chain. 6321 // Pass altered volume to effects before volume controller 6322 // and requested volume to effects after controller 6323 uint32_t lVol = newLeft; 6324 uint32_t rVol = newRight; 6325 6326 for (size_t i = 0; i < size; i++) { 6327 if ((int)i == ctrlIdx) continue; 6328 // this also works for ctrlIdx == -1 when there is no volume controller 6329 if ((int)i > ctrlIdx) { 6330 lVol = *left; 6331 rVol = *right; 6332 } 6333 mEffects[i]->setVolume(&lVol, &rVol, false); 6334 } 6335 *left = newLeft; 6336 *right = newRight; 6337 6338 return hasControl; 6339} 6340 6341status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6342{ 6343 const size_t SIZE = 256; 6344 char buffer[SIZE]; 6345 String8 result; 6346 6347 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6348 result.append(buffer); 6349 6350 bool locked = tryLock(mLock); 6351 // failed to lock - AudioFlinger is probably deadlocked 6352 if (!locked) { 6353 result.append("\tCould not lock mutex:\n"); 6354 } 6355 6356 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6357 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6358 mEffects.size(), 6359 (uint32_t)mInBuffer, 6360 (uint32_t)mOutBuffer, 6361 mActiveTrackCnt); 6362 result.append(buffer); 6363 write(fd, result.string(), result.size()); 6364 6365 for (size_t i = 0; i < mEffects.size(); ++i) { 6366 sp<EffectModule> effect = mEffects[i]; 6367 if (effect != 0) { 6368 effect->dump(fd, args); 6369 } 6370 } 6371 6372 if (locked) { 6373 mLock.unlock(); 6374 } 6375 6376 return NO_ERROR; 6377} 6378 6379#undef LOG_TAG 6380#define LOG_TAG "AudioFlinger" 6381 6382// ---------------------------------------------------------------------------- 6383 6384status_t AudioFlinger::onTransact( 6385 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6386{ 6387 return BnAudioFlinger::onTransact(code, data, reply, flags); 6388} 6389 6390}; // namespace android 6391